- Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Allow codecs to override register display
by Mark Brown 18 Jul '08

18 Jul '08

Some codecs have unusual features in their register maps such as very large registers representing arrays of coefficients. Support these codecs in the register cache sysfs file by allowing them to provide a function register_display() overriding the default output for register contents. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 14 ++++++++++---- 2 files changed, 11 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 1890d87..8b1a3e7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -410,6 +410,7 @@ struct snd_soc_codec { void *control_data; /* codec control (i2c/3wire) data */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + int (*display_register)(struct snd_soc_codec *, char *, unsigned int); hw_write_t hw_write; hw_read_t hw_read; void *reg_cache; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83f1190..49532d9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -970,10 +970,16 @@ static ssize_t codec_reg_show(struct device *dev, step = codec->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); - for (i = 0; i < codec->reg_cache_size; i += step) - count += sprintf(buf + count, "%2x: %4x\n", i, - codec->read(codec, i)); - + for (i = 0; i < codec->reg_cache_size; i += step) { + count += sprintf(buf + count, "%2x: ", i); + if (codec->display_register) + count += codec->display_register(codec, + buf + count, i); + else + count += sprintf(buf + count, "%4x", + codec->read(codec, i)); + count += sprintf(buf + count, "\n"); + } return count; } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); -- 1.5.6.3

3 3

[alsa-devel] [PATCH] send both an uncork and a trigger in _start()
by Lennart Poettering 18 Jul '08

18 Jul '08

Heya! Here's a patch for alsa-plugins: When playing very short streams, the pulse plugin needs call pa_stream_trigger() in snd_pcm_start() to make sure the stream is actually started, in addition to uncorking the stream. Lennart --- pulse/pcm_pulse.c | 21 +++++++++++++++------ 1 files changed, 15 insertions(+), 6 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index efff509..7edd157 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -67,8 +67,8 @@ static void update_ptr(snd_pcm_pulse_t *pcm) static int pulse_start(snd_pcm_ioplug_t *io) { snd_pcm_pulse_t *pcm = io->private_data; - pa_operation *o; - int err = 0; + pa_operation *o, *u; + int err = 0, err_o = 0, err_u = 0; assert(pcm); assert(pcm->p); @@ -88,11 +88,20 @@ static int pulse_start(snd_pcm_ioplug_t *io) goto finish; } - err = pulse_wait_operation(pcm->p, o); + u = pa_stream_trigger(pcm->stream, pulse_stream_success_cb, pcm->p); + if (!u) { + pa_operation_unref(o); + err = -EIO; + goto finish; + } + + err_o = pulse_wait_operation(pcm->p, o); + err_u = pulse_wait_operation(pcm->p, u); pa_operation_unref(o); + pa_operation_unref(u); - if (err < 0) { + if (err_o < 0 || err_u < 0) { err = -EIO; goto finish; } else @@ -543,7 +552,7 @@ static int pulse_hw_params(snd_pcm_ioplug_t *io, snd_pcm_hw_params_t *params) break; } } - + pa_threaded_mainloop_lock(pcm->p->mainloop); pcm->frame_size = (snd_pcm_format_physical_width(io->format) * io->channels) / 8; @@ -761,7 +770,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(pulse) pcm->io.callback = stream == SND_PCM_STREAM_PLAYBACK ? &pulse_playback_callback : &pulse_capture_callback; pcm->io.private_data = pcm; - + err = snd_pcm_ioplug_create(&pcm->io, name, stream, mode); if (err < 0) goto error; -- 1.5.6 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 3/3] fix return value of snd_pcm_rewind()/snd_pcm_forward() to return how much actually has been rewound, instead of what actually could have rewound
by Lennart Poettering 18 Jul '08

18 Jul '08

Make snd_pcm_plugin_rewind()/_forward() actually return how much has been rewound/forwarded instead of how much could have been rewounded/forwarded. This makes the code actually do what the documentation of snd_pcm_rewind() suggests. Signed-off-by: Lennart Poettering <lennart(a)poettering.net> --- src/pcm/pcm_plugin.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index b377cb2..c4f5b4a 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -222,7 +222,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t frames = plugin->client_frames(pcm, sframes); snd_pcm_mmap_appl_backward(pcm, (snd_pcm_uframes_t) frames); snd_atomic_write_end(&plugin->watom); - return n; + return (snd_pcm_sframes_t) frames; } static snd_pcm_sframes_t snd_pcm_plugin_forwardable(snd_pcm_t *pcm) @@ -255,7 +255,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_forward(snd_pcm_t *pcm, snd_pcm_uframes_ frames = plugin->client_frames(pcm, sframes); snd_pcm_mmap_appl_forward(pcm, (snd_pcm_uframes_t) frames); snd_atomic_write_end(&plugin->watom); - return n; + return (snd_pcm_sframes_t) frames; } static snd_pcm_sframes_t snd_pcm_plugin_write_areas(snd_pcm_t *pcm, -- 1.5.6 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 2/3] fix type of internally used sframes variable, to avoid unnecessary casts
by Lennart Poettering 18 Jul '08

18 Jul '08

This minor patch fixes the type of the sframes variable in snd_pcm_plugin_forward(). With this fix we need to cast less and the code is less confusing. Signed-off-by: Lennart Poettering <lennart(a)poettering.net> --- src/pcm/pcm_plugin.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index c199d8d..b377cb2 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -234,7 +234,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_forward(snd_pcm_t *pcm, snd_pcm_uframes_ { snd_pcm_plugin_t *plugin = pcm->private_data; snd_pcm_sframes_t n = snd_pcm_mmap_avail(pcm); - snd_pcm_uframes_t sframes; + snd_pcm_sframes_t sframes; if ((snd_pcm_uframes_t)n < frames) frames = n; @@ -246,8 +246,8 @@ static snd_pcm_sframes_t snd_pcm_plugin_forward(snd_pcm_t *pcm, snd_pcm_uframes_ else sframes = frames; snd_atomic_write_begin(&plugin->watom); - sframes = INTERNAL(snd_pcm_forward)(plugin->gen.slave, (snd_pcm_uframes_t) sframes); - if ((snd_pcm_sframes_t) sframes < 0) { + sframes = INTERNAL(snd_pcm_forward)(plugin->gen.slave, sframes); + if (sframes < 0) { snd_atomic_write_end(&plugin->watom); return sframes; } -- 1.5.6 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 1/3] clamp snd_pcm_rewind()/snd_pcm_forward() into the right direction
by Lennart Poettering 18 Jul '08

18 Jul '08

The clamping of the input parameter in snd_pcm_rewind()/_forward() is in the wrong direction. Signed-off-by: Lennart Poettering <lennart(a)poettering.net> --- src/pcm/pcm_plugin.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index c73a02b..c199d8d 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -203,7 +203,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t snd_pcm_sframes_t n = snd_pcm_mmap_hw_avail(pcm); snd_pcm_sframes_t sframes; - if ((snd_pcm_uframes_t)n > frames) + if ((snd_pcm_uframes_t)n < frames) frames = n; if (frames == 0) return 0; @@ -236,7 +236,7 @@ static snd_pcm_sframes_t snd_pcm_plugin_forward(snd_pcm_t *pcm, snd_pcm_uframes_ snd_pcm_sframes_t n = snd_pcm_mmap_avail(pcm); snd_pcm_uframes_t sframes; - if ((snd_pcm_uframes_t)n > frames) + if ((snd_pcm_uframes_t)n < frames) frames = n; if (frames == 0) return 0; -- 1.5.6 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

[alsa-devel] [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers
by Grant Likely 18 Jul '08

18 Jul '08

From: Grant Likely <grant.likely(a)secretlab.ca> Simple utility layer for creating ASoC machine instances based on data in the OpenFirmware device tree. OF aware platform drivers and codec drivers register themselves with this framework and the framework automatically instantiates a machine driver. This is most likely temporary glue code to work around limitations in the ASoC v1 framework. I expect ASoC v2 won't need this. --- sound/soc/Kconfig | 6 ++ sound/soc/Makefile | 1 sound/soc/soc-of.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 178 insertions(+), 0 deletions(-) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac..c5736e5 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,6 +23,12 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core. +config SND_SOC_OF + tristate "OF helpers for SoC audio support" + depends on SND_SOC + ---help--- + Add support for OpenFirmware device tree descriptions of sound device + # All the supported Soc's source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..191c2e5 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC_OF) += soc-of.o diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c new file mode 100644 index 0000000..9694979 --- /dev/null +++ b/sound/soc/soc-of.c @@ -0,0 +1,171 @@ +/* + * OF helpers for ALSA SoC Layer + * + * Copyright (C) 2008, Secret Lab Technologies Ltd. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-of.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Grant Likely <grant.likely(a)secretlab.ca>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings"); + +DEFINE_MUTEX(of_snd_soc_mutex); +LIST_HEAD(of_snd_soc_device_list); +static int of_snd_soc_next_index; + +struct of_snd_soc_device { + int id; + struct list_head list; + struct snd_soc_device device; + struct snd_soc_machine machine; + struct snd_soc_dai_link dai_link; + struct platform_device *pdev; + struct device_node *platform_node; + struct device_node *codec_node; +}; + +static struct snd_soc_ops of_snd_soc_ops = { +}; + +static struct of_snd_soc_device * +of_snd_soc_get_device(struct device_node *codec_node) +{ + struct of_snd_soc_device *of_soc; + + list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { + if (of_soc->codec_node == codec_node) + return of_soc; + } + + of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL); + if (!of_soc) + return NULL; + + /* Initialize the structure and add it to the global list */ + of_soc->codec_node = codec_node; + of_soc->id = of_snd_soc_next_index++; + of_soc->machine.dai_link = &of_soc->dai_link; + of_soc->machine.num_links = 1; + of_soc->device.machine = &of_soc->machine; + of_soc->dai_link.ops = &of_snd_soc_ops; + list_add(&of_soc->list, &of_snd_soc_device_list); + + return of_soc; +} + +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc) +{ + struct platform_device *pdev; + int rc; + + /* Only register the device if both the codec and platform have + * been registered */ + if ((!of_soc->device.codec_data) || (!of_soc->platform_node)) + return; + + pr_info("platform<-->codec match achieved; registering machine\n"); + + pdev = platform_device_alloc("soc-audio", of_soc->id); + if (!pdev) { + pr_err("of_soc: platform_device_alloc() failed\n"); + return; + } + + pdev->dev.platform_data = of_soc; + platform_set_drvdata(pdev, &of_soc->device); + of_soc->device.dev = &pdev->dev; + + /* The ASoC device is complete; register it */ + rc = platform_device_add(pdev); + if (rc) { + pr_err("of_soc: platform_device_add() failed\n"); + return; + } + +} + +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, + void *codec_data, struct snd_soc_codec_dai *dai, + struct device_node *node) +{ + struct of_snd_soc_device *of_soc; + int rc = 0; + + pr_info("registering ASoC codec driver: %s\n", node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + /* Store the codec data */ + of_soc->device.codec_data = codec_data; + of_soc->device.codec_dev = codec_dev; + of_soc->dai_link.name = node->name; + of_soc->dai_link.stream_name = node->name; + of_soc->dai_link.codec_dai = dai; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform, + struct device_node *node, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct of_snd_soc_device *of_soc; + struct device_node *codec_node; + const phandle *handle; + int len, rc = 0; + + pr_info("registering ASoC platform driver: %s\n", node->full_name); + + handle = of_get_property(node, "codec-handle", &len); + if (!handle || len < sizeof(handle)) + return -ENODEV; + codec_node = of_find_node_by_phandle(*handle); + if (!codec_node) + return -ENODEV; + pr_info("looking for codec: %s\n", codec_node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(codec_node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + of_soc->platform_node = node; + of_soc->dai_link.cpu_dai = cpu_dai; + of_soc->device.platform = platform; + of_soc->machine.name = of_soc->dai_link.cpu_dai->name; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);

8 34

Re: [alsa-devel] [PATCH] PalmTX aSoC sound support
by Marek Vasut 18 Jul '08

18 Jul '08

Dne Friday 04 of July 2008 22:26:15 Mark Brown napsal(a): > On Fri, Jul 04, 2008 at 09:49:57PM +0200, Marek Vasut wrote: > > Like Liam said, audio patches should be CCed to alsa-devel as well - a > > couple of comments from a quick scan through: > > + /* add palmtx specific widgets */ > > + for (i = 0; i < ARRAY_SIZE(palmtx_dapm_widgets); i++) > > + snd_soc_dapm_new_control(codec, &palmtx_dapm_widgets[i]); > > Ideally this should use the new snd_soc_dapm_new_controls() function... > > > + /* set up palmtx specific audio path audio_map */ > > + for (i = 0; audio_map[i][0] != NULL; i++) > > + snd_soc_dapm_connect_input(codec, audio_map[i][0], > > + audio_map[i][1], audio_map[i][2]); > > and this should use snd_soc_dapm_add_routes (which would also mean you > could drop the null terminator from audio_map()). Both are queued for > merge in the window - see ALSA git, it'll also appear in linux-next. > > > + ret = platform_device_add(palmtx_snd_device); > > + > > + if (!ret) > > + return 0; > > + > > + platform_device_put(palmtx_snd_device); > > It took me a double take to follow this - it'd be more idiomatic to say: > > if (ret != 0) > goto put_device; > > return 0; > put_device: ok, I applied some alsa patches to my branch of "devel"

4 11

[alsa-devel] [PATCH v2 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers
by Grant Likely 18 Jul '08

18 Jul '08

From: Grant Likely <grant.likely(a)secretlab.ca> Simple utility layer for creating ASoC machine instances based on data in the OpenFirmware device tree. OF aware platform drivers and codec drivers register themselves with this framework and the framework automatically instantiates a machine driver. This is most likely temporary glue code to work around limitations in the ASoC v1 framework. I expect ASoC v2 won't need this. --- include/sound/soc-of.h | 21 ++++++ sound/soc/Kconfig | 3 + sound/soc/Makefile | 1 sound/soc/soc-of.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 196 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-of.h b/include/sound/soc-of.h new file mode 100644 index 0000000..a963032 --- /dev/null +++ b/include/sound/soc-of.h @@ -0,0 +1,21 @@ +/* + * OF helpers for ALSA SoC + * + * Copyright (C) 2008, Secret Lab Technologies Ltd. + */ + +#ifndef _INCLUDE_SOC_OF_H_ +#define _INCLUDE_SOC_OF_H_ + +#include <linux/of.h> +#include <sound/soc.h> + +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, + void *codec_data, struct snd_soc_codec_dai *dai, + struct device_node *node); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform, + struct device_node *node, + struct snd_soc_cpu_dai *cpu_dai); + +#endif /* _INCLUDE_SOC_OF_H_ */ diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 18f28ac..99118ed 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -23,6 +23,9 @@ config SND_SOC This ASoC audio support can also be built as a module. If so, the module will be called snd-soc-core. +config SND_SOC_OF + tristate + # All the supported Soc's source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 782db21..191c2e5 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -2,3 +2,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ omap/ +obj-$(CONFIG_SND_SOC_OF) += soc-of.o diff --git a/sound/soc/soc-of.c b/sound/soc/soc-of.c new file mode 100644 index 0000000..0c855df --- /dev/null +++ b/sound/soc/soc-of.c @@ -0,0 +1,171 @@ +/* + * OF helpers for ALSA SoC Layer + * + * Copyright (C) 2008, Secret Lab Technologies Ltd. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/bitops.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-of.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Grant Likely <grant.likely(a)secretlab.ca>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ALSA SoC OpenFirmware bindings"); + +static DEFINE_MUTEX(of_snd_soc_mutex); +static LIST_HEAD(of_snd_soc_device_list); +static int of_snd_soc_next_index; + +struct of_snd_soc_device { + int id; + struct list_head list; + struct snd_soc_device device; + struct snd_soc_machine machine; + struct snd_soc_dai_link dai_link; + struct platform_device *pdev; + struct device_node *platform_node; + struct device_node *codec_node; +}; + +static struct snd_soc_ops of_snd_soc_ops = { +}; + +static struct of_snd_soc_device * +of_snd_soc_get_device(struct device_node *codec_node) +{ + struct of_snd_soc_device *of_soc; + + list_for_each_entry(of_soc, &of_snd_soc_device_list, list) { + if (of_soc->codec_node == codec_node) + return of_soc; + } + + of_soc = kzalloc(sizeof(struct of_snd_soc_device), GFP_KERNEL); + if (!of_soc) + return NULL; + + /* Initialize the structure and add it to the global list */ + of_soc->codec_node = codec_node; + of_soc->id = of_snd_soc_next_index++; + of_soc->machine.dai_link = &of_soc->dai_link; + of_soc->machine.num_links = 1; + of_soc->device.machine = &of_soc->machine; + of_soc->dai_link.ops = &of_snd_soc_ops; + list_add(&of_soc->list, &of_snd_soc_device_list); + + return of_soc; +} + +static void of_snd_soc_register_device(struct of_snd_soc_device *of_soc) +{ + struct platform_device *pdev; + int rc; + + /* Only register the device if both the codec and platform have + * been registered */ + if ((!of_soc->device.codec_data) || (!of_soc->platform_node)) + return; + + pr_info("platform<-->codec match achieved; registering machine\n"); + + pdev = platform_device_alloc("soc-audio", of_soc->id); + if (!pdev) { + pr_err("of_soc: platform_device_alloc() failed\n"); + return; + } + + pdev->dev.platform_data = of_soc; + platform_set_drvdata(pdev, &of_soc->device); + of_soc->device.dev = &pdev->dev; + + /* The ASoC device is complete; register it */ + rc = platform_device_add(pdev); + if (rc) { + pr_err("of_soc: platform_device_add() failed\n"); + return; + } + +} + +int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, + void *codec_data, struct snd_soc_codec_dai *dai, + struct device_node *node) +{ + struct of_snd_soc_device *of_soc; + int rc = 0; + + pr_info("registering ASoC codec driver: %s\n", node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + /* Store the codec data */ + of_soc->device.codec_data = codec_data; + of_soc->device.codec_dev = codec_dev; + of_soc->dai_link.name = node->name; + of_soc->dai_link.stream_name = node->name; + of_soc->dai_link.codec_dai = dai; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_codec); + +int of_snd_soc_register_platform(struct snd_soc_platform *platform, + struct device_node *node, + struct snd_soc_cpu_dai *cpu_dai) +{ + struct of_snd_soc_device *of_soc; + struct device_node *codec_node; + const phandle *handle; + int len, rc = 0; + + pr_info("registering ASoC platform driver: %s\n", node->full_name); + + handle = of_get_property(node, "codec-handle", &len); + if (!handle || len < sizeof(handle)) + return -ENODEV; + codec_node = of_find_node_by_phandle(*handle); + if (!codec_node) + return -ENODEV; + pr_info("looking for codec: %s\n", codec_node->full_name); + + mutex_lock(&of_snd_soc_mutex); + of_soc = of_snd_soc_get_device(codec_node); + if (!of_soc) { + rc = -ENOMEM; + goto out; + } + + of_soc->platform_node = node; + of_soc->dai_link.cpu_dai = cpu_dai; + of_soc->device.platform = platform; + of_soc->machine.name = of_soc->dai_link.cpu_dai->name; + + /* Now try to register the SoC device */ + of_snd_soc_register_device(of_soc); + + out: + mutex_unlock(&of_snd_soc_mutex); + return rc; +} +EXPORT_SYMBOL_GPL(of_snd_soc_register_platform);

7 45

[alsa-devel] [PATCH2/2] hda: digital pc-beep support hd-audio codecs.
by Matthew Ranostay 18 Jul '08

18 Jul '08

Added the respective file entry to call ../alsa-kernel/pci/hda_beep.c --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/hda_beep.c b/pci/hda/hda_beep.c new file mode 100644 index 0000000..a8e16a5 --- /dev/null +++ b/pci/hda/hda_beep.c @@ -0,0 +1,3 @@ +#define __NO_VERSION__ +#include "adriver.h" +#include "../../alsa-kernel/pci/hda/hda_beep.c"

1 0

[alsa-devel] [PATCH1/2] hda: digital pc-beep support hd-audio codecs.
by Matthew Ranostay 18 Jul '08

18 Jul '08

Added digital pc-beep support using linear tone generation for hd-codecs along with initial support for several IDT codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/Kconfig b/pci/Kconfig index 8fe5dac..c14cc6b 100644 --- a/pci/Kconfig +++ b/pci/Kconfig @@ -517,6 +517,14 @@ config SND_HDA_HWDEP This interface can be used for out-of-band communication with codecs for debugging purposes. +config SND_HDA_INPUT_BEEP + bool "Support digital beep via input layer" + depends on SND_HDA_INTEL + depends on INPUT=y || INPUT=SND_HDA_INTEL + help + Say Y here to build a digital beep interface for HD-audio + driver. This interface is used to generate digital beeps. + config SND_HDA_CODEC_REALTEK bool "Build Realtek HD-audio codec support" depends on SND_HDA_INTEL diff --git a/pci/hda/Makefile b/pci/hda/Makefile index ab0c726..6db92fd 100644 --- a/pci/hda/Makefile +++ b/pci/hda/Makefile @@ -5,6 +5,7 @@ snd-hda-intel-y := hda_intel.o snd-hda-intel-y += hda_codec.o snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o +snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o diff --git a/pci/hda/hda_beep.c b/pci/hda/hda_beep.c new file mode 100644 index 0000000..5a764c4 --- /dev/null +++ b/pci/hda/hda_beep.c @@ -0,0 +1,134 @@ +/* + * Digital Beep Input Interface for HD-audio codec + * + * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> + * Copyright (c) 2008 Embedded Alley Solutions Inc + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/input.h> +#include <linux/pci.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include "hda_beep.h" + +enum { + DIGBEEP_HZ_STEP = 46875, /* 46.875 Hz */ + DIGBEEP_HZ_MIN = 93750, /* 93.750 Hz */ + DIGBEEP_HZ_MAX = 12000000, /* 12 KHz */ +}; + +static void snd_hda_generate_beep(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, beep_work); + struct hda_codec *codec = beep->codec; + + /* generate tone */ + snd_hda_codec_write_cache(codec, beep->nid, 0, + AC_VERB_SET_BEEP_CONTROL, beep->tone); +} + +static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + struct hda_beep *beep = input_get_drvdata(dev); + + switch (code) { + case SND_BELL: + if (hz) + hz = 1000; + case SND_TONE: + hz *= 1000; /* fixed point */ + hz = hz - DIGBEEP_HZ_MIN; + if (hz < 0) + hz = 0; /* turn off PC beep*/ + else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) + hz = 0xff; + else { + hz /= DIGBEEP_HZ_STEP; + hz++; + } + break; + default: + return -1; + } + beep->tone = hz; + + /* schedule beep event */ + schedule_work(&beep->beep_work); + return 0; +} + +int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +{ + struct input_dev *input_dev; + struct hda_beep *beep; + int err; + + beep = kzalloc(sizeof(*beep), GFP_KERNEL); + if (beep == NULL) + return -ENOMEM; + snprintf(beep->phys, sizeof(beep->phys), + "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); + input_dev = input_allocate_device(); + + /* setup digital beep device */ + input_dev->name = "HDA Digital PCBeep"; + input_dev->phys = beep->phys; + input_dev->id.bustype = BUS_PCI; + + input_dev->id.vendor = codec->vendor_id >> 16; + input_dev->id.product = codec->vendor_id & 0xffff; + input_dev->id.version = 0x01; + + input_dev->evbit[0] = BIT_MASK(EV_SND); + input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + input_dev->event = snd_hda_beep_event; + input_dev->dev.parent = &codec->bus->pci->dev; + input_set_drvdata(input_dev, beep); + + err = input_register_device(input_dev); + if (err < 0) { + kfree(input_dev); + kfree(beep); + return err; + } + + /* enable linear scale */ + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_DIGI_CONVERT_2, 0x01); + + beep->nid = nid; + beep->dev = input_dev; + beep->codec = codec; + codec->beep = beep; + + INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); + return 0; +} + +void snd_hda_detach_beep_device(struct hda_codec *codec) +{ + struct hda_beep *beep = codec->beep; + if (beep) { + cancel_work_sync(&beep->beep_work); + flush_scheduled_work(); + + input_unregister_device(beep->dev); + kfree(beep); + } +} diff --git a/pci/hda/hda_beep.h b/pci/hda/hda_beep.h new file mode 100644 index 0000000..de4036e --- /dev/null +++ b/pci/hda/hda_beep.h @@ -0,0 +1,44 @@ +/* + * Digital Beep Input Interface for HD-audio codec + * + * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> + * Copyright (c) 2008 Embedded Alley Solutions Inc + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_HDA_BEEP_H +#define __SOUND_HDA_BEEP_H + +#include "hda_codec.h" + +/* beep information */ +struct hda_beep { + struct input_dev *dev; + struct hda_codec *codec; + char phys[32]; + int tone; + int nid; + struct work_struct beep_work; /* scheduled task for beep event */ +}; + +#ifdef CONFIG_SND_HDA_INPUT_BEEP +int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); +void snd_hda_detach_beep_device(struct hda_codec *codec); +#else +#define snd_hda_attach_beep_device(...) +#define snd_hda_detach_beep_device(...) +#endif +#endif diff --git a/pci/hda/hda_codec.h b/pci/hda/hda_codec.h index efc6828..3a63c44 100644 --- a/pci/hda/hda_codec.h +++ b/pci/hda/hda_codec.h @@ -449,6 +449,7 @@ enum { */ struct hda_bus; +struct hda_beep; struct hda_codec; struct hda_pcm; struct hda_pcm_stream; @@ -634,6 +635,9 @@ struct hda_codec { /* codec specific info */ void *spec; + /* beep device */ + struct hda_beep *beep; + /* widget capabilities cache */ unsigned int num_nodes; hda_nid_t start_nid; diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 08cb77f..d11fd1a 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -33,6 +33,7 @@ #include "hda_codec.h" #include "hda_local.h" #include "hda_patch.h" +#include "hda_beep.h" #define NUM_CONTROL_ALLOC 32 #define STAC_PWR_EVENT 0x20 @@ -161,6 +162,8 @@ struct sigmatel_spec { unsigned int num_dmuxes; hda_nid_t dig_in_nid; hda_nid_t mono_nid; + hda_nid_t anabeep_nid; + hda_nid_t digbeep_nid; /* pin widgets */ hda_nid_t *pin_nids; @@ -686,6 +689,8 @@ static struct hda_verb d965_core_init[] = { static struct hda_verb stac927x_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, {} }; @@ -825,8 +830,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + /* analog pc-beep replaced with digital beep support */ + /* HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), + */ HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), @@ -2603,6 +2611,34 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) "Mono Mux", spec->mono_nid); } +/* create PC beep volume controls */ +static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, + hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); + int err; + + /* check for mute support for the the amp */ + if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + "PC Beep Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + + /* check to see if there is volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, + "PC Beep Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + return 0; +} + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -2838,6 +2874,28 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; + /* setup analog beep controls */ + if (spec->anabeep_nid > 0) { + err = stac92xx_auto_create_beep_ctls(codec, + spec->anabeep_nid); + if (err < 0) + return err; + } + + /* setup digital beep controls and input device */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (spec->digbeep_nid > 0) { + hda_nid_t nid = spec->digbeep_nid; + + err = stac92xx_auto_create_beep_ctls(codec, nid); + if (err < 0) + return err; + err = snd_hda_attach_beep_device(codec, nid); + if (err < 0) + return err; + } +#endif + if (hp_speaker_swap == 1) { /* Restore the hp_outs and line_outs */ memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, @@ -3152,6 +3210,7 @@ static void stac92xx_free(struct hda_codec *codec) kfree(spec->bios_pin_configs); kfree(spec); + snd_hda_detach_beep_device(codec); } static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -3540,6 +3599,7 @@ again: spec->aloopback_mask = 0x01; spec->aloopback_shift = 8; + spec->digbeep_nid = 0x1c; spec->mux_nids = stac92hd73xx_mux_nids; spec->adc_nids = stac92hd73xx_adc_nids; spec->dmic_nids = stac92hd73xx_dmic_nids; @@ -3674,6 +3734,7 @@ again: spec->gpio_dir = 0x01; spec->gpio_data = 0x01; + spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; @@ -3845,6 +3906,7 @@ static int patch_stac927x(struct hda_codec *codec) stac92xx_set_config_regs(codec); } + spec->digbeep_nid = 0x23; spec->adc_nids = stac927x_adc_nids; spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); spec->mux_nids = stac927x_mux_nids; @@ -3965,6 +4027,7 @@ static int patch_stac9205(struct hda_codec *codec) stac92xx_set_config_regs(codec); } + spec->digbeep_nid = 0x23; spec->adc_nids = stac9205_adc_nids; spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids); spec->mux_nids = stac9205_mux_nids;

1 0

[PATCH] ALSA: opti9xx: no isapnp param for !CONFIG_PNP
by Rene Herman 18 Jul '08

18 Jul '08

"isapnp" needs CONFIG_PNP to be useful. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/isa/opti9xx/opti92x-ad1848.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index e5d9028..5dfead9 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -68,7 +68,9 @@ MODULE_SUPPORTED_DEVICE("{{OPTi,82C924 (AD1848)}," static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ //static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +#ifdef CONFIG_PNP static int isapnp = 1; /* Enable ISA PnP detection */ +#endif static long port = SNDRV_DEFAULT_PORT1; /* 0x530,0xe80,0xf40,0x604 */ static long mpu_port = SNDRV_DEFAULT_PORT1; /* 0x300,0x310,0x320,0x330 */ static long fm_port = SNDRV_DEFAULT_PORT1; /* 0x388 */ @@ -85,8 +87,10 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for opti9xx based soundcard."); //module_param(enable, bool, 0444); //MODULE_PARM_DESC(enable, "Enable opti9xx soundcard."); +#ifdef CONFIG_PNP module_param(isapnp, bool, 0444); MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif module_param(port, long, 0444); MODULE_PARM_DESC(port, "WSS port # for opti9xx driver."); module_param(mpu_port, long, 0444); -- 1.5.5 --------------070007060903020506000905--

1 0

[alsa-devel] [PATCH 9/xx] wss_lib: use wss pcm code instead of ad1848 one
by Krzysztof Helt 18 Jul '08

18 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss pcm code and kill the ad1848 pcm code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The next patch in the series of the wss_lib. It brings another code reduction and the only thing left in the ad1848_lib is detection code. There is a small code added to the snd_wss_pcm() with "temporary" comment. It is needed until the detection/initialization is merged code (the next patch I suppose). Also, the AD1848_LIB selects the WSS_LIB in the Kconfig. This is also temporary until the ad1848_lib is removed. It works on cards I tested (Opti930, Sound Galaxy with CS4231A codec and SC-6000 with the AD1848 codec). Both playback and recording was tested. diff -urp linux-alsa/include/sound/ad1848.h linux-mm/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-07-17 13:33:14.557293953 +0200 +++ linux-mm/include/sound/ad1848.h 2008-07-17 21:39:34.462143225 +0200 @@ -95,11 +95,6 @@ #define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ #define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ -/* IBM Thinkpad specific stuff */ -#define AD1848_THINKPAD_CTL_PORT1 0x15e8 -#define AD1848_THINKPAD_CTL_PORT2 0x15e9 -#define AD1848_THINKPAD_CS4248_ENABLE_BIT 0x02 - #include "wss.h" /* exported functions */ @@ -113,7 +108,4 @@ int snd_ad1848_create(struct snd_card *c unsigned short hardware, struct snd_wss **chip); -int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); -const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); - #endif /* __SOUND_AD1848_H */ diff -urp linux-alsa/include/sound/wss.h linux-mm/include/sound/wss.h --- linux-alsa/include/sound/wss.h 2008-07-17 13:33:14.561277099 +0200 +++ linux-mm/include/sound/wss.h 2008-07-17 13:19:52.643673336 +0200 @@ -71,6 +71,11 @@ #define WSS_HWSHARE_DMA1 (1<<1) #define WSS_HWSHARE_DMA2 (1<<2) +/* IBM Thinkpad specific stuff */ +#define AD1848_THINKPAD_CTL_PORT1 0x15e8 +#define AD1848_THINKPAD_CTL_PORT2 0x15e9 +#define AD1848_THINKPAD_CS4248_ENABLE_BIT 0x02 + struct snd_wss { unsigned long port; /* base i/o port */ struct resource *res_port; @@ -152,6 +157,8 @@ int snd_wss_pcm(struct snd_wss *chip, in int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer); int snd_wss_mixer(struct snd_wss *chip); +const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction); + int snd_cs4236_create(struct snd_card *card, unsigned long port, unsigned long cport, diff -urp linux-alsa/sound/isa/Kconfig linux-mm/sound/isa/Kconfig --- linux-alsa/sound/isa/Kconfig 2008-06-24 19:45:11.000000000 +0200 +++ linux-mm/sound/isa/Kconfig 2008-07-17 22:10:01.946141549 +0200 @@ -1,12 +1,13 @@ # ALSA ISA drivers -config SND_AD1848_LIB +config SND_WSS_LIB tristate select SND_PCM -config SND_WSS_LIB +config SND_AD1848_LIB tristate select SND_PCM + select SND_WSS_LIB config SND_SB_COMMON tristate diff -urp linux-alsa/sound/isa/ad1848/ad1848.c linux-mm/sound/isa/ad1848/ad1848.c --- linux-alsa/sound/isa/ad1848/ad1848.c 2008-07-17 13:33:14.589281430 +0200 +++ linux-mm/sound/isa/ad1848/ad1848.c 2008-07-17 13:07:12.435184998 +0200 @@ -102,7 +102,7 @@ static int __devinit snd_ad1848_probe(st card->private_data = chip; - error = snd_ad1848_pcm(chip, 0, &pcm); + error = snd_wss_pcm(chip, 0, &pcm); if (error < 0) goto out; diff -urp linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-mm/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-07-17 13:33:14.597275122 +0200 +++ linux-mm/sound/isa/ad1848/ad1848_lib.c 2008-07-17 21:48:14.042137359 +0200 @@ -46,34 +46,6 @@ MODULE_LICENSE("GPL"); * Some variables */ -static unsigned char freq_bits[14] = { - /* 5510 */ 0x00 | AD1848_XTAL2, - /* 6620 */ 0x0E | AD1848_XTAL2, - /* 8000 */ 0x00 | AD1848_XTAL1, - /* 9600 */ 0x0E | AD1848_XTAL1, - /* 11025 */ 0x02 | AD1848_XTAL2, - /* 16000 */ 0x02 | AD1848_XTAL1, - /* 18900 */ 0x04 | AD1848_XTAL2, - /* 22050 */ 0x06 | AD1848_XTAL2, - /* 27042 */ 0x04 | AD1848_XTAL1, - /* 32000 */ 0x06 | AD1848_XTAL1, - /* 33075 */ 0x0C | AD1848_XTAL2, - /* 37800 */ 0x08 | AD1848_XTAL2, - /* 44100 */ 0x0A | AD1848_XTAL2, - /* 48000 */ 0x0C | AD1848_XTAL1 -}; - -static unsigned int rates[14] = { - 5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050, - 27042, 32000, 33075, 37800, 44100, 48000 -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - static unsigned char snd_ad1848_original_image[16] = { 0x00, /* 00 - lic */ @@ -128,15 +100,6 @@ void snd_ad1848_out(struct snd_wss *chip EXPORT_SYMBOL(snd_ad1848_out); -static void snd_ad1848_dout(struct snd_wss *chip, - unsigned char reg, unsigned char value) -{ - snd_ad1848_wait(chip); - outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); - outb(value, chip->port + CS4231P(REG)); - mb(); -} - static unsigned char snd_ad1848_in(struct snd_wss *chip, unsigned char reg) { snd_ad1848_wait(chip); @@ -279,315 +242,6 @@ static void snd_ad1848_mce_down(struct s inb(chip->port + CS4231P(REGSEL))); } -static unsigned int snd_ad1848_get_count(unsigned char format, - unsigned int size) -{ - switch (format & 0xe0) { - case AD1848_LINEAR_16: - size >>= 1; - break; - } - if (format & AD1848_STEREO) - size >>= 1; - return size; -} - -static int snd_ad1848_trigger(struct snd_wss *chip, unsigned char what, - int channel, int cmd) -{ - int result = 0; - -#if 0 - printk("codec trigger!!! - what = %i, enable = %i, status = 0x%x\n", what, enable, inb(AD1848P(card, STATUS))); -#endif - spin_lock(&chip->reg_lock); - if (cmd == SNDRV_PCM_TRIGGER_START) { - if (chip->image[AD1848_IFACE_CTRL] & what) { - spin_unlock(&chip->reg_lock); - return 0; - } - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] |= what); - } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { - if (!(chip->image[AD1848_IFACE_CTRL] & what)) { - spin_unlock(&chip->reg_lock); - return 0; - } - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] &= ~what); - } else { - result = -EINVAL; - } - spin_unlock(&chip->reg_lock); - return result; -} - -/* - * CODEC I/O - */ - -static unsigned char snd_ad1848_get_rate(unsigned int rate) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(rates); i++) - if (rate == rates[i]) - return freq_bits[i]; - snd_BUG(); - return freq_bits[ARRAY_SIZE(rates) - 1]; -} - -static int snd_ad1848_ioctl(struct snd_pcm_substream *substream, - unsigned int cmd, void *arg) -{ - return snd_pcm_lib_ioctl(substream, cmd, arg); -} - -static unsigned char snd_ad1848_get_format(int format, int channels) -{ - unsigned char rformat; - - rformat = AD1848_LINEAR_8; - switch (format) { - case SNDRV_PCM_FORMAT_A_LAW: rformat = AD1848_ALAW_8; break; - case SNDRV_PCM_FORMAT_MU_LAW: rformat = AD1848_ULAW_8; break; - case SNDRV_PCM_FORMAT_S16_LE: rformat = AD1848_LINEAR_16; break; - } - if (channels > 1) - rformat |= AD1848_STEREO; -#if 0 - snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); -#endif - return rformat; -} - -static void snd_ad1848_calibrate_mute(struct snd_wss *chip, int mute) -{ - unsigned long flags; - - mute = mute ? 1 : 0; - spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->calibrate_mute == mute) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } - if (!mute) { - snd_ad1848_dout(chip, AD1848_LEFT_INPUT, chip->image[AD1848_LEFT_INPUT]); - snd_ad1848_dout(chip, AD1848_RIGHT_INPUT, chip->image[AD1848_RIGHT_INPUT]); - } - snd_ad1848_dout(chip, AD1848_AUX1_LEFT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX1_LEFT_INPUT]); - snd_ad1848_dout(chip, AD1848_AUX1_RIGHT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX1_RIGHT_INPUT]); - snd_ad1848_dout(chip, AD1848_AUX2_LEFT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX2_LEFT_INPUT]); - snd_ad1848_dout(chip, AD1848_AUX2_RIGHT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX2_RIGHT_INPUT]); - snd_ad1848_dout(chip, AD1848_LEFT_OUTPUT, mute ? 0x80 : chip->image[AD1848_LEFT_OUTPUT]); - snd_ad1848_dout(chip, AD1848_RIGHT_OUTPUT, mute ? 0x80 : chip->image[AD1848_RIGHT_OUTPUT]); - chip->calibrate_mute = mute; - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -static void snd_ad1848_set_data_format(struct snd_wss *chip, - struct snd_pcm_hw_params *hw_params) -{ - if (hw_params == NULL) { - chip->image[AD1848_DATA_FORMAT] = 0x20; - } else { - chip->image[AD1848_DATA_FORMAT] = - snd_ad1848_get_format(params_format(hw_params), params_channels(hw_params)) | - snd_ad1848_get_rate(params_rate(hw_params)); - } - // snd_printk(">>> pmode = 0x%x, dfr = 0x%x\n", pstr->mode, chip->image[AD1848_DATA_FORMAT]); -} - -static int snd_ad1848_open(struct snd_wss *chip, unsigned int mode) -{ - unsigned long flags; - - if (chip->mode & WSS_MODE_OPEN) - return -EAGAIN; - - snd_ad1848_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("open: (1)\n"); -#endif - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | - AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO | - AD1848_CALIB_MODE); - chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("open: (2)\n"); -#endif - - snd_ad1848_set_data_format(chip, NULL); - - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("open: (3)\n"); -#endif - - /* ok. now enable and ack CODEC IRQ */ - spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - chip->image[AD1848_PIN_CTRL] |= AD1848_IRQ_ENABLE; - snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->mode = mode; - - return 0; -} - -static void snd_ad1848_close(struct snd_wss *chip) -{ - unsigned long flags; - - if (!chip->mode) - return; - /* disable IRQ */ - spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - chip->image[AD1848_PIN_CTRL] &= ~AD1848_IRQ_ENABLE; - snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - /* now disable capture & playback */ - - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | - AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); - snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - - /* clear IRQ again */ - spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->mode = 0; -} - -/* - * ok.. exported functions.. - */ - -static int snd_ad1848_playback_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - return snd_ad1848_trigger(chip, AD1848_PLAYBACK_ENABLE, SNDRV_PCM_STREAM_PLAYBACK, cmd); -} - -static int snd_ad1848_capture_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - return snd_ad1848_trigger(chip, AD1848_CAPTURE_ENABLE, SNDRV_PCM_STREAM_CAPTURE, cmd); -} - -static int snd_ad1848_playback_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - unsigned long flags; - int err; - - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) - return err; - snd_ad1848_calibrate_mute(chip, 1); - snd_ad1848_set_data_format(chip, hw_params); - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - snd_ad1848_calibrate_mute(chip, 0); - return 0; -} - -static int snd_ad1848_playback_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int snd_ad1848_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); - - chip->p_dma_size = size; - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO); - snd_dma_program(chip->dma1, runtime->dma_addr, size, - DMA_MODE_WRITE | DMA_AUTOINIT); - count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); - snd_ad1848_out(chip, AD1848_DATA_UPR_CNT, (unsigned char) (count >> 8)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -static int snd_ad1848_capture_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - unsigned long flags; - int err; - - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) - return err; - snd_ad1848_calibrate_mute(chip, 1); - snd_ad1848_set_data_format(chip, hw_params); - snd_ad1848_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_ad1848_mce_down(chip); - snd_ad1848_calibrate_mute(chip, 0); - return 0; -} - -static int snd_ad1848_capture_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int snd_ad1848_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); - - chip->c_dma_size = size; - chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); - snd_dma_program(chip->dma2, runtime->dma_addr, size, - DMA_MODE_READ | DMA_AUTOINIT); - count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; - spin_lock_irqsave(&chip->reg_lock, flags); - snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); - snd_ad1848_out(chip, AD1848_DATA_UPR_CNT, (unsigned char) (count >> 8)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - static irqreturn_t snd_ad1848_interrupt(int irq, void *dev_id) { struct snd_wss *chip = dev_id; @@ -600,28 +254,6 @@ static irqreturn_t snd_ad1848_interrupt( return IRQ_HANDLED; } -static snd_pcm_uframes_t snd_ad1848_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) - return 0; - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_ad1848_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE)) - return 0; - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - /* */ @@ -746,6 +378,16 @@ static int snd_ad1848_probe(struct snd_w snd_ad1848_out(chip, i, *ptr++); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_ad1848_mce_up(chip); + /* init needed for WSS pcm */ + spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | + AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | + AD1848_CAPTURE_PIO | + AD1848_CALIB_MODE); + chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; + snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); + spin_unlock_irqrestore(&chip->reg_lock, flags); snd_ad1848_mce_down(chip); return 0; /* all things are ok.. */ } @@ -754,102 +396,6 @@ static int snd_ad1848_probe(struct snd_w */ -static struct snd_pcm_hardware snd_ad1848_playback = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5510, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_ad1848_capture = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5510, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -/* - - */ - -static int snd_ad1848_playback_open(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - err = snd_ad1848_open(chip, WSS_MODE_PLAY); - if (err < 0) - return err; - chip->playback_substream = substream; - runtime->hw = snd_ad1848_playback; - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return 0; -} - -static int snd_ad1848_capture_open(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - err = snd_ad1848_open(chip, WSS_MODE_RECORD); - if (err < 0) - return err; - chip->capture_substream = substream; - runtime->hw = snd_ad1848_capture; - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return 0; -} - -static int snd_ad1848_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - - chip->mode &= ~WSS_MODE_PLAY; - chip->playback_substream = NULL; - snd_ad1848_close(chip); - return 0; -} - -static int snd_ad1848_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_wss *chip = snd_pcm_substream_chip(substream); - - chip->mode &= ~WSS_MODE_RECORD; - chip->capture_substream = NULL; - snd_ad1848_close(chip); - return 0; -} - static int snd_ad1848_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); @@ -869,22 +415,6 @@ static int snd_ad1848_dev_free(struct sn return snd_ad1848_free(chip); } -static const char *snd_ad1848_chip_id(struct snd_wss *chip) -{ - switch (chip->hardware) { - case WSS_HW_AD1847: - return "AD1847"; - case WSS_HW_AD1848: - return "AD1848"; - case WSS_HW_CS4248: - return "CS4248"; - case WSS_HW_CMI8330: - return "CMI8330/C3D"; - default: - return "???"; - } -} - int snd_ad1848_create(struct snd_card *card, unsigned long port, int irq, int dma, @@ -958,65 +488,6 @@ int snd_ad1848_create(struct snd_card *c EXPORT_SYMBOL(snd_ad1848_create); -static struct snd_pcm_ops snd_ad1848_playback_ops = { - .open = snd_ad1848_playback_open, - .close = snd_ad1848_playback_close, - .ioctl = snd_ad1848_ioctl, - .hw_params = snd_ad1848_playback_hw_params, - .hw_free = snd_ad1848_playback_hw_free, - .prepare = snd_ad1848_playback_prepare, - .trigger = snd_ad1848_playback_trigger, - .pointer = snd_ad1848_playback_pointer, -}; - -static struct snd_pcm_ops snd_ad1848_capture_ops = { - .open = snd_ad1848_capture_open, - .close = snd_ad1848_capture_close, - .ioctl = snd_ad1848_ioctl, - .hw_params = snd_ad1848_capture_hw_params, - .hw_free = snd_ad1848_capture_hw_free, - .prepare = snd_ad1848_capture_prepare, - .trigger = snd_ad1848_capture_trigger, - .pointer = snd_ad1848_capture_pointer, -}; - -int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(chip->card, "AD1848", device, 1, 1, &pcm)) < 0) - return err; - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ad1848_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ad1848_capture_ops); - - pcm->private_data = chip; - pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; - strcpy(pcm->name, snd_ad1848_chip_id(chip)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64 * 1024, - chip->dma1 > 3 ? - 128 * 1024 : 64 * 1024); - - chip->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_pcm); - -const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction) -{ - return direction == SNDRV_PCM_STREAM_PLAYBACK ? - &snd_ad1848_playback_ops : &snd_ad1848_capture_ops; -} - -EXPORT_SYMBOL(snd_ad1848_get_pcm_ops); - /* * INIT part */ diff -urp linux-alsa/sound/isa/cmi8330.c linux-mm/sound/isa/cmi8330.c --- linux-alsa/sound/isa/cmi8330.c 2008-07-17 10:49:34.887464085 +0200 +++ linux-mm/sound/isa/cmi8330.c 2008-07-17 12:53:10.240510625 +0200 @@ -404,7 +404,7 @@ static int __devinit snd_cmi8330_pcm(str chip->streams[CMI_SB_STREAM].private_data = chip->sb; /* AD1848 */ - ops = snd_ad1848_get_pcm_ops(CMI_AD_STREAM); + ops = snd_wss_get_pcm_ops(CMI_AD_STREAM); chip->streams[CMI_AD_STREAM].ops = *ops; chip->streams[CMI_AD_STREAM].open = ops->open; chip->streams[CMI_AD_STREAM].ops.open = cmi_open_callbacks[CMI_AD_STREAM]; diff -urp linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-mm/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-17 13:33:14.641275632 +0200 +++ linux-mm/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-17 13:13:22.989176217 +0200 @@ -750,18 +750,15 @@ static int __devinit snd_opti9xx_probe(s #ifdef OPTi93X chip->codec = codec; #endif - error = snd_wss_pcm(codec, 0, &pcm); - if (error < 0) - return error; #else error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, WSS_HW_DETECT, &codec); if (error < 0) return error; - error = snd_ad1848_pcm(codec, 0, &pcm); +#endif + error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; -#endif error = snd_wss_mixer(codec); if (error < 0) return error; diff -urp linux-alsa/sound/isa/sc6000.c linux-mm/sound/isa/sc6000.c --- linux-alsa/sound/isa/sc6000.c 2008-07-17 13:33:14.653267537 +0200 +++ linux-mm/sound/isa/sc6000.c 2008-07-17 20:02:47.610139873 +0200 @@ -554,10 +554,10 @@ static int __devinit snd_sc6000_probe(st goto err_unmap2; card->private_data = chip; - err = snd_ad1848_pcm(chip, 0, NULL); + err = snd_wss_pcm(chip, 0, NULL); if (err < 0) { snd_printk(KERN_ERR PFX - "error creating new ad1848 PCM device\n"); + "error creating new WSS PCM device\n"); goto err_unmap2; } err = snd_wss_mixer(chip); diff -urp linux-alsa/sound/isa/sgalaxy.c linux-mm/sound/isa/sgalaxy.c --- linux-alsa/sound/isa/sgalaxy.c 2008-07-17 13:33:14.657277157 +0200 +++ linux-mm/sound/isa/sgalaxy.c 2008-07-17 13:00:01.043573560 +0200 @@ -271,8 +271,9 @@ static int __devinit snd_sgalaxy_probe(s goto _err; card->private_data = chip; - if ((err = snd_ad1848_pcm(chip, 0, NULL)) < 0) { - snd_printdd(PFX "error creating new ad1848 PCM device\n"); + err = snd_wss_pcm(chip, 0, NULL); + if (err < 0) { + snd_printdd(PFX "error creating new WSS PCM device\n"); goto _err; } err = snd_wss_mixer(chip); diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-mm/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-07-17 13:33:14.697267841 +0200 +++ linux-mm/sound/isa/wss/wss_lib.c 2008-07-17 22:07:33.562145740 +0200 @@ -363,10 +363,10 @@ static void snd_wss_busy_wait(struct snd for (timeout = 5; timeout > 0; timeout--) wss_inb(chip, CS4231P(REGSEL)); /* end of cleanup sequence */ - for (timeout = 250; + for (timeout = 2500; timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) - udelay(10); + udelay(10); } void snd_wss_mce_up(struct snd_wss *chip) @@ -396,6 +396,7 @@ void snd_wss_mce_down(struct snd_wss *ch unsigned long flags; unsigned long end_time; int timeout; + int hw_mask = WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK | WSS_HW_AD1848; snd_wss_busy_wait(chip); @@ -410,10 +411,8 @@ void snd_wss_mce_down(struct snd_wss *ch spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0 || - !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { + if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask)) return; - } /* * Wait for (possible -- during init auto-calibration may not be set) @@ -584,12 +583,14 @@ static void snd_wss_calibrate_mute(struc mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); snd_wss_dout(chip, CS4231_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); - snd_wss_dout(chip, CS4231_LEFT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); - snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, - mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); - snd_wss_dout(chip, CS4231_MONO_CTRL, - mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) { + snd_wss_dout(chip, CS4231_LEFT_LINE_IN, + mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, + mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + snd_wss_dout(chip, CS4231_MONO_CTRL, + mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); + } if (chip->hardware == WSS_HW_INTERWAVE) { snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); @@ -688,7 +689,10 @@ static void snd_wss_capture_format(struc spin_lock_irqsave(&chip->reg_lock, flags); } } - snd_wss_out(chip, CS4231_REC_FORMAT, cdfr); + if (chip->hardware & WSS_HW_AD1848_MASK) + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, cdfr); + else + snd_wss_out(chip, CS4231_REC_FORMAT, cdfr); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); } @@ -799,7 +803,9 @@ static void snd_wss_init(struct snd_wss snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + snd_wss_out(chip, CS4231_REC_FORMAT, + chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); @@ -825,18 +831,20 @@ static int snd_wss_open(struct snd_wss * } /* ok. now enable and ack CODEC IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | - CS4231_RECORD_IRQ | - CS4231_TIMER_IRQ); - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) { + snd_wss_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | + CS4231_RECORD_IRQ | CS4231_TIMER_IRQ); + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + } wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ chip->image[CS4231_PIN_CTRL] |= CS4231_IRQ_ENABLE; snd_wss_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); - snd_wss_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | - CS4231_RECORD_IRQ | - CS4231_TIMER_IRQ); - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) { + snd_wss_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | + CS4231_RECORD_IRQ | CS4231_TIMER_IRQ); + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + } spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; @@ -858,7 +866,8 @@ static void snd_wss_close(struct snd_wss /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ chip->image[CS4231_PIN_CTRL] &= ~CS4231_IRQ_ENABLE; @@ -881,7 +890,8 @@ static void snd_wss_close(struct snd_wss } /* clear IRQ again */ - snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + if (!(chip->hardware & WSS_HW_AD1848_MASK)) + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -1002,7 +1012,13 @@ static int snd_wss_capture_prepare(struc chip->c_dma_size = size; chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); - count = snd_wss_get_count(chip->image[CS4231_REC_FORMAT], count) - 1; + if (chip->hardware & WSS_HW_AD1848_MASK) + count = snd_wss_get_count(chip->image[CS4231_PLAYBK_FORMAT], + count); + else + count = snd_wss_get_count(chip->image[CS4231_REC_FORMAT], + count); + count--; if (chip->single_dma && chip->hardware != WSS_HW_INTERWAVE) { snd_wss_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); snd_wss_out(chip, CS4231_PLY_UPR_CNT, @@ -1322,10 +1338,15 @@ static int snd_wss_playback_open(struct runtime->hw = snd_wss_playback; + /* hardware limitation of older chipsets */ + if (chip->hardware & WSS_HW_AD1848_MASK) + runtime->hw.formats &= ~(SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_S16_BE); + /* hardware bug in InterWave chipset */ if (chip->hardware == WSS_HW_INTERWAVE && chip->dma1 > 3) runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_MU_LAW; - + /* hardware limitation of cheap chips */ if (chip->hardware == WSS_HW_CS4235 || chip->hardware == WSS_HW_CS4239) @@ -1360,6 +1381,11 @@ static int snd_wss_capture_open(struct s runtime->hw = snd_wss_capture; + /* hardware limitation of older chipsets */ + if (chip->hardware == WSS_HW_INTERWAVE && chip->dma1 > 3) + runtime->hw.formats &= ~(SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_S16_BE); + /* hardware limitation of cheap chips */ if (chip->hardware == WSS_HW_CS4235 || chip->hardware == WSS_HW_CS4239) @@ -1404,6 +1430,26 @@ static int snd_wss_capture_close(struct return 0; } +static void snd_wss_thinkpad_twiddle(struct snd_wss *chip, int on) +{ + int tmp; + + if (!chip->thinkpad_flag) + return; + + outb(0x1c, AD1848_THINKPAD_CTL_PORT1); + tmp = inb(AD1848_THINKPAD_CTL_PORT2); + + if (on) + /* turn it on */ + tmp |= AD1848_THINKPAD_CS4248_ENABLE_BIT; + else + /* turn it off */ + tmp &= ~AD1848_THINKPAD_CS4248_ENABLE_BIT; + + outb(tmp, AD1848_THINKPAD_CTL_PORT2); +} + #ifdef CONFIG_PM /* lowlevel suspend callback for CS4231 */ @@ -1417,6 +1463,8 @@ static void snd_wss_suspend(struct snd_w for (reg = 0; reg < 32; reg++) chip->image[reg] = snd_wss_in(chip, reg); spin_unlock_irqrestore(&chip->reg_lock, flags); + if (chip->thinkpad_flag) + snd_wss_thinkpad_twiddle(chip, 0); } /* lowlevel resume callback for CS4231 */ @@ -1426,6 +1474,8 @@ static void snd_wss_resume(struct snd_ws unsigned long flags; /* int timeout; */ + if (chip->thinkpad_flag) + snd_wss_thinkpad_twiddle(chip, 1); snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); for (reg = 0; reg < 32; reg++) { @@ -1495,21 +1545,44 @@ static int snd_wss_dev_free(struct snd_d const char *snd_wss_chip_id(struct snd_wss *chip) { switch (chip->hardware) { - case WSS_HW_CS4231: return "CS4231"; - case WSS_HW_CS4231A: return "CS4231A"; - case WSS_HW_CS4232: return "CS4232"; - case WSS_HW_CS4232A: return "CS4232A"; - case WSS_HW_CS4235: return "CS4235"; - case WSS_HW_CS4236: return "CS4236"; - case WSS_HW_CS4236B: return "CS4236B"; - case WSS_HW_CS4237B: return "CS4237B"; - case WSS_HW_CS4238B: return "CS4238B"; - case WSS_HW_CS4239: return "CS4239"; - case WSS_HW_INTERWAVE: return "AMD InterWave"; - case WSS_HW_OPL3SA2: return chip->card->shortname; - case WSS_HW_AD1845: return "AD1845"; - case WSS_HW_OPTI93X: return "OPTi 93x"; - default: return "???"; + case WSS_HW_CS4231: + return "CS4231"; + case WSS_HW_CS4231A: + return "CS4231A"; + case WSS_HW_CS4232: + return "CS4232"; + case WSS_HW_CS4232A: + return "CS4232A"; + case WSS_HW_CS4235: + return "CS4235"; + case WSS_HW_CS4236: + return "CS4236"; + case WSS_HW_CS4236B: + return "CS4236B"; + case WSS_HW_CS4237B: + return "CS4237B"; + case WSS_HW_CS4238B: + return "CS4238B"; + case WSS_HW_CS4239: + return "CS4239"; + case WSS_HW_INTERWAVE: + return "AMD InterWave"; + case WSS_HW_OPL3SA2: + return chip->card->shortname; + case WSS_HW_AD1845: + return "AD1845"; + case WSS_HW_OPTI93X: + return "OPTi 93x"; + case WSS_HW_AD1847: + return "AD1847"; + case WSS_HW_AD1848: + return "AD1848"; + case WSS_HW_CS4248: + return "CS4248"; + case WSS_HW_CMI8330: + return "CMI8330/C3D"; + default: + return "???"; } } EXPORT_SYMBOL(snd_wss_chip_id); @@ -1665,7 +1738,8 @@ int snd_wss_pcm(struct snd_wss *chip, in struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "CS4231", device, 1, 1, &pcm)) < 0) + err = snd_pcm_new(chip->card, "WSS", device, 1, 1, &pcm); + if (err < 0) return err; spin_lock_init(&chip->reg_lock); @@ -1675,6 +1749,12 @@ int snd_wss_pcm(struct snd_wss *chip, in snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops); + /* temporary */ + if (chip->hardware & WSS_HW_AD1848_MASK) { + chip->rate_constraint = snd_wss_xrate; + chip->set_playback_format = snd_wss_playback_format; + chip->set_capture_format = snd_wss_capture_format; + } /* global setup */ pcm->private_data = chip; pcm->info_flags = 0; @@ -2084,6 +2164,13 @@ int snd_wss_mixer(struct snd_wss *chip) } EXPORT_SYMBOL(snd_wss_mixer); +const struct snd_pcm_ops *snd_wss_get_pcm_ops(int direction) +{ + return direction == SNDRV_PCM_STREAM_PLAYBACK ? + &snd_wss_playback_ops : &snd_wss_capture_ops; +} +EXPORT_SYMBOL(snd_wss_get_pcm_ops); + /* * INIT part */ ---------------------------------------------------------------------- Partyjka w Chinczyka? Graj >>> http://link.interia.pl/f1e67

1 0

[alsa-devel] Weirdness with on-board CM6501 USB
by Neil Schemenauer 18 Jul '08

18 Jul '08

Hi, I have a Asus M2N-E SLI board with a CM6501 sound on-board. I asked some months ago if anyone had any advice for how to get it working: http://article.gmane.org/gmane.linux.alsa.user/29626 Apparently there are still people having problems with this card. Today I spend some more time tinkering with it and have finally got some noise out of it. The magic is a .asoundrc file as follows: pcm.!default { type plug slave.pcm "surround51:CARD=1" slave.channels 6 route_policy duplicate } ctl.!default { type hw card 1 } Note that it's card "1" because I have a second PCI sound card. I have my stereo speakers connected to the gray plug (side speaker out according to the manual). The lime plug is supposed to be for front speakers but no signal is present. The volume is turned up on all mixer controls. Anyone have ideas on why no sound comes out of the front speaker plug? Regards, Neil

1 0

[alsa-devel] Several versions of 1.0.15??
by Alan Horstmann 17 Jul '08

17 Jul '08

Sorry to have a flood of posts; I have a flood of problems, though. We are building (ARM) from 2.6.24.4 kernel, and experiencing issues like segfaults with aplay and alsamixer that might relate to incompatible versions of driver, lib, utils, so I have tried to check carefully. The kernel has 1.0.15, dated 20 Nov 07, but when I check Alsa-git: http://git.alsa-project.org/?p=alsa-kernel.git;a=history;f=include/sound/ version.h;hb=fe0a3fe324811385b64790d42079bf534798a0cd 3 tags are shown of 1.0.15, the first is 16-10-07, the last 20-11-07. However, the 1.0.15 tarballs (driver, lib, utils) at ftp://ftp.alsa-project.org/pub are dated 15-10-07. (Which I presume is what the builds will grab.) A lot of changes were commited on 16-10-07, and some up to 20-11-07. Is it possible for this to cause a problem? Should I attempt to create snapshots of 20-11-07 to build correctly with the kernel, or is it concern about nothing. Alan

1 0

[alsa-devel] [PATCH 1/2] ASoC: Refector DAPM event handler
by Mark Brown 17 Jul '08

17 Jul '08

The DAPM event callback code has many layers of indentation, taking it over 80 columns. Refactor the code to give less indentation in order to avoid checkpatch issues on further changes and exploding indentation. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 79 ++++++++++++++++++++++++++------------------------ 1 files changed, 41 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061..17698ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -586,45 +586,48 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) power_change = (w->power == power) ? 0: 1; w->power = power; + if (!power_change) + continue; + /* call any power change event handlers */ - if (power_change) { - if (w->event) { - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); - if (power) { - /* power up event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - } else { - /* power down event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - } - } else - /* no event handler */ - dapm_update_bits(w); + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + dapm_update_bits(w); + + /* power up post event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; } } } -- 1.5.6.2

1 1

[alsa-devel] snd_pcm_writei() -last value sustained
by Alan Horstmann 17 Jul '08

17 Jul '08

Working on an app I am seeing several strange effects, and this is one of them. It can be demonstrated with the pcm.c example program, and I have attempted to attach a small shot of Audacity capturing the effect. After writing to the pcm with snd_pcm_writei() a certain amount of data (I have pcm.c doing 2 periods) the value captured from the soundcard remains at the last value thereafter; clearly this card doesn't have DC elimination or this would not be seen. The sound card is ice1712 (DMX6fire); I had always noticed the PCM meters in envy24control often stayed high if something like XMMS was stopped or paused during play, but thought it a metering fault. Now it appears to be actual pcm value that 'gets stuck'. Close examination with Audacity shows it is in fact the last-but-one value that is sustained. Is this expected behaviour, configuration fault, driver bug or other problem? I had expected that after the block of data had been written out as audio the signal would return to zero straightaway. Can anyone give me any clues? Alan

2 6

[alsa-devel] [PATCH 1/2] ASoC: Refector DAPM event handler
by Mark Brown 17 Jul '08

17 Jul '08

The DAPM event callback code has many layers of indentation. Refector the code to give less indentation in order to facilitiate further refactoring. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 76 +++++++++++++++++++++++++------------------------- 1 files changed, 38 insertions(+), 38 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2c87061..9307b10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -587,44 +587,44 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) w->power = power; /* call any power change event handlers */ - if (power_change) { - if (w->event) { - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); - if (power) { - /* power up event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMU){ - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - } else { - /* power down event */ - if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - dapm_update_bits(w); - if (w->event_flags & SND_SOC_DAPM_POST_PMD) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - } - } else - /* no event handler */ - dapm_update_bits(w); + if (power_change && w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power_change && power && w->event && + w->event_flags & SND_SOC_DAPM_PRE_PMU) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (power_change && !power && w->event && + w->event_flags & SND_SOC_DAPM_PRE_PMD) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + dapm_update_bits(w); + + /* power up post event */ + if (power_change && power && w->event && + w->event_flags & SND_SOC_DAPM_POST_PMU){ + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (power_change && !power && w->event && + w->event_flags & SND_SOC_DAPM_POST_PMD) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; } } } -- 1.5.6.2

2 7

[alsa-devel] [PATCH] ALSA: ac97: copy subsys values from the PCI device
by Rene Herman 17 Jul '08

17 Jul '08

Hi. Just getting to know to the code... Why do we reread the subsys values from PCI config space in the AC97 code? (this in addition to why we store the values in the ac97 struct in the first place if they're available through ac97->pci but that's secondary) Rene.

2 2

[PATCH] ALSA: ac97: copy subsys values from the PCI device
by Rene Herman 17 Jul '08

17 Jul '08

copy the subsys values from the PCI device instead of rereading them. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ac97/ac97_codec.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 45fd290..4f4343d 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2015,8 +2015,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, #ifdef CONFIG_PCI if (ac97->pci) { - pci_read_config_word(ac97->pci, PCI_SUBSYSTEM_VENDOR_ID, &ac97->subsystem_vendor); - pci_read_config_word(ac97->pci, PCI_SUBSYSTEM_ID, &ac97->subsystem_device); + ac97->subsystem_vendor = ac97->pci->subsystem_vendor; + ac97->subsystem_device = ac97->pci->subsystem_device; } #endif if (bus->ops->reset) { -- 1.5.5 --------------000002040805080104050207--

1 0

[alsa-devel] [PATCH] ALSA: ens1370: communicate PCI device to AC97
by Rene Herman 17 Jul '08

17 Jul '08

Currently the subsys_vendor/device values always display as 0x0000 in the codec97#n/ac97#m-k file at least.

1 0

[PATCH] ALSA: ens1370: communicate PCI device to AC97
by Rene Herman 17 Jul '08

17 Jul '08

communicate the ES137x PCI device to the AC97 code for its subsys_vendor/device values Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ens1370.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 38808fb..7298fe0 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1629,6 +1629,7 @@ static int __devinit snd_ensoniq_1371_mixer(struct ensoniq *ensoniq, memset(&ac97, 0, sizeof(ac97)); ac97.private_data = ensoniq; ac97.private_free = snd_ensoniq_mixer_free_ac97; + ac97.pci = ensoniq->pci; ac97.scaps = AC97_SCAP_AUDIO; if ((err = snd_ac97_mixer(pbus, &ac97, &ensoniq->u.es1371.ac97)) < 0) return err; -- 1.5.5 --------------070204000106090400030805--

1 0

[alsa-devel] PCM/ALSA loopback (codec-->DMA_RX-->DMA_TX-->codec)
by Norbert van Bolhuis 17 Jul '08

17 Jul '08

to be able to easily check our entire analog audio in/out cicuits (with a sweep generator feeding mic in and measuring the signal from audio out) I'd like to know whether there's any low level loopback available in ALSA core (or ALSA lib). The closer to HW the better it is. This is what our analog/digital audio path look like: mic in -> analog in circuitry -> CODEC -> analog out circuitry -> out | | v ^ | | ----- CPU ----- kernel alsa core --------- alsa lib aplay/arecord The problem is there's no loopback/echo nor sidetone possible with the CODEC we use. Also the DMA between CPU and CODEC cannot be looped (no DMA_RX-->DMA_TX possible). I know it's possible to loop digital audio in user-space with arecord -D hw:0,0 .. | aplay -D hw:0,0 .. This will probably give latencies of 10-100ms (depending upon CPU, DMA priority, SW load, period_size, etc...). These are too high for the sweep generator. Is it possible to loopback a PCM in stream to a PCM out stream somehow in ALSA core/lib ? How does the latency ALSA lib example program do this ? -- This message has been scanned for viruses and is believed to be clean

1 0

[alsa-devel] [PATCH 5/xx] wss_lib: use wss constants instead of ad1848 ones
by Krzysztof Helt 17 Jul '08

17 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use wss constants for mode. Move ad1848 hardware constants to the wss.h. Move mixer tlv macros into the ad1848_lib.c from the ad1848.h. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Another step toward unification of the ad1848_lib and wss_lib. I would like to ask Rene for gathering all these patches into his git tree so they can be pushed from there once the unification is finished. This is the fifth patch. The previous four are here: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008766.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008669.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008978.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008901.html diff -urp linux-alsa/include/sound/ad1848.h linux-2.6.25/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/include/sound/ad1848.h 2008-07-08 21:43:51.000000000 +0200 @@ -104,24 +104,6 @@ #define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ #define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ -/* defines for codec.mode */ - -#define AD1848_MODE_NONE 0x0000 -#define AD1848_MODE_PLAY 0x0001 -#define AD1848_MODE_CAPTURE 0x0002 -#define AD1848_MODE_TIMER 0x0004 -#define AD1848_MODE_OPEN (AD1848_MODE_PLAY|AD1848_MODE_CAPTURE|AD1848_MODE_TIMER) -#define AD1848_MODE_RUNNING 0x0010 - -/* defines for codec.hardware */ - -#define AD1848_HW_DETECT 0x0000 /* let AD1848 driver detect chip */ -#define AD1848_HW_AD1847 0x0001 /* AD1847 chip */ -#define AD1848_HW_AD1848 0x0002 /* AD1848 chip */ -#define AD1848_HW_CS4248 0x0003 /* CS4248 chip */ -#define AD1848_HW_CMI8330 0x0004 /* CMI8330 chip */ -#define AD1848_HW_THINKPAD 0x0005 /* Thinkpad 360/750/755 */ - /* IBM Thinkpad specific stuff */ #define AD1848_THINKPAD_CTL_PORT1 0x15e8 #define AD1848_THINKPAD_CTL_PORT2 0x15e9 @@ -167,26 +149,12 @@ struct ad1848_mix_elem { .type = AD1848_MIX_SINGLE, \ .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert) } -#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_SINGLE, \ - .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ - .tlv = xtlv } - #define AD1848_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ { .name = xname, \ .index = xindex, \ .type = AD1848_MIX_DOUBLE, \ .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert) } -#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_DOUBLE, \ - .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ - .tlv = xtlv } - int snd_ad1848_add_ctl_elem(struct snd_wss *chip, const struct ad1848_mix_elem *c); diff -urp linux-alsa/include/sound/wss.h linux-2.6.25/include/sound/wss.h --- linux-alsa/include/sound/wss.h 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/include/sound/wss.h 2008-07-08 21:52:32.000000000 +0200 @@ -55,6 +55,12 @@ #define WSS_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ #define WSS_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ #define WSS_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ +#define WSS_HW_AD1848_MASK 0x0800 /* AD1848 serie (half duplex) */ +#define WSS_HW_AD1847 0x0801 /* AD1847 chip */ +#define WSS_HW_AD1848 0x0802 /* AD1848 chip */ +#define WSS_HW_CS4248 0x0803 /* CS4248 chip */ +#define WSS_HW_CMI8330 0x0804 /* CMI8330 chip */ +#define WSS_HW_THINKPAD 0x0805 /* Thinkpad 360/750/755 */ /* compatible, but clones */ #define WSS_HW_INTERWAVE 0x1000 /* InterWave chip */ #define WSS_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ diff -urp linux-alsa/sound/isa/ad1848/ad1848.c linux-2.6.25/sound/isa/ad1848/ad1848.c --- linux-alsa/sound/isa/ad1848/ad1848.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848.c 2008-07-08 18:06:22.000000000 +0200 @@ -96,7 +96,7 @@ static int __devinit snd_ad1848_probe(st return -EINVAL; error = snd_ad1848_create(card, port[n], irq[n], dma1[n], - thinkpad[n] ? AD1848_HW_THINKPAD : AD1848_HW_DETECT, &chip); + thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT, &chip); if (error < 0) goto out; diff -urp linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-2.6.25/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848_lib.c 2008-07-08 21:43:51.000000000 +0200 @@ -283,14 +283,12 @@ static int snd_ad1848_trigger(struct snd return 0; } snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] |= what); - chip->mode |= AD1848_MODE_RUNNING; } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { if (!(chip->image[AD1848_IFACE_CTRL] & what)) { spin_unlock(&chip->reg_lock); return 0; } snd_ad1848_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL] &= ~what); - chip->mode &= ~AD1848_MODE_RUNNING; } else { result = -EINVAL; } @@ -378,7 +376,7 @@ static int snd_ad1848_open(struct snd_ws { unsigned long flags; - if (chip->mode & AD1848_MODE_OPEN) + if (chip->mode & WSS_MODE_OPEN) return -EAGAIN; snd_ad1848_mce_down(chip); @@ -570,11 +568,9 @@ static irqreturn_t snd_ad1848_interrupt( { struct snd_wss *chip = dev_id; - if ((chip->mode & AD1848_MODE_PLAY) && chip->playback_substream && - (chip->mode & AD1848_MODE_RUNNING)) + if ((chip->mode & WSS_MODE_PLAY) && chip->playback_substream) snd_pcm_period_elapsed(chip->playback_substream); - if ((chip->mode & AD1848_MODE_CAPTURE) && chip->capture_substream && - (chip->mode & AD1848_MODE_RUNNING)) + if ((chip->mode & WSS_MODE_RECORD) && chip->capture_substream) snd_pcm_period_elapsed(chip->capture_substream); outb(0, AD1848P(chip, STATUS)); /* clear global interrupt bit */ return IRQ_HANDLED; @@ -690,19 +686,19 @@ static int snd_ad1848_probe(struct snd_w } if (id != 1) return -ENODEV; /* no valid device found */ - if (chip->hardware == AD1848_HW_DETECT) { + if (chip->hardware == WSS_HW_DETECT) { if (ad1847) { - chip->hardware = AD1848_HW_AD1847; + chip->hardware = WSS_HW_AD1847; } else { - chip->hardware = AD1848_HW_AD1848; + chip->hardware = WSS_HW_AD1848; rev = snd_ad1848_in(chip, AD1848_MISC_INFO); if (rev & 0x80) { - chip->hardware = AD1848_HW_CS4248; + chip->hardware = WSS_HW_CS4248; } else if ((rev & 0x0f) == 0x0a) { snd_ad1848_out(chip, AD1848_MISC_INFO, 0x40); for (i = 0; i < 16; ++i) { if (snd_ad1848_in(chip, i) != snd_ad1848_in(chip, i + 16)) { - chip->hardware = AD1848_HW_CMI8330; + chip->hardware = WSS_HW_CMI8330; break; } } @@ -782,7 +778,8 @@ static int snd_ad1848_playback_open(stru struct snd_pcm_runtime *runtime = substream->runtime; int err; - if ((err = snd_ad1848_open(chip, AD1848_MODE_PLAY)) < 0) + err = snd_ad1848_open(chip, WSS_MODE_PLAY); + if (err < 0) return err; chip->playback_substream = substream; runtime->hw = snd_ad1848_playback; @@ -798,7 +795,8 @@ static int snd_ad1848_capture_open(struc struct snd_pcm_runtime *runtime = substream->runtime; int err; - if ((err = snd_ad1848_open(chip, AD1848_MODE_CAPTURE)) < 0) + err = snd_ad1848_open(chip, WSS_MODE_RECORD); + if (err < 0) return err; chip->capture_substream = substream; runtime->hw = snd_ad1848_capture; @@ -812,7 +810,7 @@ static int snd_ad1848_playback_close(str { struct snd_wss *chip = snd_pcm_substream_chip(substream); - chip->mode &= ~AD1848_MODE_PLAY; + chip->mode &= ~WSS_MODE_PLAY; chip->playback_substream = NULL; snd_ad1848_close(chip); return 0; @@ -822,7 +820,7 @@ static int snd_ad1848_capture_close(stru { struct snd_wss *chip = snd_pcm_substream_chip(substream); - chip->mode &= ~AD1848_MODE_CAPTURE; + chip->mode &= ~WSS_MODE_RECORD; chip->capture_substream = NULL; snd_ad1848_close(chip); return 0; @@ -850,11 +848,16 @@ static int snd_ad1848_dev_free(struct sn static const char *snd_ad1848_chip_id(struct snd_wss *chip) { switch (chip->hardware) { - case AD1848_HW_AD1847: return "AD1847"; - case AD1848_HW_AD1848: return "AD1848"; - case AD1848_HW_CS4248: return "CS4248"; - case AD1848_HW_CMI8330: return "CMI8330/C3D"; - default: return "???"; + case WSS_HW_AD1847: + return "AD1847"; + case WSS_HW_AD1848: + return "AD1848"; + case WSS_HW_CS4248: + return "CS4248"; + case WSS_HW_CMI8330: + return "CMI8330/C3D"; + default: + return "???"; } } @@ -903,9 +906,9 @@ int snd_ad1848_create(struct snd_card *c chip->dma1 = dma; chip->dma2 = dma; - if (hardware == AD1848_HW_THINKPAD) { + if (hardware == WSS_HW_THINKPAD) { chip->thinkpad_flag = 1; - chip->hardware = AD1848_HW_DETECT; /* reset */ + chip->hardware = WSS_HW_DETECT; /* reset */ snd_ad1848_thinkpad_twiddle(chip, 1); } @@ -1214,6 +1217,20 @@ static const DECLARE_TLV_DB_SCALE(db_sca static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); +#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .name = xname, \ + .index = xindex, \ + .type = AD1848_MIX_SINGLE, \ + .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ + .tlv = xtlv } + +#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ +{ .name = xname, \ + .index = xindex, \ + .type = AD1848_MIX_DOUBLE, \ + .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ + .tlv = xtlv } + static struct ad1848_mix_elem snd_ad1848_controls[] = { AD1848_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), AD1848_DOUBLE_TLV("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, diff -urp linux-alsa/sound/isa/cmi8330.c linux-2.6.25/sound/isa/cmi8330.c --- linux-alsa/sound/isa/cmi8330.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/cmi8330.c 2008-07-08 21:43:51.000000000 +0200 @@ -465,12 +465,12 @@ static int __devinit snd_cmi8330_probe(s wssport[dev] + 4, wssirq[dev], wssdma[dev], - AD1848_HW_DETECT, + WSS_HW_DETECT, &acard->wss)) < 0) { snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); return err; } - if (acard->wss->hardware != AD1848_HW_CMI8330) { + if (acard->wss->hardware != WSS_HW_CMI8330) { snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); return -ENODEV; } diff -urp linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.25/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-08 18:06:22.000000000 +0200 @@ -771,7 +771,7 @@ static int __devinit snd_opti9xx_probe(s #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, - AD1848_HW_DETECT, &codec)) < 0) + WSS_HW_DETECT, &codec)) < 0) return error; if ((error = snd_ad1848_pcm(codec, 0, &pcm)) < 0) return error; diff -urp linux-alsa/sound/isa/sc6000.c linux-2.6.25/sound/isa/sc6000.c --- linux-alsa/sound/isa/sc6000.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/sc6000.c 2008-07-08 18:06:22.000000000 +0200 @@ -549,7 +549,7 @@ static int __devinit snd_sc6000_probe(st goto err_unmap2; err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, - AD1848_HW_DETECT, &chip); + WSS_HW_DETECT, &chip); if (err < 0) goto err_unmap2; card->private_data = chip; diff -urp linux-alsa/sound/isa/sgalaxy.c linux-2.6.25/sound/isa/sgalaxy.c --- linux-alsa/sound/isa/sgalaxy.c 2008-07-08 21:44:58.000000000 +0200 +++ linux-2.6.25/sound/isa/sgalaxy.c 2008-07-08 21:43:51.000000000 +0200 @@ -265,7 +265,7 @@ static int __devinit snd_sgalaxy_probe(s if ((err = snd_ad1848_create(card, wssport[dev] + 4, xirq, xdma1, - AD1848_HW_DETECT, &chip)) < 0) + WSS_HW_DETECT, &chip)) < 0) goto _err; card->private_data = chip; ---------------------------------------------------------------------- Partyjka w Chinczyka? Graj >>> http://link.interia.pl/f1e67

2 1

[alsa-devel] [PATCH] ALSA: ens1370: SRC stands for Sample Rate Converter
by Rene Herman 17 Jul '08

17 Jul '08

Trivial...

2 1

[alsa-devel] problems in building with DOXYGEN defined
by Rahul Ramasubramanian 17 Jul '08

17 Jul '08

hi i want to build alsa-lib-1.0.15 with DOXYGEN defined .. i read in the docs that if i add DOXYGEN to the PREDFINED field of doxygen.cfg , it is as good as doing a gcc -DDOXYGEN but after building it with the above option , i see that DOXYGEN still remains undefined , consequently , when i try to build alsa-utils-1.0.15 , it breaks in alsactl/name.c because it cant find the defination of function snd_timer_query_internal. This happens because the function is declared differenlty if DOXYGEN is not defined (timer_query.c). please help me in this Kind regards Rahul

1 0

[alsa-devel] [PATCH 8/xx] wss_lib: use wss mixer code instead of ad1848 one
by Krzysztof Helt 17 Jul '08

17 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss mixer code and kill the ad1848 mixer code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is the first significant code reduction. The mixer code is independent of the pcm part so the ad1848 cards work in this hybrid approach (ad1848 pcm code + wss mixer code). It allows testing of all changes from eight patches on existing hardware. I have tested this on the Gallant SC-6000 card (AD1848 codec). More tests are welcome especially on the cards using the ad1848_lib library. The next patch should kill the ad1848 pcm code so it will be big. I am working on it. diff -urp linux-alsa/include/sound/ad1848.h linux-2.6.25/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-07-08 21:55:47.000000000 +0200 +++ linux-2.6.25/include/sound/ad1848.h 2008-07-09 09:28:44.000000000 +0200 @@ -115,6 +115,5 @@ int snd_ad1848_create(struct snd_card *c int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); -int snd_ad1848_mixer(struct snd_wss *chip); #endif /* __SOUND_AD1848_H */ diff -urp linux-alsa/include/sound/wss.h linux-2.6.25/include/sound/wss.h --- linux-alsa/include/sound/wss.h 2008-07-08 21:55:42.000000000 +0200 +++ linux-2.6.25/include/sound/wss.h 2008-07-09 11:39:39.000000000 +0200 @@ -185,6 +185,27 @@ int snd_wss_put_single(struct snd_kcontr .get = snd_wss_get_double, .put = snd_wss_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define WSS_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_wss_info_single, \ + .get = snd_wss_get_single, .put = snd_wss_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } + +#define WSS_DOUBLE_TLV(xname, xindex, left_reg, right_reg, \ + shift_left, shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_wss_info_double, \ + .get = snd_wss_get_double, .put = snd_wss_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + + int snd_wss_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_wss_get_double(struct snd_kcontrol *kcontrol, diff -urp linux-alsa/sound/isa/ad1848/ad1848.c linux-2.6.25/sound/isa/ad1848/ad1848.c --- linux-alsa/sound/isa/ad1848/ad1848.c 2008-07-08 21:55:42.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848.c 2008-07-09 09:35:27.000000000 +0200 @@ -106,7 +106,7 @@ static int __devinit snd_ad1848_probe(st if (error < 0) goto out; - error = snd_ad1848_mixer(chip); + error = snd_wss_mixer(chip); if (error < 0) goto out; diff -urp linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-2.6.25/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-07-08 21:55:47.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848_lib.c 2008-07-09 12:31:37.000000000 +0200 @@ -1018,262 +1018,6 @@ const struct snd_pcm_ops *snd_ad1848_get EXPORT_SYMBOL(snd_ad1848_get_pcm_ops); /* - * MIXER part - */ - -static int snd_ad1848_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line", "Aux", "Mic", "Mix" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_ad1848_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[AD1848_LEFT_INPUT] & AD1848_MIXS_ALL) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[AD1848_RIGHT_INPUT] & AD1848_MIXS_ALL) >> 6; - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -static int snd_ad1848_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->reg_lock, flags); - left = (chip->image[AD1848_LEFT_INPUT] & ~AD1848_MIXS_ALL) | left; - right = (chip->image[AD1848_RIGHT_INPUT] & ~AD1848_MIXS_ALL) | right; - change = left != chip->image[AD1848_LEFT_INPUT] || - right != chip->image[AD1848_RIGHT_INPUT]; - snd_ad1848_out(chip, AD1848_LEFT_INPUT, left); - snd_ad1848_out(chip, AD1848_RIGHT_INPUT, right); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -static int snd_ad1848_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_ad1848_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_ad1848_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->reg_lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_ad1848_out(chip, reg, val); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -static int snd_ad1848_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_ad1848_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -static int snd_ad1848_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_wss *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->reg_lock, flags); - if (left_reg != right_reg) { - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_ad1848_out(chip, left_reg, val1); - snd_ad1848_out(chip, right_reg, val2); - } else { - val1 = (chip->image[left_reg] & ~((mask << shift_left) | (mask << shift_right))) | val1 | val2; - change = val1 != chip->image[left_reg]; - snd_ad1848_out(chip, left_reg, val1); - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); -static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); - -#define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ - .name = xname, .index = xindex, \ - .info = snd_ad1848_info_single, \ - .get = snd_ad1848_get_single, .put = snd_ad1848_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ - .tlv = { .p = (xtlv) } } - -#define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ - .name = xname, .index = xindex, \ - .info = snd_ad1848_info_double, \ - .get = snd_ad1848_get_double, .put = snd_ad1848_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ - (shift_right << 19) | (mask << 24) | (invert << 22), \ - .tlv = { .p = (xtlv) } } - - -static struct snd_kcontrol_new snd_ad1848_controls[] = { -WSS_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, - 7, 7, 1, 1), -AD1848_DOUBLE_TLV("PCM Playback Volume", 0, - AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, - db_scale_6bit), -WSS_DOUBLE("Aux Playback Switch", 0, - AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 0, - AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -WSS_DOUBLE("Aux Playback Switch", 1, - AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 1, - AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), -AD1848_DOUBLE_TLV("Capture Volume", 0, - AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0, - db_scale_rec_gain), -{ - .name = "Capture Source", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_mux, - .get = snd_ad1848_get_mux, - .put = snd_ad1848_put_mux, -}, -WSS_SINGLE("Loopback Capture Switch", 0, AD1848_LOOPBACK, 0, 1, 0), -AD1848_SINGLE_TLV("Loopback Capture Volume", 0, AD1848_LOOPBACK, 1, 63, 0, - db_scale_6bit), -}; - -int snd_ad1848_mixer(struct snd_wss *chip) -{ - struct snd_card *card; - struct snd_pcm *pcm; - unsigned int idx; - int err; - - snd_assert(chip != NULL && chip->pcm != NULL, return -EINVAL); - - pcm = chip->pcm; - card = chip->card; - - strcpy(card->mixername, pcm->name); - - for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_ad1848_controls[idx], chip)); - if (err < 0) - return err; - } - - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_mixer); - -/* * INIT part */ diff -urp linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.25/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-08 21:55:42.000000000 +0200 +++ linux-2.6.25/sound/isa/opti9xx/opti92x-ad1848.c 2008-07-09 12:36:06.000000000 +0200 @@ -753,6 +753,15 @@ static int __devinit snd_opti9xx_probe(s error = snd_wss_pcm(codec, 0, &pcm); if (error < 0) return error; +#else + error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, + chip->dma1, WSS_HW_DETECT, &codec); + if (error < 0) + return error; + error = snd_ad1848_pcm(codec, 0, &pcm); + if (error < 0) + return error; +#endif error = snd_wss_mixer(codec); if (error < 0) return error; @@ -760,7 +769,8 @@ static int __devinit snd_opti9xx_probe(s error = snd_wss_timer(codec, 0, &timer); if (error < 0) return error; -#else /* OPTI93X */ +#endif +#ifdef OPTi93X error = request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec); if (error < 0) { @@ -768,16 +778,6 @@ static int __devinit snd_opti9xx_probe(s return error; } #endif -#else - if ((error = snd_ad1848_create(card, chip->wss_base + 4, - chip->irq, chip->dma1, - WSS_HW_DETECT, &codec)) < 0) - return error; - if ((error = snd_ad1848_pcm(codec, 0, &pcm)) < 0) - return error; - if ((error = snd_ad1848_mixer(codec)) < 0) - return error; -#endif strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) diff -urp linux-alsa/sound/isa/sc6000.c linux-2.6.25/sound/isa/sc6000.c --- linux-alsa/sound/isa/sc6000.c 2008-07-08 21:55:42.000000000 +0200 +++ linux-2.6.25/sound/isa/sc6000.c 2008-07-09 09:29:55.000000000 +0200 @@ -560,9 +560,9 @@ static int __devinit snd_sc6000_probe(st "error creating new ad1848 PCM device\n"); goto err_unmap2; } - err = snd_ad1848_mixer(chip); + err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR PFX "error creating new ad1848 mixer\n"); + snd_printk(KERN_ERR PFX "error creating new WSS mixer\n"); goto err_unmap2; } err = snd_sc6000_mixer(chip); diff -urp linux-alsa/sound/isa/sgalaxy.c linux-2.6.25/sound/isa/sgalaxy.c --- linux-alsa/sound/isa/sgalaxy.c 2008-07-08 21:55:45.000000000 +0200 +++ linux-2.6.25/sound/isa/sgalaxy.c 2008-07-09 09:30:30.000000000 +0200 @@ -275,8 +275,9 @@ static int __devinit snd_sgalaxy_probe(s snd_printdd(PFX "error creating new ad1848 PCM device\n"); goto _err; } - if ((err = snd_ad1848_mixer(chip)) < 0) { - snd_printdd(PFX "error creating new ad1848 mixer\n"); + err = snd_wss_mixer(chip); + if (err < 0) { + snd_printdd(PFX "error creating new WSS mixer\n"); goto _err; } if ((err = snd_sgalaxy_mixer(chip)) < 0) { diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-2.6.25/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-07-08 21:55:47.000000000 +0200 +++ linux-2.6.25/sound/isa/wss/wss_lib.c 2008-07-09 12:36:32.000000000 +0200 @@ -33,6 +33,7 @@ #include <sound/core.h> #include <sound/wss.h> #include <sound/pcm_params.h> +#include <sound/tlv.h> #include <asm/io.h> #include <asm/dma.h> @@ -1914,16 +1915,58 @@ int snd_wss_put_double(struct snd_kcontr val1 <<= shift_left; val2 <<= shift_right; spin_lock_irqsave(&chip->reg_lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_wss_out(chip, left_reg, val1); - snd_wss_out(chip, right_reg, val2); + if (left_reg != right_reg) { + val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; + val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; + change = val1 != chip->image[left_reg] || + val2 != chip->image[right_reg]; + snd_wss_out(chip, left_reg, val1); + snd_wss_out(chip, right_reg, val2); + } else { + mask = (mask << shift_left) | (mask << shift_right); + val1 = (chip->image[left_reg] & ~mask) | val1 | val2; + change = val1 != chip->image[left_reg]; + snd_wss_out(chip, left_reg, val1); + } spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } EXPORT_SYMBOL(snd_wss_put_double); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); + +static struct snd_kcontrol_new snd_ad1848_controls[] = { +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, + 7, 7, 1, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), +WSS_DOUBLE("Aux Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), +WSS_DOUBLE("Aux Playback Switch", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE_TLV("Aux Playback Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), +{ + .name = "Capture Source", + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_wss_info_mux, + .get = snd_wss_get_mux, + .put = snd_wss_put_mux, +}, +WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0, + db_scale_6bit), +}; + static struct snd_kcontrol_new snd_wss_controls[] = { WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), @@ -2021,6 +2064,14 @@ int snd_wss_mixer(struct snd_wss *chip) if (err < 0) return err; } + else if (chip->hardware & WSS_HW_AD1848_MASK) + for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_ad1848_controls[idx], + chip)); + if (err < 0) + return err; + } else for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) { err = snd_ctl_add(card, ---------------------------------------------------------------------- Pogoda na dzis. Sprawdz >>> http://link.interia.pl/f1e42

1 0

[alsa-devel] [PATCH 7/xx] wss_lib: use CS4231P instead of AD1848P (kill the AD1848P)
by Krzysztof Helt 17 Jul '08

17 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use CS4231P instead of AD1848P (kill the AD1848P). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-alsa/include/sound/ad1848.h linux-2.6.25/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-07-08 19:37:38.000000000 +0200 +++ linux-2.6.25/include/sound/ad1848.h 2008-07-08 19:29:35.000000000 +0200 @@ -25,15 +25,6 @@ #include "pcm.h" #include <linux/interrupt.h> -/* IO ports */ - -#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) - -#define c_d_c_AD1848REGSEL 0 -#define c_d_c_AD1848REG 1 -#define c_d_c_AD1848STATUS 2 -#define c_d_c_AD1848PIO 3 - /* codec registers */ #define AD1848_LEFT_INPUT 0x00 /* left input control */ diff -urp linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-2.6.25/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-07-08 19:37:38.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848_lib.c 2008-07-08 19:43:16.000000000 +0200 @@ -103,7 +103,7 @@ static void snd_ad1848_wait(struct snd_w int timeout; for (timeout = 250; timeout > 0; timeout--) { - if ((inb(AD1848P(chip, REGSEL)) & AD1848_INIT) == 0) + if ((inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) == 0) break; udelay(100); } @@ -115,12 +115,12 @@ void snd_ad1848_out(struct snd_wss *chip { snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "auto calibration time out - " "reg = 0x%x, value = 0x%x\n", reg, value); #endif - outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); - outb(chip->image[reg] = value, AD1848P(chip, REG)); + outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); + outb(chip->image[reg] = value, chip->port + CS4231P(REG)); mb(); snd_printdd("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); @@ -132,8 +132,8 @@ static void snd_ad1848_dout(struct snd_w unsigned char reg, unsigned char value) { snd_ad1848_wait(chip); - outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); - outb(value, AD1848P(chip, REG)); + outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); + outb(value, chip->port + CS4231P(REG)); mb(); } @@ -141,37 +141,55 @@ static unsigned char snd_ad1848_in(struc { snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "auto calibration time out - " "reg = 0x%x\n", reg); #endif - outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); + outb(chip->mce_bit | reg, chip->port + CS4231P(REGSEL)); mb(); - return inb(AD1848P(chip, REG)); + return inb(chip->port + CS4231P(REG)); } #if 0 static void snd_ad1848_debug(struct snd_wss *chip) { - printk("AD1848 REGS: INDEX = 0x%02x ", inb(AD1848P(chip, REGSEL))); - printk(" STATUS = 0x%02x\n", inb(AD1848P(chip, STATUS))); - printk(" 0x00: left input = 0x%02x ", snd_ad1848_in(chip, 0x00)); - printk(" 0x08: playback format = 0x%02x\n", snd_ad1848_in(chip, 0x08)); - printk(" 0x01: right input = 0x%02x ", snd_ad1848_in(chip, 0x01)); - printk(" 0x09: iface (CFIG 1) = 0x%02x\n", snd_ad1848_in(chip, 0x09)); - printk(" 0x02: AUXA left = 0x%02x ", snd_ad1848_in(chip, 0x02)); - printk(" 0x0a: pin control = 0x%02x\n", snd_ad1848_in(chip, 0x0a)); - printk(" 0x03: AUXA right = 0x%02x ", snd_ad1848_in(chip, 0x03)); - printk(" 0x0b: init & status = 0x%02x\n", snd_ad1848_in(chip, 0x0b)); - printk(" 0x04: AUXB left = 0x%02x ", snd_ad1848_in(chip, 0x04)); - printk(" 0x0c: revision & mode = 0x%02x\n", snd_ad1848_in(chip, 0x0c)); - printk(" 0x05: AUXB right = 0x%02x ", snd_ad1848_in(chip, 0x05)); - printk(" 0x0d: loopback = 0x%02x\n", snd_ad1848_in(chip, 0x0d)); - printk(" 0x06: left output = 0x%02x ", snd_ad1848_in(chip, 0x06)); - printk(" 0x0e: data upr count = 0x%02x\n", snd_ad1848_in(chip, 0x0e)); - printk(" 0x07: right output = 0x%02x ", snd_ad1848_in(chip, 0x07)); - printk(" 0x0f: data lwr count = 0x%02x\n", snd_ad1848_in(chip, 0x0f)); + printk(KERN_DEBUG "AD1848 REGS: INDEX = 0x%02x ", + inb(chip->port + CS4231P(REGSEL))); + printk(KERN_DEBUG " STATUS = 0x%02x\n", + inb(chip->port + CS4231P(STATUS))); + printk(KERN_DEBUG " 0x00: left input = 0x%02x ", + snd_ad1848_in(chip, 0x00)); + printk(KERN_DEBUG " 0x08: playback format = 0x%02x\n", + snd_ad1848_in(chip, 0x08)); + printk(KERN_DEBUG " 0x01: right input = 0x%02x ", + snd_ad1848_in(chip, 0x01)); + printk(KERN_DEBUG " 0x09: iface (CFIG 1) = 0x%02x\n", + snd_ad1848_in(chip, 0x09)); + printk(KERN_DEBUG " 0x02: AUXA left = 0x%02x ", + snd_ad1848_in(chip, 0x02)); + printk(KERN_DEBUG " 0x0a: pin control = 0x%02x\n", + snd_ad1848_in(chip, 0x0a)); + printk(KERN_DEBUG " 0x03: AUXA right = 0x%02x ", + snd_ad1848_in(chip, 0x03)); + printk(KERN_DEBUG " 0x0b: init & status = 0x%02x\n", + snd_ad1848_in(chip, 0x0b)); + printk(KERN_DEBUG " 0x04: AUXB left = 0x%02x ", + snd_ad1848_in(chip, 0x04)); + printk(KERN_DEBUG " 0x0c: revision & mode = 0x%02x\n", + snd_ad1848_in(chip, 0x0c)); + printk(KERN_DEBUG " 0x05: AUXB right = 0x%02x ", + snd_ad1848_in(chip, 0x05)); + printk(KERN_DEBUG " 0x0d: loopback = 0x%02x\n", + snd_ad1848_in(chip, 0x0d)); + printk(KERN_DEBUG " 0x06: left output = 0x%02x ", + snd_ad1848_in(chip, 0x06)); + printk(KERN_DEBUG " 0x0e: data upr count = 0x%02x\n", + snd_ad1848_in(chip, 0x0e)); + printk(KERN_DEBUG " 0x07: right output = 0x%02x ", + snd_ad1848_in(chip, 0x07)); + printk(KERN_DEBUG " 0x0f: data lwr count = 0x%02x\n", + snd_ad1848_in(chip, 0x0f)); } #endif @@ -187,16 +205,17 @@ static void snd_ad1848_mce_up(struct snd snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit |= AD1848_MCE; - timeout = inb(AD1848P(chip, REGSEL)); + timeout = inb(chip->port + CS4231P(REGSEL)); if (timeout == 0x80) snd_printk(KERN_WARNING "mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); if (!(timeout & AD1848_MCE)) - outb(chip->mce_bit | (timeout & 0x1f), AD1848P(chip, REGSEL)); + outb(chip->mce_bit | (timeout & 0x1f), + chip->port + CS4231P(REGSEL)); spin_unlock_irqrestore(&chip->reg_lock, flags); } @@ -207,21 +226,25 @@ static void snd_ad1848_mce_down(struct s spin_lock_irqsave(&chip->reg_lock, flags); for (timeout = 5; timeout > 0; timeout--) - inb(AD1848P(chip, REGSEL)); + inb(chip->port + CS4231P(REGSEL)); /* end of cleanup sequence */ - for (timeout = 12000; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) + for (timeout = 12000; + timeout > 0 && (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT); + timeout--) udelay(100); snd_printdd("(1) timeout = %ld\n", timeout); #ifdef CONFIG_SND_DEBUG - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "mce_down [0x%lx] - auto calibration time out (0)\n", AD1848P(chip, REGSEL)); + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) + snd_printk(KERN_WARNING + "mce_down [0x%lx] - auto calibration time out (0)\n", + chip->port + CS4231P(REGSEL)); #endif chip->mce_bit &= ~AD1848_MCE; - reg = inb(AD1848P(chip, REGSEL)); - outb(chip->mce_bit | (reg & 0x1f), AD1848P(chip, REGSEL)); + reg = inb(chip->port + CS4231P(REGSEL)); + outb(chip->mce_bit | (reg & 0x1f), chip->port + CS4231P(REGSEL)); if (reg == 0x80) snd_printk(KERN_WARNING "mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); if ((reg & AD1848_MCE) == 0) { @@ -252,7 +275,8 @@ static void snd_ad1848_mce_down(struct s "mce_down - auto calibration time out (2)\n"); snd_printdd("(4) jiffies = %lu\n", jiffies); - snd_printd("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); + snd_printd("mce_down - exit = 0x%x\n", + inb(chip->port + CS4231P(REGSEL))); } static unsigned int snd_ad1848_get_count(unsigned char format, @@ -412,8 +436,8 @@ static int snd_ad1848_open(struct snd_ws /* ok. now enable and ack CODEC IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ chip->image[AD1848_PIN_CTRL] |= AD1848_IRQ_ENABLE; snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -431,8 +455,8 @@ static void snd_ad1848_close(struct snd_ return; /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ chip->image[AD1848_PIN_CTRL] &= ~AD1848_IRQ_ENABLE; snd_ad1848_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -449,8 +473,8 @@ static void snd_ad1848_close(struct snd_ /* clear IRQ again */ spin_lock_irqsave(&chip->reg_lock, flags); - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ - outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ + outb(0, chip->port + CS4231P(STATUS)); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = 0; @@ -572,7 +596,7 @@ static irqreturn_t snd_ad1848_interrupt( snd_pcm_period_elapsed(chip->playback_substream); if ((chip->mode & WSS_MODE_RECORD) && chip->capture_substream) snd_pcm_period_elapsed(chip->capture_substream); - outb(0, AD1848P(chip, STATUS)); /* clear global interrupt bit */ + outb(0, chip->port + CS4231P(STATUS)); /* clear global interrupt bit */ return IRQ_HANDLED; } @@ -638,8 +662,8 @@ static void snd_ad1848_resume(struct snd snd_ad1848_thinkpad_twiddle(chip, 1); /* clear any pendings IRQ */ - inb(AD1848P(chip, STATUS)); - outb(0, AD1848P(chip, STATUS)); + inb(chip->port + CS4231P(STATUS)); + outb(0, chip->port + CS4231P(STATUS)); mb(); snd_ad1848_mce_down(chip); @@ -662,7 +686,7 @@ static int snd_ad1848_probe(struct snd_w id = ad1847 = 0; for (i = 0; i < 1000; i++) { mb(); - if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) + if (inb(chip->port + CS4231P(REGSEL)) & AD1848_INIT) udelay(500); else { spin_lock_irqsave(&chip->reg_lock, flags); @@ -707,8 +731,8 @@ static int snd_ad1848_probe(struct snd_w } } spin_lock_irqsave(&chip->reg_lock, flags); - inb(AD1848P(chip, STATUS)); /* clear any pendings IRQ */ - outb(0, AD1848P(chip, STATUS)); + inb(chip->port + CS4231P(STATUS)); /* clear any pendings IRQ */ + outb(0, chip->port + CS4231P(STATUS)); mb(); spin_unlock_irqrestore(&chip->reg_lock, flags); diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-2.6.25/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-07-08 18:02:50.000000000 +0200 +++ linux-2.6.25/sound/isa/wss/wss_lib.c 2008-07-08 19:31:21.000000000 +0200 @@ -164,7 +164,6 @@ static inline void wss_outb(struct snd_w { outb(val, chip->port + offset); } -EXPORT_SYMBOL(snd_wss_out); static inline u8 wss_inb(struct snd_wss *chip, u8 offset) { @@ -228,6 +227,7 @@ void snd_wss_out(struct snd_wss *chip, u snd_printdd("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); } +EXPORT_SYMBOL(snd_wss_out); unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg) { ---------------------------------------------------------------------- Najciekawsze miejsca w Polsce i na swiecie! Zobacz >>> http://link.interia.pl/f1e60

1 0

[alsa-devel] [PATCH 6/xx] wss_lib: replace ad1848 mixer element macros with wss ones
by Krzysztof Helt 17 Jul '08

17 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use the wss macros instead of ad1848 ones. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-alsa/include/sound/ad1848.h linux-2.6.25/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-07-08 18:15:01.000000000 +0200 +++ linux-2.6.25/include/sound/ad1848.h 2008-07-08 19:05:46.000000000 +0200 @@ -126,36 +126,4 @@ int snd_ad1848_pcm(struct snd_wss *chip, const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); int snd_ad1848_mixer(struct snd_wss *chip); -/* exported mixer stuffs */ -enum { AD1848_MIX_SINGLE, AD1848_MIX_DOUBLE, AD1848_MIX_CAPTURE }; - -#define AD1848_MIXVAL_SINGLE(reg, shift, mask, invert) \ - ((reg) | ((shift) << 8) | ((mask) << 16) | ((invert) << 24)) -#define AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert) \ - ((left_reg) | ((right_reg) << 8) | ((shift_left) << 16) | ((shift_right) << 19) | ((mask) << 24) | ((invert) << 22)) - -/* for ease of use */ -struct ad1848_mix_elem { - const char *name; - int index; - int type; - unsigned long private_value; - const unsigned int *tlv; -}; - -#define AD1848_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_SINGLE, \ - .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert) } - -#define AD1848_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_DOUBLE, \ - .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert) } - -int snd_ad1848_add_ctl_elem(struct snd_wss *chip, - const struct ad1848_mix_elem *c); - #endif /* __SOUND_AD1848_H */ diff -urp linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-2.6.25/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-07-08 18:15:01.000000000 +0200 +++ linux-2.6.25/sound/isa/ad1848/ad1848_lib.c 2008-07-08 19:05:58.000000000 +0200 @@ -1168,86 +1168,57 @@ static int snd_ad1848_put_double(struct return change; } -/* - */ -int snd_ad1848_add_ctl_elem(struct snd_wss *chip, - const struct ad1848_mix_elem *c) -{ - static struct snd_kcontrol_new newctls[] = { - [AD1848_MIX_SINGLE] = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_single, - .get = snd_ad1848_get_single, - .put = snd_ad1848_put_single, - }, - [AD1848_MIX_DOUBLE] = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_double, - .get = snd_ad1848_get_double, - .put = snd_ad1848_put_double, - }, - [AD1848_MIX_CAPTURE] = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .info = snd_ad1848_info_mux, - .get = snd_ad1848_get_mux, - .put = snd_ad1848_put_mux, - }, - }; - struct snd_kcontrol *ctl; - int err; - - ctl = snd_ctl_new1(&newctls[c->type], chip); - if (! ctl) - return -ENOMEM; - strlcpy(ctl->id.name, c->name, sizeof(ctl->id.name)); - ctl->id.index = c->index; - ctl->private_value = c->private_value; - if (c->tlv) { - ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; - ctl->tlv.p = c->tlv; - } - if ((err = snd_ctl_add(chip->card, ctl)) < 0) - return err; - return 0; -} - -EXPORT_SYMBOL(snd_ad1848_add_ctl_elem); - static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); #define AD1848_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_SINGLE, \ - .private_value = AD1848_MIXVAL_SINGLE(reg, shift, mask, invert), \ - .tlv = xtlv } +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_ad1848_info_single, \ + .get = snd_ad1848_get_single, .put = snd_ad1848_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } #define AD1848_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert, xtlv) \ -{ .name = xname, \ - .index = xindex, \ - .type = AD1848_MIX_DOUBLE, \ - .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ - .tlv = xtlv } - -static struct ad1848_mix_elem snd_ad1848_controls[] = { -AD1848_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .name = xname, .index = xindex, \ + .info = snd_ad1848_info_double, \ + .get = snd_ad1848_get_double, .put = snd_ad1848_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + + +static struct snd_kcontrol_new snd_ad1848_controls[] = { +WSS_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, + 7, 7, 1, 1), +AD1848_DOUBLE_TLV("PCM Playback Volume", 0, + AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1, db_scale_6bit), -AD1848_DOUBLE("Aux Playback Switch", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1, +WSS_DOUBLE("Aux Playback Switch", 0, + AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +AD1848_DOUBLE_TLV("Aux Playback Volume", 0, + AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1, db_scale_5bit_12db_max), -AD1848_DOUBLE("Aux Playback Switch", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -AD1848_DOUBLE_TLV("Aux Playback Volume", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1, +WSS_DOUBLE("Aux Playback Switch", 1, + AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +AD1848_DOUBLE_TLV("Aux Playback Volume", 1, + AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1, db_scale_5bit_12db_max), -AD1848_DOUBLE_TLV("Capture Volume", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0, +AD1848_DOUBLE_TLV("Capture Volume", 0, + AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0, db_scale_rec_gain), { .name = "Capture Source", - .type = AD1848_MIX_CAPTURE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_ad1848_info_mux, + .get = snd_ad1848_get_mux, + .put = snd_ad1848_put_mux, }, -AD1848_SINGLE("Loopback Capture Switch", 0, AD1848_LOOPBACK, 0, 1, 0), +WSS_SINGLE("Loopback Capture Switch", 0, AD1848_LOOPBACK, 0, 1, 0), AD1848_SINGLE_TLV("Loopback Capture Volume", 0, AD1848_LOOPBACK, 1, 63, 0, db_scale_6bit), }; @@ -1266,9 +1237,12 @@ int snd_ad1848_mixer(struct snd_wss *chi strcpy(card->mixername, pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) - if ((err = snd_ad1848_add_ctl_elem(chip, &snd_ad1848_controls[idx])) < 0) + for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_ad1848_controls[idx], chip)); + if (err < 0) return err; + } return 0; } diff -urp linux-alsa/sound/isa/cmi8330.c linux-2.6.25/sound/isa/cmi8330.c --- linux-alsa/sound/isa/cmi8330.c 2008-07-08 18:15:01.000000000 +0200 +++ linux-2.6.25/sound/isa/cmi8330.c 2008-07-08 19:22:23.000000000 +0200 @@ -174,32 +174,48 @@ MODULE_DEVICE_TABLE(pnp_card, snd_cmi833 #endif -static struct ad1848_mix_elem snd_cmi8330_controls[] __devinitdata = { -AD1848_DOUBLE("Master Playback Volume", 0, CMI8330_MASTVOL, CMI8330_MASTVOL, 4, 0, 15, 0), -AD1848_SINGLE("Loud Playback Switch", 0, CMI8330_MUTEMUX, 6, 1, 1), -AD1848_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), -AD1848_DOUBLE("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), -AD1848_DOUBLE("Line Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, 4, 3, 1, 0), -AD1848_DOUBLE("Line Playback Volume", 0, CMI8330_LINVOL, CMI8330_LINVOL, 4, 0, 15, 0), -AD1848_DOUBLE("Line Capture Switch", 0, CMI8330_RMUX3D, CMI8330_RMUX3D, 2, 1, 1, 0), -AD1848_DOUBLE("Line Capture Volume", 0, CMI8330_LINGAIN, CMI8330_LINGAIN, 4, 0, 15, 0), -AD1848_DOUBLE("CD Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, 2, 1, 1, 0), -AD1848_DOUBLE("CD Capture Switch", 0, CMI8330_RMUX3D, CMI8330_RMUX3D, 4, 3, 1, 0), -AD1848_DOUBLE("CD Playback Volume", 0, CMI8330_CDINVOL, CMI8330_CDINVOL, 4, 0, 15, 0), -AD1848_DOUBLE("CD Capture Volume", 0, CMI8330_CDINGAIN, CMI8330_CDINGAIN, 4, 0, 15, 0), -AD1848_SINGLE("Mic Playback Switch", 0, CMI8330_MUTEMUX, 0, 1, 0), -AD1848_SINGLE("Mic Playback Volume", 0, CMI8330_OUTPUTVOL, 0, 7, 0), -AD1848_SINGLE("Mic Capture Switch", 0, CMI8330_RMUX3D, 0, 1, 0), -AD1848_SINGLE("Mic Capture Volume", 0, CMI8330_OUTPUTVOL, 5, 7, 0), -AD1848_DOUBLE("Wavetable Playback Switch", 0, CMI8330_RECMUX, CMI8330_RECMUX, 1, 0, 1, 0), -AD1848_DOUBLE("Wavetable Playback Volume", 0, CMI8330_WAVVOL, CMI8330_WAVVOL, 4, 0, 15, 0), -AD1848_DOUBLE("Wavetable Capture Switch", 0, CMI8330_RECMUX, CMI8330_RECMUX, 5, 4, 1, 0), -AD1848_DOUBLE("Wavetable Capture Volume", 0, CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0), -AD1848_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), -AD1848_SINGLE("PC Speaker Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), -AD1848_SINGLE("FM Playback Switch", 0, CMI8330_RECMUX, 3, 1, 1), -AD1848_SINGLE(SNDRV_CTL_NAME_IEC958("Input ",CAPTURE,SWITCH), 0, CMI8330_RMUX3D, 7, 1, 1), -AD1848_SINGLE(SNDRV_CTL_NAME_IEC958("Input ",PLAYBACK,SWITCH), 0, CMI8330_MUTEMUX, 7, 1, 1), +static struct snd_kcontrol_new snd_cmi8330_controls[] __devinitdata = { +WSS_DOUBLE("Master Playback Volume", 0, CMI8330_MASTVOL, CMI8330_MASTVOL, + 4, 0, 15, 0), +WSS_SINGLE("Loud Playback Switch", 0, CMI8330_MUTEMUX, 6, 1, 1), +WSS_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, + 7, 7, 1, 1), +WSS_DOUBLE("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, + 0, 0, 63, 1), +WSS_DOUBLE("Line Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, + 4, 3, 1, 0), +WSS_DOUBLE("Line Playback Volume", 0, CMI8330_LINVOL, CMI8330_LINVOL, + 4, 0, 15, 0), +WSS_DOUBLE("Line Capture Switch", 0, CMI8330_RMUX3D, CMI8330_RMUX3D, + 2, 1, 1, 0), +WSS_DOUBLE("Line Capture Volume", 0, CMI8330_LINGAIN, CMI8330_LINGAIN, + 4, 0, 15, 0), +WSS_DOUBLE("CD Playback Switch", 0, CMI8330_MUTEMUX, CMI8330_MUTEMUX, + 2, 1, 1, 0), +WSS_DOUBLE("CD Capture Switch", 0, CMI8330_RMUX3D, CMI8330_RMUX3D, 4, 3, 1, 0), +WSS_DOUBLE("CD Playback Volume", 0, CMI8330_CDINVOL, CMI8330_CDINVOL, + 4, 0, 15, 0), +WSS_DOUBLE("CD Capture Volume", 0, CMI8330_CDINGAIN, CMI8330_CDINGAIN, + 4, 0, 15, 0), +WSS_SINGLE("Mic Playback Switch", 0, CMI8330_MUTEMUX, 0, 1, 0), +WSS_SINGLE("Mic Playback Volume", 0, CMI8330_OUTPUTVOL, 0, 7, 0), +WSS_SINGLE("Mic Capture Switch", 0, CMI8330_RMUX3D, 0, 1, 0), +WSS_SINGLE("Mic Capture Volume", 0, CMI8330_OUTPUTVOL, 5, 7, 0), +WSS_DOUBLE("Wavetable Playback Switch", 0, CMI8330_RECMUX, CMI8330_RECMUX, + 1, 0, 1, 0), +WSS_DOUBLE("Wavetable Playback Volume", 0, CMI8330_WAVVOL, CMI8330_WAVVOL, + 4, 0, 15, 0), +WSS_DOUBLE("Wavetable Capture Switch", 0, CMI8330_RECMUX, CMI8330_RECMUX, + 5, 4, 1, 0), +WSS_DOUBLE("Wavetable Capture Volume", 0, CMI8330_WAVGAIN, CMI8330_WAVGAIN, + 4, 0, 15, 0), +WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), +WSS_SINGLE("PC Speaker Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), +WSS_SINGLE("FM Playback Switch", 0, CMI8330_RECMUX, 3, 1, 1), +WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0, CMI8330_RMUX3D, + 7, 1, 1), +WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0, + CMI8330_MUTEMUX, 7, 1, 1), }; #ifdef ENABLE_SB_MIXER @@ -268,7 +284,10 @@ static int __devinit snd_cmi8330_mixer(s strcpy(card->mixername, "CMI8330/C3D"); for (idx = 0; idx < ARRAY_SIZE(snd_cmi8330_controls); idx++) { - if ((err = snd_ad1848_add_ctl_elem(acard->wss, &snd_cmi8330_controls[idx])) < 0) + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cmi8330_controls[idx], + acard->wss)); + if (err < 0) return err; } diff -urp linux-alsa/sound/isa/sgalaxy.c linux-2.6.25/sound/isa/sgalaxy.c --- linux-alsa/sound/isa/sgalaxy.c 2008-07-08 18:15:01.000000000 +0200 +++ linux-2.6.25/sound/isa/sgalaxy.c 2008-07-08 19:20:27.000000000 +0200 @@ -175,9 +175,9 @@ static int __devinit snd_sgalaxy_detect( return snd_sgalaxy_setup_wss(wssport[dev], irq, dma); } -static struct ad1848_mix_elem snd_sgalaxy_controls[] = { -AD1848_DOUBLE("Aux Playback Switch", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1), -AD1848_DOUBLE("Aux Playback Volume", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) +static struct snd_kcontrol_new snd_sgalaxy_controls[] = { +WSS_DOUBLE("Aux Playback Switch", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Playback Volume", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) }; static int __devinit snd_sgalaxy_mixer(struct snd_wss *chip) @@ -210,7 +210,9 @@ static int __devinit snd_sgalaxy_mixer(s return err; /* build AUX2 input */ for (idx = 0; idx < ARRAY_SIZE(snd_sgalaxy_controls); idx++) { - if ((err = snd_ad1848_add_ctl_elem(chip, &snd_sgalaxy_controls[idx])) < 0) + err = snd_ctl_add(card, + snd_ctl_new1(&snd_sgalaxy_controls[idx], chip)); + if (err < 0) return err; } return 0; ---------------------------------------------------------------------- Zobacz cala prawde o Lukaszu Podolskim! kliknij >>> http://link.interia.pl/f1e57

1 0

[alsa-devel] ASoC: codec registers with more than 16 bits.
by Daniel Ribeiro 17 Jul '08

17 Jul '08

Hello alsa experts, I am working on a ASoC codec driver for the Motorola PCAP2 ASIC for the OpenEZX project, and i am having some trouble... PCAP2 has 25 bits registers, I access these registers via 32 bits SPI writes/reads, but soc-core/soc-dapm expects codec registers values to fit on 'unsigned short' vars. Is there any reason to this limit? I am currently faking the registers to always fit 16 bits, but this looks ugly. Would a patch to soc-core/dapm to extend this limit to 32 bits registers be accepted? Thanks! -- Daniel Ribeiro

2 1

[PATCH] ALSA: ens1370: SRC stands for Sample Rate Converter
by Rene Herman 17 Jul '08

17 Jul '08

Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ens1370.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 72d85a5..38808fb 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -522,7 +522,7 @@ static unsigned int snd_es1371_wait_src_ready(struct ensoniq * ensoniq) return r; cond_resched(); } - snd_printk(KERN_ERR "wait source ready timeout 0x%lx [0x%x]\n", + snd_printk(KERN_ERR "wait src ready timeout 0x%lx [0x%x]\n", ES_REG(ensoniq, 1371_SMPRATE), r); return 0; } -- 1.5.5 --------------080607050808050505040903--

1 0

Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Rene Herman 17 Jul '08

17 Jul '08

On 16-07-08 22:30, Sergei Steshenko wrote: > Maybe Rene meant '-R' (reverse patch) rather than '-r rejects_file'. Hey, I said I made at least one error per post -- there wouldn't be any logic in that all of a sudden not being so for the 3rd correction to another post since that, obviously, is a post in itself. I wonder what the error in this one is going to be. I should probably start putting in deliberate typos in an attempt to avoid more sarious damage. Or maybe the sleep thing will have actually worked... Rene.

1 0

Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 16 Jul '08

16 Jul '08

> From: steshenko_sergei(a)list.ru > To: b3zdomny(a)hotmail.com > CC: rene.herman(a)keyaccess.nl; tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org; alsa(a)sc8-sf-spam2.sourceforge.net > Subject: Re[2]: [Alsa-user] is this card supported by ALSA? > Date: Thu, 17 Jul 2008 00:30:15 +0400 <snip> > > Maybe Rene meant '-R' (reverse patch) rather than '-r rejects_file'. > > Regards, > Sergei. > > Spasibo, Sergei. Yes, that's what I'm thinking, too. Poka, Landis _________________________________________________________________ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_W…

1 0

[alsa-devel] Still have choppy audio using 1.0.17
by Gustavo da Silva Serra 16 Jul '08

16 Jul '08

I have been trying for many weeks to solve this problem, but with no success. After updating to Alsa 1.0.17, which include changes to aloop, I still have choppy audio. I will try to summarize what happens: - I need to use aloop, if I change to a sound card all the problems are gone; - After some time of continuous play, applications start to reproduce choppy audio. Choppy audio doesn't happens all the time; - If I repeatedly start and stop my sound capture application choppy audio occur sooner; - Start/stop my application restores the sound quality (after some tryies); - This happens with ALSA applications (xine, vlc) and OSS emulated; - I have tryied many period_size and buffer_size in asound.conf, now I am using 960 for period and 3840 for buffer. As I decrease period size, the audio quality improves, but choppy audio still happens; - I test for overrun but no errors occur. - I am using block mode, interleaved, 44100, 16 bit, 2 channels - I am using dmix, dsnoop, removing them does not correct the audio These are some of the information I have, I don't know if is is enough. If necessary I can provide any other information. What do you suggest that I do? Are there any known issues or configuration requirements? Thanks in advance.

3 17

[alsa-devel] CS5530 on Geode GX1 won't work
by georg 16 Jul '08

16 Jul '08

Hello As Alsa-User mailinglist could not help, I hopefully sent my mail to this list now. I am playing around, trying to get this sound card working for a while now. The Hardware is an embedded Geode GX1. Kernel is vanilla 2.6.25.9. Problem: ------------------------ 8<------------------------- root@tux:~# aplay foo.wav ALSA lib pcm_dmix.c:866:(snd_pcm_dmix_open) unable to open slave aplay: main:546: audio open error: Device or resource busy root@tux:~# lspci ... 00:12.2 IDE interface: Cyrix Corporation 5530 IDE [Kahlua] 00:12.3 Multimedia audio controller: Cyrix Corporation 5530 Audio [Kahlua] 00:12.4 VGA compatible controller: Cyrix Corporation 5530 Video [Kahlua] ... root@tux:~# lsmod Module Size Used by snd_cs5530 4388 0 snd_sb16_dsp 8608 1 snd_cs5530 snd_sb_common 14432 2 snd_cs5530,snd_sb16_dsp snd_pcm 63208 1 snd_sb16_dsp snd_timer 19492 1 snd_pcm snd 38804 5 snd_cs5530,snd_sb16_dsp,snd_sb_common,snd_pcm,snd_timer snd_page_alloc 7752 1 snd_pcm dmesg: ... CS5530: XpressAudio at 0x220 CS5530: MPU at 0x330 CS5530: IRQ: 5 DMA8: 1 DMA16: 5 ... root@tux:~# cat /proc/interrupts CPU0 0: 249130 XT-PIC-XT timer 1: 2 XT-PIC-XT i8042 2: 0 XT-PIC-XT cascade 5: 0 XT-PIC-XT SoundBlaster 9: 184193 XT-PIC-XT ohci_hcd:usb1, eth0 12: 1 XT-PIC-XT 14: 41917 XT-PIC-XT ide0 root@tux:~# strace aplay foo.wav ... shmctl(32768, 0x102 /* SHM_??? */, 0xbfc94a68) = 0 shmctl(32768, 0x101 /* SHM_??? */, 0xbfc94a68) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbfc947a8) = 0 ioctl(3, 0x40045532, 0xbfc947a4) = 0 open("/dev/snd/pcmC0D0p", O_RDWR|O_NONBLOCK) = -1 EBUSY (Device or resource busy) close(3) = 0 write(2, "ALSA lib pcm_dmix.c:866:(snd_pcm"..., 44ALSA lib pcm_dmix.c:866: (snd_pcm_dmix_open) ) = 44 write(2, "unable to open slave", 20unable to open slave) = 20 write(2, "\n", 1) = 1 ... root@tux:~# lsof | grep -i pcmC0D0p root@tux:~# root@tux:~# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Audio [CS5530 Audio], device 0: SB16 DSP [DSP v4.12] Subdevices: 1/1 Subdevice #0: subdevice #0 root@tux:~# aplay -L null Discard all samples (playback) or generate zero samples (capture) ------------------------ 8<------------------------- No one is using the sound device. But strace shows error 'Device or resource busy'. The thing that confuses me is the output of 'aplay -L'... Any help is appreciated :-)... Bye Georg

3 3

Re: [alsa-devel] softvol chokes on snd_pcm_rewind
by Stewart Adam 16 Jul '08

16 Jul '08

Hi, I'm using Fedora 10 development/rawhide and I'm having the same problem with softvol and alsa-lib-1.0.16 - snd_pcm_rewind() makes the audio distorted and very high-pitched. Is there a way I could help debug this? Regards, Stewart

1 0

Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 16 Jul '08

16 Jul '08

> From: steshenko_sergei(a)list.ru > To: rene.herman(a)keyaccess.nl > CC: b3zdomny(a)hotmail.com; tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org; ALSA(a)sc8-sf-spam2.sourceforge.net > Subject: Re[2]: [Alsa-user] is this card supported by ALSA? > Date: Wed, 16 Jul 2008 19:12:26 +0400 > > > > -----Original Message----- > From: Rene Herman <rene.herman(a)keyaccess.nl> > To: Landis McGauhey <b3zdomny(a)hotmail.com> > Date: Wed, 16 Jul 2008 16:44:11 +0200 > Subject: Re: [Alsa-user] is this card supported by ALSA? > > > > > On 16-07-08 16:40, Rene Herman wrote: > > > > > # moodprobe -r ens1371-ac97_reset_hack.diff > > > # moodprobe -r ens1371-ac97.diff > > > > I'm getting really sick of the fact that I seem to need to make at least > > one typo or other small error per post. "modprobe" ofcourse. > > > > Rene. > > > > > > Actually, 'moodprobe' is a really nice word :-). I love it ! > > Probably Linux distros should symlink ln -s modprobe moodprobe . > > --Sergei. LOL! Spasibo, Sergei. In that case, my psychiatrist will diagnose me as having a moodprobe disorder rather than a mood disorder! ;) _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

1 0

Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 16 Jul '08

16 Jul '08

> Date: Wed, 16 Jul 2008 11:16:32 +0200 > From: rene.herman(a)keyaccess.nl > To: b3zdomny(a)hotmail.com > CC: tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org > Subject: Re: [Alsa-user] is this card supported by ALSA? <snip> OK, here we go: # patch -p1 --dry-run < /home/bezdomny/Desktop/ens1371-ac97_reset_hack.diff= no errors # patch -p1 < /home/bezdomny/Desktop/ens1371-ac97_reset_hack.diff= no errors # make= no errors # make modules_install= no errors # modprobe -r snd-ens1371= no errors # modprobe -r es1371= no errors # modprobe snd-ens1371= no errors # alsamixer= usb mic in webcam #alsamixer -c 1= PCI card. Checked for good levels and no mutes-- all is well. # speaker-test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to open slave silence # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav= Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo but with silence running dmesg and attaching the result alsaconf= Unloading ALSA sound driver modules: snd-ens1371 snd-ac97-codec snd-usb-audio snd-pcm snd-timer snd-page-alloc snd-usb-lib snd-rawmidi snd-seq-device snd-hwdep. building card database... following cards are found on your system: ens1371 Creative Labs Ective 1938 legacy Probe legacy ISA (non-PnP) chips (OK) Configuring snd-ens1371 │ │ Do you want to modify /etc/modprobe.d/sound │ │ (and /etc/modprobe.conf if present)? (yes) "pop" in speaker as alsaconf loads snd-ens1371 running alsamixer-- no mute toggle underneath mic, that's a first and just so you know, there's no mute toggle under "master", "master-m", and "PCM", either, but that's nominal. There was one time, however, that "master" and/or "PCM" had a mute toggle. Perhaps "master" and/or "PCM" are muted and we can't tell because there are supposed to be mute toggles there, but they are missing? Something to consider... #speaker -test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 2048 to 16384 Period size range from 1024 to 1024 Using max buffer size 16384 Periods = 4 was set period_size = 1024 was set buffer_size = 16384 0 - Front Left Time per period = 2.653267 but with silence # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav=Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo but with silence. tried to run audacity, but it doesn't appear. go to process table-- there's audacity, killed it. ran xmms, changed preferences to ALSA, it runs, you can see the volume bars pumping, but it's silent. and that's about it. Not going to reboot. Rene & Takashi: I fear this is taking up too much of our time and this effort may have long passed the point of diminishing returns. Personally, my goal is to have audio with Skype, whether with OSS or ALSA, I don't care. If you want to continue to work on this for the sake of the community at large and others who have this same card and this same problem, then I'm willing and would be honored to be the guinea pig and be an active participant in the community. I'm sorry this problem is so stubborn and if you have better things to do, I understand. If you don't care to work on this anymore, I would understand and if you could perhaps just help me get sound in Skype (that is, to have /dev/dsp appear as I/O options in Skype configuration) with OSS, then I would be a happy camper indeed. thank you as always for your persistence, expertise, and patience, Landis _________________________________________________________________ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_W…

2 4

Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 16 Jul '08

16 Jul '08

> Date: Tue, 15 Jul 2008 19:02:46 +0200 > From: rene.herman(a)keyaccess.nl > To: b3zdomny(a)hotmail.com > CC: tiwai(a)suse.de; alsa-user(a)lists.sourceforge.net; alsa-devel(a)alsa-project.org > Subject: Re: [Alsa-user] is this card supported by ALSA? > > On 15-07-08 16:42, Rene Herman wrote: <snip> > Just in case... I did _say_ 'without the --dry-run' but then neglected > to actually delete it from the second line here. You _did_ delete it, right? > > > Off, > Rene. Yes, I did delete '-dry-run'. In fact, just to be doubly certain, I just re-ran the whole process and additionally ran alsaconf (and again there was a loud 'click' in the speakers when alsaconf loaded the driver): # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0= 0-0/0: 0x83848384 STS PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Capabilities : -dedicated MIC PCM IN channel- -bass & treble- DAC resolution : 16-bit ADC resolution : 20-bit 3D enhancement : Reserved 29 Current setup Mic gain : +0dB [+20dB] POP path : post 3D Sim. stereo : off 3D enhancement : off Loudness : on Mono output : Mic Mic select : Mic1 ADC/DAC loopback : off # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0+regs= 0:00 = 0000 0:02 = 0140 0:04 = 8909 0:06 = 8000 0:08 = 8009 0:0a = ffff 0:0c = 150b 0:0e = 805f 0:10 = 805f 0:12 = 9515 0:14 = 9212 0:16 = 8b0b 0:18 = 9212 0:1a = 9010 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 0000 0:28 = 000f 0:2a = ffff 0:2c = ffff 0:2e = ffff 0:30 = ffff 0:32 = ffff 0:34 = ffff 0:36 = ffff 0:38 = ffff 0:3a = ffff 0:3c = ffff 0:3e = ffff 0:40 = ffff 0:42 = ffff 0:44 = ffff 0:46 = ffff 0:48 = ffff 0:4a = ffff 0:4c = ffff 0:4e = ffff 0:50 = ffff 0:52 = ffff 0:54 = ffff 0:56 = ffff 0:58 = ffff 0:5a = ffff 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = ffff 0:68 = ffff 0:6a = ffff 0:6c = ffff 0:6e = ffff 0:70 = ffff 0:72 = ffff 0:74 = ffff 0:76 = ffff 0:78 = ffff 0:7a = ffff 0:7c = ffff 0:7e = 8384 # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav= Playing WAVE '/usr/share/sounds/startup3.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo (with silence) # speaker-test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise (with silence) xmms plays, you can see the level bars pumping, but with silence. checking /etc/modprobe.d /blacklist: es1371 is blacklisted. rebooting. # speaker-test= speaker-test 1.0.16 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to open slave Playback open error: -2,No such file or directory # aplay -D hw:AudioPCI /usr/share/sounds/startup3.wav= ALSA lib pcm_hw.c:1351:(_snd_pcm_hw_open) Invalid value for card aplay: main:564: audio open error: No such device /var/log/boot & dmesg (from after reboot) attached. Best regards, Landis cheers, Landis _________________________________________________________________ Making the world a better place one message at a time. http://www.imtalkathon.com/?source=EML_WLH_Talkathon_BetterPlace

3 4

[alsa-devel] 1.0.16->1.0.17 regression for some HDA NVidia's
by Ozan Çağlayan 16 Jul '08

16 Jul '08

Everything seems OK with those onboard audio chipsets but there's absolutely no sound. We have 2 bug reports on our bug tracking system indicating this regression for MCP51, MCP61 and MCP73. Here are the URL's for alsa-info.sh output's: MCP51: http://pastebin.ca/1062418 MCP61: http://pastebin.ca/1058482 MCP73: http://pastebin.ca/1058319 Those users have recently reported that downgrading to 1.0.16 series solves the problem. I have an ASUS mainboard which contains the MCP51 one, I can help if you want me to rebuild the driver with a patch, or anything else. Thanks. -- Ozan Caglayan <ozan_at_pardus.org.tr> http://www.pardus.org.tr/eng

3 14

[alsa-devel] Is params_format() valid in the _hw_params() function?
by Timur Tabi 16 Jul '08

16 Jul '08

My ASoC driver programs the sample size in the _prepare() callback function, because I was under the impression that the sample size cannot be known in the _hw_params() callback function. However, I just ran a test, and it appears that the params_format() macro does return the right value. Once I know the format, I can determine the same size. Is this always true, even in OSS emulation? I know very little about ALSA and OSS application programming, so I don't know what causes _hw_params() to be called vs. _prepare(). If it turns out that params_format() does work in _hw_params(), I can greatly simplify my driver. -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] SB0880 X-Fi Titanium Fatal1ty Pro (with codecgraph)
by Tony "Chainsaw" Vroon 15 Jul '08

15 Jul '08

Good evening, Just wanted to share that this card seems to be HD-Audio compatible (by virtue of its UAA compatibility claim); http://uk.europe.creative.com/products/product.asp?category=1&subcategory=2… For a while now, I've been trying to write a skeleton patch_creative.c to hook up some basic playback functions. As this has failed, I will instead post all information I can gather about the card, in the hope that such a basic patch_creative.c can be written that I can then slowly extend. I was unable to set up proper PCM paths (and thus listen to any audio at all), but did manage to get the card detected with a simple addition to hda_intel.c: + /* Creative Labs UAA-compatible */ + { PCI_DEVICE(0x1102, 0x000b), .driver_data = AZX_DRIVER_CREATIVE }, A guess at correct mixer controls would be: HDA_CODEC_VOLUME("PCM Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Surround Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x04, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x04, 2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x04, 2, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x05, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Side Playback Switch", 0x05, 0x0, HDA_OUTPUT), Please find attached the codec details and the resulting codec graph. It is a PCI-Express card, lspci information follows: 83:00.0 Audio device [0403]: Creative Labs Device [1102:000b] (rev 03) Subsystem: Creative Labs Device [1102:0043] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 505 Region 0: Memory at d0100000 (64-bit, non-prefetchable) [size=64K] Region 2: Memory at d0200000 (64-bit, non-prefetchable) [size=2M] Region 4: Memory at d4000000 (64-bit, non-prefetchable) [size=64M] Capabilities: [40] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [48] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Address: 00000000feeff00c Data: 41c9 Capabilities: [58] Express (v2) Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr+ UncorrErr- FatalErr- UnsuppReq+ AuxPwr- TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Device Serial Number ff-ff-ff-ff-ff-ff-ff-ff Capabilities: [300] Advanced Error Reporting <?> Kernel driver in use: HDA Intel Regards, Tony V.

4 9

Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Rene Herman 15 Jul '08

15 Jul '08

On 15-07-08 01:36, Landis McGauhey wrote: It seems there's just a bit too much oddness going on. Takashi, you know more about ac97. Also bringing in alsa-devel... > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0= > > 0-0/0: 0x76058384 F�S Eep? A 0x83847605 would be a SigmaTel STAC9704. And: [ ... ] > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0+regs= [ ... ] > 0:7c = 0000 > 0:7e = 8384 does't fit the above ID. Do we just have a crummy codec that needs delay between acceses somewhere or something? Here full contents again: > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0= > > 0-0/0: 0x76058384 F�S > > PCI Subsys Vendor: 0x0000 > PCI Subsys Device: 0x0000 > > Capabilities : -dedicated MIC PCM IN channel- -bass & treble- > DAC resolution : 16-bit > ADC resolution : 20-bit > 3D enhancement : Reserved 29 > > Current setup > Mic gain : +0dB [+20dB] > POP path : post 3D > Sim. stereo : off > 3D enhancement : off > Loudness : on > Mono output : MIX > Mic select : Mic2 > ADC/DAC loopback : off > > # cat /proc/asound/AudioPCI/codec97#0/ac97#0-0+regs= > > 0:00 = 0000 > 0:02 = 0140 > 0:04 = 8909 > 0:06 = 8000 > 0:08 = 8009 > 0:0a = ffff > 0:0c = 150b > 0:0e = 801f > 0:10 = 805f > 0:12 = 9515 > 0:14 = 9212 > 0:16 = 9f1f > 0:18 = 9f1f > 0:1a = 9010 > 0:1c = 0000 > 0:1e = 0000 > 0:20 = 0000 > 0:22 = 0000 > 0:24 = 0000 > 0:26 = 0000 > 0:28 = 000f > 0:2a = ffff > 0:2c = ffff > 0:2e = ffff > 0:30 = ffff > 0:32 = ffff > 0:34 = ffff > 0:36 = ffff > 0:38 = ffff > 0:3a = ffff > 0:3c = ffff > 0:3e = ffff > 0:40 = ffff > 0:42 = ffff > 0:44 = ffff > 0:46 = ffff > 0:48 = ffff > 0:4a = ffff > 0:4c = ffff > 0:4e = ffff > 0:50 = ffff > 0:52 = ffff > 0:54 = ffff > 0:56 = ffff > 0:58 = ffff > 0:5a = 0000 > 0:5c = 0000 > 0:5e = 0000 > 0:60 = 0000 > 0:62 = 0000 > 0:64 = 0000 > 0:66 = 0000 > 0:68 = 0000 > 0:6a = 0000 > 0:6c = 0000 > 0:6e = 0000 > 0:70 = 0000 > 0:72 = 0000 > 0:74 = 0000 > 0:76 = 0000 > 0:78 = 0000 > 0:7a = 0000 > 0:7c = 0000 > 0:7e = 8384 Rene.

3 15

Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Rene Herman 15 Jul '08

15 Jul '08

On 15-07-08 17:01, Sergei Steshenko wrote: > And that's why something equivalent to 'ndiswrapper' for audio is > necessary. Please note -- if you're _trying_ to be a troll, it's too obvious to have the fun last longer than this message. Rene.

1 0

[alsa-devel] [PATCH] ALSA: re-order AC97 codec ID table.
by Rene Herman 15 Jul '08

15 Jul '08

Hi Takashi. Feel free to drop if considered pointless churn but keeping it ordered makes checking easier. Rene. >From 389e02fd488d8ffdaabd6cc10cf788792258916d Mon Sep 17 00:00:00 2001 From: Rene Herman <rene.herman(a)gmail.com> Date: Tue, 15 Jul 2008 03:13:01 +0200 Subject: [PATCH] ALSA: re-order AC97 codec ID table. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ac97/ac97_codec.c | 34 +++++++++++++++++----------------- 1 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 8c49a00..f1e3a47 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -67,8 +67,8 @@ struct ac97_codec_id { }; static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { -{ 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL }, { 0x41445300, 0xffffff00, "Analog Devices", NULL, NULL }, +{ 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL }, { 0x414c4300, 0xffffff00, "Realtek", NULL, NULL }, { 0x414c4700, 0xffffff00, "Realtek", NULL, NULL }, { 0x434d4900, 0xffffff00, "C-Media Electronics", NULL, NULL }, @@ -94,11 +94,6 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { }; static const struct ac97_codec_id snd_ac97_codec_ids[] = { -{ 0x414b4d00, 0xffffffff, "AK4540", NULL, NULL }, -{ 0x414b4d01, 0xffffffff, "AK4542", NULL, NULL }, -{ 0x414b4d02, 0xffffffff, "AK4543", NULL, NULL }, -{ 0x414b4d06, 0xffffffff, "AK4544A", NULL, NULL }, -{ 0x414b4d07, 0xffffffff, "AK4545", NULL, NULL }, { 0x41445303, 0xffffffff, "AD1819", patch_ad1819, NULL }, { 0x41445340, 0xffffffff, "AD1881", patch_ad1881, NULL }, { 0x41445348, 0xffffffff, "AD1881A", patch_ad1881, NULL }, @@ -112,20 +107,25 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x41445374, 0xffffffff, "AD1981B", patch_ad1981b, NULL }, { 0x41445375, 0xffffffff, "AD1985", patch_ad1985, NULL }, { 0x41445378, 0xffffffff, "AD1986", patch_ad1986, NULL }, +{ 0x414b4d00, 0xffffffff, "AK4540", NULL, NULL }, +{ 0x414b4d01, 0xffffffff, "AK4542", NULL, NULL }, +{ 0x414b4d02, 0xffffffff, "AK4543", NULL, NULL }, +{ 0x414b4d06, 0xffffffff, "AK4544A", NULL, NULL }, +{ 0x414b4d07, 0xffffffff, "AK4545", NULL, NULL }, { 0x414c4300, 0xffffff00, "ALC100,100P", NULL, NULL }, { 0x414c4710, 0xfffffff0, "ALC200,200P", NULL, NULL }, +{ 0x414c4720, 0xfffffff0, "ALC650", patch_alc650, NULL }, { 0x414c4721, 0xffffffff, "ALC650D", NULL, NULL }, /* already patched */ { 0x414c4722, 0xffffffff, "ALC650E", NULL, NULL }, /* already patched */ { 0x414c4723, 0xffffffff, "ALC650F", NULL, NULL }, /* already patched */ -{ 0x414c4720, 0xfffffff0, "ALC650", patch_alc650, NULL }, -{ 0x414c4760, 0xfffffff0, "ALC655", patch_alc655, NULL }, -{ 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */ -{ 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL }, -{ 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL }, { 0x414c4730, 0xffffffff, "ALC101", NULL, NULL }, { 0x414c4740, 0xfffffff0, "ALC202", NULL, NULL }, { 0x414c4750, 0xfffffff0, "ALC250", NULL, NULL }, +{ 0x414c4760, 0xfffffff0, "ALC655", patch_alc655, NULL }, { 0x414c4770, 0xfffffff0, "ALC203", NULL, NULL }, +{ 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL }, +{ 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */ +{ 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL }, { 0x434d4941, 0xffffffff, "CMI9738", patch_cm9738, NULL }, { 0x434d4961, 0xffffffff, "CMI9739", patch_cm9739, NULL }, { 0x434d4969, 0xffffffff, "CMI9780", patch_cm9780, NULL }, @@ -143,7 +143,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x43525970, 0xfffffff8, "CS4202", NULL, NULL }, { 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, // SmartMC II { 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, // SmartAMC fixme: the mask might be different -{ 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, +{ 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, { 0x44543031, 0xfffffff0, "DT0398", NULL, NULL }, { 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, // same as TR28028? { 0x45838308, 0xffffffff, "ESS1988", NULL, NULL }, @@ -171,11 +171,11 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, -{ 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, -{ 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, -{ 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, -{ 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, -{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, +{ 0x574d4c03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, +{ 0x574d4c04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, +{ 0x574d4c05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, +{ 0x574d4c09, 0xffffffff, "WM9709", NULL, NULL}, +{ 0x574d4c12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, { 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, -- 1.5.5

2 3

[alsa-devel] [PATCH] ALSA: add TriTech 28023 AC97 codec ID and Wolfson 9701 name.
by Rene Herman 15 Jul '08

15 Jul '08

Hi Takashi Have the TriTech 28023 on one of my Ensoniq ENS1371 cards and on a ESS Maestro-2 card. Have a Wolfson XWM9701 on a Trident 4DWAVE-DX one. Rene. >From 8c57754f2974e30b16e5c90181f478ee75190c64 Mon Sep 17 00:00:00 2001 From: Rene Herman <rene.herman(a)gmail.com> Date: Tue, 15 Jul 2008 03:00:21 +0200 Subject: [PATCH] ALSA: add TriTech 28023 AC97 codec ID and Wolfson 9701 name. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ac97/ac97_codec.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 07364c0..8c49a00 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -161,6 +161,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, +{ 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] @@ -169,7 +170,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF { 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, -{ 0x574d4c00, 0xffffffff, "WM9701A", NULL, NULL }, +{ 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, { 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, { 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, -- 1.5.5

2 1

[alsa-devel] [ASoC] ASoC on a AT91RM9200 with a tlv320aic23b codec
by Simon Tremblay 15 Jul '08

15 Jul '08

Hello, Basically I want to use alsa on my embedded system running with a AT91RM9200. I have few questions about ASoC development. I think there is a driver for tlv320 (looks like a generic driver for tlv320xxx) in dev branch. Anybody knows if this driver works with tlv320aic23b? I saw this driver isn't tagged in any kernel version (currently I use 2.6.24 and tlv320.c and tlv320.h aren't in the codecs directory). Is it because the driver isn't tested yet? Thank you, Simon

2 1

[PATCH] ALSA: re-order AC97 codec ID table.
by Rene Herman 15 Jul '08

15 Jul '08

Signed-off-by: Rene Herman <rene.herman(a)gmail.com> --- sound/pci/ac97/ac97_codec.c | 20 ++++++++++---------- 1 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 8c49a00..f6a7d72 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -67,8 +67,8 @@ struct ac97_codec_id { }; static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { -{ 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL }, { 0x41445300, 0xffffff00, "Analog Devices", NULL, NULL }, +{ 0x414b4d00, 0xffffff00, "Asahi Kasei", NULL, NULL }, { 0x414c4300, 0xffffff00, "Realtek", NULL, NULL }, { 0x414c4700, 0xffffff00, "Realtek", NULL, NULL }, { 0x434d4900, 0xffffff00, "C-Media Electronics", NULL, NULL }, @@ -94,11 +94,6 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { }; static const struct ac97_codec_id snd_ac97_codec_ids[] = { -{ 0x414b4d00, 0xffffffff, "AK4540", NULL, NULL }, -{ 0x414b4d01, 0xffffffff, "AK4542", NULL, NULL }, -{ 0x414b4d02, 0xffffffff, "AK4543", NULL, NULL }, -{ 0x414b4d06, 0xffffffff, "AK4544A", NULL, NULL }, -{ 0x414b4d07, 0xffffffff, "AK4545", NULL, NULL }, { 0x41445303, 0xffffffff, "AD1819", patch_ad1819, NULL }, { 0x41445340, 0xffffffff, "AD1881", patch_ad1881, NULL }, { 0x41445348, 0xffffffff, "AD1881A", patch_ad1881, NULL }, @@ -112,20 +107,25 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x41445374, 0xffffffff, "AD1981B", patch_ad1981b, NULL }, { 0x41445375, 0xffffffff, "AD1985", patch_ad1985, NULL }, { 0x41445378, 0xffffffff, "AD1986", patch_ad1986, NULL }, +{ 0x414b4d00, 0xffffffff, "AK4540", NULL, NULL }, +{ 0x414b4d01, 0xffffffff, "AK4542", NULL, NULL }, +{ 0x414b4d02, 0xffffffff, "AK4543", NULL, NULL }, +{ 0x414b4d06, 0xffffffff, "AK4544A", NULL, NULL }, +{ 0x414b4d07, 0xffffffff, "AK4545", NULL, NULL }, { 0x414c4300, 0xffffff00, "ALC100,100P", NULL, NULL }, { 0x414c4710, 0xfffffff0, "ALC200,200P", NULL, NULL }, { 0x414c4721, 0xffffffff, "ALC650D", NULL, NULL }, /* already patched */ { 0x414c4722, 0xffffffff, "ALC650E", NULL, NULL }, /* already patched */ { 0x414c4723, 0xffffffff, "ALC650F", NULL, NULL }, /* already patched */ { 0x414c4720, 0xfffffff0, "ALC650", patch_alc650, NULL }, -{ 0x414c4760, 0xfffffff0, "ALC655", patch_alc655, NULL }, -{ 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */ -{ 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL }, -{ 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL }, { 0x414c4730, 0xffffffff, "ALC101", NULL, NULL }, { 0x414c4740, 0xfffffff0, "ALC202", NULL, NULL }, { 0x414c4750, 0xfffffff0, "ALC250", NULL, NULL }, +{ 0x414c4760, 0xfffffff0, "ALC655", patch_alc655, NULL }, { 0x414c4770, 0xfffffff0, "ALC203", NULL, NULL }, +{ 0x414c4781, 0xffffffff, "ALC658D", NULL, NULL }, /* already patched */ +{ 0x414c4780, 0xfffffff0, "ALC658", patch_alc655, NULL }, +{ 0x414c4790, 0xfffffff0, "ALC850", patch_alc850, NULL }, { 0x434d4941, 0xffffffff, "CMI9738", patch_cm9738, NULL }, { 0x434d4961, 0xffffffff, "CMI9739", patch_cm9739, NULL }, { 0x434d4969, 0xffffffff, "CMI9780", patch_cm9780, NULL }, -- 1.5.5 --------------000006000004070200060601--

1 0

[alsa-devel] ALSA packages version 1.0.17 final released
by Jaroslav Kysela 14 Jul '08

14 Jul '08

Hi all, all ALSA packages version 1.0.17 were released and they are available for download. Changes are described here: http://www.alsa-project.org/main/index.php/Changes_v1.0.17rc3_v1.0.17 http://www.alsa-project.org/main/index.php/Changes_v1.0.16_v1.0.17 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

Re: [alsa-devel] RME madiface
by Florian Faber 13 Jul '08

13 Jul '08

Dee & alsa community, > I have no reason to believe the MADI bus is different. However, I > suspect the stream are different. Problem solved. The MADIface uses a different internal clock than the HDSP MADI, resulting in a wrong DDS calculation. The driver with the fix will be released together with the RayDAT support, before end of july. Flo -- Machines can do the work, so people have time to think. public key 6C002249 x-hkp://wwwkeys.eu.pgp.net

1 0

[alsa-devel] [PATCH] ALSA driver for SGI HAL2 audio device
by Thomas Bogendoerfer 12 Jul '08

12 Jul '08

This patch adds a new ALSA driver for the audio device found inside many older SGI workstation (Indy, Indigo2). The hardware uses a SGI custom chip, which feeds two codec chips, an IEC chip and a synth chip. Currently only one of the codecs is supported. This driver already has the same functionality as the HAL2 OSS driver and will replace it. Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> --- Please apply for 2.6.27. sound/mips/Kconfig | 7 + sound/mips/Makefile | 2 + sound/mips/hal2.c | 947 +++++++++++++++++++++++++++++++++++++++++++++++++++ sound/mips/hal2.h | 245 +++++++++++++ 4 files changed, 1201 insertions(+), 0 deletions(-) diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 531f8ba..39e371d 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -11,5 +11,12 @@ config SND_AU1X00 help ALSA Sound driver for the Au1x00's AC97 port. +config SND_SGI_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 + help + Sound support for the SGI Indy and Indigo2 Workstation. + + endmenu diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 47afed9..63f4a9c 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -3,6 +3,8 @@ # snd-au1x00-objs := au1x00.o +snd-sgi-hal2-objs := hal2.o # Toplevel Module Dependency obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o +obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c new file mode 100644 index 0000000..db495be --- /dev/null +++ b/sound/mips/hal2.c @@ -0,0 +1,947 @@ +/* + * Driver for A2 audio system used in SGI machines + * Copyright (c) 2008 Thomas Bogendoerfer <tsbogend(a)alpha.fanken.de> + * + * Based on OSS code from Ladislav Michl <ladis(a)linux-mips.org>, which + * was based on code from Ulf Carlsson + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <asm/sgi/hpc3.h> +#include <asm/sgi/ip22.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm-indirect.h> +#include <sound/initval.h> + +#include "hal2.h" + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI HAL2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI HAL2 soundcard."); +MODULE_DESCRIPTION("ALSA driver for SGI HAL2 audio"); +MODULE_AUTHOR("Thomas Bogendoerfer"); +MODULE_LICENSE("GPL"); + + +#define H2_BLOCK_SIZE 1024 +#define H2_BUF_SIZE 16384 + +struct hal2_pbus { + struct hpc3_pbus_dmacregs *pbus; + int pbusnr; + unsigned int ctrl; /* Current state of pbus->pbdma_ctrl */ +}; + +struct hal2_desc { + struct hpc_dma_desc desc; + u32 pad; /* padding */ +}; + +struct hal2_codec { + struct snd_pcm_indirect pcm_indirect; + struct snd_pcm_substream *substream; + + unsigned char *buffer; + dma_addr_t buffer_dma; + struct hal2_desc *desc; + dma_addr_t desc_dma; + int desc_count; + struct hal2_pbus pbus; + int voices; /* mono/stereo */ + unsigned int sample_rate; + unsigned int master; /* Master frequency */ + unsigned short mod; /* MOD value */ + unsigned short inc; /* INC value */ +}; + +#define H2_MIX_OUTPUT_ATT 0 +#define H2_MIX_INPUT_GAIN 1 + +struct snd_hal2 { + struct snd_card *card; + + struct hal2_ctl_regs *ctl_regs; /* HAL2 ctl registers */ + struct hal2_aes_regs *aes_regs; /* HAL2 aes registers */ + struct hal2_vol_regs *vol_regs; /* HAL2 vol registers */ + struct hal2_syn_regs *syn_regs; /* HAL2 syn registers */ + + struct hal2_codec dac; + struct hal2_codec adc; +}; + +#define H2_INDIRECT_WAIT(regs) while (hal2_read(&regs->isr) & H2_ISR_TSTATUS); + +#define H2_READ_ADDR(addr) (addr | (1<<7)) +#define H2_WRITE_ADDR(addr) (addr) + +static inline u32 hal2_read(u32 *reg) +{ + return __raw_readl(reg); +} + +static inline void hal2_write(u32 val, u32 *reg) +{ + __raw_writel(val, reg); +} + + +static u32 hal2_i_read32(struct snd_hal2 *hal2, u16 addr) +{ + u32 ret; + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), &regs->iar); + H2_INDIRECT_WAIT(regs); + ret = hal2_read(&regs->idr0) & 0xffff; + hal2_write(H2_READ_ADDR(addr) | 0x1, &regs->iar); + H2_INDIRECT_WAIT(regs); + ret |= (hal2_read(&regs->idr0) & 0xffff) << 16; + return ret; +} + +static void hal2_i_write16(struct snd_hal2 *hal2, u16 addr, u16 val) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(val, &regs->idr0); + hal2_write(0, &regs->idr1); + hal2_write(0, &regs->idr2); + hal2_write(0, &regs->idr3); + hal2_write(H2_WRITE_ADDR(addr), &regs->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_write32(struct snd_hal2 *hal2, u16 addr, u32 val) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(val & 0xffff, &regs->idr0); + hal2_write(val >> 16, &regs->idr1); + hal2_write(0, &regs->idr2); + hal2_write(0, &regs->idr3); + hal2_write(H2_WRITE_ADDR(addr), &regs->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_setbit16(struct snd_hal2 *hal2, u16 addr, u16 bit) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), &regs->iar); + H2_INDIRECT_WAIT(regs); + hal2_write((hal2_read(&regs->idr0) & 0xffff) | bit, &regs->idr0); + hal2_write(0, &regs->idr1); + hal2_write(0, &regs->idr2); + hal2_write(0, &regs->idr3); + hal2_write(H2_WRITE_ADDR(addr), &regs->iar); + H2_INDIRECT_WAIT(regs); +} + +static void hal2_i_clearbit16(struct snd_hal2 *hal2, u16 addr, u16 bit) +{ + struct hal2_ctl_regs *regs = hal2->ctl_regs; + + hal2_write(H2_READ_ADDR(addr), &regs->iar); + H2_INDIRECT_WAIT(regs); + hal2_write((hal2_read(&regs->idr0) & 0xffff) & ~bit, &regs->idr0); + hal2_write(0, &regs->idr1); + hal2_write(0, &regs->idr2); + hal2_write(0, &regs->idr3); + hal2_write(H2_WRITE_ADDR(addr), &regs->iar); + H2_INDIRECT_WAIT(regs); +} + +static int hal2_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + uinfo->value.integer.max = 31; + break; + case H2_MIX_INPUT_GAIN: + uinfo->value.integer.max = 15; + break; + } + return 0; +} + +static int hal2_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol); + u32 tmp; + int l, r; + + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + tmp = hal2_i_read32(hal2, H2I_DAC_C2); + if (tmp & H2I_C2_MUTE) { + l = 0; + r = 0; + } else { + l = 31 - ((tmp >> H2I_C2_L_ATT_SHIFT) & 31); + r = 31 - ((tmp >> H2I_C2_R_ATT_SHIFT) & 31); + } + break; + case H2_MIX_INPUT_GAIN: + tmp = hal2_i_read32(hal2, H2I_ADC_C2); + l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15; + r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15; + break; + } + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +static int hal2_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol); + u32 old, new; + int l, r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + switch ((int)kcontrol->private_value) { + case H2_MIX_OUTPUT_ATT: + old = hal2_i_read32(hal2, H2I_DAC_C2); + new = old & ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); + if (l | r) { + l = 31 - l; + r = 31 - r; + new |= (l << H2I_C2_L_ATT_SHIFT); + new |= (r << H2I_C2_R_ATT_SHIFT); + } else + new |= H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE; + hal2_i_write32(hal2, H2I_DAC_C2, new); + break; + case H2_MIX_INPUT_GAIN: + old = hal2_i_read32(hal2, H2I_ADC_C2); + new = old & ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M); + new |= (l << H2I_C2_L_GAIN_SHIFT); + new |= (r << H2I_C2_R_GAIN_SHIFT); + hal2_i_write32(hal2, H2I_ADC_C2, new); + break; + } + return old != new; +} + +static struct snd_kcontrol_new hal2_ctrl_headphone __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = H2_MIX_OUTPUT_ATT, + .info = hal2_gain_info, + .get = hal2_gain_get, + .put = hal2_gain_put, +}; + +static struct snd_kcontrol_new hal2_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = H2_MIX_INPUT_GAIN, + .info = hal2_gain_info, + .get = hal2_gain_get, + .put = hal2_gain_put, +}; + +static int __devinit hal2_mixer_create(struct snd_hal2 *hal2) +{ + int err; + + /* mute DAC */ + hal2_i_write32(hal2, H2I_DAC_C2, + H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE); + /* mute ADC */ + hal2_i_write32(hal2, H2I_ADC_C2, 0); + + err = snd_ctl_add(hal2->card, + snd_ctl_new1(&hal2_ctrl_headphone, hal2)); + if (err < 0) + return err; + + err = snd_ctl_add(hal2->card, + snd_ctl_new1(&hal2_ctrl_mic, hal2)); + if (err < 0) + return err; + + return 0; +} + +static irqreturn_t hal2_interrupt(int irq, void *dev_id) +{ + struct snd_hal2 *hal2 = dev_id; + irqreturn_t ret = IRQ_NONE; + + /* decide what caused this interrupt */ + if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { + snd_pcm_period_elapsed(hal2->dac.substream); + ret = IRQ_HANDLED; + } + if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) { + snd_pcm_period_elapsed(hal2->adc.substream); + ret = IRQ_HANDLED; + } + return ret; +} + +static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate) +{ + unsigned short mod; + + if (44100 % rate < 48000 % rate) { + mod = 4 * 44100 / rate; + codec->master = 44100; + } else { + mod = 4 * 48000 / rate; + codec->master = 48000; + } + + codec->inc = 4; + codec->mod = mod; + rate = 4 * codec->master / mod; + + return rate; +} + +static void hal2_set_dac_rate(struct snd_hal2 *hal2) +{ + unsigned int master = hal2->dac.master; + int inc = hal2->dac.inc; + int mod = hal2->dac.mod; + + hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0); + hal2_i_write32(hal2, H2I_BRES1_C2, + ((0xffff & (inc - mod - 1)) << 16) | inc); +} + +static void hal2_set_adc_rate(struct snd_hal2 *hal2) +{ + unsigned int master = hal2->adc.master; + int inc = hal2->adc.inc; + int mod = hal2->adc.mod; + + hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0); + hal2_i_write32(hal2, H2I_BRES2_C2, + ((0xffff & (inc - mod - 1)) << 16) | inc); +} + +static void hal2_setup_dac(struct snd_hal2 *hal2) +{ + unsigned int fifobeg, fifoend, highwater, sample_size; + struct hal2_pbus *pbus = &hal2->dac.pbus; + + /* Now we set up some PBUS information. The PBUS needs information about + * what portion of the fifo it will use. If it's receiving or + * transmitting, and finally whether the stream is little endian or big + * endian. The information is written later, on the start call. + */ + sample_size = 2 * hal2->dac.voices; + /* Fifo should be set to hold exactly four samples. Highwater mark + * should be set to two samples. */ + highwater = (sample_size * 2) >> 1; /* halfwords */ + fifobeg = 0; /* playback is first */ + fifoend = (sample_size * 4) >> 3; /* doublewords */ + pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD | + (highwater << 8) | (fifobeg << 16) | (fifoend << 24); + /* We disable everything before we do anything at all */ + pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); + /* Setup the HAL2 for playback */ + hal2_set_dac_rate(hal2); + /* Set endianess */ + hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX); + /* Set DMA bus */ + hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); + /* We are using 1st Bresenham clock generator for playback */ + hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) + | (1 << H2I_C1_CLKID_SHIFT) + | (hal2->dac.voices << H2I_C1_DATAT_SHIFT)); +} + +static void hal2_setup_adc(struct snd_hal2 *hal2) +{ + unsigned int fifobeg, fifoend, highwater, sample_size; + struct hal2_pbus *pbus = &hal2->adc.pbus; + + sample_size = 2 * hal2->adc.voices; + highwater = (sample_size * 2) >> 1; /* halfwords */ + fifobeg = (4 * 4) >> 3; /* record is second */ + fifoend = (4 * 4 + sample_size * 4) >> 3; /* doublewords */ + pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD | + (highwater << 8) | (fifobeg << 16) | (fifoend << 24); + pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); + /* Setup the HAL2 for record */ + hal2_set_adc_rate(hal2); + /* Set endianess */ + hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR); + /* Set DMA bus */ + hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr)); + /* We are using 2nd Bresenham clock generator for record */ + hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT) + | (2 << H2I_C1_CLKID_SHIFT) + | (hal2->adc.voices << H2I_C1_DATAT_SHIFT)); +} + +static void hal2_start_dac(struct snd_hal2 *hal2) +{ + struct hal2_pbus *pbus = &hal2->dac.pbus; + + pbus->pbus->pbdma_dptr = hal2->dac.desc_dma; + pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; + /* enable DAC */ + hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX); +} + +static void hal2_start_adc(struct snd_hal2 *hal2) +{ + struct hal2_pbus *pbus = &hal2->adc.pbus; + + pbus->pbus->pbdma_dptr = hal2->adc.desc_dma; + pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT; + /* enable ADC */ + hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR); +} + +static inline void hal2_stop_dac(struct snd_hal2 *hal2) +{ + hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; + /* The HAL2 itself may remain enabled safely */ +} + +static inline void hal2_stop_adc(struct snd_hal2 *hal2) +{ + hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD; +} + +static int hal2_alloc_dmabuf(struct hal2_codec *codec) +{ + struct hal2_desc *desc; + dma_addr_t desc_dma, buffer_dma; + int count = H2_BUF_SIZE / H2_BLOCK_SIZE; + int i; + + codec->buffer = dma_alloc_noncoherent(NULL, H2_BUF_SIZE, + &buffer_dma, GFP_KERNEL); + if (!codec->buffer) + return -ENOMEM; + desc = dma_alloc_noncoherent(NULL, count * sizeof(struct hal2_desc), + &desc_dma, GFP_KERNEL); + if (!desc) { + dma_free_noncoherent(NULL, H2_BUF_SIZE, + codec->buffer, buffer_dma); + return -ENOMEM; + } + codec->buffer_dma = buffer_dma; + codec->desc_dma = desc_dma; + codec->desc = desc; + for (i = 0; i < count; i++) { + desc->desc.pbuf = buffer_dma + i * H2_BLOCK_SIZE; + desc->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE; + desc->desc.pnext = (i == count - 1) ? + desc_dma : desc_dma + (i + 1) * sizeof(struct hal2_desc); + desc++; + } + dma_cache_sync(NULL, codec->desc, count * sizeof(struct hal2_desc), + DMA_TO_DEVICE); + codec->desc_count = count; + return 0; +} + +static void hal2_free_dmabuf(struct hal2_codec *codec) +{ + dma_free_noncoherent(NULL, codec->desc_count * sizeof(struct hal2_desc), + codec->desc, codec->desc_dma); + dma_free_noncoherent(NULL, H2_BUF_SIZE, codec->buffer, + codec->buffer_dma); +} + +static struct snd_pcm_hardware hal2_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 1024, + .period_bytes_max = 65536, + .periods_min = 2, + .periods_max = 1024, +}; + +static int hal2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int err; + + err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (err < 0) + return err; + + return 0; +} + +static int hal2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int hal2_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + int err; + + runtime->hw = hal2_pcm_hw; + + err = hal2_alloc_dmabuf(&hal2->dac); + if (err) + return err; + return 0; +} + +static int hal2_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + hal2_free_dmabuf(&hal2->dac); + return 0; +} + +static int hal2_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct hal2_codec *dac = &hal2->dac; + + dac->voices = runtime->channels; + dac->sample_rate = hal2_compute_rate(dac, runtime->rate); + memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect)); + dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; + dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); + dac->substream = substream; + hal2_setup_dac(hal2); + return 0; +} + +static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma; + hal2->dac.pcm_indirect.hw_data = 0; + substream->ops->ack(substream); + hal2_start_dac(hal2); + break; + case SNDRV_PCM_TRIGGER_STOP: + hal2_stop_dac(hal2); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +hal2_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *dac = &hal2->dac; + + return snd_pcm_indirect_playback_pointer(substream, &dac->pcm_indirect, + dac->pbus.pbus->pbdma_bptr); +} + +static void hal2_playback_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + unsigned char *buf = hal2->dac.buffer + rec->hw_data; + + memcpy(buf, substream->runtime->dma_area + rec->sw_data, bytes); + dma_cache_sync(NULL, buf, bytes, DMA_TO_DEVICE); + +} + +static int hal2_playback_ack(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *dac = &hal2->dac; + + dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + snd_pcm_indirect_playback_transfer(substream, + &dac->pcm_indirect, + hal2_playback_transfer); + return 0; +} + +static int hal2_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + int err; + + runtime->hw = hal2_pcm_hw; + + err = hal2_alloc_dmabuf(adc); + if (err) + return err; + return 0; +} + +static int hal2_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + hal2_free_dmabuf(&hal2->adc); + return 0; +} + +static int hal2_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct hal2_codec *adc = &hal2->adc; + + adc->voices = runtime->channels; + adc->sample_rate = hal2_compute_rate(adc, runtime->rate); + memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect)); + adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE; + adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2; + adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); + adc->substream = substream; + hal2_setup_adc(hal2); + return 0; +} + +static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma; + hal2->adc.pcm_indirect.hw_data = 0; + printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma); + hal2_start_adc(hal2); + break; + case SNDRV_PCM_TRIGGER_STOP: + hal2_stop_adc(hal2); + break; + default: + return -EINVAL; + } + return 0; +} + +static snd_pcm_uframes_t +hal2_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + + return snd_pcm_indirect_capture_pointer(substream, &adc->pcm_indirect, + adc->pbus.pbus->pbdma_bptr); +} + +static void hal2_capture_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + unsigned char *buf = hal2->adc.buffer + rec->hw_data; + + dma_cache_sync(NULL, buf, bytes, DMA_FROM_DEVICE); + memcpy(substream->runtime->dma_area + rec->sw_data, buf, bytes); +} + +static int hal2_capture_ack(struct snd_pcm_substream *substream) +{ + struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); + struct hal2_codec *adc = &hal2->adc; + + snd_pcm_indirect_capture_transfer(substream, + &adc->pcm_indirect, + hal2_capture_transfer); + return 0; +} + +static struct snd_pcm_ops hal2_playback_ops = { + .open = hal2_playback_open, + .close = hal2_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hal2_pcm_hw_params, + .hw_free = hal2_pcm_hw_free, + .prepare = hal2_playback_prepare, + .trigger = hal2_playback_trigger, + .pointer = hal2_playback_pointer, + .ack = hal2_playback_ack, +}; + +static struct snd_pcm_ops hal2_capture_ops = { + .open = hal2_capture_open, + .close = hal2_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hal2_pcm_hw_params, + .hw_free = hal2_pcm_hw_free, + .prepare = hal2_capture_prepare, + .trigger = hal2_capture_trigger, + .pointer = hal2_capture_pointer, + .ack = hal2_capture_ack, +}; + +static int __devinit hal2_pcm_create(struct snd_hal2 *hal2) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(hal2->card, "SGI HAL2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = hal2; + strcpy(pcm->name, "SGI HAL2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &hal2_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &hal2_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 0, 1024 * 1024); + + return 0; +} + +static int hal2_dev_free(struct snd_device *device) +{ + struct snd_hal2 *hal2 = device->device_data; + + free_irq(SGI_HPCDMA_IRQ, hal2); + kfree(hal2); + return 0; +} + +static struct snd_device_ops hal2_ops = { + .dev_free = hal2_dev_free, +}; + +static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3, + int index) +{ + codec->pbus.pbusnr = index; + codec->pbus.pbus = &hpc3->pbdma[index]; +} + +static int hal2_detect(struct snd_hal2 *hal2) +{ + unsigned short board, major, minor; + unsigned short rev; + + /* reset HAL2 */ + hal2_write(0, &hal2->ctl_regs->isr); + + /* release reset */ + hal2_write(H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N, + &hal2->ctl_regs->isr); + + + hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE); + rev = hal2_read(&hal2->ctl_regs->rev); + if (rev & H2_REV_AUDIO_PRESENT) + return -ENODEV; + + board = (rev & H2_REV_BOARD_M) >> 12; + major = (rev & H2_REV_MAJOR_CHIP_M) >> 4; + minor = (rev & H2_REV_MINOR_CHIP_M); + + printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n", + board, major, minor); + + return 0; +} + +static int hal2_create(struct snd_card *card, struct snd_hal2 **rchip) +{ + struct snd_hal2 *hal2; + struct hpc3_regs *hpc3 = hpc3c0; + int err; + + hal2 = kzalloc(sizeof(struct snd_hal2), GFP_KERNEL); + if (!hal2) + return -ENOMEM; + + hal2->card = card; + + if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED, + "SGI HAL2", hal2)) { + printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ); + kfree(hal2); + return -EAGAIN; + } + + hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0]; + hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1]; + hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2]; + hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3]; + + if (hal2_detect(hal2) < 0) { + kfree(hal2); + return -ENODEV; + } + + hal2_init_codec(&hal2->dac, hpc3, 0); + hal2_init_codec(&hal2->adc, hpc3, 1); + + /* + * All DMA channel interfaces in HAL2 are designed to operate with + * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles + * in D5. HAL2 is a 16-bit device which can accept both big and little + * endian format. It assumes that even address bytes are on high + * portion of PBUS (15:8) and assumes that HPC3 is programmed to + * accept a live (unsynchronized) version of P_DREQ_N from HAL2. + */ +#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \ + (2 << HPC3_DMACFG_D4R_SHIFT) | \ + (2 << HPC3_DMACFG_D5R_SHIFT) | \ + (0 << HPC3_DMACFG_D3W_SHIFT) | \ + (2 << HPC3_DMACFG_D4W_SHIFT) | \ + (2 << HPC3_DMACFG_D5W_SHIFT) | \ + HPC3_DMACFG_DS16 | \ + HPC3_DMACFG_EVENHI | \ + HPC3_DMACFG_RTIME | \ + (8 << HPC3_DMACFG_BURST_SHIFT) | \ + HPC3_DMACFG_DRQLIVE) + /* + * Ignore what's mentioned in the specification and write value which + * works in The Real World (TM) + */ + hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844; + hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, hal2, &hal2_ops); + if (err < 0) { + free_irq(SGI_HPCDMA_IRQ, hal2); + kfree(hal2); + return err; + } + *rchip = hal2; + return 0; +} + +static int __devinit hal2_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_hal2 *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = hal2_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = hal2_pcm_create(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = hal2_mixer_create(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI HAL2 Audio"); + strcpy(card->shortname, "SGI HAL2 Audio"); + sprintf(card->longname, "%s irq %i", + card->shortname, + SGI_HPCDMA_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit hal2_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver hal2_driver = { + .probe = hal2_probe, + .remove = __devexit_p(hal2_remove), + .driver = { + .name = "sgihal2", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_hal2_init(void) +{ + return platform_driver_register(&hal2_driver); +} + +static void __exit alsa_card_hal2_exit(void) +{ + platform_driver_unregister(&hal2_driver); +} + +module_init(alsa_card_hal2_init); +module_exit(alsa_card_hal2_exit); diff --git a/sound/mips/hal2.h b/sound/mips/hal2.h new file mode 100644 index 0000000..f19828b --- /dev/null +++ b/sound/mips/hal2.h @@ -0,0 +1,245 @@ +#ifndef __HAL2_H +#define __HAL2_H + +/* + * Driver for HAL2 sound processors + * Copyright (c) 1999 Ulf Carlsson <ulfc(a)bun.falkenberg.se> + * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis(a)linux-mips.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/types.h> + +/* Indirect status register */ + +#define H2_ISR_TSTATUS 0x01 /* RO: transaction status 1=busy */ +#define H2_ISR_USTATUS 0x02 /* RO: utime status bit 1=armed */ +#define H2_ISR_QUAD_MODE 0x04 /* codec mode 0=indigo 1=quad */ +#define H2_ISR_GLOBAL_RESET_N 0x08 /* chip global reset 0=reset */ +#define H2_ISR_CODEC_RESET_N 0x10 /* codec/synth reset 0=reset */ + +/* Revision register */ + +#define H2_REV_AUDIO_PRESENT 0x8000 /* RO: audio present 0=present */ +#define H2_REV_BOARD_M 0x7000 /* RO: bits 14:12, board revision */ +#define H2_REV_MAJOR_CHIP_M 0x00F0 /* RO: bits 7:4, major chip revision */ +#define H2_REV_MINOR_CHIP_M 0x000F /* RO: bits 3:0, minor chip revision */ + +/* Indirect address register */ + +/* + * Address of indirect internal register to be accessed. A write to this + * register initiates read or write access to the indirect registers in the + * HAL2. Note that there af four indirect data registers for write access to + * registers larger than 16 byte. + */ + +#define H2_IAR_TYPE_M 0xF000 /* bits 15:12, type of functional */ + /* block the register resides in */ + /* 1=DMA Port */ + /* 9=Global DMA Control */ + /* 2=Bresenham */ + /* 3=Unix Timer */ +#define H2_IAR_NUM_M 0x0F00 /* bits 11:8 instance of the */ + /* blockin which the indirect */ + /* register resides */ + /* If IAR_TYPE_M=DMA Port: */ + /* 1=Synth In */ + /* 2=AES In */ + /* 3=AES Out */ + /* 4=DAC Out */ + /* 5=ADC Out */ + /* 6=Synth Control */ + /* If IAR_TYPE_M=Global DMA Control: */ + /* 1=Control */ + /* If IAR_TYPE_M=Bresenham: */ + /* 1=Bresenham Clock Gen 1 */ + /* 2=Bresenham Clock Gen 2 */ + /* 3=Bresenham Clock Gen 3 */ + /* If IAR_TYPE_M=Unix Timer: */ + /* 1=Unix Timer */ +#define H2_IAR_ACCESS_SELECT 0x0080 /* 1=read 0=write */ +#define H2_IAR_PARAM 0x000C /* Parameter Select */ +#define H2_IAR_RB_INDEX_M 0x0003 /* Read Back Index */ + /* 00:word0 */ + /* 01:word1 */ + /* 10:word2 */ + /* 11:word3 */ +/* + * HAL2 internal addressing + * + * The HAL2 has "indirect registers" (idr) which are accessed by writing to the + * Indirect Data registers. Write the address to the Indirect Address register + * to transfer the data. + * + * We define the H2IR_* to the read address and H2IW_* to the write address and + * H2I_* to be fields in whatever register is referred to. + * + * When we write to indirect registers which are larger than one word (16 bit) + * we have to fill more than one indirect register before writing. When we read + * back however we have to read several times, each time with different Read + * Back Indexes (there are defs for doing this easily). + */ + +/* + * Relay Control + */ +#define H2I_RELAY_C 0x9100 +#define H2I_RELAY_C_STATE 0x01 /* state of RELAY pin signal */ + +/* DMA port enable */ + +#define H2I_DMA_PORT_EN 0x9104 +#define H2I_DMA_PORT_EN_SY_IN 0x01 /* Synth_in DMA port */ +#define H2I_DMA_PORT_EN_AESRX 0x02 /* AES receiver DMA port */ +#define H2I_DMA_PORT_EN_AESTX 0x04 /* AES transmitter DMA port */ +#define H2I_DMA_PORT_EN_CODECTX 0x08 /* CODEC transmit DMA port */ +#define H2I_DMA_PORT_EN_CODECR 0x10 /* CODEC receive DMA port */ + +#define H2I_DMA_END 0x9108 /* global dma endian select */ +#define H2I_DMA_END_SY_IN 0x01 /* Synth_in DMA port */ +#define H2I_DMA_END_AESRX 0x02 /* AES receiver DMA port */ +#define H2I_DMA_END_AESTX 0x04 /* AES transmitter DMA port */ +#define H2I_DMA_END_CODECTX 0x08 /* CODEC transmit DMA port */ +#define H2I_DMA_END_CODECR 0x10 /* CODEC receive DMA port */ + /* 0=b_end 1=l_end */ + +#define H2I_DMA_DRV 0x910C /* global PBUS DMA enable */ + +#define H2I_SYNTH_C 0x1104 /* Synth DMA control */ + +#define H2I_AESRX_C 0x1204 /* AES RX dma control */ + +#define H2I_C_TS_EN 0x20 /* Timestamp enable */ +#define H2I_C_TS_FRMT 0x40 /* Timestamp format */ +#define H2I_C_NAUDIO 0x80 /* Sign extend */ + +/* AESRX CTL, 16 bit */ + +#define H2I_AESTX_C 0x1304 /* AES TX DMA control */ +#define H2I_AESTX_C_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ +#define H2I_AESTX_C_CLKID_M 0x18 +#define H2I_AESTX_C_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ +#define H2I_AESTX_C_DATAT_M 0x300 + +/* CODEC registers */ + +#define H2I_DAC_C1 0x1404 /* DAC DMA control, 16 bit */ +#define H2I_DAC_C2 0x1408 /* DAC DMA control, 32 bit */ +#define H2I_ADC_C1 0x1504 /* ADC DMA control, 16 bit */ +#define H2I_ADC_C2 0x1508 /* ADC DMA control, 32 bit */ + +/* Bits in CTL1 register */ + +#define H2I_C1_DMA_SHIFT 0 /* DMA channel */ +#define H2I_C1_DMA_M 0x7 +#define H2I_C1_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */ +#define H2I_C1_CLKID_M 0x18 +#define H2I_C1_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */ +#define H2I_C1_DATAT_M 0x300 + +/* Bits in CTL2 register */ + +#define H2I_C2_R_GAIN_SHIFT 0 /* right a/d input gain */ +#define H2I_C2_R_GAIN_M 0xf +#define H2I_C2_L_GAIN_SHIFT 4 /* left a/d input gain */ +#define H2I_C2_L_GAIN_M 0xf0 +#define H2I_C2_R_SEL 0x100 /* right input select */ +#define H2I_C2_L_SEL 0x200 /* left input select */ +#define H2I_C2_MUTE 0x400 /* mute */ +#define H2I_C2_DO1 0x00010000 /* digital output port bit 0 */ +#define H2I_C2_DO2 0x00020000 /* digital output port bit 1 */ +#define H2I_C2_R_ATT_SHIFT 18 /* right d/a output - */ +#define H2I_C2_R_ATT_M 0x007c0000 /* attenuation */ +#define H2I_C2_L_ATT_SHIFT 23 /* left d/a output - */ +#define H2I_C2_L_ATT_M 0x0f800000 /* attenuation */ + +#define H2I_SYNTH_MAP_C 0x1104 /* synth dma handshake ctrl */ + +/* Clock generator CTL 1, 16 bit */ + +#define H2I_BRES1_C1 0x2104 +#define H2I_BRES2_C1 0x2204 +#define H2I_BRES3_C1 0x2304 + +#define H2I_BRES_C1_SHIFT 0 /* 0=48.0 1=44.1 2=aes_rx */ +#define H2I_BRES_C1_M 0x03 + +/* Clock generator CTL 2, 32 bit */ + +#define H2I_BRES1_C2 0x2108 +#define H2I_BRES2_C2 0x2208 +#define H2I_BRES3_C2 0x2308 + +#define H2I_BRES_C2_INC_SHIFT 0 /* increment value */ +#define H2I_BRES_C2_INC_M 0xffff +#define H2I_BRES_C2_MOD_SHIFT 16 /* modcontrol value */ +#define H2I_BRES_C2_MOD_M 0xffff0000 /* modctrl=0xffff&(modinc-1) */ + +/* Unix timer, 64 bit */ + +#define H2I_UTIME 0x3104 +#define H2I_UTIME_0_LD 0xffff /* microseconds, LSB's */ +#define H2I_UTIME_1_LD0 0x0f /* microseconds, MSB's */ +#define H2I_UTIME_1_LD1 0xf0 /* tenths of microseconds */ +#define H2I_UTIME_2_LD 0xffff /* seconds, LSB's */ +#define H2I_UTIME_3_LD 0xffff /* seconds, MSB's */ + +struct hal2_ctl_regs { + u32 _unused0[4]; + u32 isr; /* 0x10 Status Register */ + u32 _unused1[3]; + u32 rev; /* 0x20 Revision Register */ + u32 _unused2[3]; + u32 iar; /* 0x30 Indirect Address Register */ + u32 _unused3[3]; + u32 idr0; /* 0x40 Indirect Data Register 0 */ + u32 _unused4[3]; + u32 idr1; /* 0x50 Indirect Data Register 1 */ + u32 _unused5[3]; + u32 idr2; /* 0x60 Indirect Data Register 2 */ + u32 _unused6[3]; + u32 idr3; /* 0x70 Indirect Data Register 3 */ +}; + +struct hal2_aes_regs { + u32 rx_stat[2]; /* Status registers */ + u32 rx_cr[2]; /* Control registers */ + u32 rx_ud[4]; /* User data window */ + u32 rx_st[24]; /* Channel status data */ + + u32 tx_stat[1]; /* Status register */ + u32 tx_cr[3]; /* Control registers */ + u32 tx_ud[4]; /* User data window */ + u32 tx_st[24]; /* Channel status data */ +}; + +struct hal2_vol_regs { + u32 right; /* Right volume */ + u32 left; /* Left volume */ +}; + +struct hal2_syn_regs { + u32 _unused0[2]; + u32 page; /* DOC Page register */ + u32 regsel; /* DOC Register selection */ + u32 dlow; /* DOC Data low */ + u32 dhigh; /* DOC Data high */ + u32 irq; /* IRQ Status */ + u32 dram; /* DRAM Access */ +}; + +#endif /* __HAL2_H */

2 1

[alsa-devel] hda-codec conexant - no external mic on FSC V5505 (with patch)
by Marek Lotke 12 Jul '08

12 Jul '08

Hi, I've tried all conextant models for my new laptop... External mic would work only with benq but then the external headphones don't work. I managed to get the external mic working with this little patch: --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -686,7 +686,7 @@ static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { static struct hda_verb cxt5045_init_verbs[] = { /* Line in, Mic */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 }, /* HP, Amp */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, Now I can switch between the mics with mixer. Still no mic auto sensing though but maybe there is none in hardware. Attached are the outputs of alsa-info.sh: default, with model benq and with default model with my patch with mixer set to external. I'll be glad to provide any info or try any patch... Thanks, Marek

2 3

[alsa-devel] [PATCH] asoc: kbuild - only show menus for the current ASoC CPU platform.
by Liam Girdwood 12 Jul '08

12 Jul '08

We don't want to see ASoC platform menus for other non selected architectures in our config. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/fsl/Kconfig | 4 ---- sound/soc/omap/Kconfig | 4 ---- 2 files changed, 0 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 19802e2..3368ace 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,3 @@ -menu "ALSA SoC audio for Freescale SOCs" - config SND_SOC_MPC8610 bool "ALSA SoC support for the MPC8610 SOC" depends on MPC8610_HPCD @@ -16,5 +14,3 @@ config SND_SOC_MPC8610_HPCD default y if MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - -endmenu diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0230d83..aea27e7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,5 +1,3 @@ -menu "SoC Audio for the Texas Instruments OMAP" - config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" depends on ARCH_OMAP && SND_SOC @@ -15,5 +13,3 @@ config SND_OMAP_SOC_N810 select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on Nokia N810. - -endmenu -- 1.5.4.3

2 1

Re: [alsa-devel] [Fwd: Re: [PATCH 2/3] ALSA SoC: Add mpc5200-psc I2S driver]
by Grant Likely 12 Jul '08

12 Jul '08

On Sat, Jul 12, 2008 at 8:31 AM, Liam Girdwood <lg(a)opensource.wolfsonmicro.com> wrote: > On Sat, 2008-07-12 at 00:53 -0600, Grant Likely wrote: >> On Wed, Jul 02, 2008 at 11:51:17AM +0100, Liam Girdwood wrote: > >> > >> > Fwiw, I usually separate out the DMA from I2S so it can be used by AC97, >> > PCM interfaces etc. If your hardware only has I2S the it's not required. >> >> Agreed. However, I don't think I have my mind sufficiently wrapped >> around ASoC yet to split it up without making a royal mess out of it. >> >> Would you be okay with merging it without the split? I know that >> at least two others people are actively playing with this driver, and >> it will be easier to collaborate on adding features if it is in >> mainline. >> > > I've no objections if it's something on your roadmap for the future. I > suppose the AC97 parts will only be written (and I2S de-merged) when > mpc5200 AC97 hardware is available. Oh, absolutely. It is definitely something on my roadmap. Thanks, g. -- Grant Likely, B.Sc., P.Eng. Secret Lab Technologies Ltd.

1 0

[alsa-devel] [PATCH v3] ALSA driver for SGI O2 audio board
by Thomas Bogendoerfer 11 Jul '08

11 Jul '08

This patch adds a new ALSA driver for the audio device found inside most of the SGI O2 workstation. The hardware uses a SGI custom chip, which feeds a AD codec chip. Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> --- Please apply for 2.6.27 Changes in v3: - removed IRQF_SHARED from request_irq, because they can't be shared Changes in v2: - removed unused volume field - spreaded some statics - switch over to use C99 field inits - use msleep_interuptible instead of long udelay - use schedule_timeout_interruptible instead of simple schedule include/sound/ad1843.h | 46 +++ sound/mips/Kconfig | 6 + sound/mips/Makefile | 2 + sound/mips/ad1843.c | 561 ++++++++++++++++++++++++++ sound/mips/sgio2audio.c | 1006 +++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 1621 insertions(+), 0 deletions(-) diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h new file mode 100644 index 0000000..b236a9d --- /dev/null +++ b/include/sound/ad1843.h @@ -0,0 +1,46 @@ +/* + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)franken.de> + */ + +#ifndef __SOUND_AD1843_H +#define __SOUND_AD1843_H + +struct snd_ad1843 { + void *chip; + int (*read)(void *chip, int reg); + int (*write)(void *chip, int reg, int val); +}; + +#define AD1843_GAIN_RECLEV 0 +#define AD1843_GAIN_LINE 1 +#define AD1843_GAIN_LINE_2 2 +#define AD1843_GAIN_MIC 3 +#define AD1843_GAIN_PCM_0 4 +#define AD1843_GAIN_PCM_1 5 +#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1) + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id); +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id); +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval); +int ad1843_get_recsrc(struct snd_ad1843 *ad1843); +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc); +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, + unsigned int id); +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843); +int ad1843_init(struct snd_ad1843 *ad1843); + +#endif /* __SOUND_AD1843_H */ diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 531f8ba..3ce743b 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -11,5 +11,11 @@ config SND_AU1X00 help ALSA Sound driver for the Au1x00's AC97 port. +config SND_SGI_O2 + tristate "SGI O2 Audio" + depends on SGI_IP32 + help + Sound support for the SGI O2 Workstation. + endmenu diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 47afed9..55624d8 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -2,7 +2,9 @@ # Makefile for ALSA # +snd-sgi-o2-objs := sgio2audio.o ad1843.o snd-au1x00-objs := au1x00.o # Toplevel Module Dependency obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o +obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c new file mode 100644 index 0000000..c624510 --- /dev/null +++ b/sound/mips/ad1843.c @@ -0,0 +1,561 @@ +/* + * AD1843 low level driver + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> + * + * inspired from vwsnd.c (SGI VW audio driver) + * Copyright 1999 Silicon Graphics, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/sched.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ad1843.h> + +/* + * AD1843 bitfield definitions. All are named as in the AD1843 data + * sheet, with ad1843_ prepended and individual bit numbers removed. + * + * E.g., bits LSS0 through LSS2 become ad1843_LSS. + * + * Only the bitfields we need are defined. + */ + +struct ad1843_bitfield { + char reg; + char lo_bit; + char nbits; +}; + +static const struct ad1843_bitfield + ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */ + ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */ + ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */ + ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */ + ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */ + ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */ + ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */ + ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */ + ad1843_RD2M = { 3, 0, 5 }, /* Right DAC 2 Mix Gain/Atten */ + ad1843_RD2MM = { 3, 7, 1 }, /* Right DAC 2 Mix Mute */ + ad1843_LD2M = { 3, 8, 5 }, /* Left DAC 2 Mix Gain/Atten */ + ad1843_LD2MM = { 3, 15, 1 }, /* Left DAC 2 Mix Mute */ + ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */ + ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */ + ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */ + ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */ + ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */ + ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */ + ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */ + ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */ + ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */ + ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */ + ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */ + ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */ + ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */ + ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */ + ad1843_MPOM = { 8, 6, 1 }, /* Mono Output Mute */ + ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */ + ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */ + ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */ + ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */ + ad1843_RDA2G = { 10, 0, 6 }, /* Right DAC2 Analog/Digital Gain */ + ad1843_RDA2GM = { 10, 7, 1 }, /* Right DAC2 Analog Mute */ + ad1843_LDA2G = { 10, 8, 6 }, /* Left DAC2 Analog/Digital Gain */ + ad1843_LDA2GM = { 10, 15, 1 }, /* Left DAC2 Analog Mute */ + ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */ + ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */ + ad1843_RDA2AM = { 12, 7, 1 }, /* Right DAC2 Digital Mute */ + ad1843_LDA2AM = { 12, 15, 1 }, /* Left DAC2 Digital Mute */ + ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */ + ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */ + ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */ + ad1843_DA2C = { 15, 10, 2 }, /* DAC2 Sample Rate Source */ + ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */ + ad1843_C2C = { 20, 0, 16 }, /* Clock 2 Sample Rate Select */ + ad1843_C3C = { 23, 0, 16 }, /* Clock 3 Sample Rate Select */ + ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */ + ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */ + ad1843_DAMIX = { 25, 14, 1 }, /* DAC Digital Mix Enable */ + ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */ + ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */ + ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */ + ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */ + ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */ + ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */ + ad1843_DA2F = { 26, 10, 2 }, /* DAC2 Data Format Select */ + ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */ + ad1843_DA2SM = { 26, 15, 1 }, /* DAC2 Stereo/Mono Mode Select */ + ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */ + ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */ + ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */ + ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */ + ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */ + ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */ + ad1843_DDMEN = { 27, 12, 1 }, /* DAC2 to DAC1 Mix Enable */ + ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */ + ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */ + ad1843_C3EN = { 28, 13, 1 }, /* Clock Generator 3 Enable */ + ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */ + +/* + * The various registers of the AD1843 use three different formats for + * specifying gain. The ad1843_gain structure parameterizes the + * formats. + */ + +struct ad1843_gain { + int negative; /* nonzero if gain is negative. */ + const struct ad1843_bitfield *lfield; + const struct ad1843_bitfield *rfield; + const struct ad1843_bitfield *lmute; + const struct ad1843_bitfield *rmute; +}; + +static const struct ad1843_gain ad1843_gain_RECLEV = { + .negative = 0, + .lfield = &ad1843_LIG, + .rfield = &ad1843_RIG +}; +static const struct ad1843_gain ad1843_gain_LINE = { + .negative = 1, + .lfield = &ad1843_LX1M, + .rfield = &ad1843_RX1M, + .lmute = &ad1843_LX1MM, + .rmute = &ad1843_RX1MM +}; +static const struct ad1843_gain ad1843_gain_LINE_2 = { + .negative = 1, + .lfield = &ad1843_LDA2G, + .rfield = &ad1843_RDA2G, + .lmute = &ad1843_LDA2GM, + .rmute = &ad1843_RDA2GM +}; +static const struct ad1843_gain ad1843_gain_MIC = { + .negative = 1, + .lfield = &ad1843_LMCM, + .rfield = &ad1843_RMCM, + .lmute = &ad1843_LMCMM, + .rmute = &ad1843_RMCMM +}; +static const struct ad1843_gain ad1843_gain_PCM_0 = { + .negative = 1, + .lfield = &ad1843_LDA1G, + .rfield = &ad1843_RDA1G, + .lmute = &ad1843_LDA1GM, + .rmute = &ad1843_RDA1GM +}; +static const struct ad1843_gain ad1843_gain_PCM_1 = { + .negative = 1, + .lfield = &ad1843_LD2M, + .rfield = &ad1843_RD2M, + .lmute = &ad1843_LD2MM, + .rmute = &ad1843_RD2MM +}; + +static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] = +{ + &ad1843_gain_RECLEV, + &ad1843_gain_LINE, + &ad1843_gain_LINE_2, + &ad1843_gain_MIC, + &ad1843_gain_PCM_0, + &ad1843_gain_PCM_1, +}; + +/* read the current value of an AD1843 bitfield. */ + +static int ad1843_read_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field) +{ + int w; + + w = ad1843->read(ad1843->chip, field->reg); + return w >> field->lo_bit & ((1 << field->nbits) - 1); +} + +/* + * write a new value to an AD1843 bitfield and return the old value. + */ + +static int ad1843_write_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field, + int newval) +{ + int w, mask, oldval, newbits; + + w = ad1843->read(ad1843->chip, field->reg); + mask = ((1 << field->nbits) - 1) << field->lo_bit; + oldval = (w & mask) >> field->lo_bit; + newbits = (newval << field->lo_bit) & mask; + w = (w & ~mask) | newbits; + ad1843->write(ad1843->chip, field->reg, w); + + return oldval; +} + +/* + * ad1843_read_multi reads multiple bitfields from the same AD1843 + * register. It uses a single read cycle to do it. (Reading the + * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20 + * microseconds.) + * + * Called like this. + * + * ad1843_read_multi(ad1843, nfields, + * &ad1843_FIELD1, &val1, + * &ad1843_FIELD2, &val2, ...); + */ + +static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + const struct ad1843_bitfield *fp; + int w = 0, mask, *value, reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int *); + if (reg == -1) { + reg = fp->reg; + w = ad1843->read(ad1843->chip, reg); + } + + mask = (1 << fp->nbits) - 1; + *value = w >> fp->lo_bit & mask; + } + va_end(ap); +} + +/* + * ad1843_write_multi stores multiple bitfields into the same AD1843 + * register. It uses one read and one write cycle to do it. + * + * Called like this. + * + * ad1843_write_multi(ad1843, nfields, + * &ad1843_FIELD1, val1, + * &ad1843_FIELF2, val2, ...); + */ + +static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + int reg; + const struct ad1843_bitfield *fp; + int value; + int w, m, mask, bits; + + mask = 0; + bits = 0; + reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int); + if (reg == -1) + reg = fp->reg; + else + BUG_ON(reg != fp->reg); + m = ((1 << fp->nbits) - 1) << fp->lo_bit; + mask |= m; + bits |= (value << fp->lo_bit) & m; + } + va_end(ap); + + if (~mask & 0xFFFF) + w = ad1843->read(ad1843->chip, reg); + else + w = 0; + w = (w & ~mask) | bits; + ad1843->write(ad1843->chip, reg, w); +} + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + int ret; + + ret = (1 << gp->lfield->nbits); + if (!gp->lmute) + ret -= 1; + return ret; +} + +/* + * ad1843_get_gain reads the specified register and extracts the gain value + * using the supplied gain type. + */ + +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id) +{ + int lg, rg, lm, rm; + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg); + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) { + ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm); + if (lm) + lg = 0; + if (rm) + rg = 0; + } + return lg << 0 | rg << 8; +} + +/* + * Set an audio channel's gain. + * + * Returns the new gain, which may be lower than the old gain. + */ + +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + int lg = (newval >> 0) & mask; + int rg = (newval >> 8) & mask; + int lm = (lg == 0) ? 1 : 0; + int rm = (rg == 0) ? 1 : 0; + + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) + ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm); + ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg); + return ad1843_get_gain(ad1843, id); +} + +/* Returns the current recording source */ + +int ad1843_get_recsrc(struct snd_ad1843 *ad1843) +{ + int val = ad1843_read_bits(ad1843, &ad1843_LSS); + + if (val < 0 || val > 2) { + val = 2; + ad1843_write_multi(ad1843, 2, + &ad1843_LSS, val, &ad1843_RSS, val); + } + return val; +} + +/* + * Set recording source. + * + * Returns newsrc on success, -errno on failure. + */ + +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc) +{ + if (newsrc < 0 || newsrc > 2) + return -EINVAL; + + ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc); + return newsrc; +} + +/* Setup ad1843 for D/A conversion. */ + +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int ad_fmt = 0, ad_mode = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_U8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + ad_fmt = 1; + break; + case SNDRV_PCM_FORMAT_MU_LAW: + ad_fmt = 2; + break; + case SNDRV_PCM_FORMAT_A_LAW: + ad_fmt = 3; + break; + default: + break; + } + + switch (channels) { + case 2: + ad_mode = 0; + break; + case 1: + ad_mode = 1; + break; + default: + break; + } + + if (id) { + ad1843_write_bits(ad1843, &ad1843_C2C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA2SM, ad_mode, + &ad1843_DA2F, ad_fmt); + } else { + ad1843_write_bits(ad1843, &ad1843_C1C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA1SM, ad_mode, + &ad1843_DA1F, ad_fmt); + } +} + +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id) +{ + if (id) + ad1843_write_bits(ad1843, &ad1843_DA2F, 1); + else + ad1843_write_bits(ad1843, &ad1843_DA1F, 1); +} + +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int da_fmt = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_U8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_S16_LE: da_fmt = 1; break; + case SNDRV_PCM_FORMAT_MU_LAW: da_fmt = 2; break; + case SNDRV_PCM_FORMAT_A_LAW: da_fmt = 3; break; + default: break; + } + + ad1843_write_bits(ad1843, &ad1843_C3C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt); +} + +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843) +{ + /* nothing to do */ +} + +/* + * Fully initialize the ad1843. As described in the AD1843 data + * sheet, section "START-UP SEQUENCE". The numbered comments are + * subsection headings from the data sheet. See the data sheet, pages + * 52-54, for more info. + * + * return 0 on success, -errno on failure. */ + +int ad1843_init(struct snd_ad1843 *ad1843) +{ + unsigned long later; + + if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) { + printk(KERN_ERR "ad1843: AD1843 won't initialize\n"); + return -EIO; + } + + ad1843_write_bits(ad1843, &ad1843_SCF, 1); + + /* 4. Put the conversion resources into standby. */ + ad1843_write_bits(ad1843, &ad1843_PDNI, 0); + later = jiffies + msecs_to_jiffies(500); + + while (ad1843_read_bits(ad1843, &ad1843_PDNO)) { + if (time_after(jiffies, later)) { + printk(KERN_ERR + "ad1843: AD1843 won't power up\n"); + return -EIO; + } + schedule_timeout_interruptible(5); + } + + /* 5. Power up the clock generators and enable clock output pins. */ + ad1843_write_multi(ad1843, 3, + &ad1843_C1EN, 1, + &ad1843_C2EN, 1, + &ad1843_C3EN, 1); + + /* 6. Configure conversion resources while they are in standby. */ + + /* DAC1/2 use clock 1/2 as source, ADC uses clock 3. Always. */ + ad1843_write_multi(ad1843, 4, + &ad1843_DA1C, 1, + &ad1843_DA2C, 2, + &ad1843_ADLC, 3, + &ad1843_ADRC, 3); + + /* 7. Enable conversion resources. */ + ad1843_write_bits(ad1843, &ad1843_ADTLK, 1); + ad1843_write_multi(ad1843, 7, + &ad1843_ANAEN, 1, + &ad1843_AAMEN, 1, + &ad1843_DA1EN, 1, + &ad1843_DA2EN, 1, + &ad1843_DDMEN, 1, + &ad1843_ADLEN, 1, + &ad1843_ADREN, 1); + + /* 8. Configure conversion resources while they are enabled. */ + + /* set gain to 0 for all channels */ + ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0); + + /* Unmute all channels. */ + /* DAC1 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0); + /* DAC2 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0); + + /* Set default recording source to Line In and set + * mic gain to +20 dB. + */ + ad1843_set_recsrc(ad1843, 2); + ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1); + + /* Set Speaker Out level to +/- 4V and unmute it. */ + ad1843_write_multi(ad1843, 3, + &ad1843_HPOS, 1, + &ad1843_HPOM, 0, + &ad1843_MPOM, 0); + + return 0; +} diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c new file mode 100644 index 0000000..4c63504 --- /dev/null +++ b/sound/mips/sgio2audio.c @@ -0,0 +1,1006 @@ +/* + * Sound driver for Silicon Graphics O2 Workstations A/V board audio. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> + * Mxier part taken from mace_audio.c: + * Copyright 2007 Thorben Jändling <tj.trevelyan(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/spinlock.h> +#include <linux/gfp.h> +#include <linux/vmalloc.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <asm/ip32/ip32_ints.h> +#include <asm/ip32/mace.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#define SNDRV_GET_ID +#include <sound/initval.h> +#include <sound/ad1843.h> + + +MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier(a)linux-mips.org>"); +MODULE_DESCRIPTION("SGI O2 Audio"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard."); + + +#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */ +#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */ + +#define CODEC_CONTROL_WORD_SHIFT 0 +#define CODEC_CONTROL_READ BIT(16) +#define CODEC_CONTROL_ADDRESS_SHIFT 17 + +#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */ +#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */ +#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */ +#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */ +#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */ +#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */ +#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */ +#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */ + +#define CHANNEL_RING_SHIFT 12 +#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT) +#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1) + +#define CHANNEL_LEFT_SHIFT 40 +#define CHANNEL_RIGHT_SHIFT 8 + +struct snd_sgio2audio_chan { + int idx; + struct snd_pcm_substream *substream; + int pos; + snd_pcm_uframes_t size; + spinlock_t lock; +}; + +/* definition of the chip-specific record */ +struct snd_sgio2audio { + struct snd_card *card; + + /* codec */ + struct snd_ad1843 ad1843; + spinlock_t ad1843_lock; + + /* channels */ + struct snd_sgio2audio_chan channel[3]; + + /* resources */ + void *ring_base; + dma_addr_t ring_base_dma; +}; + +/* AD1843 access */ + +/* + * read_ad1843_reg returns the current contents of a 16 bit AD1843 register. + * + * Returns unsigned register value on success, -errno on failure. + */ +static int read_ad1843_reg(void *priv, int reg) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + CODEC_CONTROL_READ, &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + val = readq(&mace->perif.audio.codec_read); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return val; +} + +/* + * write_ad1843_reg writes the specified value to a 16 bit AD1843 register. + */ +static int write_ad1843_reg(void *priv, int reg, int word) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + (word << CODEC_CONTROL_WORD_SHIFT), + &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return 0; +} + +static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843, + (int)kcontrol->private_value); + return 0; +} + +static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int vol; + + vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value); + + ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF; + ucontrol->value.integer.value[1] = vol & 0xFF; + + return 0; +} + +static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newvol, oldvol; + + oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value); + newvol = (ucontrol->value.integer.value[0] << 8) | + ucontrol->value.integer.value[1]; + + newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value, + newvol); + + return newvol != oldvol; +} + +static int sgio2audio_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "Cam Mic", "Mic", "Line" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int sgio2audio_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843); + return 0; +} + +static int sgio2audio_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newsrc, oldsrc; + + oldsrc = ad1843_get_recsrc(&chip->ad1843); + newsrc = ad1843_set_recsrc(&chip->ad1843, + ucontrol->value.enumerated.item[0]); + + return newsrc != oldsrc; +} + +/* dac1/pcm0 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_0, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* dac2/pcm1 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_1, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_RECLEV, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level source control */ +static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = sgio2audio_source_info, + .get = sgio2audio_source_get, + .put = sgio2audio_source_put, +}; + +/* line mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* cd mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE_2, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* mic mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_MIC, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + + +static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip) +{ + int err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_recsource, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_line, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_cd, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_mic, chip)); + if (err < 0) + return err; + + return 0; +} + +/* low-level audio interface DMA */ + +/* get data out of bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + unsigned long src_base, src_pos, dst_mask; + unsigned char *dst_base; + int dst_pos; + u64 *src; + s16 *dst; + u64 x; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT); + src_pos = readq(&mace->perif.audio.chan[ch].read_ptr); + dst_base = runtime->dma_area; + dst_pos = chip->channel[ch].pos; + dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (u64 *)(src_base + src_pos); + dst = (s16 *)(dst_base + dst_pos); + + x = *src; + dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff; + dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff; + + src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK; + dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask; + count -= sizeof(u64); + } + + writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */ + chip->channel[ch].pos = dst_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +/* put some DMA data in bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + s64 l, r; + unsigned long dst_base, dst_pos, src_mask; + unsigned char *src_base; + int src_pos; + u64 *dst; + s16 *src; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT); + dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr); + src_base = runtime->dma_area; + src_pos = chip->channel[ch].pos; + src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (s16 *)(src_base + src_pos); + dst = (u64 *)(dst_base + dst_pos); + + l = src[0]; /* sign extend */ + r = src[1]; /* sign extend */ + + *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) | + ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT); + + dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK; + src_pos = (src_pos + 2 * sizeof(s16)) & src_mask; + count -= sizeof(u64); + } + + writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */ + chip->channel[ch].pos = src_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + + /* reset DMA channel */ + writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control); + udelay(10); + writeq(0, &mace->perif.audio.chan[ch].control); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* push a full buffer */ + snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32); + } + /* set DMA to wake on 50% empty and enable interrupt */ + writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50, + &mace->perif.audio.chan[ch].control); + return 0; +} + +static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + writeq(0, &mace->perif.audio.chan[chan->idx].control); + return 0; +} + +static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + + /* empty the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_pull_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + /* fill the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_push_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + + substream = chan->substream; + snd_sgio2audio_dma_stop(substream); + snd_sgio2audio_dma_start(substream); + return IRQ_HANDLED; +} + +/* PCM part */ +/* PCM hardware definition */ +static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 32768, + .period_bytes_max = 65536, + .periods_min = 1, + .periods_max = 1024, +}; + +/* PCM playback open callback */ +static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[1]; + return 0; +} + +static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[2]; + return 0; +} + +/* PCM capture open callback */ +static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[0]; + return 0; +} + +/* PCM close callback */ +static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->private_data = NULL; + return 0; +} + + +/* hw_params callback */ +static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int size = params_buffer_bytes(hw_params); + + /* alloc virtual 'dma' area */ + if (runtime->dma_area) + vfree(runtime->dma_area); + runtime->dma_area = vmalloc(size); + if (runtime->dma_area == NULL) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* hw_free callback */ +static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) +{ + if (substream->runtime->dma_area) + vfree(substream->runtime->dma_area); + substream->runtime->dma_area = NULL; + return 0; +} + +/* prepare callback */ +static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + unsigned long flags; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + /* Setup the pseudo-dma transfer pointers. */ + chip->channel[ch].pos = 0; + chip->channel[ch].size = 0; + chip->channel[ch].substream = substream; + + /* set AD1843 format */ + /* hardware format is always S16_LE */ + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + ad1843_setup_dac(&chip->ad1843, + ch - 1, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + case SNDRV_PCM_STREAM_CAPTURE: + ad1843_setup_adc(&chip->ad1843, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + } + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return 0; +} + +/* trigger callback */ +static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* start the PCM engine */ + snd_sgio2audio_dma_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* stop the PCM engine */ + snd_sgio2audio_dma_stop(substream); + break; + default: + return -EINVAL; + } + return 0; +} + +/* pointer callback */ +static snd_pcm_uframes_t +snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + /* get the current hardware pointer */ + return bytes_to_frames(substream->runtime, + chip->channel[chan->idx].pos); +} + +/* get the physical page pointer on the given offset */ +static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} + +/* operators */ +static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { + .open = snd_sgio2audio_playback1_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { + .open = snd_sgio2audio_playback2_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_capture_ops = { + .open = snd_sgio2audio_capture_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +/* + * definitions of capture are omitted here... + */ + +/* create a pcm device */ +static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC1"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback1_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_sgio2audio_capture_ops); + + /* create second pcm device with one outputs and no input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback2_ops); + + return 0; +} + +static struct { + int idx; + int irq; + irqreturn_t (*isr)(int, void *); + const char *desc; +} snd_sgio2_isr_table[] = { + { + .idx = 0, + .irq = MACEISA_AUDIO1_DMAT_IRQ, + .isr = snd_sgio2audio_dma_in_isr, + .desc = "Capture DMA Channel 0" + }, { + .idx = 0, + .irq = MACEISA_AUDIO1_OF_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Capture Overflow" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 1" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 1" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 2" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 2" + } +}; + +/* ALSA driver */ + +static int snd_sgio2audio_free(struct snd_sgio2audio *chip) +{ + int i; + + /* reset interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + + /* release IRQ's */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) + free_irq(snd_sgio2_isr_table[i].irq, + &chip->channel[snd_sgio2_isr_table[i].idx]); + + dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + chip->ring_base, chip->ring_base_dma); + + /* release card data */ + kfree(chip); + return 0; +} + +static int snd_sgio2audio_dev_free(struct snd_device *device) +{ + struct snd_sgio2audio *chip = device->device_data; + + return snd_sgio2audio_free(chip); +} + +static struct snd_device_ops ops = { + .dev_free = snd_sgio2audio_dev_free, +}; + +static int __devinit snd_sgio2audio_create(struct snd_card *card, + struct snd_sgio2audio **rchip) +{ + struct snd_sgio2audio *chip; + int i, err; + + *rchip = NULL; + + /* check if a codec is attached to the interface */ + /* (Audio or Audio/Video board present) */ + if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT)) + return -ENOENT; + + chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + &chip->ring_base_dma, GFP_USER); + if (chip->ring_base == NULL) { + printk(KERN_ERR + "sgio2audio: could not allocate ring buffers\n"); + kfree(chip); + return -ENOMEM; + } + + spin_lock_init(&chip->ad1843_lock); + + /* initialize channels */ + for (i = 0; i < 3; i++) { + spin_lock_init(&chip->channel[i].lock); + chip->channel[i].idx = i; + } + + /* allocate IRQs */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) { + if (request_irq(snd_sgio2_isr_table[i].irq, + snd_sgio2_isr_table[i].isr, + 0, + snd_sgio2_isr_table[i].desc, + &chip->channel[snd_sgio2_isr_table[i].idx])) { + snd_sgio2audio_free(chip); + printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n", + snd_sgio2_isr_table[i].irq); + return -EBUSY; + } + } + + /* reset the interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + msleep_interruptible(1); /* give time to recover */ + + /* set ring base */ + writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase); + + /* attach the AD1843 codec */ + chip->ad1843.read = read_ad1843_reg; + chip->ad1843.write = write_ad1843_reg; + chip->ad1843.chip = chip; + + /* initialize the AD1843 codec */ + err = ad1843_init(&chip->ad1843); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + *rchip = chip; + return 0; +} + +static int __devinit snd_sgio2audio_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_sgio2audio *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = snd_sgio2audio_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = snd_sgio2audio_new_pcm(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sgio2audio_new_mixer(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI O2 Audio"); + strcpy(card->shortname, "SGI O2 Audio"); + sprintf(card->longname, "%s irq %i-%i", + card->shortname, + MACEISA_AUDIO1_DMAT_IRQ, + MACEISA_AUDIO3_MERR_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit snd_sgio2audio_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver sgio2audio_driver = { + .probe = snd_sgio2audio_probe, + .remove = __devexit_p(snd_sgio2audio_remove), + .driver = { + .name = "sgio2audio", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_sgio2audio_init(void) +{ + return platform_driver_register(&sgio2audio_driver); +} + +static void __exit alsa_card_sgio2audio_exit(void) +{ + platform_driver_unregister(&sgio2audio_driver); +} + +module_init(alsa_card_sgio2audio_init) +module_exit(alsa_card_sgio2audio_exit)

1 0

[alsa-devel] read function
by Gustavo da Silva Serra 11 Jul '08

11 Jul '08

Hi, I am trying to understand how aloop interacts with alsa. I understand that the interface between alsa and a driver is an audio buffer and pointers, offsets inside this buffer. However, I can't discover where these pointers are being used. In pcm_lib.c I inserted some debug messages in read/write functions, but they are never called. My question is: how is that the snd_pcm_readi, for example, from libasound reaches the driver. Thanks.

1 0

[alsa-devel] RME madiface
by D. Sen 11 Jul '08

11 Jul '08

Hello, I have been trying to use the hdspm driver with a recently purchased RME Madiface device. The functionality that I require is to play various files through various channels under program (script) control. I am happy to report that it *almost* works - in that I can use my simple userspace programs to play files. However, there are subtle problems - in that the external MADI A-to-D (an Euphonix 703) does not seem to lock into the sampling rate (something that works under windows). Its very close to achieving full functionality (for my needs) - but just that tiny bit away from being fully useable. Any ideas? Havent had much luck contacting RME and Winfried Ritsch (who wrote/maintains the code). Information on the the hardware and/or windows code would help to figure out how to modify the driver (which was originally written for the rme 9652/aes32 devices). Regards, D. Sen

2 4

[alsa-devel] [PATCH] ALSA driver for SGI O2 audio board
by Thomas Bogendoerfer 11 Jul '08

11 Jul '08

This patch adds a new ALSA driver for the audio device found inside most of the SGI O2 workstation. The hardware uses a SGI custom chip, which feeds a AD codec chip. Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> --- Please apply for 2.6.27. include/sound/ad1843.h | 46 +++ sound/mips/Kconfig | 6 + sound/mips/Makefile | 2 + sound/mips/ad1843.c | 539 +++++++++++++++++++++++++ sound/mips/sgio2audio.c | 1013 +++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 1606 insertions(+), 0 deletions(-) diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h new file mode 100644 index 0000000..b236a9d --- /dev/null +++ b/include/sound/ad1843.h @@ -0,0 +1,46 @@ +/* + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)franken.de> + */ + +#ifndef __SOUND_AD1843_H +#define __SOUND_AD1843_H + +struct snd_ad1843 { + void *chip; + int (*read)(void *chip, int reg); + int (*write)(void *chip, int reg, int val); +}; + +#define AD1843_GAIN_RECLEV 0 +#define AD1843_GAIN_LINE 1 +#define AD1843_GAIN_LINE_2 2 +#define AD1843_GAIN_MIC 3 +#define AD1843_GAIN_PCM_0 4 +#define AD1843_GAIN_PCM_1 5 +#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1) + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id); +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id); +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval); +int ad1843_get_recsrc(struct snd_ad1843 *ad1843); +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc); +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, + unsigned int id); +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843); +int ad1843_init(struct snd_ad1843 *ad1843); + +#endif /* __SOUND_AD1843_H */ diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 531f8ba..a3e202e 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -11,5 +11,11 @@ config SND_AU1X00 help ALSA Sound driver for the Au1x00's AC97 port. +config SND_SGI_O2 + tristate "SGI O2 Audio" + depends on SND && SGI_IP32 + help + Sound support for the SGI O2 Workstation. + endmenu diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 47afed9..55624d8 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -2,7 +2,9 @@ # Makefile for ALSA # +snd-sgi-o2-objs := sgio2audio.o ad1843.o snd-au1x00-objs := au1x00.o # Toplevel Module Dependency obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o +obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c new file mode 100644 index 0000000..76bdf68 --- /dev/null +++ b/sound/mips/ad1843.c @@ -0,0 +1,539 @@ +/* + * AD1843 low level driver + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> + * + * inspired from vwsnd.c (SGI VW audio driver) + * Copyright 1999 Silicon Graphics, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/sched.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ad1843.h> + +/* + * AD1843 bitfield definitions. All are named as in the AD1843 data + * sheet, with ad1843_ prepended and individual bit numbers removed. + * + * E.g., bits LSS0 through LSS2 become ad1843_LSS. + * + * Only the bitfields we need are defined. + */ + +struct ad1843_bitfield { + char reg; + char lo_bit; + char nbits; +}; + +static const struct ad1843_bitfield + ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */ + ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */ + ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */ + ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */ + ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */ + ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */ + ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */ + ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */ + ad1843_RD2M = { 3, 0, 5 }, /* Right DAC 2 Mix Gain/Atten */ + ad1843_RD2MM = { 3, 7, 1 }, /* Right DAC 2 Mix Mute */ + ad1843_LD2M = { 3, 8, 5 }, /* Left DAC 2 Mix Gain/Atten */ + ad1843_LD2MM = { 3, 15, 1 }, /* Left DAC 2 Mix Mute */ + ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */ + ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */ + ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */ + ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */ + ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */ + ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */ + ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */ + ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */ + ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */ + ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */ + ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */ + ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */ + ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */ + ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */ + ad1843_MPOM = { 8, 6, 1 }, /* Mono Output Mute */ + ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */ + ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */ + ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */ + ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */ + ad1843_RDA2G = { 10, 0, 6 }, /* Right DAC2 Analog/Digital Gain */ + ad1843_RDA2GM = { 10, 7, 1 }, /* Right DAC2 Analog Mute */ + ad1843_LDA2G = { 10, 8, 6 }, /* Left DAC2 Analog/Digital Gain */ + ad1843_LDA2GM = { 10, 15, 1 }, /* Left DAC2 Analog Mute */ + ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */ + ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */ + ad1843_RDA2AM = { 12, 7, 1 }, /* Right DAC2 Digital Mute */ + ad1843_LDA2AM = { 12, 15, 1 }, /* Left DAC2 Digital Mute */ + ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */ + ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */ + ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */ + ad1843_DA2C = { 15, 10, 2 }, /* DAC2 Sample Rate Source */ + ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */ + ad1843_C2C = { 20, 0, 16 }, /* Clock 2 Sample Rate Select */ + ad1843_C3C = { 23, 0, 16 }, /* Clock 3 Sample Rate Select */ + ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */ + ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */ + ad1843_DAMIX = { 25, 14, 1 }, /* DAC Digital Mix Enable */ + ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */ + ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */ + ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */ + ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */ + ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */ + ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */ + ad1843_DA2F = { 26, 10, 2 }, /* DAC2 Data Format Select */ + ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */ + ad1843_DA2SM = { 26, 15, 1 }, /* DAC2 Stereo/Mono Mode Select */ + ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */ + ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */ + ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */ + ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */ + ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */ + ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */ + ad1843_DDMEN = { 27, 12, 1 }, /* DAC2 to DAC1 Mix Enable */ + ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */ + ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */ + ad1843_C3EN = { 28, 13, 1 }, /* Clock Generator 3 Enable */ + ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */ + +/* + * The various registers of the AD1843 use three different formats for + * specifying gain. The ad1843_gain structure parameterizes the + * formats. + */ + +struct ad1843_gain { + int negative; /* nonzero if gain is negative. */ + const struct ad1843_bitfield *lfield; + const struct ad1843_bitfield *rfield; + const struct ad1843_bitfield *lmute; + const struct ad1843_bitfield *rmute; +}; + +const struct ad1843_gain ad1843_gain_RECLEV = { + 0, &ad1843_LIG, &ad1843_RIG +}; +const struct ad1843_gain ad1843_gain_LINE = { + 1, &ad1843_LX1M, &ad1843_RX1M, &ad1843_LX1MM, &ad1843_RX1MM +}; +const struct ad1843_gain ad1843_gain_LINE_2 = { + 1, &ad1843_LDA2G, &ad1843_RDA2G, &ad1843_LDA2GM, &ad1843_RDA2GM +}; +const struct ad1843_gain ad1843_gain_MIC = { + 1, &ad1843_LMCM, &ad1843_RMCM, &ad1843_LMCMM, &ad1843_RMCMM +}; +const struct ad1843_gain ad1843_gain_PCM_0 = { + 1, &ad1843_LDA1G, &ad1843_RDA1G, &ad1843_LDA1GM, &ad1843_RDA1GM +}; +const struct ad1843_gain ad1843_gain_PCM_1 = { + 1, &ad1843_LD2M, &ad1843_RD2M, &ad1843_LD2MM, &ad1843_RD2MM +}; + +const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] = +{ + &ad1843_gain_RECLEV, + &ad1843_gain_LINE, + &ad1843_gain_LINE_2, + &ad1843_gain_MIC, + &ad1843_gain_PCM_0, + &ad1843_gain_PCM_1, +}; + +/* read the current value of an AD1843 bitfield. */ + +static int ad1843_read_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field) +{ + int w; + + w = ad1843->read(ad1843->chip, field->reg); + return w >> field->lo_bit & ((1 << field->nbits) - 1); +} + +/* + * write a new value to an AD1843 bitfield and return the old value. + */ + +static int ad1843_write_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field, + int newval) +{ + int w, mask, oldval, newbits; + + w = ad1843->read(ad1843->chip, field->reg); + mask = ((1 << field->nbits) - 1) << field->lo_bit; + oldval = (w & mask) >> field->lo_bit; + newbits = (newval << field->lo_bit) & mask; + w = (w & ~mask) | newbits; + ad1843->write(ad1843->chip, field->reg, w); + + return oldval; +} + +/* + * ad1843_read_multi reads multiple bitfields from the same AD1843 + * register. It uses a single read cycle to do it. (Reading the + * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20 + * microseconds.) + * + * Called like this. + * + * ad1843_read_multi(ad1843, nfields, + * &ad1843_FIELD1, &val1, + * &ad1843_FIELD2, &val2, ...); + */ + +static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + const struct ad1843_bitfield *fp; + int w = 0, mask, *value, reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int *); + if (reg == -1) { + reg = fp->reg; + w = ad1843->read(ad1843->chip, reg); + } + + mask = (1 << fp->nbits) - 1; + *value = w >> fp->lo_bit & mask; + } + va_end(ap); +} + +/* + * ad1843_write_multi stores multiple bitfields into the same AD1843 + * register. It uses one read and one write cycle to do it. + * + * Called like this. + * + * ad1843_write_multi(ad1843, nfields, + * &ad1843_FIELD1, val1, + * &ad1843_FIELF2, val2, ...); + */ + +static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + int reg; + const struct ad1843_bitfield *fp; + int value; + int w, m, mask, bits; + + mask = 0; + bits = 0; + reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int); + if (reg == -1) + reg = fp->reg; + else + BUG_ON(reg != fp->reg); + m = ((1 << fp->nbits) - 1) << fp->lo_bit; + mask |= m; + bits |= (value << fp->lo_bit) & m; + } + va_end(ap); + + if (~mask & 0xFFFF) + w = ad1843->read(ad1843->chip, reg); + else + w = 0; + w = (w & ~mask) | bits; + ad1843->write(ad1843->chip, reg, w); +} + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + int ret; + + ret = (1 << gp->lfield->nbits); + if (!gp->lmute) + ret -= 1; + return ret; +} + +/* + * ad1843_get_gain reads the specified register and extracts the gain value + * using the supplied gain type. + */ + +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id) +{ + int lg, rg, lm, rm; + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg); + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) { + ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm); + if (lm) + lg = 0; + if (rm) + rg = 0; + } + return lg << 0 | rg << 8; +} + +/* + * Set an audio channel's gain. + * + * Returns the new gain, which may be lower than the old gain. + */ + +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + int lg = (newval >> 0) & mask; + int rg = (newval >> 8) & mask; + int lm = (lg == 0) ? 1 : 0; + int rm = (rg == 0) ? 1 : 0; + + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) + ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm); + ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg); + return ad1843_get_gain(ad1843, id); +} + +/* Returns the current recording source */ + +int ad1843_get_recsrc(struct snd_ad1843 *ad1843) +{ + int val = ad1843_read_bits(ad1843, &ad1843_LSS); + + if (val < 0 || val > 2) { + val = 2; + ad1843_write_multi(ad1843, 2, + &ad1843_LSS, val, &ad1843_RSS, val); + } + return val; +} + +/* + * Set recording source. + * + * Returns newsrc on success, -errno on failure. + */ + +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc) +{ + if (newsrc < 0 || newsrc > 2) + return -EINVAL; + + ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc); + return newsrc; +} + +/* Setup ad1843 for D/A conversion. */ + +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int ad_fmt = 0, ad_mode = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_U8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + ad_fmt = 1; + break; + case SNDRV_PCM_FORMAT_MU_LAW: + ad_fmt = 2; + break; + case SNDRV_PCM_FORMAT_A_LAW: + ad_fmt = 3; + break; + default: + break; + } + + switch (channels) { + case 2: + ad_mode = 0; + break; + case 1: + ad_mode = 1; + break; + default: + break; + } + + if (id) { + ad1843_write_bits(ad1843, &ad1843_C2C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA2SM, ad_mode, + &ad1843_DA2F, ad_fmt); + } else { + ad1843_write_bits(ad1843, &ad1843_C1C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA1SM, ad_mode, + &ad1843_DA1F, ad_fmt); + } +} + +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id) +{ + if (id) + ad1843_write_bits(ad1843, &ad1843_DA2F, 1); + else + ad1843_write_bits(ad1843, &ad1843_DA1F, 1); +} + +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int da_fmt = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_U8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_S16_LE: da_fmt = 1; break; + case SNDRV_PCM_FORMAT_MU_LAW: da_fmt = 2; break; + case SNDRV_PCM_FORMAT_A_LAW: da_fmt = 3; break; + default: break; + } + + ad1843_write_bits(ad1843, &ad1843_C3C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt); +} + +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843) +{ + /* nothing to do */ +} + +/* + * Fully initialize the ad1843. As described in the AD1843 data + * sheet, section "START-UP SEQUENCE". The numbered comments are + * subsection headings from the data sheet. See the data sheet, pages + * 52-54, for more info. + * + * return 0 on success, -errno on failure. */ + +int ad1843_init(struct snd_ad1843 *ad1843) +{ + unsigned long later; + + if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) { + printk(KERN_ERR "ad1843: AD1843 won't initialize\n"); + return -EIO; + } + + ad1843_write_bits(ad1843, &ad1843_SCF, 1); + + /* 4. Put the conversion resources into standby. */ + ad1843_write_bits(ad1843, &ad1843_PDNI, 0); + later = jiffies + HZ / 2; /* roughly half a second */ + + while (ad1843_read_bits(ad1843, &ad1843_PDNO)) { + if (time_after(jiffies, later)) { + printk(KERN_ERR + "ad1843: AD1843 won't power up\n"); + return -EIO; + } + schedule(); + } + + /* 5. Power up the clock generators and enable clock output pins. */ + ad1843_write_multi(ad1843, 3, + &ad1843_C1EN, 1, + &ad1843_C2EN, 1, + &ad1843_C3EN, 1); + + /* 6. Configure conversion resources while they are in standby. */ + + /* DAC1/2 use clock 1/2 as source, ADC uses clock 3. Always. */ + ad1843_write_multi(ad1843, 4, + &ad1843_DA1C, 1, + &ad1843_DA2C, 2, + &ad1843_ADLC, 3, + &ad1843_ADRC, 3); + + /* 7. Enable conversion resources. */ + ad1843_write_bits(ad1843, &ad1843_ADTLK, 1); + ad1843_write_multi(ad1843, 7, + &ad1843_ANAEN, 1, + &ad1843_AAMEN, 1, + &ad1843_DA1EN, 1, + &ad1843_DA2EN, 1, + &ad1843_DDMEN, 1, + &ad1843_ADLEN, 1, + &ad1843_ADREN, 1); + + /* 8. Configure conversion resources while they are enabled. */ + + /* set gain to 0 for all channels */ + ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0); + + /* Unmute all channels. */ + /* DAC1 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0); + /* DAC2 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0); + + /* Set default recording source to Line In and set + * mic gain to +20 dB. + */ + ad1843_set_recsrc(ad1843, 2); + ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1); + + /* Set Speaker Out level to +/- 4V and unmute it. */ + ad1843_write_multi(ad1843, 3, + &ad1843_HPOS, 1, + &ad1843_HPOM, 0, + &ad1843_MPOM, 0); + + return 0; +} diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c new file mode 100644 index 0000000..6630af3 --- /dev/null +++ b/sound/mips/sgio2audio.c @@ -0,0 +1,1013 @@ +/* + * Sound driver for Silicon Graphics O2 Workstations A/V board audio. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> + * Mxier part taken from mace_audio.c: + * Copyright 2007 Thorben Jändling <tj.trevelyan(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/spinlock.h> +#include <linux/gfp.h> +#include <linux/vmalloc.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <asm/ip32/ip32_ints.h> +#include <asm/ip32/mace.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#define SNDRV_GET_ID +#include <sound/initval.h> +#include <sound/ad1843.h> + + +MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier(a)linux-mips.org>"); +MODULE_DESCRIPTION("SGI O2 Audio"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard."); + + +#define SGIO2AUDIO_MAX_VOLUME 31 + +#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */ +#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */ + +#define CODEC_CONTROL_WORD_SHIFT 0 +#define CODEC_CONTROL_READ BIT(16) +#define CODEC_CONTROL_ADDRESS_SHIFT 17 + +#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */ +#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */ +#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */ +#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */ +#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */ +#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */ +#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */ +#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */ + +#define CHANNEL_RING_SHIFT 12 +#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT) +#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1) + +#define CHANNEL_LEFT_SHIFT 40 +#define CHANNEL_RIGHT_SHIFT 8 + +struct snd_sgio2audio_chan { + int idx; + struct snd_pcm_substream *substream; + int pos; + snd_pcm_uframes_t size; + spinlock_t lock; +}; + +/* definition of the chip-specific record */ +struct snd_sgio2audio { + struct snd_card *card; + + /* codec */ + struct snd_ad1843 ad1843; + spinlock_t ad1843_lock; + + /* channels */ + struct snd_sgio2audio_chan channel[3]; + + /* properties */ + int volume; + + /* resources */ + void *ring_base; + dma_addr_t ring_base_dma; +}; + +/* AD1843 access */ + +/* + * read_ad1843_reg returns the current contents of a 16 bit AD1843 register. + * + * Returns unsigned register value on success, -errno on failure. + */ +static int read_ad1843_reg(void *priv, int reg) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + CODEC_CONTROL_READ, &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + val = readq(&mace->perif.audio.codec_read); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return val; +} + +/* + * write_ad1843_reg writes the specified value to a 16 bit AD1843 register. + */ +static int write_ad1843_reg(void *priv, int reg, int word) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + (word << CODEC_CONTROL_WORD_SHIFT), + &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return 0; +} + +static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843, + (int)kcontrol->private_value); + return 0; +} + +static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int vol; + + vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value); + + ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF; + ucontrol->value.integer.value[1] = vol & 0xFF; + + return 0; +} + +static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newvol, oldvol; + + oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value); + newvol = (ucontrol->value.integer.value[0] << 8) | + ucontrol->value.integer.value[1]; + + newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value, + newvol); + + return newvol != oldvol; +} + +static int sgio2audio_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "Cam Mic", "Mic", "Line" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int sgio2audio_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843); + return 0; +} + +static int sgio2audio_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newsrc, oldsrc; + + oldsrc = ad1843_get_recsrc(&chip->ad1843); + newsrc = ad1843_set_recsrc(&chip->ad1843, + ucontrol->value.enumerated.item[0]); + + return newsrc != oldsrc; +} + +/* dac1/pcm0 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_0, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* dac2/pcm1 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_1, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_RECLEV, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level source control */ +static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = sgio2audio_source_info, + .get = sgio2audio_source_get, + .put = sgio2audio_source_put, +}; + +/* line mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* cd mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE_2, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* mic mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_MIC, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + + +static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip) +{ + int err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_recsource, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_line, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_cd, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_mic, chip)); + if (err < 0) + return err; + + return 0; +} + +/* low-level audio interface DMA */ + +/* get data out of bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + unsigned long src_base, src_pos, dst_mask; + unsigned char *dst_base; + int dst_pos; + u64 *src; + s16 *dst; + u64 x; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT); + src_pos = readq(&mace->perif.audio.chan[ch].read_ptr); + dst_base = runtime->dma_area; + dst_pos = chip->channel[ch].pos; + dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (u64 *)(src_base + src_pos); + dst = (s16 *)(dst_base + dst_pos); + + x = *src; + dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff; + dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff; + + src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK; + dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask; + count -= sizeof(u64); + } + + writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */ + chip->channel[ch].pos = dst_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +/* put some DMA data in bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + s64 l, r; + unsigned long dst_base, dst_pos, src_mask; + unsigned char *src_base; + int src_pos; + u64 *dst; + s16 *src; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT); + dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr); + src_base = runtime->dma_area; + src_pos = chip->channel[ch].pos; + src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (s16 *)(src_base + src_pos); + dst = (u64 *)(dst_base + dst_pos); + + l = src[0]; /* sign extend */ + r = src[1]; /* sign extend */ + + *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) | + ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT); + + dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK; + src_pos = (src_pos + 2 * sizeof(s16)) & src_mask; + count -= sizeof(u64); + } + + writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */ + chip->channel[ch].pos = src_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + + /* reset DMA channel */ + writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control); + udelay(10); + writeq(0, &mace->perif.audio.chan[ch].control); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* push a full buffer */ + snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32); + } + /* set DMA to wake on 50% empty and enable interrupt */ + writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50, + &mace->perif.audio.chan[ch].control); + return 0; +} + +static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + writeq(0, &mace->perif.audio.chan[chan->idx].control); + return 0; +} + +static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + + /* empty the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_pull_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + /* fill the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_push_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + + substream = chan->substream; + snd_sgio2audio_dma_stop(substream); + snd_sgio2audio_dma_start(substream); + return IRQ_HANDLED; +} + +/* PCM part */ +/* PCM hardware definition */ +static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 32768, + .period_bytes_max = 65536, + .periods_min = 1, + .periods_max = 1024, +}; + +/* PCM playback open callback */ +static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[1]; + return 0; +} + +static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[2]; + return 0; +} + +/* PCM capture open callback */ +static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[0]; + return 0; +} + +/* PCM close callback */ +static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->private_data = NULL; + return 0; +} + + +/* hw_params callback */ +static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int size = params_buffer_bytes(hw_params); + + /* alloc virtual 'dma' area */ + if (runtime->dma_area) + vfree(runtime->dma_area); + runtime->dma_area = vmalloc(size); + if (runtime->dma_area == NULL) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* hw_free callback */ +static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) +{ + if (substream->runtime->dma_area) + vfree(substream->runtime->dma_area); + substream->runtime->dma_area = NULL; + return 0; +} + +/* prepare callback */ +static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + unsigned long flags; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + /* Setup the pseudo-dma transfer pointers. */ + chip->channel[ch].pos = 0; + chip->channel[ch].size = 0; + chip->channel[ch].substream = substream; + + /* set AD1843 format */ + /* hardware format is always S16_LE */ + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + ad1843_setup_dac(&chip->ad1843, + ch - 1, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + case SNDRV_PCM_STREAM_CAPTURE: + ad1843_setup_adc(&chip->ad1843, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + } + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return 0; +} + +/* trigger callback */ +static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* start the PCM engine */ + snd_sgio2audio_dma_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* stop the PCM engine */ + snd_sgio2audio_dma_stop(substream); + break; + default: + return -EINVAL; + } + return 0; +} + +/* pointer callback */ +static snd_pcm_uframes_t +snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + /* get the current hardware pointer */ + return bytes_to_frames(substream->runtime, + chip->channel[chan->idx].pos); +} + +/* get the physical page pointer on the given offset */ +static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} + +/* operators */ +static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { + .open = snd_sgio2audio_playback1_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { + .open = snd_sgio2audio_playback2_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_capture_ops = { + .open = snd_sgio2audio_capture_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +/* + * definitions of capture are omitted here... + */ + +/* create a pcm device */ +static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC1"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback1_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_sgio2audio_capture_ops); + + /* create second pcm device with one outputs and no input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback2_ops); + + return 0; +} + +static struct { + int idx; + int irq; + irqreturn_t (*isr)(int, void *); + const char *desc; +} snd_sgio2_isr_table[] = { + { + .idx = 0, + .irq = MACEISA_AUDIO1_DMAT_IRQ, + .isr = snd_sgio2audio_dma_in_isr, + .desc = "Capture DMA Channel 0" + }, { + .idx = 0, + .irq = MACEISA_AUDIO1_OF_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Capture Overflow" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 1" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 1" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 2" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 2" + } +}; + +/* ALSA driver */ + +static int snd_sgio2audio_free(struct snd_sgio2audio *chip) +{ + int i; + + /* reset interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + + /* release IRQ's */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) + free_irq(snd_sgio2_isr_table[i].irq, + &chip->channel[snd_sgio2_isr_table[i].idx]); + + dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + chip->ring_base, chip->ring_base_dma); + + /* release card data */ + kfree(chip); + return 0; +} + +static int snd_sgio2audio_dev_free(struct snd_device *device) +{ + struct snd_sgio2audio *chip = device->device_data; + + return snd_sgio2audio_free(chip); +} + +static struct snd_device_ops ops = { + .dev_free = snd_sgio2audio_dev_free, +}; + +static int __devinit snd_sgio2audio_create(struct snd_card *card, + struct snd_sgio2audio **rchip) +{ + struct snd_sgio2audio *chip; + int i, err; + + *rchip = NULL; + + /* check if a codec is attached to the interface */ + /* (Audio or Audio/Video board present) */ + if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT)) + return -ENOENT; + + chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + &chip->ring_base_dma, GFP_USER); + if (chip->ring_base == NULL) { + printk(KERN_ERR + "sgio2audio: could not allocate ring buffers\n"); + kfree(chip); + return -ENOMEM; + } + + spin_lock_init(&chip->ad1843_lock); + + chip->volume = SGIO2AUDIO_MAX_VOLUME; + + /* initialize channels */ + for (i = 0; i < 3; i++) { + spin_lock_init(&chip->channel[i].lock); + chip->channel[i].idx = i; + } + + /* allocate IRQs */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) { + if (request_irq(snd_sgio2_isr_table[i].irq, + snd_sgio2_isr_table[i].isr, + IRQF_SHARED, + snd_sgio2_isr_table[i].desc, + &chip->channel[snd_sgio2_isr_table[i].idx])) { + snd_sgio2audio_free(chip); + printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n", + snd_sgio2_isr_table[i].irq); + return -EBUSY; + } + } + + /* reset the interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + udelay(100); /* give time to recover */ + + /* set ring base */ + writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase); + + /* attach the AD1843 codec */ + chip->ad1843.read = read_ad1843_reg; + chip->ad1843.write = write_ad1843_reg; + chip->ad1843.chip = chip; + + /* initialize the AD1843 codec */ + err = ad1843_init(&chip->ad1843); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + *rchip = chip; + return 0; +} + +static int __devinit snd_sgio2audio_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_sgio2audio *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = snd_sgio2audio_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = snd_sgio2audio_new_pcm(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sgio2audio_new_mixer(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI O2 Audio"); + strcpy(card->shortname, "SGI O2 Audio"); + sprintf(card->longname, "%s irq %i-%i", + card->shortname, + MACEISA_AUDIO1_DMAT_IRQ, + MACEISA_AUDIO3_MERR_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit snd_sgio2audio_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver sgio2audio_driver = { + .probe = snd_sgio2audio_probe, + .remove = __devexit_p(snd_sgio2audio_remove), + .driver = { + .name = "sgio2audio", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_sgio2audio_init(void) +{ + return platform_driver_register(&sgio2audio_driver); +} + +static void __exit alsa_card_sgio2audio_exit(void) +{ + platform_driver_unregister(&sgio2audio_driver); +} + +module_init(alsa_card_sgio2audio_init) +module_exit(alsa_card_sgio2audio_exit)

3 7

[alsa-devel] Error compiling ALSA driver
by DSP Tech 11 Jul '08

11 Jul '08

Hi All, I am trying to compile ALSA driver for arm platform. But it keeps complaining that "__LINUX_ARM_ARCH__" is undeclared. But the linux kernel is properly built. Do I have to include any specific header file for this? The error message is as follows: ============== /linux-2.6.23//include/linux/calc64.h: In function 'do_div_llr': /linux-2.6.23//include/linux/calc64.h:25: error: '__LINUX_ARM_ARCH__' undeclared (first use in this function)/linux-2.6.23//include/linux/calc64.h:25: error: (Each undeclared identifier is reported only once============== The environmental variable CROSS_COMPILE is set to /extra/arm-2007q1/bin/arm-none-linux-gnueabi- The configuration options provided is: ./configure --target=arm-linux --host=arm-linux --build=i686-pc-linux-gnu --prefix=/modules/sound/alsa-driver-1.0.16/ --exec-prefix=/modules/sound/alsa-driver-1.0.16/ --with-kernel=/linux-2.6.23/ --with-build=/linux-2.6.23/ --with-oss=yes Is ther any mistake in the way I am building? Thanks _________________________________________________________________ Searching for the best deals on travel? Visit MSN Travel. http://msn.coxandkings.co.in/cnk/cnk.do

1 0

[alsa-devel] [PATCH] alsa: correct kcalloc usage
by Milton Miller 11 Jul '08

11 Jul '08

kcalloc is supposed to be called with the count as its first argument and the element size as the second. Signed-off-by: Milton Miller <miltonm(a)bga.com> --- Both arguments are size_t so does not affect correctness. This callsite is during module_init and therefore not performance critical. Another patch will optimize the case when the count is variable but the size is fixed. alsa-devel on bcc due to (subscribers-only) diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 7efb838..06d13e7 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1302,8 +1302,8 @@ snd_nm256_mixer(struct nm256 *chip) .read = snd_nm256_ac97_read, }; - chip->ac97_regs = kcalloc(sizeof(short), - ARRAY_SIZE(nm256_ac97_init_val), GFP_KERNEL); + chip->ac97_regs = kcalloc(ARRAY_SIZE(nm256_ac97_init_val), + sizeof(short), GFP_KERNEL); if (! chip->ac97_regs) return -ENOMEM;

1 0

[alsa-devel] [PATCH] hda: 92hd71bxx PC Beep
by Matthew Ranostay 10 Jul '08

10 Jul '08

Added volume controls for the analog PC Beep on 92hd71bxx codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index c4f3489..42908e6 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -825,6 +825,9 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */

2 1

[alsa-devel] S24_LE is 3 bytes ? in alsa-lib example pcm.c
by Norbert van Bolhuis 10 Jul '08

10 Jul '08

as far as I know: sample format S24_LE is 24bit sample in 4 bytes sample format S24_3LE is 24bit sample in 3 bytes (why else have separate definitions), right ? In the ALSA-LIb example /test/pcm.c however (http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html) there is no diff between the 2 sample formats. I noticed this by simply showing the bytes presented to snd_pcm_writei See: S32_LE (for reference) ---------------------- ./pcm -c 2 -o S32_LE -f 4000 Playback device is plughw:0,0 Stream parameters are 48000Hz, S32_LE, 2 channels Sine wave rate is 4000.0000Hz Using transfer method: write write_loop: going to write: period_size=4096 samples data[00:31]= 00000000 00000000 ffffff3f ffffff3f a0ebd96e a0ebd96e ffffff7f ffffff7f data[32:63]= a0ebd96e a0ebd96e ffffff3f ffffff3f 00000000 00000000 010000c0 010000c0 S24_3LE (as expected) ---------------------- ./pcm -c 2 -o S24_3LE -f 4000 Playback device is plughw:0,0 Stream parameters are 48000Hz, S24_3LE, 2 channels Sine wave rate is 4000.0000Hz Using transfer method: write write_loop: going to write: period_size=4096 samples data[00:31]= 00000000 0000ffff 3fffff3f ead96eea d96effff 7fffff7f ead96eea d96effff data[32:63]= 3fffff3f 00000000 00000100 c00100c0 16269116 26910100 80010080 16269116 S24_LE ---------------------- ./pcm -c 2 -o S24_LE -f 4000 Playback device is plughw:0,0 Stream parameters are 48000Hz, S24_LE, 2 channels Sine wave rate is 4000.0000Hz Using transfer method: write write_loop: going to write: period_size=4096 samples data[00:31]= 00000000 0000ffff 3fffff3f ead96eea d96effff 7fffff7f ead96eea d96effff data[32:63]= 3fffff3f 00000000 00000100 c00100c0 16269116 26910100 80010080 16269116 I would expect to see data[00:31]= 0x00000000 00000000 00ffff3f 00ffff3f 00ebd96e 00ebd96e 00ffff7f 00ffff7f data[32:63]= 0x00ebd96e 00ebd96e 00ffff3f 00ffff3f 00000000 00000000 000000c0 000000c0 for S24_LE do I miss something ? is this is a fault in /test/pcm.c ? -- This message has been scanned for viruses and is believed to be clean

4 5

[alsa-devel] [PATCH - asihpi 1/4] V3.10.1. Add hpi_RDS enum.
by linux＠audioscience.com 10 Jul '08

10 Jul '08

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> diff --git a/pci/asihpi/hpi.h b/pci/asihpi/hpi.h index a58bf71..06953bc 100644 --- a/pci/asihpi/hpi.h +++ b/pci/asihpi/hpi.h @@ -41,12 +41,12 @@ i.e 3.05.02 is a development version #define HPI_VERSION_CONSTRUCTOR(maj, min, rel) \ ((maj << 16) + (min << 8) + rel) -#define HPI_VER_MAJOR(v) (int)(v >> 16) -#define HPI_VER_MINOR(v) (int)((v >> 8) & 0xFF) -#define HPI_VER_RELEASE(v) (int)(v & 0xFF) +#define HPI_VER_MAJOR(v) ((int)(v >> 16)) +#define HPI_VER_MINOR(v) ((int)((v >> 8) & 0xFF)) +#define HPI_VER_RELEASE(v) ((int)(v & 0xFF)) /* Use single digits for versions less that 10 to avoid octal. */ -#define HPI_VER HPI_VERSION_CONSTRUCTOR(3L, 10, 0) +#define HPI_VER HPI_VERSION_CONSTRUCTOR(3L, 10, 1) #ifdef _DOX_ONLY_ /*****************************************************************************/ @@ -1052,6 +1052,13 @@ Contains either 1 or 0. */ Contains either 1 or 0. */ #define HPI_PAD_TA_ACTIVE HPI_CTL_ATTR(PAD, 8) +/** Data types for PTY string translation. + */ +enum eHPI_RDS_type { + HPI_RDS_DATATYPE_RDS = 0, /**< RDS bitstream. */ + HPI_RDS_DATATYPE_RDBS = 1, /**< RDBS bitstream. */ + HPI_RDS_DATATYPE_FUTURE = 2 /**< Future bitstream. */ +}; /** \} */ /** \defgroup tuner_bands Tuner bands -- 1.5.4.3

3 5

[alsa-devel] ALSA Problem - Broken Pipe
by Kristin Low 10 Jul '08

10 Jul '08

Hi all, I apologize if this isn't the right place to ask for help, but no one who I've asked knows how to help and I'm somewhat new to Linux. I'm running Ubuntu Hardy Heron and haven't changed any config files (practically a fresh install) however ALSA does not work properly and I assume that because of this, Pulseaudio doesn't work either (it breaks when ALSA does). Playing a song, I can hear the song for a random length of time and then it cuts out. If I run VLC from a terminal, it claims "alsa output error: write error (broken pipe)" as the cause and continues to write that over the screen until I force VLC to quit. If I restart the song again, the same thing occurs. Pulseaudio cuts out and says it gets a POLLERR from ALSA. When I login, only part of the login sounds play and are cut just the same as any songs or audio files I try. It seems like my sound card is detected fine, and OSS works when I select to use it instead. It's on a laptop, a Toshiba Satellite a70. Below I've listed the alsa-info.txt as per https://wiki.ubuntu.com/DebuggingSoundProblems. Thanks, Kristin name=kristin&type=33&description=/tmp/alsa-info.txt&expiry=&s=Submit+Post&content= !!################################ !!ALSA Information Script v 0.4.48 !!################################ !!Script ran on: Tue Jul 8 16:13:38 EDT 2008 !!Linux Distribution !!------------------ Ubuntu 8.04.1 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 8.04.1" !!Kernel Information !!------------------ Kernel release: 2.6.24-19-generic Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.16 Library version: Utilities version: 1.0.15 !!Loaded ALSA modules !!------------------- snd_atiixp snd_atiixp_modem !!Soundcards recognised by ALSA !!----------------------------- 0 [IXP ]: ATIIXP - ATI IXP ATI IXP rev 0 with ALC250 at 0xd0004400, irq 17 1 [Modem ]: ATIIXP-MODEM - ATI IXP Modem ATI IXP Modem rev 1 at 0xd0004800, irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:14.5 Multimedia audio controller: ATI Technologies Inc IXP150 AC'97 Audio Controller 02:02.0 Ethernet controller: Atheros Communications Inc. AR5212/AR5213 Multiprotocol MAC/baseband processor (rev 01) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:14.5 0401: 1002:4341 Subsystem: 1179:ff01 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 !!Loaded sound module options !!-------------------------- !!Module: snd_atiixp ac97_clock : 48000 ac97_codec : -1 ac97_quirk : <NULL> enable : N id : <NULL> index : -1 spdif_aclink : Y !!Module: snd_atiixp_modem ac97_clock : 48000 enable : N id : <NULL> index : -2 !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: Realtek ALC250 rev 2 PCI Subsys Vendor: 0x1179 PCI Subsys Device: 0xff01 Revision : 0x00 Compat. Class : 0x00 Subsys. Vendor ID: 0xffff Subsys. ID : 0xffff Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 18-bit 3D enhancement : No 3D Stereo Enhancement Current setup Mic gain : +0dB [+0dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Double rate slots: 10/11 Extended ID : codec=0 rev=2 AMAP DSA=0 SPDIF DRA VRA Extended status : SPDIF=3/4 VRA PCM front DAC : 44100Hz PCM ADC : 44100Hz SPDIF Control : Consumer PCM Category=0x2 Generation=1 Rate=48kHz Gain Inverted Buffer delay Location Master Out : 0.0 dBV - 23/fs Rear I/O Panel AUX Out : 0.0 dBV - 23/fs Rear I/O Panel Center/LFE Out : 0.0 dBV - 23/fs Rear I/O Panel SPDIF Out : 0.0 dBV - 23/fs Rear I/O Panel Phone In : 0.0 dBV - 23/fs Rear I/O Panel Mic 1 : 0.0 dBV - 23/fs Rear I/O Panel Mic 2 : 0.0 dBV - 23/fs Rear I/O Panel Line In : 0.0 dBV - 23/fs Rear I/O Panel CD In : 0.0 dBV - 23/fs Rear I/O Panel Video In : 0.0 dBV - 23/fs Rear I/O Panel Aux In : 0.0 dBV - 23/fs Rear I/O Panel Mono Out : 0.0 dBV - 23/fs Rear I/O Panel 0:00 = 0190 0:02 = 0000 0:04 = 1212 0:06 = 0008 0:08 = 0000 0:0a = 801e 0:0c = 801f 0:0e = 831f 0:10 = 9f1f 0:12 = 0505 0:14 = 0000 0:16 = 9f1f 0:18 = 0606 0:1a = 0000 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 000f 0:28 = 0a07 0:2a = 0001 0:2c = ac44 0:2e = 0000 0:30 = 0000 0:32 = ac44 0:34 = bb80 0:36 = 0000 0:38 = 0000 0:3a = 2824 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0aea 0:6a = 4400 0:6c = 4601 0:6e = 0015 0:70 = 0008 0:72 = 0000 0:74 = 0100 0:76 = 0304 0:78 = 0c03 0:7a = 6002 0:7c = 414c 0:7e = 4752 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 2008-07-08 12:08 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 32 2008-07-08 12:08 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 24 2008-07-08 14:11 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 2008-07-08 15:15 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 56 2008-07-08 12:08 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 48 2008-07-08 12:08 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 1 2008-07-08 12:08 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 2008-07-08 12:08 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: IXP [ATI IXP], device 0: ATI IXP AC97 [ATI IXP AC97] Subdevices: 0/1 Subdevice #0: subdevice #0 card 1: Modem [ATI IXP Modem], device 0: ATI IXP MC97 [ATI IXP MC97] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: IXP [ATI IXP], device 0: ATI IXP AC97 [ATI IXP AC97] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Modem [ATI IXP Modem], device 0: ATI IXP MC97 [ATI IXP MC97] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [IXP] Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 63 Mono: Front Left: Playback 63 [100%] [0.00dB] [on] Front Right: Playback 63 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 23 [74%] [-12.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 63 Mono: Front Left: Playback 45 [71%] [-27.00dB] [on] Front Right: Playback 45 [71%] [-27.00dB] [on] Simple mixer control '3D Control - Switch',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 25 [81%] [3.00dB] [on] Front Right: Playback 25 [81%] [3.00dB] [on] Simple mixer control 'PCM Out Path & Mute',0 Capabilities: enum Items: 'pre 3D' 'post 3D' Item0: 'pre 3D' Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 26 [84%] [4.50dB] [on] Capture [off] Front Right: Playback 26 [84%] [4.50dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [on] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Playback AC97-SPSA',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 0 [0%] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 0 [0%] [-45.00dB] [off] Simple mixer control 'Aux',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 0 [0%] [0.00dB] [on] Front Right: Capture 0 [0%] [0.00dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!-------Mixer controls for card 1 [Modem] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Modem Speaker',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!------------- --startcollapse-- state.IXP { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 63' iface MIXER name 'Master Playback Volume' value.0 63 value.1 63 } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Headphone Playback Switch' value true } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 63' iface MIXER name 'Headphone Playback Volume' value.0 45 value.1 45 } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Mono Playback Switch' value true } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Master Mono Playback Volume' value 23 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PC Speaker Playback Switch' value false } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' iface MIXER name 'PC Speaker Playback Volume' value 0 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Playback Switch' value false } control.10 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Phone Playback Volume' value 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Playback Switch' value false } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost (+20dB)' value false } control.14 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Playback Switch' value false } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'CD Playback Switch' value true } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'CD Playback Volume' value.0 26 value.1 26 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Aux Playback Switch' value false } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Aux Playback Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PCM Playback Switch' value true } control.21 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'PCM Playback Volume' value.0 25 value.1 25 } control.22 { comment.access 'read write' comment.type ENUMERATED comment.count 2 comment.item.0 Mic comment.item.1 CD comment.item.2 Video comment.item.3 Aux comment.item.4 Line comment.item.5 Mix comment.item.6 'Mix Mono' comment.item.7 Phone iface MIXER name 'Capture Source' value.0 Mic value.1 Mic } control.23 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Capture Switch' value true } control.24 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.25 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'pre 3D' comment.item.1 'post 3D' iface MIXER name 'PCM Out Path & Mute' value 'pre 3D' } control.26 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name '3D Control - Switch' value false } control.27 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic1 } control.28 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.29 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value cf00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.30 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.31 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.32 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 3' iface MIXER name 'IEC958 Playback AC97-SPSA' value 0 } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value true } } state.Modem { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Off-hook Switch' value false } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Caller ID Switch' value false } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' iface MIXER name 'Modem Speaker Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module ipv6 af_packet radeon drm rfcomm l2cap bluetooth ppdev acpi_cpufreq speedstep_lib cpufreq_powersave cpufreq_conservative cpufreq_userspace cpufreq_ondemand cpufreq_stats freq_table dock sbs sbshc iptable_filter ip_tables x_tables sbp2 lp joydev pcmcia wlan_scan_sta ath_rate_sample snd_atiixp parport_pc parport snd_seq_dummy evdev snd_atiixp_modem video output snd_ac97_codec serio_raw snd_seq_oss sdhci battery irda container ac97_bus snd_pcm_oss snd_mixer_oss ac crc_ccitt ath_pci snd_seq_midi mmc_core yenta_socket rsrc_nonstatic pcmcia_core psmouse wlan snd_rawmidi snd_seq_midi_event ath_hal button snd_pcm snd_seq snd_timer snd_seq_device i2c_piix4 ati_agp shpchp pci_hotplug pcspkr i2c_core agpgart snd soundcore snd_page_alloc ext3 jbd mbcache sg usbhid hid sr_mod cdrom sd_mod atiixp ide_core pata_acpi pata_atiixp 8139too ohci1394 ata_generic 8139cp mii ieee1394 libata scsi_mod ehci_hcd ohci_hcd usbcore thermal processor fan fbcon tileblit font bitblit softcursor fuse

2 2

[alsa-devel] [PATCH] Add support for Mitac Mio A701 smartphone
by Robert Jarzmik 10 Jul '08

10 Jul '08

This patch serie aims at integration of the Mitac Mio A701 smartphone's sound system into alsa sound system. It should be noted that this serie applies upon asoc-v2-dev tree, commit id e02f9491e1e6d9720e0caffdb8e5edc259f41946, and should not hit alsa tree before 2.6.27, because A701 official support will be included at that version through arm tree (perhaps even 2.6.28, depending on Russell's time). Happy review. -- Robert

4 11

[alsa-devel] ASoC: AC97 codec PM
by Manuel Lauss 09 Jul '08

09 Jul '08

Hi Liam, This simple patch adds suspend/resume support to the ASoC AC97 codec. Tested on Au1200, works fine for me(TM). Thanks! Manuel Lauss --- From: Manuel Lauss <mano(a)roarinelk.homelinux.net> Simple suspend/resume for AC97 ASoC codec. Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- sound/soc/codecs/ac97.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2a1ffe3..fb9ac9f 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -146,9 +146,34 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_suspend(socdev->codec->ac97); + + return 0; +} + +static int ac97_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_ac97_resume(socdev->codec->ac97); + + return 0; +} +#else +#define ac97_soc_suspend NULL +#define ac97_soc_resume NULL +#endif + struct snd_soc_codec_device soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .remove = ac97_soc_remove, + .suspend = ac97_soc_suspend, + .resume = ac97_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); -- 1.5.6.1

3 19

[alsa-devel] [PATCH v2] ALSA driver for SGI O2 audio board
by Thomas Bogendoerfer 09 Jul '08

09 Jul '08

This patch adds a new ALSA driver for the audio device found inside most of the SGI O2 workstation. The hardware uses a SGI custom chip, which feeds a AD codec chip. Signed-off-by: Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> --- Changes in v2: - removed unused volume field - spreaded some statics - switch over to use C99 field inits - use msleep_interuptible instead of long udelay - use schedule_timeout_interruptible instead of simple schedule include/sound/ad1843.h | 46 +++ sound/mips/Kconfig | 6 + sound/mips/Makefile | 2 + sound/mips/ad1843.c | 561 ++++++++++++++++++++++++++ sound/mips/sgio2audio.c | 1006 +++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 1621 insertions(+), 0 deletions(-) diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h new file mode 100644 index 0000000..b236a9d --- /dev/null +++ b/include/sound/ad1843.h @@ -0,0 +1,46 @@ +/* + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)franken.de> + */ + +#ifndef __SOUND_AD1843_H +#define __SOUND_AD1843_H + +struct snd_ad1843 { + void *chip; + int (*read)(void *chip, int reg); + int (*write)(void *chip, int reg, int val); +}; + +#define AD1843_GAIN_RECLEV 0 +#define AD1843_GAIN_LINE 1 +#define AD1843_GAIN_LINE_2 2 +#define AD1843_GAIN_MIC 3 +#define AD1843_GAIN_PCM_0 4 +#define AD1843_GAIN_PCM_1 5 +#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1) + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id); +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id); +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval); +int ad1843_get_recsrc(struct snd_ad1843 *ad1843); +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc); +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, + unsigned int id); +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels); +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843); +int ad1843_init(struct snd_ad1843 *ad1843); + +#endif /* __SOUND_AD1843_H */ diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 531f8ba..3ce743b 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -11,5 +11,11 @@ config SND_AU1X00 help ALSA Sound driver for the Au1x00's AC97 port. +config SND_SGI_O2 + tristate "SGI O2 Audio" + depends on SGI_IP32 + help + Sound support for the SGI O2 Workstation. + endmenu diff --git a/sound/mips/Makefile b/sound/mips/Makefile index 47afed9..55624d8 100644 --- a/sound/mips/Makefile +++ b/sound/mips/Makefile @@ -2,7 +2,9 @@ # Makefile for ALSA # +snd-sgi-o2-objs := sgio2audio.o ad1843.o snd-au1x00-objs := au1x00.o # Toplevel Module Dependency obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o +obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c new file mode 100644 index 0000000..c624510 --- /dev/null +++ b/sound/mips/ad1843.c @@ -0,0 +1,561 @@ +/* + * AD1843 low level driver + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> + * + * inspired from vwsnd.c (SGI VW audio driver) + * Copyright 1999 Silicon Graphics, Inc. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/sched.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ad1843.h> + +/* + * AD1843 bitfield definitions. All are named as in the AD1843 data + * sheet, with ad1843_ prepended and individual bit numbers removed. + * + * E.g., bits LSS0 through LSS2 become ad1843_LSS. + * + * Only the bitfields we need are defined. + */ + +struct ad1843_bitfield { + char reg; + char lo_bit; + char nbits; +}; + +static const struct ad1843_bitfield + ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */ + ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */ + ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */ + ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */ + ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */ + ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */ + ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */ + ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */ + ad1843_RD2M = { 3, 0, 5 }, /* Right DAC 2 Mix Gain/Atten */ + ad1843_RD2MM = { 3, 7, 1 }, /* Right DAC 2 Mix Mute */ + ad1843_LD2M = { 3, 8, 5 }, /* Left DAC 2 Mix Gain/Atten */ + ad1843_LD2MM = { 3, 15, 1 }, /* Left DAC 2 Mix Mute */ + ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */ + ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */ + ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */ + ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */ + ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */ + ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */ + ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */ + ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */ + ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */ + ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */ + ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */ + ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */ + ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */ + ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */ + ad1843_MPOM = { 8, 6, 1 }, /* Mono Output Mute */ + ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */ + ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */ + ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */ + ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */ + ad1843_RDA2G = { 10, 0, 6 }, /* Right DAC2 Analog/Digital Gain */ + ad1843_RDA2GM = { 10, 7, 1 }, /* Right DAC2 Analog Mute */ + ad1843_LDA2G = { 10, 8, 6 }, /* Left DAC2 Analog/Digital Gain */ + ad1843_LDA2GM = { 10, 15, 1 }, /* Left DAC2 Analog Mute */ + ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */ + ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */ + ad1843_RDA2AM = { 12, 7, 1 }, /* Right DAC2 Digital Mute */ + ad1843_LDA2AM = { 12, 15, 1 }, /* Left DAC2 Digital Mute */ + ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */ + ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */ + ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */ + ad1843_DA2C = { 15, 10, 2 }, /* DAC2 Sample Rate Source */ + ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */ + ad1843_C2C = { 20, 0, 16 }, /* Clock 2 Sample Rate Select */ + ad1843_C3C = { 23, 0, 16 }, /* Clock 3 Sample Rate Select */ + ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */ + ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */ + ad1843_DAMIX = { 25, 14, 1 }, /* DAC Digital Mix Enable */ + ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */ + ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */ + ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */ + ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */ + ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */ + ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */ + ad1843_DA2F = { 26, 10, 2 }, /* DAC2 Data Format Select */ + ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */ + ad1843_DA2SM = { 26, 15, 1 }, /* DAC2 Stereo/Mono Mode Select */ + ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */ + ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */ + ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */ + ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */ + ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */ + ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */ + ad1843_DDMEN = { 27, 12, 1 }, /* DAC2 to DAC1 Mix Enable */ + ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */ + ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */ + ad1843_C3EN = { 28, 13, 1 }, /* Clock Generator 3 Enable */ + ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */ + +/* + * The various registers of the AD1843 use three different formats for + * specifying gain. The ad1843_gain structure parameterizes the + * formats. + */ + +struct ad1843_gain { + int negative; /* nonzero if gain is negative. */ + const struct ad1843_bitfield *lfield; + const struct ad1843_bitfield *rfield; + const struct ad1843_bitfield *lmute; + const struct ad1843_bitfield *rmute; +}; + +static const struct ad1843_gain ad1843_gain_RECLEV = { + .negative = 0, + .lfield = &ad1843_LIG, + .rfield = &ad1843_RIG +}; +static const struct ad1843_gain ad1843_gain_LINE = { + .negative = 1, + .lfield = &ad1843_LX1M, + .rfield = &ad1843_RX1M, + .lmute = &ad1843_LX1MM, + .rmute = &ad1843_RX1MM +}; +static const struct ad1843_gain ad1843_gain_LINE_2 = { + .negative = 1, + .lfield = &ad1843_LDA2G, + .rfield = &ad1843_RDA2G, + .lmute = &ad1843_LDA2GM, + .rmute = &ad1843_RDA2GM +}; +static const struct ad1843_gain ad1843_gain_MIC = { + .negative = 1, + .lfield = &ad1843_LMCM, + .rfield = &ad1843_RMCM, + .lmute = &ad1843_LMCMM, + .rmute = &ad1843_RMCMM +}; +static const struct ad1843_gain ad1843_gain_PCM_0 = { + .negative = 1, + .lfield = &ad1843_LDA1G, + .rfield = &ad1843_RDA1G, + .lmute = &ad1843_LDA1GM, + .rmute = &ad1843_RDA1GM +}; +static const struct ad1843_gain ad1843_gain_PCM_1 = { + .negative = 1, + .lfield = &ad1843_LD2M, + .rfield = &ad1843_RD2M, + .lmute = &ad1843_LD2MM, + .rmute = &ad1843_RD2MM +}; + +static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] = +{ + &ad1843_gain_RECLEV, + &ad1843_gain_LINE, + &ad1843_gain_LINE_2, + &ad1843_gain_MIC, + &ad1843_gain_PCM_0, + &ad1843_gain_PCM_1, +}; + +/* read the current value of an AD1843 bitfield. */ + +static int ad1843_read_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field) +{ + int w; + + w = ad1843->read(ad1843->chip, field->reg); + return w >> field->lo_bit & ((1 << field->nbits) - 1); +} + +/* + * write a new value to an AD1843 bitfield and return the old value. + */ + +static int ad1843_write_bits(struct snd_ad1843 *ad1843, + const struct ad1843_bitfield *field, + int newval) +{ + int w, mask, oldval, newbits; + + w = ad1843->read(ad1843->chip, field->reg); + mask = ((1 << field->nbits) - 1) << field->lo_bit; + oldval = (w & mask) >> field->lo_bit; + newbits = (newval << field->lo_bit) & mask; + w = (w & ~mask) | newbits; + ad1843->write(ad1843->chip, field->reg, w); + + return oldval; +} + +/* + * ad1843_read_multi reads multiple bitfields from the same AD1843 + * register. It uses a single read cycle to do it. (Reading the + * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20 + * microseconds.) + * + * Called like this. + * + * ad1843_read_multi(ad1843, nfields, + * &ad1843_FIELD1, &val1, + * &ad1843_FIELD2, &val2, ...); + */ + +static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + const struct ad1843_bitfield *fp; + int w = 0, mask, *value, reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int *); + if (reg == -1) { + reg = fp->reg; + w = ad1843->read(ad1843->chip, reg); + } + + mask = (1 << fp->nbits) - 1; + *value = w >> fp->lo_bit & mask; + } + va_end(ap); +} + +/* + * ad1843_write_multi stores multiple bitfields into the same AD1843 + * register. It uses one read and one write cycle to do it. + * + * Called like this. + * + * ad1843_write_multi(ad1843, nfields, + * &ad1843_FIELD1, val1, + * &ad1843_FIELF2, val2, ...); + */ + +static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...) +{ + va_list ap; + int reg; + const struct ad1843_bitfield *fp; + int value; + int w, m, mask, bits; + + mask = 0; + bits = 0; + reg = -1; + + va_start(ap, argcount); + while (--argcount >= 0) { + fp = va_arg(ap, const struct ad1843_bitfield *); + value = va_arg(ap, int); + if (reg == -1) + reg = fp->reg; + else + BUG_ON(reg != fp->reg); + m = ((1 << fp->nbits) - 1) << fp->lo_bit; + mask |= m; + bits |= (value << fp->lo_bit) & m; + } + va_end(ap); + + if (~mask & 0xFFFF) + w = ad1843->read(ad1843->chip, reg); + else + w = 0; + w = (w & ~mask) | bits; + ad1843->write(ad1843->chip, reg, w); +} + +int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + int ret; + + ret = (1 << gp->lfield->nbits); + if (!gp->lmute) + ret -= 1; + return ret; +} + +/* + * ad1843_get_gain reads the specified register and extracts the gain value + * using the supplied gain type. + */ + +int ad1843_get_gain(struct snd_ad1843 *ad1843, int id) +{ + int lg, rg, lm, rm; + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg); + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) { + ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm); + if (lm) + lg = 0; + if (rm) + rg = 0; + } + return lg << 0 | rg << 8; +} + +/* + * Set an audio channel's gain. + * + * Returns the new gain, which may be lower than the old gain. + */ + +int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval) +{ + const struct ad1843_gain *gp = ad1843_gain[id]; + unsigned short mask = (1 << gp->lfield->nbits) - 1; + + int lg = (newval >> 0) & mask; + int rg = (newval >> 8) & mask; + int lm = (lg == 0) ? 1 : 0; + int rm = (rg == 0) ? 1 : 0; + + if (gp->negative) { + lg = mask - lg; + rg = mask - rg; + } + if (gp->lmute) + ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm); + ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg); + return ad1843_get_gain(ad1843, id); +} + +/* Returns the current recording source */ + +int ad1843_get_recsrc(struct snd_ad1843 *ad1843) +{ + int val = ad1843_read_bits(ad1843, &ad1843_LSS); + + if (val < 0 || val > 2) { + val = 2; + ad1843_write_multi(ad1843, 2, + &ad1843_LSS, val, &ad1843_RSS, val); + } + return val; +} + +/* + * Set recording source. + * + * Returns newsrc on success, -errno on failure. + */ + +int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc) +{ + if (newsrc < 0 || newsrc > 2) + return -EINVAL; + + ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc); + return newsrc; +} + +/* Setup ad1843 for D/A conversion. */ + +void ad1843_setup_dac(struct snd_ad1843 *ad1843, + unsigned int id, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int ad_fmt = 0, ad_mode = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_U8: + ad_fmt = 0; + break; + case SNDRV_PCM_FORMAT_S16_LE: + ad_fmt = 1; + break; + case SNDRV_PCM_FORMAT_MU_LAW: + ad_fmt = 2; + break; + case SNDRV_PCM_FORMAT_A_LAW: + ad_fmt = 3; + break; + default: + break; + } + + switch (channels) { + case 2: + ad_mode = 0; + break; + case 1: + ad_mode = 1; + break; + default: + break; + } + + if (id) { + ad1843_write_bits(ad1843, &ad1843_C2C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA2SM, ad_mode, + &ad1843_DA2F, ad_fmt); + } else { + ad1843_write_bits(ad1843, &ad1843_C1C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_DA1SM, ad_mode, + &ad1843_DA1F, ad_fmt); + } +} + +void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id) +{ + if (id) + ad1843_write_bits(ad1843, &ad1843_DA2F, 1); + else + ad1843_write_bits(ad1843, &ad1843_DA1F, 1); +} + +void ad1843_setup_adc(struct snd_ad1843 *ad1843, + unsigned int framerate, + snd_pcm_format_t fmt, + unsigned int channels) +{ + int da_fmt = 0; + + switch (fmt) { + case SNDRV_PCM_FORMAT_S8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_U8: da_fmt = 0; break; + case SNDRV_PCM_FORMAT_S16_LE: da_fmt = 1; break; + case SNDRV_PCM_FORMAT_MU_LAW: da_fmt = 2; break; + case SNDRV_PCM_FORMAT_A_LAW: da_fmt = 3; break; + default: break; + } + + ad1843_write_bits(ad1843, &ad1843_C3C, framerate); + ad1843_write_multi(ad1843, 2, + &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt); +} + +void ad1843_shutdown_adc(struct snd_ad1843 *ad1843) +{ + /* nothing to do */ +} + +/* + * Fully initialize the ad1843. As described in the AD1843 data + * sheet, section "START-UP SEQUENCE". The numbered comments are + * subsection headings from the data sheet. See the data sheet, pages + * 52-54, for more info. + * + * return 0 on success, -errno on failure. */ + +int ad1843_init(struct snd_ad1843 *ad1843) +{ + unsigned long later; + + if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) { + printk(KERN_ERR "ad1843: AD1843 won't initialize\n"); + return -EIO; + } + + ad1843_write_bits(ad1843, &ad1843_SCF, 1); + + /* 4. Put the conversion resources into standby. */ + ad1843_write_bits(ad1843, &ad1843_PDNI, 0); + later = jiffies + msecs_to_jiffies(500); + + while (ad1843_read_bits(ad1843, &ad1843_PDNO)) { + if (time_after(jiffies, later)) { + printk(KERN_ERR + "ad1843: AD1843 won't power up\n"); + return -EIO; + } + schedule_timeout_interruptible(5); + } + + /* 5. Power up the clock generators and enable clock output pins. */ + ad1843_write_multi(ad1843, 3, + &ad1843_C1EN, 1, + &ad1843_C2EN, 1, + &ad1843_C3EN, 1); + + /* 6. Configure conversion resources while they are in standby. */ + + /* DAC1/2 use clock 1/2 as source, ADC uses clock 3. Always. */ + ad1843_write_multi(ad1843, 4, + &ad1843_DA1C, 1, + &ad1843_DA2C, 2, + &ad1843_ADLC, 3, + &ad1843_ADRC, 3); + + /* 7. Enable conversion resources. */ + ad1843_write_bits(ad1843, &ad1843_ADTLK, 1); + ad1843_write_multi(ad1843, 7, + &ad1843_ANAEN, 1, + &ad1843_AAMEN, 1, + &ad1843_DA1EN, 1, + &ad1843_DA2EN, 1, + &ad1843_DDMEN, 1, + &ad1843_ADLEN, 1, + &ad1843_ADREN, 1); + + /* 8. Configure conversion resources while they are enabled. */ + + /* set gain to 0 for all channels */ + ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0); + ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0); + + /* Unmute all channels. */ + /* DAC1 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0); + /* DAC2 */ + ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0); + + /* Set default recording source to Line In and set + * mic gain to +20 dB. + */ + ad1843_set_recsrc(ad1843, 2); + ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1); + + /* Set Speaker Out level to +/- 4V and unmute it. */ + ad1843_write_multi(ad1843, 3, + &ad1843_HPOS, 1, + &ad1843_HPOM, 0, + &ad1843_MPOM, 0); + + return 0; +} diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c new file mode 100644 index 0000000..501b07c --- /dev/null +++ b/sound/mips/sgio2audio.c @@ -0,0 +1,1006 @@ +/* + * Sound driver for Silicon Graphics O2 Workstations A/V board audio. + * + * Copyright 2003 Vivien Chappelier <vivien.chappelier(a)linux-mips.org> + * Copyright 2008 Thomas Bogendoerfer <tsbogend(a)alpha.franken.de> + * Mxier part taken from mace_audio.c: + * Copyright 2007 Thorben Jändling <tj.trevelyan(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/spinlock.h> +#include <linux/gfp.h> +#include <linux/vmalloc.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <asm/ip32/ip32_ints.h> +#include <asm/ip32/mace.h> + +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#define SNDRV_GET_ID +#include <sound/initval.h> +#include <sound/ad1843.h> + + +MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier(a)linux-mips.org>"); +MODULE_DESCRIPTION("SGI O2 Audio"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard."); + + +#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */ +#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */ + +#define CODEC_CONTROL_WORD_SHIFT 0 +#define CODEC_CONTROL_READ BIT(16) +#define CODEC_CONTROL_ADDRESS_SHIFT 17 + +#define CHANNEL_CONTROL_RESET BIT(10) /* 1: reset channel */ +#define CHANNEL_DMA_ENABLE BIT(9) /* 1: enable DMA transfer */ +#define CHANNEL_INT_THRESHOLD_DISABLED (0 << 5) /* interrupt disabled */ +#define CHANNEL_INT_THRESHOLD_25 (1 << 5) /* int on buffer >25% full */ +#define CHANNEL_INT_THRESHOLD_50 (2 << 5) /* int on buffer >50% full */ +#define CHANNEL_INT_THRESHOLD_75 (3 << 5) /* int on buffer >75% full */ +#define CHANNEL_INT_THRESHOLD_EMPTY (4 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */ +#define CHANNEL_INT_THRESHOLD_FULL (6 << 5) /* int on buffer empty */ +#define CHANNEL_INT_THRESHOLD_NOT_FULL (7 << 5) /* int on buffer !empty */ + +#define CHANNEL_RING_SHIFT 12 +#define CHANNEL_RING_SIZE (1 << CHANNEL_RING_SHIFT) +#define CHANNEL_RING_MASK (CHANNEL_RING_SIZE - 1) + +#define CHANNEL_LEFT_SHIFT 40 +#define CHANNEL_RIGHT_SHIFT 8 + +struct snd_sgio2audio_chan { + int idx; + struct snd_pcm_substream *substream; + int pos; + snd_pcm_uframes_t size; + spinlock_t lock; +}; + +/* definition of the chip-specific record */ +struct snd_sgio2audio { + struct snd_card *card; + + /* codec */ + struct snd_ad1843 ad1843; + spinlock_t ad1843_lock; + + /* channels */ + struct snd_sgio2audio_chan channel[3]; + + /* resources */ + void *ring_base; + dma_addr_t ring_base_dma; +}; + +/* AD1843 access */ + +/* + * read_ad1843_reg returns the current contents of a 16 bit AD1843 register. + * + * Returns unsigned register value on success, -errno on failure. + */ +static int read_ad1843_reg(void *priv, int reg) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + CODEC_CONTROL_READ, &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + val = readq(&mace->perif.audio.codec_read); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return val; +} + +/* + * write_ad1843_reg writes the specified value to a 16 bit AD1843 register. + */ +static int write_ad1843_reg(void *priv, int reg, int word) +{ + struct snd_sgio2audio *chip = priv; + int val; + unsigned long flags; + + spin_lock_irqsave(&chip->ad1843_lock, flags); + + writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) | + (word << CODEC_CONTROL_WORD_SHIFT), + &mace->perif.audio.codec_control); + wmb(); + val = readq(&mace->perif.audio.codec_control); /* flush bus */ + udelay(200); + + spin_unlock_irqrestore(&chip->ad1843_lock, flags); + return 0; +} + +static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843, + (int)kcontrol->private_value); + return 0; +} + +static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int vol; + + vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value); + + ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF; + ucontrol->value.integer.value[1] = vol & 0xFF; + + return 0; +} + +static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newvol, oldvol; + + oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value); + newvol = (ucontrol->value.integer.value[0] << 8) | + ucontrol->value.integer.value[1]; + + newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value, + newvol); + + return newvol != oldvol; +} + +static int sgio2audio_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "Cam Mic", "Mic", "Line" + }; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item >= 3) + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int sgio2audio_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843); + return 0; +} + +static int sgio2audio_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol); + int newsrc, oldsrc; + + oldsrc = ad1843_get_recsrc(&chip->ad1843); + newsrc = ad1843_set_recsrc(&chip->ad1843, + ucontrol->value.enumerated.item[0]); + + return newsrc != oldsrc; +} + +/* dac1/pcm0 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_0, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* dac2/pcm1 mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_PCM_1, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_RECLEV, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* record level source control */ +static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = sgio2audio_source_info, + .get = sgio2audio_source_get, + .put = sgio2audio_source_put, +}; + +/* line mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* cd mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Playback Volume", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_LINE_2, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + +/* mic mixer control */ +static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .private_value = AD1843_GAIN_MIC, + .info = sgio2audio_gain_info, + .get = sgio2audio_gain_get, + .put = sgio2audio_gain_put, +}; + + +static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip) +{ + int err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_recsource, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_line, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_cd, chip)); + if (err < 0) + return err; + + err = snd_ctl_add(chip->card, + snd_ctl_new1(&sgio2audio_ctrl_mic, chip)); + if (err < 0) + return err; + + return 0; +} + +/* low-level audio interface DMA */ + +/* get data out of bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + unsigned long src_base, src_pos, dst_mask; + unsigned char *dst_base; + int dst_pos; + u64 *src; + s16 *dst; + u64 x; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT); + src_pos = readq(&mace->perif.audio.chan[ch].read_ptr); + dst_base = runtime->dma_area; + dst_pos = chip->channel[ch].pos; + dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (u64 *)(src_base + src_pos); + dst = (s16 *)(dst_base + dst_pos); + + x = *src; + dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff; + dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff; + + src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK; + dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask; + count -= sizeof(u64); + } + + writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */ + chip->channel[ch].pos = dst_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +/* put some DMA data in bounce buffer, count must be a multiple of 32 */ +/* returns 1 if a period has elapsed */ +static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip, + unsigned int ch, unsigned int count) +{ + int ret; + s64 l, r; + unsigned long dst_base, dst_pos, src_mask; + unsigned char *src_base; + int src_pos; + u64 *dst; + s16 *src; + unsigned long flags; + struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT); + dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr); + src_base = runtime->dma_area; + src_pos = chip->channel[ch].pos; + src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1; + + /* check if a period has elapsed */ + chip->channel[ch].size += (count >> 3); /* in frames */ + ret = chip->channel[ch].size >= runtime->period_size; + chip->channel[ch].size %= runtime->period_size; + + while (count) { + src = (s16 *)(src_base + src_pos); + dst = (u64 *)(dst_base + dst_pos); + + l = src[0]; /* sign extend */ + r = src[1]; /* sign extend */ + + *dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) | + ((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT); + + dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK; + src_pos = (src_pos + 2 * sizeof(s16)) & src_mask; + count -= sizeof(u64); + } + + writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */ + chip->channel[ch].pos = src_pos; + + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return ret; +} + +static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + + /* reset DMA channel */ + writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control); + udelay(10); + writeq(0, &mace->perif.audio.chan[ch].control); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* push a full buffer */ + snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32); + } + /* set DMA to wake on 50% empty and enable interrupt */ + writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50, + &mace->perif.audio.chan[ch].control); + return 0; +} + +static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + writeq(0, &mace->perif.audio.chan[chan->idx].control); + return 0; +} + +static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + + /* empty the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_pull_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + struct snd_sgio2audio *chip; + int count, ch; + + substream = chan->substream; + chip = snd_pcm_substream_chip(substream); + ch = chan->idx; + /* fill the ring */ + count = CHANNEL_RING_SIZE - + readq(&mace->perif.audio.chan[ch].depth) - 32; + if (snd_sgio2audio_dma_push_frag(chip, ch, count)) + snd_pcm_period_elapsed(substream); + + return IRQ_HANDLED; +} + +static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id) +{ + struct snd_sgio2audio_chan *chan = dev_id; + struct snd_pcm_substream *substream; + + substream = chan->substream; + snd_sgio2audio_dma_stop(substream); + snd_sgio2audio_dma_start(substream); + return IRQ_HANDLED; +} + +/* PCM part */ +/* PCM hardware definition */ +static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536, + .period_bytes_min = 32768, + .period_bytes_max = 65536, + .periods_min = 1, + .periods_max = 1024, +}; + +/* PCM playback open callback */ +static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[1]; + return 0; +} + +static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[2]; + return 0; +} + +/* PCM capture open callback */ +static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sgio2audio_pcm_hw; + runtime->private_data = &chip->channel[0]; + return 0; +} + +/* PCM close callback */ +static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->private_data = NULL; + return 0; +} + + +/* hw_params callback */ +static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int size = params_buffer_bytes(hw_params); + + /* alloc virtual 'dma' area */ + if (runtime->dma_area) + vfree(runtime->dma_area); + runtime->dma_area = vmalloc(size); + if (runtime->dma_area == NULL) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* hw_free callback */ +static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) +{ + if (substream->runtime->dma_area) + vfree(substream->runtime->dma_area); + substream->runtime->dma_area = NULL; + return 0; +} + +/* prepare callback */ +static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + int ch = chan->idx; + unsigned long flags; + + spin_lock_irqsave(&chip->channel[ch].lock, flags); + + /* Setup the pseudo-dma transfer pointers. */ + chip->channel[ch].pos = 0; + chip->channel[ch].size = 0; + chip->channel[ch].substream = substream; + + /* set AD1843 format */ + /* hardware format is always S16_LE */ + switch (substream->stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + ad1843_setup_dac(&chip->ad1843, + ch - 1, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + case SNDRV_PCM_STREAM_CAPTURE: + ad1843_setup_adc(&chip->ad1843, + runtime->rate, + SNDRV_PCM_FORMAT_S16_LE, + runtime->channels); + break; + } + spin_unlock_irqrestore(&chip->channel[ch].lock, flags); + return 0; +} + +/* trigger callback */ +static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* start the PCM engine */ + snd_sgio2audio_dma_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* stop the PCM engine */ + snd_sgio2audio_dma_stop(substream); + break; + default: + return -EINVAL; + } + return 0; +} + +/* pointer callback */ +static snd_pcm_uframes_t +snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream); + struct snd_sgio2audio_chan *chan = substream->runtime->private_data; + + /* get the current hardware pointer */ + return bytes_to_frames(substream->runtime, + chip->channel[chan->idx].pos); +} + +/* get the physical page pointer on the given offset */ +static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} + +/* operators */ +static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { + .open = snd_sgio2audio_playback1_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { + .open = snd_sgio2audio_playback2_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +static struct snd_pcm_ops snd_sgio2audio_capture_ops = { + .open = snd_sgio2audio_capture_open, + .close = snd_sgio2audio_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sgio2audio_pcm_hw_params, + .hw_free = snd_sgio2audio_pcm_hw_free, + .prepare = snd_sgio2audio_pcm_prepare, + .trigger = snd_sgio2audio_pcm_trigger, + .pointer = snd_sgio2audio_pcm_pointer, + .page = snd_sgio2audio_page, +}; + +/* + * definitions of capture are omitted here... + */ + +/* create a pcm device */ +static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip) +{ + struct snd_pcm *pcm; + int err; + + /* create first pcm device with one outputs and one input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC1"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback1_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_sgio2audio_capture_ops); + + /* create second pcm device with one outputs and no input */ + err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SGI O2 DAC2"); + + /* set operators */ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_sgio2audio_playback2_ops); + + return 0; +} + +static struct { + int idx; + int irq; + irqreturn_t (*isr)(int, void *); + const char *desc; +} snd_sgio2_isr_table[] = { + { + .idx = 0, + .irq = MACEISA_AUDIO1_DMAT_IRQ, + .isr = snd_sgio2audio_dma_in_isr, + .desc = "Capture DMA Channel 0" + }, { + .idx = 0, + .irq = MACEISA_AUDIO1_OF_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Capture Overflow" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 1" + }, { + .idx = 1, + .irq = MACEISA_AUDIO2_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 1" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_DMAT_IRQ, + .isr = snd_sgio2audio_dma_out_isr, + .desc = "Playback DMA Channel 2" + }, { + .idx = 2, + .irq = MACEISA_AUDIO3_MERR_IRQ, + .isr = snd_sgio2audio_error_isr, + .desc = "Memory Error Channel 2" + } +}; + +/* ALSA driver */ + +static int snd_sgio2audio_free(struct snd_sgio2audio *chip) +{ + int i; + + /* reset interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + + /* release IRQ's */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) + free_irq(snd_sgio2_isr_table[i].irq, + &chip->channel[snd_sgio2_isr_table[i].idx]); + + dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + chip->ring_base, chip->ring_base_dma); + + /* release card data */ + kfree(chip); + return 0; +} + +static int snd_sgio2audio_dev_free(struct snd_device *device) +{ + struct snd_sgio2audio *chip = device->device_data; + + return snd_sgio2audio_free(chip); +} + +static struct snd_device_ops ops = { + .dev_free = snd_sgio2audio_dev_free, +}; + +static int __devinit snd_sgio2audio_create(struct snd_card *card, + struct snd_sgio2audio **rchip) +{ + struct snd_sgio2audio *chip; + int i, err; + + *rchip = NULL; + + /* check if a codec is attached to the interface */ + /* (Audio or Audio/Video board present) */ + if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT)) + return -ENOENT; + + chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE, + &chip->ring_base_dma, GFP_USER); + if (chip->ring_base == NULL) { + printk(KERN_ERR + "sgio2audio: could not allocate ring buffers\n"); + kfree(chip); + return -ENOMEM; + } + + spin_lock_init(&chip->ad1843_lock); + + /* initialize channels */ + for (i = 0; i < 3; i++) { + spin_lock_init(&chip->channel[i].lock); + chip->channel[i].idx = i; + } + + /* allocate IRQs */ + for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) { + if (request_irq(snd_sgio2_isr_table[i].irq, + snd_sgio2_isr_table[i].isr, + IRQF_SHARED, + snd_sgio2_isr_table[i].desc, + &chip->channel[snd_sgio2_isr_table[i].idx])) { + snd_sgio2audio_free(chip); + printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n", + snd_sgio2_isr_table[i].irq); + return -EBUSY; + } + } + + /* reset the interface */ + writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control); + udelay(1); + writeq(0, &mace->perif.audio.control); + msleep_interruptible(1); /* give time to recover */ + + /* set ring base */ + writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase); + + /* attach the AD1843 codec */ + chip->ad1843.read = read_ad1843_reg; + chip->ad1843.write = write_ad1843_reg; + chip->ad1843.chip = chip; + + /* initialize the AD1843 codec */ + err = ad1843_init(&chip->ad1843); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sgio2audio_free(chip); + return err; + } + *rchip = chip; + return 0; +} + +static int __devinit snd_sgio2audio_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct snd_sgio2audio *chip; + int err; + + card = snd_card_new(index, id, THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + err = snd_sgio2audio_create(card, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + snd_card_set_dev(card, &pdev->dev); + + err = snd_sgio2audio_new_pcm(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sgio2audio_new_mixer(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "SGI O2 Audio"); + strcpy(card->shortname, "SGI O2 Audio"); + sprintf(card->longname, "%s irq %i-%i", + card->shortname, + MACEISA_AUDIO1_DMAT_IRQ, + MACEISA_AUDIO3_MERR_IRQ); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pdev, card); + return 0; +} + +static int __exit snd_sgio2audio_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + + snd_card_free(card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver sgio2audio_driver = { + .probe = snd_sgio2audio_probe, + .remove = __devexit_p(snd_sgio2audio_remove), + .driver = { + .name = "sgio2audio", + .owner = THIS_MODULE, + } +}; + +static int __init alsa_card_sgio2audio_init(void) +{ + return platform_driver_register(&sgio2audio_driver); +} + +static void __exit alsa_card_sgio2audio_exit(void) +{ + platform_driver_unregister(&sgio2audio_driver); +} + +module_init(alsa_card_sgio2audio_init) +module_exit(alsa_card_sgio2audio_exit)

1 0

[alsa-devel] [PATCH 2/2] ASoC: pxa2xx-ac97: fix warning due to missing argument in fuction declaration
by Dmitry Baryshkov 09 Jul '08

09 Jul '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/pxa2xx-ac97.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index b458667..ce1835e 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -356,7 +356,7 @@ static int pxa2xx_ac97_probe(struct platform_device *pdev, return ret; } -static void pxa2xx_ac97_remove(struct platform_device *pdev) +static void pxa2xx_ac97_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); -- 1.5.6 -- With best wishes Dmitry

3 2

[alsa-devel] [PATCH] Add Yamaha KX49 (USB MIDI controller) to usbquirks.h
by Richard Chan 09 Jul '08

09 Jul '08

This patch is for the Yamaha USB MIDI controller KX49. http://www.yamahasynth.com/products/kx/index.html It has a 3-port MIDI interface and an HID interface (it has a tiny keyboard subset). --------------dmesg---------------------------------------------------------------- usb 6-2: new full speed USB device using uhci_hcd and address 3 usb 6-2: configuration #1 chosen from 1 choice input: YAMAHA Corporation YAMAHA KX as /devices/pci0000:00/0000:00:1d.1/usb6/6-2/6-2:1.1/input/input12 input,hidraw1: USB HID v1.00 Keyboard [YAMAHA Corporation YAMAHA KX] on usb-0000:00:1d.1-2 usb 6-2: New USB device found, idVendor=0499, idProduct=104e usb 6-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 6-2: Product: YAMAHA KX usb 6-2: Manufacturer: YAMAHA Corporation ALSA /home/richard/tmp/usb/usbmidi.c:1312: created 3 output and 3 input ports usbcore: registered new interface driver snd-usb-audio -------------end dmesg-------------------------------------------------------------- diff -u -r alsa-kernel/usb/usbquirks.h b/usb/usbquirks.h --- alsa-kernel/usb/usbquirks.h 2008-07-08 21:35:42.000000000 +0800 +++ b/usb/usbquirks.h 2008-07-09 20:45:20.000000000 +0800 @@ -210,6 +210,7 @@ YAMAHA_DEVICE(0x1043, NULL), YAMAHA_DEVICE(0x1044, NULL), YAMAHA_DEVICE(0x1045, NULL), +YAMAHA_DEVICE(0x104E, "KX49"), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), diff -u -r alsa-kernel/usb/usbquirks.h b/usb/usbquirks.h --- alsa-kernel/usb/usbquirks.h 2008-07-08 21:35:42.000000000 +0800 +++ b/usb/usbquirks.h 2008-07-09 20:45:20.000000000 +0800 @@ -210,6 +210,7 @@ YAMAHA_DEVICE(0x1043, NULL), YAMAHA_DEVICE(0x1044, NULL), YAMAHA_DEVICE(0x1045, NULL), +YAMAHA_DEVICE(0x104E, "KX49"), YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), Bus 006 Device 003: ID 0499:104e Yamaha Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0499 Yamaha Corp. idProduct 0x104e bcdDevice 1.00 iManufacturer 1 YAMAHA Corporation iProduct 2 YAMAHA KX iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 109 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 255 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 42 00 ** UNRECOGNIZED: 06 24 02 02 01 00 ** UNRECOGNIZED: 06 24 02 02 02 00 ** UNRECOGNIZED: 06 24 02 02 03 00 ** UNRECOGNIZED: 09 24 03 02 01 01 01 01 00 ** UNRECOGNIZED: 09 24 03 02 02 01 01 01 00 ** UNRECOGNIZED: 09 24 03 02 03 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 1 Boot Interface Subclass bInterfaceProtocol 1 Keyboard iInterface 3 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 45 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 10 Device Status: 0x0001 Self Powered

2 1

[alsa-devel] FWIW: is this right
by Justin Mattock 09 Jul '08

09 Jul '08

Hello; FWIW I think this is the right area for this question: basically I have a bug filed:http://bugzilla.kernel.org/show_bug.cgi?id=10954 and am trying to fix it or find a work around. what I'm seeing in dmesg is: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0x010f1c00 I can't seem to reproduce this issue, only seems to appear every 100 +/- boots. any ideas.(attached is dmesg of the bug). regards; -- Justin P. Mattock

3 4

[alsa-devel] [PATCH 1/2] tosa: fix compilation with new DAPM API
by Dmitry Baryshkov 09 Jul '08

09 Jul '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/tosa.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index b59a8df..fe6cca9 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -151,7 +151,7 @@ SND_SOC_DAPM_SPK("Speaker", NULL), }; /* tosa audio map */ -static const snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route audio_map[] = { /* headphone connected to HPOUTL, HPOUTR */ {"Headphone Jack", NULL, "HPOUTL"}, @@ -203,7 +203,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) } /* add tosa specific widgets */ - snd_soc_dapm_new_controls(codec, &tosa_dapm_widgets, + snd_soc_dapm_new_controls(codec, tosa_dapm_widgets, ARRAY_SIZE(tosa_dapm_widgets)); /* set up tosa specific audio path audio_map */ -- 1.5.6 -- With best wishes Dmitry

3 3

[alsa-devel] [PATCH] fix patch_ucb1400 for proper resume
by Mike Rapoport 09 Jul '08

09 Jul '08

Replace 'snd_ac97_write' with snd_ac97_write_cache' in pacth_ucb1400 to allow proper codec wakeup. Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1292dce..b821874 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3633,7 +3633,7 @@ static int patch_ucb1400(struct snd_ac97 * ac97) { ac97->build_ops = &patch_ucb1400_ops; /* enable headphone driver and smart low power mode by default */ - snd_ac97_write(ac97, 0x6a, 0x0050); - snd_ac97_write(ac97, 0x6c, 0x0030); + snd_ac97_write_cache(ac97, 0x6a, 0x0050); + snd_ac97_write_cache(ac97, 0x6c, 0x0030); return 0; } -- Sincerely yours, Mike.

1 1

[alsa-devel] soc-of.h file
by dinesh 08 Jul '08

08 Jul '08

I need to use function of_snd_soc_register_platform of soc-of.o i got soc-of.c file from patch [PATCH 1/3] ALSA SoC: Add OpenFirmware helper for matching bus and codec drivers with but i am not able to find soc-of.h file please provide this to me. Regards, Dinesh Dua

2 1

[alsa-devel] [PATCH] asoc: at32 - DAI struct merge and enable_pin() change.
by Liam Girdwood 08 Jul '08

08 Jul '08

This adds support for the recent DAI struct merge and new DAPM pin API. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at32/playpaq_wm8510.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index fd62f25..fee5f8e 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -341,15 +341,15 @@ static int playpaq_wm8510_init(struct snd_soc_codec *codec) } - /* always connected endpoints */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); - snd_soc_dapm_sync_endpoints(codec); + /* always connected pins */ + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_sync(codec); /* Make CSB show PLL rate */ - codec->dai->dai_ops.set_clkdiv(codec->dai, WM8510_OPCLKDIV, + snd_soc_dai_set_clkdiv(codec->dai, WM8510_OPCLKDIV, WM8510_OPCLKDIV_1 | 4); return 0; -- 1.5.4.2

1 0

[alsa-devel] [PATCH] asoc: n810 - fix build error.
by Liam Girdwood 08 Jul '08

08 Jul '08

This patch adds a missing " that was recently introduced (removed) in the DAI struct merge patch series. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/omap/n810.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index e53c055..02cec96 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -63,7 +63,7 @@ static void n810_ext_control(struct snd_soc_codec *codec) if (n810_dmic_func) snd_soc_dapm_enable_pin(codec, "DMic"); else - snd_soc_dapm_disable_pin(codec, "DMic); + snd_soc_dapm_disable_pin(codec, "DMic"); snd_soc_dapm_sync(codec); } -- 1.5.4.2

1 0

[alsa-devel] [PATCH 02/02] asoc: machines - add Digital Audio Interface (DAI) control functions.
by Liam Girdwood 08 Jul '08

08 Jul '08

This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct within the machine drivers. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at32/playpaq_wm8510.c | 16 ++++----- sound/soc/at91/eti_b1_wm8731.c | 18 +++++----- sound/soc/davinci/davinci-evm.c | 6 ++-- sound/soc/fsl/mpc8610_hpcd.c | 58 +++++++++++++++--------------------- sound/soc/omap/n810.c | 6 ++-- sound/soc/pxa/corgi.c | 8 ++-- sound/soc/pxa/poodle.c | 8 ++-- sound/soc/pxa/spitz.c | 8 ++-- sound/soc/s3c24xx/neo1973_wm8753.c | 26 ++++++++-------- 9 files changed, 71 insertions(+), 83 deletions(-) diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index 18ac7d7..fd62f25 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -210,14 +210,14 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, /* * set CPU and CODEC DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, fmt); + ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CODEC DAI format (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, fmt); + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { pr_warning("playpaq_wm8510: " "Failed to set CPU DAI format (%d)\n", @@ -233,14 +233,13 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, cd = playpaq_wm8510_calc_ssc_clock(params, cpu_dai); pr_debug("playpaq_wm8510: cmr_div = %d, period = %d\n", cd.cmr_div, cd.period); - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, - AT32_SSC_CMR_DIV, cd.cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_CMR_DIV, cd.cmr_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CPU CMR_DIV (%d)\n", ret); return ret; } - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT32_SSC_TCMR_PERIOD, cd.period); if (ret < 0) { pr_warning("playpaq_wm8510: " @@ -260,7 +259,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #if !defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_BCLKDIV, bclk); if (ret < 0) { pr_warning ("playpaq_wm8510: Failed to set CODEC DAI BCLKDIV (%d)\n", @@ -270,7 +269,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - ret = codec_dai->dai_ops.set_pll(codec_dai, 0, + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_get_rate(CODEC_CLK), pll_out); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", @@ -279,8 +278,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, } - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, - WM8510_MCLKDIV, mclk_div); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8510_MCLKDIV, mclk_div); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC MCLKDIV (%d)\n", ret); diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 9d42139..d532de9 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -58,13 +58,13 @@ static int eti_b1_startup(struct snd_pcm_substream *substream) int ret; /* cpu clock is the AT91 master clock sent to the SSC */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, AT91_SYSCLK_MCK, + ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK, 60000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* codec system clock is supplied by PCK1, set to 12MHz */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -96,13 +96,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, int cmr_div, period; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; @@ -141,17 +141,17 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, } /* set the MCK divider for BCLK */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div); if (ret < 0) return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* set the BCLK divider for DACLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_TCMR_PERIOD, period); } else { /* set the BCLK divider for ADCLRC */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, + ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_RCMR_PERIOD, period); } if (ret < 0) @@ -163,13 +163,13 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4249e6a..5e2c306 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -38,19 +38,19 @@ static int evm_hw_params(struct snd_pcm_substream *substream, int ret = 0; /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF); if (ret < 0) return ret; /* set the codec system clock */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + ret = snd_soc_dai_set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 59d7e49..4bdc9d8 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -103,55 +103,45 @@ static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) int ret = 0; /* Tell the CPU driver what the serial protocol is. */ - if (cpu_dai->dai_ops.set_fmt) { - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(cpu_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver audio format\n"); + return ret; } /* Tell the codec driver what the serial protocol is. */ - if (codec_dai->dai_ops.set_fmt) { - ret = codec_dai->dai_ops.set_fmt(codec_dai, - machine_data->dai_format); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver audio format\n"); - return ret; - } + ret = snd_soc_dai_set_fmt(codec_dai, machine_data->dai_format); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver audio format\n"); + return ret; } /* * Tell the CPU driver what the clock frequency is, and whether it's a * slave or master. */ - if (cpu_dai->dai_ops.set_sysclk) { - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, 0, - machine_data->clk_frequency, - machine_data->cpu_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set CPU driver clock parameters\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, + machine_data->clk_frequency, + machine_data->cpu_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set CPU driver clock parameters\n"); + return ret; } /* * Tell the codec driver what the MCLK frequency is, and whether it's * a slave or master. */ - if (codec_dai->dai_ops.set_sysclk) { - ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(substream->pcm->card->dev, - "could not set codec driver clock params\n"); - return ret; - } + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + machine_data->clk_frequency, + machine_data->codec_clk_direction); + if (ret < 0) { + dev_err(substream->pcm->card->dev, + "could not set codec driver clock params\n"); + return ret; } return 0; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index d1233c0..e53c055 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -91,7 +91,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, int err; /* Set codec DAI configuration */ - err = codec_dai->dai_ops.set_fmt(codec_dai, + err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -99,7 +99,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set cpu DAI configuration */ - err = cpu_dai->dai_ops.set_fmt(cpu_dai, + err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); @@ -107,7 +107,7 @@ static int n810_hw_params(struct snd_pcm_substream *substream, return err; /* Set the codec system clock for DAC and ADC */ - err = codec_dai->dai_ops.set_sysclk(codec_dai, 0, 12000000, + err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); return err; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 782afbf..c029446 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -143,25 +143,25 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index ce25b6b..65a4e9a 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -122,25 +122,25 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index fd1abc7..6438579 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -141,25 +141,25 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8750_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set the I2S system clock as input (unused) */ - ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 51a4ce3..4d7a9aa 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -108,44 +108,44 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set MCLK division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS); if (ret < 0) return ret; /* set codec BCLK division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); if (ret < 0) return ret; /* set prescaler division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; /* codec PLL input is PCLK/4 */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -161,7 +161,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); } /* @@ -194,24 +194,24 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, /* todo: gg check mode (DSP_B) against CSR datasheet */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set codec PCM division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); if (ret < 0) return ret; /* configue and enable PLL for 12.288MHz output */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -227,7 +227,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { -- 1.5.4.2

1 0

[alsa-devel] [PATCH 01/02] asoc: core - add Digital Audio Interface (DAI) control functions.
by Liam Girdwood 08 Jul '08

08 Jul '08

This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 21 ++++++++ sound/soc/soc-core.c | 140 +++++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 153 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 778e57e..1890d87 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -256,6 +256,27 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + /* *Controls */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d626b4..83f1190 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -434,8 +434,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) else { codec_dai->pop_wait = 0; cancel_delayed_work(&socdev->delayed_work); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } } else { /* no delayed work - do we need to power up codec */ @@ -454,8 +453,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_START); snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + snd_soc_dai_digital_mute(codec_dai, 0); } else { /* codec already powered - power on widgets */ @@ -467,8 +465,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dapm_stream_event(codec, codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_START); - if (codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 0); + + snd_soc_dai_digital_mute(codec_dai, 0); } } @@ -566,8 +564,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* apply codec digital mute */ - if (!codec->active && codec_dai->dai_ops.digital_mute) - codec_dai->dai_ops.digital_mute(codec_dai, 1); + if (!codec->active) + snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ if (machine->ops && machine->ops->hw_free) @@ -1703,6 +1701,132 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); +/** + * snd_soc_dai_set_sysclk - configure DAI system or master clock. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the DAI master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); + +/** + * snd_soc_dai_set_clkdiv - configure DAI clock dividers. + * @dai: DAI + * @clk_id: DAI specific clock divider ID + * @div: new clock divisor. + * + * Configures the clock dividers. This is used to derive the best DAI bit and + * frame clocks from the system or master clock. It's best to set the DAI bit + * and frame clocks as low as possible to save system power. + */ +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div) +{ + if (dai->dai_ops.set_clkdiv) + return dai->dai_ops.set_clkdiv(dai, div_id, div); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); + +/** + * snd_soc_dai_set_pll - configure DAI PLL. + * @dai: DAI + * @pll_id: DAI specific PLL ID + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + if (dai->dai_ops.set_pll) + return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); + +/** + * snd_soc_dai_set_fmt - configure DAI hardware audio format. + * @dai: DAI + * @clk_id: DAI specific clock ID + * @fmt: SND_SOC_DAIFMT_ format value. + * + * Configures the DAI hardware format and clocking. + */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + if (dai->dai_ops.set_fmt) + return dai->dai_ops.set_fmt(dai, fmt); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); + +/** + * snd_soc_dai_set_tdm_slot - configure DAI TDM. + * @dai: DAI + * @mask: DAI specific mask representing used slots. + * @slots: Number of slots in use. + * + * Configures a DAI for TDM operation. Both mask and slots are codec and DAI + * specific. + */ +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tdm_slot(dai, mask, slots); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); + +/** + * snd_soc_dai_set_tristate - configure DAI system or master clock. + * @dai: DAI + * @tristate: tristate enable + * + * Tristates the DAI so that others can use it. + */ +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + if (dai->dai_ops.set_sysclk) + return dai->dai_ops.set_tristate(dai, tristate); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); + +/** + * snd_soc_dai_digital_mute - configure DAI system or master clock. + * @dai: DAI + * @mute: mute enable + * + * Mutes the DAI DAC. + */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) +{ + if (dai->dai_ops.digital_mute) + return dai->dai_ops.digital_mute(dai, mute); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); + static int __devinit snd_soc_init(void) { printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); -- 1.5.4.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: Switch DAPM to use of standard DEBUG macro
by Mark Brown 08 Jul '08

08 Jul '08

DAPM contains debug output controlled by a DAPM_DEBUG macro. Change this to be controlled by the standard DEBUG, dropping the custom dbg() macro as we go. Also fix the error printed when configuring an unknown pin to be an unconditionally displayed error rather than debug output. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 22 ++++++++++------------ 1 files changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 94296b5..d18ebc6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -45,13 +45,10 @@ #include <sound/initval.h> /* debug */ -#define DAPM_DEBUG 0 -#if DAPM_DEBUG +#ifdef DEBUG #define dump_dapm(codec, action) dbg_dump_dapm(codec, action) -#define dbg(format, arg...) printk(format, ## arg) #else #define dump_dapm(codec, action) -#define dbg(format, arg...) #endif /* dapm power sequences - make this per codec in the future */ @@ -233,7 +230,8 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(); } - dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); + pr_debug("reg %x old %x new %x change %d\n", widget->reg, + old, new, change); return change; } @@ -591,8 +589,8 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* call any power change event handlers */ if (power_change) { if (w->event) { - dbg("power %s event for %s flags %x\n", - w->power ? "on" : "off", w->name, w->event_flags); + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", w->name, w->event_flags); if (power) { /* power up event */ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { @@ -634,7 +632,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) return ret; } -#if DAPM_DEBUG +#ifdef DEBUG static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) { struct snd_soc_dapm_widget *w; @@ -887,13 +885,13 @@ static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, list_for_each_entry(w, &codec->dapm_widgets, list) { if (!strcmp(w->name, pin)) { - dbg("dapm: %s: pin %s\n", codec->name, pin); + pr_debug("dapm: %s: pin %s\n", codec->name, pin); w->connected = status; return 0; } } - dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + pr_err("dapm: %s: configuring unknown pin %s\n", codec->name, pin); return -EINVAL; } @@ -1397,8 +1395,8 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, { if (!w->sname) continue; - dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname, - stream, event); + pr_debug("widget %s\n %s stream %s event %d\n", + w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { case SND_SOC_DAPM_STREAM_START: -- 1.5.6

2 2

[alsa-devel] [PATCH] hda: removed redundant gpio_mask
by Matthew Ranostay 08 Jul '08

08 Jul '08

An gpio_mask value was defined twice needlessly. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index c4f3489..a6d1388 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -3669,7 +3669,6 @@ again: /* GPIO0 High = EAPD */ spec->gpio_mask = 0x01; spec->gpio_dir = 0x01; - spec->gpio_mask = 0x01; spec->gpio_data = 0x01; spec->mux_nids = stac92hd71bxx_mux_nids;

1 0

[alsa-devel] [PATCH 01/10] asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 71 ++++++++++++++++--------------------------------- sound/soc/soc-core.c | 50 +++++++++++++++++----------------- 2 files changed, 48 insertions(+), 73 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 340223a..778e57e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -221,8 +221,7 @@ struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; -struct snd_soc_codec_dai; -struct snd_soc_cpu_dai; +struct snd_soc_dai; struct snd_soc_codec; struct snd_soc_machine_config; struct soc_enum; @@ -317,50 +316,24 @@ struct snd_soc_ops { /* ASoC DAI ops */ struct snd_soc_dai_ops { /* DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_codec_dai *codec_dai, + int (*set_sysclk)(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_codec_dai *codec_dai, + int (*set_pll)(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_codec_dai *codec_dai, - int div_id, int div); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); /* DAI format configuration */ - int (*set_fmt)(struct snd_soc_codec_dai *codec_dai, - unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_codec_dai *codec_dai, + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_codec_dai *, int tristate); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* digital mute */ - int (*digital_mute)(struct snd_soc_codec_dai *, int mute); + int (*digital_mute)(struct snd_soc_dai *dai, int mute); }; -/* SoC Codec DAI */ -struct snd_soc_codec_dai { - char *name; - int id; - unsigned char type; - - /* DAI capabilities */ - struct snd_soc_pcm_stream playback; - struct snd_soc_pcm_stream capture; - - /* DAI runtime info */ - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; - - /* DAI private data */ - void *private_data; -}; - -/* SoC CPU DAI */ -struct snd_soc_cpu_dai { - +/* SoC DAI (Digital Audio Interface) */ +struct snd_soc_dai { /* DAI description */ char *name; unsigned int id; @@ -368,13 +341,13 @@ struct snd_soc_cpu_dai { /* DAI callbacks */ int (*probe)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); void (*remove)(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai); + struct snd_soc_dai *dai); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* ops */ struct snd_soc_ops ops; @@ -386,7 +359,9 @@ struct snd_soc_cpu_dai { /* DAI runtime info */ struct snd_pcm_runtime *runtime; - unsigned char active:1; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; void *dma_data; /* DAI private data */ @@ -428,7 +403,7 @@ struct snd_soc_codec { struct delayed_work delayed_work; /* codec DAI's */ - struct snd_soc_codec_dai *dai; + struct snd_soc_dai *dai; unsigned int num_dai; }; @@ -447,12 +422,12 @@ struct snd_soc_platform { int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); int (*suspend)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); int (*resume)(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai); + struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, + int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, struct snd_pcm *); void (*pcm_free)(struct snd_pcm *); @@ -466,8 +441,8 @@ struct snd_soc_dai_link { char *stream_name; /* Stream name */ /* DAI */ - struct snd_soc_codec_dai *codec_dai; - struct snd_soc_cpu_dai *cpu_dai; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; /* machine stream operations */ struct snd_soc_ops *ops; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bdbbc6a..4d626b4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -134,8 +134,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -272,7 +272,7 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_device *socdev = container_of(work, struct snd_soc_device, delayed_work.work); struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; mutex_lock(&pcm_mutex); @@ -323,8 +323,8 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -384,8 +384,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; int ret = 0; @@ -489,8 +489,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); @@ -559,8 +559,8 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; struct snd_soc_codec *codec = socdev->codec; mutex_lock(&pcm_mutex); @@ -594,8 +594,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_dai_link *machine = rtd->dai; struct snd_soc_platform *platform = socdev->platform; - struct snd_soc_cpu_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_codec_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; if (codec_dai->ops.trigger) { @@ -651,7 +651,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 1); } @@ -664,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) machine->suspend_pre(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) @@ -690,7 +690,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) codec_dev->suspend(pdev, state); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->suspend(pdev, cpu_dai); } @@ -726,7 +726,7 @@ static void soc_resume_deferred(struct work_struct *work) machine->resume_pre(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); } @@ -747,13 +747,13 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < machine->num_links; i++) { - struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; + struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; if (dai->dai_ops.digital_mute && dai->playback.active) dai->dai_ops.digital_mute(dai, 0); } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) @@ -803,7 +803,7 @@ static int soc_probe(struct platform_device *pdev) } for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->probe) { ret = cpu_dai->probe(pdev, cpu_dai); if (ret < 0) @@ -838,7 +838,7 @@ platform_err: cpu_dai_err: for (i--; i >= 0; i--) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -867,7 +867,7 @@ static int soc_remove(struct platform_device *pdev) codec_dev->remove(pdev); for (i = 0; i < machine->num_links; i++) { - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai; if (cpu_dai->remove) cpu_dai->remove(pdev, cpu_dai); } @@ -895,8 +895,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { struct snd_soc_codec *codec = socdev->codec; - struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = dai_link->codec_dai; + struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; struct snd_soc_pcm_runtime *rtd; struct snd_pcm *pcm; char new_name[64]; @@ -1211,7 +1211,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; #ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_codec_dai *codec_dai; + struct snd_soc_dai *codec_dai; int i; #endif -- 1.5.4.2

2 1

[alsa-devel] [PATCH] asoc: core - refactored DAPM pin control API.
by Liam Girdwood 07 Jul '08

07 Jul '08

Refactored snd_soc_dapm_set_endpoint() to snd_soc_dapm_enable_pin() and snd_soc_dapm_disable_pin(). Renamed snd_soc_dapm_sync_endpoints() to snd_soc_dapm_sync(). Renamed snd_soc_dapm_get_endpoint_status() to snd_soc_dapm_get_pin_status(). Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 11 ++-- sound/soc/at91/eti_b1_wm8731.c | 10 ++-- sound/soc/codecs/tlv320aic3x.c | 4 +- sound/soc/davinci/davinci-evm.c | 16 +++--- sound/soc/omap/n810.c | 21 ++++-- sound/soc/pxa/corgi.c | 42 +++++++------ sound/soc/pxa/poodle.c | 24 ++++---- sound/soc/pxa/spitz.c | 62 ++++++++++---------- sound/soc/pxa/tosa.c | 30 +++++---- sound/soc/s3c24xx/neo1973_wm8753.c | 116 ++++++++++++++++++------------------ sound/soc/sh/exmmb-ac97.c | 2 +- sound/soc/sh/exmmb-i2s.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/soc-dapm.c | 81 ++++++++++++++++--------- 14 files changed, 227 insertions(+), 196 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index f8223fa..64adbe5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -227,12 +227,11 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); -/* dapm audio endpoint control */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *pin, int status); -int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, - char *pin); -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec); +/* dapm audio pin control and status */ +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_sync(struct snd_soc_codec *codec); /* dapm widget types */ enum snd_soc_dapm_type { diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 83614dd..c8b6bc2 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -216,14 +216,14 @@ static int eti_b1_wm8731_init(struct snd_soc_codec *codec) snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Int Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 528c26a..b735ddb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -29,7 +29,7 @@ * --------------------------------------- * * Hence the machine layer should disable unsupported inputs/outputs by - * snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0), etc. + * snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc. */ #include <linux/module.h> @@ -206,7 +206,7 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, } if (found) - snd_soc_dapm_sync_endpoints(widget->codec); + snd_soc_dapm_sync(widget->codec); } ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 4c70a0e..091eae3 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -103,17 +103,17 @@ static int evm_aic3x_init(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); /* not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* always connected */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line In", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c168a64..182d690 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -49,10 +49,17 @@ static int n810_jack_func; static void n810_ext_control(struct snd_soc_codec *codec) { - snd_soc_dapm_set_endpoint(codec, "Ext Spk", n810_spk_func); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", n810_jack_func); + if (n810_spk_func) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); + + if (n810_jack_func) + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int n810_startup(struct snd_pcm_substream *substream) @@ -204,9 +211,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec) int i, err; /* Not connected */ - snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); - snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); - snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + snd_soc_dapm_disable_pin(codec, "MONO_LOUT"); + snd_soc_dapm_disable_pin(codec, "HPLCOM"); + snd_soc_dapm_disable_pin(codec, "HPRCOM"); /* Add N810 specific controls */ for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { @@ -223,7 +230,7 @@ static int n810_aic33_init(struct snd_soc_codec *codec) /* Set up N810 specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index edeea63..db18ef6 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -50,47 +50,51 @@ static int corgi_spk_func; static void corgi_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic = 0, line = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: - hp = 1; /* set = unmute headphone */ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_MIC: - mic = 1; /* reset = mute headphone */ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_LINE: - line = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case CORGI_HEADSET: - hs = 1; - mic = 1; reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (corgi_spk_func == CORGI_SPK_ON) - spk = 1; - - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic); - snd_soc_dapm_set_endpoint(codec, "Line Jack", line); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int corgi_startup(struct snd_pcm_substream *substream) @@ -285,8 +289,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { @@ -303,7 +307,7 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) /* Set up corgi specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 810f1fe..36cbf69 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -48,8 +48,6 @@ static int poodle_spk_func; static void poodle_ext_control(struct snd_soc_codec *codec) { - int spk = 0; - /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -57,23 +55,23 @@ static void poodle_ext_control(struct snd_soc_codec *codec) POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); } else { locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 0); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); } - if (poodle_spk_func == POODLE_SPK_ON) - spk = 1; - /* set the enpoints to their new connetion states */ - snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); + if (poodle_spk_func == POODLE_SPK_ON) + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + else + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int poodle_startup(struct snd_pcm_substream *substream) @@ -248,9 +246,9 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); - snd_soc_dapm_set_endpoint(codec, "MICIN", 1); + snd_soc_dapm_disable_pin(codec, "LLINEIN"); + snd_soc_dapm_disable_pin(codec, "RLINEIN"); + snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { @@ -267,7 +265,7 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) /* Set up poodle specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 092b5c7..ec18163 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -51,60 +51,60 @@ static int spitz_spk_func; static void spitz_ext_control(struct snd_soc_codec *codec) { if (spitz_spk_func == SPITZ_SPK_ON) - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); else - snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0); + snd_soc_dapm_disable_pin(codec, "Ext Spk"); /* set up jack connection */ switch (spitz_jack_func) { case SPITZ_HP: /* enable and unmute hp jack, disable mic bias */ - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; case SPITZ_HP_OFF: /* jack removed, everything off */ - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); - snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); + snd_soc_dapm_disable_pin(codec, "Mic Jack"); + snd_soc_dapm_disable_pin(codec, "Line Jack"); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); break; } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int spitz_startup(struct snd_pcm_substream *substream) @@ -291,13 +291,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) int i, err; /* NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); - snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONO", 0); + snd_soc_dapm_disable_pin(codec, "RINPUT1"); + snd_soc_dapm_disable_pin(codec, "LINPUT2"); + snd_soc_dapm_disable_pin(codec, "RINPUT2"); + snd_soc_dapm_disable_pin(codec, "LINPUT3"); + snd_soc_dapm_disable_pin(codec, "RINPUT3"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONO"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { @@ -314,7 +314,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) /* Set up spitz specific audio paths */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 465ff0f..dba7689 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -52,29 +52,31 @@ static int tosa_spk_func; static void tosa_ext_control(struct snd_soc_codec *codec) { - int spk = 0, mic_int = 0, hp = 0, hs = 0; - /* set up jack connection */ switch (tosa_jack_func) { case TOSA_HP: - hp = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_MIC_INT: - mic_int = 1; + snd_soc_dapm_enable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_disable_pin(codec, "Headset Jack"); break; case TOSA_HEADSET: - hs = 1; + snd_soc_dapm_disable_pin(codec, "Mic (Internal)"); + snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Headset Jack"); break; } if (tosa_spk_func == TOSA_SPK_ON) - spk = 1; + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); - snd_soc_dapm_set_endpoint(codec, "Speaker", spk); - snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int); - snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); - snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); } static int tosa_startup(struct snd_pcm_substream *substream) @@ -191,8 +193,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) { int i, err; - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "MONOOUT"); /* add tosa specific controls */ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { @@ -209,7 +211,7 @@ static int tosa_ac97_init(struct snd_soc_codec *codec) /* set up tosa specific audio path audio_map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 3485123..f053e85 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -250,77 +250,77 @@ static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) switch (neo1973_scenario) { case NEO_AUDIO_OFF: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; default: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } @@ -511,12 +511,12 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); - snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT4", 0); - snd_soc_dapm_set_endpoint(codec, "LINE1", 0); - snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + snd_soc_dapm_disable_pin(codec, "LOUT2"); + snd_soc_dapm_disable_pin(codec, "ROUT2"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "OUT4"); + snd_soc_dapm_disable_pin(codec, "LINE1"); + snd_soc_dapm_disable_pin(codec, "LINE2"); /* set endpoints to default mode */ @@ -539,7 +539,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) err = snd_soc_dapm_add_routes(codec, dapm_routes, ARRAY_SIZE(dapm_routes)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/sh/exmmb-ac97.c b/sound/soc/sh/exmmb-ac97.c index e8d710f..b0b4a20 100644 --- a/sound/soc/sh/exmmb-ac97.c +++ b/sound/soc/sh/exmmb-ac97.c @@ -46,7 +46,7 @@ static int exm7760_ac97_machine_init(struct snd_soc_codec *codec) { MSG("machine_init() enter\n"); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); MSG("machine_init() leave\n"); return 0; diff --git a/sound/soc/sh/exmmb-i2s.c b/sound/soc/sh/exmmb-i2s.c index db1341e..f253f94 100644 --- a/sound/soc/sh/exmmb-i2s.c +++ b/sound/soc/sh/exmmb-i2s.c @@ -116,7 +116,7 @@ static int exm7760_i2s_machine_init(struct snd_soc_codec *codec) cs4251x_gpio_mode(codec, CS4251X_RXP_6, CS4251X_GPIO_MODE_GPOLOW, 0, 0); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 2f91de8..846d1b3 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -25,7 +25,7 @@ extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) { - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9e83357..0813c08 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -857,8 +857,25 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) } } +static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, + char *pin, int status) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!strcmp(w->name, pin)) { + dbg("dapm: %s: pin %s\n", codec->name, pin); + w->connected = status; + return 0; + } + } + + dbg("dapm: %s: configuring unknown pin %s\n", codec->name, pin); + return -EINVAL; +} + /** - * snd_soc_dapm_sync_endpoints - scan and power dapm paths + * snd_soc_dapm_sync - scan and power dapm paths * @codec: audio codec * * Walks all dapm audio paths and powers widgets according to their @@ -866,11 +883,11 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) * * Returns 0 for success. */ -int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) +int snd_soc_dapm_sync(struct snd_soc_codec *codec) { return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); +EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, const char *sink, const char *control, const char *source) @@ -1418,53 +1435,57 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, } /** - * snd_soc_dapm_set_endpoint - set audio endpoint status - * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) - * @status: point status - * - * Set audio endpoint status - connected or disconnected. + * snd_soc_dapm_enable_pin - enable pin. + * @snd_soc_codec: SoC codec + * @pin: pin name * - * Returns 0 for success else error. + * Enables input/output pin and it's parents or children widgets iff there is + * a valid audio route and active audio stream. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. */ -int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, - char *endpoint, int status) +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) { - struct snd_soc_dapm_widget *w; - - list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) { - w->connected = status; - return 0; - } - } + return snd_soc_dapm_set_pin(codec, pin, 1); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); - return -ENODEV; +/** + * snd_soc_dapm_disable_pin - disable pin. + * @codec: SoC codec + * @pin: pin name + * + * Disables input/output pin and it's parents or children widgets. + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) +{ + return snd_soc_dapm_set_pin(codec, pin, 0); } -EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** - * snd_soc_dapm_get_endpoint_status - get audio endpoint status + * snd_soc_dapm_get_pin_status - get audio pin status * @codec: audio codec - * @endpoint: audio signal endpoint (or start point) + * @pin: audio signal pin endpoint (or start point) * - * Get audio endpoint status - connected or disconnected. + * Get audio pin status - connected or disconnected. * - * Returns status + * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_get_endpoint_status(struct snd_soc_codec *codec, - char *endpoint) +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) { struct snd_soc_dapm_widget *w; list_for_each_entry(w, &codec->dapm_widgets, list) { - if (!strcmp(w->name, endpoint)) + if (!strcmp(w->name, pin)) return w->connected; } return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_endpoint_status); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); /** * snd_soc_dapm_free - free dapm resources -- 1.5.4.2

2 4

[alsa-devel] [PATCH 10/10] asoc: sh - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the SuperH platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/sh/ssi.c | 8 ++++---- 4 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 7a3ce80..9faa126 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -326,7 +326,7 @@ static void camelot_pcm_free(struct snd_pcm *pcm) } static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index b7b676b..df7bc34 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -266,7 +266,7 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE -struct snd_soc_cpu_dai sh4_hac_dai[] = { +struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index 846d1b3..92bfaf4 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -20,7 +20,7 @@ #define IPSEL 0xFE400034 /* platform specific structs can be declared here */ -extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_dai sh4_hac_dai[2]; extern struct snd_soc_platform sh7760_soc_platform; static int machine_init(struct snd_soc_codec *codec) diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 3388bc3..55c3464 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -208,7 +208,7 @@ static int ssi_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, +static int ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; @@ -222,7 +222,7 @@ static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, * This divider is used to generate the SSI_SCK (I2S bitclock) from the * clock at the HAC_BIT_CLK ("oversampling clock") pin. */ -static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +static int ssi_set_clkdiv(struct snd_soc_dai *dai, int did, int div) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr; @@ -245,7 +245,7 @@ static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) return 0; } -static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; unsigned long ssicr = SSIREG(SSICR); @@ -332,7 +332,7 @@ static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) -struct snd_soc_cpu_dai sh4_ssi_dai[] = { +struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, -- 1.5.4.2

1 0

[alsa-devel] [PATCH 09/10] asoc: s3c24xx - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the S3C24xx platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/neo1973_wm8753.c | 12 ++++++------ sound/soc/s3c24xx/s3c2412-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c2412-i2s.h | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- sound/soc/s3c24xx/s3c24xx-ac97.h | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c24xx-i2s.h | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 8 files changed, 29 insertions(+), 29 deletions(-) diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index f053e85..51a4ce3 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -66,8 +66,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int pll_out = 0, bclk = 0; int ret = 0; unsigned long iis_clkrate; @@ -156,7 +156,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -176,7 +176,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; unsigned int pcmdiv = 0; int ret = 0; unsigned long iis_clkrate; @@ -222,7 +222,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -546,7 +546,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) /* * BT Codec DAI */ -static struct snd_soc_cpu_dai bt_dai = { +static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c463a82..ee4676e 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -295,7 +295,7 @@ static inline int s3c2412_snd_is_clkmaster(void) /* * Set S3C2412 I2S DAI format */ -static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -500,7 +500,7 @@ EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); /* * Set S3C2412 Clock source */ -static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); @@ -528,7 +528,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C2412 Clock dividers */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -602,7 +602,7 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); static int s3c2412_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -648,7 +648,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c2412_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; u32 iismod; @@ -676,7 +676,7 @@ static int s3c2412_i2s_suspend(struct platform_device *dev, } static int s3c2412_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -708,7 +708,7 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c2412_i2s_dai = { +struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index 27f48e1..aac08a2 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -24,7 +24,7 @@ extern struct clk *s3c2412_get_iisclk(void); -extern struct snd_soc_cpu_dai s3c2412_i2s_dai; +extern struct snd_soc_dai s3c2412_i2s_dai; struct s3c2412_rate_calc { unsigned int clk_div; /* for prescaler */ diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 533565b..783349b 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -210,7 +210,7 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { }; static int s3c2443_ac97_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { int ret; u32 ac_glbctrl; @@ -262,7 +262,7 @@ static int s3c2443_ac97_probe(struct platform_device *pdev, } static void s3c2443_ac97_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); @@ -274,7 +274,7 @@ static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; @@ -316,7 +316,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -352,7 +352,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_cpu_dai s3c2443_ac97_dai[] = { +struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h index bf03e8e..a96dcad 100644 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -26,6 +26,6 @@ #define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 #endif -extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; +extern struct snd_soc_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 42e96b5..3975242 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -205,7 +205,7 @@ static inline int s3c24xx_snd_is_clkmaster(void) /* * Set S3C24xx I2S DAI format */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -313,7 +313,7 @@ exit_err: /* * Set S3C24xx Clock source */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -339,7 +339,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C24xx Clock dividers */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { u32 reg; @@ -378,7 +378,7 @@ u32 s3c24xx_i2s_get_clockrate(void) EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -411,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c24xx_i2s_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); @@ -426,7 +426,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev, } static int s3c24xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); @@ -449,7 +449,7 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c24xx_i2s_dai = { +struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h index 537b4ec..726d91c 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ b/sound/soc/s3c24xx/s3c24xx-i2s.h @@ -32,6 +32,6 @@ u32 s3c24xx_i2s_get_clockrate(void); -extern struct snd_soc_cpu_dai s3c24xx_i2s_dai; +extern struct snd_soc_dai s3c24xx_i2s_dai; #endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index ef59974..cef79b3 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -429,7 +429,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; static int s3c24xx_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; -- 1.5.4.2

1 0

[alsa-devel] [PATCH 08/10] asoc: pxa - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the PXA platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 4 ++-- sound/soc/pxa/pxa2xx-ac97.c | 14 +++++++------- sound/soc/pxa/pxa2xx-ac97.h | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 14 +++++++------- sound/soc/pxa/pxa2xx-i2s.h | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- 8 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index db18ef6..782afbf 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -123,8 +123,8 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 36cbf69..ce25b6b 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -102,8 +102,8 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index cb94795..b458667 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -283,7 +283,7 @@ static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { #ifdef CONFIG_PM static int pxa2xx_ac97_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { GCR |= GCR_ACLINK_OFF; clk_disable(ac97_clk); @@ -291,7 +291,7 @@ static int pxa2xx_ac97_suspend(struct platform_device *pdev, } static int pxa2xx_ac97_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); @@ -311,7 +311,7 @@ static int pxa2xx_ac97_resume(struct platform_device *pdev, #endif static int pxa2xx_ac97_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { int ret; @@ -373,7 +373,7 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; @@ -387,7 +387,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; @@ -401,7 +401,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -419,7 +419,7 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. */ -struct snd_soc_cpu_dai pxa_ac97_dai[] = { +struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97", .id = 0, diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h index b8ccfee..e390de8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.h +++ b/sound/soc/pxa/pxa2xx-ac97.h @@ -14,7 +14,7 @@ #define PXA2XX_DAI_AC97_AUX 1 #define PXA2XX_DAI_AC97_MIC 2 -extern struct snd_soc_cpu_dai pxa_ac97_dai[3]; +extern struct snd_soc_dai pxa_ac97_dai[3]; /* platform data */ extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 35090c2..9c06553 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -77,7 +77,7 @@ static struct pxa2xx_gpio gpio_bus[] = { static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (!cpu_dai->active) { SACR0 |= SACR0_RST; @@ -98,7 +98,7 @@ static int pxa_i2s_wait(void) return 0; } -static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { /* interface format */ @@ -124,7 +124,7 @@ static int pxa2xx_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { if (clk_id != PXA2XX_I2S_SYSCLK) @@ -140,7 +140,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); @@ -237,7 +237,7 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int pxa2xx_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -255,7 +255,7 @@ static int pxa2xx_i2s_suspend(struct platform_device *dev, } static int pxa2xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { if (!dai->active) return 0; @@ -280,7 +280,7 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai pxa_i2s_dai = { +struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/pxa/pxa2xx-i2s.h b/sound/soc/pxa/pxa2xx-i2s.h index 4435bd9..e2def44 100644 --- a/sound/soc/pxa/pxa2xx-i2s.h +++ b/sound/soc/pxa/pxa2xx-i2s.h @@ -15,6 +15,6 @@ /* I2S clock */ #define PXA2XX_I2S_SYSCLK 0 -extern struct snd_soc_cpu_dai pxa_i2s_dai; +extern struct snd_soc_dai pxa_i2s_dai; #endif diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 01ad7bf..2df03ee 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -330,7 +330,7 @@ static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; -int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ec18163..fd1abc7 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -121,8 +121,8 @@ static int spitz_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int clk = 0; int ret = 0; -- 1.5.4.2

1 0

[alsa-devel] [PATCH 07/10] asoc: omap - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the Omap platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 16 ++++++++-------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 2 +- 4 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 74f4599..d1233c0 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -86,8 +86,8 @@ static int n810_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int err; /* Set codec DAI configuration */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 40d87e6..00b0c9d 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -103,7 +103,7 @@ static const unsigned long omap2420_mcbsp_port[][2] = {}; static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -116,7 +116,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); if (!cpu_dai->active) { @@ -128,7 +128,7 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); int err = 0; @@ -157,7 +157,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; @@ -223,7 +223,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, * This must be called before _set_clkdiv and _set_sysclk since McBSP register * cache is initialized here */ -static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -292,7 +292,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, return 0; } -static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); @@ -347,7 +347,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } -static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -376,7 +376,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, return err; } -struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS] = { +struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = { { .name = "omap-mcbsp-dai", .id = 0, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 9965fd4..ed8afb5 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -44,6 +44,6 @@ enum omap_mcbsp_div { */ #define NUM_LINKS 1 -extern struct snd_soc_cpu_dai omap_mcbsp_dai[NUM_LINKS]; +extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS]; #endif diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 6237020..e092f3d 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -316,7 +316,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -int omap_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; -- 1.5.4.2

1 0

[alsa-devel] [PATCH 06/10] asoc: fsl - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the Freescale PPC platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_dma.h | 2 +- sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++------------ sound/soc/fsl/fsl_ssi.h | 4 ++-- sound/soc/fsl/mpc8610_hpcd.c | 4 ++-- 5 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 78de716..da2bc59 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -282,7 +282,7 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * once for each .dai_link in the machine driver's snd_soc_machine * structure. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_codec_dai *dai, +static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { static u64 fsl_dma_dmamask = DMA_BIT_MASK(32); diff --git a/sound/soc/fsl/fsl_dma.h b/sound/soc/fsl/fsl_dma.h index 430a6ce..385d4a4 100644 --- a/sound/soc/fsl/fsl_dma.h +++ b/sound/soc/fsl/fsl_dma.h @@ -126,7 +126,7 @@ struct fsl_dma_link_descriptor { u8 res[4]; /* Reserved */ } __attribute__ ((aligned(32), packed)); -/* DMA information needed to create a snd_soc_cpu_dai object +/* DMA information needed to create a snd_soc_dai object * * ssi_stx_phys: bus address of SSI STX register to use * ssi_srx_phys: bus address of SSI SRX register to use diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f588545..71bff33 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -82,7 +82,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; - struct snd_soc_cpu_dai cpu_dai; + struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; struct { @@ -479,7 +479,7 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) * @freq: the frequency of the given clock ID, currently ignored * @dir: SND_SOC_CLOCK_IN (clock slave) or SND_SOC_CLOCK_OUT (clock master) */ -static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int fsl_ssi_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -497,7 +497,7 @@ static int fsl_ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, * * @format: one of SND_SOC_DAIFMT_xxx */ -static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) +static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) { return (format == SND_SOC_DAIFMT_I2S) ? 0 : -EINVAL; } @@ -505,7 +505,7 @@ static int fsl_ssi_set_fmt(struct snd_soc_cpu_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ -static struct snd_soc_cpu_dai fsl_ssi_dai_template = { +static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ .channels_min = 2, @@ -569,15 +569,15 @@ static ssize_t fsl_sysfs_ssi_show(struct device *dev, } /** - * fsl_ssi_create_dai: create a snd_soc_cpu_dai structure + * fsl_ssi_create_dai: create a snd_soc_dai structure * - * This function is called by the machine driver to create a snd_soc_cpu_dai + * This function is called by the machine driver to create a snd_soc_dai * structure. The function creates an ssi_private object, which contains - * the snd_soc_cpu_dai. It also creates the sysfs statistics device. + * the snd_soc_dai. It also creates the sysfs statistics device. */ -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) { - struct snd_soc_cpu_dai *fsl_ssi_dai; + struct snd_soc_dai *fsl_ssi_dai; struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr; @@ -588,7 +588,7 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) return NULL; } memcpy(&ssi_private->cpu_dai, &fsl_ssi_dai_template, - sizeof(struct snd_soc_cpu_dai)); + sizeof(struct snd_soc_dai)); fsl_ssi_dai = &ssi_private->cpu_dai; dev_attr = &ssi_private->dev_attr; @@ -623,11 +623,11 @@ struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) EXPORT_SYMBOL_GPL(fsl_ssi_create_dai); /** - * fsl_ssi_destroy_dai: destroy the snd_soc_cpu_dai object + * fsl_ssi_destroy_dai: destroy the snd_soc_dai object * * This function undoes the operations of fsl_ssi_create_dai() */ -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai) +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) { struct fsl_ssi_private *ssi_private = container_of(fsl_ssi_dai, struct fsl_ssi_private, cpu_dai); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index c5ce88e..83b44d7 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -217,8 +217,8 @@ struct fsl_ssi_info { struct device *dev; }; -struct snd_soc_cpu_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); -void fsl_ssi_destroy_dai(struct snd_soc_cpu_dai *fsl_ssi_dai); +struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); +void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai); #endif diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 8820c3f..59d7e49 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -96,8 +96,8 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct mpc8610_hpcd_data *machine_data = rtd->socdev->dev->platform_data; int ret = 0; -- 1.5.4.2

1 0

[alsa-devel] [PATCH 05/10] asoc: codecs - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the codec drivers. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ac97.h | 2 +- sound/soc/codecs/ak4535.c | 8 ++++---- sound/soc/codecs/ak4535.h | 2 +- sound/soc/codecs/cs4270.c | 8 ++++---- sound/soc/codecs/cs4270.h | 2 +- sound/soc/codecs/tlv320aic3x.c | 8 ++++---- sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/codecs/uda1380.c | 6 +++--- sound/soc/codecs/uda1380.h | 2 +- sound/soc/codecs/wm8510.c | 10 +++++----- sound/soc/codecs/wm8510.h | 2 +- sound/soc/codecs/wm8731.c | 8 ++++---- sound/soc/codecs/wm8731.h | 2 +- sound/soc/codecs/wm8750.c | 8 ++++---- sound/soc/codecs/wm8750.h | 2 +- sound/soc/codecs/wm8753.c | 28 ++++++++++++++-------------- sound/soc/codecs/wm8753.h | 2 +- sound/soc/codecs/wm8990.c | 12 ++++++------ sound/soc/codecs/wm8990.h | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9712.h | 2 +- sound/soc/codecs/wm9713.c | 10 +++++----- sound/soc/codecs/wm9713.h | 2 +- 24 files changed, 67 insertions(+), 67 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index e4516f3..61fd96c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -41,7 +41,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ac97_dai = { +struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97, .playback = { diff --git a/sound/soc/codecs/ac97.h b/sound/soc/codecs/ac97.h index 2bf6d69..281aa42 100644 --- a/sound/soc/codecs/ac97.h +++ b/sound/soc/codecs/ac97.h @@ -14,6 +14,6 @@ #define __LINUX_SND_SOC_AC97_H extern struct snd_soc_codec_device soc_codec_dev_ac97; -extern struct snd_soc_codec_dai ac97_dai; +extern struct snd_soc_dai ac97_dai; #endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 469266e..b26003c 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -329,7 +329,7 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec) return 0; } -static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -369,7 +369,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -394,7 +394,7 @@ static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +static int ak4535_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; @@ -436,7 +436,7 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai ak4535_dai = { +struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h index fc686dd..e9fe30e 100644 --- a/sound/soc/codecs/ak4535.h +++ b/sound/soc/codecs/ak4535.h @@ -40,7 +40,7 @@ struct ak4535_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_dai ak4535_dai; extern struct snd_soc_codec_device soc_codec_dev_ak4535; #endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e73fcfd..9deb8c7 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -201,7 +201,7 @@ static struct { * driver what the input settings can be. This would need to be implemented * for stand-alone mode to work. */ -static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -251,7 +251,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, * data for playback only, but ASoC currently does not support different * formats for playback vs. record. */ -static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; @@ -471,7 +471,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, * board does not have the MUTEA or MUTEB pins connected to such circuitry, * then this function will do nothing. */ -static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +static int cs4270_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; int reg6; @@ -667,7 +667,7 @@ error: #endif /* USE_I2C*/ -struct snd_soc_codec_dai cs4270_dai = { +struct snd_soc_dai cs4270_dai = { .name = "CS4270", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h index 0ced49b..adc6cd9 100644 --- a/sound/soc/codecs/cs4270.h +++ b/sound/soc/codecs/cs4270.h @@ -16,7 +16,7 @@ * The ASoC codec DAI structure for the CS4270. Assign this structure to * the .codec_dai field of your machine driver's snd_soc_dai_link structure. */ -extern struct snd_soc_codec_dai cs4270_dai; +extern struct snd_soc_dai cs4270_dai; /* * The ASoC codec device structure for the CS4270. Assign this structure diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 954d39b..b1dce5f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -814,7 +814,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, return 0; } -static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) +static int aic3x_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u8 ldac_reg = aic3x_read_reg_cache(codec, LDAC_VOL) & ~MUTE_ON; @@ -831,7 +831,7 @@ static int aic3x_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -841,7 +841,7 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return 0; } -static int aic3x_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -990,7 +990,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai aic3x_dai = { +struct snd_soc_dai aic3x_dai = { .name = "aic3x", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index e600946..d76c079 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -228,7 +228,7 @@ struct aic3x_setup_data { unsigned int gpio_func[2]; }; -extern struct snd_soc_codec_dai aic3x_dai; +extern struct snd_soc_dai aic3x_dai; extern struct snd_soc_codec_device soc_codec_dev_aic3x; #endif /* _AIC3X_H */ diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 6d5335b..a52d6d9 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -372,7 +372,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) return 0; } -static int uda1380_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -499,7 +499,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) uda1380_write(codec, UDA1380_CLK, clk); } -static int uda1380_mute(struct snd_soc_codec_dai *codec_dai, int mute) +static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; @@ -542,7 +542,7 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai uda1380_dai[] = { +struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", .playback = { diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h index f9d885c..50c603e 100644 --- a/sound/soc/codecs/uda1380.h +++ b/sound/soc/codecs/uda1380.h @@ -83,7 +83,7 @@ struct uda1380_setup_data { #define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ #define UDA1380_DAI_CAPTURE 2 /* capture DAI */ -extern struct snd_soc_codec_dai uda1380_dai[3]; +extern struct snd_soc_dai uda1380_dai[3]; extern struct snd_soc_codec_device soc_codec_dev_uda1380; #endif /* _UDA1380_H */ diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index b549f67..67325fd 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -332,7 +332,7 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -368,7 +368,7 @@ static int wm8510_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Configure WM8510 clock dividers. */ -static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -402,7 +402,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8510_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -510,7 +510,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8510_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8510_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; @@ -554,7 +554,7 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_codec_dai wm8510_dai = { +struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h index c862e7b..f5d2e42 100644 --- a/sound/soc/codecs/wm8510.h +++ b/sound/soc/codecs/wm8510.h @@ -97,7 +97,7 @@ struct wm8510_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8510_dai; +extern struct snd_soc_dai wm8510_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8510; #endif diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 3ff42ad..369d39c 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -318,7 +318,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream) } } -static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; @@ -330,7 +330,7 @@ static int wm8731_mute(struct snd_soc_codec_dai *dai, int mute) return 0; } -static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -349,7 +349,7 @@ static int wm8731_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, } -static int wm8731_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -443,7 +443,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8731_dai = { +struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 5bcab6a..99f2e3c 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -38,7 +38,7 @@ struct wm8731_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8731_dai; +extern struct snd_soc_dai wm8731_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8731; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index eb460c9..e23cb09 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -536,7 +536,7 @@ static inline int get_coeff(int mclk, int rate) return -EINVAL; } -static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -554,7 +554,7 @@ static int wm8750_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, return -EINVAL; } -static int wm8750_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -647,7 +647,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8750_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; @@ -692,7 +692,7 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) -struct snd_soc_codec_dai wm8750_dai = { +struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/wm8750.h b/sound/soc/codecs/wm8750.h index a97a54a..8ef30e6 100644 --- a/sound/soc/codecs/wm8750.h +++ b/sound/soc/codecs/wm8750.h @@ -61,7 +61,7 @@ struct wm8750_setup_data { unsigned short i2c_address; }; -extern struct snd_soc_codec_dai wm8750_dai; +extern struct snd_soc_dai wm8750_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8750; #endif diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be01a73..8604809 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -740,7 +740,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; @@ -863,7 +863,7 @@ static int get_coeff(int mclk, int rate) /* * Clock after PLL and dividers */ -static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -890,7 +890,7 @@ static int wm8753_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -960,7 +960,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, /* * Set's PCM dai fmt and BCLK. */ -static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1026,7 +1026,7 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8753_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1054,7 +1054,7 @@ static int wm8753_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, /* * Set's HiFi DAC format. */ -static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1087,7 +1087,7 @@ static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, /* * Set's I2S DAI format. */ -static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1195,7 +1195,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1210,7 +1210,7 @@ static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_pcm_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0) @@ -1218,7 +1218,7 @@ static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1233,7 +1233,7 @@ static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1250,7 +1250,7 @@ static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return wm8753_i2s_set_dai_fmt(codec_dai, fmt); } -static int wm8753_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8753_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7; @@ -1316,7 +1316,7 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ -static const struct snd_soc_codec_dai wm8753_all_dai[] = { +static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", .id = 1, @@ -1456,7 +1456,7 @@ static const struct snd_soc_codec_dai wm8753_all_dai[] = { }, }; -struct snd_soc_codec_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[2]; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index 95e2a1f..44f5f1f 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -120,7 +120,7 @@ struct wm8753_setup_data { #define WM8753_DAI_HIFI 0 #define WM8753_DAI_VOICE 1 -extern struct snd_soc_codec_dai wm8753_dai[2]; +extern struct snd_soc_dai wm8753_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm8753; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index a1371b7..3ecce51 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1029,7 +1029,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { u16 reg; @@ -1065,7 +1065,7 @@ static int wm8990_set_dai_pll(struct snd_soc_codec_dai *codec_dai, /* * Clock after PLL and dividers */ -static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1078,7 +1078,7 @@ static int wm8990_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, /* * Set's ADC and Voice DAC format. */ -static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1131,7 +1131,7 @@ static int wm8990_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm8990_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1196,7 +1196,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8990_mute(struct snd_soc_codec_dai *dai, int mute) +static int wm8990_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; u16 val; @@ -1329,7 +1329,7 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ -struct snd_soc_codec_dai wm8990_dai = { +struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", .id = 1, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index bf9f882..6bea574 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -825,7 +825,7 @@ struct wm8990_setup_data { #define WM8990_ADCCLK_DIV 2 #define WM8990_BCLK_DIV 3 -extern struct snd_soc_codec_dai wm8990_dai; +extern struct snd_soc_dai wm8990_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8990; #endif /* __WM8990REGISTERDEFS_H__ */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 4739011..9fc8edd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -532,7 +532,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) -struct snd_soc_codec_dai wm9712_dai[] = { +struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9712.h b/sound/soc/codecs/wm9712.h index 719105d..d29e8a1 100644 --- a/sound/soc/codecs/wm9712.h +++ b/sound/soc/codecs/wm9712.h @@ -8,7 +8,7 @@ #define WM9712_DAI_AC97_HIFI 0 #define WM9712_DAI_AC97_AUX 1 -extern struct snd_soc_codec_dai wm9712_dai[2]; +extern struct snd_soc_dai wm9712_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_wm9712; #endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a480618..38d1fe0 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -789,7 +789,7 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -800,7 +800,7 @@ static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, * Tristate the PCM DAI lines, tristate can be disabled by calling * wm9713_set_dai_fmt() */ -static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai, int tristate) { struct snd_soc_codec *codec = codec_dai->codec; @@ -816,7 +816,7 @@ static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, * Configure WM9713 clock dividers. * Voice DAC needs 256 FS */ -static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { struct snd_soc_codec *codec = codec_dai->codec; @@ -858,7 +858,7 @@ static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, return 0; } -static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, +static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1018,7 +1018,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) -struct snd_soc_codec_dai wm9713_dai[] = { +struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", .type = SND_SOC_DAI_AC97_BUS, diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h index d357b6c..63b8d81 100644 --- a/sound/soc/codecs/wm9713.h +++ b/sound/soc/codecs/wm9713.h @@ -46,7 +46,7 @@ #define WM9713_DAI_PCM_VOICE 2 extern struct snd_soc_codec_device soc_codec_dev_wm9713; -extern struct snd_soc_codec_dai wm9713_dai[3]; +extern struct snd_soc_dai wm9713_dai[3]; int wm9713_reset(struct snd_soc_codec *codec, int try_warm); -- 1.5.4.2

1 0

[alsa-devel] [PATCH 04/10] asoc: davinci - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the DaVinci platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/davinci/davinci-evm.c | 4 ++-- sound/soc/davinci/davinci-i2s.c | 14 +++++++------- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 091eae3..4249e6a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,8 +33,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; /* set codec DAI configuration */ diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c3b545c..5ebf1ff 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -147,7 +147,7 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) static int davinci_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; cpu_dai->dma_data = dev->dma_params[substream->stream]; @@ -155,7 +155,7 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream) return 0; } -static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcbsp_dev *dev = cpu_dai->private_data; @@ -296,11 +296,11 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) } static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -358,11 +358,11 @@ err_release_region: } static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -378,7 +378,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -struct snd_soc_cpu_dai davinci_i2s_dai = { +struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 9592d17..c5b0918 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,6 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H -extern struct snd_soc_cpu_dai davinci_i2s_dai; +extern struct snd_soc_dai davinci_i2s_dai; #endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 6a76927..6a5e56a 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -350,7 +350,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret; -- 1.5.4.2

1 0

[alsa-devel] [PATCH 03/10] asoc: at91 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the AT91 platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at91/at91-pcm.c | 6 +++--- sound/soc/at91/at91-ssc.c | 12 ++++++------ sound/soc/at91/at91-ssc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 8 ++++---- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index ccac6bd..d47492b 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -318,7 +318,7 @@ static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, static u64 at91_pcm_dmamask = 0xffffffff; static int at91_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -367,7 +367,7 @@ static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; @@ -392,7 +392,7 @@ static int at91_pcm_suspend(struct platform_device *pdev, } static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index bc35d00..c3625b6 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -281,7 +281,7 @@ static void at91_ssc_shutdown(struct snd_pcm_substream *substream) /* * Record the SSC system clock rate. */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* @@ -303,7 +303,7 @@ static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record the DAI format for use in hw_params(). */ -static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -315,7 +315,7 @@ static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params(). */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -634,7 +634,7 @@ static int at91_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -662,7 +662,7 @@ static int at91_ssc_suspend(struct platform_device *pdev, } static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -700,7 +700,7 @@ static int at91_ssc_resume(struct platform_device *pdev, #define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_cpu_dai at91_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at91-ssc0", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index b188f97..6b7bf38 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -21,7 +21,7 @@ #define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at91_ssc_dai[]; +extern struct snd_soc_dai at91_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index ad971e7..9d42139 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -53,8 +53,8 @@ static struct clk *pllb_clk; static int eti_b1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* cpu clock is the AT91 master clock sent to the SSC */ @@ -87,8 +87,8 @@ static int eti_b1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; #ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE -- 1.5.4.2

1 0

[alsa-devel] [PATCH 02/10] asoc: at32 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for AT32 platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at32/at32-pcm.c | 6 +++--- sound/soc/at32/at32-ssc.c | 12 ++++++------ sound/soc/at32/at32-ssc.h | 2 +- sound/soc/at32/playpaq_wm8510.c | 6 +++--- 4 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c index cf76e89..435f1da 100644 --- a/sound/soc/at32/at32-pcm.c +++ b/sound/soc/at32/at32-pcm.c @@ -366,7 +366,7 @@ static struct snd_pcm_ops at32_pcm_ops = { static u64 at32_pcm_dmamask = 0xffffffff; static int at32_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -422,7 +422,7 @@ static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; @@ -447,7 +447,7 @@ static int at32_pcm_suspend(struct platform_device *pdev, static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c index 0ca4410..4ef6492 100644 --- a/sound/soc/at32/at32-ssc.c +++ b/sound/soc/at32/at32-ssc.c @@ -336,7 +336,7 @@ static void at32_ssc_shutdown(struct snd_pcm_substream *substream) /* * Set the SSC system clock rate */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* TODO: What the heck do I do here? */ @@ -348,7 +348,7 @@ static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record DAI format for use by hw_params() */ -static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -362,7 +362,7 @@ static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params() */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -670,7 +670,7 @@ static int at32_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; @@ -699,7 +699,7 @@ static int at32_ssc_suspend(struct platform_device *pdev, static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; u32 cr; @@ -746,7 +746,7 @@ static int at32_ssc_resume(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) -struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at32-ssc0", .id = 0, diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h index 3c6901a..3c052db 100644 --- a/sound/soc/at32/at32-ssc.h +++ b/sound/soc/at32/at32-ssc.h @@ -52,7 +52,7 @@ struct at32_ssc_info { #define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at32_ssc_dai[]; +extern struct snd_soc_dai at32_ssc_dai[]; diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index d6b9fd5..18ac7d7 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -82,7 +82,7 @@ static struct clk *_pll0; #if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( struct snd_pcm_hw_params *params, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; @@ -132,8 +132,8 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; unsigned int pll_out = 0, bclk = 0, mclk_div = 0; -- 1.5.4.2

1 0

[alsa-devel] Repeated calls to trigger and prepare in ALSA driver
by DSP Tech 07 Jul '08

07 Jul '08

Hi All, I am trying to develop ALSA PCM driver for my hardware board which has I2S-DMA interface. I ran a OSS test application. I find that trigger() and prepare() calls are being called repeatedly. Consequently, the DMA tries to reset and start playback for every buffer. Have any of you faced this problem? Any clue in this matter would be helpful. Thanks _________________________________________________________________ Tried the new MSN Messenger? It’s cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

2 1

[alsa-devel] ASOC: codec_reg_show and variable length registers
by Jon Smirl 07 Jul '08

07 Jul '08

codec_reg_show() can't handle my TAS5504 codec. The codec has 104 (sparse out of 256) registers with variable lengths from 1 to 32 bytes. Should the register cache really be part of the core? The core could still make the sysfs entries and use a callback to print them. reg_cache_size/reg_cache_step would then become private. It's the sprintf that is the problem. I need to push that into my driver so that I can print out variable lengths. Or we need to make the core use an indirect table that encodes the lengths. -- Jon Smirl jonsmirl(a)gmail.com

2 3

[alsa-devel] HDA Intel ALC885 - Need new or edited model option
by Constrabus Gegami 06 Jul '08

06 Jul '08

I need a new or edited model option for this card: HDA Intel card in Alienware m15x laptop That uses codec: ALC885 Detailed specifications on the card: http://www.alienware.com/Configurator_Pages/Details_Pages/Details_Page.aspx… alsa-info.sh genterated output: http://pastebin.ca/1061774 (i switched back to no model options before i ran the script, i dont know if it leaves behind anything or if it even matters) Jack Observations: -1x auto-sensing headphone jack -1x auto-sensing microphone jack All current available model options either present me with: -working switchable internal speakers with non working headphone jack -working switchable headphone jack with non working internal speakers -non working headphone jack and internal speakers -working non switchable internal speakers with non working heaphone -working non switchable headphones with non working speakers -no working sound What is needed: - working switchable internal speakers with working headphone and mic jack (preferably switchable) Distro: Gentoo amd64 using alsa driver package (not in kernel modules) I hope that this will benefit others as well as I. If any more information is needed, please email me. -- --Constrabus

1 0

[alsa-devel] [PATCH] ASoC: Fix warning from strict_strtoul()
by Mark Brown 06 Jul '08

06 Jul '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9e83357..456d78a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -808,7 +808,11 @@ static ssize_t dapm_pop_time_store(struct device *dev, const char *buf, size_t count) { - if (strict_strtoul(buf, 10, &pop_time) < 0) + unsigned long val; + + if (strict_strtoul(buf, 10, &val) >= 0) + pop_time = val; + else printk(KERN_ERR "Unable to parse pop_time setting\n"); return count; -- 1.5.6

2 4

[alsa-devel] EMU10K1 Problem. Im stumped.
by Travis Place 05 Jul '08

05 Jul '08

Just had a user on #alsa with an EMU10K1 problem. To summarize, the card shows up in linux (but doesnt work), yet the card works perfectly in windows. The correct modules are loading, but ALSA doesnt see the card. It had the user unload/reload snd-emu10k1 and check dmesg output, and the relevant result is as follows. EMU10K1_Audigy: probe of 0000:00:0a.0 failed with error -16 There complete alsa-info.sh output can be found at http://pastebin.ca/1062377 I have had the user run this both with and without the onboard AC97 card enabled (just to be sure). What seems odd to me, is that the card is at 00:0a.0 and the SB Live revision is showing as (rev 0a). Just doesnt look right to me.. The card is an Creative SBLive! 'SB0220'. I have one of these exact cards here, if its needed for testing, but its never caused me any issues. Anyway, any help would be appreciated. Thanks, Travis Place (wishie)

2 1

[alsa-devel] Sound distortions with Realtek ALC889A
by Matthias Dahl 04 Jul '08

04 Jul '08

Hello. Due to a hw defect, I recently bought a Gigabyte X48-DQ6 board which utilizes a Realtek ALC889A for sound output. At first glance everything seemed fine but during day-to-day use I figured that during playback of a 5.1 file, I've to turn up the volume quite a bit to match normal 2.0 playback or the volume levels of my old board (ADI1988B). And to make matters worse, the surround channel produces (static) noise on the center channel. If I mute every channel but the surround channel, the distortions are clearly hearable. During normal 5.1 playback, they are only hearable during scenes with a lot of talk or quieter scenes which is still very annoying unfortunately. Muting the surround channel also stops the distortions on the center channel. Also the model is not properly recognized, I've to pass model=6stack-dig to get all channels properly working with digital out. By the way, this is with a 2.6.26-rc7 x86_64 kernel running a alsa git checkout from last monday. I'd appreciate any help or hints I can get. Thanks in advance. Best regards, Matthias Dahl

1 1

[alsa-devel] Using BD in place of DMA in soc driver
by dinesh 04 Jul '08

04 Jul '08

Hi i want to use mpc8323 ucc tdm as sound soc device but i dont know how to configure its buffer descriptor because every soc device available uses DMA. Please help.

2 2

[alsa-devel] add TV-CARD to alsa driver
by g000fy g000fy 04 Jul '08

04 Jul '08

Hi: im responing to: "If you have an unknown card, please mail the ID and board name to <alsa-devel[AT]alsa-project[DOT]org>, regardless of whether audio capture works or not, so that future versions of this driver know about your card." i have an ATI TV WONDER VE (PCI) card. im using slackware 12.1 with default kernel "root@cpu99:~# uname -a Linux cpu99 2.6.24.5-smp #2 SMP Wed Apr 30 13:41:38 CDT 2008 i686 AMD Athlon(tm) XP 2600+ AuthenticAMD GNU/Linux" im following this project here: http://www.domenech.org/bt878a-adc/alsa-e.htm if you wouldnt mind telling me which version of alsa and with what config settings to get that to work i would much appreciate it. "dmesg | grep alsa" reports the following root@cpu99:~# dmesg | grep alsa ALSA /root/building_area/ALSA/alsa-driver-1.0.16/alsa-driver-1.0.16/pci/bt87x.c:876: unknown card 0x878-0x1002:0x03 ALSA /root/building_area/ALSA/alsa-driver-1.0.16/alsa-driver-1.0.16/pci/bt87x.c:879: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> ALSA /root/building_area/ALSA/alsa-driver-1.0.16/alsa-driver-1.0.16/pci/bt87x.c:947: bt87x0: Using board 0, analog, digital (rate 32000 Hz) root@cpu99:~# here is my /etc/modprobe.d/sound contents: alias snd-card-0 snd-intel8x0 alias sound-slot-0 snd-intel8x0 alias snd-card-1 snd-bt87x alias sound-slot-1 snd-bt87x options snd-bt87x index=1 load_all=1 when i built the driver i used: "./configure --with-cards=intel8x0,bt87x" as per the project guide i updated these alsa alsa-driver-1.0.16, alsa-lib-1.0.16, alsa-utils-1.0.16 and alsa-oss-1.0.15 here is some more info http://img53.imageshack.us/img53/6453/snapshot2fh8.jpg http://img124.imageshack.us/img124/6185/alsamixerc1ok3.jpg http://www.imagecoast.com/gallery.php?u=en1gma&g=tv-card oot@cpu99:~# ls /dev/snd* controlC0 controlC1 pcmC0D0c pcmC0D0p pcmC0D1c pcmC0D2p pcmC1D0c pcmC1D1c seq timer root@cpu99:~# ls /dev/mixer* /dev/mixer /dev/mixer1 root@cpu99:~# ls /dev/dsp* /dev/dsp /dev/dsp1 root@cpu99:~# cat /proc/asound/cards 0 [nForce2 ]: NFORCE - NVidia nForce2 NVidia nForce2 with ALC650F at irq 17 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xe4000000, irq 19 basically i just want the oscilloscope project to work. I followed the oscilloscope project guide pretty good. http://www.domenech.org/bt878a-adc/alsa-e.htm patched "aplay.c" configured the "alsa-driver" like it said. alsamixer -c1, alsaconf, kmix, audacity and baudline dont pick up the "FM", "CAPTURE", "MIC" and "CAPTURE BOOST" i think. anyhow i apprecaite your work and if you could help me on this i would apprecaite it. _________________________________________________________________ Watch “Cause Effect,” a show about real people making a real difference. Learn more. http://im.live.com/Messenger/IM/MTV/?source=text_watchcause

1 0

[alsa-devel] GIT repo (connection refused)
by Aldrin Martoq 04 Jul '08

04 Jul '08

Hello! http://www.alsa-project.org/main/index.php/GIT_Server#Occasional_Developers Says but doesn't work: $ git clone git://git.alsa-project.org/alsa-driver.git alsa-driver Initialized empty Git repository in /tmp/alsa-driver/.git/ git.alsa-project.org[0: 212.20.107.51]: errno=Connection refused fatal: unable to connect a socket (Connection refused) fetch-pack from 'git://git.alsa-project.org/alsa-driver.git' failed. Well, this works: git clone http://git.alsa-project.org/http/alsa-driver.git alsa-driver Is this the right repository for submitting patches? Thanks! -- Aldrin Martoq <amartoq(a)dcc.uchile.cl> http://aldrinvideopodcast.podshow.com/

3 4

[alsa-devel] Fixed period
by Gustavo da Silva Serra 03 Jul '08

03 Jul '08

Why is that the emu10k1 driver has the number of capture periods fixed to 2? Thanks.

2 2

[alsa-devel] [PATCH 2/2] wss: use stuct snd_wss instead of snd_ad1848
by Krzysztof Helt 03 Jul '08

03 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The snd_wss is superset of the snd_ad1848 so kill the latter and replace it with the snd_wss. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Another step toward unification of all WSS compatible chips handling. diff -urp linux-alsa/include/sound/ad1848.h linux-mm/include/sound/ad1848.h --- linux-alsa/include/sound/ad1848.h 2008-06-15 13:57:46.000000000 +0200 +++ linux-mm/include/sound/ad1848.h 2008-06-30 20:51:00.072617979 +0200 @@ -127,48 +127,22 @@ #define AD1848_THINKPAD_CTL_PORT2 0x15e9 #define AD1848_THINKPAD_CS4248_ENABLE_BIT 0x02 -struct snd_ad1848 { - unsigned long port; /* i/o port */ - struct resource *res_port; - int irq; /* IRQ line */ - int dma; /* data DMA */ - unsigned short version; /* version of CODEC chip */ - unsigned short mode; /* see to AD1848_MODE_XXXX */ - unsigned short hardware; /* see to AD1848_HW_XXXX */ - unsigned short single_dma:1; /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ - - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - struct snd_card *card; - - unsigned char image[32]; /* SGalaxy needs an access to extended registers */ - int mce_bit; - int calibrate_mute; - int dma_size; - int thinkpad_flag; /* Thinkpad CS4248 needs some extra help */ - -#ifdef CONFIG_PM - void (*suspend)(struct snd_ad1848 *chip); - void (*resume)(struct snd_ad1848 *chip); -#endif - - spinlock_t reg_lock; -}; +#include "wss.h" /* exported functions */ -void snd_ad1848_out(struct snd_ad1848 *chip, unsigned char reg, unsigned char value); +void snd_ad1848_out(struct snd_wss *chip, unsigned char reg, + unsigned char value); int snd_ad1848_create(struct snd_card *card, unsigned long port, int irq, int dma, unsigned short hardware, - struct snd_ad1848 ** chip); + struct snd_wss **chip); -int snd_ad1848_pcm(struct snd_ad1848 * chip, int device, struct snd_pcm **rpcm); +int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); const struct snd_pcm_ops *snd_ad1848_get_pcm_ops(int direction); -int snd_ad1848_mixer(struct snd_ad1848 * chip); +int snd_ad1848_mixer(struct snd_wss *chip); /* exported mixer stuffs */ enum { AD1848_MIX_SINGLE, AD1848_MIX_DOUBLE, AD1848_MIX_CAPTURE }; @@ -213,6 +187,7 @@ struct ad1848_mix_elem { .private_value = AD1848_MIXVAL_DOUBLE(left_reg, right_reg, shift_left, shift_right, mask, invert), \ .tlv = xtlv } -int snd_ad1848_add_ctl_elem(struct snd_ad1848 *chip, const struct ad1848_mix_elem *c); +int snd_ad1848_add_ctl_elem(struct snd_wss *chip, + const struct ad1848_mix_elem *c); #endif /* __SOUND_AD1848_H */ diff -urp linux-alsa/include/sound/wss.h linux-mm/include/sound/wss.h --- linux-alsa/include/sound/wss.h 2008-06-30 20:44:04.150659912 +0200 +++ linux-mm/include/sound/wss.h 2008-06-30 20:23:37.161281486 +0200 @@ -78,7 +78,8 @@ struct snd_wss { unsigned short hardware; /* see to WSS_HW_XXXX */ unsigned short hwshare; /* shared resources */ unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ - ebus_flag:1; /* SPARC: EBUS present */ + ebus_flag:1, /* SPARC: EBUS present */ + thinkpad_flag:1; /* Thinkpad CS4248 needs extra help */ struct snd_card *card; struct snd_pcm *pcm; diff -urp linux-alsa/sound/isa/ad1848/ad1848.c linux-mm/sound/isa/ad1848/ad1848.c --- linux-alsa/sound/isa/ad1848/ad1848.c 2008-06-30 20:44:04.202659855 +0200 +++ linux-mm/sound/isa/ad1848/ad1848.c 2008-06-30 21:02:19.840898902 +0200 @@ -87,7 +87,7 @@ static int __devinit snd_ad1848_match(st static int __devinit snd_ad1848_probe(struct device *dev, unsigned int n) { struct snd_card *card; - struct snd_ad1848 *chip; + struct snd_wss *chip; struct snd_pcm *pcm; int error; @@ -142,7 +142,7 @@ static int __devexit snd_ad1848_remove(s static int snd_ad1848_suspend(struct device *dev, unsigned int n, pm_message_t state) { struct snd_card *card = dev_get_drvdata(dev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->suspend(chip); @@ -152,7 +152,7 @@ static int snd_ad1848_suspend(struct dev static int snd_ad1848_resume(struct device *dev, unsigned int n) { struct snd_card *card = dev_get_drvdata(dev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; chip->resume(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D0); diff -urp linux-alsa/sound/isa/ad1848/ad1848_lib.c linux-mm/sound/isa/ad1848/ad1848_lib.c --- linux-alsa/sound/isa/ad1848/ad1848_lib.c 2008-06-15 12:11:37.000000000 +0200 +++ linux-mm/sound/isa/ad1848/ad1848_lib.c 2008-06-30 21:06:39.307115291 +0200 @@ -98,7 +98,7 @@ static unsigned char snd_ad1848_original * Basic I/O functions */ -static void snd_ad1848_wait(struct snd_ad1848 *chip) +static void snd_ad1848_wait(struct snd_wss *chip) { int timeout; @@ -109,7 +109,7 @@ static void snd_ad1848_wait(struct snd_a } } -void snd_ad1848_out(struct snd_ad1848 *chip, +void snd_ad1848_out(struct snd_wss *chip, unsigned char reg, unsigned char value) { @@ -128,7 +128,7 @@ void snd_ad1848_out(struct snd_ad1848 *c EXPORT_SYMBOL(snd_ad1848_out); -static void snd_ad1848_dout(struct snd_ad1848 *chip, +static void snd_ad1848_dout(struct snd_wss *chip, unsigned char reg, unsigned char value) { snd_ad1848_wait(chip); @@ -137,7 +137,7 @@ static void snd_ad1848_dout(struct snd_a mb(); } -static unsigned char snd_ad1848_in(struct snd_ad1848 *chip, unsigned char reg) +static unsigned char snd_ad1848_in(struct snd_wss *chip, unsigned char reg) { snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG @@ -152,7 +152,7 @@ static unsigned char snd_ad1848_in(struc #if 0 -static void snd_ad1848_debug(struct snd_ad1848 *chip) +static void snd_ad1848_debug(struct snd_wss *chip) { printk("AD1848 REGS: INDEX = 0x%02x ", inb(AD1848P(chip, REGSEL))); printk(" STATUS = 0x%02x\n", inb(AD1848P(chip, STATUS))); @@ -180,7 +180,7 @@ static void snd_ad1848_debug(struct snd_ * AD1848 detection / MCE routines */ -static void snd_ad1848_mce_up(struct snd_ad1848 *chip) +static void snd_ad1848_mce_up(struct snd_wss *chip) { unsigned long flags; int timeout; @@ -200,7 +200,7 @@ static void snd_ad1848_mce_up(struct snd spin_unlock_irqrestore(&chip->reg_lock, flags); } -static void snd_ad1848_mce_down(struct snd_ad1848 *chip) +static void snd_ad1848_mce_down(struct snd_wss *chip) { unsigned long flags, timeout; int reg; @@ -268,7 +268,7 @@ static unsigned int snd_ad1848_get_count return size; } -static int snd_ad1848_trigger(struct snd_ad1848 *chip, unsigned char what, +static int snd_ad1848_trigger(struct snd_wss *chip, unsigned char what, int channel, int cmd) { int result = 0; @@ -337,7 +337,7 @@ static unsigned char snd_ad1848_get_form return rformat; } -static void snd_ad1848_calibrate_mute(struct snd_ad1848 *chip, int mute) +static void snd_ad1848_calibrate_mute(struct snd_wss *chip, int mute) { unsigned long flags; @@ -361,7 +361,8 @@ static void snd_ad1848_calibrate_mute(st spin_unlock_irqrestore(&chip->reg_lock, flags); } -static void snd_ad1848_set_data_format(struct snd_ad1848 *chip, struct snd_pcm_hw_params *hw_params) +static void snd_ad1848_set_data_format(struct snd_wss *chip, + struct snd_pcm_hw_params *hw_params) { if (hw_params == NULL) { chip->image[AD1848_DATA_FORMAT] = 0x20; @@ -373,7 +374,7 @@ static void snd_ad1848_set_data_format(s // snd_printk(">>> pmode = 0x%x, dfr = 0x%x\n", pstr->mode, chip->image[AD1848_DATA_FORMAT]); } -static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode) +static int snd_ad1848_open(struct snd_wss *chip, unsigned int mode) { unsigned long flags; @@ -424,7 +425,7 @@ static int snd_ad1848_open(struct snd_ad return 0; } -static void snd_ad1848_close(struct snd_ad1848 *chip) +static void snd_ad1848_close(struct snd_wss *chip) { unsigned long flags; @@ -464,21 +465,21 @@ static void snd_ad1848_close(struct snd_ static int snd_ad1848_playback_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); return snd_ad1848_trigger(chip, AD1848_PLAYBACK_ENABLE, SNDRV_PCM_STREAM_PLAYBACK, cmd); } static int snd_ad1848_capture_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); return snd_ad1848_trigger(chip, AD1848_CAPTURE_ENABLE, SNDRV_PCM_STREAM_CAPTURE, cmd); } static int snd_ad1848_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); unsigned long flags; int err; @@ -502,15 +503,16 @@ static int snd_ad1848_playback_hw_free(s static int snd_ad1848_playback_prepare(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma_size = size; + chip->p_dma_size = size; chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO); - snd_dma_program(chip->dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); + snd_dma_program(chip->dma1, runtime->dma_addr, size, + DMA_MODE_WRITE | DMA_AUTOINIT); count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; spin_lock_irqsave(&chip->reg_lock, flags); snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); @@ -522,7 +524,7 @@ static int snd_ad1848_playback_prepare(s static int snd_ad1848_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); unsigned long flags; int err; @@ -546,15 +548,16 @@ static int snd_ad1848_capture_hw_free(st static int snd_ad1848_capture_prepare(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma_size = size; + chip->c_dma_size = size; chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); - snd_dma_program(chip->dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); + snd_dma_program(chip->dma2, runtime->dma_addr, size, + DMA_MODE_READ | DMA_AUTOINIT); count = snd_ad1848_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; spin_lock_irqsave(&chip->reg_lock, flags); snd_ad1848_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); @@ -565,7 +568,7 @@ static int snd_ad1848_capture_prepare(st static irqreturn_t snd_ad1848_interrupt(int irq, void *dev_id) { - struct snd_ad1848 *chip = dev_id; + struct snd_wss *chip = dev_id; if ((chip->mode & AD1848_MODE_PLAY) && chip->playback_substream && (chip->mode & AD1848_MODE_RUNNING)) @@ -579,23 +582,23 @@ static irqreturn_t snd_ad1848_interrupt( static snd_pcm_uframes_t snd_ad1848_playback_pointer(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); size_t ptr; if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) return 0; - ptr = snd_dma_pointer(chip->dma, chip->dma_size); + ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } static snd_pcm_uframes_t snd_ad1848_capture_pointer(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); size_t ptr; if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE)) return 0; - ptr = snd_dma_pointer(chip->dma, chip->dma_size); + ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -603,8 +606,8 @@ static snd_pcm_uframes_t snd_ad1848_capt */ -static void snd_ad1848_thinkpad_twiddle(struct snd_ad1848 *chip, int on) { - +static void snd_ad1848_thinkpad_twiddle(struct snd_wss *chip, int on) +{ int tmp; if (!chip->thinkpad_flag) return; @@ -624,14 +627,14 @@ static void snd_ad1848_thinkpad_twiddle( } #ifdef CONFIG_PM -static void snd_ad1848_suspend(struct snd_ad1848 *chip) +static void snd_ad1848_suspend(struct snd_wss *chip) { snd_pcm_suspend_all(chip->pcm); if (chip->thinkpad_flag) snd_ad1848_thinkpad_twiddle(chip, 0); } -static void snd_ad1848_resume(struct snd_ad1848 *chip) +static void snd_ad1848_resume(struct snd_wss *chip) { int i; @@ -651,7 +654,7 @@ static void snd_ad1848_resume(struct snd } #endif /* CONFIG_PM */ -static int snd_ad1848_probe(struct snd_ad1848 * chip) +static int snd_ad1848_probe(struct snd_wss *chip) { unsigned long flags; int i, id, rev, ad1847; @@ -775,7 +778,7 @@ static struct snd_pcm_hardware snd_ad184 static int snd_ad1848_playback_open(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -783,15 +786,15 @@ static int snd_ad1848_playback_open(stru return err; chip->playback_substream = substream; runtime->hw = snd_ad1848_playback; - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.period_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); return 0; } static int snd_ad1848_capture_open(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -799,15 +802,15 @@ static int snd_ad1848_capture_open(struc return err; chip->capture_substream = substream; runtime->hw = snd_ad1848_capture; - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma, &runtime->hw.period_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); return 0; } static int snd_ad1848_playback_close(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); chip->mode &= ~AD1848_MODE_PLAY; chip->playback_substream = NULL; @@ -817,7 +820,7 @@ static int snd_ad1848_playback_close(str static int snd_ad1848_capture_close(struct snd_pcm_substream *substream) { - struct snd_ad1848 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); chip->mode &= ~AD1848_MODE_CAPTURE; chip->capture_substream = NULL; @@ -825,14 +828,14 @@ static int snd_ad1848_capture_close(stru return 0; } -static int snd_ad1848_free(struct snd_ad1848 *chip) +static int snd_ad1848_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); - if (chip->dma >= 0) { - snd_dma_disable(chip->dma); - free_dma(chip->dma); + if (chip->dma1 >= 0) { + snd_dma_disable(chip->dma1); + free_dma(chip->dma1); } kfree(chip); return 0; @@ -840,11 +843,11 @@ static int snd_ad1848_free(struct snd_ad static int snd_ad1848_dev_free(struct snd_device *device) { - struct snd_ad1848 *chip = device->device_data; + struct snd_wss *chip = device->device_data; return snd_ad1848_free(chip); } -static const char *snd_ad1848_chip_id(struct snd_ad1848 *chip) +static const char *snd_ad1848_chip_id(struct snd_wss *chip) { switch (chip->hardware) { case AD1848_HW_AD1847: return "AD1847"; @@ -859,12 +862,12 @@ int snd_ad1848_create(struct snd_card *c unsigned long port, int irq, int dma, unsigned short hardware, - struct snd_ad1848 ** rchip) + struct snd_wss **rchip) { static struct snd_device_ops ops = { .dev_free = snd_ad1848_dev_free, }; - struct snd_ad1848 *chip; + struct snd_wss *chip; int err; *rchip = NULL; @@ -875,7 +878,9 @@ int snd_ad1848_create(struct snd_card *c chip->card = card; chip->port = port; chip->irq = -1; - chip->dma = -1; + chip->dma1 = -1; + chip->dma2 = -1; + chip->single_dma = 1; chip->hardware = hardware; memcpy(&chip->image, &snd_ad1848_original_image, sizeof(snd_ad1848_original_image)); @@ -895,7 +900,8 @@ int snd_ad1848_create(struct snd_card *c snd_ad1848_free(chip); return -EBUSY; } - chip->dma = dma; + chip->dma1 = dma; + chip->dma2 = dma; if (hardware == AD1848_HW_THINKPAD) { chip->thinkpad_flag = 1; @@ -947,7 +953,7 @@ static struct snd_pcm_ops snd_ad1848_cap .pointer = snd_ad1848_capture_pointer, }; -int snd_ad1848_pcm(struct snd_ad1848 *chip, int device, struct snd_pcm **rpcm) +int snd_ad1848_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; @@ -964,7 +970,9 @@ int snd_ad1848_pcm(struct snd_ad1848 *ch snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, chip->dma > 3 ? 128*1024 : 64*1024); + 64 * 1024, + chip->dma1 > 3 ? + 128 * 1024 : 64 * 1024); chip->pcm = pcm; if (rpcm) @@ -1003,7 +1011,7 @@ static int snd_ad1848_info_mux(struct sn static int snd_ad1848_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); @@ -1015,7 +1023,7 @@ static int snd_ad1848_get_mux(struct snd static int snd_ad1848_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; unsigned short left, right; int change; @@ -1049,7 +1057,7 @@ static int snd_ad1848_info_single(struct static int snd_ad1848_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -1066,7 +1074,7 @@ static int snd_ad1848_get_single(struct static int snd_ad1848_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -1100,7 +1108,7 @@ static int snd_ad1848_info_double(struct static int snd_ad1848_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -1122,7 +1130,7 @@ static int snd_ad1848_get_double(struct static int snd_ad1848_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_ad1848 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -1159,7 +1167,7 @@ static int snd_ad1848_put_double(struct /* */ -int snd_ad1848_add_ctl_elem(struct snd_ad1848 *chip, +int snd_ad1848_add_ctl_elem(struct snd_wss *chip, const struct ad1848_mix_elem *c) { static struct snd_kcontrol_new newctls[] = { @@ -1227,7 +1235,7 @@ AD1848_SINGLE_TLV("Loopback Capture Volu db_scale_6bit), }; -int snd_ad1848_mixer(struct snd_ad1848 *chip) +int snd_ad1848_mixer(struct snd_wss *chip) { struct snd_card *card; struct snd_pcm *pcm; diff -urp linux-alsa/sound/isa/cmi8330.c linux-mm/sound/isa/cmi8330.c --- linux-alsa/sound/isa/cmi8330.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-mm/sound/isa/cmi8330.c 2008-06-30 20:30:25.363059784 +0200 @@ -151,7 +151,7 @@ struct snd_cmi8330 { struct pnp_dev *play; #endif struct snd_card *card; - struct snd_ad1848 *wss; + struct snd_wss *wss; struct snd_sb *sb; struct snd_pcm *pcm; diff -urp linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-mm/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-30 20:44:04.386667569 +0200 +++ linux-mm/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-30 21:12:15.015144288 +0200 @@ -702,14 +702,10 @@ static int __devinit snd_opti9xx_probe(s static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; struct snd_opti9xx *chip = card->private_data; -#if defined(CS4231) || defined(OPTi93X) struct snd_wss *codec; #ifdef CS4231 struct snd_timer *timer; #endif -#else - struct snd_ad1848 *codec; -#endif struct snd_pcm *pcm; struct snd_rawmidi *rmidi; struct snd_hwdep *synth; diff -urp linux-alsa/sound/isa/sc6000.c linux-mm/sound/isa/sc6000.c --- linux-alsa/sound/isa/sc6000.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-mm/sound/isa/sc6000.c 2008-06-30 20:36:53.332343377 +0200 @@ -397,7 +397,7 @@ static int __devinit sc6000_init_board(c return 0; } -static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip) +static int __devinit snd_sc6000_mixer(struct snd_wss *chip) { struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; @@ -483,7 +483,7 @@ static int __devinit snd_sc6000_probe(st int xirq = irq[dev]; int xdma = dma[dev]; struct snd_card *card; - struct snd_ad1848 *chip; + struct snd_wss *chip; struct snd_opl3 *opl3; char __iomem *vport; char __iomem *vmss_port; diff -urp linux-alsa/sound/isa/sgalaxy.c linux-mm/sound/isa/sgalaxy.c --- linux-alsa/sound/isa/sgalaxy.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-mm/sound/isa/sgalaxy.c 2008-06-30 20:38:56.415298307 +0200 @@ -180,7 +180,7 @@ AD1848_DOUBLE("Aux Playback Switch", 0, AD1848_DOUBLE("Aux Playback Volume", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) }; -static int __devinit snd_sgalaxy_mixer(struct snd_ad1848 *chip) +static int __devinit snd_sgalaxy_mixer(struct snd_wss *chip) { struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; @@ -237,7 +237,7 @@ static int __devinit snd_sgalaxy_probe(s static int possible_dmas[] = {1, 3, 0, -1}; int err, xirq, xdma1; struct snd_card *card; - struct snd_ad1848 *chip; + struct snd_wss *chip; card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) @@ -312,7 +312,7 @@ static int snd_sgalaxy_suspend(struct de pm_message_t state) { struct snd_card *card = dev_get_drvdata(pdev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->suspend(chip); @@ -322,7 +322,7 @@ static int snd_sgalaxy_suspend(struct de static int snd_sgalaxy_resume(struct device *pdev, unsigned int n) { struct snd_card *card = dev_get_drvdata(pdev); - struct snd_ad1848 *chip = card->private_data; + struct snd_wss *chip = card->private_data; chip->resume(chip); snd_ad1848_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); ---------------------------------------------------------------------- Masz czas tylko do 10 lipca. Przystap on-line do ING OFE! Sprawdz >> http://link.interia.pl/f1e54

3 5

[alsa-devel] Alsa 32-64 bit issue
by Wolfgang Rosenauer 03 Jul '08

03 Jul '08

Hi, I'm new to to this list (and also Alsa) so please forgive me some newbie questions you probably find ;-) I have an application which does simple PCM output to Alsa. It works so far (if pulseaudio is not used but that's another issue). But if I try to run the app (which is compiled for 32bit) on a 64bit Linux system it fails at snd_pcm_start() with the following error: "DatenÃ¼bergabe unterbrochen (broken pipe)" So is that an app, system or kernel issue? Sound output from other 32bit applications (notably flash) works in general so I guess something in the app is broken? What could be the issue? Thanks, Wolfgang

3 5

[alsa-devel] [PATCH 1/2] ASoC: Advertise 16000Hz rate for WM9713 PCM interface
by Mark Brown 02 Jul '08

02 Jul '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9713.c | 19 ++++++++++++++----- 1 files changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 0db9637..a480618 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1001,9 +1001,18 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } -#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) #define WM9713_PCM_FORMATS \ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ @@ -1051,13 +1060,13 @@ struct snd_soc_codec_dai wm9713_dai[] = { .stream_name = "Voice Playback", .channels_min = 1, .channels_max = 1, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .capture = { .stream_name = "Voice Capture", .channels_min = 1, .channels_max = 2, - .rates = WM9713_RATES, + .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .ops = { .hw_params = wm9713_pcm_hw_params, -- 1.5.6

2 2

[alsa-devel] [PATCH 1/2] wss: rename cs4321_foo to wss_foo
by Krzysztof Helt 01 Jul '08

01 Jul '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Rename functions and structures from the former cs4321_lib to names more corresponding with the new name: wss_lib. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Another step toward unification of all WSS compatible chips handling. This patch is huge because it contains formatting fixes in all places the function names changed (and maybe few other as well). There are few issues left in the patch but I don't know how to fix them and do not lose common sense. diff -urp linux-alsa/include/sound/wss.h linux-mm/include/sound/wss.h --- linux-alsa/include/sound/wss.h 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/include/sound/wss.h 2008-06-25 23:45:58.063543700 +0200 @@ -1,5 +1,5 @@ -#ifndef __SOUND_CS4231_H -#define __SOUND_CS4231_H +#ifndef __SOUND_WSS_H +#define __SOUND_WSS_H /* * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> @@ -30,42 +30,42 @@ /* defines for codec.mode */ -#define CS4231_MODE_NONE 0x0000 -#define CS4231_MODE_PLAY 0x0001 -#define CS4231_MODE_RECORD 0x0002 -#define CS4231_MODE_TIMER 0x0004 -#define CS4231_MODE_OPEN (CS4231_MODE_PLAY|CS4231_MODE_RECORD|CS4231_MODE_TIMER) +#define WSS_MODE_NONE 0x0000 +#define WSS_MODE_PLAY 0x0001 +#define WSS_MODE_RECORD 0x0002 +#define WSS_MODE_TIMER 0x0004 +#define WSS_MODE_OPEN (WSS_MODE_PLAY|WSS_MODE_RECORD|WSS_MODE_TIMER) /* defines for codec.hardware */ -#define CS4231_HW_DETECT 0x0000 /* let CS4231 driver detect chip */ -#define CS4231_HW_DETECT3 0x0001 /* allow mode 3 */ -#define CS4231_HW_TYPE_MASK 0xff00 /* type mask */ -#define CS4231_HW_CS4231_MASK 0x0100 /* CS4231 serie */ -#define CS4231_HW_CS4231 0x0100 /* CS4231 chip */ -#define CS4231_HW_CS4231A 0x0101 /* CS4231A chip */ -#define CS4231_HW_AD1845 0x0102 /* AD1845 chip */ -#define CS4231_HW_CS4232_MASK 0x0200 /* CS4232 serie (has control ports) */ -#define CS4231_HW_CS4232 0x0200 /* CS4232 */ -#define CS4231_HW_CS4232A 0x0201 /* CS4232A */ -#define CS4231_HW_CS4236 0x0202 /* CS4236 */ -#define CS4231_HW_CS4236B_MASK 0x0400 /* CS4236B serie (has extended control regs) */ -#define CS4231_HW_CS4235 0x0400 /* CS4235 - Crystal Clear (tm) stereo enhancement */ -#define CS4231_HW_CS4236B 0x0401 /* CS4236B */ -#define CS4231_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ -#define CS4231_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ -#define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ +#define WSS_HW_DETECT 0x0000 /* let CS4231 driver detect chip */ +#define WSS_HW_DETECT3 0x0001 /* allow mode 3 */ +#define WSS_HW_TYPE_MASK 0xff00 /* type mask */ +#define WSS_HW_CS4231_MASK 0x0100 /* CS4231 serie */ +#define WSS_HW_CS4231 0x0100 /* CS4231 chip */ +#define WSS_HW_CS4231A 0x0101 /* CS4231A chip */ +#define WSS_HW_AD1845 0x0102 /* AD1845 chip */ +#define WSS_HW_CS4232_MASK 0x0200 /* CS4232 serie (has control ports) */ +#define WSS_HW_CS4232 0x0200 /* CS4232 */ +#define WSS_HW_CS4232A 0x0201 /* CS4232A */ +#define WSS_HW_CS4236 0x0202 /* CS4236 */ +#define WSS_HW_CS4236B_MASK 0x0400 /* CS4236B serie (has extended control regs) */ +#define WSS_HW_CS4235 0x0400 /* CS4235 - Crystal Clear (tm) stereo enhancement */ +#define WSS_HW_CS4236B 0x0401 /* CS4236B */ +#define WSS_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ +#define WSS_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ +#define WSS_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ /* compatible, but clones */ -#define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ -#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ -#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ +#define WSS_HW_INTERWAVE 0x1000 /* InterWave chip */ +#define WSS_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define WSS_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ /* defines for codec.hwshare */ -#define CS4231_HWSHARE_IRQ (1<<0) -#define CS4231_HWSHARE_DMA1 (1<<1) -#define CS4231_HWSHARE_DMA2 (1<<2) +#define WSS_HWSHARE_IRQ (1<<0) +#define WSS_HWSHARE_DMA1 (1<<1) +#define WSS_HWSHARE_DMA2 (1<<2) -struct snd_cs4231 { +struct snd_wss { unsigned long port; /* base i/o port */ struct resource *res_port; unsigned long cport; /* control base i/o port (CS4236) */ @@ -74,8 +74,8 @@ struct snd_cs4231 { int dma1; /* playback DMA */ int dma2; /* record DMA */ unsigned short version; /* version of CODEC chip */ - unsigned short mode; /* see to CS4231_MODE_XXXX */ - unsigned short hardware; /* see to CS4231_HW_XXXX */ + unsigned short mode; /* see to WSS_MODE_XXXX */ + unsigned short hardware; /* see to WSS_HW_XXXX */ unsigned short hwshare; /* shared resources */ unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ ebus_flag:1; /* SPARC: EBUS present */ @@ -100,43 +100,50 @@ struct snd_cs4231 { struct mutex open_mutex; int (*rate_constraint) (struct snd_pcm_runtime *runtime); - void (*set_playback_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char pdfr); - void (*set_capture_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char cdfr); - void (*trigger) (struct snd_cs4231 *chip, unsigned int what, int start); + void (*set_playback_format) (struct snd_wss *chip, + struct snd_pcm_hw_params *hw_params, + unsigned char pdfr); + void (*set_capture_format) (struct snd_wss *chip, + struct snd_pcm_hw_params *hw_params, + unsigned char cdfr); + void (*trigger) (struct snd_wss *chip, unsigned int what, int start); #ifdef CONFIG_PM - void (*suspend) (struct snd_cs4231 *chip); - void (*resume) (struct snd_cs4231 *chip); + void (*suspend) (struct snd_wss *chip); + void (*resume) (struct snd_wss *chip); #endif void *dma_private_data; - int (*claim_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); - int (*release_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); + int (*claim_dma) (struct snd_wss *chip, + void *dma_private_data, int dma); + int (*release_dma) (struct snd_wss *chip, + void *dma_private_data, int dma); }; /* exported functions */ -void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); -unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg); -void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); -unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg); -void snd_cs4231_mce_up(struct snd_cs4231 *chip); -void snd_cs4231_mce_down(struct snd_cs4231 *chip); +void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char val); +unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg); +void snd_cs4236_ext_out(struct snd_wss *chip, + unsigned char reg, unsigned char val); +unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg); +void snd_wss_mce_up(struct snd_wss *chip); +void snd_wss_mce_down(struct snd_wss *chip); -void snd_cs4231_overrange(struct snd_cs4231 *chip); +void snd_wss_overrange(struct snd_wss *chip); -irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); +irqreturn_t snd_wss_interrupt(int irq, void *dev_id); -const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); +const char *snd_wss_chip_id(struct snd_wss *chip); -int snd_cs4231_create(struct snd_card *card, +int snd_wss_create(struct snd_card *card, unsigned long port, unsigned long cport, int irq, int dma1, int dma2, unsigned short hardware, unsigned short hwshare, - struct snd_cs4231 ** rchip); -int snd_cs4231_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); -int snd_cs4231_timer(struct snd_cs4231 * chip, int device, struct snd_timer **rtimer); -int snd_cs4231_mixer(struct snd_cs4231 * chip); + struct snd_wss **rchip); +int snd_wss_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); +int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer); +int snd_wss_mixer(struct snd_wss *chip); int snd_cs4236_create(struct snd_card *card, unsigned long port, @@ -144,32 +151,38 @@ int snd_cs4236_create(struct snd_card *c int irq, int dma1, int dma2, unsigned short hardware, unsigned short hwshare, - struct snd_cs4231 ** rchip); -int snd_cs4236_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); -int snd_cs4236_mixer(struct snd_cs4231 * chip); + struct snd_wss **rchip); +int snd_cs4236_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm); +int snd_cs4236_mixer(struct snd_wss *chip); /* * mixer library */ -#define CS4231_SINGLE(xname, xindex, reg, shift, mask, invert) \ +#define WSS_SINGLE(xname, xindex, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_single, \ - .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \ + .info = snd_wss_info_single, \ + .get = snd_wss_get_single, .put = snd_wss_put_single, \ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } -int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_wss_info_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_wss_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_wss_put_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); -#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +#define WSS_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_double, \ - .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \ + .info = snd_wss_info_double, \ + .get = snd_wss_get_double, .put = snd_wss_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } -int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_wss_info_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_wss_get_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_wss_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); -#endif /* __SOUND_CS4231_H */ +#endif /* __SOUND_WSS_H */ diff -urp linux-alsa/sound/isa/ad1848/ad1848.c linux-mm/sound/isa/ad1848/ad1848.c --- linux-alsa/sound/isa/ad1848/ad1848.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/ad1848/ad1848.c 2008-06-25 18:41:45.531158672 +0200 @@ -70,15 +70,15 @@ static int __devinit snd_ad1848_match(st return 0; if (port[n] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + snd_printk(KERN_ERR "%s: please specify port\n", dev_name(dev)); return 0; } if (irq[n] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + snd_printk(KERN_ERR "%s: please specify irq\n", dev_name(dev)); return 0; } if (dma1[n] == SNDRV_AUTO_DMA) { - snd_printk(KERN_ERR "%s: please specify dma1\n", dev->bus_id); + snd_printk(KERN_ERR "%s: please specify dma1\n", dev_name(dev)); return 0; } return 1; diff -urp linux-alsa/sound/isa/azt2320.c linux-mm/sound/isa/azt2320.c --- linux-alsa/sound/isa/azt2320.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/azt2320.c 2008-06-25 18:41:45.599174329 +0200 @@ -76,7 +76,7 @@ struct snd_card_azt2320 { int dev_no; struct pnp_dev *dev; struct pnp_dev *devmpu; - struct snd_cs4231 *chip; + struct snd_wss *chip; }; static struct pnp_card_device_id snd_azt2320_pnpids[] = { @@ -181,7 +181,7 @@ static int __devinit snd_card_azt2320_pr int error; struct snd_card *card; struct snd_card_azt2320 *acard; - struct snd_cs4231 *chip; + struct snd_wss *chip; struct snd_opl3 *opl3; if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE, @@ -200,11 +200,12 @@ static int __devinit snd_card_azt2320_pr return error; } - if ((error = snd_cs4231_create(card, wss_port[dev], -1, - irq[dev], - dma1[dev], - dma2[dev], - CS4231_HW_DETECT, 0, &chip)) < 0) { + error = snd_wss_create(card, wss_port[dev], -1, + irq[dev], + dma1[dev], + dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (error < 0) { snd_card_free(card); return error; } @@ -214,15 +215,18 @@ static int __devinit snd_card_azt2320_pr sprintf(card->longname, "%s, WSS at 0x%lx, irq %i, dma %i&%i", card->shortname, chip->port, irq[dev], dma1[dev], dma2[dev]); - if ((error = snd_cs4231_pcm(chip, 0, NULL)) < 0) { + error = snd_wss_pcm(chip, 0, NULL); + if (error < 0) { snd_card_free(card); return error; } - if ((error = snd_cs4231_mixer(chip)) < 0) { + error = snd_wss_mixer(chip); + if (error < 0) { snd_card_free(card); return error; } - if ((error = snd_cs4231_timer(chip, 0, NULL)) < 0) { + error = snd_wss_timer(chip, 0, NULL); + if (error < 0) { snd_card_free(card); return error; } @@ -293,7 +297,7 @@ static int snd_azt2320_pnp_suspend(struc { struct snd_card *card = pnp_get_card_drvdata(pcard); struct snd_card_azt2320 *acard = card->private_data; - struct snd_cs4231 *chip = acard->chip; + struct snd_wss *chip = acard->chip; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->suspend(chip); @@ -304,7 +308,7 @@ static int snd_azt2320_pnp_resume(struct { struct snd_card *card = pnp_get_card_drvdata(pcard); struct snd_card_azt2320 *acard = card->private_data; - struct snd_cs4231 *chip = acard->chip; + struct snd_wss *chip = acard->chip; chip->resume(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D0); diff -urp linux-alsa/sound/isa/cs423x/cs4231.c linux-mm/sound/isa/cs423x/cs4231.c --- linux-alsa/sound/isa/cs423x/cs4231.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/cs423x/cs4231.c 2008-06-25 18:41:45.647162309 +0200 @@ -91,7 +91,7 @@ static int __devinit snd_cs4231_match(st static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) { struct snd_card *card; - struct snd_cs4231 *chip; + struct snd_wss *chip; struct snd_pcm *pcm; int error; @@ -99,14 +99,14 @@ static int __devinit snd_cs4231_probe(st if (!card) return -EINVAL; - error = snd_cs4231_create(card, port[n], -1, irq[n], dma1[n], dma2[n], - CS4231_HW_DETECT, 0, &chip); + error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], dma2[n], + WSS_HW_DETECT, 0, &chip); if (error < 0) goto out; card->private_data = chip; - error = snd_cs4231_pcm(chip, 0, &pcm); + error = snd_wss_pcm(chip, 0, &pcm); if (error < 0) goto out; @@ -118,11 +118,11 @@ static int __devinit snd_cs4231_probe(st if (dma2[n] >= 0) sprintf(card->longname + strlen(card->longname), "&%d", dma2[n]); - error = snd_cs4231_mixer(chip); + error = snd_wss_mixer(chip); if (error < 0) goto out; - error = snd_cs4231_timer(chip, 0, NULL); + error = snd_wss_timer(chip, 0, NULL); if (error < 0) goto out; @@ -160,7 +160,7 @@ static int __devexit snd_cs4231_remove(s static int snd_cs4231_suspend(struct device *dev, unsigned int n, pm_message_t state) { struct snd_card *card = dev_get_drvdata(dev); - struct snd_cs4231 *chip = card->private_data; + struct snd_wss *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->suspend(chip); @@ -170,7 +170,7 @@ static int snd_cs4231_suspend(struct dev static int snd_cs4231_resume(struct device *dev, unsigned int n) { struct snd_card *card = dev_get_drvdata(dev); - struct snd_cs4231 *chip = card->private_data; + struct snd_wss *chip = card->private_data; chip->resume(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D0); diff -urp linux-alsa/sound/isa/cs423x/cs4236.c linux-mm/sound/isa/cs423x/cs4236.c --- linux-alsa/sound/isa/cs423x/cs4236.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/cs423x/cs4236.c 2008-06-25 18:41:45.663172931 +0200 @@ -134,7 +134,7 @@ static int pnp_registered; #endif /* CONFIG_PNP */ struct snd_card_cs4236 { - struct snd_cs4231 *chip; + struct snd_wss *chip; struct resource *res_sb_port; #ifdef CONFIG_PNP struct pnp_dev *wss; @@ -395,7 +395,7 @@ static int __devinit snd_cs423x_probe(st { struct snd_card_cs4236 *acard; struct snd_pcm *pcm; - struct snd_cs4231 *chip; + struct snd_wss *chip; struct snd_opl3 *opl3; int err; @@ -407,41 +407,37 @@ static int __devinit snd_cs423x_probe(st } #ifdef CS4232 - if ((err = snd_cs4231_create(card, - port[dev], - cport[dev], - irq[dev], - dma1[dev], - dma2[dev], - CS4231_HW_DETECT, - 0, - &chip)) < 0) + err = snd_wss_create(card, + port[dev], cport[dev], irq[dev], + dma1[dev], dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (err < 0) return err; acard->chip = chip; - if ((err = snd_cs4231_pcm(chip, 0, &pcm)) < 0) + err = snd_wss_pcm(chip, 0, &pcm); + if (err < 0) return err; - if ((err = snd_cs4231_mixer(chip)) < 0) + err = snd_wss_mixer(chip); + if (err < 0) return err; #else /* CS4236 */ - if ((err = snd_cs4236_create(card, - port[dev], - cport[dev], - irq[dev], - dma1[dev], - dma2[dev], - CS4231_HW_DETECT, - 0, - &chip)) < 0) + err = snd_cs4236_create(card, + port[dev], cport[dev], + irq[dev], dma1[dev], dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (err < 0) return err; acard->chip = chip; - if ((err = snd_cs4236_pcm(chip, 0, &pcm)) < 0) + err = snd_cs4236_pcm(chip, 0, &pcm); + if (err < 0) return err; - if ((err = snd_cs4236_mixer(chip)) < 0) + err = snd_cs4236_mixer(chip); + if (err < 0) return err; #endif strcpy(card->driver, pcm->name); @@ -454,7 +450,8 @@ static int __devinit snd_cs423x_probe(st if (dma2[dev] >= 0) sprintf(card->longname + strlen(card->longname), "&%d", dma2[dev]); - if ((err = snd_cs4231_timer(chip, 0, NULL)) < 0) + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) return err; if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { diff -urp linux-alsa/sound/isa/cs423x/cs4236_lib.c linux-mm/sound/isa/cs423x/cs4236_lib.c --- linux-alsa/sound/isa/cs423x/cs4236_lib.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/cs423x/cs4236_lib.c 2008-06-25 23:32:58.556208225 +0200 @@ -121,13 +121,14 @@ static unsigned char snd_cs4236_ext_map[ * */ -static void snd_cs4236_ctrl_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val) +static void snd_cs4236_ctrl_out(struct snd_wss *chip, + unsigned char reg, unsigned char val) { outb(reg, chip->cport + 3); outb(chip->cimage[reg] = val, chip->cport + 4); } -static unsigned char snd_cs4236_ctrl_in(struct snd_cs4231 *chip, unsigned char reg) +static unsigned char snd_cs4236_ctrl_in(struct snd_wss *chip, unsigned char reg) { outb(reg, chip->cport + 3); return inb(chip->cport + 4); @@ -180,44 +181,52 @@ static unsigned char divisor_to_rate_reg } } -static void snd_cs4236_playback_format(struct snd_cs4231 *chip, struct snd_pcm_hw_params *params, unsigned char pdfr) +static void snd_cs4236_playback_format(struct snd_wss *chip, + struct snd_pcm_hw_params *params, + unsigned char pdfr) { unsigned long flags; unsigned char rate = divisor_to_rate_register(params->rate_den); spin_lock_irqsave(&chip->reg_lock, flags); /* set fast playback format change and clean playback FIFO */ - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x10); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, pdfr & 0xf0); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] & ~0x10); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] | 0x10); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, pdfr & 0xf0); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] & ~0x10); snd_cs4236_ext_out(chip, CS4236_DAC_RATE, rate); spin_unlock_irqrestore(&chip->reg_lock, flags); } -static void snd_cs4236_capture_format(struct snd_cs4231 *chip, struct snd_pcm_hw_params *params, unsigned char cdfr) +static void snd_cs4236_capture_format(struct snd_wss *chip, + struct snd_pcm_hw_params *params, + unsigned char cdfr) { unsigned long flags; unsigned char rate = divisor_to_rate_register(params->rate_den); spin_lock_irqsave(&chip->reg_lock, flags); /* set fast capture format change and clean capture FIFO */ - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x20); - snd_cs4231_out(chip, CS4231_REC_FORMAT, cdfr & 0xf0); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] & ~0x20); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] | 0x20); + snd_wss_out(chip, CS4231_REC_FORMAT, cdfr & 0xf0); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] & ~0x20); snd_cs4236_ext_out(chip, CS4236_ADC_RATE, rate); spin_unlock_irqrestore(&chip->reg_lock, flags); } #ifdef CONFIG_PM -static void snd_cs4236_suspend(struct snd_cs4231 *chip) +static void snd_cs4236_suspend(struct snd_wss *chip) { int reg; unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); for (reg = 0; reg < 32; reg++) - chip->image[reg] = snd_cs4231_in(chip, reg); + chip->image[reg] = snd_wss_in(chip, reg); for (reg = 0; reg < 18; reg++) chip->eimage[reg] = snd_cs4236_ext_in(chip, CS4236_I23VAL(reg)); for (reg = 2; reg < 9; reg++) @@ -225,12 +234,12 @@ static void snd_cs4236_suspend(struct sn spin_unlock_irqrestore(&chip->reg_lock, flags); } -static void snd_cs4236_resume(struct snd_cs4231 *chip) +static void snd_cs4236_resume(struct snd_wss *chip) { int reg; unsigned long flags; - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); for (reg = 0; reg < 32; reg++) { switch (reg) { @@ -240,7 +249,7 @@ static void snd_cs4236_resume(struct snd case 29: /* why? CS4235 - master right */ break; default: - snd_cs4231_out(chip, reg, chip->image[reg]); + snd_wss_out(chip, reg, chip->image[reg]); break; } } @@ -255,7 +264,7 @@ static void snd_cs4236_resume(struct snd } } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); } #endif /* CONFIG_PM */ @@ -266,24 +275,26 @@ int snd_cs4236_create(struct snd_card *c int irq, int dma1, int dma2, unsigned short hardware, unsigned short hwshare, - struct snd_cs4231 ** rchip) + struct snd_wss **rchip) { - struct snd_cs4231 *chip; + struct snd_wss *chip; unsigned char ver1, ver2; unsigned int reg; int err; *rchip = NULL; - if (hardware == CS4231_HW_DETECT) - hardware = CS4231_HW_DETECT3; + if (hardware == WSS_HW_DETECT) + hardware = WSS_HW_DETECT3; if (cport < 0x100) { snd_printk("please, specify control port for CS4236+ chips\n"); return -ENODEV; } - if ((err = snd_cs4231_create(card, port, cport, irq, dma1, dma2, hardware, hwshare, &chip)) < 0) + err = snd_wss_create(card, port, cport, + irq, dma1, dma2, hardware, hwshare, &chip); + if (err < 0) return err; - if (!(chip->hardware & CS4231_HW_CS4236B_MASK)) { + if (!(chip->hardware & WSS_HW_CS4236B_MASK)) { snd_printk("CS4236+: MODE3 and extended registers not available, hardware=0x%x\n",chip->hardware); snd_device_free(card, chip); return -ENODEV; @@ -330,20 +341,20 @@ int snd_cs4236_create(struct snd_card *c snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), snd_cs4236_ext_map[reg]); /* initialize compatible but more featured registers */ - snd_cs4231_out(chip, CS4231_LEFT_INPUT, 0x40); - snd_cs4231_out(chip, CS4231_RIGHT_INPUT, 0x40); - snd_cs4231_out(chip, CS4231_AUX1_LEFT_INPUT, 0xff); - snd_cs4231_out(chip, CS4231_AUX1_RIGHT_INPUT, 0xff); - snd_cs4231_out(chip, CS4231_AUX2_LEFT_INPUT, 0xdf); - snd_cs4231_out(chip, CS4231_AUX2_RIGHT_INPUT, 0xdf); - snd_cs4231_out(chip, CS4231_RIGHT_LINE_IN, 0xff); - snd_cs4231_out(chip, CS4231_LEFT_LINE_IN, 0xff); - snd_cs4231_out(chip, CS4231_RIGHT_LINE_IN, 0xff); + snd_wss_out(chip, CS4231_LEFT_INPUT, 0x40); + snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x40); + snd_wss_out(chip, CS4231_AUX1_LEFT_INPUT, 0xff); + snd_wss_out(chip, CS4231_AUX1_RIGHT_INPUT, 0xff); + snd_wss_out(chip, CS4231_AUX2_LEFT_INPUT, 0xdf); + snd_wss_out(chip, CS4231_AUX2_RIGHT_INPUT, 0xdf); + snd_wss_out(chip, CS4231_RIGHT_LINE_IN, 0xff); + snd_wss_out(chip, CS4231_LEFT_LINE_IN, 0xff); + snd_wss_out(chip, CS4231_RIGHT_LINE_IN, 0xff); switch (chip->hardware) { - case CS4231_HW_CS4235: - case CS4231_HW_CS4239: - snd_cs4231_out(chip, CS4235_LEFT_MASTER, 0xff); - snd_cs4231_out(chip, CS4235_RIGHT_MASTER, 0xff); + case WSS_HW_CS4235: + case WSS_HW_CS4239: + snd_wss_out(chip, CS4235_LEFT_MASTER, 0xff); + snd_wss_out(chip, CS4235_RIGHT_MASTER, 0xff); break; } @@ -351,12 +362,13 @@ int snd_cs4236_create(struct snd_card *c return 0; } -int snd_cs4236_pcm(struct snd_cs4231 *chip, int device, struct snd_pcm **rpcm) +int snd_cs4236_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; - if ((err = snd_cs4231_pcm(chip, device, &pcm)) < 0) + err = snd_wss_pcm(chip, device, &pcm); + if (err < 0) return err; pcm->info_flags &= ~SNDRV_PCM_INFO_JOINT_DUPLEX; if (rpcm) @@ -387,7 +399,7 @@ static int snd_cs4236_info_single(struct static int snd_cs4236_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -404,7 +416,7 @@ static int snd_cs4236_get_single(struct static int snd_cs4236_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -433,7 +445,7 @@ static int snd_cs4236_put_single(struct static int snd_cs4236_get_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -450,7 +462,7 @@ static int snd_cs4236_get_singlec(struct static int snd_cs4236_put_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -490,7 +502,7 @@ static int snd_cs4236_info_double(struct static int snd_cs4236_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -512,7 +524,7 @@ static int snd_cs4236_get_double(struct static int snd_cs4236_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -555,7 +567,7 @@ static int snd_cs4236_put_double(struct static int snd_cs4236_get_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -577,7 +589,7 @@ static int snd_cs4236_get_double1(struct static int snd_cs4236_put_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -600,7 +612,7 @@ static int snd_cs4236_put_double1(struct val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; val2 = (chip->eimage[CS4236_REG(right_reg)] & ~(mask << shift_right)) | val2; change = val1 != chip->image[left_reg] || val2 != chip->eimage[CS4236_REG(right_reg)]; - snd_cs4231_out(chip, left_reg, val1); + snd_wss_out(chip, left_reg, val1); snd_cs4236_ext_out(chip, right_reg, val2); spin_unlock_irqrestore(&chip->reg_lock, flags); return change; @@ -619,7 +631,7 @@ static inline int snd_cs4236_mixer_maste static int snd_cs4236_get_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); @@ -631,7 +643,7 @@ static int snd_cs4236_get_master_digital static int snd_cs4236_put_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int change; unsigned short val1, val2; @@ -678,7 +690,7 @@ static inline int snd_cs4235_mixer_outpu static int snd_cs4235_get_output_accu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); @@ -690,7 +702,7 @@ static int snd_cs4235_get_output_accu(st static int snd_cs4235_put_output_accu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int change; unsigned short val1, val2; @@ -701,108 +713,160 @@ static int snd_cs4235_put_output_accu(st val1 = (chip->image[CS4235_LEFT_MASTER] & ~(3 << 5)) | val1; val2 = (chip->image[CS4235_RIGHT_MASTER] & ~(3 << 5)) | val2; change = val1 != chip->image[CS4235_LEFT_MASTER] || val2 != chip->image[CS4235_RIGHT_MASTER]; - snd_cs4231_out(chip, CS4235_LEFT_MASTER, val1); - snd_cs4231_out(chip, CS4235_RIGHT_MASTER, val2); + snd_wss_out(chip, CS4235_LEFT_MASTER, val1); + snd_wss_out(chip, CS4235_RIGHT_MASTER, val2); spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } static struct snd_kcontrol_new snd_cs4236_controls[] = { -CS4236_DOUBLE("Master Digital Playback Switch", 0, CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), -CS4236_DOUBLE("Master Digital Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), +CS4236_DOUBLE("Master Digital Playback Switch", 0, + CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), +CS4236_DOUBLE("Master Digital Capture Switch", 0, + CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), CS4236_MASTER_DIGITAL("Master Digital Volume", 0), -CS4236_DOUBLE("Capture Boost Volume", 0, CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE("Capture Boost Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), -CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), - -CS4236_DOUBLE("DSP Playback Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), -CS4236_DOUBLE("DSP Playback Volume", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1), - -CS4236_DOUBLE("FM Playback Switch", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1), -CS4236_DOUBLE("FM Playback Volume", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1), - -CS4236_DOUBLE("Wavetable Playback Switch", 0, CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), -CS4236_DOUBLE("Wavetable Playback Volume", 0, CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1), - -CS4231_DOUBLE("Synth Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -CS4231_DOUBLE("Synth Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), -CS4231_DOUBLE("Synth Capture Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -CS4231_DOUBLE("Synth Capture Bypass", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 5, 5, 1, 1), - -CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), -CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), +WSS_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), + +CS4236_DOUBLE("DSP Playback Switch", 0, + CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), +CS4236_DOUBLE("DSP Playback Volume", 0, + CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1), + +CS4236_DOUBLE("FM Playback Switch", 0, + CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1), +CS4236_DOUBLE("FM Playback Volume", 0, + CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1), + +CS4236_DOUBLE("Wavetable Playback Switch", 0, + CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), +CS4236_DOUBLE("Wavetable Playback Volume", 0, + CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1), + +WSS_DOUBLE("Synth Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE("Synth Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE("Synth Capture Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), +WSS_DOUBLE("Synth Capture Bypass", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 5, 5, 1, 1), + +CS4236_DOUBLE("Mic Playback Switch", 0, + CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), +CS4236_DOUBLE("Mic Capture Switch", 0, + CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), -CS4236_DOUBLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), - -CS4231_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Line Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -CS4231_DOUBLE("Line Capture Bypass", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 5, 5, 1, 1), - -CS4231_DOUBLE("CD Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("CD Volume", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("CD Capture Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), - -CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), -CS4236_DOUBLE1("Mono Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -CS4231_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +CS4236_DOUBLE("Mic Playback Boost", 0, + CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), -CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -CS4231_DOUBLE("Analog Loopback Capture Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), - -CS4231_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE("Line Capture Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), +WSS_DOUBLE("Line Capture Bypass", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 5, 5, 1, 1), + +WSS_DOUBLE("CD Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("CD Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE("CD Capture Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), + +CS4236_DOUBLE1("Mono Output Playback Switch", 0, + CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), +CS4236_DOUBLE1("Mono Playback Switch", 0, + CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), +WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), + +WSS_DOUBLE("Capture Volume", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE("Analog Loopback Capture Switch", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), + +WSS_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, + CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) }; static struct snd_kcontrol_new snd_cs4235_controls[] = { -CS4231_DOUBLE("Master Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -CS4231_DOUBLE("Master Volume", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), +WSS_DOUBLE("Master Switch", 0, + CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), +WSS_DOUBLE("Master Volume", 0, + CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), CS4235_OUTPUT_ACCU("Playback Volume", 0), -CS4236_DOUBLE("Master Digital Playback Switch", 0, CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), -CS4236_DOUBLE("Master Digital Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), +CS4236_DOUBLE("Master Digital Playback Switch", 0, + CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), +CS4236_DOUBLE("Master Digital Capture Switch", 0, + CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), CS4236_MASTER_DIGITAL("Master Digital Volume", 0), -CS4231_DOUBLE("Master Digital Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -CS4231_DOUBLE("Master Digital Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -CS4231_DOUBLE("Master Digital Volume", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), - -CS4236_DOUBLE("Capture Volume", 0, CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), - -CS4231_DOUBLE("PCM Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE("Master Digital Playback Switch", 1, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE("Master Digital Capture Switch", 1, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), +WSS_DOUBLE("Master Digital Volume", 1, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), + +CS4236_DOUBLE("Capture Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), + +WSS_DOUBLE("PCM Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("PCM Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), CS4236_DOUBLE("DSP Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), CS4236_DOUBLE("FM Switch", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1), -CS4236_DOUBLE("Wavetable Switch", 0, CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), +CS4236_DOUBLE("Wavetable Switch", 0, + CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), -CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), -CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), +CS4236_DOUBLE("Mic Capture Switch", 0, + CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), +CS4236_DOUBLE("Mic Playback Switch", 0, + CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0), -CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -CS4231_DOUBLE("Aux Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), - -CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -CS4231_DOUBLE("Aux Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), - -CS4236_DOUBLE1("Master Mono Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), - -CS4236_DOUBLE1("Mono Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -CS4231_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_DOUBLE("Aux Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Capture Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), +WSS_DOUBLE("Aux Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), + +WSS_DOUBLE("Aux Playback Switch", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Capture Switch", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), +WSS_DOUBLE("Aux Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), + +CS4236_DOUBLE1("Master Mono Switch", 0, + CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), + +CS4236_DOUBLE1("Mono Switch", 0, + CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), +WSS_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -CS4231_DOUBLE("Analog Loopback Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), +WSS_DOUBLE("Analog Loopback Switch", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), }; #define CS4236_IEC958_ENABLE(xname, xindex) \ @@ -813,14 +877,14 @@ CS4231_DOUBLE("Analog Loopback Switch", static int snd_cs4236_get_iec958_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = chip->image[CS4231_ALT_FEATURE_1] & 0x02 ? 1 : 0; #if 0 printk("get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", - snd_cs4231_in(chip, CS4231_ALT_FEATURE_1), + snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), snd_cs4236_ctrl_in(chip, 5), @@ -833,7 +897,7 @@ static int snd_cs4236_get_iec958_switch( static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int change; unsigned short enable, val; @@ -841,23 +905,23 @@ static int snd_cs4236_put_iec958_switch( enable = ucontrol->value.integer.value[0] & 1; mutex_lock(&chip->mce_mutex); - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); val = (chip->image[CS4231_ALT_FEATURE_1] & ~0x0e) | (0<<2) | (enable << 1); change = val != chip->image[CS4231_ALT_FEATURE_1]; - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, val); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, val); val = snd_cs4236_ctrl_in(chip, 4) | 0xc0; snd_cs4236_ctrl_out(chip, 4, val); udelay(100); val &= ~0x40; snd_cs4236_ctrl_out(chip, 4, val); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); mutex_unlock(&chip->mce_mutex); #if 0 printk("set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", - snd_cs4231_in(chip, CS4231_ALT_FEATURE_1), + snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), snd_cs4236_ctrl_in(chip, 5), @@ -896,7 +960,7 @@ CS4236_SINGLEC("3D Control - Volume", 0, CS4236_SINGLEC("3D Control - IEC958", 0, 3, 5, 1, 0) }; -int snd_cs4236_mixer(struct snd_cs4231 *chip) +int snd_cs4236_mixer(struct snd_wss *chip) { struct snd_card *card; unsigned int idx, count; @@ -905,10 +969,10 @@ int snd_cs4236_mixer(struct snd_cs4231 * snd_assert(chip != NULL && chip->card != NULL, return -EINVAL); card = chip->card; - strcpy(card->mixername, snd_cs4231_chip_id(chip)); + strcpy(card->mixername, snd_wss_chip_id(chip)); - if (chip->hardware == CS4231_HW_CS4235 || - chip->hardware == CS4231_HW_CS4239) { + if (chip->hardware == WSS_HW_CS4235 || + chip->hardware == WSS_HW_CS4239) { for (idx = 0; idx < ARRAY_SIZE(snd_cs4235_controls); idx++) { if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4235_controls[idx], chip))) < 0) return err; @@ -920,16 +984,16 @@ int snd_cs4236_mixer(struct snd_cs4231 * } } switch (chip->hardware) { - case CS4231_HW_CS4235: - case CS4231_HW_CS4239: + case WSS_HW_CS4235: + case WSS_HW_CS4239: count = ARRAY_SIZE(snd_cs4236_3d_controls_cs4235); kcontrol = snd_cs4236_3d_controls_cs4235; break; - case CS4231_HW_CS4237B: + case WSS_HW_CS4237B: count = ARRAY_SIZE(snd_cs4236_3d_controls_cs4237); kcontrol = snd_cs4236_3d_controls_cs4237; break; - case CS4231_HW_CS4238B: + case WSS_HW_CS4238B: count = ARRAY_SIZE(snd_cs4236_3d_controls_cs4238); kcontrol = snd_cs4236_3d_controls_cs4238; break; @@ -941,8 +1005,8 @@ int snd_cs4236_mixer(struct snd_cs4231 * if ((err = snd_ctl_add(card, snd_ctl_new1(kcontrol, chip))) < 0) return err; } - if (chip->hardware == CS4231_HW_CS4237B || - chip->hardware == CS4231_HW_CS4238B) { + if (chip->hardware == WSS_HW_CS4237B || + chip->hardware == WSS_HW_CS4238B) { for (idx = 0; idx < ARRAY_SIZE(snd_cs4236_iec958_controls); idx++) { if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4236_iec958_controls[idx], chip))) < 0) return err; diff -urp linux-alsa/sound/isa/gus/gusmax.c linux-mm/sound/isa/gus/gusmax.c --- linux-alsa/sound/isa/gus/gusmax.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/gus/gusmax.c 2008-06-25 18:41:45.819179911 +0200 @@ -75,7 +75,7 @@ struct snd_gusmax { int irq; struct snd_card *card; struct snd_gus_card *gus; - struct snd_cs4231 *cs4231; + struct snd_wss *wss; unsigned short gus_status_reg; unsigned short pcm_status_reg; }; @@ -117,7 +117,7 @@ static irqreturn_t snd_gusmax_interrupt( } if (inb(maxcard->pcm_status_reg) & 0x01) { /* IRQ bit is set? */ handled = 1; - snd_cs4231_interrupt(irq, maxcard->cs4231); + snd_wss_interrupt(irq, maxcard->wss); loop++; } } while (loop && --max > 0); @@ -140,10 +140,7 @@ static void __devinit snd_gusmax_init(in outb(gus->max_cntrl_val, GUSP(gus, MAXCNTRLPORT)); } -#define CS4231_PRIVATE( left, right, shift, mute ) \ - ((left << 24)|(right << 16)|(shift<<8)|mute) - -static int __devinit snd_gusmax_mixer(struct snd_cs4231 *chip) +static int __devinit snd_gusmax_mixer(struct snd_wss *chip) { struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; @@ -214,7 +211,7 @@ static int __devinit snd_gusmax_probe(st int xirq, xdma1, xdma2, err; struct snd_card *card; struct snd_gus_card *gus = NULL; - struct snd_cs4231 *cs4231; + struct snd_wss *wss; struct snd_gusmax *maxcard; card = snd_card_new(index[dev], id[dev], THIS_MODULE, @@ -301,33 +298,39 @@ static int __devinit snd_gusmax_probe(st } maxcard->irq = xirq; - if ((err = snd_cs4231_create(card, - gus->gf1.port + 0x10c, -1, xirq, - xdma2 < 0 ? xdma1 : xdma2, xdma1, - CS4231_HW_DETECT, - CS4231_HWSHARE_IRQ | - CS4231_HWSHARE_DMA1 | - CS4231_HWSHARE_DMA2, - &cs4231)) < 0) + err = snd_wss_create(card, + gus->gf1.port + 0x10c, -1, xirq, + xdma2 < 0 ? xdma1 : xdma2, xdma1, + WSS_HW_DETECT, + WSS_HWSHARE_IRQ | + WSS_HWSHARE_DMA1 | + WSS_HWSHARE_DMA2, + &wss); + if (err < 0) goto _err; - if ((err = snd_cs4231_pcm(cs4231, 0, NULL)) < 0) + err = snd_wss_pcm(wss, 0, NULL); + if (err < 0) goto _err; - if ((err = snd_cs4231_mixer(cs4231)) < 0) + err = snd_wss_mixer(wss); + if (err < 0) goto _err; - if ((err = snd_cs4231_timer(cs4231, 2, NULL)) < 0) + err = snd_wss_timer(wss, 2, NULL); + if (err < 0) goto _err; if (pcm_channels[dev] > 0) { if ((err = snd_gf1_pcm_new(gus, 1, 1, NULL)) < 0) goto _err; } - if ((err = snd_gusmax_mixer(cs4231)) < 0) + err = snd_gusmax_mixer(wss); + if (err < 0) goto _err; - if ((err = snd_gf1_rawmidi_new(gus, 0, NULL)) < 0) + err = snd_gf1_rawmidi_new(gus, 0, NULL); + if (err < 0) goto _err; sprintf(card->longname + strlen(card->longname), " at 0x%lx, irq %i, dma %i", gus->gf1.port, xirq, xdma1); @@ -336,11 +339,12 @@ static int __devinit snd_gusmax_probe(st snd_card_set_dev(card, pdev); - if ((err = snd_card_register(card)) < 0) + err = snd_card_register(card); + if (err < 0) goto _err; maxcard->gus = gus; - maxcard->cs4231 = cs4231; + maxcard->wss = wss; dev_set_drvdata(pdev, card); return 0; diff -urp linux-alsa/sound/isa/gus/interwave.c linux-mm/sound/isa/gus/interwave.c --- linux-alsa/sound/isa/gus/interwave.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/gus/interwave.c 2008-06-25 23:23:41.623536555 +0200 @@ -118,7 +118,7 @@ struct snd_interwave { int irq; struct snd_card *card; struct snd_gus_card *gus; - struct snd_cs4231 *cs4231; + struct snd_wss *wss; #ifdef SNDRV_STB struct resource *i2c_res; #endif @@ -312,7 +312,7 @@ static irqreturn_t snd_interwave_interru } if (inb(iwcard->pcm_status_reg) & 0x01) { /* IRQ bit is set? */ handled = 1; - snd_cs4231_interrupt(irq, iwcard->cs4231); + snd_wss_interrupt(irq, iwcard->wss); loop++; } } while (loop && --max > 0); @@ -498,13 +498,17 @@ static void __devinit snd_interwave_init } static struct snd_kcontrol_new snd_interwave_controls[] = { -CS4231_DOUBLE("Master Playback Switch", 0, CS4231_LINE_LEFT_OUTPUT, CS4231_LINE_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Master Playback Volume", 0, CS4231_LINE_LEFT_OUTPUT, CS4231_LINE_RIGHT_OUTPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Mic Playback Switch", 0, CS4231_LEFT_MIC_INPUT, CS4231_RIGHT_MIC_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Mic Playback Volume", 0, CS4231_LEFT_MIC_INPUT, CS4231_RIGHT_MIC_INPUT, 0, 0, 31, 1) +WSS_DOUBLE("Master Playback Switch", 0, + CS4231_LINE_LEFT_OUTPUT, CS4231_LINE_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("Master Playback Volume", 0, + CS4231_LINE_LEFT_OUTPUT, CS4231_LINE_RIGHT_OUTPUT, 0, 0, 31, 1), +WSS_DOUBLE("Mic Playback Switch", 0, + CS4231_LEFT_MIC_INPUT, CS4231_RIGHT_MIC_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Mic Playback Volume", 0, + CS4231_LEFT_MIC_INPUT, CS4231_RIGHT_MIC_INPUT, 0, 0, 31, 1) }; -static int __devinit snd_interwave_mixer(struct snd_cs4231 *chip) +static int __devinit snd_interwave_mixer(struct snd_wss *chip) { struct snd_card *card = chip->card; struct snd_ctl_elem_id id1, id2; @@ -527,10 +531,10 @@ static int __devinit snd_interwave_mixer for (idx = 0; idx < ARRAY_SIZE(snd_interwave_controls); idx++) if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_interwave_controls[idx], chip))) < 0) return err; - snd_cs4231_out(chip, CS4231_LINE_LEFT_OUTPUT, 0x9f); - snd_cs4231_out(chip, CS4231_LINE_RIGHT_OUTPUT, 0x9f); - snd_cs4231_out(chip, CS4231_LEFT_MIC_INPUT, 0x9f); - snd_cs4231_out(chip, CS4231_RIGHT_MIC_INPUT, 0x9f); + snd_wss_out(chip, CS4231_LINE_LEFT_OUTPUT, 0x9f); + snd_wss_out(chip, CS4231_LINE_RIGHT_OUTPUT, 0x9f); + snd_wss_out(chip, CS4231_LEFT_MIC_INPUT, 0x9f); + snd_wss_out(chip, CS4231_RIGHT_MIC_INPUT, 0x9f); /* reassign AUXA to SYNTHESIZER */ strcpy(id1.name, "Aux Playback Switch"); strcpy(id2.name, "Synth Playback Switch"); @@ -642,7 +646,7 @@ static int __devinit snd_interwave_probe { int xirq, xdma1, xdma2; struct snd_interwave *iwcard = card->private_data; - struct snd_cs4231 *cs4231; + struct snd_wss *wss; struct snd_gus_card *gus; #ifdef SNDRV_STB struct snd_i2c_bus *i2c_bus; @@ -684,33 +688,39 @@ static int __devinit snd_interwave_probe } iwcard->irq = xirq; - if ((err = snd_cs4231_create(card, - gus->gf1.port + 0x10c, -1, xirq, - xdma2 < 0 ? xdma1 : xdma2, xdma1, - CS4231_HW_INTERWAVE, - CS4231_HWSHARE_IRQ | - CS4231_HWSHARE_DMA1 | - CS4231_HWSHARE_DMA2, - &cs4231)) < 0) + err = snd_wss_create(card, + gus->gf1.port + 0x10c, -1, xirq, + xdma2 < 0 ? xdma1 : xdma2, xdma1, + WSS_HW_INTERWAVE, + WSS_HWSHARE_IRQ | + WSS_HWSHARE_DMA1 | + WSS_HWSHARE_DMA2, + &wss); + if (err < 0) return err; - if ((err = snd_cs4231_pcm(cs4231, 0, &pcm)) < 0) + err = snd_wss_pcm(wss, 0, &pcm); + if (err < 0) return err; sprintf(pcm->name + strlen(pcm->name), " rev %c", gus->revision + 'A'); strcat(pcm->name, " (codec)"); - if ((err = snd_cs4231_timer(cs4231, 2, NULL)) < 0) + err = snd_wss_timer(wss, 2, NULL); + if (err < 0) return err; - if ((err = snd_cs4231_mixer(cs4231)) < 0) + err = snd_wss_mixer(wss); + if (err < 0) return err; if (pcm_channels[dev] > 0) { - if ((err = snd_gf1_pcm_new(gus, 1, 1, NULL)) < 0) + err = snd_gf1_pcm_new(gus, 1, 1, NULL); + if (err < 0) return err; } - if ((err = snd_interwave_mixer(cs4231)) < 0) + err = snd_interwave_mixer(wss); + if (err < 0) return err; #ifdef SNDRV_STB @@ -754,10 +764,11 @@ static int __devinit snd_interwave_probe if (xdma2 >= 0) sprintf(card->longname + strlen(card->longname), "&%d", xdma2); - if ((err = snd_card_register(card)) < 0) + err = snd_card_register(card); + if (err < 0) return err; - iwcard->cs4231 = cs4231; + iwcard->wss = wss; iwcard->gus = gus; return 0; } diff -urp linux-alsa/sound/isa/opl3sa2.c linux-mm/sound/isa/opl3sa2.c --- linux-alsa/sound/isa/opl3sa2.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/opl3sa2.c 2008-06-25 23:20:43.230193930 +0200 @@ -133,7 +133,7 @@ struct snd_opl3sa2 { spinlock_t reg_lock; struct snd_hwdep *synth; struct snd_rawmidi *rmidi; - struct snd_cs4231 *cs4231; + struct snd_wss *wss; unsigned char ctlregs[0x20]; int ymode; /* SL added */ struct snd_kcontrol *master_switch; @@ -318,7 +318,7 @@ static irqreturn_t snd_opl3sa2_interrupt if (status & 0x07) { /* TI,CI,PI */ handled = 1; - snd_cs4231_interrupt(irq, chip->cs4231); + snd_wss_interrupt(irq, chip->wss); } if (status & 0x40) { /* hardware volume change */ @@ -573,7 +573,7 @@ static int snd_opl3sa2_suspend(struct sn struct snd_opl3sa2 *chip = card->private_data; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - chip->cs4231->suspend(chip->cs4231); + chip->wss->suspend(chip->wss); /* power down */ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3); @@ -597,8 +597,8 @@ static int snd_opl3sa2_resume(struct snd for (i = 0x12; i <= 0x16; i++) snd_opl3sa2_write(chip, i, chip->ctlregs[i]); } - /* restore cs4231 */ - chip->cs4231->resume(chip->cs4231); + /* restore wss */ + chip->wss->resume(chip->wss); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; @@ -659,7 +659,7 @@ static int __devinit snd_opl3sa2_probe(s { int xirq, xdma1, xdma2; struct snd_opl3sa2 *chip; - struct snd_cs4231 *cs4231; + struct snd_wss *wss; struct snd_opl3 *opl3; int err; @@ -679,23 +679,25 @@ static int __devinit snd_opl3sa2_probe(s return -ENODEV; } chip->irq = xirq; - if ((err = snd_cs4231_create(card, - wss_port[dev] + 4, -1, - xirq, xdma1, xdma2, - CS4231_HW_OPL3SA2, - CS4231_HWSHARE_IRQ, - &cs4231)) < 0) { + err = snd_wss_create(card, + wss_port[dev] + 4, -1, + xirq, xdma1, xdma2, + WSS_HW_OPL3SA2, WSS_HWSHARE_IRQ, &wss); + if (err < 0) { snd_printd("Oops, WSS not detected at 0x%lx\n", wss_port[dev] + 4); return err; } - chip->cs4231 = cs4231; - if ((err = snd_cs4231_pcm(cs4231, 0, NULL)) < 0) + chip->wss = wss; + err = snd_wss_pcm(wss, 0, NULL); + if (err < 0) return err; - if ((err = snd_cs4231_mixer(cs4231)) < 0) + err = snd_wss_mixer(wss); + if (err < 0) return err; if ((err = snd_opl3sa2_mixer(chip)) < 0) return err; - if ((err = snd_cs4231_timer(cs4231, 0, NULL)) < 0) + err = snd_wss_timer(wss, 0, NULL); + if (err < 0) return err; if (fm_port[dev] >= 0x340 && fm_port[dev] < 0x400) { if ((err = snd_opl3_create(card, fm_port[dev], diff -urp linux-alsa/sound/isa/opti9xx/miro.c linux-mm/sound/isa/opti9xx/miro.c --- linux-alsa/sound/isa/opti9xx/miro.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/opti9xx/miro.c 2008-06-25 23:17:51.009039938 +0200 @@ -1221,7 +1221,7 @@ static int __devinit snd_miro_probe(stru int error; struct snd_miro *miro; - struct snd_cs4231 *codec; + struct snd_wss *codec; struct snd_timer *timer; struct snd_card *card; struct snd_pcm *pcm; @@ -1310,29 +1310,32 @@ static int __devinit snd_miro_probe(stru } } - if ((error = snd_miro_configure(miro))) { + error = snd_miro_configure(miro); + if (error) { snd_card_free(card); return error; } - if ((error = snd_cs4231_create(card, miro->wss_base + 4, -1, - miro->irq, miro->dma1, miro->dma2, - CS4231_HW_AD1845, - 0, - &codec)) < 0) { + error = snd_wss_create(card, miro->wss_base + 4, -1, + miro->irq, miro->dma1, miro->dma2, + WSS_HW_AD1845, 0, &codec); + if (error < 0) { snd_card_free(card); return error; } - if ((error = snd_cs4231_pcm(codec, 0, &pcm)) < 0) { + error = snd_wss_pcm(codec, 0, &pcm); + if (error < 0) { snd_card_free(card); return error; } - if ((error = snd_cs4231_mixer(codec)) < 0) { + error = snd_wss_mixer(codec); + if (error < 0) { snd_card_free(card); return error; } - if ((error = snd_cs4231_timer(codec, 0, &timer)) < 0) { + error = snd_wss_timer(codec, 0, &timer); + if (error < 0) { snd_card_free(card); return error; } diff -urp linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-mm/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-25 23:13:58.190075936 +0200 @@ -135,7 +135,7 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; - struct snd_cs4231 *codec; + struct snd_wss *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -558,7 +558,7 @@ __skip_mpu: static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_cs4231 *codec = dev_id; + struct snd_wss *codec = dev_id; struct snd_opti9xx *chip = codec->card->private_data; unsigned char status; @@ -566,7 +566,7 @@ static irqreturn_t snd_opti93x_interrupt if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); if ((status & OPTi93X_IRQ_CAPTURE) && codec->capture_substream) { - snd_cs4231_overrange(codec); + snd_wss_overrange(codec); snd_pcm_period_elapsed(codec->capture_substream); } outb(0x00, OPTi93X_PORT(codec, STATUS)); @@ -687,7 +687,7 @@ static void snd_card_opti9xx_free(struct if (chip) { #ifdef OPTi93X - struct snd_cs4231 *codec = chip->codec; + struct snd_wss *codec = chip->codec; if (codec->irq > 0) { disable_irq(codec->irq); free_irq(codec->irq, codec); @@ -703,7 +703,7 @@ static int __devinit snd_opti9xx_probe(s int error; struct snd_opti9xx *chip = card->private_data; #if defined(CS4231) || defined(OPTi93X) - struct snd_cs4231 *codec; + struct snd_wss *codec; #ifdef CS4231 struct snd_timer *timer; #endif @@ -730,33 +730,39 @@ static int __devinit snd_opti9xx_probe(s #endif if (chip->wss_base == SNDRV_AUTO_PORT) { - if ((chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4)) < 0) { + chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); + if (chip->wss_base < 0) { snd_printk("unable to find a free WSS port\n"); return -EBUSY; } } - if ((error = snd_opti9xx_configure(chip))) + error = snd_opti9xx_configure(chip); + if (error) return error; #if defined(CS4231) || defined(OPTi93X) - if ((error = snd_cs4231_create(card, chip->wss_base + 4, -1, - chip->irq, chip->dma1, chip->dma2, + error = snd_wss_create(card, chip->wss_base + 4, -1, + chip->irq, chip->dma1, chip->dma2, #ifdef CS4231 - CS4231_HW_DETECT, 0, + WSS_HW_DETECT, 0, #else /* OPTi93x */ - CS4231_HW_OPTI93X, CS4231_HWSHARE_IRQ, + WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #endif - &codec)) < 0) + &codec); + if (error < 0) return error; #ifdef OPTi93X chip->codec = codec; #endif - if ((error = snd_cs4231_pcm(codec, 0, &pcm)) < 0) + error = snd_wss_pcm(codec, 0, &pcm); + if (error < 0) return error; - if ((error = snd_cs4231_mixer(codec)) < 0) + error = snd_wss_mixer(codec); + if (error < 0) return error; #ifdef CS4231 - if ((error = snd_cs4231_timer(codec, 0, &timer)) < 0) + error = snd_wss_timer(codec, 0, &timer); + if (error < 0) return error; #else /* OPTI93X */ error = request_irq(chip->irq, snd_opti93x_interrupt, diff -urp linux-alsa/sound/isa/sscape.c linux-mm/sound/isa/sscape.c --- linux-alsa/sound/isa/sscape.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/sscape.c 2008-06-25 19:28:01.434867234 +0200 @@ -147,7 +147,7 @@ struct soundscape { enum card_type type; struct resource *io_res; struct resource *wss_res; - struct snd_cs4231 *chip; + struct snd_wss *chip; struct snd_mpu401 *mpu; struct snd_hwdep *hw; @@ -726,7 +726,7 @@ static int sscape_midi_info(struct snd_k static int sscape_midi_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *uctl) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kctl); + struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; register struct soundscape *s = get_card_soundscape(card); unsigned long flags; @@ -746,7 +746,7 @@ static int sscape_midi_get(struct snd_kc static int sscape_midi_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *uctl) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kctl); + struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; register struct soundscape *s = get_card_soundscape(card); unsigned long flags; @@ -958,7 +958,9 @@ static int __devinit create_mpu401(struc * Override for the CS4231 playback format function. * The AD1845 has much simpler format and rate selection. */ -static void ad1845_playback_format(struct snd_cs4231 * chip, struct snd_pcm_hw_params *params, unsigned char format) +static void ad1845_playback_format(struct snd_wss *chip, + struct snd_pcm_hw_params *params, + unsigned char format) { unsigned long flags; unsigned rate = params_rate(params); @@ -983,9 +985,9 @@ static void ad1845_playback_format(struc * NOTE: We seem to need to write to the MSB before the LSB * to get the correct sample frequency. */ - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0)); - snd_cs4231_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_cs4231_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0)); + snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); + snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); spin_unlock_irqrestore(&chip->reg_lock, flags); } @@ -994,7 +996,9 @@ static void ad1845_playback_format(struc * Override for the CS4231 capture format function. * The AD1845 has much simpler format and rate selection. */ -static void ad1845_capture_format(struct snd_cs4231 * chip, struct snd_pcm_hw_params *params, unsigned char format) +static void ad1845_capture_format(struct snd_wss *chip, + struct snd_pcm_hw_params *params, + unsigned char format) { unsigned long flags; unsigned rate = params_rate(params); @@ -1019,9 +1023,9 @@ static void ad1845_capture_format(struct * NOTE: We seem to need to write to the MSB before the LSB * to get the correct sample frequency. */ - snd_cs4231_out(chip, CS4231_REC_FORMAT, (format & 0xf0)); - snd_cs4231_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_cs4231_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); + snd_wss_out(chip, CS4231_REC_FORMAT, (format & 0xf0)); + snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); + snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); spin_unlock_irqrestore(&chip->reg_lock, flags); } @@ -1036,7 +1040,7 @@ static int __devinit create_ad1845(struc int irq, int dma1, int dma2) { register struct soundscape *sscape = get_card_soundscape(card); - struct snd_cs4231 *chip; + struct snd_wss *chip; int err; if (sscape->type == SSCAPE_VIVO) @@ -1045,9 +1049,9 @@ static int __devinit create_ad1845(struc if (dma1 == dma2) dma2 = -1; - err = snd_cs4231_create(card, + err = snd_wss_create(card, port, -1, irq, dma1, dma2, - CS4231_HW_DETECT, CS4231_HWSHARE_DMA1, &chip); + WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; @@ -1063,11 +1067,11 @@ static int __devinit create_ad1845(struc * #define AD1845_IFACE_CONFIG \ (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); + snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); */ if (sscape->type != SSCAPE_VIVO) { @@ -1077,11 +1081,11 @@ static int __devinit create_ad1845(struc * be 14.31818 MHz, because we must set this register * to get the playback to sound correct ... */ - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); + snd_wss_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); /* * More custom configuration: @@ -1089,28 +1093,28 @@ static int __devinit create_ad1845(struc * b) enable frequency selection (for capture/playback) */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_MISC_INFO, + snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2 | 0x10); - val = snd_cs4231_in(chip, AD1845_PWR_DOWN_CTRL); - snd_cs4231_out(chip, AD1845_PWR_DOWN_CTRL, + val = snd_wss_in(chip, AD1845_PWR_DOWN_CTRL); + snd_wss_out(chip, AD1845_PWR_DOWN_CTRL, val | AD1845_FREQ_SEL_ENABLE); spin_unlock_irqrestore(&chip->reg_lock, flags); } - err = snd_cs4231_pcm(chip, 0, &pcm); + err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) { snd_printk(KERN_ERR "sscape: No PCM device " "for AD1845 chip\n"); goto _error; } - err = snd_cs4231_mixer(chip); + err = snd_wss_mixer(chip); if (err < 0) { snd_printk(KERN_ERR "sscape: No mixer device " "for AD1845 chip\n"); goto _error; } - err = snd_cs4231_timer(chip, 0, NULL); + err = snd_wss_timer(chip, 0, NULL); if (err < 0) { snd_printk(KERN_ERR "sscape: No timer device " "for AD1845 chip\n"); diff -urp linux-alsa/sound/isa/wavefront/wavefront.c linux-mm/sound/isa/wavefront/wavefront.c --- linux-alsa/sound/isa/wavefront/wavefront.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/wavefront/wavefront.c 2008-06-25 23:46:59.269375541 +0200 @@ -1,6 +1,6 @@ /* * ALSA card-level driver for Turtle Beach Wavefront cards - * (Maui,Tropez,Tropez+) + * (Maui,Tropez,Tropez+) * * Copyright (c) 1997-1999 by Paul Barton-Davis <pbd(a)op.net> * @@ -320,8 +320,8 @@ snd_wavefront_new_midi (struct snd_card snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_wavefront_midi_input); rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | - SNDRV_RAWMIDI_INFO_INPUT | - SNDRV_RAWMIDI_INFO_DUPLEX; + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; return rmidi; } @@ -364,7 +364,7 @@ static int __devinit snd_wavefront_probe (struct snd_card *card, int dev) { snd_wavefront_card_t *acard = card->private_data; - struct snd_cs4231 *chip; + struct snd_wss *chip; struct snd_hwdep *wavefront_synth; struct snd_rawmidi *ics2115_internal_rmidi = NULL; struct snd_rawmidi *ics2115_external_rmidi = NULL; @@ -373,21 +373,20 @@ snd_wavefront_probe (struct snd_card *ca /* --------- PCM --------------- */ - if ((err = snd_cs4231_create (card, - cs4232_pcm_port[dev], - -1, - cs4232_pcm_irq[dev], - dma1[dev], - dma2[dev], - CS4231_HW_DETECT, 0, &chip)) < 0) { - snd_printk (KERN_ERR "can't allocate CS4231 device\n"); + err = snd_wss_create(card, cs4232_pcm_port[dev], -1, + cs4232_pcm_irq[dev], dma1[dev], dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (err < 0) { + snd_printk(KERN_ERR "can't allocate WSS device\n"); return err; } - if ((err = snd_cs4231_pcm (chip, 0, NULL)) < 0) + err = snd_wss_pcm(chip, 0, NULL); + if (err < 0) return err; - if ((err = snd_cs4231_timer (chip, 0, NULL)) < 0) + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) return err; /* ---------- OPL3 synth --------- */ @@ -395,24 +394,24 @@ snd_wavefront_probe (struct snd_card *ca if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3; - if ((err = snd_opl3_create(card, - fm_port[dev], - fm_port[dev] + 2, - OPL3_HW_OPL3_CS, - 0, &opl3)) < 0) { + err = snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3_CS, 0, &opl3); + if (err < 0) { snd_printk (KERN_ERR "can't allocate or detect OPL3 synth\n"); return err; } - if ((err = snd_opl3_hwdep_new(opl3, hw_dev, 1, NULL)) < 0) + err = snd_opl3_hwdep_new(opl3, hw_dev, 1, NULL); + if (err < 0) return err; hw_dev++; } /* ------- ICS2115 Wavetable synth ------- */ - if ((acard->wavefront.res_base = request_region(ics2115_port[dev], 16, - "ICS2115")) == NULL) { + acard->wavefront.res_base = request_region(ics2115_port[dev], 16, + "ICS2115"); + if (acard->wavefront.res_base == NULL) { snd_printk(KERN_ERR "unable to grab ICS2115 i/o region 0x%lx-0x%lx\n", ics2115_port[dev], ics2115_port[dev] + 16 - 1); return -EBUSY; @@ -426,7 +425,8 @@ snd_wavefront_probe (struct snd_card *ca acard->wavefront.irq = ics2115_irq[dev]; acard->wavefront.base = ics2115_port[dev]; - if ((wavefront_synth = snd_wavefront_new_synth (card, hw_dev, acard)) == NULL) { + wavefront_synth = snd_wavefront_new_synth(card, hw_dev, acard); + if (wavefront_synth == NULL) { snd_printk (KERN_ERR "can't create WaveFront synth device\n"); return -ENOMEM; } @@ -437,7 +437,8 @@ snd_wavefront_probe (struct snd_card *ca /* --------- Mixer ------------ */ - if ((err = snd_cs4231_mixer(chip)) < 0) { + err = snd_wss_mixer(chip); + if (err < 0) { snd_printk (KERN_ERR "can't allocate mixer device\n"); return err; } @@ -445,11 +446,11 @@ snd_wavefront_probe (struct snd_card *ca /* -------- CS4232 MPU-401 interface -------- */ if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) { - if ((err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232, - cs4232_mpu_port[dev], 0, - cs4232_mpu_irq[dev], - IRQF_DISABLED, - NULL)) < 0) { + err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232, + cs4232_mpu_port[dev], 0, + cs4232_mpu_irq[dev], IRQF_DISABLED, + NULL); + if (err < 0) { snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n"); return err; } @@ -602,7 +603,7 @@ static struct isa_driver snd_wavefront_d #ifdef CONFIG_PNP static int __devinit snd_wavefront_pnp_detect(struct pnp_card_link *pcard, - const struct pnp_card_device_id *pid) + const struct pnp_card_device_id *pid) { static int dev; struct snd_card *card; diff -urp linux-alsa/sound/isa/wss/wss_lib.c linux-mm/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-06-24 19:45:16.000000000 +0200 +++ linux-mm/sound/isa/wss/wss_lib.c 2008-06-25 19:11:56.104196128 +0200 @@ -6,7 +6,7 @@ * - sometimes record brokes playback with WSS portion of * Yamaha OPL3-SA3 chip * - CS4231 (GUS MAX) - still trouble with occasional noises - * - broken initialization? + * - broken initialization? * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -78,12 +78,13 @@ static struct snd_pcm_hw_constraint_list .mask = 0, }; -static int snd_cs4231_xrate(struct snd_pcm_runtime *runtime) +static int snd_wss_xrate(struct snd_pcm_runtime *runtime) { - return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); } -static unsigned char snd_cs4231_original_image[32] = +static unsigned char snd_wss_original_image[32] = { 0x00, /* 00/00 - lic */ 0x00, /* 01/01 - ric */ @@ -159,150 +160,192 @@ static unsigned char snd_opti93x_origina * Basic I/O functions */ -static inline void cs4231_outb(struct snd_cs4231 *chip, u8 offset, u8 val) +static inline void wss_outb(struct snd_wss *chip, u8 offset, u8 val) { outb(val, chip->port + offset); } +EXPORT_SYMBOL(snd_wss_out); -static inline u8 cs4231_inb(struct snd_cs4231 *chip, u8 offset) +static inline u8 wss_inb(struct snd_wss *chip, u8 offset) { return inb(chip->port + offset); } -static void snd_cs4231_wait(struct snd_cs4231 *chip) +static void snd_wss_wait(struct snd_wss *chip) { int timeout; for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(100); } -static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, +static void snd_wss_outm(struct snd_wss *chip, unsigned char reg, unsigned char mask, unsigned char value) { unsigned char tmp = (chip->image[reg] & mask) | value; - snd_cs4231_wait(chip); + snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif chip->image[reg] = tmp; if (!chip->calibrate_mute) { - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); wmb(); - cs4231_outb(chip, CS4231P(REG), tmp); + wss_outb(chip, CS4231P(REG), tmp); mb(); } } -static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +static void snd_wss_dout(struct snd_wss *chip, unsigned char reg, + unsigned char value) { int timeout; for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(10); - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - cs4231_outb(chip, CS4231P(REG), value); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + wss_outb(chip, CS4231P(REG), value); mb(); } -void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char value) { - snd_cs4231_wait(chip); + snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - cs4231_outb(chip, CS4231P(REG), value); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + wss_outb(chip, CS4231P(REG), value); chip->image[reg] = value; mb(); snd_printdd("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); } -unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) +unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg) { - snd_cs4231_wait(chip); + snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); #endif - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); mb(); - return cs4231_inb(chip, CS4231P(REG)); + return wss_inb(chip, CS4231P(REG)); } +EXPORT_SYMBOL(snd_wss_in); -void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val) +void snd_cs4236_ext_out(struct snd_wss *chip, unsigned char reg, + unsigned char val) { - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); - cs4231_outb(chip, CS4231P(REG), reg | (chip->image[CS4236_EXT_REG] & 0x01)); - cs4231_outb(chip, CS4231P(REG), val); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); + wss_outb(chip, CS4231P(REG), + reg | (chip->image[CS4236_EXT_REG] & 0x01)); + wss_outb(chip, CS4231P(REG), val); chip->eimage[CS4236_REG(reg)] = val; #if 0 printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val); #endif } +EXPORT_SYMBOL(snd_cs4236_ext_out); -unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg) +unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg) { - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); - cs4231_outb(chip, CS4231P(REG), reg | (chip->image[CS4236_EXT_REG] & 0x01)); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); + wss_outb(chip, CS4231P(REG), + reg | (chip->image[CS4236_EXT_REG] & 0x01)); #if 1 - return cs4231_inb(chip, CS4231P(REG)); + return wss_inb(chip, CS4231P(REG)); #else { unsigned char res; - res = cs4231_inb(chip, CS4231P(REG)); + res = wss_inb(chip, CS4231P(REG)); printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res); return res; } #endif } +EXPORT_SYMBOL(snd_cs4236_ext_in); #if 0 -static void snd_cs4231_debug(struct snd_cs4231 *chip) +static void snd_wss_debug(struct snd_wss *chip) { - printk("CS4231 REGS: INDEX = 0x%02x ", cs4231_inb(chip, CS4231P(REGSEL))); - printk(" STATUS = 0x%02x\n", cs4231_inb(chip, CS4231P(STATUS))); - printk(" 0x00: left input = 0x%02x ", snd_cs4231_in(chip, 0x00)); - printk(" 0x10: alt 1 (CFIG 2) = 0x%02x\n", snd_cs4231_in(chip, 0x10)); - printk(" 0x01: right input = 0x%02x ", snd_cs4231_in(chip, 0x01)); - printk(" 0x11: alt 2 (CFIG 3) = 0x%02x\n", snd_cs4231_in(chip, 0x11)); - printk(" 0x02: GF1 left input = 0x%02x ", snd_cs4231_in(chip, 0x02)); - printk(" 0x12: left line in = 0x%02x\n", snd_cs4231_in(chip, 0x12)); - printk(" 0x03: GF1 right input = 0x%02x ", snd_cs4231_in(chip, 0x03)); - printk(" 0x13: right line in = 0x%02x\n", snd_cs4231_in(chip, 0x13)); - printk(" 0x04: CD left input = 0x%02x ", snd_cs4231_in(chip, 0x04)); - printk(" 0x14: timer low = 0x%02x\n", snd_cs4231_in(chip, 0x14)); - printk(" 0x05: CD right input = 0x%02x ", snd_cs4231_in(chip, 0x05)); - printk(" 0x15: timer high = 0x%02x\n", snd_cs4231_in(chip, 0x15)); - printk(" 0x06: left output = 0x%02x ", snd_cs4231_in(chip, 0x06)); - printk(" 0x16: left MIC (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x16)); - printk(" 0x07: right output = 0x%02x ", snd_cs4231_in(chip, 0x07)); - printk(" 0x17: right MIC (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x17)); - printk(" 0x08: playback format = 0x%02x ", snd_cs4231_in(chip, 0x08)); - printk(" 0x18: IRQ status = 0x%02x\n", snd_cs4231_in(chip, 0x18)); - printk(" 0x09: iface (CFIG 1) = 0x%02x ", snd_cs4231_in(chip, 0x09)); - printk(" 0x19: left line out = 0x%02x\n", snd_cs4231_in(chip, 0x19)); - printk(" 0x0a: pin control = 0x%02x ", snd_cs4231_in(chip, 0x0a)); - printk(" 0x1a: mono control = 0x%02x\n", snd_cs4231_in(chip, 0x1a)); - printk(" 0x0b: init & status = 0x%02x ", snd_cs4231_in(chip, 0x0b)); - printk(" 0x1b: right line out = 0x%02x\n", snd_cs4231_in(chip, 0x1b)); - printk(" 0x0c: revision & mode = 0x%02x ", snd_cs4231_in(chip, 0x0c)); - printk(" 0x1c: record format = 0x%02x\n", snd_cs4231_in(chip, 0x1c)); - printk(" 0x0d: loopback = 0x%02x ", snd_cs4231_in(chip, 0x0d)); - printk(" 0x1d: var freq (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x1d)); - printk(" 0x0e: ply upr count = 0x%02x ", snd_cs4231_in(chip, 0x0e)); - printk(" 0x1e: ply lwr count = 0x%02x\n", snd_cs4231_in(chip, 0x1e)); - printk(" 0x0f: rec upr count = 0x%02x ", snd_cs4231_in(chip, 0x0f)); - printk(" 0x1f: rec lwr count = 0x%02x\n", snd_cs4231_in(chip, 0x1f)); + printk(KERN_DEBUG "CS4231 REGS: INDEX = 0x%02x ", + wss_inb(chip, CS4231P(REGSEL))); + printk(KERN_DEBUG " STATUS = 0x%02x\n", + wss_inb(chip, CS4231P(STATUS))); + printk(KERN_DEBUG " 0x00: left input = 0x%02x ", + snd_wss_in(chip, 0x00)); + printk(KERN_DEBUG " 0x10: alt 1 (CFIG 2) = 0x%02x\n", + snd_wss_in(chip, 0x10)); + printk(KERN_DEBUG " 0x01: right input = 0x%02x ", + snd_wss_in(chip, 0x01)); + printk(KERN_DEBUG " 0x11: alt 2 (CFIG 3) = 0x%02x\n", + snd_wss_in(chip, 0x11)); + printk(KERN_DEBUG " 0x02: GF1 left input = 0x%02x ", + snd_wss_in(chip, 0x02)); + printk(KERN_DEBUG " 0x12: left line in = 0x%02x\n", + snd_wss_in(chip, 0x12)); + printk(KERN_DEBUG " 0x03: GF1 right input = 0x%02x ", + snd_wss_in(chip, 0x03)); + printk(KERN_DEBUG " 0x13: right line in = 0x%02x\n", + snd_wss_in(chip, 0x13)); + printk(KERN_DEBUG " 0x04: CD left input = 0x%02x ", + snd_wss_in(chip, 0x04)); + printk(KERN_DEBUG " 0x14: timer low = 0x%02x\n", + snd_wss_in(chip, 0x14)); + printk(KERN_DEBUG " 0x05: CD right input = 0x%02x ", + snd_wss_in(chip, 0x05)); + printk(KERN_DEBUG " 0x15: timer high = 0x%02x\n", + snd_wss_in(chip, 0x15)); + printk(KERN_DEBUG " 0x06: left output = 0x%02x ", + snd_wss_in(chip, 0x06)); + printk(KERN_DEBUG " 0x16: left MIC (PnP) = 0x%02x\n", + snd_wss_in(chip, 0x16)); + printk(KERN_DEBUG " 0x07: right output = 0x%02x ", + snd_wss_in(chip, 0x07)); + printk(KERN_DEBUG " 0x17: right MIC (PnP) = 0x%02x\n", + snd_wss_in(chip, 0x17)); + printk(KERN_DEBUG " 0x08: playback format = 0x%02x ", + snd_wss_in(chip, 0x08)); + printk(KERN_DEBUG " 0x18: IRQ status = 0x%02x\n", + snd_wss_in(chip, 0x18)); + printk(KERN_DEBUG " 0x09: iface (CFIG 1) = 0x%02x ", + snd_wss_in(chip, 0x09)); + printk(KERN_DEBUG " 0x19: left line out = 0x%02x\n", + snd_wss_in(chip, 0x19)); + printk(KERN_DEBUG " 0x0a: pin control = 0x%02x ", + snd_wss_in(chip, 0x0a)); + printk(KERN_DEBUG " 0x1a: mono control = 0x%02x\n", + snd_wss_in(chip, 0x1a)); + printk(KERN_DEBUG " 0x0b: init & status = 0x%02x ", + snd_wss_in(chip, 0x0b)); + printk(KERN_DEBUG " 0x1b: right line out = 0x%02x\n", + snd_wss_in(chip, 0x1b)); + printk(KERN_DEBUG " 0x0c: revision & mode = 0x%02x ", + snd_wss_in(chip, 0x0c)); + printk(KERN_DEBUG " 0x1c: record format = 0x%02x\n", + snd_wss_in(chip, 0x1c)); + printk(KERN_DEBUG " 0x0d: loopback = 0x%02x ", + snd_wss_in(chip, 0x0d)); + printk(KERN_DEBUG " 0x1d: var freq (PnP) = 0x%02x\n", + snd_wss_in(chip, 0x1d)); + printk(KERN_DEBUG " 0x0e: ply upr count = 0x%02x ", + snd_wss_in(chip, 0x0e)); + printk(KERN_DEBUG " 0x1e: ply lwr count = 0x%02x\n", + snd_wss_in(chip, 0x1e)); + printk(KERN_DEBUG " 0x0f: rec upr count = 0x%02x ", + snd_wss_in(chip, 0x0f)); + printk(KERN_DEBUG " 0x1f: rec lwr count = 0x%02x\n", + snd_wss_in(chip, 0x1f)); } #endif @@ -311,61 +354,63 @@ static void snd_cs4231_debug(struct snd_ * CS4231 detection / MCE routines */ -static void snd_cs4231_busy_wait(struct snd_cs4231 *chip) +static void snd_wss_busy_wait(struct snd_wss *chip) { int timeout; /* huh.. looks like this sequence is proper for CS4231A chip (GUS MAX) */ for (timeout = 5; timeout > 0; timeout--) - cs4231_inb(chip, CS4231P(REGSEL)); + wss_inb(chip, CS4231P(REGSEL)); /* end of cleanup sequence */ for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout > 0 && (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(10); } -void snd_cs4231_mce_up(struct snd_cs4231 *chip) +void snd_wss_mce_up(struct snd_wss *chip) { unsigned long flags; int timeout; - snd_cs4231_wait(chip); + snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit |= CS4231_MCE; - timeout = cs4231_inb(chip, CS4231P(REGSEL)); + timeout = wss_inb(chip, CS4231P(REGSEL)); if (timeout == 0x80) snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); if (!(timeout & CS4231_MCE)) - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); + wss_outb(chip, CS4231P(REGSEL), + chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); } +EXPORT_SYMBOL(snd_wss_mce_up); -void snd_cs4231_mce_down(struct snd_cs4231 *chip) +void snd_wss_mce_down(struct snd_wss *chip) { unsigned long flags; unsigned long end_time; int timeout; - snd_cs4231_busy_wait(chip); + snd_wss_busy_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit &= ~CS4231_MCE; - timeout = cs4231_inb(chip, CS4231P(REGSEL)); - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); + timeout = wss_inb(chip, CS4231P(REGSEL)); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); if ((timeout & CS4231_MCE) == 0 || - !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { + !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { return; } @@ -380,7 +425,7 @@ void snd_cs4231_mce_down(struct snd_cs42 /* check condition up to 250 ms */ end_time = jiffies + msecs_to_jiffies(250); - while (snd_cs4231_in(chip, CS4231_TEST_INIT) & + while (snd_wss_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { if (time_after(jiffies, end_time)) { @@ -395,7 +440,7 @@ void snd_cs4231_mce_down(struct snd_cs42 /* check condition up to 100 ms */ end_time = jiffies + msecs_to_jiffies(100); - while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { + while (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { if (time_after(jiffies, end_time)) { snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); return; @@ -404,10 +449,11 @@ void snd_cs4231_mce_down(struct snd_cs42 } snd_printdd("(3) jiffies = %lu\n", jiffies); - snd_printd("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); + snd_printd("mce_down - exit = 0x%x\n", wss_inb(chip, CS4231P(REGSEL))); } +EXPORT_SYMBOL(snd_wss_mce_down); -static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size) +static unsigned int snd_wss_get_count(unsigned char format, unsigned int size) { switch (format & 0xe0) { case CS4231_LINEAR_16: @@ -422,19 +468,15 @@ static unsigned int snd_cs4231_get_count return size; } -static int snd_cs4231_trigger(struct snd_pcm_substream *substream, +static int snd_wss_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); int result = 0; unsigned int what; struct snd_pcm_substream *s; int do_start; -#if 0 - printk("codec trigger!!! - what = %i, enable = %i, status = 0x%x\n", what, enable, cs4231_inb(chip, CS4231P(STATUS))); -#endif - switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -466,10 +508,10 @@ static int snd_cs4231_trigger(struct snd if (chip->trigger) chip->trigger(chip, what, 0); } - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); spin_unlock(&chip->reg_lock); #if 0 - snd_cs4231_debug(chip); + snd_wss_debug(chip); #endif return result; } @@ -478,7 +520,7 @@ static int snd_cs4231_trigger(struct snd * CODEC I/O */ -static unsigned char snd_cs4231_get_rate(unsigned int rate) +static unsigned char snd_wss_get_rate(unsigned int rate) { int i; @@ -489,9 +531,9 @@ static unsigned char snd_cs4231_get_rate return freq_bits[ARRAY_SIZE(rates) - 1]; } -static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip, - int format, - int channels) +static unsigned char snd_wss_get_format(struct snd_wss *chip, + int format, + int channels) { unsigned char rformat; @@ -511,7 +553,7 @@ static unsigned char snd_cs4231_get_form return rformat; } -static void snd_cs4231_calibrate_mute(struct snd_cs4231 *chip, int mute) +static void snd_wss_calibrate_mute(struct snd_wss *chip, int mute) { unsigned long flags; @@ -522,30 +564,46 @@ static void snd_cs4231_calibrate_mute(st return; } if (!mute) { - snd_cs4231_dout(chip, CS4231_LEFT_INPUT, chip->image[CS4231_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_INPUT, chip->image[CS4231_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); - } - snd_cs4231_dout(chip, CS4231_AUX1_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX1_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LEFT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_LEFT_LINE_IN, mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); - snd_cs4231_dout(chip, CS4231_RIGHT_LINE_IN, mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); - snd_cs4231_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); - if (chip->hardware == CS4231_HW_INTERWAVE) { - snd_cs4231_dout(chip, CS4231_LEFT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); - snd_cs4231_dout(chip, CS4231_LINE_LEFT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_LINE_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); + snd_wss_dout(chip, CS4231_LEFT_INPUT, + chip->image[CS4231_LEFT_INPUT]); + snd_wss_dout(chip, CS4231_RIGHT_INPUT, + chip->image[CS4231_RIGHT_INPUT]); + snd_wss_dout(chip, CS4231_LOOPBACK, + chip->image[CS4231_LOOPBACK]); + } + snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, + mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); + snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, + mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); + snd_wss_dout(chip, CS4231_AUX2_LEFT_INPUT, + mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); + snd_wss_dout(chip, CS4231_AUX2_RIGHT_INPUT, + mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); + snd_wss_dout(chip, CS4231_LEFT_OUTPUT, + mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); + snd_wss_dout(chip, CS4231_RIGHT_OUTPUT, + mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + snd_wss_dout(chip, CS4231_LEFT_LINE_IN, + mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + snd_wss_dout(chip, CS4231_RIGHT_LINE_IN, + mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + snd_wss_dout(chip, CS4231_MONO_CTRL, + mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); + if (chip->hardware == WSS_HW_INTERWAVE) { + snd_wss_dout(chip, CS4231_LEFT_MIC_INPUT, + mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); + snd_wss_dout(chip, CS4231_RIGHT_MIC_INPUT, + mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); + snd_wss_dout(chip, CS4231_LINE_LEFT_OUTPUT, + mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); + snd_wss_dout(chip, CS4231_LINE_RIGHT_OUTPUT, + mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); } chip->calibrate_mute = mute; spin_unlock_irqrestore(&chip->reg_lock, flags); } -static void snd_cs4231_playback_format(struct snd_cs4231 *chip, +static void snd_wss_playback_format(struct snd_wss *chip, struct snd_pcm_hw_params *params, unsigned char pdfr) { @@ -553,79 +611,87 @@ static void snd_cs4231_playback_format(s int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_cs4231_calibrate_mute(chip, 1); - if (chip->hardware == CS4231_HW_CS4231A || - (chip->hardware & CS4231_HW_CS4232_MASK)) { + snd_wss_calibrate_mute(chip, 1); + if (chip->hardware == WSS_HW_CS4231A || + (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (pdfr & 0x0f)) { /* rate is same? */ - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x10); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x10); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] | 0x10); + chip->image[CS4231_PLAYBK_FORMAT] = pdfr; + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, + chip->image[CS4231_PLAYBK_FORMAT]); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] &= ~0x10); udelay(100); /* Fixes audible clicks at least on GUS MAX */ full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->hardware != CS4231_HW_INTERWAVE && !chip->single_dma) { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, + if (chip->hardware != WSS_HW_INTERWAVE && !chip->single_dma) { + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ? (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) : - pdfr); + pdfr); } else { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, + chip->image[CS4231_PLAYBK_FORMAT] = pdfr); } spin_unlock_irqrestore(&chip->reg_lock, flags); - if (chip->hardware == CS4231_HW_OPL3SA2) + if (chip->hardware == WSS_HW_OPL3SA2) udelay(100); /* this seems to help */ - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); } - snd_cs4231_calibrate_mute(chip, 0); + snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } -static void snd_cs4231_capture_format(struct snd_cs4231 *chip, - struct snd_pcm_hw_params *params, - unsigned char cdfr) +static void snd_wss_capture_format(struct snd_wss *chip, + struct snd_pcm_hw_params *params, + unsigned char cdfr) { unsigned long flags; int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_cs4231_calibrate_mute(chip, 1); - if (chip->hardware == CS4231_HW_CS4231A || - (chip->hardware & CS4231_HW_CS4232_MASK)) { + snd_wss_calibrate_mute(chip, 1); + if (chip->hardware == WSS_HW_CS4231A || + (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (cdfr & 0x0f) || /* rate is same? */ (chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x20); - snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT] = cdfr); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] | 0x20); + snd_wss_out(chip, CS4231_REC_FORMAT, + chip->image[CS4231_REC_FORMAT] = cdfr); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->hardware != CS4231_HW_INTERWAVE) { + if (chip->hardware != WSS_HW_INTERWAVE) { if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, ((chip->single_dma ? cdfr : chip->image[CS4231_PLAYBK_FORMAT]) & 0xf0) | (cdfr & 0x0f)); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - snd_cs4231_mce_up(chip); + snd_wss_mce_down(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); } } - snd_cs4231_out(chip, CS4231_REC_FORMAT, cdfr); + snd_wss_out(chip, CS4231_REC_FORMAT, cdfr); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); } - snd_cs4231_calibrate_mute(chip, 0); + snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -633,130 +699,143 @@ static void snd_cs4231_capture_format(st * Timer interface */ -static unsigned long snd_cs4231_timer_resolution(struct snd_timer * timer) +static unsigned long snd_wss_timer_resolution(struct snd_timer *timer) { - struct snd_cs4231 *chip = snd_timer_chip(timer); - if (chip->hardware & CS4231_HW_CS4236B_MASK) + struct snd_wss *chip = snd_timer_chip(timer); + if (chip->hardware & WSS_HW_CS4236B_MASK) return 14467; else return chip->image[CS4231_PLAYBK_FORMAT] & 1 ? 9969 : 9920; } -static int snd_cs4231_timer_start(struct snd_timer * timer) +static int snd_wss_timer_start(struct snd_timer *timer) { unsigned long flags; unsigned int ticks; - struct snd_cs4231 *chip = snd_timer_chip(timer); + struct snd_wss *chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); ticks = timer->sticks; if ((chip->image[CS4231_ALT_FEATURE_1] & CS4231_TIMER_ENABLE) == 0 || (unsigned char)(ticks >> 8) != chip->image[CS4231_TIMER_HIGH] || (unsigned char)ticks != chip->image[CS4231_TIMER_LOW]) { - snd_cs4231_out(chip, CS4231_TIMER_HIGH, chip->image[CS4231_TIMER_HIGH] = (unsigned char) (ticks >> 8)); - snd_cs4231_out(chip, CS4231_TIMER_LOW, chip->image[CS4231_TIMER_LOW] = (unsigned char) ticks); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | CS4231_TIMER_ENABLE); + chip->image[CS4231_TIMER_HIGH] = (unsigned char) (ticks >> 8); + snd_wss_out(chip, CS4231_TIMER_HIGH, + chip->image[CS4231_TIMER_HIGH]); + chip->image[CS4231_TIMER_LOW] = (unsigned char) ticks; + snd_wss_out(chip, CS4231_TIMER_LOW, + chip->image[CS4231_TIMER_LOW]); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] | + CS4231_TIMER_ENABLE); } spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } -static int snd_cs4231_timer_stop(struct snd_timer * timer) +static int snd_wss_timer_stop(struct snd_timer *timer) { unsigned long flags; - struct snd_cs4231 *chip = snd_timer_chip(timer); + struct snd_wss *chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE); + snd_wss_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE); spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } -static void snd_cs4231_init(struct snd_cs4231 *chip) +static void snd_wss_init(struct snd_wss *chip) { unsigned long flags; - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE snd_printk("init: (1)\n"); #endif - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO | - CS4231_CALIB_MODE); + chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | + CS4231_PLAYBACK_PIO | + CS4231_RECORD_ENABLE | + CS4231_RECORD_PIO | + CS4231_CALIB_MODE); chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE snd_printk("init: (2)\n"); #endif - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); + snd_wss_out(chip, + CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (3) - afei = 0x%x\n", chip->image[CS4231_ALT_FEATURE_1]); + snd_printk("init: (3) - afei = 0x%x\n", + chip->image[CS4231_ALT_FEATURE_1]); #endif spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_2, chip->image[CS4231_ALT_FEATURE_2]); + snd_wss_out(chip, CS4231_ALT_FEATURE_2, + chip->image[CS4231_ALT_FEATURE_2]); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT]); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, + chip->image[CS4231_PLAYBK_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE snd_printk("init: (4)\n"); #endif - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); + snd_wss_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE snd_printk("init: (5)\n"); #endif } -static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode) +static int snd_wss_open(struct snd_wss *chip, unsigned int mode) { unsigned long flags; mutex_lock(&chip->open_mutex); if ((chip->mode & mode) || - ((chip->mode & CS4231_MODE_OPEN) && chip->single_dma)) { + ((chip->mode & WSS_MODE_OPEN) && chip->single_dma)) { mutex_unlock(&chip->open_mutex); return -EAGAIN; } - if (chip->mode & CS4231_MODE_OPEN) { + if (chip->mode & WSS_MODE_OPEN) { chip->mode |= mode; mutex_unlock(&chip->open_mutex); return 0; } /* ok. now enable and ack CODEC IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | + snd_wss_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | CS4231_RECORD_IRQ | CS4231_TIMER_IRQ); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ chip->image[CS4231_PIN_CTRL] |= CS4231_IRQ_ENABLE; - snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | + snd_wss_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); + snd_wss_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | CS4231_RECORD_IRQ | CS4231_TIMER_IRQ); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; @@ -764,48 +843,49 @@ static int snd_cs4231_open(struct snd_cs return 0; } -static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode) +static void snd_wss_close(struct snd_wss *chip, unsigned int mode) { unsigned long flags; mutex_lock(&chip->open_mutex); chip->mode &= ~mode; - if (chip->mode & CS4231_MODE_OPEN) { + if (chip->mode & WSS_MODE_OPEN) { mutex_unlock(&chip->open_mutex); return; } - snd_cs4231_calibrate_mute(chip, 1); + snd_wss_calibrate_mute(chip, 1); /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ chip->image[CS4231_PIN_CTRL] &= ~CS4231_IRQ_ENABLE; - snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); + snd_wss_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); /* now disable record & playback */ if (chip->image[CS4231_IFACE_CTRL] & (CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | CS4231_RECORD_ENABLE | CS4231_RECORD_PIO)) { spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + snd_wss_out(chip, CS4231_IFACE_CTRL, + chip->image[CS4231_IFACE_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); } /* clear IRQ again */ - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + snd_wss_out(chip, CS4231_IRQ_STATUS, 0); + wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_calibrate_mute(chip, 0); + snd_wss_calibrate_mute(chip, 0); chip->mode = 0; mutex_unlock(&chip->open_mutex); @@ -815,59 +895,60 @@ static void snd_cs4231_close(struct snd_ * timer open/close */ -static int snd_cs4231_timer_open(struct snd_timer * timer) +static int snd_wss_timer_open(struct snd_timer *timer) { - struct snd_cs4231 *chip = snd_timer_chip(timer); - snd_cs4231_open(chip, CS4231_MODE_TIMER); + struct snd_wss *chip = snd_timer_chip(timer); + snd_wss_open(chip, WSS_MODE_TIMER); return 0; } -static int snd_cs4231_timer_close(struct snd_timer * timer) +static int snd_wss_timer_close(struct snd_timer *timer) { - struct snd_cs4231 *chip = snd_timer_chip(timer); - snd_cs4231_close(chip, CS4231_MODE_TIMER); + struct snd_wss *chip = snd_timer_chip(timer); + snd_wss_close(chip, WSS_MODE_TIMER); return 0; } -static struct snd_timer_hardware snd_cs4231_timer_table = +static struct snd_timer_hardware snd_wss_timer_table = { .flags = SNDRV_TIMER_HW_AUTO, .resolution = 9945, .ticks = 65535, - .open = snd_cs4231_timer_open, - .close = snd_cs4231_timer_close, - .c_resolution = snd_cs4231_timer_resolution, - .start = snd_cs4231_timer_start, - .stop = snd_cs4231_timer_stop, + .open = snd_wss_timer_open, + .close = snd_wss_timer_close, + .c_resolution = snd_wss_timer_resolution, + .start = snd_wss_timer_start, + .stop = snd_wss_timer_stop, }; /* * ok.. exported functions.. */ -static int snd_cs4231_playback_hw_params(struct snd_pcm_substream *substream, +static int snd_wss_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); unsigned char new_pdfr; int err; if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) return err; - new_pdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | - snd_cs4231_get_rate(params_rate(hw_params)); + new_pdfr = snd_wss_get_format(chip, params_format(hw_params), + params_channels(hw_params)) | + snd_wss_get_rate(params_rate(hw_params)); chip->set_playback_format(chip, hw_params, new_pdfr); return 0; } -static int snd_cs4231_playback_hw_free(struct snd_pcm_substream *substream) +static int snd_wss_playback_hw_free(struct snd_pcm_substream *substream) { return snd_pcm_lib_free_pages(substream); } -static int snd_cs4231_playback_prepare(struct snd_pcm_substream *substream) +static int snd_wss_playback_prepare(struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; unsigned int size = snd_pcm_lib_buffer_bytes(substream); @@ -877,39 +958,40 @@ static int snd_cs4231_playback_prepare(s chip->p_dma_size = size; chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO); snd_dma_program(chip->dma1, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); - count = snd_cs4231_get_count(chip->image[CS4231_PLAYBK_FORMAT], count) - 1; - snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); + count = snd_wss_get_count(chip->image[CS4231_PLAYBK_FORMAT], count) - 1; + snd_wss_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); + snd_wss_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); spin_unlock_irqrestore(&chip->reg_lock, flags); #if 0 - snd_cs4231_debug(chip); + snd_wss_debug(chip); #endif return 0; } -static int snd_cs4231_capture_hw_params(struct snd_pcm_substream *substream, +static int snd_wss_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); unsigned char new_cdfr; int err; if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) return err; - new_cdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | - snd_cs4231_get_rate(params_rate(hw_params)); + new_cdfr = snd_wss_get_format(chip, params_format(hw_params), + params_channels(hw_params)) | + snd_wss_get_rate(params_rate(hw_params)); chip->set_capture_format(chip, hw_params, new_cdfr); return 0; } -static int snd_cs4231_capture_hw_free(struct snd_pcm_substream *substream) +static int snd_wss_capture_hw_free(struct snd_pcm_substream *substream) { return snd_pcm_lib_free_pages(substream); } -static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream) +static int snd_wss_capture_prepare(struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned long flags; unsigned int size = snd_pcm_lib_buffer_bytes(substream); @@ -919,49 +1001,52 @@ static int snd_cs4231_capture_prepare(st chip->c_dma_size = size; chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); - count = snd_cs4231_get_count(chip->image[CS4231_REC_FORMAT], count) - 1; - if (chip->single_dma && chip->hardware != CS4231_HW_INTERWAVE) { - snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); + count = snd_wss_get_count(chip->image[CS4231_REC_FORMAT], count) - 1; + if (chip->single_dma && chip->hardware != WSS_HW_INTERWAVE) { + snd_wss_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); + snd_wss_out(chip, CS4231_PLY_UPR_CNT, + (unsigned char) (count >> 8)); } else { - snd_cs4231_out(chip, CS4231_REC_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_REC_UPR_CNT, (unsigned char) (count >> 8)); + snd_wss_out(chip, CS4231_REC_LWR_CNT, (unsigned char) count); + snd_wss_out(chip, CS4231_REC_UPR_CNT, + (unsigned char) (count >> 8)); } spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } -void snd_cs4231_overrange(struct snd_cs4231 *chip) +void snd_wss_overrange(struct snd_wss *chip) { unsigned long flags; unsigned char res; spin_lock_irqsave(&chip->reg_lock, flags); - res = snd_cs4231_in(chip, CS4231_TEST_INIT); + res = snd_wss_in(chip, CS4231_TEST_INIT); spin_unlock_irqrestore(&chip->reg_lock, flags); if (res & (0x08 | 0x02)) /* detect overrange only above 0dB; may be user selectable? */ chip->capture_substream->runtime->overrange++; } +EXPORT_SYMBOL(snd_wss_overrange); -irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id) +irqreturn_t snd_wss_interrupt(int irq, void *dev_id) { - struct snd_cs4231 *chip = dev_id; + struct snd_wss *chip = dev_id; unsigned char status; - status = snd_cs4231_in(chip, CS4231_IRQ_STATUS); + status = snd_wss_in(chip, CS4231_IRQ_STATUS); if (status & CS4231_TIMER_IRQ) { if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); } - if (chip->single_dma && chip->hardware != CS4231_HW_INTERWAVE) { + if (chip->single_dma && chip->hardware != WSS_HW_INTERWAVE) { if (status & CS4231_PLAYBACK_IRQ) { - if (chip->mode & CS4231_MODE_PLAY) { + if (chip->mode & WSS_MODE_PLAY) { if (chip->playback_substream) snd_pcm_period_elapsed(chip->playback_substream); } - if (chip->mode & CS4231_MODE_RECORD) { + if (chip->mode & WSS_MODE_RECORD) { if (chip->capture_substream) { - snd_cs4231_overrange(chip); + snd_wss_overrange(chip); snd_pcm_period_elapsed(chip->capture_substream); } } @@ -973,21 +1058,23 @@ irqreturn_t snd_cs4231_interrupt(int irq } if (status & CS4231_RECORD_IRQ) { if (chip->capture_substream) { - snd_cs4231_overrange(chip); + snd_wss_overrange(chip); snd_pcm_period_elapsed(chip->capture_substream); } } } spin_lock(&chip->reg_lock); - snd_cs4231_outm(chip, CS4231_IRQ_STATUS, ~CS4231_ALL_IRQS | ~status, 0); + snd_wss_outm(chip, CS4231_IRQ_STATUS, ~CS4231_ALL_IRQS | ~status, 0); spin_unlock(&chip->reg_lock); return IRQ_HANDLED; } +EXPORT_SYMBOL(snd_wss_interrupt); -static snd_pcm_uframes_t snd_cs4231_playback_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t snd_wss_playback_pointer + (struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); size_t ptr; if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) @@ -996,9 +1083,10 @@ static snd_pcm_uframes_t snd_cs4231_play return bytes_to_frames(substream->runtime, ptr); } -static snd_pcm_uframes_t snd_cs4231_capture_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t snd_wss_capture_pointer + (struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); size_t ptr; if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE)) @@ -1011,7 +1099,7 @@ static snd_pcm_uframes_t snd_cs4231_capt */ -static int snd_cs4231_probe(struct snd_cs4231 *chip) +static int snd_wss_probe(struct snd_wss *chip) { unsigned long flags; int i, id, rev; @@ -1019,103 +1107,104 @@ static int snd_cs4231_probe(struct snd_c unsigned int hw; #if 0 - snd_cs4231_debug(chip); + snd_wss_debug(chip); #endif id = 0; for (i = 0; i < 50; i++) { mb(); - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) udelay(2000); else { spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - id = snd_cs4231_in(chip, CS4231_MISC_INFO) & 0x0f; + snd_wss_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + id = snd_wss_in(chip, CS4231_MISC_INFO) & 0x0f; spin_unlock_irqrestore(&chip->reg_lock, flags); if (id == 0x0a) break; /* this is valid value */ } } - snd_printdd("cs4231: port = 0x%lx, id = 0x%x\n", chip->port, id); + snd_printdd("wss: port = 0x%lx, id = 0x%x\n", chip->port, id); if (id != 0x0a) return -ENODEV; /* no valid device found */ - if (((hw = chip->hardware) & CS4231_HW_TYPE_MASK) == CS4231_HW_DETECT) { - rev = snd_cs4231_in(chip, CS4231_VERSION) & 0xe7; + hw = chip->hardware; + if ((hw & WSS_HW_TYPE_MASK) == WSS_HW_DETECT) { + rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; snd_printdd("CS4231: VERSION (I25) = 0x%x\n", rev); if (rev == 0x80) { - unsigned char tmp = snd_cs4231_in(chip, 23); - snd_cs4231_out(chip, 23, ~tmp); - if (snd_cs4231_in(chip, 23) != tmp) - chip->hardware = CS4231_HW_AD1845; + unsigned char tmp = snd_wss_in(chip, 23); + snd_wss_out(chip, 23, ~tmp); + if (snd_wss_in(chip, 23) != tmp) + chip->hardware = WSS_HW_AD1845; else - chip->hardware = CS4231_HW_CS4231; + chip->hardware = WSS_HW_CS4231; } else if (rev == 0xa0) { - chip->hardware = CS4231_HW_CS4231A; + chip->hardware = WSS_HW_CS4231A; } else if (rev == 0xa2) { - chip->hardware = CS4231_HW_CS4232; + chip->hardware = WSS_HW_CS4232; } else if (rev == 0xb2) { - chip->hardware = CS4231_HW_CS4232A; + chip->hardware = WSS_HW_CS4232A; } else if (rev == 0x83) { - chip->hardware = CS4231_HW_CS4236; + chip->hardware = WSS_HW_CS4236; } else if (rev == 0x03) { - chip->hardware = CS4231_HW_CS4236B; + chip->hardware = WSS_HW_CS4236B; } else { snd_printk("unknown CS chip with version 0x%x\n", rev); return -ENODEV; /* unknown CS4231 chip? */ } } spin_lock_irqsave(&chip->reg_lock, flags); - cs4231_inb(chip, CS4231P(STATUS)); /* clear any pendings IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); + wss_inb(chip, CS4231P(STATUS)); /* clear any pendings IRQ */ + wss_outb(chip, CS4231P(STATUS), 0); mb(); spin_unlock_irqrestore(&chip->reg_lock, flags); chip->image[CS4231_MISC_INFO] = CS4231_MODE2; switch (chip->hardware) { - case CS4231_HW_INTERWAVE: + case WSS_HW_INTERWAVE: chip->image[CS4231_MISC_INFO] = CS4231_IW_MODE3; break; - case CS4231_HW_CS4235: - case CS4231_HW_CS4236B: - case CS4231_HW_CS4237B: - case CS4231_HW_CS4238B: - case CS4231_HW_CS4239: - if (hw == CS4231_HW_DETECT3) + case WSS_HW_CS4235: + case WSS_HW_CS4236B: + case WSS_HW_CS4237B: + case WSS_HW_CS4238B: + case WSS_HW_CS4239: + if (hw == WSS_HW_DETECT3) chip->image[CS4231_MISC_INFO] = CS4231_4236_MODE3; else - chip->hardware = CS4231_HW_CS4236; + chip->hardware = WSS_HW_CS4236; break; } chip->image[CS4231_IFACE_CTRL] = (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | (chip->single_dma ? CS4231_SINGLE_DMA : 0); - if (chip->hardware != CS4231_HW_OPTI93X) { + if (chip->hardware != WSS_HW_OPTI93X) { chip->image[CS4231_ALT_FEATURE_1] = 0x80; chip->image[CS4231_ALT_FEATURE_2] = - chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01; } ptr = (unsigned char *) &chip->image; - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); for (i = 0; i < 32; i++) /* ok.. fill all CS4231 registers */ - snd_cs4231_out(chip, i, *ptr++); + snd_wss_out(chip, i, *ptr++); spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_up(chip); - snd_cs4231_mce_down(chip); + snd_wss_mce_up(chip); + snd_wss_mce_down(chip); mdelay(2); /* ok.. try check hardware version for CS4236+ chips */ - if ((hw & CS4231_HW_TYPE_MASK) == CS4231_HW_DETECT) { - if (chip->hardware == CS4231_HW_CS4236B) { + if ((hw & WSS_HW_TYPE_MASK) == WSS_HW_DETECT) { + if (chip->hardware == WSS_HW_CS4236B) { rev = snd_cs4236_ext_in(chip, CS4236_VERSION); snd_cs4236_ext_out(chip, CS4236_VERSION, 0xff); id = snd_cs4236_ext_in(chip, CS4236_VERSION); snd_cs4236_ext_out(chip, CS4236_VERSION, rev); snd_printdd("CS4231: ext version; rev = 0x%x, id = 0x%x\n", rev, id); if ((id & 0x1f) == 0x1d) { /* CS4235 */ - chip->hardware = CS4231_HW_CS4235; + chip->hardware = WSS_HW_CS4235; switch (id >> 5) { case 4: case 5: @@ -1130,13 +1219,13 @@ static int snd_cs4231_probe(struct snd_c case 5: case 6: case 7: - chip->hardware = CS4231_HW_CS4236B; + chip->hardware = WSS_HW_CS4236B; break; default: snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id); } } else if ((id & 0x1f) == 0x08) { /* CS4237B */ - chip->hardware = CS4231_HW_CS4237B; + chip->hardware = WSS_HW_CS4237B; switch (id >> 5) { case 4: case 5: @@ -1147,7 +1236,7 @@ static int snd_cs4231_probe(struct snd_c snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id); } } else if ((id & 0x1f) == 0x09) { /* CS4238B */ - chip->hardware = CS4231_HW_CS4238B; + chip->hardware = WSS_HW_CS4238B; switch (id >> 5) { case 5: case 6: @@ -1157,7 +1246,7 @@ static int snd_cs4231_probe(struct snd_c snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id); } } else if ((id & 0x1f) == 0x1e) { /* CS4239 */ - chip->hardware = CS4231_HW_CS4239; + chip->hardware = WSS_HW_CS4239; switch (id >> 5) { case 4: case 5: @@ -1178,7 +1267,7 @@ static int snd_cs4231_probe(struct snd_c */ -static struct snd_pcm_hardware snd_cs4231_playback = +static struct snd_pcm_hardware snd_wss_playback = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | @@ -1199,7 +1288,7 @@ static struct snd_pcm_hardware snd_cs423 .fifo_size = 0, }; -static struct snd_pcm_hardware snd_cs4231_capture = +static struct snd_pcm_hardware snd_wss_capture = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | @@ -1224,21 +1313,21 @@ static struct snd_pcm_hardware snd_cs423 */ -static int snd_cs4231_playback_open(struct snd_pcm_substream *substream) +static int snd_wss_playback_open(struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; - runtime->hw = snd_cs4231_playback; + runtime->hw = snd_wss_playback; /* hardware bug in InterWave chipset */ - if (chip->hardware == CS4231_HW_INTERWAVE && chip->dma1 > 3) + if (chip->hardware == WSS_HW_INTERWAVE && chip->dma1 > 3) runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_MU_LAW; /* hardware limitation of cheap chips */ - if (chip->hardware == CS4231_HW_CS4235 || - chip->hardware == CS4231_HW_CS4239) + if (chip->hardware == WSS_HW_CS4235 || + chip->hardware == WSS_HW_CS4239) runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); @@ -1249,7 +1338,8 @@ static int snd_cs4231_playback_open(stru return err; } - if ((err = snd_cs4231_open(chip, CS4231_MODE_PLAY)) < 0) { + err = snd_wss_open(chip, WSS_MODE_PLAY); + if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma1); snd_free_pages(runtime->dma_area, runtime->dma_bytes); @@ -1261,17 +1351,17 @@ static int snd_cs4231_playback_open(stru return 0; } -static int snd_cs4231_capture_open(struct snd_pcm_substream *substream) +static int snd_wss_capture_open(struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; int err; - runtime->hw = snd_cs4231_capture; + runtime->hw = snd_wss_capture; /* hardware limitation of cheap chips */ - if (chip->hardware == CS4231_HW_CS4235 || - chip->hardware == CS4231_HW_CS4239) + if (chip->hardware == WSS_HW_CS4235 || + chip->hardware == WSS_HW_CS4239) runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); @@ -1282,7 +1372,8 @@ static int snd_cs4231_capture_open(struc return err; } - if ((err = snd_cs4231_open(chip, CS4231_MODE_RECORD)) < 0) { + err = snd_wss_open(chip, WSS_MODE_RECORD); + if (err < 0) { if (chip->release_dma) chip->release_dma(chip, chip->dma_private_data, chip->dma2); snd_free_pages(runtime->dma_area, runtime->dma_bytes); @@ -1294,28 +1385,28 @@ static int snd_cs4231_capture_open(struc return 0; } -static int snd_cs4231_playback_close(struct snd_pcm_substream *substream) +static int snd_wss_playback_close(struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); chip->playback_substream = NULL; - snd_cs4231_close(chip, CS4231_MODE_PLAY); + snd_wss_close(chip, WSS_MODE_PLAY); return 0; } -static int snd_cs4231_capture_close(struct snd_pcm_substream *substream) +static int snd_wss_capture_close(struct snd_pcm_substream *substream) { - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_wss *chip = snd_pcm_substream_chip(substream); chip->capture_substream = NULL; - snd_cs4231_close(chip, CS4231_MODE_RECORD); + snd_wss_close(chip, WSS_MODE_RECORD); return 0; } #ifdef CONFIG_PM /* lowlevel suspend callback for CS4231 */ -static void snd_cs4231_suspend(struct snd_cs4231 *chip) +static void snd_wss_suspend(struct snd_wss *chip) { int reg; unsigned long flags; @@ -1323,67 +1414,68 @@ static void snd_cs4231_suspend(struct sn snd_pcm_suspend_all(chip->pcm); spin_lock_irqsave(&chip->reg_lock, flags); for (reg = 0; reg < 32; reg++) - chip->image[reg] = snd_cs4231_in(chip, reg); + chip->image[reg] = snd_wss_in(chip, reg); spin_unlock_irqrestore(&chip->reg_lock, flags); } /* lowlevel resume callback for CS4231 */ -static void snd_cs4231_resume(struct snd_cs4231 *chip) +static void snd_wss_resume(struct snd_wss *chip) { int reg; unsigned long flags; /* int timeout; */ - snd_cs4231_mce_up(chip); + snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); for (reg = 0; reg < 32; reg++) { switch (reg) { case CS4231_VERSION: break; default: - snd_cs4231_out(chip, reg, chip->image[reg]); + snd_wss_out(chip, reg, chip->image[reg]); break; } } spin_unlock_irqrestore(&chip->reg_lock, flags); #if 1 - snd_cs4231_mce_down(chip); + snd_wss_mce_down(chip); #else /* The following is a workaround to avoid freeze after resume on TP600E. - This is the first half of copy of snd_cs4231_mce_down(), but doesn't + This is the first half of copy of snd_wss_mce_down(), but doesn't include rescheduling. -- iwai */ - snd_cs4231_busy_wait(chip); + snd_wss_busy_wait(chip); spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit &= ~CS4231_MCE; - timeout = cs4231_inb(chip, CS4231P(REGSEL)); - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); + timeout = wss_inb(chip, CS4231P(REGSEL)); + wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port); if ((timeout & CS4231_MCE) == 0 || - !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { + !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { return; } - snd_cs4231_busy_wait(chip); + snd_wss_busy_wait(chip); #endif } #endif /* CONFIG_PM */ -static int snd_cs4231_free(struct snd_cs4231 *chip) +static int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); if (chip->irq >= 0) { disable_irq(chip->irq); - if (!(chip->hwshare & CS4231_HWSHARE_IRQ)) + if (!(chip->hwshare & WSS_HWSHARE_IRQ)) free_irq(chip->irq, (void *) chip); } - if (!(chip->hwshare & CS4231_HWSHARE_DMA1) && chip->dma1 >= 0) { + if (!(chip->hwshare & WSS_HWSHARE_DMA1) && chip->dma1 >= 0) { snd_dma_disable(chip->dma1); free_dma(chip->dma1); } - if (!(chip->hwshare & CS4231_HWSHARE_DMA2) && chip->dma2 >= 0 && chip->dma2 != chip->dma1) { + if (!(chip->hwshare & WSS_HWSHARE_DMA2) && + chip->dma2 >= 0 && chip->dma2 != chip->dma1) { snd_dma_disable(chip->dma2); free_dma(chip->dma2); } @@ -1393,39 +1485,40 @@ static int snd_cs4231_free(struct snd_cs return 0; } -static int snd_cs4231_dev_free(struct snd_device *device) +static int snd_wss_dev_free(struct snd_device *device) { - struct snd_cs4231 *chip = device->device_data; - return snd_cs4231_free(chip); + struct snd_wss *chip = device->device_data; + return snd_wss_free(chip); } -const char *snd_cs4231_chip_id(struct snd_cs4231 *chip) +const char *snd_wss_chip_id(struct snd_wss *chip) { switch (chip->hardware) { - case CS4231_HW_CS4231: return "CS4231"; - case CS4231_HW_CS4231A: return "CS4231A"; - case CS4231_HW_CS4232: return "CS4232"; - case CS4231_HW_CS4232A: return "CS4232A"; - case CS4231_HW_CS4235: return "CS4235"; - case CS4231_HW_CS4236: return "CS4236"; - case CS4231_HW_CS4236B: return "CS4236B"; - case CS4231_HW_CS4237B: return "CS4237B"; - case CS4231_HW_CS4238B: return "CS4238B"; - case CS4231_HW_CS4239: return "CS4239"; - case CS4231_HW_INTERWAVE: return "AMD InterWave"; - case CS4231_HW_OPL3SA2: return chip->card->shortname; - case CS4231_HW_AD1845: return "AD1845"; - case CS4231_HW_OPTI93X: return "OPTi 93x"; + case WSS_HW_CS4231: return "CS4231"; + case WSS_HW_CS4231A: return "CS4231A"; + case WSS_HW_CS4232: return "CS4232"; + case WSS_HW_CS4232A: return "CS4232A"; + case WSS_HW_CS4235: return "CS4235"; + case WSS_HW_CS4236: return "CS4236"; + case WSS_HW_CS4236B: return "CS4236B"; + case WSS_HW_CS4237B: return "CS4237B"; + case WSS_HW_CS4238B: return "CS4238B"; + case WSS_HW_CS4239: return "CS4239"; + case WSS_HW_INTERWAVE: return "AMD InterWave"; + case WSS_HW_OPL3SA2: return chip->card->shortname; + case WSS_HW_AD1845: return "AD1845"; + case WSS_HW_OPTI93X: return "OPTi 93x"; default: return "???"; } } +EXPORT_SYMBOL(snd_wss_chip_id); -static int snd_cs4231_new(struct snd_card *card, +static int snd_wss_new(struct snd_card *card, unsigned short hardware, unsigned short hwshare, - struct snd_cs4231 ** rchip) + struct snd_wss **rchip) { - struct snd_cs4231 *chip; + struct snd_wss *chip; *rchip = NULL; chip = kzalloc(sizeof(*chip), GFP_KERNEL); @@ -1438,35 +1531,35 @@ static int snd_cs4231_new(struct snd_car mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); chip->card = card; - chip->rate_constraint = snd_cs4231_xrate; - chip->set_playback_format = snd_cs4231_playback_format; - chip->set_capture_format = snd_cs4231_capture_format; - if (chip->hardware == CS4231_HW_OPTI93X) + chip->rate_constraint = snd_wss_xrate; + chip->set_playback_format = snd_wss_playback_format; + chip->set_capture_format = snd_wss_capture_format; + if (chip->hardware == WSS_HW_OPTI93X) memcpy(&chip->image, &snd_opti93x_original_image, sizeof(snd_opti93x_original_image)); else - memcpy(&chip->image, &snd_cs4231_original_image, - sizeof(snd_cs4231_original_image)); + memcpy(&chip->image, &snd_wss_original_image, + sizeof(snd_wss_original_image)); - *rchip = chip; - return 0; + *rchip = chip; + return 0; } -int snd_cs4231_create(struct snd_card *card, - unsigned long port, - unsigned long cport, +int snd_wss_create(struct snd_card *card, + unsigned long port, + unsigned long cport, int irq, int dma1, int dma2, unsigned short hardware, unsigned short hwshare, - struct snd_cs4231 ** rchip) + struct snd_wss **rchip) { static struct snd_device_ops ops = { - .dev_free = snd_cs4231_dev_free, + .dev_free = snd_wss_dev_free, }; - struct snd_cs4231 *chip; + struct snd_wss *chip; int err; - err = snd_cs4231_new(card, hardware, hwshare, &chip); + err = snd_wss_new(card, hardware, hwshare, &chip); if (err < 0) return err; @@ -1475,32 +1568,36 @@ int snd_cs4231_create(struct snd_card *c chip->dma2 = -1; if ((chip->res_port = request_region(port, 4, "CS4231")) == NULL) { - snd_printk(KERN_ERR "cs4231: can't grab port 0x%lx\n", port); - snd_cs4231_free(chip); + snd_printk(KERN_ERR "wss: can't grab port 0x%lx\n", port); + snd_wss_free(chip); return -EBUSY; } chip->port = port; if ((long)cport >= 0 && (chip->res_cport = request_region(cport, 8, "CS4232 Control")) == NULL) { - snd_printk(KERN_ERR "cs4231: can't grab control port 0x%lx\n", cport); - snd_cs4231_free(chip); + snd_printk(KERN_ERR + "wss: can't grab control port 0x%lx\n", cport); + snd_wss_free(chip); return -ENODEV; } chip->cport = cport; - if (!(hwshare & CS4231_HWSHARE_IRQ) && request_irq(irq, snd_cs4231_interrupt, IRQF_DISABLED, "CS4231", (void *) chip)) { - snd_printk(KERN_ERR "cs4231: can't grab IRQ %d\n", irq); - snd_cs4231_free(chip); - return -EBUSY; - } + if (!(hwshare & WSS_HWSHARE_IRQ)) + if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED, + "CS4231", (void *) chip)) { + snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq); + snd_wss_free(chip); + return -EBUSY; + } chip->irq = irq; - if (!(hwshare & CS4231_HWSHARE_DMA1) && request_dma(dma1, "CS4231 - 1")) { - snd_printk(KERN_ERR "cs4231: can't grab DMA1 %d\n", dma1); - snd_cs4231_free(chip); + if (!(hwshare & WSS_HWSHARE_DMA1) && request_dma(dma1, "CS4231 - 1")) { + snd_printk(KERN_ERR "wss: can't grab DMA1 %d\n", dma1); + snd_wss_free(chip); return -EBUSY; } chip->dma1 = dma1; - if (!(hwshare & CS4231_HWSHARE_DMA2) && dma1 != dma2 && dma2 >= 0 && request_dma(dma2, "CS4231 - 2")) { - snd_printk(KERN_ERR "cs4231: can't grab DMA2 %d\n", dma2); - snd_cs4231_free(chip); + if (!(hwshare & WSS_HWSHARE_DMA2) && dma1 != dma2 && + dma2 >= 0 && request_dma(dma2, "CS4231 - 2")) { + snd_printk(KERN_ERR "wss: can't grab DMA2 %d\n", dma2); + snd_wss_free(chip); return -EBUSY; } if (dma1 == dma2 || dma2 < 0) { @@ -1510,14 +1607,14 @@ int snd_cs4231_create(struct snd_card *c chip->dma2 = dma2; /* global setup */ - if (snd_cs4231_probe(chip) < 0) { - snd_cs4231_free(chip); + if (snd_wss_probe(chip) < 0) { + snd_wss_free(chip); return -ENODEV; } - snd_cs4231_init(chip); + snd_wss_init(chip); #if 0 - if (chip->hardware & CS4231_HW_CS4232_MASK) { + if (chip->hardware & WSS_HW_CS4232_MASK) { if (chip->res_cport == NULL) snd_printk("CS4232 control port features are not accessible\n"); } @@ -1525,43 +1622,44 @@ int snd_cs4231_create(struct snd_card *c /* Register device */ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_cs4231_free(chip); + snd_wss_free(chip); return err; } #ifdef CONFIG_PM /* Power Management */ - chip->suspend = snd_cs4231_suspend; - chip->resume = snd_cs4231_resume; + chip->suspend = snd_wss_suspend; + chip->resume = snd_wss_resume; #endif *rchip = chip; return 0; } +EXPORT_SYMBOL(snd_wss_create); -static struct snd_pcm_ops snd_cs4231_playback_ops = { - .open = snd_cs4231_playback_open, - .close = snd_cs4231_playback_close, +static struct snd_pcm_ops snd_wss_playback_ops = { + .open = snd_wss_playback_open, + .close = snd_wss_playback_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_cs4231_playback_hw_params, - .hw_free = snd_cs4231_playback_hw_free, - .prepare = snd_cs4231_playback_prepare, - .trigger = snd_cs4231_trigger, - .pointer = snd_cs4231_playback_pointer, + .hw_params = snd_wss_playback_hw_params, + .hw_free = snd_wss_playback_hw_free, + .prepare = snd_wss_playback_prepare, + .trigger = snd_wss_trigger, + .pointer = snd_wss_playback_pointer, }; -static struct snd_pcm_ops snd_cs4231_capture_ops = { - .open = snd_cs4231_capture_open, - .close = snd_cs4231_capture_close, +static struct snd_pcm_ops snd_wss_capture_ops = { + .open = snd_wss_capture_open, + .close = snd_wss_capture_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_cs4231_capture_hw_params, - .hw_free = snd_cs4231_capture_hw_free, - .prepare = snd_cs4231_capture_prepare, - .trigger = snd_cs4231_trigger, - .pointer = snd_cs4231_capture_pointer, + .hw_params = snd_wss_capture_hw_params, + .hw_free = snd_wss_capture_hw_free, + .prepare = snd_wss_capture_prepare, + .trigger = snd_wss_trigger, + .pointer = snd_wss_capture_pointer, }; -int snd_cs4231_pcm(struct snd_cs4231 *chip, int device, struct snd_pcm **rpcm) +int snd_wss_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; @@ -1573,17 +1671,17 @@ int snd_cs4231_pcm(struct snd_cs4231 *ch mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4231_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cs4231_capture_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_wss_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_wss_capture_ops); /* global setup */ pcm->private_data = chip; pcm->info_flags = 0; if (chip->single_dma) pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX; - if (chip->hardware != CS4231_HW_INTERWAVE) + if (chip->hardware != WSS_HW_INTERWAVE) pcm->info_flags |= SNDRV_PCM_INFO_JOINT_DUPLEX; - strcpy(pcm->name, snd_cs4231_chip_id(chip)); + strcpy(pcm->name, snd_wss_chip_id(chip)); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), @@ -1594,14 +1692,15 @@ int snd_cs4231_pcm(struct snd_cs4231 *ch *rpcm = pcm; return 0; } +EXPORT_SYMBOL(snd_wss_pcm); -static void snd_cs4231_timer_free(struct snd_timer *timer) +static void snd_wss_timer_free(struct snd_timer *timer) { - struct snd_cs4231 *chip = timer->private_data; + struct snd_wss *chip = timer->private_data; chip->timer = NULL; } -int snd_cs4231_timer(struct snd_cs4231 *chip, int device, struct snd_timer **rtimer) +int snd_wss_timer(struct snd_wss *chip, int device, struct snd_timer **rtimer) { struct snd_timer *timer; struct snd_timer_id tid; @@ -1615,21 +1714,23 @@ int snd_cs4231_timer(struct snd_cs4231 * tid.subdevice = 0; if ((err = snd_timer_new(chip->card, "CS4231", &tid, &timer)) < 0) return err; - strcpy(timer->name, snd_cs4231_chip_id(chip)); + strcpy(timer->name, snd_wss_chip_id(chip)); timer->private_data = chip; - timer->private_free = snd_cs4231_timer_free; - timer->hw = snd_cs4231_timer_table; + timer->private_free = snd_wss_timer_free; + timer->hw = snd_wss_timer_table; chip->timer = timer; if (rtimer) *rtimer = timer; return 0; } +EXPORT_SYMBOL(snd_wss_timer); /* * MIXER part */ -static int snd_cs4231_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +static int snd_wss_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { static char *texts[4] = { "Line", "Aux", "Mic", "Mix" @@ -1641,7 +1742,7 @@ static int snd_cs4231_info_mux(struct sn "Line", "Synth", "Mic", "Mix" }; char **ptexts = texts; - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); snd_assert(chip->card != NULL, return -EINVAL); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -1652,16 +1753,17 @@ static int snd_cs4231_info_mux(struct sn if (!strcmp(chip->card->driver, "GUS MAX")) ptexts = gusmax_texts; switch (chip->hardware) { - case CS4231_HW_INTERWAVE: ptexts = gusmax_texts; break; - case CS4231_HW_OPL3SA2: ptexts = opl3sa_texts; break; + case WSS_HW_INTERWAVE: ptexts = gusmax_texts; break; + case WSS_HW_OPL3SA2: ptexts = opl3sa_texts; break; } strcpy(uinfo->value.enumerated.name, ptexts[uinfo->value.enumerated.item]); return 0; } -static int snd_cs4231_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_wss_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); @@ -1671,9 +1773,10 @@ static int snd_cs4231_get_mux(struct snd return 0; } -static int snd_cs4231_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_wss_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; unsigned short left, right; int change; @@ -1687,14 +1790,15 @@ static int snd_cs4231_put_mux(struct snd left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left; right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right; change = left != chip->image[CS4231_LEFT_INPUT] || - right != chip->image[CS4231_RIGHT_INPUT]; - snd_cs4231_out(chip, CS4231_LEFT_INPUT, left); - snd_cs4231_out(chip, CS4231_RIGHT_INPUT, right); + right != chip->image[CS4231_RIGHT_INPUT]; + snd_wss_out(chip, CS4231_LEFT_INPUT, left); + snd_wss_out(chip, CS4231_RIGHT_INPUT, right); spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } -int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +int snd_wss_info_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 16) & 0xff; @@ -1704,10 +1808,12 @@ int snd_cs4231_info_single(struct snd_kc uinfo->value.integer.max = mask; return 0; } +EXPORT_SYMBOL(snd_wss_info_single); -int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +int snd_wss_get_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -1721,10 +1827,12 @@ int snd_cs4231_get_single(struct snd_kco ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; return 0; } +EXPORT_SYMBOL(snd_wss_get_single); -int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +int snd_wss_put_single(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int reg = kcontrol->private_value & 0xff; int shift = (kcontrol->private_value >> 8) & 0xff; @@ -1740,12 +1848,14 @@ int snd_cs4231_put_single(struct snd_kco spin_lock_irqsave(&chip->reg_lock, flags); val = (chip->image[reg] & ~(mask << shift)) | val; change = val != chip->image[reg]; - snd_cs4231_out(chip, reg, val); + snd_wss_out(chip, reg, val); spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } +EXPORT_SYMBOL(snd_wss_put_single); -int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +int snd_wss_info_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 24) & 0xff; @@ -1755,10 +1865,12 @@ int snd_cs4231_info_double(struct snd_kc uinfo->value.integer.max = mask; return 0; } +EXPORT_SYMBOL(snd_wss_info_double); -int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +int snd_wss_get_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -1777,10 +1889,12 @@ int snd_cs4231_get_double(struct snd_kco } return 0; } +EXPORT_SYMBOL(snd_wss_get_double); -int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +int snd_wss_put_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + struct snd_wss *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; int left_reg = kcontrol->private_value & 0xff; int right_reg = (kcontrol->private_value >> 8) & 0xff; @@ -1803,81 +1917,91 @@ int snd_cs4231_put_double(struct snd_kco val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_cs4231_out(chip, left_reg, val1); - snd_cs4231_out(chip, right_reg, val2); + snd_wss_out(chip, left_reg, val1); + snd_wss_out(chip, right_reg, val2); spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } +EXPORT_SYMBOL(snd_wss_put_double); -static struct snd_kcontrol_new snd_cs4231_controls[] = { -CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), -CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -CS4231_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), -CS4231_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +static struct snd_kcontrol_new snd_wss_controls[] = { +WSS_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +WSS_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +WSS_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE("Aux Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE("Aux Playback Switch", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("Aux Playback Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), +WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), +WSS_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +WSS_DOUBLE("Capture Volume", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", - .info = snd_cs4231_info_mux, - .get = snd_cs4231_get_mux, - .put = snd_cs4231_put_mux, + .info = snd_wss_info_mux, + .get = snd_wss_get_mux, + .put = snd_wss_put_mux, }, -CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) +WSS_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +WSS_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) }; - + static struct snd_kcontrol_new snd_opti93x_controls[] = { -CS4231_DOUBLE("Master Playback Switch", 0, +WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -CS4231_DOUBLE("Master Playback Volume", 0, +WSS_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), -CS4231_DOUBLE("PCM Playback Switch", 0, +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, +WSS_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -CS4231_DOUBLE("FM Playback Switch", 0, +WSS_DOUBLE("FM Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("FM Playback Volume", 0, +WSS_DOUBLE("FM Playback Volume", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Line Playback Switch", 0, +WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -CS4231_DOUBLE("Line Playback Volume", 0, +WSS_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), -CS4231_DOUBLE("Mic Playback Switch", 0, +WSS_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Mic Playback Volume", 0, +WSS_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Mic Boost", 0, +WSS_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -CS4231_DOUBLE("CD Playback Switch", 0, +WSS_DOUBLE("CD Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("CD Playback Volume", 0, +WSS_DOUBLE("CD Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Aux Playback Switch", 0, +WSS_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 0, +WSS_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Capture Volume", 0, +WSS_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", - .info = snd_cs4231_info_mux, - .get = snd_cs4231_get_mux, - .put = snd_cs4231_put_mux, + .info = snd_wss_info_mux, + .get = snd_wss_get_mux, + .put = snd_wss_put_mux, } }; -int snd_cs4231_mixer(struct snd_cs4231 *chip) +int snd_wss_mixer(struct snd_wss *chip) { struct snd_card *card; unsigned int idx; @@ -1889,7 +2013,7 @@ int snd_cs4231_mixer(struct snd_cs4231 * strcpy(card->mixername, chip->pcm->name); - if (chip->hardware == CS4231_HW_OPTI93X) + if (chip->hardware == WSS_HW_OPTI93X) for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { err = snd_ctl_add(card, snd_ctl_new1(&snd_opti93x_controls[idx], @@ -1898,48 +2022,29 @@ int snd_cs4231_mixer(struct snd_cs4231 * return err; } else - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) { err = snd_ctl_add(card, - snd_ctl_new1(&snd_cs4231_controls[idx], + snd_ctl_new1(&snd_wss_controls[idx], chip)); if (err < 0) return err; } return 0; } - -EXPORT_SYMBOL(snd_cs4231_out); -EXPORT_SYMBOL(snd_cs4231_in); -EXPORT_SYMBOL(snd_cs4236_ext_out); -EXPORT_SYMBOL(snd_cs4236_ext_in); -EXPORT_SYMBOL(snd_cs4231_mce_up); -EXPORT_SYMBOL(snd_cs4231_mce_down); -EXPORT_SYMBOL(snd_cs4231_overrange); -EXPORT_SYMBOL(snd_cs4231_interrupt); -EXPORT_SYMBOL(snd_cs4231_chip_id); -EXPORT_SYMBOL(snd_cs4231_create); -EXPORT_SYMBOL(snd_cs4231_pcm); -EXPORT_SYMBOL(snd_cs4231_mixer); -EXPORT_SYMBOL(snd_cs4231_timer); -EXPORT_SYMBOL(snd_cs4231_info_single); -EXPORT_SYMBOL(snd_cs4231_get_single); -EXPORT_SYMBOL(snd_cs4231_put_single); -EXPORT_SYMBOL(snd_cs4231_info_double); -EXPORT_SYMBOL(snd_cs4231_get_double); -EXPORT_SYMBOL(snd_cs4231_put_double); +EXPORT_SYMBOL(snd_wss_mixer); /* * INIT part */ -static int __init alsa_cs4231_init(void) +static int __init alsa_wss_init(void) { return 0; } -static void __exit alsa_cs4231_exit(void) +static void __exit alsa_wss_exit(void) { } -module_init(alsa_cs4231_init) -module_exit(alsa_cs4231_exit) +module_init(alsa_wss_init) +module_exit(alsa_wss_exit) ---------------------------------------------------------------------- Masz czas tylko do 10 lipca. Przystap on-line do ING OFE! Sprawdz >> http://link.interia.pl/f1e54

1 0

[alsa-devel] [PATCH] Add better checks for pastebin.ca availability
by Travis Place 01 Jul '08

01 Jul '08

At the moment, it seems that a) pastebin.ca (the host) is UP, and b) the WEB SERVER on it is DOWN. So ive implemented (even if messy) an extra check for this scenario. I would consider using another pastebin service, but we have it pretty good with slepp (the pastebin.ca owner) as he has allowed us to have our own paste 'type' (which enables the areas to collapse, highlighting of certain things, etc). Changelog: Added an extra check to see if the Web Server is functioning on our pastebin service. Now we check that the host is up, and the webserver itself is alive. Patch Attached. Signed-off-by: Travis Place <wishie(a)wishie.net>

2 1

[alsa-devel] Low output from a Dell XPS M1530 with a STAC 9228 codec
by Anders Rønningen 01 Jul '08

01 Jul '08

The output on the mentioned setup is much lower in Ubuntu than in Vista. I have gotten support in #alsa, so I'm fairly sure my alsamixer settings are correct. I have also tried different model options in alsa-base, but without any luck. The volume is much lower both from the internal speakers and from the headphone outputs on the card. The behaviour fits the description at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3785 My alsa-info output is at http://paste.ubuntu.com/24221/ Any idea what can be done to debug/solve this? Someone on a Dell forum suggested that maybe the volume ranges for the STAC9228 card could be wrong. What model-option am I supposed to use for this card? I've seen this info: STAC9227/9228/9229/927x ref Reference board 3stack D965 3stack 5stack D965 5stack + SPDIF dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup but I'm not sure which one fits my setup. -- Anders Rønningen

1 0

Re: [alsa-devel] [ALSA] hda-codec - Fix laptop models for Cxt5045
by Jaroslav Kysela 01 Jul '08

01 Jul '08

On Mon, 30 Jun 2008, Mark T.B. Carroll wrote: > Greetings. I thought I should mention that your patch > http://kerneltrap.org/mailarchive/git-commits-head/2008/2/1/670694 > broke headphone sound for me on a Toshiba Satellite P105-S6197, > sound card PCI ID 8086:27d8 with Conexant CX20549 codec. The speakers > silence when I plug the headphones in but no sound comes from them. > > Details are at http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=488063 > but half of my Debian bug reports go ignored for years. > > A sufficient workaround for me is to comment out the `default:' in the > `switch (board_config)' in the function `patch_cxt5045' in > sound/pci/hda/patch_conexant.c. Then everything works fine again. > > The same problem appears to be reported by evilviper77 at > http://bbs.archlinux.org/viewtopic.php?pid=373770 - they have a > Toshiba P105-9339. CCing to alsa-devel mailing list. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

3 8

[alsa-devel] Audigy bay protocol
by Tony "Chainsaw" Vroon 01 Jul '08

01 Jul '08

Good afternoon, Having spent some time looking at the MIDI that the internal/external Audigy bays emit, I noticed that they report more then is currently used by the LIRC driver. There is one thing missing in here, which is how to control the bay power LED (which is supposed to blink when the card is muted) and the CMSS LED. I am not sure whether this is controlled through MIDI SysEx sequences or through other means. Due to an annoying hardware failure in my Audigy 4 Pro I can not continue this endeavour, but I want to share what I have. Based on this, perhaps one can extend the EMU10K1/2 driver a bit with headphone muting and volume control through the bay. With thanks to the libdouze author, Arnaud de Bossoreille de Ribou. Midi System Exclusive ( extended manufacturer id == 0x2021 ) Payload for 61 bay message: 61 00 01 00 XX YY XX 00 = button push event 01 = volume knob event 02 = jack sense event For XX=00 button push: Bit 1: CMSS Bit 2: Mute For XX=01 volume knob: Bit 1: Always set Bit 7: Set for downward adjustment, unset for upward adjustment For XX=02 jack sense: YY contains jacks Bit 1: headphone Bit 2: digital out (back of bay) Response to initialization: Midi System Exclusive ( extended manufacturer id == 0x2021, payload == | 60 00 02 03 | ) <-- Unknown, bay version? Midi System Exclusive ( extended manufacturer id == 0x2021, payload == | 61 00 01 00 02 01 | ) <-- Jack sense update I hope it is of use to someone. Regards, Tony V.

2 2

Re: [alsa-devel] looping S/PDIF data
by Rene Herman 01 Jul '08

01 Jul '08

On 17-06-08 21:55, Pierre Ossman wrote: > Ping > > On Sat, 31 May 2008 11:26:06 +0200 > Pierre Ossman <drzeus-list(a)drzeus.cx> wrote: > >> I have a minor annoyance with the snd_trident driver that I was hoping >> you might have a fix for: >> >> Pausing the output does not seem to take on the S/PDIF output. Analog >> outputs properly stop in their tracks, but the S/PDIF port keeps >> looping the current data buffer, which gives a rather unpleasant end >> result. :) Try the alsa-devel list... Rene.

4 15

[alsa-devel] [PATCH] [alsa-utils] Fix build with library options disabled
by Sean McNamara 01 Jul '08

01 Jul '08

Hello, It's been a perennial problem with alsa-utils that there is no configure option to disable the seq and amidi apps. These apps depend on how you configured alsa-lib: If you pass --disable-seq --disable-instr to the alsa-lib configure, there does not currently exist a configure-time option to allow alsa-utils to build against this particular alsa-lib. Of course, you could hack makefiles and stuff so it doesn't build seq and amidi, but I thought it would be nice to commit changes to the configure.in and Makefile.am to upstream, so that we can disable seq and amidi the "right" way. The attached patches are diffed against the latest revision of alsa-utils. With this change, you can: 1. Disable building amidi by passing --disable-midi to the configure in alsa-utils. 2. Disable building seq by passing --disable-seq to the configure in alsa-utils. With this change, we can get a "good" build of alsa-utils (albeit, missing two of the utilities, which is intended) in the following circumstances: 1. seq and instr are DISABLED in alsa-lib; seq and midi are DISABLED in alsa-utils 2. seq and instr are ENABLED in alsa-lib; seq and midi are ENABLED in alsa-utils (the default). 3. seq and instr are ENABLED in alsa-lib; seq and midi are DISABLED in alsa-utils. Of course, it does not build cleanly if seq and instr are DISABLED in alsa-lib, but seq and midi are ENABLED in alsa-utils. Patches are attached -- please let me know if there any issues, and if not, could one of the committers please commit them. Thanks, Sean

2 1

[alsa-devel] [PATCH] Add some system information to alsa-info.sh output
by Travis Place 01 Jul '08

01 Jul '08

First some information about this.. I was planning on using dmidecode to fetch this information, but that required 1) the script be run as root, and 2) requires the user to have dmidecode installed (most people wouldnt). So we fetch from /sys/ (if available). Patch attached. Changelog: Fetch the board_vendor, product_version and product_name from DMI in /sys/ (if available). Increased alsa-info.sh version to 0.4.49. Signed-off-by: Travis Place <wishie(a)wishie.net>

3 6

[alsa-devel] [PATCH/RFC 1/2] Add jack reporting API for ALSA
by Mark Brown 01 Jul '08

01 Jul '08

Currently very few systems provide information about jack status to user space, even though many have hardware facilities to do detection. Those systems that do use an input device with the existing SW_HEADPHONE_INSERT switch type to do so, often independently of ALSA. This patch introduces a standard method for representing jacks to user space into ALSA. It allows drivers to register jacks for a sound card with the input subsystem, binding the input device to the card to help user space associate the input devices with their sound cards. The created input devices are named in the form "card longname jack" where jack is provided by the driver when allocating a jack. The existing user space API with SW_HEADPHONE_INSERT is preserved. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/core.h | 1 + include/sound/jack.h | 51 +++++++++++++++++++ sound/core/Kconfig | 5 ++ sound/core/Makefile | 2 + sound/core/jack.c | 136 ++++++++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 195 insertions(+), 0 deletions(-) create mode 100644 include/sound/jack.h create mode 100644 sound/core/jack.c diff --git a/include/sound/core.h b/include/sound/core.h index ee2d7e1..8c87713 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -53,6 +53,7 @@ typedef int __bitwise snd_device_type_t; #define SNDRV_DEV_INFO ((__force snd_device_type_t) 0x1006) #define SNDRV_DEV_BUS ((__force snd_device_type_t) 0x1007) #define SNDRV_DEV_CODEC ((__force snd_device_type_t) 0x1008) +#define SNDRV_DEV_JACK ((__force snd_device_type_t) 0x1009) #define SNDRV_DEV_LOWLEVEL ((__force snd_device_type_t) 0x2000) typedef int __bitwise snd_device_state_t; diff --git a/include/sound/jack.h b/include/sound/jack.h new file mode 100644 index 0000000..e5ac26c --- /dev/null +++ b/include/sound/jack.h @@ -0,0 +1,51 @@ +#ifndef __SOUND_JACK_H +#define __SOUND_JACK_H + +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics plc + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <sound/core.h> + +struct input_dev; + +/** + * Jack types which can be reported. These values are used as a + * bitmask. + */ +enum snd_jack_types { + SND_JACK_HEADPHONE = 0x0001, +}; + +struct snd_jack { + struct input_dev *input_dev; + int registered; + int type; + const char *id; + char name[100]; +}; + +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jack); + +void snd_jack_report(struct snd_jack *jack, int status); + +#endif diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 829ca38..4e8e52d 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -16,6 +16,11 @@ config SND_RAWMIDI tristate depends on SND +config SND_JACK + tristate + depends on SND + depends on INPUT + config SND_SEQUENCER tristate "Sequencer support" depends on SND diff --git a/sound/core/Makefile b/sound/core/Makefile index 5a01c76..e1cb6d6 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -20,9 +20,11 @@ snd-rawmidi-objs := rawmidi.o snd-timer-objs := timer.o snd-rtctimer-objs := rtctimer.o snd-hwdep-objs := hwdep.o +snd-jack-objs := jack.o obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o +obj-$(CONFIG_SND_JACK) += snd-jack.o obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o snd-page-alloc.o diff --git a/sound/core/jack.c b/sound/core/jack.c new file mode 100644 index 0000000..1a52997 --- /dev/null +++ b/sound/core/jack.c @@ -0,0 +1,136 @@ +/* + * Jack abstraction layer + * + * Copyright 2008 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/input.h> +#include <sound/jack.h> +#include <sound/core.h> + +static int snd_jack_dev_free(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + + /* If the input device is registered with the input subsystem + * then we need to use a different deallocator. */ + if (jack->registered) + input_unregister_device(jack->input_dev); + else + input_free_device(jack->input_dev); + + kfree(jack); + + return 0; +} + +static int snd_jack_dev_register(struct snd_device *device) +{ + struct snd_jack *jack = device->device_data; + struct snd_card *card = device->card; + int err; + + snprintf(jack->name, sizeof(jack->name), "%s %s", + card->longname, jack->id); + jack->input_dev->name = jack->name; + + err = input_register_device(jack->input_dev); + if (err == 0) + jack->registered = 1; + + return err; +} + +/** + * snd_jack_new - Create a new jack + * @card: the card instance + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jjack: Used to provide the allocated jack object to the caller. + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jjack will be initialised. + */ +int snd_jack_new(struct snd_card *card, const char *id, int type, + struct snd_jack **jjack) +{ + struct snd_jack *jack; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_jack_dev_free, + .dev_register = snd_jack_dev_register, + }; + + jack = kzalloc(sizeof(struct snd_jack), GFP_KERNEL); + if (jack == NULL) + return -ENOMEM; + + jack->id = id; + + jack->input_dev = input_allocate_device(); + if (jack->input_dev == NULL) { + err = -ENOMEM; + goto fail_input; + } + + jack->input_dev->phys = "ALSA"; + jack->input_dev->dev.parent = card->dev; + + jack->type = type; + + if (type & SND_JACK_HEADPHONE) + input_set_capability(jack->input_dev, EV_SW, + SW_HEADPHONE_INSERT); + + err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); + if (err < 0) + goto fail_input; + + *jjack = jack; + + return 0; + +fail_input: + input_free_device(jack->input_dev); + kfree(jack); + return err; +} +EXPORT_SYMBOL(snd_jack_new); + +/** + * snd_jack_report - Report the current status of a jack + * + * @jack: The jack to report status for + * @status: The current status of the jack + */ +void snd_jack_report(struct snd_jack *jack, int status) +{ + if (jack->type & SND_JACK_HEADPHONE) + input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, + status & SND_JACK_HEADPHONE); + + input_sync(jack->input_dev); +} +EXPORT_SYMBOL(snd_jack_report); + +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("Jack detection support for ALSA"); +MODULE_LICENSE("GPL"); -- 1.5.5.1

3 12

[alsa-devel] [PATCH] autoconf problems
by ben.stanley＠exemail.com.au 01 Jul '08

01 Jul '08

Dear List, A month ago I was having trouble building alsa-drivers on RHEL5. The following patch fixes the problems. The root of the problem is that including the header include/linux/config.h produces a warning on RHEL5. This interferes with several autoconf tests that treat warnings as errors. Therefore it seemed best to globally check use of #include <linux/config.h> in the same pattern as was already done for init_utsname. Furthermore, the check for is_power_of_2 would produce 2 warnings: one for the use of an undeclared function (desired), and another warning for an un-used variable. Still the test would succeed, because the required condition is flagged by warnings, not errors. This was fixed by following the pattern from init_utsname. Signed-off-by: Ben Stanley <Ben.Stanley(a)exemail.com.au> --- a/configure.in 2008-07-01 13:09:02.000000000 +1000 +++ b/configure.in 2008-07-01 13:26:12.000000000 +1000 @@ -1567,7 +1567,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/kernel.h> #include <linux/string.h> ],[ @@ -1595,7 +1599,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/kernel.h> ],[ char buf[128]; @@ -1622,7 +1630,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/kernel.h> void f(const char *fmt, ...) { @@ -1656,7 +1668,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/kernel.h> ],[ char buf[128]; @@ -1683,7 +1699,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/kernel.h> ],[ unsigned int i; @@ -1710,7 +1730,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/time.h> #include <linux/sched.h> #include <linux/interrupt.h> @@ -1740,7 +1764,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/kmod.h> ],[ request_module("abcd", "def", "ixj"); @@ -1766,7 +1794,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/fs.h> #include <linux/proc_fs.h> ],[ @@ -1794,7 +1826,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/pci.h> ],[ int (*func)(); @@ -1820,7 +1856,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/pci.h> ],[ int (*func)(); @@ -1846,7 +1886,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/delay.h> ],[ void (*func)(unsigned int); @@ -1874,7 +1918,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/delay.h> ],[ unsigned long (*func)(unsigned int); @@ -1902,7 +1950,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/jiffies.h> #include <linux/time.h> ],[ @@ -1930,7 +1982,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/tty.h> ],[ struct tty_struct tty; @@ -1957,7 +2013,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/videodev.h> #include "media/v4l2-dev.h" ],[ @@ -1995,7 +2055,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/mm.h> ],[ static pgprot_t p; @@ -2020,7 +2084,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/mm.h> ],[ static pgprot_t p; @@ -2046,7 +2114,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/slab.h> ],[ void * (*func)(size_t, size_t, int); @@ -2073,7 +2145,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/string.h> ],[ char * (*func)(const char *, int); @@ -2100,7 +2176,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/slab.h> ],[ void * (*func)(size_t, int); @@ -2128,7 +2208,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/workqueue.h> ],[ create_workqueue("abcd", 0); @@ -2155,7 +2239,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/pci.h> ],[ struct pci_dev dev; @@ -2183,7 +2271,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/pci.h> ],[ struct pci_dev *pci; @@ -2209,7 +2301,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/sound.h> ],[ void (*func)(struct file_operations *, int, struct device *); @@ -2453,7 +2549,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/device.h> ],[ struct class_simple *(*foo)(struct module *, char *) = class_simple_create; @@ -2482,7 +2582,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/device.h> ],[ static struct device_driver driver; @@ -2511,7 +2615,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/mm.h> ],[ void (*func)(struct vm_area_struct *, pte_t, unsigned long) = print_bad_pte; @@ -2539,7 +2647,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/device.h> ],[ struct class_device dev; @@ -2568,7 +2680,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/interrupt.h> ],[ irq_handler_t handle = 0; @@ -2596,7 +2712,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/types.h> #include <linux/gfp.h> ],[ @@ -2623,7 +2743,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/types.h> #include <linux/gfp.h> ],[ @@ -2650,7 +2774,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/types.h> #include <linux/mm.h> ],[ @@ -2716,14 +2844,19 @@ is_power_of_2="0" ac_save_CFLAGS="$CFLAGS" ac_save_CC="$CC" - CFLAGS="$KERNEL_CHECK_CFLAGS" + CFLAGS="$KERNEL_CHECK_CFLAGS -Werror" CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/log2.h> ],[ int a = is_power_of_2(2); + a = 2; ], AC_MSG_RESULT(yes);is_power_of_2="1", AC_MSG_RESULT(no);is_power_of_2="0", @@ -2750,7 +2883,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/bitops.h> ],[ unsigned long (*x)(unsigned long); @@ -2779,7 +2916,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/pnp.h> ],[ struct pnp_driver pnp; @@ -2808,7 +2949,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/device.h> ],[ struct device *(*foo)() = device_create_drvdata; @@ -2836,7 +2981,11 @@ CC=$KCC AC_TRY_COMPILE([ #define __KERNEL__ +#ifdef CONFIG_HAVE_DEPRECATED_CONFIG_H +#include <linux/autoconf.h> +#else #include <linux/config.h> +#endif #include <linux/fs.h> ],[ struct file_operations ops; @@ -3005,4 +3154,4 @@ dnl avoid changing include/sound/version.h if test "$update_versoin" != "yes"; then touch alsa-kernel/include/version.h -fi \ No newline at end of file +fi

2 1

[alsa-devel] ALSA with Ubuntu 8.04
by Eliot Blennerhassett 28 Jun '08

28 Jun '08

This just an FYI In Ubuntu 8.04, ubuntu has packaged alsa-drivers including some out of tree drivers in a separate package: linux-ubuntu-modules-$kernelver This gets installed in /lib/modules/$kernelver/ubuntu/sound The only things installed in /lib/modules/$kernelver/sound are soundcore.ko and ac97_bus.ko However, when you do a default "make install" from alsa-driver source, the modules get installed in /lib/modules/$kernelver/kernel/sound So, now there are 2 copies of the drivers in different places. I have taken to removing (moving away to say /root) /lib/modules/$kernelver/ubuntu/sound to avoid any confusion and some odd module versioning problems that I was having. regards Eliot.

4 3

[alsa-devel] softvol chokes on snd_pcm_rewind
by Lennart Poettering 28 Jun '08

28 Jun '08

Hey! On 1.0.16 softvol sometimes chokes on snd_pcm_rewind(): the command suceeds but the sound is heavily distorted from then on. Takashi, Jaroslav, is this known? Any short-term workaround? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] emu20k1 in 1.0.17?
by daedra 28 Jun '08

28 Jun '08

I was just wondering if there is a chance of the emu20k1 driver making it's way into 1.0.17?

3 3

[alsa-devel] seg fault with 1.0.17rc2
by Jerry Geis 27 Jun '08

27 Jun '08

#0 0xb7e892ff in memcpy () from /lib/tls/libc.so.6 #1 0xb74e40a9 in snd_pcm_area_copy (dst_area=0xb7125b70, dst_offset=0, src_area=0x81dc1c0, src_offset=170, samples=0, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2589 #2 0xb74e438d in snd_pcm_areas_copy (dst_areas=0xb7125b7c, dst_offset=0, src_areas=0x81dc1cc, src_offset=170, channels=1, frames=122, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2736 #3 0xb74ee0a1 in snd_pcm_mmap_read_areas (pcm=0x81de3b8, areas=0xb7125b70, offset=0, size=122) at pcm_mmap.c:151 #4 0xb74e350d in snd1_pcm_read_areas (pcm=0x81de3b8, areas=0xb7125b70, offset=0, size=122, func=0xb74edfd0 <snd_pcm_mmap_read_areas>) at pcm.c:6501 #5 0xb74ed2df in snd_pcm_mmap_readi (pcm=0x81de3b8, buffer=0xb6352a04, size=122) at pcm_mmap.c:236 #6 0xb74dbb14 in snd_pcm_readi (pcm=0x81de110, buffer=0xb6352a04, size=61) at pcm_local.h:526 #7 0xb756279b in alsa_read (chan=0x821c928) at chan_alsa.c:683 #8 0x080855b1 in __ast_read (chan=0x821c928, dropaudio=0) at channel.c:2100 I am using a usb-audio device. Bus 003 Device 001: ID 0000:0000 Bus 003 Device 002: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter Bus 002 Device 002: ID 13fe:1f00 Bus 002 Device 001: ID 0000:0000 Bus 004 Device 001: ID 0000:0000 Bus 001 Device 001: ID 0000:0000 I am using this as my asound.conf file pcm.usb-audio { type hw card 0 } ctl.usb-audio { type hw card 0 } I am using debian etch 2.6.18-6-486 #1 Fri Jun 6 21:47:01 UTC 2008 i486 GNU/Linux It works for a couple times then I get the seg fault. I am using the Console output from asterisk. help? Thanks, Jerry

2 9

[alsa-devel] ALSA 1.0.17rc3 release (driver only)
by Jaroslav Kysela 27 Jun '08

27 Jun '08

Hello all, ALSA driver package v1.0.17rc3 is released. I hope that all remaining bugs for compilation with 2.4 kernels are fixed. Changes are available here: http://www.alsa-project.org/main/index.php/Changes_v1.0.17rc2_v1.0.17rc3 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

Re: [alsa-devel] [alsa-cvslog] alsa-kmirror: ALSA kernel mirror repository branch, master now at 9d46f4a919532c3f29a4ca1df3a4ce4686b11f37
by Thierry Vignaud 27 Jun '08

27 Jun '08

noreply-git(a)alsa-project.org writes: > Hello, > > This is an automated email from the git hooks/update script, it was > generated because a ref change was pushed to the repository. > > Updating branch, master, > via 9d46f4a919532c3f29a4ca1df3a4ce4686b11f37 (commit) > from fc5102d6c6520f48d39faf572881bc520c41aeb7 (commit) > > - Log ----------------------------------------------------------------- > commit 9d46f4a919532c3f29a4ca1df3a4ce4686b11f37 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Tue May 20 13:25:35 2008 +0200 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Tue May 20 13:25:35 2008 +0200 > > removed .hgtags can we make the log appears in subject instead of useless sha1 ID?

5 16

[alsa-devel] Building trouble - redefinition of struct timeval
by Caloro Maurizio 27 Jun '08

27 Jun '08

Please i need help for following error message "redefinition of struct timeval - /include/global.h" the configure script run without any Error, if i compile "alsa-lib-1.0.17rc2" on I386 Netbsd 4.0 i receive folowing error break. "./configure --prefix=/usr --sysconfdir=/etc --build=i386-unknown-netbsdelf4.0 --host=i386-unknown-netbsdelf4.0 --target=i386-unknown-netbsdelf4.0 --with-pythonlibs=lpython2.5 --with-pythonincludes=-I/usr/include/python2.5 --enable-shared=no --enable-static=yes" & gmake gmake ... > Making all in controll gmake[1]: Entering directory `/usr/source/alsa-lib-1.0.17rc2/src/control' if /bin/ksh ../../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I../../include -I../../include -g -O2 -MT cards.lo -MD -MP -MF ".deps/cards.Tpo" -c -o cards.lo cards.c; \ then mv -f ".deps/cards.Tpo" ".deps/cards.Plo"; else rm -f ".deps/cards.Tpo"; exit 1; fi gcc -DHAVE_CONFIG_H -I. -I. -I../../include -I../../include -g -O2 -MT cards.lo -MD -MP -MF .deps/cards.Tpo -c cards.c -o cards.o In file included from ../../include/local.h:109, from control_local.h:22, from cards.c:35: ../../include/global.h:138: error: redefinition of 'struct timeval' ../../include/global.h:143: error: redefinition of 'struct timespec' gmake[1]: *** [cards.lo] Error 1 gmake[1]: Leaving directory `/usr/source/alsa-lib-1.0.17rc2/src/control' gmake: *** [all-recursive] Error 1 i have see that this its a possible handle with Posix_C_Source but i dont found the correct answer for me trouble.... Please for any Help i'am happy. thanks MC -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

2 5

[alsa-devel] [alsa-kernel][PATCH] ASoC: TLV320AIC3X: Add mixer control for ADC highpass filter
by Jarkko Nikula 27 Jun '08

27 Jun '08

Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ sound/soc/codecs/tlv320aic3x.h | 2 ++ 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4f0bf26..d138306 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -223,6 +223,8 @@ static const char *aic3x_right_hpcom_mux[] = { "differential of HPROUT", "constant VCM", "single-ended", "differential of HPLCOM", "external feedback" }; static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; +static const char *aic3x_adc_hpf[] = + { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -232,6 +234,7 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; #define LINE1R_ENUM 5 #define LINE2L_ENUM 6 #define LINE2R_ENUM 7 +#define ADC_HPF_ENUM 8 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -242,6 +245,7 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), + SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; static const struct snd_kcontrol_new aic3x_snd_controls[] = { @@ -292,6 +296,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Input */ SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), + + SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index c1dd1ac..e600946 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -37,6 +37,8 @@ #define AIC3X_ASD_INTF_CTRLB 9 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 +/* Audio codec digital filter control register */ +#define AIC3X_CODEC_DFILT_CTRL 12 /* ADC PGA Gain control registers */ #define LADC_VOL 15 -- 1.5.5.4

2 1

[alsa-devel] [PATCH] Add another VIA audio chip to ac97 quirks list
by walt 27 Jun '08

27 Jun '08

Author: Walter Sheets <w41ter(a)gmail.com> Date: Thu Jun 26 10:39:19 PDT 2008 x86: Add VIA audio device #1841 to ac97_quirk list Signed-off-by: Walter Sheets <w41ter(a)gmail.com> --- Takashi, feel free to modify this in any way you need to. Thanks. diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index b585cc3..aa5c2e0 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1746,6 +1746,12 @@ static struct ac97_quirk ac97_quirks[] = { }, { .subvendor = 0x1019, + .subdevice = 0x1841, + .name = "ECS K7VTA3", + .type = AC97_TUNE_HP_ONLY + }, + { + .subvendor = 0x1019, .subdevice = 0x0a81, .name = "ECS K7VTA3", .type = AC97_TUNE_HP_ONLY

2 1

[alsa-devel] sw parameter boundary
by Norbert van Bolhuis 26 Jun '08

26 Jun '08

is the boundary sw parameter still used ? where is it for ? in the PCM kernel core (e.g. snd_pcm_playback_avail) it clearly serves as a pointers wrap point, but which one ? The one at the end of the DMA buffer or the one at the end of a process (virtual) address range (4 GB) ? -- This message has been scanned for viruses and is believed to be clean

2 4

[alsa-devel] ECS K7VTA3: Mixer channels reversed
by walt 26 Jun '08

26 Jun '08

Hi list, I just replaced a mobo with an ECS K7VTA3 and I find that the Headphones control is now controlling the master channel and the Master control does nothing. I tried 'modprobe via82xx ac97_quirk=hp_only' but that makes the audio device unusable (although alsamixer still seems happy to set the individual channels). Any other quirks I should try? Here is some hardware info: ALSA sound/pci/via82xx.c:2409: via82xx: DXS white list for ESC K8 found 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) Subsystem: Elitegroup Computer Systems Unknown device 1841 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin C routed to IRQ 5 Region 0: I/O ports at dc00 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: VIA 82xx Audio Kernel modules: snd-via82xx I'm running Linus' git kernel, current as of today, and alsa 1.0.16.

2 5

[alsa-devel] seg fault in libasound
by Jerry Geis 26 Jun '08

26 Jun '08

I am running asterisk on debian Etch , i486 with math emulation in kernel. Verified by /boot/config-file. I am getting a seg fault in alsa. First I ran straight debian with alsa install. I got the same seg fault. Second I installed 1.0.16 and I still get the seg fault. Third I installed 1.0.17rc2 and I still get the seg fault. Posted to users first and stan suggested I post to devel. Below is the trace. I am only using the Console/Dsp from asterisk at this point. Which is the interface to the soundcard and alsa. What shall I do? Jerry -------------------------- Program received signal SIGABRT, Aborted. [Switching to Thread -1224033360 (LWP 2440)] 0xb7d77947 in raise () from /lib/tls/libc.so.6 (gdb) where #0 0xb7d77947 in raise () from /lib/tls/libc.so.6 #1 0xb7d790c9 in abort () from /lib/tls/libc.so.6 #2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6 #3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0, src_area=0x81cb7ec, src_offset=5223, samples=816, format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499 #4 0xb744b7b7 in snd_pcm_dsnoop_sync_ptr (pcm=0x81cb930) at pcm_dsnoop.c:77 #5 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cb930) at pcm.c:932 #6 0xb7422b55 in snd1_pcm_generic_hwsync (pcm=0x81cc1c8) at pcm_generic.c:143 #7 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc1c8) at pcm.c:932 #8 0xb742fb4d in snd_pcm_rate_hwsync (pcm=0x81cc3f8) at pcm_rate.c:624 #9 0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc3f8) at pcm.c:932 #10 0xb7417817 in snd1_pcm_read_areas (pcm=0x81cc3f8, areas=0xb70a5b70, offset=0, size=160, func=0xb7421e10 <snd_pcm_mmap_read_areas>) at pcm.c:6376 #11 0xb742111f in snd_pcm_mmap_readi (pcm=0x81cc3f8, buffer=0xb74a5a40, size=160) at pcm_mmap.c:236 #12 0xb740fff4 in snd_pcm_readi (pcm=0x81cc038, buffer=0xb74a5a40, size=2440) at pcm_local.h:521 #13 0xb749579b in alsa_read (chan=0x8213f18) at chan_alsa.c:683 #14 0x080855b1 in __ast_read (chan=0x8213f18, dropaudio=0) at channel.c:2100 #15 0x08087725 in ast_channel_bridge (c0=0x8212858, c1=0x817cf68, config=0xb70a6eac, fo=0xb70a5fe8, rc=0xb70a5fe4) at channel.c:2380 #16 0xb7c75592 in ast_bridge_call (chan=0x8212858, peer=0x8213f18, config=0xb70a6eac) at res_features.c:1490 #17 0xb75c38fd in dial_exec_full (chan=0x8212858, data=0xb70a8fd8, peerflags=0xb70a6f84, continue_exec=0x0) at app_dial.c:1773 #18 0xb75c519d in dial_exec (chan=0x8212858, data=0xb70a8fd8) at app_dial.c:1827 #19 0x080c6e3f in pbx_extension_helper (c=0x8212858, con=0x0, context=0x82129d8 "smvoice", exten=0x8212a28 "s", priority=3, label=0x0, callerid=0x8215450 "404", action=E_SPAWN) at pbx.c:537 #20 0x080c9185 in __ast_pbx_run (c=0x8212858) at pbx.c:2317 #21 0x080c9ebe in pbx_thread (data=0x8212858) at pbx.c:2636 #22 0x080f51d2 in dummy_start (data=0x8213178) at utils.c:909 #23 0xb7eff0bd in start_thread () from /lib/tls/libpthread.so.0 #24 0xb7e1b01e in clone () from /lib/tls/libc.so.6

2 4

[alsa-devel] [asoc-dev][RFC/PATCH 0/2] Generic DAPM register modifier widget
by Jarkko Nikula 26 Jun '08

26 Jun '08

Hi At least tlv320aic3x codec has several use cases where multiple register bits are needed to change when widget power state changes and where these are not worth/safe to expose into user space. Like mic bias voltage. Currently there is one use case for this multiple bit change in tlv320aic3x: SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 6, 0), SND_SOC_DAPM_MICBIAS("Mic Bias AVDD", MICBIAS_CTRL, 7, 0), There are two problems with that: 1. Two widgets having the same name and thus snd_soc_dapm_add_route will add only first one them into audio path 2. Even then bits aren't modified at once and thus some fault bit configuration might be present while power state is changing. So I came with this idea of generic register modifiers widget and event handler for it. -- Jarkko

2 8

[alsa-devel] No sound on MacBook 3.1
by Marco Vermeulen 26 Jun '08

26 Jun '08

Hi there, I hope that someone can help me, as I've tried everything conceivable to get sound working on my Macbook 3.1. I've visited many a forum, and eventually came across this list. My last hope :) I purchased a Macbook 3.1 earlier this year. On installing Linux, I haven't been able to get any sound going. I've tried several distributions: Ubuntu Hardy, Fedora 8/9, and now openSuse. I came across a script that uploads local relevant information to a paste bin, so here's a link: http://pastebin.ca/1055610 Please help, I'm desperate to get this going on my machine! Thanks & Regards, Marco.

4 7

[alsa-devel] support for RME HDSPe RayDAT
by Matthijs ten Berge 26 Jun '08

26 Jun '08

Hello all, <intro:> As you might already know, RME has released a successor for the HDSP 9652, the HDSPe RayDAT (http://www.rme-audio.de/en_products_hdspe_raydat.php). It is based on the 9652's design, with some new/changed features (e.g. more channels and lower latencies), but the most important hardware change is that it is now a PCI-express card. I'm currently in the process of choosing which audio interface to buy, the 'old' HDSP 9652 or the new RayDAT (I haven't found any other card with 3 or more ADAT in/outs in the same price range or below, but suggestions are welcome). Of course I can stay on the safe side and opt for the HDSP 9652, which has stable and complete driver support (right, eh?). But on the other hand the RayDAT has better features and future, but without the guarantee that it is or will be supported it can only serve as a very costly paper weight (and not even a good one I suppose). I wrote an e-mail to RME, but a bit too early, the card was not released at that moment. See below for their reply. <question:> Are there any plans to add support for the RME RayDAT to the HDSP driver/tools? Or perhaps somebody is already working on it right now? It is not mentioned in the Wiki matrix atm. I wish I could just do it myself, but (concerning kernel programming) I never did more than a 'hello world' kernel module, so I don't think this would be a good idea... (I do have coding experience, but not in-kernel).

2 3

Re: [alsa-devel] Alsa Issue - Sound playback starts on left speaker only
by Takashi Iwai 26 Jun '08

26 Jun '08

At Thu, 26 Jun 2008 14:43:30 +0800, <Ashwin.Chandra(a)corel.com> wrote: > > Hello, > I have found an issue with the mixer devices. I had been controlling > only one elem, "LineOut", but there is another elem named "iSpeaker" > which is the one that shows only left speaker active in alsamixer. I am > not sure why I would need to control both of them. I would expect > "Lineout" to be the master control. But anyway I can fix the problem in > my application by un setting mute on the right channel at the beginning > of the application. Good to hear that you found out the culprit. > Do you have any idea about this though? (or have seen this before)? There have been similar bug reports, but your finding isn't unknown (and thus important). Could you run alsa-info.sh with --no-upload option for both working and non-working cases, and attach the generated files? The script is found at http://www.alsa-project.org/alsa-info.sh thanks, Takashi > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, June 25, 2008 11:26 PM > To: Ashwin Chandra > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > speaker only > > At Thu, 26 Jun 2008 03:28:42 +0800, > <Ashwin.Chandra(a)corel.com> wrote: > > > > Hi Takashi, > > Actually now I am noticing the problem (left speaker only) with > > mplayer/smplayer in their OSS emulation as well. It seems to happen > once > > I run my player first, then after that the other players exhibit the > > problem. > > In this case, mplayer will choose the ALSA mode automatically, I > guess. So the problem appears to be ALSA-mode specific, possibly > about dmix. > > Try to define the following in ~/.asoundrc: > pcm.!default "plughw" > > then it won't use dmix as default any more. (Then you can use only > one app at the same time, of course.) Check whether the problem can > be reproduced with this setting. > > > Takashi > > > > Sometimes after muting and unmuting my player, the problem goes > > away and also away on the players. It is almost like my player is > > causing bad internal state for the audio driver. Do you have any > > suggestions to work around this? > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Wednesday, June 25, 2008 3:23 AM > > To: Ashwin Chandra > > Cc: alsa-devel(a)alsa-project.org > > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > > speaker only > > > > At Wed, 25 Jun 2008 01:52:16 +0800, > > <Ashwin.Chandra(a)corel.com> wrote: > > > > > > I have an Intel ALC861VD. Other players (mplayer, smplayer, aplay) > > don't > > > seem to exhibit this problem. Is there any workaround I can do in my > > > application so that upon first time sound playback, I can get it to > > play > > > on both channels? > > > > Check /proc/asound/card0/pcm0p/sub0/* files and compare between > > working and non-working states. I guess the working case uses OSS > > emulation or 16bit format. > > > > If so, try to add the following to ~/.asoundrc: > > > > defaults.pcm.dmix.format S16_LE > > > > > > Takashi > > > > > > > > -----Original Message----- > > > From: stan [mailto:ghjeold_i_mwee@cox.net] > > > Sent: Tuesday, June 24, 2008 1:25 AM > > > To: Ashwin Chandra > > > Cc: alsa-devel(a)alsa-project.org > > > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > > > speaker only > > > > > > Ashwin.Chandra(a)corel.com wrote: > > > > > > > > > > > > Hello, > > > > > > > > I am having an issue with my playback application where upon > > playback > > > > start, only the left speaker is outputted. But if I mute and > unmute > > > > (either through my application or using alsamixer), the sound > > playback > > > > resumes on both left and right speakers. This only happens on a > > > > particular system too (out of 5 systems). But I know it is my > issue > > > > because mplayer or smplayer works fine. > > > > > > > > > > > > > > > > I believe I am setting the correct HW params (ie. 2 channels), > etc. > > > > > > > > > > > > > > > > Any ideas? > > > > > > > > > > > > > > > > Thanks! > > > > > > > > > > > The ICE1724 driver has exactly this behavior. It is a known bug and > > has > > > > > > been there for years. > > > You don't say what chip is having the problem, but I would guess > that > > > this is it. Most people > > > just program a script to adjust the volume at startup to cure it. > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > >

1 0

[alsa-devel] ALSA 1.0.17rc2 release
by Jaroslav Kysela 26 Jun '08

26 Jun '08

Hello all, ALSA 1.0.17rc2 packages (except tools, oss, python - no changes) were released and are available for download on ALSA server. Changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.17rc1_v1.0.17rc2 The alsa-driver compilation with 2.4 kernels should be fixed in this release. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

5 21

Re: [alsa-devel] Alsa Issue - Sound playback starts on left speaker only
by Takashi Iwai 26 Jun '08

26 Jun '08

At Thu, 26 Jun 2008 03:28:42 +0800, <Ashwin.Chandra(a)corel.com> wrote: > > Hi Takashi, > Actually now I am noticing the problem (left speaker only) with > mplayer/smplayer in their OSS emulation as well. It seems to happen once > I run my player first, then after that the other players exhibit the > problem. In this case, mplayer will choose the ALSA mode automatically, I guess. So the problem appears to be ALSA-mode specific, possibly about dmix. Try to define the following in ~/.asoundrc: pcm.!default "plughw" then it won't use dmix as default any more. (Then you can use only one app at the same time, of course.) Check whether the problem can be reproduced with this setting. Takashi > Sometimes after muting and unmuting my player, the problem goes > away and also away on the players. It is almost like my player is > causing bad internal state for the audio driver. Do you have any > suggestions to work around this? > > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, June 25, 2008 3:23 AM > To: Ashwin Chandra > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > speaker only > > At Wed, 25 Jun 2008 01:52:16 +0800, > <Ashwin.Chandra(a)corel.com> wrote: > > > > I have an Intel ALC861VD. Other players (mplayer, smplayer, aplay) > don't > > seem to exhibit this problem. Is there any workaround I can do in my > > application so that upon first time sound playback, I can get it to > play > > on both channels? > > Check /proc/asound/card0/pcm0p/sub0/* files and compare between > working and non-working states. I guess the working case uses OSS > emulation or 16bit format. > > If so, try to add the following to ~/.asoundrc: > > defaults.pcm.dmix.format S16_LE > > > Takashi > > > > > -----Original Message----- > > From: stan [mailto:ghjeold_i_mwee@cox.net] > > Sent: Tuesday, June 24, 2008 1:25 AM > > To: Ashwin Chandra > > Cc: alsa-devel(a)alsa-project.org > > Subject: Re: [alsa-devel] Alsa Issue - Sound playback starts on left > > speaker only > > > > Ashwin.Chandra(a)corel.com wrote: > > > > > > > > > Hello, > > > > > > I am having an issue with my playback application where upon > playback > > > start, only the left speaker is outputted. But if I mute and unmute > > > (either through my application or using alsamixer), the sound > playback > > > resumes on both left and right speakers. This only happens on a > > > particular system too (out of 5 systems). But I know it is my issue > > > because mplayer or smplayer works fine. > > > > > > > > > > > > I believe I am setting the correct HW params (ie. 2 channels), etc. > > > > > > > > > > > > Any ideas? > > > > > > > > > > > > Thanks! > > > > > > > > The ICE1724 driver has exactly this behavior. It is a known bug and > has > > > > been there for years. > > You don't say what chip is having the problem, but I would guess that > > this is it. Most people > > just program a script to adjust the volume at startup to cure it. > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > >

1 0

[alsa-devel] Alsa Issue - Sound playback starts on left speaker only
by Ashwin.Chandra＠corel.com 26 Jun '08

26 Jun '08

Hello, I am having an issue with my playback application where upon playback start, only the left speaker is outputted. But if I mute and unmute (either through my application or using alsamixer), the sound playback resumes on both left and right speakers. This only happens on a particular system too (out of 5 systems). But I know it is my issue because mplayer or smplayer works fine. I believe I am setting the correct HW params (ie. 2 channels), etc. Any ideas? Thanks!

4 9

[alsa-devel] IBM X41 laptop, no headphone detection after s2ram cycle
by Dmitry E. Mikhailov 25 Jun '08

25 Jun '08

Hi, For about a year I got a following problem. After s2ram cycle I got only sound from internal speaker, even if the headphones are plugged in. s2disk (hibernate) restores the functionality. Sometimes (rare!) I got the functionality after s2ram. If I repeat s2ram, it stops working again. I compared ALSA registers before and after: they are all the same. It was Ok when I got Fedora7 with kernel 2.6.18 installed. But even if I restore 2.6.18 now, things don't go right. Before the Switch from 'another proprietary OS' it was also Ok. With Fedora9 I got a famous 'alsasound' service. It's restart doesn't help. Reloading modules doesn't help. Unload prior to s2ram and load after it doesn't help. Rolling ALSA back doesn't seem to help but I could do it wrong way. Tested kernels: 2.6.18.something, 2.6.19, 2.6.20, 2.6.20.4, 2.6.22, 2.6.22.something, 2.6.23, 2.6.23.6, 2.6.23.12, 2.6.23.12, 2.6.25, 2.6.25.3, 2.6.25.14 (current). Or it's a hardware failure? As a last resort I'll reinstall /\/\$ \/\/in to check. Best regards, Dmitry Mikhailov

1 0

[alsa-devel] A few general questions (my first driver): Marian MARC4
by Joshua Boyd 25 Jun '08

25 Jun '08

I have a Marian Marc4 card. I have the documentation for this card, which has been on their web site for quite some time. I want to write a driver for this card. This is my first ALSA driver, but it isn't my first kernel driver. Anyway, I can't google any mention of this family of cards on this mailing list, which seems a bit odd. I did find some mentions on linux-audio-developer, about the documentation, but no mention there of prior work on this family. However, I did find a MARC8 driver on sourceforge (http:// sourceforge.net/projects/marc8-driver). Does anyone know anything about this? I'm vary curious about why I've never seen it mentioned here before. The license is listed as GPL. The Marian Marc8 was formerly known as the Sek'd Siena, but again I can't find any real mention of that in the archives. Does anyone have any idea why the MARC8 wasn't mentioned and merged into the main ALSA sources? I did go ahead and add some Marian product information to the support matrix on the alsa wiki, including a link to the sourceforge page. My plan is to add MARC4 to the MARC8 driver. The register layout is basically the same, adjusted for the 4 versus 8 channels. Both cards use the same PLX chip. The biggest difference is in the layout of the play/capture RAM area, which is because of the 4 versus 8 channels. It seems silly to make two separate drivers. I would like to see my results eventually merged into the main ALSA. I'm hoping that if I get these into the main tree that I would then be able to get some support from Marian in the form of clearance (and a card) to do the Trace D4 and D4 SRC, and possibly even documentation and hardware for the UCON CX. Anyway, moving on to my specific questions. First, is it appropriate to put support for two device IDs in the same file? The examples I tried looking at ended up being a single device id, differentiated by a subvendor ID. Second, is it wrong to use the same structures, but with an enum field in the main snd_marc device structure that points out which card I'm talking about, then else where switching on that enum to figure out how many channels I need to operate on? Third, once I get this functional on my MARC4 card, how hard is it likely to be to get it accepted? I don't expect low standards for code that is accepted, but I'd like to have some idea how picky the process may be.

2 1

[alsa-devel] [asoc-dev][RFC/PATCH 0/3] Add digital mic support to TLV320AIC3X and N810
by Jarkko Nikula 25 Jun '08

25 Jun '08

Hi These patches are built on top of my recent generic register modifier widget patches: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008778.html Basically AIC3x will capture data either from its ADC converter or from external digital mic. Therefore I named this input selection in n810.c as "Input Select" == "ADC" | "Digital Mic". -- Jarkko

2 4

[alsa-devel] Problems compiling Davinci ASoC driver in 2.6.26-rc6
by Felix Radensky 25 Jun '08

25 Jun '08

Hi, I wanted to test Davinci ASoC driver on Davinci EVM evaluation board, but encountered the following compilation errors: CC sound/soc/codecs/tlv320aic3x.o sound/soc/codecs/tlv320aic3x.c:1012: warning: 'aic3x_init' defined but not used LD sound/soc/codecs/snd-soc-tlv320aic3x.o LD sound/soc/codecs/built-in.o CC sound/soc/davinci/davinci-i2s.o sound/soc/davinci/davinci-i2s.c:402: warning: initialization from incompatible pointer type CC sound/soc/davinci/davinci-pcm.o sound/soc/davinci/davinci-pcm.c: In function `davinci_pcm_enqueue_dma': sound/soc/davinci/davinci-pcm.c:99: error: implicit declaration of function `davinci_set_dma_src_params' sound/soc/davinci/davinci-pcm.c:99: error: `INCR' undeclared (first use in this function) sound/soc/davinci/davinci-pcm.c:99: error: (Each undeclared identifier is reported only once sound/soc/davinci/davinci-pcm.c:99: error: for each function it appears in.) sound/soc/davinci/davinci-pcm.c:99: error: `W8BIT' undeclared (first use in this function) sound/soc/davinci/davinci-pcm.c:100: error: implicit declaration of function `davinci_set_dma_dest_params' sound/soc/davinci/davinci-pcm.c:101: error: implicit declaration of function `davinci_set_dma_src_index' sound/soc/davinci/davinci-pcm.c:102: error: implicit declaration of function `davinci_set_dma_dest_index' sound/soc/davinci/davinci-pcm.c:103: error: implicit declaration of function `davinci_set_dma_transfer_params' sound/soc/davinci/davinci-pcm.c:103: error: `ASYNC' undeclared (first use in this function) sound/soc/davinci/davinci-pcm.c: In function `davinci_pcm_dma_irq': sound/soc/davinci/davinci-pcm.c:117: error: `DMA_COMPLETE' undeclared (first use in this function) sound/soc/davinci/davinci-pcm.c: In function `davinci_pcm_dma_request': sound/soc/davinci/davinci-pcm.c:134: error: `TCC_ANY' undeclared (first use in this function) sound/soc/davinci/davinci-pcm.c:143: error: implicit declaration of function `davinci_request_dma' sound/soc/davinci/davinci-pcm.c:145: error: `EVENTQ_0' undeclared (first use in this function) sound/soc/davinci/davinci-pcm.c:150: error: `PARAM_ANY' undeclared (first use in this function) sound/soc/davinci/davinci-pcm.c:153: error: implicit declaration of function `davinci_free_dma' sound/soc/davinci/davinci-pcm.c:158: error: implicit declaration of function `davinci_dma_link_lch' sound/soc/davinci/davinci-pcm.c: In function `davinci_pcm_trigger': sound/soc/davinci/davinci-pcm.c:174: error: implicit declaration of function `davinci_start_dma' sound/soc/davinci/davinci-pcm.c:179: error: implicit declaration of function `davinci_stop_dma' sound/soc/davinci/davinci-pcm.c: In function `davinci_pcm_prepare': sound/soc/davinci/davinci-pcm.c:194: error: storage size of 'temp' isn't known sound/soc/davinci/davinci-pcm.c:200: error: implicit declaration of function `davinci_get_dma_params' sound/soc/davinci/davinci-pcm.c:201: error: implicit declaration of function `davinci_set_dma_params' sound/soc/davinci/davinci-pcm.c:194: warning: unused variable `temp' sound/soc/davinci/davinci-pcm.c: In function `davinci_pcm_pointer': sound/soc/davinci/davinci-pcm.c:217: error: implicit declaration of function `davinci_dma_getposition' sound/soc/davinci/davinci-pcm.c: In function `davinci_pcm_close': sound/soc/davinci/davinci-pcm.c:262: error: implicit declaration of function `davinci_dma_unlink_lch' make[3]: *** [sound/soc/davinci/davinci-pcm.o] Error 1 make[2]: *** [sound/soc/davinci] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Are there plans to fix this before release of 2.6.26 ? Thanks. Felix.

5 5

[alsa-devel] [RESEND][PATCH 1/2] wss_lib: move cs4231_lib into wss_lib
by Krzysztof Helt 24 Jun '08

24 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Move the file sound/isa/cs423x/cs4231_lib.c into sound/isa/cs423x/wss_lib.c This is the first step toward merging all libraries for Windows Sound System compatible chips into a single library. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is second version of the patch with makefile copyright changed to Jaroslav Kysela. Only the first patch from two is resend (the other one was already ok). Regards, Krzysztof diff -urNp linux-alsa/sound/isa/Kconfig linux-mm/sound/isa/Kconfig --- linux-alsa/sound/isa/Kconfig 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/Kconfig 2008-06-15 22:39:55.165643480 +0200 @@ -4,7 +4,7 @@ config SND_AD1848_LIB tristate select SND_PCM -config SND_CS4231_LIB +config SND_WSS_LIB tristate select SND_PCM @@ -86,7 +86,7 @@ config SND_AZT2320 select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for soundcards based on the Aztech Systems AZT2320 chip. @@ -108,7 +108,7 @@ config SND_CMI8330 config SND_CS4231 tristate "Generic Cirrus Logic CS4231 driver" select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for CS4231 chips from Cirrus Logic - Crystal Semiconductors. @@ -120,7 +120,7 @@ config SND_CS4232 tristate "Generic Cirrus Logic CS4232 driver" select SND_OPL3_LIB select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for CS4232 chips from Cirrus Logic - Crystal Semiconductors. @@ -132,7 +132,7 @@ config SND_CS4236 tristate "Generic Cirrus Logic CS4236+ driver" select SND_OPL3_LIB select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y to include support for CS4235,CS4236,CS4237B,CS4238B, CS4239 chips from Cirrus Logic - Crystal Semiconductors. @@ -228,7 +228,7 @@ config SND_GUSEXTREME config SND_GUSMAX tristate "Gravis UltraSound MAX" select SND_RAWMIDI - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for Gravis UltraSound MAX soundcards. @@ -240,7 +240,7 @@ config SND_INTERWAVE tristate "AMD InterWave, Gravis UltraSound PnP" depends on PNP select SND_RAWMIDI - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for AMD InterWave based soundcards (Gravis UltraSound Plug & Play, STB SoundRage32, @@ -253,7 +253,7 @@ config SND_INTERWAVE_STB tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)" depends on PNP select SND_RAWMIDI - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for AMD InterWave based soundcards with a TEA6330T bass and treble regulator @@ -266,7 +266,7 @@ config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" select SND_OPL3_LIB select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for Yamaha OPL3-SA2 and OPL3-SA3 chips. @@ -292,7 +292,7 @@ config SND_OPTI92X_CS4231 select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for soundcards based on Opti 82C92x chips and using a CS4231 codec. @@ -304,7 +304,7 @@ config SND_OPTI93X tristate "OPTi 82C93x" select SND_OPL3_LIB select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for soundcards based on Opti 82C93x chips. @@ -315,7 +315,7 @@ config SND_OPTI93X config SND_MIRO tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver" select SND_OPL4_LIB - select SND_CS4231_LIB + select SND_WSS_LIB select SND_MPU401_UART select SND_PCM help @@ -393,7 +393,7 @@ config SND_SSCAPE tristate "Ensoniq SoundScape PnP driver" select SND_HWDEP select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for Ensoniq SoundScape PnP soundcards. @@ -406,7 +406,7 @@ config SND_WAVEFRONT select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART - select SND_CS4231_LIB + select SND_WSS_LIB help Say Y here to include support for Turtle Beach Maui, Tropez and Tropez+ soundcards based on the Wavefront chip. diff -urNp linux-alsa/sound/isa/Makefile linux-mm/sound/isa/Makefile --- linux-alsa/sound/isa/Makefile 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/Makefile 2008-06-15 22:49:45.869642293 +0200 @@ -27,4 +27,4 @@ obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \ - sb/ wavefront/ + sb/ wavefront/ wss/ diff -urNp linux-alsa/sound/isa/cs423x/Makefile linux-mm/sound/isa/cs423x/Makefile --- linux-alsa/sound/isa/cs423x/Makefile 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/cs423x/Makefile 2008-06-15 22:50:20.033648983 +0200 @@ -3,14 +3,12 @@ # Copyright (c) 2001 by Jaroslav Kysela <perex(a)perex.cz> # -snd-cs4231-lib-objs := cs4231_lib.o snd-cs4236-lib-objs := cs4236_lib.o snd-cs4231-objs := cs4231.o snd-cs4232-objs := cs4232.o snd-cs4236-objs := cs4236.o # Toplevel Module Dependency -obj-$(CONFIG_SND_CS4231_LIB) += snd-cs4231-lib.o obj-$(CONFIG_SND_CS4231) += snd-cs4231.o obj-$(CONFIG_SND_CS4232) += snd-cs4232.o obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o diff -urNp linux-alsa/sound/isa/cs423x/cs4231_lib.c linux-mm/sound/isa/cs423x/cs4231_lib.c --- linux-alsa/sound/isa/cs423x/cs4231_lib.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/cs423x/cs4231_lib.c 1970-01-01 01:00:00.000000000 +0100 @@ -1,1945 +0,0 @@ -/* - * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * Routines for control of CS4231(A)/CS4232/InterWave & compatible chips - * - * Bugs: - * - sometimes record brokes playback with WSS portion of - * Yamaha OPL3-SA3 chip - * - CS4231 (GUS MAX) - still trouble with occasional noises - * - broken initialization? - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include <linux/delay.h> -#include <linux/pm.h> -#include <linux/init.h> -#include <linux/interrupt.h> -#include <linux/slab.h> -#include <linux/ioport.h> -#include <sound/core.h> -#include <sound/cs4231.h> -#include <sound/pcm_params.h> - -#include <asm/io.h> -#include <asm/dma.h> -#include <asm/irq.h> - -MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); -MODULE_DESCRIPTION("Routines for control of CS4231(A)/CS4232/InterWave & compatible chips"); -MODULE_LICENSE("GPL"); - -#if 0 -#define SNDRV_DEBUG_MCE -#endif - -/* - * Some variables - */ - -static unsigned char freq_bits[14] = { - /* 5510 */ 0x00 | CS4231_XTAL2, - /* 6620 */ 0x0E | CS4231_XTAL2, - /* 8000 */ 0x00 | CS4231_XTAL1, - /* 9600 */ 0x0E | CS4231_XTAL1, - /* 11025 */ 0x02 | CS4231_XTAL2, - /* 16000 */ 0x02 | CS4231_XTAL1, - /* 18900 */ 0x04 | CS4231_XTAL2, - /* 22050 */ 0x06 | CS4231_XTAL2, - /* 27042 */ 0x04 | CS4231_XTAL1, - /* 32000 */ 0x06 | CS4231_XTAL1, - /* 33075 */ 0x0C | CS4231_XTAL2, - /* 37800 */ 0x08 | CS4231_XTAL2, - /* 44100 */ 0x0A | CS4231_XTAL2, - /* 48000 */ 0x0C | CS4231_XTAL1 -}; - -static unsigned int rates[14] = { - 5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050, - 27042, 32000, 33075, 37800, 44100, 48000 -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static int snd_cs4231_xrate(struct snd_pcm_runtime *runtime) -{ - return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); -} - -static unsigned char snd_cs4231_original_image[32] = -{ - 0x00, /* 00/00 - lic */ - 0x00, /* 01/01 - ric */ - 0x9f, /* 02/02 - la1ic */ - 0x9f, /* 03/03 - ra1ic */ - 0x9f, /* 04/04 - la2ic */ - 0x9f, /* 05/05 - ra2ic */ - 0xbf, /* 06/06 - loc */ - 0xbf, /* 07/07 - roc */ - 0x20, /* 08/08 - pdfr */ - CS4231_AUTOCALIB, /* 09/09 - ic */ - 0x00, /* 0a/10 - pc */ - 0x00, /* 0b/11 - ti */ - CS4231_MODE2, /* 0c/12 - mi */ - 0xfc, /* 0d/13 - lbc */ - 0x00, /* 0e/14 - pbru */ - 0x00, /* 0f/15 - pbrl */ - 0x80, /* 10/16 - afei */ - 0x01, /* 11/17 - afeii */ - 0x9f, /* 12/18 - llic */ - 0x9f, /* 13/19 - rlic */ - 0x00, /* 14/20 - tlb */ - 0x00, /* 15/21 - thb */ - 0x00, /* 16/22 - la3mic/reserved */ - 0x00, /* 17/23 - ra3mic/reserved */ - 0x00, /* 18/24 - afs */ - 0x00, /* 19/25 - lamoc/version */ - 0xcf, /* 1a/26 - mioc */ - 0x00, /* 1b/27 - ramoc/reserved */ - 0x20, /* 1c/28 - cdfr */ - 0x00, /* 1d/29 - res4 */ - 0x00, /* 1e/30 - cbru */ - 0x00, /* 1f/31 - cbrl */ -}; - -static unsigned char snd_opti93x_original_image[32] = -{ - 0x00, /* 00/00 - l_mixout_outctrl */ - 0x00, /* 01/01 - r_mixout_outctrl */ - 0x88, /* 02/02 - l_cd_inctrl */ - 0x88, /* 03/03 - r_cd_inctrl */ - 0x88, /* 04/04 - l_a1/fm_inctrl */ - 0x88, /* 05/05 - r_a1/fm_inctrl */ - 0x80, /* 06/06 - l_dac_inctrl */ - 0x80, /* 07/07 - r_dac_inctrl */ - 0x00, /* 08/08 - ply_dataform_reg */ - 0x00, /* 09/09 - if_conf */ - 0x00, /* 0a/10 - pin_ctrl */ - 0x00, /* 0b/11 - err_init_reg */ - 0x0a, /* 0c/12 - id_reg */ - 0x00, /* 0d/13 - reserved */ - 0x00, /* 0e/14 - ply_upcount_reg */ - 0x00, /* 0f/15 - ply_lowcount_reg */ - 0x88, /* 10/16 - reserved/l_a1_inctrl */ - 0x88, /* 11/17 - reserved/r_a1_inctrl */ - 0x88, /* 12/18 - l_line_inctrl */ - 0x88, /* 13/19 - r_line_inctrl */ - 0x88, /* 14/20 - l_mic_inctrl */ - 0x88, /* 15/21 - r_mic_inctrl */ - 0x80, /* 16/22 - l_out_outctrl */ - 0x80, /* 17/23 - r_out_outctrl */ - 0x00, /* 18/24 - reserved */ - 0x00, /* 19/25 - reserved */ - 0x00, /* 1a/26 - reserved */ - 0x00, /* 1b/27 - reserved */ - 0x00, /* 1c/28 - cap_dataform_reg */ - 0x00, /* 1d/29 - reserved */ - 0x00, /* 1e/30 - cap_upcount_reg */ - 0x00 /* 1f/31 - cap_lowcount_reg */ -}; - -/* - * Basic I/O functions - */ - -static inline void cs4231_outb(struct snd_cs4231 *chip, u8 offset, u8 val) -{ - outb(val, chip->port + offset); -} - -static inline u8 cs4231_inb(struct snd_cs4231 *chip, u8 offset) -{ - return inb(chip->port + offset); -} - -static void snd_cs4231_wait(struct snd_cs4231 *chip) -{ - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); -} - -static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - unsigned char tmp = (chip->image[reg] & mask) | value; - - snd_cs4231_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - chip->image[reg] = tmp; - if (!chip->calibrate_mute) { - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - wmb(); - cs4231_outb(chip, CS4231P(REG), tmp); - mb(); - } -} - -static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) -{ - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(10); - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - cs4231_outb(chip, CS4231P(REG), value); - mb(); -} - -void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) -{ - snd_cs4231_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - cs4231_outb(chip, CS4231P(REG), value); - chip->image[reg] = value; - mb(); - snd_printdd("codec out - reg 0x%x = 0x%x\n", - chip->mce_bit | reg, value); -} - -unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) -{ - snd_cs4231_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); -#endif - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - mb(); - return cs4231_inb(chip, CS4231P(REG)); -} - -void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val) -{ - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); - cs4231_outb(chip, CS4231P(REG), reg | (chip->image[CS4236_EXT_REG] & 0x01)); - cs4231_outb(chip, CS4231P(REG), val); - chip->eimage[CS4236_REG(reg)] = val; -#if 0 - printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val); -#endif -} - -unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg) -{ - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); - cs4231_outb(chip, CS4231P(REG), reg | (chip->image[CS4236_EXT_REG] & 0x01)); -#if 1 - return cs4231_inb(chip, CS4231P(REG)); -#else - { - unsigned char res; - res = cs4231_inb(chip, CS4231P(REG)); - printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res); - return res; - } -#endif -} - -#if 0 - -static void snd_cs4231_debug(struct snd_cs4231 *chip) -{ - printk("CS4231 REGS: INDEX = 0x%02x ", cs4231_inb(chip, CS4231P(REGSEL))); - printk(" STATUS = 0x%02x\n", cs4231_inb(chip, CS4231P(STATUS))); - printk(" 0x00: left input = 0x%02x ", snd_cs4231_in(chip, 0x00)); - printk(" 0x10: alt 1 (CFIG 2) = 0x%02x\n", snd_cs4231_in(chip, 0x10)); - printk(" 0x01: right input = 0x%02x ", snd_cs4231_in(chip, 0x01)); - printk(" 0x11: alt 2 (CFIG 3) = 0x%02x\n", snd_cs4231_in(chip, 0x11)); - printk(" 0x02: GF1 left input = 0x%02x ", snd_cs4231_in(chip, 0x02)); - printk(" 0x12: left line in = 0x%02x\n", snd_cs4231_in(chip, 0x12)); - printk(" 0x03: GF1 right input = 0x%02x ", snd_cs4231_in(chip, 0x03)); - printk(" 0x13: right line in = 0x%02x\n", snd_cs4231_in(chip, 0x13)); - printk(" 0x04: CD left input = 0x%02x ", snd_cs4231_in(chip, 0x04)); - printk(" 0x14: timer low = 0x%02x\n", snd_cs4231_in(chip, 0x14)); - printk(" 0x05: CD right input = 0x%02x ", snd_cs4231_in(chip, 0x05)); - printk(" 0x15: timer high = 0x%02x\n", snd_cs4231_in(chip, 0x15)); - printk(" 0x06: left output = 0x%02x ", snd_cs4231_in(chip, 0x06)); - printk(" 0x16: left MIC (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x16)); - printk(" 0x07: right output = 0x%02x ", snd_cs4231_in(chip, 0x07)); - printk(" 0x17: right MIC (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x17)); - printk(" 0x08: playback format = 0x%02x ", snd_cs4231_in(chip, 0x08)); - printk(" 0x18: IRQ status = 0x%02x\n", snd_cs4231_in(chip, 0x18)); - printk(" 0x09: iface (CFIG 1) = 0x%02x ", snd_cs4231_in(chip, 0x09)); - printk(" 0x19: left line out = 0x%02x\n", snd_cs4231_in(chip, 0x19)); - printk(" 0x0a: pin control = 0x%02x ", snd_cs4231_in(chip, 0x0a)); - printk(" 0x1a: mono control = 0x%02x\n", snd_cs4231_in(chip, 0x1a)); - printk(" 0x0b: init & status = 0x%02x ", snd_cs4231_in(chip, 0x0b)); - printk(" 0x1b: right line out = 0x%02x\n", snd_cs4231_in(chip, 0x1b)); - printk(" 0x0c: revision & mode = 0x%02x ", snd_cs4231_in(chip, 0x0c)); - printk(" 0x1c: record format = 0x%02x\n", snd_cs4231_in(chip, 0x1c)); - printk(" 0x0d: loopback = 0x%02x ", snd_cs4231_in(chip, 0x0d)); - printk(" 0x1d: var freq (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x1d)); - printk(" 0x0e: ply upr count = 0x%02x ", snd_cs4231_in(chip, 0x0e)); - printk(" 0x1e: ply lwr count = 0x%02x\n", snd_cs4231_in(chip, 0x1e)); - printk(" 0x0f: rec upr count = 0x%02x ", snd_cs4231_in(chip, 0x0f)); - printk(" 0x1f: rec lwr count = 0x%02x\n", snd_cs4231_in(chip, 0x1f)); -} - -#endif - -/* - * CS4231 detection / MCE routines - */ - -static void snd_cs4231_busy_wait(struct snd_cs4231 *chip) -{ - int timeout; - - /* huh.. looks like this sequence is proper for CS4231A chip (GUS MAX) */ - for (timeout = 5; timeout > 0; timeout--) - cs4231_inb(chip, CS4231P(REGSEL)); - /* end of cleanup sequence */ - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(10); -} - -void snd_cs4231_mce_up(struct snd_cs4231 *chip) -{ - unsigned long flags; - int timeout; - - snd_cs4231_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_up - auto calibration time out (0)\n"); -#endif - spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit |= CS4231_MCE; - timeout = cs4231_inb(chip, CS4231P(REGSEL)); - if (timeout == 0x80) - snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); - if (!(timeout & CS4231_MCE)) - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -void snd_cs4231_mce_down(struct snd_cs4231 *chip) -{ - unsigned long flags; - unsigned long end_time; - int timeout; - - snd_cs4231_busy_wait(chip); - -#ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); -#endif - spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit &= ~CS4231_MCE; - timeout = cs4231_inb(chip, CS4231P(REGSEL)); - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (timeout == 0x80) - snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0 || - !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { - return; - } - - /* - * Wait for (possible -- during init auto-calibration may not be set) - * calibration process to start. Needs upto 5 sample periods on AD1848 - * which at the slowest possible rate of 5.5125 kHz means 907 us. - */ - msleep(1); - - snd_printdd("(1) jiffies = %lu\n", jiffies); - - /* check condition up to 250 ms */ - end_time = jiffies + msecs_to_jiffies(250); - while (snd_cs4231_in(chip, CS4231_TEST_INIT) & - CS4231_CALIB_IN_PROGRESS) { - - if (time_after(jiffies, end_time)) { - snd_printk(KERN_ERR "mce_down - " - "auto calibration time out (2)\n"); - return; - } - msleep(1); - } - - snd_printdd("(2) jiffies = %lu\n", jiffies); - - /* check condition up to 100 ms */ - end_time = jiffies + msecs_to_jiffies(100); - while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { - if (time_after(jiffies, end_time)) { - snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); - return; - } - msleep(1); - } - - snd_printdd("(3) jiffies = %lu\n", jiffies); - snd_printd("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); -} - -static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size) -{ - switch (format & 0xe0) { - case CS4231_LINEAR_16: - case CS4231_LINEAR_16_BIG: - size >>= 1; - break; - case CS4231_ADPCM_16: - return size >> 2; - } - if (format & CS4231_STEREO) - size >>= 1; - return size; -} - -static int snd_cs4231_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - int result = 0; - unsigned int what; - struct snd_pcm_substream *s; - int do_start; - -#if 0 - printk("codec trigger!!! - what = %i, enable = %i, status = 0x%x\n", what, enable, cs4231_inb(chip, CS4231P(STATUS))); -#endif - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - do_start = 1; break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - do_start = 0; break; - default: - return -EINVAL; - } - - what = 0; - snd_pcm_group_for_each_entry(s, substream) { - if (s == chip->playback_substream) { - what |= CS4231_PLAYBACK_ENABLE; - snd_pcm_trigger_done(s, substream); - } else if (s == chip->capture_substream) { - what |= CS4231_RECORD_ENABLE; - snd_pcm_trigger_done(s, substream); - } - } - spin_lock(&chip->reg_lock); - if (do_start) { - chip->image[CS4231_IFACE_CTRL] |= what; - if (chip->trigger) - chip->trigger(chip, what, 1); - } else { - chip->image[CS4231_IFACE_CTRL] &= ~what; - if (chip->trigger) - chip->trigger(chip, what, 0); - } - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); - spin_unlock(&chip->reg_lock); -#if 0 - snd_cs4231_debug(chip); -#endif - return result; -} - -/* - * CODEC I/O - */ - -static unsigned char snd_cs4231_get_rate(unsigned int rate) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(rates); i++) - if (rate == rates[i]) - return freq_bits[i]; - // snd_BUG(); - return freq_bits[ARRAY_SIZE(rates) - 1]; -} - -static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip, - int format, - int channels) -{ - unsigned char rformat; - - rformat = CS4231_LINEAR_8; - switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: rformat = CS4231_ULAW_8; break; - case SNDRV_PCM_FORMAT_A_LAW: rformat = CS4231_ALAW_8; break; - case SNDRV_PCM_FORMAT_S16_LE: rformat = CS4231_LINEAR_16; break; - case SNDRV_PCM_FORMAT_S16_BE: rformat = CS4231_LINEAR_16_BIG; break; - case SNDRV_PCM_FORMAT_IMA_ADPCM: rformat = CS4231_ADPCM_16; break; - } - if (channels > 1) - rformat |= CS4231_STEREO; -#if 0 - snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); -#endif - return rformat; -} - -static void snd_cs4231_calibrate_mute(struct snd_cs4231 *chip, int mute) -{ - unsigned long flags; - - mute = mute ? 1 : 0; - spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->calibrate_mute == mute) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } - if (!mute) { - snd_cs4231_dout(chip, CS4231_LEFT_INPUT, chip->image[CS4231_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_INPUT, chip->image[CS4231_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); - } - snd_cs4231_dout(chip, CS4231_AUX1_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX1_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LEFT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_LEFT_LINE_IN, mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); - snd_cs4231_dout(chip, CS4231_RIGHT_LINE_IN, mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); - snd_cs4231_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); - if (chip->hardware == CS4231_HW_INTERWAVE) { - snd_cs4231_dout(chip, CS4231_LEFT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); - snd_cs4231_dout(chip, CS4231_LINE_LEFT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_LINE_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); - } - chip->calibrate_mute = mute; - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -static void snd_cs4231_playback_format(struct snd_cs4231 *chip, - struct snd_pcm_hw_params *params, - unsigned char pdfr) -{ - unsigned long flags; - int full_calib = 1; - - mutex_lock(&chip->mce_mutex); - snd_cs4231_calibrate_mute(chip, 1); - if (chip->hardware == CS4231_HW_CS4231A || - (chip->hardware & CS4231_HW_CS4232_MASK)) { - spin_lock_irqsave(&chip->reg_lock, flags); - if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (pdfr & 0x0f)) { /* rate is same? */ - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x10); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x10); - udelay(100); /* Fixes audible clicks at least on GUS MAX */ - full_calib = 0; - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - } - if (full_calib) { - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->hardware != CS4231_HW_INTERWAVE && !chip->single_dma) { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, - (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ? - (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) : - pdfr); - } else { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (chip->hardware == CS4231_HW_OPL3SA2) - udelay(100); /* this seems to help */ - snd_cs4231_mce_down(chip); - } - snd_cs4231_calibrate_mute(chip, 0); - mutex_unlock(&chip->mce_mutex); -} - -static void snd_cs4231_capture_format(struct snd_cs4231 *chip, - struct snd_pcm_hw_params *params, - unsigned char cdfr) -{ - unsigned long flags; - int full_calib = 1; - - mutex_lock(&chip->mce_mutex); - snd_cs4231_calibrate_mute(chip, 1); - if (chip->hardware == CS4231_HW_CS4231A || - (chip->hardware & CS4231_HW_CS4232_MASK)) { - spin_lock_irqsave(&chip->reg_lock, flags); - if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (cdfr & 0x0f) || /* rate is same? */ - (chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x20); - snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT] = cdfr); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); - full_calib = 0; - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - } - if (full_calib) { - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - if (chip->hardware != CS4231_HW_INTERWAVE) { - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, - ((chip->single_dma ? cdfr : chip->image[CS4231_PLAYBK_FORMAT]) & 0xf0) | - (cdfr & 0x0f)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - } - } - snd_cs4231_out(chip, CS4231_REC_FORMAT, cdfr); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - } - snd_cs4231_calibrate_mute(chip, 0); - mutex_unlock(&chip->mce_mutex); -} - -/* - * Timer interface - */ - -static unsigned long snd_cs4231_timer_resolution(struct snd_timer * timer) -{ - struct snd_cs4231 *chip = snd_timer_chip(timer); - if (chip->hardware & CS4231_HW_CS4236B_MASK) - return 14467; - else - return chip->image[CS4231_PLAYBK_FORMAT] & 1 ? 9969 : 9920; -} - -static int snd_cs4231_timer_start(struct snd_timer * timer) -{ - unsigned long flags; - unsigned int ticks; - struct snd_cs4231 *chip = snd_timer_chip(timer); - spin_lock_irqsave(&chip->reg_lock, flags); - ticks = timer->sticks; - if ((chip->image[CS4231_ALT_FEATURE_1] & CS4231_TIMER_ENABLE) == 0 || - (unsigned char)(ticks >> 8) != chip->image[CS4231_TIMER_HIGH] || - (unsigned char)ticks != chip->image[CS4231_TIMER_LOW]) { - snd_cs4231_out(chip, CS4231_TIMER_HIGH, chip->image[CS4231_TIMER_HIGH] = (unsigned char) (ticks >> 8)); - snd_cs4231_out(chip, CS4231_TIMER_LOW, chip->image[CS4231_TIMER_LOW] = (unsigned char) ticks); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | CS4231_TIMER_ENABLE); - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -static int snd_cs4231_timer_stop(struct snd_timer * timer) -{ - unsigned long flags; - struct snd_cs4231 *chip = snd_timer_chip(timer); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -static void snd_cs4231_init(struct snd_cs4231 *chip) -{ - unsigned long flags; - - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("init: (1)\n"); -#endif - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO | - CS4231_CALIB_MODE); - chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("init: (2)\n"); -#endif - - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("init: (3) - afei = 0x%x\n", chip->image[CS4231_ALT_FEATURE_1]); -#endif - - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_2, chip->image[CS4231_ALT_FEATURE_2]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("init: (4)\n"); -#endif - - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - -#ifdef SNDRV_DEBUG_MCE - snd_printk("init: (5)\n"); -#endif -} - -static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode) -{ - unsigned long flags; - - mutex_lock(&chip->open_mutex); - if ((chip->mode & mode) || - ((chip->mode & CS4231_MODE_OPEN) && chip->single_dma)) { - mutex_unlock(&chip->open_mutex); - return -EAGAIN; - } - if (chip->mode & CS4231_MODE_OPEN) { - chip->mode |= mode; - mutex_unlock(&chip->open_mutex); - return 0; - } - /* ok. now enable and ack CODEC IRQ */ - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | - CS4231_RECORD_IRQ | - CS4231_TIMER_IRQ); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - chip->image[CS4231_PIN_CTRL] |= CS4231_IRQ_ENABLE; - snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | - CS4231_RECORD_IRQ | - CS4231_TIMER_IRQ); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->mode = mode; - mutex_unlock(&chip->open_mutex); - return 0; -} - -static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode) -{ - unsigned long flags; - - mutex_lock(&chip->open_mutex); - chip->mode &= ~mode; - if (chip->mode & CS4231_MODE_OPEN) { - mutex_unlock(&chip->open_mutex); - return; - } - snd_cs4231_calibrate_mute(chip, 1); - - /* disable IRQ */ - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - chip->image[CS4231_PIN_CTRL] &= ~CS4231_IRQ_ENABLE; - snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); - - /* now disable record & playback */ - - if (chip->image[CS4231_IFACE_CTRL] & (CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO)) { - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_down(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - } - - /* clear IRQ again */ - snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - spin_unlock_irqrestore(&chip->reg_lock, flags); - - snd_cs4231_calibrate_mute(chip, 0); - - chip->mode = 0; - mutex_unlock(&chip->open_mutex); -} - -/* - * timer open/close - */ - -static int snd_cs4231_timer_open(struct snd_timer * timer) -{ - struct snd_cs4231 *chip = snd_timer_chip(timer); - snd_cs4231_open(chip, CS4231_MODE_TIMER); - return 0; -} - -static int snd_cs4231_timer_close(struct snd_timer * timer) -{ - struct snd_cs4231 *chip = snd_timer_chip(timer); - snd_cs4231_close(chip, CS4231_MODE_TIMER); - return 0; -} - -static struct snd_timer_hardware snd_cs4231_timer_table = -{ - .flags = SNDRV_TIMER_HW_AUTO, - .resolution = 9945, - .ticks = 65535, - .open = snd_cs4231_timer_open, - .close = snd_cs4231_timer_close, - .c_resolution = snd_cs4231_timer_resolution, - .start = snd_cs4231_timer_start, - .stop = snd_cs4231_timer_stop, -}; - -/* - * ok.. exported functions.. - */ - -static int snd_cs4231_playback_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - unsigned char new_pdfr; - int err; - - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) - return err; - new_pdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | - snd_cs4231_get_rate(params_rate(hw_params)); - chip->set_playback_format(chip, hw_params, new_pdfr); - return 0; -} - -static int snd_cs4231_playback_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int snd_cs4231_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); - - spin_lock_irqsave(&chip->reg_lock, flags); - chip->p_dma_size = size; - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO); - snd_dma_program(chip->dma1, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); - count = snd_cs4231_get_count(chip->image[CS4231_PLAYBK_FORMAT], count) - 1; - snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); - spin_unlock_irqrestore(&chip->reg_lock, flags); -#if 0 - snd_cs4231_debug(chip); -#endif - return 0; -} - -static int snd_cs4231_capture_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - unsigned char new_cdfr; - int err; - - if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) - return err; - new_cdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | - snd_cs4231_get_rate(params_rate(hw_params)); - chip->set_capture_format(chip, hw_params, new_cdfr); - return 0; -} - -static int snd_cs4231_capture_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); - - spin_lock_irqsave(&chip->reg_lock, flags); - chip->c_dma_size = size; - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); - snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); - count = snd_cs4231_get_count(chip->image[CS4231_REC_FORMAT], count) - 1; - if (chip->single_dma && chip->hardware != CS4231_HW_INTERWAVE) { - snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); - } else { - snd_cs4231_out(chip, CS4231_REC_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_REC_UPR_CNT, (unsigned char) (count >> 8)); - } - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -void snd_cs4231_overrange(struct snd_cs4231 *chip) -{ - unsigned long flags; - unsigned char res; - - spin_lock_irqsave(&chip->reg_lock, flags); - res = snd_cs4231_in(chip, CS4231_TEST_INIT); - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (res & (0x08 | 0x02)) /* detect overrange only above 0dB; may be user selectable? */ - chip->capture_substream->runtime->overrange++; -} - -irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id) -{ - struct snd_cs4231 *chip = dev_id; - unsigned char status; - - status = snd_cs4231_in(chip, CS4231_IRQ_STATUS); - if (status & CS4231_TIMER_IRQ) { - if (chip->timer) - snd_timer_interrupt(chip->timer, chip->timer->sticks); - } - if (chip->single_dma && chip->hardware != CS4231_HW_INTERWAVE) { - if (status & CS4231_PLAYBACK_IRQ) { - if (chip->mode & CS4231_MODE_PLAY) { - if (chip->playback_substream) - snd_pcm_period_elapsed(chip->playback_substream); - } - if (chip->mode & CS4231_MODE_RECORD) { - if (chip->capture_substream) { - snd_cs4231_overrange(chip); - snd_pcm_period_elapsed(chip->capture_substream); - } - } - } - } else { - if (status & CS4231_PLAYBACK_IRQ) { - if (chip->playback_substream) - snd_pcm_period_elapsed(chip->playback_substream); - } - if (status & CS4231_RECORD_IRQ) { - if (chip->capture_substream) { - snd_cs4231_overrange(chip); - snd_pcm_period_elapsed(chip->capture_substream); - } - } - } - - spin_lock(&chip->reg_lock); - snd_cs4231_outm(chip, CS4231_IRQ_STATUS, ~CS4231_ALL_IRQS | ~status, 0); - spin_unlock(&chip->reg_lock); - return IRQ_HANDLED; -} - -static snd_pcm_uframes_t snd_cs4231_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) - return 0; - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_cs4231_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE)) - return 0; - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -/* - - */ - -static int snd_cs4231_probe(struct snd_cs4231 *chip) -{ - unsigned long flags; - int i, id, rev; - unsigned char *ptr; - unsigned int hw; - -#if 0 - snd_cs4231_debug(chip); -#endif - id = 0; - for (i = 0; i < 50; i++) { - mb(); - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - udelay(2000); - else { - spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - id = snd_cs4231_in(chip, CS4231_MISC_INFO) & 0x0f; - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (id == 0x0a) - break; /* this is valid value */ - } - } - snd_printdd("cs4231: port = 0x%lx, id = 0x%x\n", chip->port, id); - if (id != 0x0a) - return -ENODEV; /* no valid device found */ - - if (((hw = chip->hardware) & CS4231_HW_TYPE_MASK) == CS4231_HW_DETECT) { - rev = snd_cs4231_in(chip, CS4231_VERSION) & 0xe7; - snd_printdd("CS4231: VERSION (I25) = 0x%x\n", rev); - if (rev == 0x80) { - unsigned char tmp = snd_cs4231_in(chip, 23); - snd_cs4231_out(chip, 23, ~tmp); - if (snd_cs4231_in(chip, 23) != tmp) - chip->hardware = CS4231_HW_AD1845; - else - chip->hardware = CS4231_HW_CS4231; - } else if (rev == 0xa0) { - chip->hardware = CS4231_HW_CS4231A; - } else if (rev == 0xa2) { - chip->hardware = CS4231_HW_CS4232; - } else if (rev == 0xb2) { - chip->hardware = CS4231_HW_CS4232A; - } else if (rev == 0x83) { - chip->hardware = CS4231_HW_CS4236; - } else if (rev == 0x03) { - chip->hardware = CS4231_HW_CS4236B; - } else { - snd_printk("unknown CS chip with version 0x%x\n", rev); - return -ENODEV; /* unknown CS4231 chip? */ - } - } - spin_lock_irqsave(&chip->reg_lock, flags); - cs4231_inb(chip, CS4231P(STATUS)); /* clear any pendings IRQ */ - cs4231_outb(chip, CS4231P(STATUS), 0); - mb(); - spin_unlock_irqrestore(&chip->reg_lock, flags); - - chip->image[CS4231_MISC_INFO] = CS4231_MODE2; - switch (chip->hardware) { - case CS4231_HW_INTERWAVE: - chip->image[CS4231_MISC_INFO] = CS4231_IW_MODE3; - break; - case CS4231_HW_CS4235: - case CS4231_HW_CS4236B: - case CS4231_HW_CS4237B: - case CS4231_HW_CS4238B: - case CS4231_HW_CS4239: - if (hw == CS4231_HW_DETECT3) - chip->image[CS4231_MISC_INFO] = CS4231_4236_MODE3; - else - chip->hardware = CS4231_HW_CS4236; - break; - } - - chip->image[CS4231_IFACE_CTRL] = - (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | - (chip->single_dma ? CS4231_SINGLE_DMA : 0); - if (chip->hardware != CS4231_HW_OPTI93X) { - chip->image[CS4231_ALT_FEATURE_1] = 0x80; - chip->image[CS4231_ALT_FEATURE_2] = - chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; - } - ptr = (unsigned char *) &chip->image; - snd_cs4231_mce_down(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - for (i = 0; i < 32; i++) /* ok.. fill all CS4231 registers */ - snd_cs4231_out(chip, i, *ptr++); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_cs4231_mce_up(chip); - snd_cs4231_mce_down(chip); - - mdelay(2); - - /* ok.. try check hardware version for CS4236+ chips */ - if ((hw & CS4231_HW_TYPE_MASK) == CS4231_HW_DETECT) { - if (chip->hardware == CS4231_HW_CS4236B) { - rev = snd_cs4236_ext_in(chip, CS4236_VERSION); - snd_cs4236_ext_out(chip, CS4236_VERSION, 0xff); - id = snd_cs4236_ext_in(chip, CS4236_VERSION); - snd_cs4236_ext_out(chip, CS4236_VERSION, rev); - snd_printdd("CS4231: ext version; rev = 0x%x, id = 0x%x\n", rev, id); - if ((id & 0x1f) == 0x1d) { /* CS4235 */ - chip->hardware = CS4231_HW_CS4235; - switch (id >> 5) { - case 4: - case 5: - case 6: - break; - default: - snd_printk("unknown CS4235 chip (enhanced version = 0x%x)\n", id); - } - } else if ((id & 0x1f) == 0x0b) { /* CS4236/B */ - switch (id >> 5) { - case 4: - case 5: - case 6: - case 7: - chip->hardware = CS4231_HW_CS4236B; - break; - default: - snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id); - } - } else if ((id & 0x1f) == 0x08) { /* CS4237B */ - chip->hardware = CS4231_HW_CS4237B; - switch (id >> 5) { - case 4: - case 5: - case 6: - case 7: - break; - default: - snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id); - } - } else if ((id & 0x1f) == 0x09) { /* CS4238B */ - chip->hardware = CS4231_HW_CS4238B; - switch (id >> 5) { - case 5: - case 6: - case 7: - break; - default: - snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id); - } - } else if ((id & 0x1f) == 0x1e) { /* CS4239 */ - chip->hardware = CS4231_HW_CS4239; - switch (id >> 5) { - case 4: - case 5: - case 6: - break; - default: - snd_printk("unknown CS4239 chip (enhanced version = 0x%x)\n", id); - } - } else { - snd_printk("unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", id); - } - } - } - return 0; /* all things are ok.. */ -} - -/* - - */ - -static struct snd_pcm_hardware snd_cs4231_playback = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5510, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_cs4231_capture = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5510, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -/* - - */ - -static int snd_cs4231_playback_open(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - runtime->hw = snd_cs4231_playback; - - /* hardware bug in InterWave chipset */ - if (chip->hardware == CS4231_HW_INTERWAVE && chip->dma1 > 3) - runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_MU_LAW; - - /* hardware limitation of cheap chips */ - if (chip->hardware == CS4231_HW_CS4235 || - chip->hardware == CS4231_HW_CS4239) - runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; - - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); - - if (chip->claim_dma) { - if ((err = chip->claim_dma(chip, chip->dma_private_data, chip->dma1)) < 0) - return err; - } - - if ((err = snd_cs4231_open(chip, CS4231_MODE_PLAY)) < 0) { - if (chip->release_dma) - chip->release_dma(chip, chip->dma_private_data, chip->dma1); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); - return err; - } - chip->playback_substream = substream; - snd_pcm_set_sync(substream); - chip->rate_constraint(runtime); - return 0; -} - -static int snd_cs4231_capture_open(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int err; - - runtime->hw = snd_cs4231_capture; - - /* hardware limitation of cheap chips */ - if (chip->hardware == CS4231_HW_CS4235 || - chip->hardware == CS4231_HW_CS4239) - runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; - - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); - - if (chip->claim_dma) { - if ((err = chip->claim_dma(chip, chip->dma_private_data, chip->dma2)) < 0) - return err; - } - - if ((err = snd_cs4231_open(chip, CS4231_MODE_RECORD)) < 0) { - if (chip->release_dma) - chip->release_dma(chip, chip->dma_private_data, chip->dma2); - snd_free_pages(runtime->dma_area, runtime->dma_bytes); - return err; - } - chip->capture_substream = substream; - snd_pcm_set_sync(substream); - chip->rate_constraint(runtime); - return 0; -} - -static int snd_cs4231_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - - chip->playback_substream = NULL; - snd_cs4231_close(chip, CS4231_MODE_PLAY); - return 0; -} - -static int snd_cs4231_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); - - chip->capture_substream = NULL; - snd_cs4231_close(chip, CS4231_MODE_RECORD); - return 0; -} - -#ifdef CONFIG_PM - -/* lowlevel suspend callback for CS4231 */ -static void snd_cs4231_suspend(struct snd_cs4231 *chip) -{ - int reg; - unsigned long flags; - - snd_pcm_suspend_all(chip->pcm); - spin_lock_irqsave(&chip->reg_lock, flags); - for (reg = 0; reg < 32; reg++) - chip->image[reg] = snd_cs4231_in(chip, reg); - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* lowlevel resume callback for CS4231 */ -static void snd_cs4231_resume(struct snd_cs4231 *chip) -{ - int reg; - unsigned long flags; - /* int timeout; */ - - snd_cs4231_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - for (reg = 0; reg < 32; reg++) { - switch (reg) { - case CS4231_VERSION: - break; - default: - snd_cs4231_out(chip, reg, chip->image[reg]); - break; - } - } - spin_unlock_irqrestore(&chip->reg_lock, flags); -#if 1 - snd_cs4231_mce_down(chip); -#else - /* The following is a workaround to avoid freeze after resume on TP600E. - This is the first half of copy of snd_cs4231_mce_down(), but doesn't - include rescheduling. -- iwai - */ - snd_cs4231_busy_wait(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit &= ~CS4231_MCE; - timeout = cs4231_inb(chip, CS4231P(REGSEL)); - cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (timeout == 0x80) - snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0 || - !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { - return; - } - snd_cs4231_busy_wait(chip); -#endif -} -#endif /* CONFIG_PM */ - -static int snd_cs4231_free(struct snd_cs4231 *chip) -{ - release_and_free_resource(chip->res_port); - release_and_free_resource(chip->res_cport); - if (chip->irq >= 0) { - disable_irq(chip->irq); - if (!(chip->hwshare & CS4231_HWSHARE_IRQ)) - free_irq(chip->irq, (void *) chip); - } - if (!(chip->hwshare & CS4231_HWSHARE_DMA1) && chip->dma1 >= 0) { - snd_dma_disable(chip->dma1); - free_dma(chip->dma1); - } - if (!(chip->hwshare & CS4231_HWSHARE_DMA2) && chip->dma2 >= 0 && chip->dma2 != chip->dma1) { - snd_dma_disable(chip->dma2); - free_dma(chip->dma2); - } - if (chip->timer) - snd_device_free(chip->card, chip->timer); - kfree(chip); - return 0; -} - -static int snd_cs4231_dev_free(struct snd_device *device) -{ - struct snd_cs4231 *chip = device->device_data; - return snd_cs4231_free(chip); -} - -const char *snd_cs4231_chip_id(struct snd_cs4231 *chip) -{ - switch (chip->hardware) { - case CS4231_HW_CS4231: return "CS4231"; - case CS4231_HW_CS4231A: return "CS4231A"; - case CS4231_HW_CS4232: return "CS4232"; - case CS4231_HW_CS4232A: return "CS4232A"; - case CS4231_HW_CS4235: return "CS4235"; - case CS4231_HW_CS4236: return "CS4236"; - case CS4231_HW_CS4236B: return "CS4236B"; - case CS4231_HW_CS4237B: return "CS4237B"; - case CS4231_HW_CS4238B: return "CS4238B"; - case CS4231_HW_CS4239: return "CS4239"; - case CS4231_HW_INTERWAVE: return "AMD InterWave"; - case CS4231_HW_OPL3SA2: return chip->card->shortname; - case CS4231_HW_AD1845: return "AD1845"; - case CS4231_HW_OPTI93X: return "OPTi 93x"; - default: return "???"; - } -} - -static int snd_cs4231_new(struct snd_card *card, - unsigned short hardware, - unsigned short hwshare, - struct snd_cs4231 ** rchip) -{ - struct snd_cs4231 *chip; - - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - chip->hardware = hardware; - chip->hwshare = hwshare; - - spin_lock_init(&chip->reg_lock); - mutex_init(&chip->mce_mutex); - mutex_init(&chip->open_mutex); - chip->card = card; - chip->rate_constraint = snd_cs4231_xrate; - chip->set_playback_format = snd_cs4231_playback_format; - chip->set_capture_format = snd_cs4231_capture_format; - if (chip->hardware == CS4231_HW_OPTI93X) - memcpy(&chip->image, &snd_opti93x_original_image, - sizeof(snd_opti93x_original_image)); - else - memcpy(&chip->image, &snd_cs4231_original_image, - sizeof(snd_cs4231_original_image)); - - *rchip = chip; - return 0; -} - -int snd_cs4231_create(struct snd_card *card, - unsigned long port, - unsigned long cport, - int irq, int dma1, int dma2, - unsigned short hardware, - unsigned short hwshare, - struct snd_cs4231 ** rchip) -{ - static struct snd_device_ops ops = { - .dev_free = snd_cs4231_dev_free, - }; - struct snd_cs4231 *chip; - int err; - - err = snd_cs4231_new(card, hardware, hwshare, &chip); - if (err < 0) - return err; - - chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - - if ((chip->res_port = request_region(port, 4, "CS4231")) == NULL) { - snd_printk(KERN_ERR "cs4231: can't grab port 0x%lx\n", port); - snd_cs4231_free(chip); - return -EBUSY; - } - chip->port = port; - if ((long)cport >= 0 && (chip->res_cport = request_region(cport, 8, "CS4232 Control")) == NULL) { - snd_printk(KERN_ERR "cs4231: can't grab control port 0x%lx\n", cport); - snd_cs4231_free(chip); - return -ENODEV; - } - chip->cport = cport; - if (!(hwshare & CS4231_HWSHARE_IRQ) && request_irq(irq, snd_cs4231_interrupt, IRQF_DISABLED, "CS4231", (void *) chip)) { - snd_printk(KERN_ERR "cs4231: can't grab IRQ %d\n", irq); - snd_cs4231_free(chip); - return -EBUSY; - } - chip->irq = irq; - if (!(hwshare & CS4231_HWSHARE_DMA1) && request_dma(dma1, "CS4231 - 1")) { - snd_printk(KERN_ERR "cs4231: can't grab DMA1 %d\n", dma1); - snd_cs4231_free(chip); - return -EBUSY; - } - chip->dma1 = dma1; - if (!(hwshare & CS4231_HWSHARE_DMA2) && dma1 != dma2 && dma2 >= 0 && request_dma(dma2, "CS4231 - 2")) { - snd_printk(KERN_ERR "cs4231: can't grab DMA2 %d\n", dma2); - snd_cs4231_free(chip); - return -EBUSY; - } - if (dma1 == dma2 || dma2 < 0) { - chip->single_dma = 1; - chip->dma2 = chip->dma1; - } else - chip->dma2 = dma2; - - /* global setup */ - if (snd_cs4231_probe(chip) < 0) { - snd_cs4231_free(chip); - return -ENODEV; - } - snd_cs4231_init(chip); - -#if 0 - if (chip->hardware & CS4231_HW_CS4232_MASK) { - if (chip->res_cport == NULL) - snd_printk("CS4232 control port features are not accessible\n"); - } -#endif - - /* Register device */ - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_cs4231_free(chip); - return err; - } - -#ifdef CONFIG_PM - /* Power Management */ - chip->suspend = snd_cs4231_suspend; - chip->resume = snd_cs4231_resume; -#endif - - *rchip = chip; - return 0; -} - -static struct snd_pcm_ops snd_cs4231_playback_ops = { - .open = snd_cs4231_playback_open, - .close = snd_cs4231_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_cs4231_playback_hw_params, - .hw_free = snd_cs4231_playback_hw_free, - .prepare = snd_cs4231_playback_prepare, - .trigger = snd_cs4231_trigger, - .pointer = snd_cs4231_playback_pointer, -}; - -static struct snd_pcm_ops snd_cs4231_capture_ops = { - .open = snd_cs4231_capture_open, - .close = snd_cs4231_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_cs4231_capture_hw_params, - .hw_free = snd_cs4231_capture_hw_free, - .prepare = snd_cs4231_capture_prepare, - .trigger = snd_cs4231_trigger, - .pointer = snd_cs4231_capture_pointer, -}; - -int snd_cs4231_pcm(struct snd_cs4231 *chip, int device, struct snd_pcm **rpcm) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(chip->card, "CS4231", device, 1, 1, &pcm)) < 0) - return err; - - spin_lock_init(&chip->reg_lock); - mutex_init(&chip->mce_mutex); - mutex_init(&chip->open_mutex); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4231_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cs4231_capture_ops); - - /* global setup */ - pcm->private_data = chip; - pcm->info_flags = 0; - if (chip->single_dma) - pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX; - if (chip->hardware != CS4231_HW_INTERWAVE) - pcm->info_flags |= SNDRV_PCM_INFO_JOINT_DUPLEX; - strcpy(pcm->name, snd_cs4231_chip_id(chip)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, chip->dma1 > 3 || chip->dma2 > 3 ? 128*1024 : 64*1024); - - chip->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -static void snd_cs4231_timer_free(struct snd_timer *timer) -{ - struct snd_cs4231 *chip = timer->private_data; - chip->timer = NULL; -} - -int snd_cs4231_timer(struct snd_cs4231 *chip, int device, struct snd_timer **rtimer) -{ - struct snd_timer *timer; - struct snd_timer_id tid; - int err; - - /* Timer initialization */ - tid.dev_class = SNDRV_TIMER_CLASS_CARD; - tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; - tid.card = chip->card->number; - tid.device = device; - tid.subdevice = 0; - if ((err = snd_timer_new(chip->card, "CS4231", &tid, &timer)) < 0) - return err; - strcpy(timer->name, snd_cs4231_chip_id(chip)); - timer->private_data = chip; - timer->private_free = snd_cs4231_timer_free; - timer->hw = snd_cs4231_timer_table; - chip->timer = timer; - if (rtimer) - *rtimer = timer; - return 0; -} - -/* - * MIXER part - */ - -static int snd_cs4231_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line", "Aux", "Mic", "Mix" - }; - static char *opl3sa_texts[4] = { - "Line", "CD", "Mic", "Mix" - }; - static char *gusmax_texts[4] = { - "Line", "Synth", "Mic", "Mix" - }; - char **ptexts = texts; - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - - snd_assert(chip->card != NULL, return -EINVAL); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - if (!strcmp(chip->card->driver, "GUS MAX")) - ptexts = gusmax_texts; - switch (chip->hardware) { - case CS4231_HW_INTERWAVE: ptexts = gusmax_texts; break; - case CS4231_HW_OPL3SA2: ptexts = opl3sa_texts; break; - } - strcpy(uinfo->value.enumerated.name, ptexts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_cs4231_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[CS4231_LEFT_INPUT] & CS4231_MIXS_ALL) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[CS4231_RIGHT_INPUT] & CS4231_MIXS_ALL) >> 6; - spin_unlock_irqrestore(&chip->reg_lock, flags); - return 0; -} - -static int snd_cs4231_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->reg_lock, flags); - left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left; - right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right; - change = left != chip->image[CS4231_LEFT_INPUT] || - right != chip->image[CS4231_RIGHT_INPUT]; - snd_cs4231_out(chip, CS4231_LEFT_INPUT, left); - snd_cs4231_out(chip, CS4231_RIGHT_INPUT, right); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->reg_lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_cs4231_out(chip, reg, val); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->reg_lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->reg_lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_cs4231_out(chip, left_reg, val1); - snd_cs4231_out(chip, right_reg, val2); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return change; -} - -static struct snd_kcontrol_new snd_cs4231_controls[] = { -CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), -CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -CS4231_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), -CS4231_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_cs4231_info_mux, - .get = snd_cs4231_get_mux, - .put = snd_cs4231_put_mux, -}, -CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) -}; - -static struct snd_kcontrol_new snd_opti93x_controls[] = { -CS4231_DOUBLE("Master Playback Switch", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -CS4231_DOUBLE("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), -CS4231_DOUBLE("PCM Playback Switch", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -CS4231_DOUBLE("FM Playback Switch", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Line Playback Switch", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -CS4231_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), -CS4231_DOUBLE("Mic Playback Switch", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -CS4231_DOUBLE("CD Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Aux Playback Switch", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -CS4231_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_cs4231_info_mux, - .get = snd_cs4231_get_mux, - .put = snd_cs4231_put_mux, -} -}; - -int snd_cs4231_mixer(struct snd_cs4231 *chip) -{ - struct snd_card *card; - unsigned int idx; - int err; - - snd_assert(chip != NULL && chip->pcm != NULL, return -EINVAL); - - card = chip->card; - - strcpy(card->mixername, chip->pcm->name); - - if (chip->hardware == CS4231_HW_OPTI93X) - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_opti93x_controls[idx], - chip)); - if (err < 0) - return err; - } - else - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_cs4231_controls[idx], - chip)); - if (err < 0) - return err; - } - return 0; -} - -EXPORT_SYMBOL(snd_cs4231_out); -EXPORT_SYMBOL(snd_cs4231_in); -EXPORT_SYMBOL(snd_cs4236_ext_out); -EXPORT_SYMBOL(snd_cs4236_ext_in); -EXPORT_SYMBOL(snd_cs4231_mce_up); -EXPORT_SYMBOL(snd_cs4231_mce_down); -EXPORT_SYMBOL(snd_cs4231_overrange); -EXPORT_SYMBOL(snd_cs4231_interrupt); -EXPORT_SYMBOL(snd_cs4231_chip_id); -EXPORT_SYMBOL(snd_cs4231_create); -EXPORT_SYMBOL(snd_cs4231_pcm); -EXPORT_SYMBOL(snd_cs4231_mixer); -EXPORT_SYMBOL(snd_cs4231_timer); -EXPORT_SYMBOL(snd_cs4231_info_single); -EXPORT_SYMBOL(snd_cs4231_get_single); -EXPORT_SYMBOL(snd_cs4231_put_single); -EXPORT_SYMBOL(snd_cs4231_info_double); -EXPORT_SYMBOL(snd_cs4231_get_double); -EXPORT_SYMBOL(snd_cs4231_put_double); - -/* - * INIT part - */ - -static int __init alsa_cs4231_init(void) -{ - return 0; -} - -static void __exit alsa_cs4231_exit(void) -{ -} - -module_init(alsa_cs4231_init) -module_exit(alsa_cs4231_exit) diff -urNp linux-alsa/sound/isa/wss/Makefile linux-mm/sound/isa/wss/Makefile --- linux-alsa/sound/isa/wss/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ linux-mm/sound/isa/wss/Makefile 2008-06-15 22:38:44.705642447 +0200 @@ -0,0 +1,10 @@ +# +# Makefile for ALSA +# Copyright (c) 2008 by Jaroslav Kysela <perex(a)perex.cz> +# + +snd-wss-lib-objs := wss_lib.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_WSS_LIB) += snd-wss-lib.o + diff -urNp linux-alsa/sound/isa/wss/wss_lib.c linux-mm/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-mm/sound/isa/wss/wss_lib.c 2008-06-15 22:38:10.265643406 +0200 @@ -0,0 +1,1945 @@ +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Routines for control of CS4231(A)/CS4232/InterWave & compatible chips + * + * Bugs: + * - sometimes record brokes playback with WSS portion of + * Yamaha OPL3-SA3 chip + * - CS4231 (GUS MAX) - still trouble with occasional noises + * - broken initialization? + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/slab.h> +#include <linux/ioport.h> +#include <sound/core.h> +#include <sound/cs4231.h> +#include <sound/pcm_params.h> + +#include <asm/io.h> +#include <asm/dma.h> +#include <asm/irq.h> + +MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); +MODULE_DESCRIPTION("Routines for control of CS4231(A)/CS4232/InterWave & compatible chips"); +MODULE_LICENSE("GPL"); + +#if 0 +#define SNDRV_DEBUG_MCE +#endif + +/* + * Some variables + */ + +static unsigned char freq_bits[14] = { + /* 5510 */ 0x00 | CS4231_XTAL2, + /* 6620 */ 0x0E | CS4231_XTAL2, + /* 8000 */ 0x00 | CS4231_XTAL1, + /* 9600 */ 0x0E | CS4231_XTAL1, + /* 11025 */ 0x02 | CS4231_XTAL2, + /* 16000 */ 0x02 | CS4231_XTAL1, + /* 18900 */ 0x04 | CS4231_XTAL2, + /* 22050 */ 0x06 | CS4231_XTAL2, + /* 27042 */ 0x04 | CS4231_XTAL1, + /* 32000 */ 0x06 | CS4231_XTAL1, + /* 33075 */ 0x0C | CS4231_XTAL2, + /* 37800 */ 0x08 | CS4231_XTAL2, + /* 44100 */ 0x0A | CS4231_XTAL2, + /* 48000 */ 0x0C | CS4231_XTAL1 +}; + +static unsigned int rates[14] = { + 5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050, + 27042, 32000, 33075, 37800, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static int snd_cs4231_xrate(struct snd_pcm_runtime *runtime) +{ + return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); +} + +static unsigned char snd_cs4231_original_image[32] = +{ + 0x00, /* 00/00 - lic */ + 0x00, /* 01/01 - ric */ + 0x9f, /* 02/02 - la1ic */ + 0x9f, /* 03/03 - ra1ic */ + 0x9f, /* 04/04 - la2ic */ + 0x9f, /* 05/05 - ra2ic */ + 0xbf, /* 06/06 - loc */ + 0xbf, /* 07/07 - roc */ + 0x20, /* 08/08 - pdfr */ + CS4231_AUTOCALIB, /* 09/09 - ic */ + 0x00, /* 0a/10 - pc */ + 0x00, /* 0b/11 - ti */ + CS4231_MODE2, /* 0c/12 - mi */ + 0xfc, /* 0d/13 - lbc */ + 0x00, /* 0e/14 - pbru */ + 0x00, /* 0f/15 - pbrl */ + 0x80, /* 10/16 - afei */ + 0x01, /* 11/17 - afeii */ + 0x9f, /* 12/18 - llic */ + 0x9f, /* 13/19 - rlic */ + 0x00, /* 14/20 - tlb */ + 0x00, /* 15/21 - thb */ + 0x00, /* 16/22 - la3mic/reserved */ + 0x00, /* 17/23 - ra3mic/reserved */ + 0x00, /* 18/24 - afs */ + 0x00, /* 19/25 - lamoc/version */ + 0xcf, /* 1a/26 - mioc */ + 0x00, /* 1b/27 - ramoc/reserved */ + 0x20, /* 1c/28 - cdfr */ + 0x00, /* 1d/29 - res4 */ + 0x00, /* 1e/30 - cbru */ + 0x00, /* 1f/31 - cbrl */ +}; + +static unsigned char snd_opti93x_original_image[32] = +{ + 0x00, /* 00/00 - l_mixout_outctrl */ + 0x00, /* 01/01 - r_mixout_outctrl */ + 0x88, /* 02/02 - l_cd_inctrl */ + 0x88, /* 03/03 - r_cd_inctrl */ + 0x88, /* 04/04 - l_a1/fm_inctrl */ + 0x88, /* 05/05 - r_a1/fm_inctrl */ + 0x80, /* 06/06 - l_dac_inctrl */ + 0x80, /* 07/07 - r_dac_inctrl */ + 0x00, /* 08/08 - ply_dataform_reg */ + 0x00, /* 09/09 - if_conf */ + 0x00, /* 0a/10 - pin_ctrl */ + 0x00, /* 0b/11 - err_init_reg */ + 0x0a, /* 0c/12 - id_reg */ + 0x00, /* 0d/13 - reserved */ + 0x00, /* 0e/14 - ply_upcount_reg */ + 0x00, /* 0f/15 - ply_lowcount_reg */ + 0x88, /* 10/16 - reserved/l_a1_inctrl */ + 0x88, /* 11/17 - reserved/r_a1_inctrl */ + 0x88, /* 12/18 - l_line_inctrl */ + 0x88, /* 13/19 - r_line_inctrl */ + 0x88, /* 14/20 - l_mic_inctrl */ + 0x88, /* 15/21 - r_mic_inctrl */ + 0x80, /* 16/22 - l_out_outctrl */ + 0x80, /* 17/23 - r_out_outctrl */ + 0x00, /* 18/24 - reserved */ + 0x00, /* 19/25 - reserved */ + 0x00, /* 1a/26 - reserved */ + 0x00, /* 1b/27 - reserved */ + 0x00, /* 1c/28 - cap_dataform_reg */ + 0x00, /* 1d/29 - reserved */ + 0x00, /* 1e/30 - cap_upcount_reg */ + 0x00 /* 1f/31 - cap_lowcount_reg */ +}; + +/* + * Basic I/O functions + */ + +static inline void cs4231_outb(struct snd_cs4231 *chip, u8 offset, u8 val) +{ + outb(val, chip->port + offset); +} + +static inline u8 cs4231_inb(struct snd_cs4231 *chip, u8 offset) +{ + return inb(chip->port + offset); +} + +static void snd_cs4231_wait(struct snd_cs4231 *chip) +{ + int timeout; + + for (timeout = 250; + timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout--) + udelay(100); +} + +static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, + unsigned char mask, unsigned char value) +{ + unsigned char tmp = (chip->image[reg] & mask) | value; + + snd_cs4231_wait(chip); +#ifdef CONFIG_SND_DEBUG + if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); +#endif + chip->image[reg] = tmp; + if (!chip->calibrate_mute) { + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + wmb(); + cs4231_outb(chip, CS4231P(REG), tmp); + mb(); + } +} + +static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +{ + int timeout; + + for (timeout = 250; + timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout--) + udelay(10); + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + cs4231_outb(chip, CS4231P(REG), value); + mb(); +} + +void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +{ + snd_cs4231_wait(chip); +#ifdef CONFIG_SND_DEBUG + if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); +#endif + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + cs4231_outb(chip, CS4231P(REG), value); + chip->image[reg] = value; + mb(); + snd_printdd("codec out - reg 0x%x = 0x%x\n", + chip->mce_bit | reg, value); +} + +unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) +{ + snd_cs4231_wait(chip); +#ifdef CONFIG_SND_DEBUG + if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); +#endif + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); + mb(); + return cs4231_inb(chip, CS4231P(REG)); +} + +void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val) +{ + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); + cs4231_outb(chip, CS4231P(REG), reg | (chip->image[CS4236_EXT_REG] & 0x01)); + cs4231_outb(chip, CS4231P(REG), val); + chip->eimage[CS4236_REG(reg)] = val; +#if 0 + printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val); +#endif +} + +unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg) +{ + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | 0x17); + cs4231_outb(chip, CS4231P(REG), reg | (chip->image[CS4236_EXT_REG] & 0x01)); +#if 1 + return cs4231_inb(chip, CS4231P(REG)); +#else + { + unsigned char res; + res = cs4231_inb(chip, CS4231P(REG)); + printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res); + return res; + } +#endif +} + +#if 0 + +static void snd_cs4231_debug(struct snd_cs4231 *chip) +{ + printk("CS4231 REGS: INDEX = 0x%02x ", cs4231_inb(chip, CS4231P(REGSEL))); + printk(" STATUS = 0x%02x\n", cs4231_inb(chip, CS4231P(STATUS))); + printk(" 0x00: left input = 0x%02x ", snd_cs4231_in(chip, 0x00)); + printk(" 0x10: alt 1 (CFIG 2) = 0x%02x\n", snd_cs4231_in(chip, 0x10)); + printk(" 0x01: right input = 0x%02x ", snd_cs4231_in(chip, 0x01)); + printk(" 0x11: alt 2 (CFIG 3) = 0x%02x\n", snd_cs4231_in(chip, 0x11)); + printk(" 0x02: GF1 left input = 0x%02x ", snd_cs4231_in(chip, 0x02)); + printk(" 0x12: left line in = 0x%02x\n", snd_cs4231_in(chip, 0x12)); + printk(" 0x03: GF1 right input = 0x%02x ", snd_cs4231_in(chip, 0x03)); + printk(" 0x13: right line in = 0x%02x\n", snd_cs4231_in(chip, 0x13)); + printk(" 0x04: CD left input = 0x%02x ", snd_cs4231_in(chip, 0x04)); + printk(" 0x14: timer low = 0x%02x\n", snd_cs4231_in(chip, 0x14)); + printk(" 0x05: CD right input = 0x%02x ", snd_cs4231_in(chip, 0x05)); + printk(" 0x15: timer high = 0x%02x\n", snd_cs4231_in(chip, 0x15)); + printk(" 0x06: left output = 0x%02x ", snd_cs4231_in(chip, 0x06)); + printk(" 0x16: left MIC (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x16)); + printk(" 0x07: right output = 0x%02x ", snd_cs4231_in(chip, 0x07)); + printk(" 0x17: right MIC (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x17)); + printk(" 0x08: playback format = 0x%02x ", snd_cs4231_in(chip, 0x08)); + printk(" 0x18: IRQ status = 0x%02x\n", snd_cs4231_in(chip, 0x18)); + printk(" 0x09: iface (CFIG 1) = 0x%02x ", snd_cs4231_in(chip, 0x09)); + printk(" 0x19: left line out = 0x%02x\n", snd_cs4231_in(chip, 0x19)); + printk(" 0x0a: pin control = 0x%02x ", snd_cs4231_in(chip, 0x0a)); + printk(" 0x1a: mono control = 0x%02x\n", snd_cs4231_in(chip, 0x1a)); + printk(" 0x0b: init & status = 0x%02x ", snd_cs4231_in(chip, 0x0b)); + printk(" 0x1b: right line out = 0x%02x\n", snd_cs4231_in(chip, 0x1b)); + printk(" 0x0c: revision & mode = 0x%02x ", snd_cs4231_in(chip, 0x0c)); + printk(" 0x1c: record format = 0x%02x\n", snd_cs4231_in(chip, 0x1c)); + printk(" 0x0d: loopback = 0x%02x ", snd_cs4231_in(chip, 0x0d)); + printk(" 0x1d: var freq (PnP) = 0x%02x\n", snd_cs4231_in(chip, 0x1d)); + printk(" 0x0e: ply upr count = 0x%02x ", snd_cs4231_in(chip, 0x0e)); + printk(" 0x1e: ply lwr count = 0x%02x\n", snd_cs4231_in(chip, 0x1e)); + printk(" 0x0f: rec upr count = 0x%02x ", snd_cs4231_in(chip, 0x0f)); + printk(" 0x1f: rec lwr count = 0x%02x\n", snd_cs4231_in(chip, 0x1f)); +} + +#endif + +/* + * CS4231 detection / MCE routines + */ + +static void snd_cs4231_busy_wait(struct snd_cs4231 *chip) +{ + int timeout; + + /* huh.. looks like this sequence is proper for CS4231A chip (GUS MAX) */ + for (timeout = 5; timeout > 0; timeout--) + cs4231_inb(chip, CS4231P(REGSEL)); + /* end of cleanup sequence */ + for (timeout = 250; + timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout--) + udelay(10); +} + +void snd_cs4231_mce_up(struct snd_cs4231 *chip) +{ + unsigned long flags; + int timeout; + + snd_cs4231_wait(chip); +#ifdef CONFIG_SND_DEBUG + if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + snd_printk("mce_up - auto calibration time out (0)\n"); +#endif + spin_lock_irqsave(&chip->reg_lock, flags); + chip->mce_bit |= CS4231_MCE; + timeout = cs4231_inb(chip, CS4231P(REGSEL)); + if (timeout == 0x80) + snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); + if (!(timeout & CS4231_MCE)) + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); + spin_unlock_irqrestore(&chip->reg_lock, flags); +} + +void snd_cs4231_mce_down(struct snd_cs4231 *chip) +{ + unsigned long flags; + unsigned long end_time; + int timeout; + + snd_cs4231_busy_wait(chip); + +#ifdef CONFIG_SND_DEBUG + if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); +#endif + spin_lock_irqsave(&chip->reg_lock, flags); + chip->mce_bit &= ~CS4231_MCE; + timeout = cs4231_inb(chip, CS4231P(REGSEL)); + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (timeout == 0x80) + snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); + if ((timeout & CS4231_MCE) == 0 || + !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { + return; + } + + /* + * Wait for (possible -- during init auto-calibration may not be set) + * calibration process to start. Needs upto 5 sample periods on AD1848 + * which at the slowest possible rate of 5.5125 kHz means 907 us. + */ + msleep(1); + + snd_printdd("(1) jiffies = %lu\n", jiffies); + + /* check condition up to 250 ms */ + end_time = jiffies + msecs_to_jiffies(250); + while (snd_cs4231_in(chip, CS4231_TEST_INIT) & + CS4231_CALIB_IN_PROGRESS) { + + if (time_after(jiffies, end_time)) { + snd_printk(KERN_ERR "mce_down - " + "auto calibration time out (2)\n"); + return; + } + msleep(1); + } + + snd_printdd("(2) jiffies = %lu\n", jiffies); + + /* check condition up to 100 ms */ + end_time = jiffies + msecs_to_jiffies(100); + while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { + if (time_after(jiffies, end_time)) { + snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); + return; + } + msleep(1); + } + + snd_printdd("(3) jiffies = %lu\n", jiffies); + snd_printd("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); +} + +static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size) +{ + switch (format & 0xe0) { + case CS4231_LINEAR_16: + case CS4231_LINEAR_16_BIG: + size >>= 1; + break; + case CS4231_ADPCM_16: + return size >> 2; + } + if (format & CS4231_STEREO) + size >>= 1; + return size; +} + +static int snd_cs4231_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + int result = 0; + unsigned int what; + struct snd_pcm_substream *s; + int do_start; + +#if 0 + printk("codec trigger!!! - what = %i, enable = %i, status = 0x%x\n", what, enable, cs4231_inb(chip, CS4231P(STATUS))); +#endif + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + do_start = 1; break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + do_start = 0; break; + default: + return -EINVAL; + } + + what = 0; + snd_pcm_group_for_each_entry(s, substream) { + if (s == chip->playback_substream) { + what |= CS4231_PLAYBACK_ENABLE; + snd_pcm_trigger_done(s, substream); + } else if (s == chip->capture_substream) { + what |= CS4231_RECORD_ENABLE; + snd_pcm_trigger_done(s, substream); + } + } + spin_lock(&chip->reg_lock); + if (do_start) { + chip->image[CS4231_IFACE_CTRL] |= what; + if (chip->trigger) + chip->trigger(chip, what, 1); + } else { + chip->image[CS4231_IFACE_CTRL] &= ~what; + if (chip->trigger) + chip->trigger(chip, what, 0); + } + snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + spin_unlock(&chip->reg_lock); +#if 0 + snd_cs4231_debug(chip); +#endif + return result; +} + +/* + * CODEC I/O + */ + +static unsigned char snd_cs4231_get_rate(unsigned int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rates); i++) + if (rate == rates[i]) + return freq_bits[i]; + // snd_BUG(); + return freq_bits[ARRAY_SIZE(rates) - 1]; +} + +static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip, + int format, + int channels) +{ + unsigned char rformat; + + rformat = CS4231_LINEAR_8; + switch (format) { + case SNDRV_PCM_FORMAT_MU_LAW: rformat = CS4231_ULAW_8; break; + case SNDRV_PCM_FORMAT_A_LAW: rformat = CS4231_ALAW_8; break; + case SNDRV_PCM_FORMAT_S16_LE: rformat = CS4231_LINEAR_16; break; + case SNDRV_PCM_FORMAT_S16_BE: rformat = CS4231_LINEAR_16_BIG; break; + case SNDRV_PCM_FORMAT_IMA_ADPCM: rformat = CS4231_ADPCM_16; break; + } + if (channels > 1) + rformat |= CS4231_STEREO; +#if 0 + snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); +#endif + return rformat; +} + +static void snd_cs4231_calibrate_mute(struct snd_cs4231 *chip, int mute) +{ + unsigned long flags; + + mute = mute ? 1 : 0; + spin_lock_irqsave(&chip->reg_lock, flags); + if (chip->calibrate_mute == mute) { + spin_unlock_irqrestore(&chip->reg_lock, flags); + return; + } + if (!mute) { + snd_cs4231_dout(chip, CS4231_LEFT_INPUT, chip->image[CS4231_LEFT_INPUT]); + snd_cs4231_dout(chip, CS4231_RIGHT_INPUT, chip->image[CS4231_RIGHT_INPUT]); + snd_cs4231_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); + } + snd_cs4231_dout(chip, CS4231_AUX1_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); + snd_cs4231_dout(chip, CS4231_AUX1_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); + snd_cs4231_dout(chip, CS4231_AUX2_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); + snd_cs4231_dout(chip, CS4231_AUX2_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); + snd_cs4231_dout(chip, CS4231_LEFT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); + snd_cs4231_dout(chip, CS4231_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + snd_cs4231_dout(chip, CS4231_LEFT_LINE_IN, mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); + snd_cs4231_dout(chip, CS4231_RIGHT_LINE_IN, mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); + snd_cs4231_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); + if (chip->hardware == CS4231_HW_INTERWAVE) { + snd_cs4231_dout(chip, CS4231_LEFT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_LEFT_MIC_INPUT]); + snd_cs4231_dout(chip, CS4231_RIGHT_MIC_INPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_MIC_INPUT]); + snd_cs4231_dout(chip, CS4231_LINE_LEFT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LINE_LEFT_OUTPUT]); + snd_cs4231_dout(chip, CS4231_LINE_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LINE_RIGHT_OUTPUT]); + } + chip->calibrate_mute = mute; + spin_unlock_irqrestore(&chip->reg_lock, flags); +} + +static void snd_cs4231_playback_format(struct snd_cs4231 *chip, + struct snd_pcm_hw_params *params, + unsigned char pdfr) +{ + unsigned long flags; + int full_calib = 1; + + mutex_lock(&chip->mce_mutex); + snd_cs4231_calibrate_mute(chip, 1); + if (chip->hardware == CS4231_HW_CS4231A || + (chip->hardware & CS4231_HW_CS4232_MASK)) { + spin_lock_irqsave(&chip->reg_lock, flags); + if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (pdfr & 0x0f)) { /* rate is same? */ + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x10); + snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x10); + udelay(100); /* Fixes audible clicks at least on GUS MAX */ + full_calib = 0; + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + } + if (full_calib) { + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + if (chip->hardware != CS4231_HW_INTERWAVE && !chip->single_dma) { + snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, + (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ? + (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) : + pdfr); + } else { + snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (chip->hardware == CS4231_HW_OPL3SA2) + udelay(100); /* this seems to help */ + snd_cs4231_mce_down(chip); + } + snd_cs4231_calibrate_mute(chip, 0); + mutex_unlock(&chip->mce_mutex); +} + +static void snd_cs4231_capture_format(struct snd_cs4231 *chip, + struct snd_pcm_hw_params *params, + unsigned char cdfr) +{ + unsigned long flags; + int full_calib = 1; + + mutex_lock(&chip->mce_mutex); + snd_cs4231_calibrate_mute(chip, 1); + if (chip->hardware == CS4231_HW_CS4231A || + (chip->hardware & CS4231_HW_CS4232_MASK)) { + spin_lock_irqsave(&chip->reg_lock, flags); + if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (cdfr & 0x0f) || /* rate is same? */ + (chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x20); + snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT] = cdfr); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); + full_calib = 0; + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + } + if (full_calib) { + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + if (chip->hardware != CS4231_HW_INTERWAVE) { + if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { + snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, + ((chip->single_dma ? cdfr : chip->image[CS4231_PLAYBK_FORMAT]) & 0xf0) | + (cdfr & 0x0f)); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + } + } + snd_cs4231_out(chip, CS4231_REC_FORMAT, cdfr); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); + } + snd_cs4231_calibrate_mute(chip, 0); + mutex_unlock(&chip->mce_mutex); +} + +/* + * Timer interface + */ + +static unsigned long snd_cs4231_timer_resolution(struct snd_timer * timer) +{ + struct snd_cs4231 *chip = snd_timer_chip(timer); + if (chip->hardware & CS4231_HW_CS4236B_MASK) + return 14467; + else + return chip->image[CS4231_PLAYBK_FORMAT] & 1 ? 9969 : 9920; +} + +static int snd_cs4231_timer_start(struct snd_timer * timer) +{ + unsigned long flags; + unsigned int ticks; + struct snd_cs4231 *chip = snd_timer_chip(timer); + spin_lock_irqsave(&chip->reg_lock, flags); + ticks = timer->sticks; + if ((chip->image[CS4231_ALT_FEATURE_1] & CS4231_TIMER_ENABLE) == 0 || + (unsigned char)(ticks >> 8) != chip->image[CS4231_TIMER_HIGH] || + (unsigned char)ticks != chip->image[CS4231_TIMER_LOW]) { + snd_cs4231_out(chip, CS4231_TIMER_HIGH, chip->image[CS4231_TIMER_HIGH] = (unsigned char) (ticks >> 8)); + snd_cs4231_out(chip, CS4231_TIMER_LOW, chip->image[CS4231_TIMER_LOW] = (unsigned char) ticks); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | CS4231_TIMER_ENABLE); + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + return 0; +} + +static int snd_cs4231_timer_stop(struct snd_timer * timer) +{ + unsigned long flags; + struct snd_cs4231 *chip = snd_timer_chip(timer); + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE); + spin_unlock_irqrestore(&chip->reg_lock, flags); + return 0; +} + +static void snd_cs4231_init(struct snd_cs4231 *chip) +{ + unsigned long flags; + + snd_cs4231_mce_down(chip); + +#ifdef SNDRV_DEBUG_MCE + snd_printk("init: (1)\n"); +#endif + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | + CS4231_RECORD_ENABLE | CS4231_RECORD_PIO | + CS4231_CALIB_MODE); + chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; + snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); + +#ifdef SNDRV_DEBUG_MCE + snd_printk("init: (2)\n"); +#endif + + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); + +#ifdef SNDRV_DEBUG_MCE + snd_printk("init: (3) - afei = 0x%x\n", chip->image[CS4231_ALT_FEATURE_1]); +#endif + + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_2, chip->image[CS4231_ALT_FEATURE_2]); + spin_unlock_irqrestore(&chip->reg_lock, flags); + + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT]); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); + +#ifdef SNDRV_DEBUG_MCE + snd_printk("init: (4)\n"); +#endif + + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT]); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); + +#ifdef SNDRV_DEBUG_MCE + snd_printk("init: (5)\n"); +#endif +} + +static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode) +{ + unsigned long flags; + + mutex_lock(&chip->open_mutex); + if ((chip->mode & mode) || + ((chip->mode & CS4231_MODE_OPEN) && chip->single_dma)) { + mutex_unlock(&chip->open_mutex); + return -EAGAIN; + } + if (chip->mode & CS4231_MODE_OPEN) { + chip->mode |= mode; + mutex_unlock(&chip->open_mutex); + return 0; + } + /* ok. now enable and ack CODEC IRQ */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | + CS4231_RECORD_IRQ | + CS4231_TIMER_IRQ); + snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); + cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + chip->image[CS4231_PIN_CTRL] |= CS4231_IRQ_ENABLE; + snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); + snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | + CS4231_RECORD_IRQ | + CS4231_TIMER_IRQ); + snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); + spin_unlock_irqrestore(&chip->reg_lock, flags); + + chip->mode = mode; + mutex_unlock(&chip->open_mutex); + return 0; +} + +static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode) +{ + unsigned long flags; + + mutex_lock(&chip->open_mutex); + chip->mode &= ~mode; + if (chip->mode & CS4231_MODE_OPEN) { + mutex_unlock(&chip->open_mutex); + return; + } + snd_cs4231_calibrate_mute(chip, 1); + + /* disable IRQ */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); + cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + chip->image[CS4231_PIN_CTRL] &= ~CS4231_IRQ_ENABLE; + snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); + + /* now disable record & playback */ + + if (chip->image[CS4231_IFACE_CTRL] & (CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | + CS4231_RECORD_ENABLE | CS4231_RECORD_PIO)) { + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | + CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); + snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_down(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + } + + /* clear IRQ again */ + snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); + cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ + spin_unlock_irqrestore(&chip->reg_lock, flags); + + snd_cs4231_calibrate_mute(chip, 0); + + chip->mode = 0; + mutex_unlock(&chip->open_mutex); +} + +/* + * timer open/close + */ + +static int snd_cs4231_timer_open(struct snd_timer * timer) +{ + struct snd_cs4231 *chip = snd_timer_chip(timer); + snd_cs4231_open(chip, CS4231_MODE_TIMER); + return 0; +} + +static int snd_cs4231_timer_close(struct snd_timer * timer) +{ + struct snd_cs4231 *chip = snd_timer_chip(timer); + snd_cs4231_close(chip, CS4231_MODE_TIMER); + return 0; +} + +static struct snd_timer_hardware snd_cs4231_timer_table = +{ + .flags = SNDRV_TIMER_HW_AUTO, + .resolution = 9945, + .ticks = 65535, + .open = snd_cs4231_timer_open, + .close = snd_cs4231_timer_close, + .c_resolution = snd_cs4231_timer_resolution, + .start = snd_cs4231_timer_start, + .stop = snd_cs4231_timer_stop, +}; + +/* + * ok.. exported functions.. + */ + +static int snd_cs4231_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + unsigned char new_pdfr; + int err; + + if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) + return err; + new_pdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | + snd_cs4231_get_rate(params_rate(hw_params)); + chip->set_playback_format(chip, hw_params, new_pdfr); + return 0; +} + +static int snd_cs4231_playback_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_cs4231_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + spin_lock_irqsave(&chip->reg_lock, flags); + chip->p_dma_size = size; + chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO); + snd_dma_program(chip->dma1, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); + count = snd_cs4231_get_count(chip->image[CS4231_PLAYBK_FORMAT], count) - 1; + snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); + snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); + spin_unlock_irqrestore(&chip->reg_lock, flags); +#if 0 + snd_cs4231_debug(chip); +#endif + return 0; +} + +static int snd_cs4231_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + unsigned char new_cdfr; + int err; + + if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) + return err; + new_cdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | + snd_cs4231_get_rate(params_rate(hw_params)); + chip->set_capture_format(chip, hw_params, new_cdfr); + return 0; +} + +static int snd_cs4231_capture_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long flags; + unsigned int size = snd_pcm_lib_buffer_bytes(substream); + unsigned int count = snd_pcm_lib_period_bytes(substream); + + spin_lock_irqsave(&chip->reg_lock, flags); + chip->c_dma_size = size; + chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); + snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); + count = snd_cs4231_get_count(chip->image[CS4231_REC_FORMAT], count) - 1; + if (chip->single_dma && chip->hardware != CS4231_HW_INTERWAVE) { + snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); + snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); + } else { + snd_cs4231_out(chip, CS4231_REC_LWR_CNT, (unsigned char) count); + snd_cs4231_out(chip, CS4231_REC_UPR_CNT, (unsigned char) (count >> 8)); + } + spin_unlock_irqrestore(&chip->reg_lock, flags); + return 0; +} + +void snd_cs4231_overrange(struct snd_cs4231 *chip) +{ + unsigned long flags; + unsigned char res; + + spin_lock_irqsave(&chip->reg_lock, flags); + res = snd_cs4231_in(chip, CS4231_TEST_INIT); + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (res & (0x08 | 0x02)) /* detect overrange only above 0dB; may be user selectable? */ + chip->capture_substream->runtime->overrange++; +} + +irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id) +{ + struct snd_cs4231 *chip = dev_id; + unsigned char status; + + status = snd_cs4231_in(chip, CS4231_IRQ_STATUS); + if (status & CS4231_TIMER_IRQ) { + if (chip->timer) + snd_timer_interrupt(chip->timer, chip->timer->sticks); + } + if (chip->single_dma && chip->hardware != CS4231_HW_INTERWAVE) { + if (status & CS4231_PLAYBACK_IRQ) { + if (chip->mode & CS4231_MODE_PLAY) { + if (chip->playback_substream) + snd_pcm_period_elapsed(chip->playback_substream); + } + if (chip->mode & CS4231_MODE_RECORD) { + if (chip->capture_substream) { + snd_cs4231_overrange(chip); + snd_pcm_period_elapsed(chip->capture_substream); + } + } + } + } else { + if (status & CS4231_PLAYBACK_IRQ) { + if (chip->playback_substream) + snd_pcm_period_elapsed(chip->playback_substream); + } + if (status & CS4231_RECORD_IRQ) { + if (chip->capture_substream) { + snd_cs4231_overrange(chip); + snd_pcm_period_elapsed(chip->capture_substream); + } + } + } + + spin_lock(&chip->reg_lock); + snd_cs4231_outm(chip, CS4231_IRQ_STATUS, ~CS4231_ALL_IRQS | ~status, 0); + spin_unlock(&chip->reg_lock); + return IRQ_HANDLED; +} + +static snd_pcm_uframes_t snd_cs4231_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + size_t ptr; + + if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) + return 0; + ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); + return bytes_to_frames(substream->runtime, ptr); +} + +static snd_pcm_uframes_t snd_cs4231_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + size_t ptr; + + if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE)) + return 0; + ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); + return bytes_to_frames(substream->runtime, ptr); +} + +/* + + */ + +static int snd_cs4231_probe(struct snd_cs4231 *chip) +{ + unsigned long flags; + int i, id, rev; + unsigned char *ptr; + unsigned int hw; + +#if 0 + snd_cs4231_debug(chip); +#endif + id = 0; + for (i = 0; i < 50; i++) { + mb(); + if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + udelay(2000); + else { + spin_lock_irqsave(&chip->reg_lock, flags); + snd_cs4231_out(chip, CS4231_MISC_INFO, CS4231_MODE2); + id = snd_cs4231_in(chip, CS4231_MISC_INFO) & 0x0f; + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (id == 0x0a) + break; /* this is valid value */ + } + } + snd_printdd("cs4231: port = 0x%lx, id = 0x%x\n", chip->port, id); + if (id != 0x0a) + return -ENODEV; /* no valid device found */ + + if (((hw = chip->hardware) & CS4231_HW_TYPE_MASK) == CS4231_HW_DETECT) { + rev = snd_cs4231_in(chip, CS4231_VERSION) & 0xe7; + snd_printdd("CS4231: VERSION (I25) = 0x%x\n", rev); + if (rev == 0x80) { + unsigned char tmp = snd_cs4231_in(chip, 23); + snd_cs4231_out(chip, 23, ~tmp); + if (snd_cs4231_in(chip, 23) != tmp) + chip->hardware = CS4231_HW_AD1845; + else + chip->hardware = CS4231_HW_CS4231; + } else if (rev == 0xa0) { + chip->hardware = CS4231_HW_CS4231A; + } else if (rev == 0xa2) { + chip->hardware = CS4231_HW_CS4232; + } else if (rev == 0xb2) { + chip->hardware = CS4231_HW_CS4232A; + } else if (rev == 0x83) { + chip->hardware = CS4231_HW_CS4236; + } else if (rev == 0x03) { + chip->hardware = CS4231_HW_CS4236B; + } else { + snd_printk("unknown CS chip with version 0x%x\n", rev); + return -ENODEV; /* unknown CS4231 chip? */ + } + } + spin_lock_irqsave(&chip->reg_lock, flags); + cs4231_inb(chip, CS4231P(STATUS)); /* clear any pendings IRQ */ + cs4231_outb(chip, CS4231P(STATUS), 0); + mb(); + spin_unlock_irqrestore(&chip->reg_lock, flags); + + chip->image[CS4231_MISC_INFO] = CS4231_MODE2; + switch (chip->hardware) { + case CS4231_HW_INTERWAVE: + chip->image[CS4231_MISC_INFO] = CS4231_IW_MODE3; + break; + case CS4231_HW_CS4235: + case CS4231_HW_CS4236B: + case CS4231_HW_CS4237B: + case CS4231_HW_CS4238B: + case CS4231_HW_CS4239: + if (hw == CS4231_HW_DETECT3) + chip->image[CS4231_MISC_INFO] = CS4231_4236_MODE3; + else + chip->hardware = CS4231_HW_CS4236; + break; + } + + chip->image[CS4231_IFACE_CTRL] = + (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | + (chip->single_dma ? CS4231_SINGLE_DMA : 0); + if (chip->hardware != CS4231_HW_OPTI93X) { + chip->image[CS4231_ALT_FEATURE_1] = 0x80; + chip->image[CS4231_ALT_FEATURE_2] = + chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + } + ptr = (unsigned char *) &chip->image; + snd_cs4231_mce_down(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + for (i = 0; i < 32; i++) /* ok.. fill all CS4231 registers */ + snd_cs4231_out(chip, i, *ptr++); + spin_unlock_irqrestore(&chip->reg_lock, flags); + snd_cs4231_mce_up(chip); + snd_cs4231_mce_down(chip); + + mdelay(2); + + /* ok.. try check hardware version for CS4236+ chips */ + if ((hw & CS4231_HW_TYPE_MASK) == CS4231_HW_DETECT) { + if (chip->hardware == CS4231_HW_CS4236B) { + rev = snd_cs4236_ext_in(chip, CS4236_VERSION); + snd_cs4236_ext_out(chip, CS4236_VERSION, 0xff); + id = snd_cs4236_ext_in(chip, CS4236_VERSION); + snd_cs4236_ext_out(chip, CS4236_VERSION, rev); + snd_printdd("CS4231: ext version; rev = 0x%x, id = 0x%x\n", rev, id); + if ((id & 0x1f) == 0x1d) { /* CS4235 */ + chip->hardware = CS4231_HW_CS4235; + switch (id >> 5) { + case 4: + case 5: + case 6: + break; + default: + snd_printk("unknown CS4235 chip (enhanced version = 0x%x)\n", id); + } + } else if ((id & 0x1f) == 0x0b) { /* CS4236/B */ + switch (id >> 5) { + case 4: + case 5: + case 6: + case 7: + chip->hardware = CS4231_HW_CS4236B; + break; + default: + snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id); + } + } else if ((id & 0x1f) == 0x08) { /* CS4237B */ + chip->hardware = CS4231_HW_CS4237B; + switch (id >> 5) { + case 4: + case 5: + case 6: + case 7: + break; + default: + snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id); + } + } else if ((id & 0x1f) == 0x09) { /* CS4238B */ + chip->hardware = CS4231_HW_CS4238B; + switch (id >> 5) { + case 5: + case 6: + case 7: + break; + default: + snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id); + } + } else if ((id & 0x1f) == 0x1e) { /* CS4239 */ + chip->hardware = CS4231_HW_CS4239; + switch (id >> 5) { + case 4: + case 5: + case 6: + break; + default: + snd_printk("unknown CS4239 chip (enhanced version = 0x%x)\n", id); + } + } else { + snd_printk("unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", id); + } + } + } + return 0; /* all things are ok.. */ +} + +/* + + */ + +static struct snd_pcm_hardware snd_cs4231_playback = +{ + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 5510, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_cs4231_capture = +{ + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), + .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, + .rate_min = 5510, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = 64, + .period_bytes_max = (128*1024), + .periods_min = 1, + .periods_max = 1024, + .fifo_size = 0, +}; + +/* + + */ + +static int snd_cs4231_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + runtime->hw = snd_cs4231_playback; + + /* hardware bug in InterWave chipset */ + if (chip->hardware == CS4231_HW_INTERWAVE && chip->dma1 > 3) + runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_MU_LAW; + + /* hardware limitation of cheap chips */ + if (chip->hardware == CS4231_HW_CS4235 || + chip->hardware == CS4231_HW_CS4239) + runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; + + snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.period_bytes_max); + + if (chip->claim_dma) { + if ((err = chip->claim_dma(chip, chip->dma_private_data, chip->dma1)) < 0) + return err; + } + + if ((err = snd_cs4231_open(chip, CS4231_MODE_PLAY)) < 0) { + if (chip->release_dma) + chip->release_dma(chip, chip->dma_private_data, chip->dma1); + snd_free_pages(runtime->dma_area, runtime->dma_bytes); + return err; + } + chip->playback_substream = substream; + snd_pcm_set_sync(substream); + chip->rate_constraint(runtime); + return 0; +} + +static int snd_cs4231_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + runtime->hw = snd_cs4231_capture; + + /* hardware limitation of cheap chips */ + if (chip->hardware == CS4231_HW_CS4235 || + chip->hardware == CS4231_HW_CS4239) + runtime->hw.formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE; + + snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); + snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.period_bytes_max); + + if (chip->claim_dma) { + if ((err = chip->claim_dma(chip, chip->dma_private_data, chip->dma2)) < 0) + return err; + } + + if ((err = snd_cs4231_open(chip, CS4231_MODE_RECORD)) < 0) { + if (chip->release_dma) + chip->release_dma(chip, chip->dma_private_data, chip->dma2); + snd_free_pages(runtime->dma_area, runtime->dma_bytes); + return err; + } + chip->capture_substream = substream; + snd_pcm_set_sync(substream); + chip->rate_constraint(runtime); + return 0; +} + +static int snd_cs4231_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + + chip->playback_substream = NULL; + snd_cs4231_close(chip, CS4231_MODE_PLAY); + return 0; +} + +static int snd_cs4231_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); + + chip->capture_substream = NULL; + snd_cs4231_close(chip, CS4231_MODE_RECORD); + return 0; +} + +#ifdef CONFIG_PM + +/* lowlevel suspend callback for CS4231 */ +static void snd_cs4231_suspend(struct snd_cs4231 *chip) +{ + int reg; + unsigned long flags; + + snd_pcm_suspend_all(chip->pcm); + spin_lock_irqsave(&chip->reg_lock, flags); + for (reg = 0; reg < 32; reg++) + chip->image[reg] = snd_cs4231_in(chip, reg); + spin_unlock_irqrestore(&chip->reg_lock, flags); +} + +/* lowlevel resume callback for CS4231 */ +static void snd_cs4231_resume(struct snd_cs4231 *chip) +{ + int reg; + unsigned long flags; + /* int timeout; */ + + snd_cs4231_mce_up(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + for (reg = 0; reg < 32; reg++) { + switch (reg) { + case CS4231_VERSION: + break; + default: + snd_cs4231_out(chip, reg, chip->image[reg]); + break; + } + } + spin_unlock_irqrestore(&chip->reg_lock, flags); +#if 1 + snd_cs4231_mce_down(chip); +#else + /* The following is a workaround to avoid freeze after resume on TP600E. + This is the first half of copy of snd_cs4231_mce_down(), but doesn't + include rescheduling. -- iwai + */ + snd_cs4231_busy_wait(chip); + spin_lock_irqsave(&chip->reg_lock, flags); + chip->mce_bit &= ~CS4231_MCE; + timeout = cs4231_inb(chip, CS4231P(REGSEL)); + cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (timeout == 0x80) + snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port); + if ((timeout & CS4231_MCE) == 0 || + !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { + return; + } + snd_cs4231_busy_wait(chip); +#endif +} +#endif /* CONFIG_PM */ + +static int snd_cs4231_free(struct snd_cs4231 *chip) +{ + release_and_free_resource(chip->res_port); + release_and_free_resource(chip->res_cport); + if (chip->irq >= 0) { + disable_irq(chip->irq); + if (!(chip->hwshare & CS4231_HWSHARE_IRQ)) + free_irq(chip->irq, (void *) chip); + } + if (!(chip->hwshare & CS4231_HWSHARE_DMA1) && chip->dma1 >= 0) { + snd_dma_disable(chip->dma1); + free_dma(chip->dma1); + } + if (!(chip->hwshare & CS4231_HWSHARE_DMA2) && chip->dma2 >= 0 && chip->dma2 != chip->dma1) { + snd_dma_disable(chip->dma2); + free_dma(chip->dma2); + } + if (chip->timer) + snd_device_free(chip->card, chip->timer); + kfree(chip); + return 0; +} + +static int snd_cs4231_dev_free(struct snd_device *device) +{ + struct snd_cs4231 *chip = device->device_data; + return snd_cs4231_free(chip); +} + +const char *snd_cs4231_chip_id(struct snd_cs4231 *chip) +{ + switch (chip->hardware) { + case CS4231_HW_CS4231: return "CS4231"; + case CS4231_HW_CS4231A: return "CS4231A"; + case CS4231_HW_CS4232: return "CS4232"; + case CS4231_HW_CS4232A: return "CS4232A"; + case CS4231_HW_CS4235: return "CS4235"; + case CS4231_HW_CS4236: return "CS4236"; + case CS4231_HW_CS4236B: return "CS4236B"; + case CS4231_HW_CS4237B: return "CS4237B"; + case CS4231_HW_CS4238B: return "CS4238B"; + case CS4231_HW_CS4239: return "CS4239"; + case CS4231_HW_INTERWAVE: return "AMD InterWave"; + case CS4231_HW_OPL3SA2: return chip->card->shortname; + case CS4231_HW_AD1845: return "AD1845"; + case CS4231_HW_OPTI93X: return "OPTi 93x"; + default: return "???"; + } +} + +static int snd_cs4231_new(struct snd_card *card, + unsigned short hardware, + unsigned short hwshare, + struct snd_cs4231 ** rchip) +{ + struct snd_cs4231 *chip; + + *rchip = NULL; + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + chip->hardware = hardware; + chip->hwshare = hwshare; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->mce_mutex); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->rate_constraint = snd_cs4231_xrate; + chip->set_playback_format = snd_cs4231_playback_format; + chip->set_capture_format = snd_cs4231_capture_format; + if (chip->hardware == CS4231_HW_OPTI93X) + memcpy(&chip->image, &snd_opti93x_original_image, + sizeof(snd_opti93x_original_image)); + else + memcpy(&chip->image, &snd_cs4231_original_image, + sizeof(snd_cs4231_original_image)); + + *rchip = chip; + return 0; +} + +int snd_cs4231_create(struct snd_card *card, + unsigned long port, + unsigned long cport, + int irq, int dma1, int dma2, + unsigned short hardware, + unsigned short hwshare, + struct snd_cs4231 ** rchip) +{ + static struct snd_device_ops ops = { + .dev_free = snd_cs4231_dev_free, + }; + struct snd_cs4231 *chip; + int err; + + err = snd_cs4231_new(card, hardware, hwshare, &chip); + if (err < 0) + return err; + + chip->irq = -1; + chip->dma1 = -1; + chip->dma2 = -1; + + if ((chip->res_port = request_region(port, 4, "CS4231")) == NULL) { + snd_printk(KERN_ERR "cs4231: can't grab port 0x%lx\n", port); + snd_cs4231_free(chip); + return -EBUSY; + } + chip->port = port; + if ((long)cport >= 0 && (chip->res_cport = request_region(cport, 8, "CS4232 Control")) == NULL) { + snd_printk(KERN_ERR "cs4231: can't grab control port 0x%lx\n", cport); + snd_cs4231_free(chip); + return -ENODEV; + } + chip->cport = cport; + if (!(hwshare & CS4231_HWSHARE_IRQ) && request_irq(irq, snd_cs4231_interrupt, IRQF_DISABLED, "CS4231", (void *) chip)) { + snd_printk(KERN_ERR "cs4231: can't grab IRQ %d\n", irq); + snd_cs4231_free(chip); + return -EBUSY; + } + chip->irq = irq; + if (!(hwshare & CS4231_HWSHARE_DMA1) && request_dma(dma1, "CS4231 - 1")) { + snd_printk(KERN_ERR "cs4231: can't grab DMA1 %d\n", dma1); + snd_cs4231_free(chip); + return -EBUSY; + } + chip->dma1 = dma1; + if (!(hwshare & CS4231_HWSHARE_DMA2) && dma1 != dma2 && dma2 >= 0 && request_dma(dma2, "CS4231 - 2")) { + snd_printk(KERN_ERR "cs4231: can't grab DMA2 %d\n", dma2); + snd_cs4231_free(chip); + return -EBUSY; + } + if (dma1 == dma2 || dma2 < 0) { + chip->single_dma = 1; + chip->dma2 = chip->dma1; + } else + chip->dma2 = dma2; + + /* global setup */ + if (snd_cs4231_probe(chip) < 0) { + snd_cs4231_free(chip); + return -ENODEV; + } + snd_cs4231_init(chip); + +#if 0 + if (chip->hardware & CS4231_HW_CS4232_MASK) { + if (chip->res_cport == NULL) + snd_printk("CS4232 control port features are not accessible\n"); + } +#endif + + /* Register device */ + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { + snd_cs4231_free(chip); + return err; + } + +#ifdef CONFIG_PM + /* Power Management */ + chip->suspend = snd_cs4231_suspend; + chip->resume = snd_cs4231_resume; +#endif + + *rchip = chip; + return 0; +} + +static struct snd_pcm_ops snd_cs4231_playback_ops = { + .open = snd_cs4231_playback_open, + .close = snd_cs4231_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_cs4231_playback_hw_params, + .hw_free = snd_cs4231_playback_hw_free, + .prepare = snd_cs4231_playback_prepare, + .trigger = snd_cs4231_trigger, + .pointer = snd_cs4231_playback_pointer, +}; + +static struct snd_pcm_ops snd_cs4231_capture_ops = { + .open = snd_cs4231_capture_open, + .close = snd_cs4231_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_cs4231_capture_hw_params, + .hw_free = snd_cs4231_capture_hw_free, + .prepare = snd_cs4231_capture_prepare, + .trigger = snd_cs4231_trigger, + .pointer = snd_cs4231_capture_pointer, +}; + +int snd_cs4231_pcm(struct snd_cs4231 *chip, int device, struct snd_pcm **rpcm) +{ + struct snd_pcm *pcm; + int err; + + if ((err = snd_pcm_new(chip->card, "CS4231", device, 1, 1, &pcm)) < 0) + return err; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->mce_mutex); + mutex_init(&chip->open_mutex); + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4231_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cs4231_capture_ops); + + /* global setup */ + pcm->private_data = chip; + pcm->info_flags = 0; + if (chip->single_dma) + pcm->info_flags |= SNDRV_PCM_INFO_HALF_DUPLEX; + if (chip->hardware != CS4231_HW_INTERWAVE) + pcm->info_flags |= SNDRV_PCM_INFO_JOINT_DUPLEX; + strcpy(pcm->name, snd_cs4231_chip_id(chip)); + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), + 64*1024, chip->dma1 > 3 || chip->dma2 > 3 ? 128*1024 : 64*1024); + + chip->pcm = pcm; + if (rpcm) + *rpcm = pcm; + return 0; +} + +static void snd_cs4231_timer_free(struct snd_timer *timer) +{ + struct snd_cs4231 *chip = timer->private_data; + chip->timer = NULL; +} + +int snd_cs4231_timer(struct snd_cs4231 *chip, int device, struct snd_timer **rtimer) +{ + struct snd_timer *timer; + struct snd_timer_id tid; + int err; + + /* Timer initialization */ + tid.dev_class = SNDRV_TIMER_CLASS_CARD; + tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; + tid.card = chip->card->number; + tid.device = device; + tid.subdevice = 0; + if ((err = snd_timer_new(chip->card, "CS4231", &tid, &timer)) < 0) + return err; + strcpy(timer->name, snd_cs4231_chip_id(chip)); + timer->private_data = chip; + timer->private_free = snd_cs4231_timer_free; + timer->hw = snd_cs4231_timer_table; + chip->timer = timer; + if (rtimer) + *rtimer = timer; + return 0; +} + +/* + * MIXER part + */ + +static int snd_cs4231_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + static char *texts[4] = { + "Line", "Aux", "Mic", "Mix" + }; + static char *opl3sa_texts[4] = { + "Line", "CD", "Mic", "Mix" + }; + static char *gusmax_texts[4] = { + "Line", "Synth", "Mic", "Mix" + }; + char **ptexts = texts; + struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + + snd_assert(chip->card != NULL, return -EINVAL); + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 2; + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + if (!strcmp(chip->card->driver, "GUS MAX")) + ptexts = gusmax_texts; + switch (chip->hardware) { + case CS4231_HW_INTERWAVE: ptexts = gusmax_texts; break; + case CS4231_HW_OPL3SA2: ptexts = opl3sa_texts; break; + } + strcpy(uinfo->value.enumerated.name, ptexts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_cs4231_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + unsigned long flags; + + spin_lock_irqsave(&chip->reg_lock, flags); + ucontrol->value.enumerated.item[0] = (chip->image[CS4231_LEFT_INPUT] & CS4231_MIXS_ALL) >> 6; + ucontrol->value.enumerated.item[1] = (chip->image[CS4231_RIGHT_INPUT] & CS4231_MIXS_ALL) >> 6; + spin_unlock_irqrestore(&chip->reg_lock, flags); + return 0; +} + +static int snd_cs4231_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + unsigned long flags; + unsigned short left, right; + int change; + + if (ucontrol->value.enumerated.item[0] > 3 || + ucontrol->value.enumerated.item[1] > 3) + return -EINVAL; + left = ucontrol->value.enumerated.item[0] << 6; + right = ucontrol->value.enumerated.item[1] << 6; + spin_lock_irqsave(&chip->reg_lock, flags); + left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left; + right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right; + change = left != chip->image[CS4231_LEFT_INPUT] || + right != chip->image[CS4231_RIGHT_INPUT]; + snd_cs4231_out(chip, CS4231_LEFT_INPUT, left); + snd_cs4231_out(chip, CS4231_RIGHT_INPUT, right); + spin_unlock_irqrestore(&chip->reg_lock, flags); + return change; +} + +int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} + +int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + + spin_lock_irqsave(&chip->reg_lock, flags); + ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (invert) + ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; + return 0; +} + +int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int mask = (kcontrol->private_value >> 16) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + int change; + unsigned short val; + + val = (ucontrol->value.integer.value[0] & mask); + if (invert) + val = mask - val; + val <<= shift; + spin_lock_irqsave(&chip->reg_lock, flags); + val = (chip->image[reg] & ~(mask << shift)) | val; + change = val != chip->image[reg]; + snd_cs4231_out(chip, reg, val); + spin_unlock_irqrestore(&chip->reg_lock, flags); + return change; +} + +int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} + +int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x07; + int shift_right = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + + spin_lock_irqsave(&chip->reg_lock, flags); + ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; + ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (invert) { + ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; + } + return 0; +} + +int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int left_reg = kcontrol->private_value & 0xff; + int right_reg = (kcontrol->private_value >> 8) & 0xff; + int shift_left = (kcontrol->private_value >> 16) & 0x07; + int shift_right = (kcontrol->private_value >> 19) & 0x07; + int mask = (kcontrol->private_value >> 24) & 0xff; + int invert = (kcontrol->private_value >> 22) & 1; + int change; + unsigned short val1, val2; + + val1 = ucontrol->value.integer.value[0] & mask; + val2 = ucontrol->value.integer.value[1] & mask; + if (invert) { + val1 = mask - val1; + val2 = mask - val2; + } + val1 <<= shift_left; + val2 <<= shift_right; + spin_lock_irqsave(&chip->reg_lock, flags); + val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; + val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; + change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; + snd_cs4231_out(chip, left_reg, val1); + snd_cs4231_out(chip, right_reg, val2); + spin_unlock_irqrestore(&chip->reg_lock, flags); + return change; +} + +static struct snd_kcontrol_new snd_cs4231_controls[] = { +CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), +CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +CS4231_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), +CS4231_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +}, +CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) +}; + +static struct snd_kcontrol_new snd_opti93x_controls[] = { +CS4231_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +CS4231_DOUBLE("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +CS4231_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +CS4231_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("CD Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Capture Volume", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +} +}; + +int snd_cs4231_mixer(struct snd_cs4231 *chip) +{ + struct snd_card *card; + unsigned int idx; + int err; + + snd_assert(chip != NULL && chip->pcm != NULL, return -EINVAL); + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + if (chip->hardware == CS4231_HW_OPTI93X) + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], + chip)); + if (err < 0) + return err; + } + else + for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], + chip)); + if (err < 0) + return err; + } + return 0; +} + +EXPORT_SYMBOL(snd_cs4231_out); +EXPORT_SYMBOL(snd_cs4231_in); +EXPORT_SYMBOL(snd_cs4236_ext_out); +EXPORT_SYMBOL(snd_cs4236_ext_in); +EXPORT_SYMBOL(snd_cs4231_mce_up); +EXPORT_SYMBOL(snd_cs4231_mce_down); +EXPORT_SYMBOL(snd_cs4231_overrange); +EXPORT_SYMBOL(snd_cs4231_interrupt); +EXPORT_SYMBOL(snd_cs4231_chip_id); +EXPORT_SYMBOL(snd_cs4231_create); +EXPORT_SYMBOL(snd_cs4231_pcm); +EXPORT_SYMBOL(snd_cs4231_mixer); +EXPORT_SYMBOL(snd_cs4231_timer); +EXPORT_SYMBOL(snd_cs4231_info_single); +EXPORT_SYMBOL(snd_cs4231_get_single); +EXPORT_SYMBOL(snd_cs4231_put_single); +EXPORT_SYMBOL(snd_cs4231_info_double); +EXPORT_SYMBOL(snd_cs4231_get_double); +EXPORT_SYMBOL(snd_cs4231_put_double); + +/* + * INIT part + */ + +static int __init alsa_cs4231_init(void) +{ + return 0; +} + +static void __exit alsa_cs4231_exit(void) +{ +} + +module_init(alsa_cs4231_init) +module_exit(alsa_cs4231_exit) ---------------------------------------------------------------------- Konkurs dla zmotoryzowanych! Wygraj narz�dzia samochodowe! kliknij >>> http://link.interia.pl/f1e44

2 1

Re: [alsa-devel] [PATCH 1/2] wss_lib: move cs4231_lib into wss_lib
by Rene Herman 24 Jun '08

24 Jun '08

On 19-06-08 23:07, Krzysztof Helt wrote: > From: Krzysztof Helt <krzysztof.h1(a)wp.pl> > > Move the file sound/isa/cs423x/cs4231_lib.c > into sound/isa/cs423x/wss_lib.c > This is the first step toward merging all libraries > for Windows Sound System compatible chips > into a single library. > > Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> Acked-by: Rene Herman <rene.herman(a)gmail.com> Only small request: > diff -urNp linux-alsa/sound/isa/wss/Makefile linux-mm/sound/isa/wss/Makefile > --- linux-alsa/sound/isa/wss/Makefile 1970-01-01 01:00:00.000000000 +0100 > +++ linux-mm/sound/isa/wss/Makefile 2008-06-15 22:38:44.705642447 +0200 > @@ -0,0 +1,10 @@ > +# > +# Makefile for ALSA > +# Copyright (c) 2008 by Krzysztof Helt <krzysztof.h1(a)wp.pl> > +# > + > +snd-wss-lib-objs := wss_lib.o > + > +# Toplevel Module Dependency > +obj-$(CONFIG_SND_WSS_LIB) += snd-wss-lib.o > + Could you maybe "donate" the copyright on this Makefile to Jaroslav Kysela? Basically, this single file is too trivial (and uncreative due to being rather fully dictated by the build-system as a whole) to be protectable under copyright in the first place but the ALSA build system as a whole IS copyrightable and Jaroslav holds the copyright on all others (except David Miller on the one sparc specific one). It's likely not a practical issue really but I feel it's probably good to keep this concentrated. Rene.

3 3

[alsa-devel] alc262 - hda intel driver - power save not working
by Harsha priya gupta 24 Jun '08

24 Jun '08

Hi, Have been working on this to get the power save option to work for ALC262 using intel hda driver. (ALSA 2.16v, Kernel-2.6.22.18-1) Once i load the driver, the power_save_controller shows 'y' indicating its enabled and power_save value is set to'10' Still i dont see the driver going into d3 state when there is no playback or capture happening. I also made sure that i play a file, close the player and mute the controls. I wait for 10 seconds and still see the power consumed is the same whereas I expect the power to come down. Please help me with more clues as to what i could be doing wrong or missing here. Any clue will be greatly appreciated as i am needing to resolve this ASAP. -- -Harsha

2 3

[alsa-devel] wrong print debugging message code
by Jin-Young Park 24 Jun '08

24 Jun '08

Hello, I written ASoC Audio driver for our company chip in linux kernel 2.6.24.1. New kernel 2.6.25.7 released before several days. So, I merge my ASoC Audio dirver to kernel 2.6.25.7. then I found wrong code that print debuuging message code in several files. These file location is kernel_src/sound/soc/codecs for SoC audio codec drvier files These files include follow define code for print debug message. ------------------------------------------------------------------------------------------------------------------------ #define XXXX_DEBUG 0 #ifdef XXXX_DEBUG #define dbg(format, arg...) \ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) #else #define dbg(format, arg...) do {} while (0) #endif ------------------------------------------------------------------------------------------------------------------------ I think "#ifdef XXXX_DEBUG" line is wrong. Because, XXXX_DEBUG was already defined 0. I known that "#ifdef" is only check defined or not defined, 0 or 1(false or true) are not check. "#if XXXX_DEBUG" code is more correct than "#ifdef XXXX_DEBUG". thanks

3 7

Re: [alsa-devel] Audiowerk2 driver doesnt check pci subsystem id.
by Cédric Brégardis 23 Jun '08

23 Jun '08

Hi At Monday 23 June 2008 10:02:17, Mike Lampard wrote: > Hi > > Is there any chance you could add a pci subsystem id check to your > Audiowerk2 alsa driver? As you know, at the moment it claims all saa7146 > devices as AW2's, which is a pain for me because I also have DVB hardware > that uses the SAA7146.. Indeed, it is a problem. However, I am not sure that the subsystem id is working correctly for AW2. In any case the behavior on my system is strange. Here is the result of the command 'lspci -x' [...] 02:0b.0 Multimedia controller: Philips Semiconductors SAA7146 (rev 01) 00: 31 11 46 71 02 00 80 02 01 00 80 04 00 40 00 00 10: 00 bc af fe 00 00 00 00 00 00 00 00 00 00 00 00 20: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 30: 00 00 00 00 00 00 00 00 00 00 00 00 0b 01 00 00 [...] SAA7146 datasheet says that subsystem ID is at address 0x2C of configuration space regsiters, and: "this value will be loaded after a PCI reset from external hardware using the I2C-bus; the default value is 0000H" So there is no specific subsystem id on my AW2: it is the default value. I have noted that the card has an eeprom on i2c (reference 8585c) but it seams it doesn't help. The solution could be to suppose that the AW2 subid is 0. But if another card has the same behavior, there will be a problem. Mike, could you give us the result of 'lspci -x' command on your system please ? And I rely upon the list to correct me if I made wrong assumption ;-). Cedric. > > Cheers & Good Work, > > Mike

1 0

[alsa-devel] [PATCH] Fix register programming in Freescale MPC8610 HPCD sound driver
by Timur Tabi 23 Jun '08

23 Jun '08

Fix the Freescale MPC8610 HPCD sound driver so that it programs the DMACR and PMUXCR registers in the global utilities correctly. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch is for 2.6.26. A similar fix has already been made in the ASOC V2 repository. sound/soc/fsl/mpc8610_hpcd.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a00aac7..8820c3f 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -58,9 +58,9 @@ static int mpc8610_hpcd_machine_probe(struct platform_device *sound_device) sound_device->dev.platform_data; /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], CCSR_GUTS_DMACR_DEV_SSI); guts_set_pmuxcr_dma(machine_data->guts, machine_data->dma_id, @@ -170,9 +170,9 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) /* Restore the signal routing */ - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[0], 0); - guts_set_dmacr(machine_data->guts, machine_data->dma_id + 1, + guts_set_dmacr(machine_data->guts, machine_data->dma_id, machine_data->dma_channel_id[1], 0); switch (machine_data->ssi_id) { @@ -182,7 +182,7 @@ int mpc8610_hpcd_machine_remove(struct platform_device *sound_device) break; case 1: clrsetbits_be32(&machine_data->guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); + CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); break; } -- 1.5.5

3 9

[alsa-devel] AUDIO: Split WM8750 IO routines from core for SPI
by Ben Dooks 23 Jun '08

23 Jun '08

Add SPI support to the WM8750 ASoC codec by splitting the I2C out of the WM8750 codec and adding an standard SPI driver to the system. Signed-off-by: Ben Dooks <ben-linux(a)fluff.org> Index: linux-2.6.26-rc4-quilt3/sound/soc/codecs/Kconfig =================================================================== --- linux-2.6.26-rc4-quilt3.orig/sound/soc/codecs/Kconfig 2008-05-26 23:17:51.000000000 +0100 +++ linux-2.6.26-rc4-quilt3/sound/soc/codecs/Kconfig 2008-06-02 15:26:21.000000000 +0100 @@ -10,6 +10,16 @@ config SND_SOC_WM8750 tristate depends on SND_SOC +config SND_SOC_WM8750_I2C + tristate + select SND_SOC_WM8750 + depends on SND_SOC && (CONFIG_I2C || CONFIG_I2C_MODULE) + +config SND_SOC_WM8750_SPI + tristate + select SND_SOC_WM8750 + depends on SND_SOC && CONFIG_SPI + config SND_SOC_WM8753 tristate depends on SND_SOC Index: linux-2.6.26-rc4-quilt3/sound/soc/codecs/Makefile =================================================================== --- linux-2.6.26-rc4-quilt3.orig/sound/soc/codecs/Makefile 2008-05-26 23:17:51.000000000 +0100 +++ linux-2.6.26-rc4-quilt3/sound/soc/codecs/Makefile 2008-06-02 15:26:57.000000000 +0100 @@ -1,6 +1,8 @@ snd-soc-ac97-objs := ac97.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o +snd-soc-wm8750-spi-objs := wm8750-spi.o +snd-soc-wm8750-i2c-objs := wm8750-i2c.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o @@ -10,6 +12,8 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x. obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o +obj-$(CONFIG_SND_SOC_WM8750_I2C) += snd-soc-wm8750-i2c.o +obj-$(CONFIG_SND_SOC_WM8750_SPI) += snd-soc-wm8750-spi.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o Index: linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750-i2c.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750-i2c.c 2008-06-02 15:42:19.000000000 +0100 @@ -0,0 +1,175 @@ +/* + * wm8750-i2c.c -- WM8750 ALSA SoC I2C Driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie <richard(a)openedhand.com> + * + * Based on WM8753.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8750.h" + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static struct snd_soc_device *wm8750_socdev; + +/* + * WM8731 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver wm8750_i2c_driver; +static struct i2c_client client_template; + +static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = wm8750_socdev; + struct wm8750_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + err("failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = wm8750_init(socdev); + if (ret < 0) { + err("failed to initialise WM8750\n"); + goto err; + } + + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int wm8750_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int wm8750_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, wm8750_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver wm8750_i2c_driver = { + .driver = { + .name = "WM8750 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_WM8750, + .attach_adapter = wm8750_i2c_attach, + .detach_client = wm8750_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "WM8750", + .driver = &wm8750_i2c_driver, +}; + +static int wm8750_probe_i2c(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8750_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec; + + codec = wm8750_probe(pdev); + if (codec == NULL) { + dev_err(&pdev->dev, "No memory for CODEC device\n"); + return -ENODEV; + } + + /* initialise our interface */ + + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } + + return 0; +} + +static int wm8750_remove_i2c(struct platform_device *pdev) +{ + int ret; + + ret = wm8750_remove(pdev); + if (ret) + return ret; + + i2c_del_driver(&wm8750_i2c_driver); + wm8750_free(pdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8750_i2c = { + .probe = wm8750_probe_i2c, + .remove = wm8750_remove_i2c, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; + +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750_i2c); + +MODULE_DESCRIPTION("ASoC WM8750 I2C access driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); Index: linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750.c =================================================================== --- linux-2.6.26-rc4-quilt3.orig/sound/soc/codecs/wm8750.c 2008-05-26 23:17:51.000000000 +0100 +++ linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750.c 2008-06-02 19:59:08.000000000 +0100 @@ -17,7 +17,6 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> -#include <linux/i2c.h> #include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -751,7 +750,7 @@ static void wm8750_work(struct work_stru wm8750_dapm_event(codec, codec->dapm_state); } -static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) +int wm8750_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; @@ -760,7 +759,9 @@ static int wm8750_suspend(struct platfor return 0; } -static int wm8750_resume(struct platform_device *pdev) +EXPORT_SYMBOL_GPL(wm8750_suspend); + +int wm8750_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; @@ -790,11 +791,13 @@ static int wm8750_resume(struct platform return 0; } +EXPORT_SYMBOL_GPL(wm8750_resume); + /* * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev) +int wm8750_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; int reg, ret = 0; @@ -860,139 +863,36 @@ pcm_err: return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8750_socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -/* - * WM8731 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8750_i2c_driver; -static struct i2c_client client_template; - -static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) -{ - struct snd_soc_device *socdev = wm8750_socdev; - struct wm8750_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; - int ret; - - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; - } - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = i2c_attach_client(i2c); - if (ret < 0) { - err("failed to attach codec at addr %x\n", addr); - goto err; - } - - ret = wm8750_init(socdev); - if (ret < 0) { - err("failed to initialise WM8750\n"); - goto err; - } - return ret; - -err: - kfree(codec); - kfree(i2c); - return ret; -} - -static int wm8750_i2c_detach(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); - kfree(codec->reg_cache); - kfree(client); - return 0; -} - -static int wm8750_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8750_codec_probe); -} - -/* corgi i2c codec control layer */ -static struct i2c_driver wm8750_i2c_driver = { - .driver = { - .name = "WM8750 I2C Codec", - .owner = THIS_MODULE, - }, - .id = I2C_DRIVERID_WM8750, - .attach_adapter = wm8750_i2c_attach, - .detach_client = wm8750_i2c_detach, - .command = NULL, -}; - -static struct i2c_client client_template = { - .name = "WM8750", - .driver = &wm8750_i2c_driver, -}; -#endif +EXPORT_SYMBOL_GPL(wm8750_init); -static int wm8750_probe(struct platform_device *pdev) +struct snd_soc_codec *wm8750_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8750_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec; struct wm8750_priv *wm8750; - int ret = 0; info("WM8750 Audio Codec %s", WM8750_VERSION); + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) - return -ENOMEM; + return NULL; wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); if (wm8750 == NULL) { kfree(codec); - return -ENOMEM; + return NULL; } codec->private_data = wm8750; socdev->codec = codec; + wm8750_socdev = socdev; mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - wm8750_socdev = socdev; INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; - codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); - } -#else - /* Add other interfaces here */ -#endif - - return ret; + return codec; } /* @@ -1015,32 +915,33 @@ static int run_delayed_work(struct delay } /* power down chip */ -static int wm8750_remove(struct platform_device *pdev) +int wm8750_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; if (codec->control_data) wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); + run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm8750_i2c_driver); -#endif - kfree(codec->private_data); - kfree(codec); return 0; } -struct snd_soc_codec_device soc_codec_dev_wm8750 = { - .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); +EXPORT_SYMBOL_GPL(wm8750_remove); + +void wm8750_free(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + kfree(codec->private_data); + kfree(codec); +} + +EXPORT_SYMBOL_GPL(wm8750_free); MODULE_DESCRIPTION("ASoC WM8750 driver"); MODULE_AUTHOR("Liam Girdwood"); Index: linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750.h =================================================================== --- linux-2.6.26-rc4-quilt3.orig/sound/soc/codecs/wm8750.h 2008-01-24 22:58:37.000000000 +0000 +++ linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750.h 2008-06-02 15:26:21.000000000 +0100 @@ -62,6 +62,18 @@ struct wm8750_setup_data { }; extern struct snd_soc_codec_dai wm8750_dai; -extern struct snd_soc_codec_device soc_codec_dev_wm8750; + +extern struct snd_soc_codec_device soc_codec_dev_wm8750_i2c; +extern struct snd_soc_codec_device soc_codec_dev_wm8750_spi; + +/* internal items for the interface modules, not for SoC drivers */ + +extern struct snd_soc_codec *wm8750_probe(struct platform_device *pdev); +extern int wm8750_init(struct snd_soc_device *socdev); + +extern void wm8750_free(struct platform_device *pdev); +extern int wm8750_suspend(struct platform_device *pdev, pm_message_t state); +extern int wm8750_resume(struct platform_device *pdev); +extern int wm8750_remove(struct platform_device *pdev); #endif Index: linux-2.6.26-rc4-quilt3/sound/soc/pxa/spitz.c =================================================================== --- linux-2.6.26-rc4-quilt3.orig/sound/soc/pxa/spitz.c 2008-05-26 23:17:51.000000000 +0100 +++ linux-2.6.26-rc4-quilt3/sound/soc/pxa/spitz.c 2008-06-02 15:26:21.000000000 +0100 @@ -351,7 +351,7 @@ static struct wm8750_setup_data spitz_wm static struct snd_soc_device spitz_snd_devdata = { .machine = &snd_soc_machine_spitz, .platform = &pxa2xx_soc_platform, - .codec_dev = &soc_codec_dev_wm8750, + .codec_dev = &soc_codec_dev_wm8750_i2c, .codec_data = &spitz_wm8750_setup, }; Index: linux-2.6.26-rc4-quilt3/sound/soc/pxa/Kconfig =================================================================== --- linux-2.6.26-rc4-quilt3.orig/sound/soc/pxa/Kconfig 2008-04-17 09:49:21.000000000 +0100 +++ linux-2.6.26-rc4-quilt3/sound/soc/pxa/Kconfig 2008-06-02 15:26:21.000000000 +0100 @@ -31,6 +31,7 @@ config SND_PXA2XX_SOC_SPITZ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 select SND_PXA2XX_SOC_I2S + select SND_SOC_WM8750_I2C select SND_SOC_WM8750 help Say Y if you want to add support for SoC audio on Sharp Index: linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750-spi.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ linux-2.6.26-rc4-quilt3/sound/soc/codecs/wm8750-spi.c 2008-06-02 19:58:55.000000000 +0100 @@ -0,0 +1,161 @@ +/* sound/asoc/codecs/wm8750-spi.c + * + * Copyright 2007 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> + +#include <linux/spi/spi.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8750.h" + +static struct snd_soc_device *wm8750_socdev; + +struct wm8750_spi { + struct snd_soc_device *socdev; + struct spi_device *dev; + struct spi_message message; + struct spi_transfer transfer; +}; + +static int wm8750_spi_write(void *ctrl, const char *data, int size) +{ + struct wm8750_spi *spi = ctrl; + + spi_message_init(&spi->message); + spi_message_add_tail(&spi->transfer, &spi->message); + + spi->transfer.tx_buf = data; + spi->transfer.len = size; + spi->transfer.bits_per_word = 8; + + return spi_sync(spi->dev, &spi->message); +} + +static int __devinit wm8750_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8750_socdev; + struct snd_soc_codec *codec = socdev->codec; + struct wm8750_spi *spidata; + int ret; + + dev_info(&spi->dev, "probing device\n"); + + spidata = kmalloc(sizeof(struct wm8750_spi), GFP_KERNEL); + if (spidata == NULL) { + dev_err(&spi->dev, "failed to allocate state\n"); + kfree(codec); + return -ENOMEM; + } + + spidata->dev = spi; + spidata->socdev = socdev; + codec->control_data = spi; + + dev_set_drvdata(&spi->dev, spi); + + ret = wm8750_init(socdev); + if (ret < 0) { + dev_err(&spi->dev, "failed to initialise codec\n"); + kfree(spi); + kfree(codec); + return ret; + } + + dev_info(&spi->dev, "CODEC initialised\n"); + + return 0; +} + +static int __devexit wm8750_spi_remove(struct spi_device *spi) +{ + struct wm8750_spi *spidata = dev_get_drvdata(&spi->dev); + struct snd_soc_codec *codec = spidata->socdev->codec; + + dev_info(&spi->dev, "removing device\n"); + + kfree(codec->reg_cache); + kfree(spi); + + return 0; +} + +static struct spi_driver wm8750_spi_driver = { + .driver = { + .name = "WM8750", + .owner = THIS_MODULE, + }, + .probe = wm8750_spi_probe, + .remove = __devexit_p(wm8750_spi_remove), +}; + + +static int wm8750_probe_spi(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = wm8750_probe(pdev); + if (codec == NULL) { + dev_err(&pdev->dev, "No memory for CODEC device\n"); + return -ENODEV; + } + + wm8750_socdev = socdev; + + codec->hw_write = wm8750_spi_write; + + ret = spi_register_driver(&wm8750_spi_driver); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register spi driver\n"); + return ret; + } + + return 0; +} + +static int wm8750_remove_spi(struct platform_device *pdev) +{ + int ret; + + ret = wm8750_remove(pdev); + if (ret) + return ret; + + spi_unregister_driver(&wm8750_spi_driver); + wm8750_free(pdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8750_spi = { + .probe = wm8750_probe_spi, + .remove = wm8750_remove_spi, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; + +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750_spi); + +MODULE_DESCRIPTION("ASoC WM8750 SPI access driver"); +MODULE_AUTHOR("Ben Dooks"); +MODULE_LICENSE("GPL"); -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

3 2

[alsa-devel] [PATCH 1/2] ASoC: Add AK4535 driver
by Mark Brown 23 Jun '08

23 Jun '08

From: Richard Purdie <richard(a)openedhand.com> The AK4535 codec is included in some HP iPAQ systems. This driver was originally written by Richard Purdie and with some bug fixes from Milan Plzik. While out of tree it has also had some mechanical updates for new APIs and current best practices from Liam Girdwood, Graeme Gregory and Mark Brown. Signed-off-by: Richard Purdie <richard(a)openedhand.com> Signed-off-by: Milan Plzik <milan.plzik(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4535.c | 696 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4535.h | 46 +++ 4 files changed, 747 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4535.c create mode 100644 sound/soc/codecs/ak4535.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f439138..1db04a2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2,6 +2,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_AK4535 + tristate + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d9826f3..d7b97ab 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ak4535-objs := ak4535.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8731-objs := wm8731.o @@ -11,6 +12,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c new file mode 100644 index 0000000..469266e --- /dev/null +++ b/sound/soc/codecs/ak4535.c @@ -0,0 +1,696 @@ +/* + * ak4535.c -- AK4535 ALSA Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie <richard(a)openedhand.com> + * + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4535.h" + +#define AUDIO_NAME "ak4535" +#define AK4535_VERSION "0.3" + +struct snd_soc_codec_device soc_codec_dev_ak4535; + +/* codec private data */ +struct ak4535_priv { + unsigned int sysclk; +}; + +/* + * ak4535 register cache + */ +static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { + 0x0000, 0x0080, 0x0000, 0x0003, + 0x0002, 0x0000, 0x0011, 0x0001, + 0x0000, 0x0040, 0x0036, 0x0010, + 0x0000, 0x0000, 0x0057, 0x0000, +}; + +/* + * read ak4535 register cache + */ +static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return -1; + return cache[reg]; +} + +static inline unsigned int ak4535_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 data; + data = reg; + + if (codec->hw_write(codec->control_data, &data, 1) != 1) + return -EIO; + + if (codec->hw_read(codec->control_data, &data, 1) != 1) + return -EIO; + + return data; +}; + +/* + * write ak4535 register cache + */ +static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4535_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the AK4535 register space + */ +static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4535 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + ak4535_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int ak4535_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4535_CACHEREGNUM; i++) + r |= ak4535_write(codec, i, cache[i]); + + return r; +}; + +static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; +static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; +static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; +static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"}; +static const char *ak4535_mic_select[] = {"Internal", "External"}; + +static const struct soc_enum ak4535_enum[] = { + SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain), + SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out), + SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out), + SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp), + SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select), +}; + +static const struct snd_kcontrol_new ak4535_snd_controls[] = { + SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0), + SOC_ENUM("Mono 1 Output", ak4535_enum[1]), + SOC_ENUM("Mono 1 Gain", ak4535_enum[0]), + SOC_ENUM("Headphone Output", ak4535_enum[2]), + SOC_ENUM("Playback Deemphasis", ak4535_enum[3]), + SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0), + SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0), + SOC_ENUM("Mic Select", ak4535_enum[4]), + SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0), + SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0), + SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0), + SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0), + SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0), + SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0), + SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0), + SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1), + SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1), + SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0), + SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), +}; + +/* add non dapm controls */ +static int ak4535_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* Mono 1 Mixer */ +static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), + SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0), +}; + +/* Stereo Mixer */ +static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0), + SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0), +}; + +/* Input Mixer */ +static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = { + SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0), + SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0), +}; + +/* Input mux */ +static const struct snd_kcontrol_new ak4535_input_mux_control = + SOC_DAPM_ENUM("Input Select", ak4535_enum[4]); + +/* HP L switch */ +static const struct snd_kcontrol_new ak4535_hpl_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1); + +/* HP R switch */ +static const struct snd_kcontrol_new ak4535_hpr_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1); + +/* mono 2 switch */ +static const struct snd_kcontrol_new ak4535_mono2_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0); + +/* Line out switch */ +static const struct snd_kcontrol_new ak4535_line_control = + SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0); + +/* ak4535 dapm widgets */ +static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_stereo_mixer_controls[0], + ARRAY_SIZE(ak4535_stereo_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_mono1_mixer_controls[0], + ARRAY_SIZE(ak4535_mono1_mixer_controls)), + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, + &ak4535_input_mixer_controls[0], + ARRAY_SIZE(ak4535_input_mixer_controls)), + SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, + &ak4535_input_mux_control), + SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0), + SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, + &ak4535_mono2_control), + /* speaker powersave bit */ + SND_SOC_DAPM_PGA("Speaker Enable", AK4535_MODE2, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, + &ak4535_line_control), + SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpl_control), + SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0, + &ak4535_hpr_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPP"), + SND_SOC_DAPM_OUTPUT("SPN"), + SND_SOC_DAPM_OUTPUT("MOUT1"), + SND_SOC_DAPM_OUTPUT("MOUT2"), + SND_SOC_DAPM_OUTPUT("MICOUT"), + SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 0), + SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0), + SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0), + SND_SOC_DAPM_INPUT("MICIN"), + SND_SOC_DAPM_INPUT("MICEXT"), + SND_SOC_DAPM_INPUT("AUX"), + SND_SOC_DAPM_INPUT("MIN"), + SND_SOC_DAPM_INPUT("AIN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /*stereo mixer */ + {"Stereo Mixer", "Playback Switch", "DAC"}, + {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, + {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, + + /* mono1 mixer */ + {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"}, + {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, + + /* Mic */ + {"Mic", NULL, "AIN"}, + {"Input Mux", "Internal", "Mic Int Bias"}, + {"Input Mux", "External", "Mic Ext Bias"}, + {"Mic Int Bias", NULL, "MICIN"}, + {"Mic Ext Bias", NULL, "MICEXT"}, + {"MICOUT", NULL, "Input Mux"}, + + /* line out */ + {"LOUT", NULL, "Line Out Enable"}, + {"ROUT", NULL, "Line Out Enable"}, + {"Line Out Enable", "Switch", "Line Out"}, + {"Line Out", NULL, "Stereo Mixer"}, + + /* mono1 out */ + {"MOUT1", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono1 Mixer"}, + + /* left HP */ + {"HPL", NULL, "Left HP Enable"}, + {"Left HP Enable", "Switch", "HP L Amp"}, + {"HP L Amp", NULL, "Stereo Mixer"}, + + /* right HP */ + {"HPR", NULL, "Right HP Enable"}, + {"Right HP Enable", "Switch", "HP R Amp"}, + {"HP R Amp", NULL, "Stereo Mixer"}, + + /* speaker */ + {"SPP", NULL, "Speaker Enable"}, + {"SPN", NULL, "Speaker Enable"}, + {"Speaker Enable", "Switch", "Spk Amp"}, + {"Spk Amp", NULL, "MIN"}, + + /* mono 2 */ + {"MOUT2", NULL, "Mono 2 Enable"}, + {"Mono 2 Enable", "Switch", "Stereo Mixer"}, + + /* Aux In */ + {"Aux In", NULL, "AUX"}, + + /* ADC */ + {"ADC", NULL, "Input Mixer"}, + {"Input Mixer", "Mic Capture Switch", "Mic"}, + {"Input Mixer", "Aux Capture Switch", "Aux In"}, +}; + +static int ak4535_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4535_dapm_widgets, + ARRAY_SIZE(ak4535_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4535_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4535_priv *ak4535 = codec->private_data; + + ak4535->sysclk = freq; + return 0; +} + +static int ak4535_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct ak4535_priv *ak4535 = codec->private_data; + u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); + int rate = params_rate(params), fs = 256; + + if (rate) + fs = ak4535->sysclk / rate; + + /* set fs */ + switch (fs) { + case 1024: + mode2 |= (0x2 << 5); + break; + case 512: + mode2 |= (0x1 << 5); + break; + case 256: + break; + } + + /* set rate */ + ak4535_write(codec, AK4535_MODE2, mode2); + return 0; +} + +static int ak4535_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode1 = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + mode1 = 0x0002; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode1 = 0x0001; + break; + default: + return -EINVAL; + } + + /* use 32 fs for BCLK to save power */ + mode1 |= 0x4; + + ak4535_write(codec, AK4535_MODE1, mode1); + return 0; +} + +static int ak4535_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; + if (!mute) + ak4535_write(codec, AK4535_DAC, mute_reg); + else + ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); + return 0; +} + +static int ak4535_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 i; + + switch (level) { + case SND_SOC_BIAS_ON: + ak4535_mute(codec->dai, 0); + break; + case SND_SOC_BIAS_PREPARE: + ak4535_mute(codec->dai, 1); + break; + case SND_SOC_BIAS_STANDBY: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i | 0x80); + i = ak4535_read_reg_cache(codec, AK4535_PM2); + ak4535_write(codec, AK4535_PM2, i & (~0x80)); + break; + case SND_SOC_BIAS_OFF: + i = ak4535_read_reg_cache(codec, AK4535_PM1); + ak4535_write(codec, AK4535_PM1, i & (~0x80)); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4535_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + +struct snd_soc_codec_dai ak4535_dai = { + .name = "AK4535", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4535_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .hw_params = ak4535_hw_params, + }, + .dai_ops = { + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, + }, +}; +EXPORT_SYMBOL_GPL(ak4535_dai); + +static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ak4535_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + ak4535_sync(codec); + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + ak4535_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialise the AK4535 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4535_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "AK4535"; + codec->owner = THIS_MODULE; + codec->read = ak4535_read_reg_cache; + codec->write = ak4535_write; + codec->set_bias_level = ak4535_set_bias_level; + codec->dai = &ak4535_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(ak4535_reg); + codec->reg_cache = kmemdup(ak4535_reg, sizeof(ak4535_reg), GFP_KERNEL); + + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ak4535_add_controls(codec); + ak4535_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4535: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + + return ret; +} + +static struct snd_soc_device *ak4535_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */ + +static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; + +/* Magic definition of all other variables and things */ +I2C_CLIENT_INSMOD; + +static struct i2c_driver ak4535_i2c_driver; +static struct i2c_client client_template; + +/* If the i2c layer weren't so broken, we could pass this kind of data + around */ +static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) +{ + struct snd_soc_device *socdev = ak4535_socdev; + struct ak4535_setup_data *setup = socdev->codec_data; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c; + int ret; + + if (addr != setup->i2c_address) + return -ENODEV; + + client_template.adapter = adap; + client_template.addr = addr; + + i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); + if (i2c == NULL) { + kfree(codec); + return -ENOMEM; + } + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = i2c_attach_client(i2c); + if (ret < 0) { + printk(KERN_ERR "failed to attach codec at addr %x\n", addr); + goto err; + } + + ret = ak4535_init(socdev); + if (ret < 0) { + printk(KERN_ERR "failed to initialise AK4535\n"); + goto err; + } + return ret; + +err: + kfree(codec); + kfree(i2c); + return ret; +} + +static int ak4535_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + i2c_detach_client(client); + kfree(codec->reg_cache); + kfree(client); + return 0; +} + +static int ak4535_i2c_attach(struct i2c_adapter *adap) +{ + return i2c_probe(adap, &addr_data, ak4535_codec_probe); +} + +/* corgi i2c codec control layer */ +static struct i2c_driver ak4535_i2c_driver = { + .driver = { + .name = "AK4535 I2C Codec", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_AK4535, + .attach_adapter = ak4535_i2c_attach, + .detach_client = ak4535_i2c_detach, + .command = NULL, +}; + +static struct i2c_client client_template = { + .name = "AK4535", + .driver = &ak4535_i2c_driver, +}; +#endif + +static int ak4535_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct ak4535_setup_data *setup; + struct snd_soc_codec *codec; + struct ak4535_priv *ak4535; + int ret = 0; + + printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + ak4535 = kzalloc(sizeof(struct ak4535_priv), GFP_KERNEL); + if (ak4535 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = ak4535; + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ak4535_socdev = socdev; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + normal_i2c[0] = setup->i2c_address; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = (hw_read_t)i2c_master_recv; + ret = i2c_add_driver(&ak4535_i2c_driver); + if (ret != 0) + printk(KERN_ERR "can't add i2c driver"); + } +#else + /* Add other interfaces here */ +#endif + return ret; +} + +/* power down chip */ +static int ak4535_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4535_i2c_driver); +#endif + kfree(codec->private_data); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4535 = { + .probe = ak4535_probe, + .remove = ak4535_remove, + .suspend = ak4535_suspend, + .resume = ak4535_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); + +MODULE_DESCRIPTION("Soc AK4535 driver"); +MODULE_AUTHOR("Richard Purdie"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h new file mode 100644 index 0000000..fc686dd --- /dev/null +++ b/sound/soc/codecs/ak4535.h @@ -0,0 +1,46 @@ +/* + * ak4535.h -- AK4535 Soc Audio driver + * + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie <richard(a)openedhand.com> + * + * Based on wm8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4535_H +#define _AK4535_H + +/* AK4535 register space */ + +#define AK4535_PM1 0x0 +#define AK4535_PM2 0x1 +#define AK4535_SIG1 0x2 +#define AK4535_SIG2 0x3 +#define AK4535_MODE1 0x4 +#define AK4535_MODE2 0x5 +#define AK4535_DAC 0x6 +#define AK4535_MIC 0x7 +#define AK4535_TIMER 0x8 +#define AK4535_ALC1 0x9 +#define AK4535_ALC2 0xa +#define AK4535_PGA 0xb +#define AK4535_LATT 0xc +#define AK4535_RATT 0xd +#define AK4535_VOL 0xe +#define AK4535_STATUS 0xf + +#define AK4535_CACHEREGNUM 0x10 + +struct ak4535_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_codec_dai ak4535_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4535; + +#endif -- 1.5.6

1 1

[alsa-devel] [Fwd: [Alsa-user] Intel Poulsbo(SCH) chipset does not recognize Adi 1986 codec under linux]
by stan 23 Jun '08

23 Jun '08

Sent to wrong list.

1 0

[alsa-devel] [PATCH] Added SSID for 'Fujitsu Siemens Amilo M1451G' laptop
by Travis Place 23 Jun '08

23 Jun '08

Changelog: Add the SSID for the "Fujitsu Siemens Amilo M1451G" laptop to patch_realtek.c , so that it uses ALC880_FUJITSU by default. --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -3074,6 +3074,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), + SND_PCI_QUIRK(0x1734, 0x1094, "Fujitsu Siemens Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC", ALC880_UNIWILL), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x1854, 0x0018, "LG LW20", ALC880_LG_LW), Signed-off-by: Travis Place <wishie(a)wishie.net>

2 1

[alsa-devel] JIVE: Add ASoC audio support
by Ben Dooks 23 Jun '08

23 Jun '08

Add support for the Jive's WM8750 codec attached to the S3C2412 IIS controller. Signed-off-by: Ben Dooks <ben-linux(a)fluff.org> Index: linux-2.6.26-rc5-quilt2/sound/soc/s3c24xx/Kconfig =================================================================== --- linux-2.6.26-rc5-quilt2.orig/sound/soc/s3c24xx/Kconfig 2008-06-10 12:31:40.000000000 +0100 +++ linux-2.6.26-rc5-quilt2/sound/soc/s3c24xx/Kconfig 2008-06-12 11:07:04.000000000 +0100 @@ -28,6 +28,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753 Say Y if you want to add support for SoC audio on smdk2440 with the WM8753. +config SND_S3C24XX_SOC_JIVE_WM8750 + tristate "SoC I2S Audio support for Jive" + depends on SND_S3C24XX_SOC && MACH_JIVE + select SND_SOC_WM8750 + select SND_SOC_WM8750_SPI + select SND_S3C2412_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the Jive. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 Index: linux-2.6.26-rc5-quilt2/sound/soc/s3c24xx/Makefile =================================================================== --- linux-2.6.26-rc5-quilt2.orig/sound/soc/s3c24xx/Makefile 2008-06-10 12:31:40.000000000 +0100 +++ linux-2.6.26-rc5-quilt2/sound/soc/s3c24xx/Makefile 2008-06-12 11:07:04.000000000 +0100 @@ -10,10 +10,12 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += sn obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o # S3C24XX Machine Support +snd-soc-jive-wm8750-objs := jive_wm8750.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o Index: linux-2.6.26-rc5-quilt2/sound/soc/s3c24xx/jive_wm8750.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ linux-2.6.26-rc5-quilt2/sound/soc/s3c24xx/jive_wm8750.c 2008-06-12 11:07:04.000000000 +0100 @@ -0,0 +1,266 @@ +/* sound/soc/s3c24xx/jive_wm8705.c + * + * Copyright 2007 Simtec Electronics + * + * Based on sound/soc/pxa/spitz.c + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "s3c24xx-pcm.h" +#include "s3c2412-i2s.h" + +#include "../codecs/wm8750.h" + +static const char *audio_map[][2] = { + [0] = { "Headphone Jack", "LOUT1" }, + [1] = { "Headphone Jack", "ROUT1" }, + [2] = { "Internal Speaker", "LOUT2" }, + [3] = { "Internal Speaker", "ROUT2" }, + [4] = { "LINPUT1", "Line Input" }, + [5] = { "RINPUT1", "Line Input" }, + [6] = { NULL, NULL }, +}; + +static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Internal Speaker", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static int jive_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->socdev->codec; + + snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Internal Speaker", 1); + snd_soc_dapm_set_endpoint(codec, "Line In", 1); + + snd_soc_dapm_sync_endpoints(codec); + + return 0; +} + +static inline int codec_set_dai_fmt(struct snd_soc_codec_dai *dai, + unsigned int format) +{ + return dai->dai_ops.set_fmt(dai, format); +} + +static inline int codec_set_dai_sysclk(struct snd_soc_codec_dai *dai, + unsigned int clk, + unsigned int arg, + unsigned int arg2) +{ + return dai->dai_ops.set_sysclk(dai, clk, arg, arg2); +} + +static inline int cpu_set_dai_fmt(struct snd_soc_cpu_dai *dai, + unsigned int format) +{ + return dai->dai_ops.set_fmt(dai, format); +} + +static inline int cpu_set_dai_sysclk(struct snd_soc_cpu_dai *dai, + unsigned int clk, + unsigned int arg, + unsigned int arg2) +{ + return dai->dai_ops.set_sysclk(dai, clk, arg, arg2); +} + +static inline int cpu_set_dai_clkdiv(struct snd_soc_cpu_dai *dai, + unsigned int clk, + unsigned int arg) +{ + return dai->dai_ops.set_clkdiv(dai, clk, arg); +} + + + +static int jive_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c2412_rate_calc div; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + s3c2412_iis_calc_rate(&div, NULL, params_rate(params), + s3c2412_get_iisclk()); + + /* set codec DAI configuration */ + ret = codec_set_dai_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = cpu_set_dai_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_set_dai_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_set_dai_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + ret = cpu_set_dai_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops jive_ops = { + .startup = jive_startup, + .hw_params = jive_hw_params, +}; + +static const struct snd_kcontrol_new wm8750_jive_controls[] = { + +}; + +static int jive_wm8750_init(struct snd_soc_codec *codec) +{ + int i, err; + + /* These endpoints are not being used. */ + snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); + snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); + snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); + snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); + snd_soc_dapm_set_endpoint(codec, "OUT3", 0); + snd_soc_dapm_set_endpoint(codec, "MONO", 0); + + /* Add jive specific controls */ + for (i = 0; i < ARRAY_SIZE(wm8750_jive_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8750_jive_controls[i], codec, NULL)); + if (err < 0) + return err; + } + + /* Add jive specific widgets */ + for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { + snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); + } + + /* Set up jive specific audio path audio_map */ + for (i = 0; audio_map[i][0] != NULL; i++) { + printk(KERN_INFO "mapping %s => %s\n", audio_map[i][0], audio_map[i][1]); + snd_soc_dapm_connect_input(codec, audio_map[i][0], NULL, + audio_map[i][1]); + } + + snd_soc_dapm_sync_endpoints(codec); + return 0; +} + +static struct snd_soc_dai_link jive_dai = { + .name = "wm8750", + .stream_name = "WM8750", + .cpu_dai = &s3c2412_i2s_dai, + .codec_dai = &wm8750_dai, + .init = jive_wm8750_init, + .ops = &jive_ops, +}; + +/* jive audio machine driver */ +static struct snd_soc_machine snd_soc_machine_jive = { + .name = "Jive", + .dai_link = &jive_dai, + .num_links = 1, +}; + +/* jive audio private data */ +static struct wm8750_setup_data jive_wm8750_setup = { +}; + +/* jive audio subsystem */ +static struct snd_soc_device jive_snd_devdata = { + .machine = &snd_soc_machine_jive, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_wm8750_spi, + .codec_data = &jive_wm8750_setup, +}; + +static struct platform_device *jive_snd_device; + +static int __init jive_init(void) +{ + int ret; + + printk("JIVE WM8750 Audio support\n"); + + if (!machine_is_jive()) + return 0; + + jive_snd_device = platform_device_alloc("soc-audio", -1); + if (!jive_snd_device) + return -ENOMEM; + + platform_set_drvdata(jive_snd_device, &jive_snd_devdata); + jive_snd_devdata.dev = &jive_snd_device->dev; + ret = platform_device_add(jive_snd_device); + + if (ret) + platform_device_put(jive_snd_device); + + return ret; +} + +static void __exit jive_exit(void) +{ + platform_device_unregister(jive_snd_device); +} + +module_init(jive_init); +module_exit(jive_exit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Jive"); +MODULE_LICENSE("GPL"); -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] Hoping to enhance cs5535 driver - initial question
by Vince Negri 23 Jun '08

23 Jun '08

Hi All, I'm looking to enhance the cs5535 driver (Geode companion device) so that it can support 5.1 operation when used with an appropriate AC97 codec. AFAIK, the existing ALSA driver only supports 2-channel operation. Having looked at the existing driver, and Takashi Iwai's guide, there is still something I'm not too clear on - I hope you can set me straight. According to the AMD datasheet, when used in 5.1 mode the cs5535 makes use of 4 distinct DMAs - the first (BM0) is the main L/R pair, and this is the one used by the existing driver. There is another DMA for L/R surround, another for the centre channel, and the fourth DMA feeds the LFE channel. Two stereos and two monos. I'm comfortable with the low-level bit bashing: what I'm unsure of is what is necessary to represent all of this to the upper layers so that the ALSA clients see me as a 6-channel device. Do I need to represent the 4 DMAs as substreams of the playback pcm, for example? Or have I completely misunderstood what a substream is? Alternatively, pointers to an existing driver that faces the same issue would be very welcome. Vince

2 1

[alsa-devel] sb_mixer.c/save_mixer() awfully buggy?
by Andreas Mohr 23 Jun '08

23 Jun '08

Hi all, with my ALS4000, I'm hitting this upon resume: ALSA sound/isa/sb/sb_mixer.c:936: BUG? (num_regs > (sizeof(chip->saved_regs) / sizeof( (chip->saved_regs)[0]) + (sizeof(char[1 - 2 * !!(__builtin_types_compatible_p(typeof(c hip->saved_regs), typeof(&chip->saved_regs[0])))]) - 1))) Pid: 4498, comm: my_sleep Not tainted 2.6.26-rc3 #1 [<e097ac8c>] restore_mixer+0x6c/0x70 [snd_sb_common] [<e097ad07>] snd_sbmixer_resume+0x77/0x80 [snd_sb_common] [<e098030a>] snd_als4000_resume+0x5a/0xd0 [snd_als4000] [<c0281a72>] pci_device_resume+0x22/0x60 [<c0303e76>] device_resume+0x86/0x100 [<c0145bb7>] hibernation_snapshot+0xc7/0x180 [<c0144f9f>] ? freeze_processes+0x3f/0x80 [<c0145d4c>] hibernate+0xdc/0x180 [<c0144980>] ? state_store+0x0/0xd0 [<c0144a3f>] state_store+0xbf/0xd0 [<c0144980>] ? state_store+0x0/0xd0 [<c02743d4>] kobj_attr_store+0x24/0x30 [<c01aaf5b>] sysfs_write_file+0xbb/0x110 [<c016ee56>] vfs_write+0x96/0x130 [<c01aaea0>] ? sysfs_write_file+0x0/0x110 [<c016f3cd>] sys_write+0x3d/0x70 [<c0103026>] syscall_call+0x7/0xb ======================= triggered by this line: static void restore_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs) { unsigned char *val = chip->saved_regs; snd_assert(num_regs > ARRAY_SIZE(chip->saved_regs), return); for (; num_regs; num_regs--) snd_sbmixer_write(chip, *regs++, *val++); } Uhmm, forgive my utterly dumb state of mind, but isn't this assert _exactly_ wrong? We assert that the number of regs to copy is _higher_ than the array size of the content we want to copy, IOW we want to make _certain_ that we'll hit a nice array overflow or segmentation violation later. Doesn't make too much - well, exactly zero, zilch - sense to me. Dito save_mixer(). Forgive me if this is already fixed in ALSA head (still exists in current 2.6.26-rc7). Sounds like 2.6.26(!) material to me if it is indeed buggy. Thanks, Andreas Mohr

2 1

[alsa-devel] note on Suse 10.3 build
by randall 23 Jun '08

23 Jun '08

To all: Had to rebuild the Suse 10.3 kernel again to get back the NVidia MCP61 (RealTek ALC662) on board sound. Turned on two SND parameters in the kernel config file: CONFIG_SND_DEBUG=y CONFIG_SND_DEBUG_DETECT=y and enabled a module: CONFIG_SND_SOC=m The kernal is 2.6.22.18-0.2 x86_64 architecture. The sound file under /etc/modprobe.d is ------------------- snip ---------------------- options snd-hda-intel enable=1 index=0 model=3stack-6ch-dig position_fix=1 # CvwD.9hpqWdunz16:MCP61 High Definition Audio alias snd-card-0 snd-hda-intel ------------- end snippet ------------------- and the 3stack 6 channel SPIF digital setting for the model type works for the NVidia MCP61 setup. There were NO changes to the modprobe.conf file. Not sure why the kernel has to be rebuilt to restore the onboard sound, any ideas? The 3 changes are the only ones I've made, when running menuconfig. Motherboard is Elite Computer Systems (ECS) GF6100-M754 with NVidia GeForce6100/nForce 405 running the AMD Athlon 64 3200+ processor. and hwinfo --sound 18: PCI 05.0: 0403 Audio device [Created at pci.301] UDI: /org/freedesktop/Hal/devices/pci_10de_3f0 Unique ID: CvwD.9hpqWdunz16 SysFS ID: /devices/pci0000:00/0000:00:05.0 SysFS BusID: 0000:00:05.0 Hardware Class: sound Model: "Elitegroup MCP61 High Definition Audio" Vendor: pci 0x10de "nVidia Corporation" Device: pci 0x03f0 "MCP61 High Definition Audio" SubVendor: pci 0x1019 "Elitegroup Computer Systems" SubDevice: pci 0x2151 Revision: 0xa2 Driver: "HDA Intel" Driver Modules: "snd_hda_intel" Memory Range: 0xfe028000-0xfe02bfff (rw,non-prefetchable) IRQ: 22 (7841 events) Module Alias: "pci:v000010DEd000003F0sv00001019sd00002151bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=no, avail=yes, need=no, active=unknown Again, any ideas as to why the kernel has to be regenerated to enable on-board sound? - Randall

2 3

[alsa-devel] asoc-v2 ac97-wm9713 suspend/resume
by Oliver Ford 23 Jun '08

23 Jun '08

I have been porting linux to the iPAQ 214 which uses a wm9713 codec on the ac97 of a pxa310. It uses it for audio and touchscreen. I'm working from the git tree of arm.linux.org.uk pxa branch and have merged in the asoc-v2-dev of wolfson-asoc (all last thursday). I have the audio and touchscreen working via a platform driver I've put together (sound/soc/pxa/hpipaq214-audio.c) from copying sound/soc/pxa/zylonite.c etc a few months ago. I'm now trying to get the audio working through suspend/resume. To save as much power as possible in D3 suspend the wm9713 power is completely turned off in hpipaq214-audio.c via a gpio. This is in suspend_late() and resume_early() to make sure the power is back before the AC97/WM9713 drivers try to wake stuff up. To make the touchscreen work after suspend I had to add a cold reset to sound/soc/wm9713.c : wm9713_resume(). Also I have to remove the calls to snd_soc_card_suspend_pcms() and snd_soc_card_resume_pcms() from my hpipaq214-audio.c's normal suspend/resume functions. These calls were originally in sound/soc/pxa/zylonite.c but I notice there is now no suspend/resume functions in there. Some of the other platform audio drivers call snd_soc_suspend_pcms() some call snd_soc_suspend() and some snd_soc_card_suspend_pcms(). Only snd_soc_suspend_pcms() seems to actually exists. If I do actually call snd_soc_card_suspend/resume_pcms() then any use of the audio or touchscreen later on freezes the process doing it. Which am I actually supposed to do, and is there a more 'proper' way way of bringing the wm9713 back up from complete power down on resume? Oliver Ford

3 4

[alsa-devel] [PATCH 2/2] wss_lib: rename cs4231.h into wss.h
by Krzysztof Helt 23 Jun '08

23 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Rename file include/sound/cs4231.h into include/sound/wss.h This is a step toward merging all libraries for Windows Sound System compatible chips into a single library. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch does not pass the checkpatch check. It is done to preserve "move" operation, so the old and a new file are identical and differ only by name. diff -urpN linux-alsa/include/sound/cs4231.h linux-mm/include/sound/cs4231.h --- linux-alsa/include/sound/cs4231.h 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/include/sound/cs4231.h 1970-01-01 01:00:00.000000000 +0100 @@ -1,175 +0,0 @@ -#ifndef __SOUND_CS4231_H -#define __SOUND_CS4231_H - -/* - * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * Definitions for CS4231 & InterWave chips & compatible chips - * - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - */ - -#include "control.h" -#include "pcm.h" -#include "timer.h" - -#include "cs4231-regs.h" - -/* defines for codec.mode */ - -#define CS4231_MODE_NONE 0x0000 -#define CS4231_MODE_PLAY 0x0001 -#define CS4231_MODE_RECORD 0x0002 -#define CS4231_MODE_TIMER 0x0004 -#define CS4231_MODE_OPEN (CS4231_MODE_PLAY|CS4231_MODE_RECORD|CS4231_MODE_TIMER) - -/* defines for codec.hardware */ - -#define CS4231_HW_DETECT 0x0000 /* let CS4231 driver detect chip */ -#define CS4231_HW_DETECT3 0x0001 /* allow mode 3 */ -#define CS4231_HW_TYPE_MASK 0xff00 /* type mask */ -#define CS4231_HW_CS4231_MASK 0x0100 /* CS4231 serie */ -#define CS4231_HW_CS4231 0x0100 /* CS4231 chip */ -#define CS4231_HW_CS4231A 0x0101 /* CS4231A chip */ -#define CS4231_HW_AD1845 0x0102 /* AD1845 chip */ -#define CS4231_HW_CS4232_MASK 0x0200 /* CS4232 serie (has control ports) */ -#define CS4231_HW_CS4232 0x0200 /* CS4232 */ -#define CS4231_HW_CS4232A 0x0201 /* CS4232A */ -#define CS4231_HW_CS4236 0x0202 /* CS4236 */ -#define CS4231_HW_CS4236B_MASK 0x0400 /* CS4236B serie (has extended control regs) */ -#define CS4231_HW_CS4235 0x0400 /* CS4235 - Crystal Clear (tm) stereo enhancement */ -#define CS4231_HW_CS4236B 0x0401 /* CS4236B */ -#define CS4231_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ -#define CS4231_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ -#define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ -/* compatible, but clones */ -#define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ -#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ -#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ - -/* defines for codec.hwshare */ -#define CS4231_HWSHARE_IRQ (1<<0) -#define CS4231_HWSHARE_DMA1 (1<<1) -#define CS4231_HWSHARE_DMA2 (1<<2) - -struct snd_cs4231 { - unsigned long port; /* base i/o port */ - struct resource *res_port; - unsigned long cport; /* control base i/o port (CS4236) */ - struct resource *res_cport; - int irq; /* IRQ line */ - int dma1; /* playback DMA */ - int dma2; /* record DMA */ - unsigned short version; /* version of CODEC chip */ - unsigned short mode; /* see to CS4231_MODE_XXXX */ - unsigned short hardware; /* see to CS4231_HW_XXXX */ - unsigned short hwshare; /* shared resources */ - unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ - ebus_flag:1; /* SPARC: EBUS present */ - - struct snd_card *card; - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - struct snd_timer *timer; - - unsigned char image[32]; /* registers image */ - unsigned char eimage[32]; /* extended registers image */ - unsigned char cimage[16]; /* control registers image */ - int mce_bit; - int calibrate_mute; - int sw_3d_bit; - unsigned int p_dma_size; - unsigned int c_dma_size; - - spinlock_t reg_lock; - struct mutex mce_mutex; - struct mutex open_mutex; - - int (*rate_constraint) (struct snd_pcm_runtime *runtime); - void (*set_playback_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char pdfr); - void (*set_capture_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char cdfr); - void (*trigger) (struct snd_cs4231 *chip, unsigned int what, int start); -#ifdef CONFIG_PM - void (*suspend) (struct snd_cs4231 *chip); - void (*resume) (struct snd_cs4231 *chip); -#endif - void *dma_private_data; - int (*claim_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); - int (*release_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); -}; - -/* exported functions */ - -void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); -unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg); -void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); -unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg); -void snd_cs4231_mce_up(struct snd_cs4231 *chip); -void snd_cs4231_mce_down(struct snd_cs4231 *chip); - -void snd_cs4231_overrange(struct snd_cs4231 *chip); - -irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); - -const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); - -int snd_cs4231_create(struct snd_card *card, - unsigned long port, - unsigned long cport, - int irq, int dma1, int dma2, - unsigned short hardware, - unsigned short hwshare, - struct snd_cs4231 ** rchip); -int snd_cs4231_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); -int snd_cs4231_timer(struct snd_cs4231 * chip, int device, struct snd_timer **rtimer); -int snd_cs4231_mixer(struct snd_cs4231 * chip); - -int snd_cs4236_create(struct snd_card *card, - unsigned long port, - unsigned long cport, - int irq, int dma1, int dma2, - unsigned short hardware, - unsigned short hwshare, - struct snd_cs4231 ** rchip); -int snd_cs4236_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); -int snd_cs4236_mixer(struct snd_cs4231 * chip); - -/* - * mixer library - */ - -#define CS4231_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_single, \ - .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - -#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_double, \ - .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } - -int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); -int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); - -#endif /* __SOUND_CS4231_H */ diff -urpN linux-alsa/include/sound/snd_wavefront.h linux-mm/include/sound/snd_wavefront.h --- linux-alsa/include/sound/snd_wavefront.h 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/include/sound/snd_wavefront.h 2008-06-15 23:04:42.657642437 +0200 @@ -1,7 +1,6 @@ #ifndef __SOUND_SND_WAVEFRONT_H__ #define __SOUND_SND_WAVEFRONT_H__ -#include "cs4231.h" #include "mpu401.h" #include "hwdep.h" #include "rawmidi.h" diff -urpN linux-alsa/include/sound/wss.h linux-mm/include/sound/wss.h --- linux-alsa/include/sound/wss.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-mm/include/sound/wss.h 2008-06-14 00:05:02.000000000 +0200 @@ -0,0 +1,175 @@ +#ifndef __SOUND_CS4231_H +#define __SOUND_CS4231_H + +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Definitions for CS4231 & InterWave chips & compatible chips + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include "control.h" +#include "pcm.h" +#include "timer.h" + +#include "cs4231-regs.h" + +/* defines for codec.mode */ + +#define CS4231_MODE_NONE 0x0000 +#define CS4231_MODE_PLAY 0x0001 +#define CS4231_MODE_RECORD 0x0002 +#define CS4231_MODE_TIMER 0x0004 +#define CS4231_MODE_OPEN (CS4231_MODE_PLAY|CS4231_MODE_RECORD|CS4231_MODE_TIMER) + +/* defines for codec.hardware */ + +#define CS4231_HW_DETECT 0x0000 /* let CS4231 driver detect chip */ +#define CS4231_HW_DETECT3 0x0001 /* allow mode 3 */ +#define CS4231_HW_TYPE_MASK 0xff00 /* type mask */ +#define CS4231_HW_CS4231_MASK 0x0100 /* CS4231 serie */ +#define CS4231_HW_CS4231 0x0100 /* CS4231 chip */ +#define CS4231_HW_CS4231A 0x0101 /* CS4231A chip */ +#define CS4231_HW_AD1845 0x0102 /* AD1845 chip */ +#define CS4231_HW_CS4232_MASK 0x0200 /* CS4232 serie (has control ports) */ +#define CS4231_HW_CS4232 0x0200 /* CS4232 */ +#define CS4231_HW_CS4232A 0x0201 /* CS4232A */ +#define CS4231_HW_CS4236 0x0202 /* CS4236 */ +#define CS4231_HW_CS4236B_MASK 0x0400 /* CS4236B serie (has extended control regs) */ +#define CS4231_HW_CS4235 0x0400 /* CS4235 - Crystal Clear (tm) stereo enhancement */ +#define CS4231_HW_CS4236B 0x0401 /* CS4236B */ +#define CS4231_HW_CS4237B 0x0402 /* CS4237B - SRS 3D */ +#define CS4231_HW_CS4238B 0x0403 /* CS4238B - QSOUND 3D */ +#define CS4231_HW_CS4239 0x0404 /* CS4239 - Crystal Clear (tm) stereo enhancement */ +/* compatible, but clones */ +#define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ +#define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ + +/* defines for codec.hwshare */ +#define CS4231_HWSHARE_IRQ (1<<0) +#define CS4231_HWSHARE_DMA1 (1<<1) +#define CS4231_HWSHARE_DMA2 (1<<2) + +struct snd_cs4231 { + unsigned long port; /* base i/o port */ + struct resource *res_port; + unsigned long cport; /* control base i/o port (CS4236) */ + struct resource *res_cport; + int irq; /* IRQ line */ + int dma1; /* playback DMA */ + int dma2; /* record DMA */ + unsigned short version; /* version of CODEC chip */ + unsigned short mode; /* see to CS4231_MODE_XXXX */ + unsigned short hardware; /* see to CS4231_HW_XXXX */ + unsigned short hwshare; /* shared resources */ + unsigned short single_dma:1, /* forced single DMA mode (GUS 16-bit daughter board) or dma1 == dma2 */ + ebus_flag:1; /* SPARC: EBUS present */ + + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + struct snd_timer *timer; + + unsigned char image[32]; /* registers image */ + unsigned char eimage[32]; /* extended registers image */ + unsigned char cimage[16]; /* control registers image */ + int mce_bit; + int calibrate_mute; + int sw_3d_bit; + unsigned int p_dma_size; + unsigned int c_dma_size; + + spinlock_t reg_lock; + struct mutex mce_mutex; + struct mutex open_mutex; + + int (*rate_constraint) (struct snd_pcm_runtime *runtime); + void (*set_playback_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char pdfr); + void (*set_capture_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char cdfr); + void (*trigger) (struct snd_cs4231 *chip, unsigned int what, int start); +#ifdef CONFIG_PM + void (*suspend) (struct snd_cs4231 *chip); + void (*resume) (struct snd_cs4231 *chip); +#endif + void *dma_private_data; + int (*claim_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); + int (*release_dma) (struct snd_cs4231 *chip, void *dma_private_data, int dma); +}; + +/* exported functions */ + +void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); +unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg); +void snd_cs4236_ext_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char val); +unsigned char snd_cs4236_ext_in(struct snd_cs4231 *chip, unsigned char reg); +void snd_cs4231_mce_up(struct snd_cs4231 *chip); +void snd_cs4231_mce_down(struct snd_cs4231 *chip); + +void snd_cs4231_overrange(struct snd_cs4231 *chip); + +irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); + +const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); + +int snd_cs4231_create(struct snd_card *card, + unsigned long port, + unsigned long cport, + int irq, int dma1, int dma2, + unsigned short hardware, + unsigned short hwshare, + struct snd_cs4231 ** rchip); +int snd_cs4231_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); +int snd_cs4231_timer(struct snd_cs4231 * chip, int device, struct snd_timer **rtimer); +int snd_cs4231_mixer(struct snd_cs4231 * chip); + +int snd_cs4236_create(struct snd_card *card, + unsigned long port, + unsigned long cport, + int irq, int dma1, int dma2, + unsigned short hardware, + unsigned short hwshare, + struct snd_cs4231 ** rchip); +int snd_cs4236_pcm(struct snd_cs4231 * chip, int device, struct snd_pcm **rpcm); +int snd_cs4236_mixer(struct snd_cs4231 * chip); + +/* + * mixer library + */ + +#define CS4231_SINGLE(xname, xindex, reg, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_cs4231_info_single, \ + .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } + +int snd_cs4231_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_cs4231_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_cs4231_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); + +#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_cs4231_info_double, \ + .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } + +int snd_cs4231_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); +int snd_cs4231_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_cs4231_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); + +#endif /* __SOUND_CS4231_H */ diff -urpN linux-alsa/sound/isa/azt2320.c linux-mm/sound/isa/azt2320.c --- linux-alsa/sound/isa/azt2320.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/azt2320.c 2008-06-15 23:03:04.309647551 +0200 @@ -38,7 +38,7 @@ #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/initval.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl3.h> diff -urpN linux-alsa/sound/isa/cs423x/cs4231.c linux-mm/sound/isa/cs423x/cs4231.c --- linux-alsa/sound/isa/cs423x/cs4231.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/cs423x/cs4231.c 2008-06-15 23:02:29.865646338 +0200 @@ -27,7 +27,7 @@ #include <linux/wait.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> diff -urpN linux-alsa/sound/isa/cs423x/cs4236.c linux-mm/sound/isa/cs423x/cs4236.c --- linux-alsa/sound/isa/cs423x/cs4236.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/cs423x/cs4236.c 2008-06-15 23:02:54.801647075 +0200 @@ -26,7 +26,7 @@ #include <linux/pnp.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl3.h> #include <sound/initval.h> diff -urpN linux-alsa/sound/isa/cs423x/cs4236_lib.c linux-mm/sound/isa/cs423x/cs4236_lib.c --- linux-alsa/sound/isa/cs423x/cs4236_lib.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/cs423x/cs4236_lib.c 2008-06-15 23:02:45.493647684 +0200 @@ -85,7 +85,7 @@ #include <linux/time.h> #include <linux/wait.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/asoundef.h> MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); diff -urpN linux-alsa/sound/isa/gus/gusmax.c linux-mm/sound/isa/gus/gusmax.c --- linux-alsa/sound/isa/gus/gusmax.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/gus/gusmax.c 2008-06-15 23:01:38.317642279 +0200 @@ -28,7 +28,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/gus.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> diff -urpN linux-alsa/sound/isa/gus/interwave.c linux-mm/sound/isa/gus/interwave.c --- linux-alsa/sound/isa/gus/interwave.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/gus/interwave.c 2008-06-15 23:01:47.769643095 +0200 @@ -32,7 +32,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/gus.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #ifdef SNDRV_STB #include <sound/tea6330t.h> #endif diff -urpN linux-alsa/sound/isa/opl3sa2.c linux-mm/sound/isa/opl3sa2.c --- linux-alsa/sound/isa/opl3sa2.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/opl3sa2.c 2008-06-15 23:01:58.585642441 +0200 @@ -28,7 +28,7 @@ #include <linux/pnp.h> #include <linux/moduleparam.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl3.h> #include <sound/initval.h> diff -urpN linux-alsa/sound/isa/opti9xx/miro.c linux-mm/sound/isa/opti9xx/miro.c --- linux-alsa/sound/isa/opti9xx/miro.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/opti9xx/miro.c 2008-06-15 23:03:49.161643750 +0200 @@ -32,7 +32,7 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/opl4.h> #include <sound/control.h> diff -urpN linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c linux-mm/sound/isa/opti9xx/opti92x-ad1848.c --- linux-alsa/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/opti9xx/opti92x-ad1848.c 2008-06-15 23:01:29.393645689 +0200 @@ -34,7 +34,7 @@ #include <asm/dma.h> #include <sound/core.h> #if defined(CS4231) || defined(OPTi93X) -#include <sound/cs4231.h> +#include <sound/wss.h> #else #include <sound/ad1848.h> #endif /* CS4231 */ diff -urpN linux-alsa/sound/isa/sscape.c linux-mm/sound/isa/sscape.c --- linux-alsa/sound/isa/sscape.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/sscape.c 2008-06-15 23:01:15.653643218 +0200 @@ -31,7 +31,7 @@ #include <asm/dma.h> #include <sound/core.h> #include <sound/hwdep.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> diff -urpN linux-alsa/sound/isa/wavefront/wavefront.c linux-mm/sound/isa/wavefront/wavefront.c --- linux-alsa/sound/isa/wavefront/wavefront.c 2008-06-14 00:05:02.000000000 +0200 +++ linux-mm/sound/isa/wavefront/wavefront.c 2008-06-15 23:04:59.941642387 +0200 @@ -29,6 +29,7 @@ #include <sound/core.h> #include <sound/initval.h> #include <sound/opl3.h> +#include <sound/wss.h> #include <sound/snd_wavefront.h> MODULE_AUTHOR("Paul Barton-Davis <pbd(a)op.net>"); diff -urpN linux-alsa/sound/isa/wss/wss_lib.c linux-mm/sound/isa/wss/wss_lib.c --- linux-alsa/sound/isa/wss/wss_lib.c 2008-06-15 23:05:34.053647670 +0200 +++ linux-mm/sound/isa/wss/wss_lib.c 2008-06-15 23:00:53.449648387 +0200 @@ -31,7 +31,7 @@ #include <linux/slab.h> #include <linux/ioport.h> #include <sound/core.h> -#include <sound/cs4231.h> +#include <sound/wss.h> #include <sound/pcm_params.h> #include <asm/io.h> ---------------------------------------------------------------------- Gdzie warto jechac na urlop? Sprawdz! kliknij >>> http://link.interia.pl/f1e32

2 1

[alsa-devel] Repaired version ([PATCH] PCI168 snd-azt3328: some more fixups)
by Andreas Mohr 23 Jun '08

23 Jun '08

Hi, update: darn, I just noticed that those "unknown" play/rec IRQ types actually do turn up on dmesg _sometimes_ (a bit too often), thus a printk KERN_WARNING wasn't quite such a good idea... (revert it to optional debug logging, will possibly figure out its meaning later) ---- another update: - fix problem with codec register 0x6a being write-only by adding a software shadow register (caused annoying noise after module loading due to _toggling_ between gameport and audio bits instead of configuring them properly) - rename several "Wave" mixer controls to "PCM", since this is what Wine and several other apps are looking for (IOW, _requiring_) and this is what AC97 specs use as naming, too, thus I'd guess it's what these controls are - cleanup, small optimizations Patch diffed against patched 2.6.26-rc6 (patched with my last patch submission, that is, IOW it should apply to current ALSA tree easily). checkpatch.pl tested, runtime tested. Hopefully this will come in time for 1.0.17 and/or the next kernel merge before my previous patch hits the streets ;) (to have this all in one go) Thanks! Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- azt3328.c.my_org 2008-06-14 08:20:36.000000000 +0200 +++ azt3328.c 2008-06-22 23:02:57.000000000 +0200 @@ -289,6 +289,12 @@ struct pci_dev *pci; int irq; + /* register 0x6a is write-only, thus need to remember setting. + * If we need to add more registers here, then we might try to fold this + * into some transparent combined shadow register handling with + * CONFIG_PM register storage below, but that's slightly difficult. */ + u16 shadow_reg_codec_6AH; + #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ @@ -324,21 +330,6 @@ return 0; } -static int -snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set) -{ - u16 prev = inw(reg), new; - - new = (do_set) ? (prev|mask) : (prev & ~mask); - /* we need to always write the new value no matter whether it differs or not, - * since some register bits don't indicate their setting */ - outw(new, reg); - if (new != prev) - return 1; - - return 0; -} - static inline void snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { @@ -662,7 +653,7 @@ "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; - const char *p = NULL; + const char * const *p = NULL; snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -673,20 +664,20 @@ if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - p = texts1[uinfo->value.enumerated.item]; + p = texts1; break; case 9: /* mono sel source */ - p = texts2[uinfo->value.enumerated.item]; + p = texts2; break; case 15: /* PCM Out Path */ - p = texts4[uinfo->value.enumerated.item]; + p = texts4; break; } } else if (reg.reg == IDX_MIXER_REC_SELECT) - p = texts3[uinfo->value.enumerated.item]; + p = texts3; - strcpy(uinfo->value.enumerated.name, p); + strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]); return 0; } @@ -745,9 +736,11 @@ static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Master Playback Switch", IDX_MIXER_PLAY_MASTER, 15, 1), AZF3328_MIXER_VOL_STEREO("Master Playback Volume", IDX_MIXER_PLAY_MASTER, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), - AZF3328_MIXER_VOL_STEREO("Wave Playback Volume", IDX_MIXER_WAVEOUT, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave 3D Bypass Playback Switch", IDX_MIXER_ADVCTL2, 7, 1), + AZF3328_MIXER_SWITCH("PCM Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), + AZF3328_MIXER_VOL_STEREO("PCM Playback Volume", + IDX_MIXER_WAVEOUT, 0x1f, 1), + AZF3328_MIXER_SWITCH("PCM 3D Bypass Playback Switch", + IDX_MIXER_ADVCTL2, 7, 1), AZF3328_MIXER_SWITCH("FM Playback Switch", IDX_MIXER_FMSYNTH, 15, 1), AZF3328_MIXER_VOL_STEREO("FM Playback Volume", IDX_MIXER_FMSYNTH, 0x1f, 1), AZF3328_MIXER_SWITCH("CD Playback Switch", IDX_MIXER_CDAUDIO, 15, 1), @@ -874,7 +867,7 @@ static void snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, unsigned reg, - unsigned int bitrate, + enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) @@ -958,15 +951,29 @@ snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); } +static void +snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip, + unsigned bitmask, + int enable +) +{ + if (enable) + chip->shadow_reg_codec_6AH &= ~bitmask; + else + chip->shadow_reg_codec_6AH |= bitmask; + snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n", + bitmask, enable, chip->shadow_reg_codec_6AH); + snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH); +} + static inline void snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) { + snd_azf3328_dbgplay("codec_enable %d\n", enable); /* no idea what exactly is being done here, but I strongly assume it's * PM related */ - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_PAUSE_PLAYBACK_BIT8, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable ); } @@ -1404,10 +1411,8 @@ static inline void snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) { - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_SOMETHING2_GAMEPORT, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_SOMETHING2_GAMEPORT, enable ); } @@ -1525,8 +1530,6 @@ { struct gameport *gp; - int io_port = chip->game_io; - chip->gameport = gp = gameport_allocate_port(); if (!gp) { printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); @@ -1536,7 +1539,7 @@ gameport_set_name(gp, "AZF3328 Gameport"); gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = io_port; + gp->io = chip->game_io; gameport_set_port_data(gp, chip); gp->open = snd_azf3328_gameport_open; @@ -1577,6 +1580,15 @@ /******************************************************************/ +static inline void +snd_azf3328_irq_log_unknown_type(u8 which) +{ + snd_azf3328_dbgplay( + "azt3328: unknown IRQ type (%x) occurred, please report!\n", + which + ); +} + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { @@ -1594,11 +1606,14 @@ )) return IRQ_NONE; /* must be interrupt for another device */ - snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count++ /* debug-only */, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), - status); + snd_azf3328_dbgplay( + "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, " + "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", + irq_count++ /* debug-only */, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), + status + ); if (status & IRQ_TIMER) { /* snd_azf3328_dbgplay("timer %ld\n", @@ -1631,9 +1646,9 @@ ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_RECORDING) { spin_lock(&chip->reg_lock); @@ -1653,9 +1668,9 @@ ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); @@ -2311,6 +2326,10 @@ for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + + /* manually store the one currently relevant write-only reg, too */ + chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH; + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) --- azt3328.h.my_org 2008-06-14 08:18:51.000000000 +0200 +++ azt3328.h 2008-06-22 14:03:30.000000000 +0200 @@ -1,7 +1,8 @@ #ifndef __SOUND_AZT3328_H #define __SOUND_AZT3328_H -/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 */ +/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 + * "WRITE_ONLY" == register does not indicate actual bit values */ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ @@ -76,7 +77,7 @@ #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 /* define frequency helpers, for maximum value safety */ -enum { +enum azf_freq_t { #define AZF_FREQ(rate) AZF_FREQ_##rate = rate AZF_FREQ(4000), AZF_FREQ(4800), @@ -150,11 +151,18 @@ #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ /* umm, nope, behaviour of these bits changes depending on what we wrote - * to 0x6b!! */ + * to 0x6b!! + * And they change upon playback/stop, too: + * Writing a value to 0x68 will display this exact value during playback, + * too but when stopped it can fall back to a rather different + * seemingly random value). Hmm, possibly this is a register which + * has a remote shadow which needs proper device supply which only exists + * in case playback is active? Or is this driver-induced? + */ /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); * actually inhibits PCM playback!!! maybe power management??: */ -#define IDX_IO_6AH 0x6A +#define IDX_IO_6AH 0x6A /* WRITE_ONLY! */ /* bit 5: enabling this will activate permanent counting of bytes 2/3 * at gameport I/O (0xb402/3) (equal values each) and cause * gameport legacy I/O at 0x0200 to be _DISABLED_!

1 0

[alsa-devel] [PATCH] PCI168 snd-azt3328: some more fixups
by Andreas Mohr 23 Jun '08

23 Jun '08

Hi, another update: - fix problem with codec register 0x6a being write-only by adding a software shadow register (caused annoying noise after module loading due to _toggling_ between gameport and audio bits instead of configuring them properly) - rename several "Wave" mixer controls to "PCM", since this is what Wine and several other apps are looking for (IOW, _requiring_) and this is what AC97 specs use as naming, too, thus I'd guess it's what these controls are - cleanup, small optimizations Patch diffed against patched 2.6.26-rc6 (patched with my last patch submission, that is, IOW it should apply to current ALSA tree easily). checkpatch.pl tested, runtime tested. Hopefully this will come in time for 1.0.17 and/or the next kernel merge before my previous patch hits the streets ;) (to have this all in one go) Thanks! Signed-off-by: Andreas Mohr <andi(a)lisas.de> --- azt3328.c.my_org 2008-06-14 08:20:36.000000000 +0200 +++ azt3328.c 2008-06-22 14:05:55.000000000 +0200 @@ -289,6 +289,12 @@ struct pci_dev *pci; int irq; + /* register 0x6a is write-only, thus need to remember setting. + * If we need to add more registers here, then we might try to fold this + * into some transparent combined shadow register handling with + * CONFIG_PM register storage below, but that's slightly difficult. */ + u16 shadow_reg_codec_6AH; + #ifdef CONFIG_PM /* register value containers for power management * Note: not always full I/O range preserved (just like Win driver!) */ @@ -324,21 +330,6 @@ return 0; } -static int -snd_azf3328_io_reg_setw(unsigned reg, u16 mask, int do_set) -{ - u16 prev = inw(reg), new; - - new = (do_set) ? (prev|mask) : (prev & ~mask); - /* we need to always write the new value no matter whether it differs or not, - * since some register bits don't indicate their setting */ - outw(new, reg); - if (new != prev) - return 1; - - return 0; -} - static inline void snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) { @@ -662,7 +653,7 @@ "pre 3D", "post 3D" }; struct azf3328_mixer_reg reg; - const char *p = NULL; + const char * const *p = NULL; snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; @@ -673,20 +664,20 @@ if (reg.reg == IDX_MIXER_ADVCTL2) { switch(reg.lchan_shift) { case 8: /* modem out sel */ - p = texts1[uinfo->value.enumerated.item]; + p = texts1; break; case 9: /* mono sel source */ - p = texts2[uinfo->value.enumerated.item]; + p = texts2; break; case 15: /* PCM Out Path */ - p = texts4[uinfo->value.enumerated.item]; + p = texts4; break; } } else if (reg.reg == IDX_MIXER_REC_SELECT) - p = texts3[uinfo->value.enumerated.item]; + p = texts3; - strcpy(uinfo->value.enumerated.name, p); + strcpy(uinfo->value.enumerated.name, p[uinfo->value.enumerated.item]); return 0; } @@ -745,9 +736,11 @@ static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Master Playback Switch", IDX_MIXER_PLAY_MASTER, 15, 1), AZF3328_MIXER_VOL_STEREO("Master Playback Volume", IDX_MIXER_PLAY_MASTER, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), - AZF3328_MIXER_VOL_STEREO("Wave Playback Volume", IDX_MIXER_WAVEOUT, 0x1f, 1), - AZF3328_MIXER_SWITCH("Wave 3D Bypass Playback Switch", IDX_MIXER_ADVCTL2, 7, 1), + AZF3328_MIXER_SWITCH("PCM Playback Switch", IDX_MIXER_WAVEOUT, 15, 1), + AZF3328_MIXER_VOL_STEREO("PCM Playback Volume", + IDX_MIXER_WAVEOUT, 0x1f, 1), + AZF3328_MIXER_SWITCH("PCM 3D Bypass Playback Switch", + IDX_MIXER_ADVCTL2, 7, 1), AZF3328_MIXER_SWITCH("FM Playback Switch", IDX_MIXER_FMSYNTH, 15, 1), AZF3328_MIXER_VOL_STEREO("FM Playback Volume", IDX_MIXER_FMSYNTH, 0x1f, 1), AZF3328_MIXER_SWITCH("CD Playback Switch", IDX_MIXER_CDAUDIO, 15, 1), @@ -874,7 +867,7 @@ static void snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, unsigned reg, - unsigned int bitrate, + enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) @@ -958,15 +951,29 @@ snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); } +static void +snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip, + unsigned bitmask, + int enable +) +{ + if (enable) + chip->shadow_reg_codec_6AH &= ~bitmask; + else + chip->shadow_reg_codec_6AH |= bitmask; + snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n", + bitmask, enable, chip->shadow_reg_codec_6AH); + snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH); +} + static inline void snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) { + snd_azf3328_dbgplay("codec_enable %d\n", enable); /* no idea what exactly is being done here, but I strongly assume it's * PM related */ - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_PAUSE_PLAYBACK_BIT8, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable ); } @@ -1404,10 +1411,8 @@ static inline void snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) { - snd_azf3328_io_reg_setw( - chip->codec_io+IDX_IO_6AH, - IO_6A_SOMETHING2_GAMEPORT, - !enable + snd_azf3328_codec_reg_6AH_update( + chip, IO_6A_SOMETHING2_GAMEPORT, enable ); } @@ -1525,8 +1530,6 @@ { struct gameport *gp; - int io_port = chip->game_io; - chip->gameport = gp = gameport_allocate_port(); if (!gp) { printk(KERN_ERR "azt3328: cannot alloc memory for gameport\n"); @@ -1536,7 +1539,7 @@ gameport_set_name(gp, "AZF3328 Gameport"); gameport_set_phys(gp, "pci%s/gameport0", pci_name(chip->pci)); gameport_set_dev_parent(gp, &chip->pci->dev); - gp->io = io_port; + gp->io = chip->game_io; gameport_set_port_data(gp, chip); gp->open = snd_azf3328_gameport_open; @@ -1577,6 +1580,15 @@ /******************************************************************/ +static inline void +snd_azf3328_irq_log_unknown_type(u8 which) +{ + printk(KERN_WARNING + "azt3328: unknown IRQ type (%x) occurred, please report!\n", + which + ); +} + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { @@ -1594,11 +1606,14 @@ )) return IRQ_NONE; /* must be interrupt for another device */ - snd_azf3328_dbgplay("Interrupt %ld!\nIDX_IO_PLAY_FLAGS %04x, IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", - irq_count++ /* debug-only */, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), - status); + snd_azf3328_dbgplay( + "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, " + "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", + irq_count++ /* debug-only */, + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), + snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), + status + ); if (status & IRQ_TIMER) { /* snd_azf3328_dbgplay("timer %ld\n", @@ -1631,9 +1646,9 @@ ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_PLAY_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_RECORDING) { spin_lock(&chip->reg_lock); @@ -1653,9 +1668,9 @@ ) ); } else - snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n"); + printk(KERN_WARNING "azt3328: irq handler problem!\n"); if (which & IRQ_REC_SOMETHING) - snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n"); + snd_azf3328_irq_log_unknown_type(which); } if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); @@ -2311,6 +2326,10 @@ for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + + /* manually store the one currently relevant write-only reg, too */ + chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH; + for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) --- azt3328.h.my_org 2008-06-14 08:18:51.000000000 +0200 +++ azt3328.h 2008-06-22 14:03:30.000000000 +0200 @@ -1,7 +1,8 @@ #ifndef __SOUND_AZT3328_H #define __SOUND_AZT3328_H -/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 */ +/* "PU" == "power-up value", as tested on PCI168 PCI rev. 10 + * "WRITE_ONLY" == register does not indicate actual bit values */ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ @@ -76,7 +77,7 @@ #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 /* define frequency helpers, for maximum value safety */ -enum { +enum azf_freq_t { #define AZF_FREQ(rate) AZF_FREQ_##rate = rate AZF_FREQ(4000), AZF_FREQ(4800), @@ -150,11 +151,18 @@ #define IO_68_RANDOM_TOGGLE1 0x0100 /* toggles randomly */ #define IO_68_RANDOM_TOGGLE2 0x0200 /* toggles randomly */ /* umm, nope, behaviour of these bits changes depending on what we wrote - * to 0x6b!! */ + * to 0x6b!! + * And they change upon playback/stop, too: + * Writing a value to 0x68 will display this exact value during playback, + * too but when stopped it can fall back to a rather different + * seemingly random value). Hmm, possibly this is a register which + * has a remote shadow which needs proper device supply which only exists + * in case playback is active? Or is this driver-induced? + */ /* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); * actually inhibits PCM playback!!! maybe power management??: */ -#define IDX_IO_6AH 0x6A +#define IDX_IO_6AH 0x6A /* WRITE_ONLY! */ /* bit 5: enabling this will activate permanent counting of bytes 2/3 * at gameport I/O (0xb402/3) (equal values each) and cause * gameport legacy I/O at 0x0200 to be _DISABLED_!

1 0

[alsa-devel] RME HDSP + Multiface
by Martin Horn 22 Jun '08

22 Jun '08

Hello everybody, I want to use a RME Multiface soundcard with PCMCIA Cardbus Card on a Dell Notebook with Ubuntu. I installed all alsa drivers and other available tools for the RME card successfully and if I do: aplay -D plughw:1 track1.wav I can hear sound and see playback on PCM channels 1+2 in the hdspmixer, which is a special softwaremixer for this card. Now the card has 8 in and 8 output channels, and though I am able to send the sound from PCM channels 1+2 to all physical outputs via the routing functionalities of the hdspmixer, I need 6 more PCM channels to play through all 8 physical channels independently. If I do: aplay -l or: cat /proc/asound/devices I get only one digital playback device which is the 2 channels already working as plughw:1. Where are my other PCM channels?? Thank you for any help!!! Martin

1 0

[alsa-devel] snd-hda-intel[ALC883] scratching and stuttering sound on MCP78S board
by Hans-Frieder Vogt 22 Jun '08

22 Jun '08

I am experiencing severely scratching and stuttering sound on my nVidia MCP78S (GeForce 8200) based Asus M3N78-EMH HDMI mainboard, which has an ALC883 on board, running with an Athlon X2 4850e dual-core CPU. It sounds, as if the sound buffers are not filled in time and are therefore repeated many times. The problem seems to be kernel / alsa-driver related, because I experience the problem also when running runlevel 1 with only a minimum processes. Strange enough, the sound is OK when I do a larger compilation in parallel (kernel make with make -j 2), but the sound is not OK when I run burnK7 (two instances to load both CPU cores) => What is different in CPU loading or scheduling between these two load cases? I tried several things so far: Debian kernel image 2.6.25-2-amd64 (2.6.25-5) self-compiled 2.6.25.x (with x in nothing to 5), with and without -rt patches self-compiled 2.6.26-rc5, -rc6, -rc7 latest alsa-driver 1.0.17rc2 please find the alsa-info output here: http://pastebin.ca/1052359 attached is my latest kernel setup (2.6.26-rc7, alsa driver modules now replaced by 1.0.17-rc2). Please tell me your suggestions what the problem may be or how I could debug the problem. Thanks. -- -- Hans-Frieder Vogt e-mail: hfvogt <at> gmx .dot. net

1 0

[alsa-devel] [PATCH] alsa: add annotations to bitwise type snd_pcm_hw_param_t
by Harvey Harrison 22 Jun '08

22 Jun '08

Fully half of all alsa sparse warnings are from snd_pcm_hw_param_t degrading to integer type, this goes a long way towards eliminating them. Signed-off-by: Harvey Harrison <harvey.harrison(a)gmail.com> --- include/sound/asound.h | 12 ++++++++---- include/sound/pcm.h | 32 ++++++++++++++++---------------- 2 files changed, 24 insertions(+), 20 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 3eaf155..0309da2 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -302,6 +302,8 @@ typedef int __bitwise snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_SUBFORMAT ((__force snd_pcm_hw_param_t) 2) /* Subformat */ #define SNDRV_PCM_HW_PARAM_FIRST_MASK SNDRV_PCM_HW_PARAM_ACCESS #define SNDRV_PCM_HW_PARAM_LAST_MASK SNDRV_PCM_HW_PARAM_SUBFORMAT +#define SNDRV_PCM_HW_PARAM_MASK_INDEX(var) \ + ((__force int)(var) - (__force int)SNDRV_PCM_HW_PARAM_FIRST_MASK) #define SNDRV_PCM_HW_PARAM_SAMPLE_BITS ((__force snd_pcm_hw_param_t) 8) /* Bits per sample */ #define SNDRV_PCM_HW_PARAM_FRAME_BITS ((__force snd_pcm_hw_param_t) 9) /* Bits per frame */ @@ -317,6 +319,8 @@ typedef int __bitwise snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAM_TICK_TIME ((__force snd_pcm_hw_param_t) 19) /* Approx tick duration in us */ #define SNDRV_PCM_HW_PARAM_FIRST_INTERVAL SNDRV_PCM_HW_PARAM_SAMPLE_BITS #define SNDRV_PCM_HW_PARAM_LAST_INTERVAL SNDRV_PCM_HW_PARAM_TICK_TIME +#define SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var) \ + ((__force int)(var) - (__force int)SNDRV_PCM_HW_PARAM_FIRST_INTERVAL) #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ @@ -336,11 +340,11 @@ struct snd_mask { struct snd_pcm_hw_params { unsigned int flags; - struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - - SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; + struct snd_mask masks[ + SNDRV_PCM_HW_PARAM_MASK_INDEX(SNDRV_PCM_HW_PARAM_LAST_MASK) + 1]; struct snd_mask mres[5]; /* reserved masks */ - struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; + struct snd_interval intervals[ + SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(SNDRV_PCM_HW_PARAM_LAST_INTERVAL) + 1]; struct snd_interval ires[9]; /* reserved intervals */ unsigned int rmask; /* W: requested masks */ unsigned int cmask; /* R: changed masks */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 51d58cc..5315b53 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -209,10 +209,10 @@ struct snd_pcm_hw_rule { }; struct snd_pcm_hw_constraints { - struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - - SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; - struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; + struct snd_mask masks[ + SNDRV_PCM_HW_PARAM_MASK_INDEX(SNDRV_PCM_HW_PARAM_LAST_MASK) + 1]; + struct snd_interval intervals[ + SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(SNDRV_PCM_HW_PARAM_LAST_INTERVAL) + 1]; unsigned int rules_num; unsigned int rules_all; struct snd_pcm_hw_rule *rules; @@ -221,13 +221,13 @@ struct snd_pcm_hw_constraints { static inline struct snd_mask *constrs_mask(struct snd_pcm_hw_constraints *constrs, snd_pcm_hw_param_t var) { - return &constrs->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; + return &constrs->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; } static inline struct snd_interval *constrs_interval(struct snd_pcm_hw_constraints *constrs, snd_pcm_hw_param_t var) { - return &constrs->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; + return &constrs->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; } struct snd_ratnum { @@ -761,40 +761,40 @@ static inline void snd_pcm_trigger_done(struct snd_pcm_substream *substream, substream->runtime->trigger_master = master; } -static inline int hw_is_mask(int var) +static inline int hw_is_mask(snd_pcm_hw_param_t var) { - return var >= SNDRV_PCM_HW_PARAM_FIRST_MASK && - var <= SNDRV_PCM_HW_PARAM_LAST_MASK; + return (__force int)var >= (__force int)SNDRV_PCM_HW_PARAM_FIRST_MASK && + (__force int)var <= (__force int)SNDRV_PCM_HW_PARAM_LAST_MASK; } -static inline int hw_is_interval(int var) +static inline int hw_is_interval(snd_pcm_hw_param_t var) { - return var >= SNDRV_PCM_HW_PARAM_FIRST_INTERVAL && - var <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; + return (__force int)var >= (__force int)SNDRV_PCM_HW_PARAM_FIRST_INTERVAL && + (__force int)var <= (__force int)SNDRV_PCM_HW_PARAM_LAST_INTERVAL; } static inline struct snd_mask *hw_param_mask(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return &params->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; + return &params->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; } static inline struct snd_interval *hw_param_interval(struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return &params->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; + return &params->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; } static inline const struct snd_mask *hw_param_mask_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return &params->masks[var - SNDRV_PCM_HW_PARAM_FIRST_MASK]; + return &params->masks[SNDRV_PCM_HW_PARAM_MASK_INDEX(var)]; } static inline const struct snd_interval *hw_param_interval_c(const struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var) { - return &params->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; + return &params->intervals[SNDRV_PCM_HW_PARAM_INTERVAL_INDEX(var)]; } #define params_access(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS)) -- 1.5.6.rc3.295.gdca2e

4 3

[alsa-devel] Nonblocking non-synchronized io plugin?
by Aryeh Leib Taurog 22 Jun '08

22 Jun '08

Hello, I am trying to develop an io plugin to route audio input/output to/from a network application I'm developing. I'm working on openSUSE 10.2, which I think means I'm using ALSA 1.0.13rc2 I am running into two problems: 1. ALSA calls my transfer function with io->nonblock = 0 and seems to be quite picky about the timing of the function. I'm having trouble getting the timing correct, and it seems to be causing ALSA to stop/crash. Is there any way that I can get ALSA to handle the timing for me and call my transfer function with io->nonblock = 1 instead? 2. ALSA seems to require that the capture and playback transfers be synchronized. That seems like a simple enough expectation, but for some reason, I am having a hard time providing this. I tried splitting the capture and playback up in .asoundrc as below, but it doesn't seem to help. (Actually I can't be sure, since it could just be a timing issue). Is there a way to separate the capture from the playback so it won't require them to be synchronized? Basically, I want the ioplug to have as much latitude as possible to just transfer the data and return as soon as it can. I guess if the calls are non-blocking, then they also won't need to be syncronized. Please correct me if I'm wrong. .asoundrc: ----------------------------- pcm.read { type myplug } pcm.write { type myplug } ctl.myapp { type myplug } pcm.myapp { type asym playback.pcm "write" capture.pcm "read" } pcm.!default myapp ------------------------------ Thanks very much Please let me know if I need to clarify anything. Aryeh Leib Taurog

1 0

[alsa-devel] 2.6.26-rc6 snd-pcsp: "Timer resolution is not sufficient"
by Rene Herman 21 Jun '08

21 Jun '08

Hi Stas. What's this? 2.6.26-rc6: === input: PC Speaker as /class/input/input5 PCSP: Timer resolution is not sufficient (4000250nS) PCSP: Make sure you have HPET and ACPI enabled. PC-Speaker initialization failed. pcspkr: probe of pcspkr failed with error -5 === A HPET I do not have, ACPI is enabled and the driver worked fine on a previous -rc. Rene.

2 4

[alsa-devel] Audio only I/O on audio-video things.
by Tom Watson 21 Jun '08

21 Jun '08

In my musings, I bought a video input device that by its nature also has audio (stereo) inputs (the video was baseband NTSC/PAL). I was wondering if one could configure one of these devices to be an 'audio only' and be used with ALSA? I understand that there might be a conflict with which set of drivers (V4L, or ALSA) latches onto the device, but I could control that thru some configuration file. I just need a "cheap" stereo audio input device for my laptop, and since I already have the video unit, and it HAS audio inputs, I thought it might work. The capture device is USB if that matters. Am I dreaming about this being able to work? Is the audio samples that are generated by such a device other than 16bit PCM? Has anyone done such a thing (I don't see it mentioned much)? Any other thoughts? Thanks. -- Tom Watson tsw(a)johana.com

1 0

Re: [alsa-devel] Avoid overload of PCM volume on Cx5045 codec
by Andreas Påhlsson 20 Jun '08

20 Jun '08

Hello. I've had a problem with low volume on my laptop and while scratching this itch I found the patch titled "hda-codec - Avoid overload of PCM volume on Cx5045 codec" I reverted this patch and of course got a lot more volume. The description states: "The PCM volume of Cx5045 codec has overload that isn't useful but rather harmful. Add a hack to override the amp info to set the max level 0 dB.". Does anybody know where to find more information on a better default value for the max level setting? regards, Andreas Påhlsson

2 5

[alsa-devel] Maximum buffer size on HDA
by Lennart Poettering 20 Jun '08

20 Jun '08

Heya, the HDA driver allows a playback buffer size of something like 64K onyl. Is this a driver limitation or a limitation of the hardware? Skimming through the docs I cannot see anything like this, and I got the idea that the audio buffer is maintained in system memory, not sound card memory, so why this limit? I'd like to buffer more than 64K, especially for surround playback in PA. Something like 2s or so. I am kind of disappointed I can't do this right now on HDA. Thanks, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 9

[alsa-devel] [PATCH 2/2] ad1848: replace duplicated constants in cs4231
by Krzysztof Helt 20 Jun '08

20 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Use AD1848 register names in the cs4231 library as well. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch contains just mindless replacement of names and should not contain any changes in the code. diff -urp linux-ref/include/sound/cs4231-regs.h linux-mm/include/sound/cs4231-regs.h --- linux-ref/include/sound/cs4231-regs.h 2008-06-15 12:16:32.452543595 +0200 +++ linux-mm/include/sound/cs4231-regs.h 2008-06-15 07:23:22.033524688 +0200 @@ -22,33 +22,14 @@ * */ -/* IO ports */ +#include "ad1848-regs.h" -#define CS4231P(x) (c_d_c_CS4231##x) +/* IO ports */ -#define c_d_c_CS4231REGSEL 0 -#define c_d_c_CS4231REG 1 -#define c_d_c_CS4231STATUS 2 -#define c_d_c_CS4231PIO 3 +#define CS4231P(x) (c_d_c_AD1848##x) /* codec registers */ -#define CS4231_LEFT_INPUT 0x00 /* left input control */ -#define CS4231_RIGHT_INPUT 0x01 /* right input control */ -#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */ -#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */ -#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */ -#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define CS4231_PIN_CTRL 0x0a /* pin control */ -#define CS4231_TEST_INIT 0x0b /* test and initialization */ -#define CS4231_MISC_INFO 0x0c /* miscellaneous information */ -#define CS4231_LOOPBACK 0x0d /* loopback control */ -#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ -#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ #define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */ #define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */ #define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */ @@ -76,16 +57,6 @@ #define CS4231_REC_UPR_CNT 0x1e /* record upper count */ #define CS4231_REC_LWR_CNT 0x1f /* record lower count */ -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define CS4231_INIT 0x80 /* CODEC is initializing */ -#define CS4231_MCE 0x40 /* mode change enable */ -#define CS4231_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */ - /* definitions for codec irq status */ #define CS4231_PLAYBACK_IRQ 0x10 @@ -97,48 +68,10 @@ #define CS4231_PLY_OVERRUN 0x02 #define CS4231_PLY_UNDERRUN 0x01 -/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */ - -#define CS4231_ENABLE_MIC_GAIN 0x20 - -#define CS4231_MIXS_LINE 0x00 -#define CS4231_MIXS_AUX1 0x40 -#define CS4231_MIXS_MIC 0x80 -#define CS4231_MIXS_ALL 0xc0 - /* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */ -#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */ -#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */ -#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ #define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */ #define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */ -#define CS4231_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define CS4231_XTAL1 0x00 /* 24.576 crystal */ -#define CS4231_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - CS4231_IFACE_CTRL */ - -#define CS4231_RECORD_PIO 0x80 /* record PIO enable */ -#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */ -#define CS4231_AUTOCALIB 0x08 /* auto calibrate */ -#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */ -#define CS4231_RECORD_ENABLE 0x02 /* record enable */ -#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - CS4231_PIN_CTRL */ - -#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */ -#define CS4231_XCTL1 0x40 /* external control #1 */ -#define CS4231_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - CS4231_TEST_INIT */ - -#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */ /* definitions for misc control register - CS4231_MISC_INFO */ diff -urp linux-ref/sound/isa/cs423x/cs4231_lib.c linux-mm/sound/isa/cs423x/cs4231_lib.c --- linux-ref/sound/isa/cs423x/cs4231_lib.c 2008-06-15 12:16:32.480545191 +0200 +++ linux-mm/sound/isa/cs423x/cs4231_lib.c 2008-06-15 12:13:35.000000000 +0200 @@ -51,20 +51,20 @@ MODULE_LICENSE("GPL"); */ static unsigned char freq_bits[14] = { - /* 5510 */ 0x00 | CS4231_XTAL2, - /* 6620 */ 0x0E | CS4231_XTAL2, - /* 8000 */ 0x00 | CS4231_XTAL1, - /* 9600 */ 0x0E | CS4231_XTAL1, - /* 11025 */ 0x02 | CS4231_XTAL2, - /* 16000 */ 0x02 | CS4231_XTAL1, - /* 18900 */ 0x04 | CS4231_XTAL2, - /* 22050 */ 0x06 | CS4231_XTAL2, - /* 27042 */ 0x04 | CS4231_XTAL1, - /* 32000 */ 0x06 | CS4231_XTAL1, - /* 33075 */ 0x0C | CS4231_XTAL2, - /* 37800 */ 0x08 | CS4231_XTAL2, - /* 44100 */ 0x0A | CS4231_XTAL2, - /* 48000 */ 0x0C | CS4231_XTAL1 + /* 5510 */ 0x00 | AD1848_XTAL2, + /* 6620 */ 0x0E | AD1848_XTAL2, + /* 8000 */ 0x00 | AD1848_XTAL1, + /* 9600 */ 0x0E | AD1848_XTAL1, + /* 11025 */ 0x02 | AD1848_XTAL2, + /* 16000 */ 0x02 | AD1848_XTAL1, + /* 18900 */ 0x04 | AD1848_XTAL2, + /* 22050 */ 0x06 | AD1848_XTAL2, + /* 27042 */ 0x04 | AD1848_XTAL1, + /* 32000 */ 0x06 | AD1848_XTAL1, + /* 33075 */ 0x0C | AD1848_XTAL2, + /* 37800 */ 0x08 | AD1848_XTAL2, + /* 44100 */ 0x0A | AD1848_XTAL2, + /* 48000 */ 0x0C | AD1848_XTAL1 }; static unsigned int rates[14] = { @@ -94,7 +94,7 @@ static unsigned char snd_cs4231_original 0xbf, /* 06/06 - loc */ 0xbf, /* 07/07 - roc */ 0x20, /* 08/08 - pdfr */ - CS4231_AUTOCALIB, /* 09/09 - ic */ + AD1848_AUTOCALIB, /* 09/09 - ic */ 0x00, /* 0a/10 - pc */ 0x00, /* 0b/11 - ti */ CS4231_MODE2, /* 0c/12 - mi */ @@ -174,7 +174,7 @@ static void snd_cs4231_wait(struct snd_c int timeout; for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT); timeout--) udelay(100); } @@ -186,7 +186,7 @@ static void snd_cs4231_outm(struct snd_c snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT) snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif chip->image[reg] = tmp; @@ -203,7 +203,7 @@ static void snd_cs4231_dout(struct snd_c int timeout; for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT); timeout--) udelay(10); cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); @@ -215,7 +215,7 @@ void snd_cs4231_out(struct snd_cs4231 *c { snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT) snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); @@ -230,7 +230,7 @@ unsigned char snd_cs4231_in(struct snd_c { snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT) snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); #endif cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); @@ -320,7 +320,7 @@ static void snd_cs4231_busy_wait(struct cs4231_inb(chip, CS4231P(REGSEL)); /* end of cleanup sequence */ for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); + timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT); timeout--) udelay(10); } @@ -332,15 +332,15 @@ void snd_cs4231_mce_up(struct snd_cs4231 snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT) snd_printk("mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit |= CS4231_MCE; + chip->mce_bit |= AD1848_MCE; timeout = cs4231_inb(chip, CS4231P(REGSEL)); if (timeout == 0x80) snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); - if (!(timeout & CS4231_MCE)) + if (!(timeout & AD1848_MCE)) cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); } @@ -354,17 +354,17 @@ void snd_cs4231_mce_down(struct snd_cs42 snd_cs4231_busy_wait(chip); #ifdef CONFIG_SND_DEBUG - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT) snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); #endif spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit &= ~CS4231_MCE; + chip->mce_bit &= ~AD1848_MCE; timeout = cs4231_inb(chip, CS4231P(REGSEL)); cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0 || + if ((timeout & AD1848_MCE) == 0 || !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { return; } @@ -380,8 +380,8 @@ void snd_cs4231_mce_down(struct snd_cs42 /* check condition up to 250 ms */ end_time = jiffies + msecs_to_jiffies(250); - while (snd_cs4231_in(chip, CS4231_TEST_INIT) & - CS4231_CALIB_IN_PROGRESS) { + while (snd_cs4231_in(chip, AD1848_TEST_INIT) & + AD1848_CALIB_IN_PROGRESS) { if (time_after(jiffies, end_time)) { snd_printk(KERN_ERR "mce_down - " @@ -395,7 +395,7 @@ void snd_cs4231_mce_down(struct snd_cs42 /* check condition up to 100 ms */ end_time = jiffies + msecs_to_jiffies(100); - while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { + while (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT) { if (time_after(jiffies, end_time)) { snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); return; @@ -410,14 +410,14 @@ void snd_cs4231_mce_down(struct snd_cs42 static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size) { switch (format & 0xe0) { - case CS4231_LINEAR_16: + case AD1848_LINEAR_16: case CS4231_LINEAR_16_BIG: size >>= 1; break; case CS4231_ADPCM_16: return size >> 2; } - if (format & CS4231_STEREO) + if (format & AD1848_STEREO) size >>= 1; return size; } @@ -449,24 +449,24 @@ static int snd_cs4231_trigger(struct snd what = 0; snd_pcm_group_for_each_entry(s, substream) { if (s == chip->playback_substream) { - what |= CS4231_PLAYBACK_ENABLE; + what |= AD1848_PLAYBACK_ENABLE; snd_pcm_trigger_done(s, substream); } else if (s == chip->capture_substream) { - what |= CS4231_RECORD_ENABLE; + what |= AD1848_CAPTURE_ENABLE; snd_pcm_trigger_done(s, substream); } } spin_lock(&chip->reg_lock); if (do_start) { - chip->image[CS4231_IFACE_CTRL] |= what; + chip->image[AD1848_IFACE_CTRL] |= what; if (chip->trigger) chip->trigger(chip, what, 1); } else { - chip->image[CS4231_IFACE_CTRL] &= ~what; + chip->image[AD1848_IFACE_CTRL] &= ~what; if (chip->trigger) chip->trigger(chip, what, 0); } - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + snd_cs4231_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); spin_unlock(&chip->reg_lock); #if 0 snd_cs4231_debug(chip); @@ -495,16 +495,16 @@ static unsigned char snd_cs4231_get_form { unsigned char rformat; - rformat = CS4231_LINEAR_8; + rformat = AD1848_LINEAR_8; switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: rformat = CS4231_ULAW_8; break; - case SNDRV_PCM_FORMAT_A_LAW: rformat = CS4231_ALAW_8; break; - case SNDRV_PCM_FORMAT_S16_LE: rformat = CS4231_LINEAR_16; break; + case SNDRV_PCM_FORMAT_MU_LAW: rformat = AD1848_ULAW_8; break; + case SNDRV_PCM_FORMAT_A_LAW: rformat = AD1848_ALAW_8; break; + case SNDRV_PCM_FORMAT_S16_LE: rformat = AD1848_LINEAR_16; break; case SNDRV_PCM_FORMAT_S16_BE: rformat = CS4231_LINEAR_16_BIG; break; case SNDRV_PCM_FORMAT_IMA_ADPCM: rformat = CS4231_ADPCM_16; break; } if (channels > 1) - rformat |= CS4231_STEREO; + rformat |= AD1848_STEREO; #if 0 snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); #endif @@ -522,16 +522,16 @@ static void snd_cs4231_calibrate_mute(st return; } if (!mute) { - snd_cs4231_dout(chip, CS4231_LEFT_INPUT, chip->image[CS4231_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_INPUT, chip->image[CS4231_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); - } - snd_cs4231_dout(chip, CS4231_AUX1_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX1_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_LEFT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_RIGHT_INPUT, mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LEFT_OUTPUT, mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_OUTPUT, mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + snd_cs4231_dout(chip, AD1848_LEFT_INPUT, chip->image[AD1848_LEFT_INPUT]); + snd_cs4231_dout(chip, AD1848_RIGHT_INPUT, chip->image[AD1848_RIGHT_INPUT]); + snd_cs4231_dout(chip, AD1848_LOOPBACK, chip->image[AD1848_LOOPBACK]); + } + snd_cs4231_dout(chip, AD1848_AUX1_LEFT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX1_LEFT_INPUT]); + snd_cs4231_dout(chip, AD1848_AUX1_RIGHT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX1_RIGHT_INPUT]); + snd_cs4231_dout(chip, AD1848_AUX2_LEFT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX2_LEFT_INPUT]); + snd_cs4231_dout(chip, AD1848_AUX2_RIGHT_INPUT, mute ? 0x80 : chip->image[AD1848_AUX2_RIGHT_INPUT]); + snd_cs4231_dout(chip, AD1848_LEFT_OUTPUT, mute ? 0x80 : chip->image[AD1848_LEFT_OUTPUT]); + snd_cs4231_dout(chip, AD1848_RIGHT_OUTPUT, mute ? 0x80 : chip->image[AD1848_RIGHT_OUTPUT]); snd_cs4231_dout(chip, CS4231_LEFT_LINE_IN, mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); snd_cs4231_dout(chip, CS4231_RIGHT_LINE_IN, mute ? 0x80 : chip->image[CS4231_RIGHT_LINE_IN]); snd_cs4231_dout(chip, CS4231_MONO_CTRL, mute ? 0xc0 : chip->image[CS4231_MONO_CTRL]); @@ -557,9 +557,9 @@ static void snd_cs4231_playback_format(s if (chip->hardware == CS4231_HW_CS4231A || (chip->hardware & CS4231_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); - if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (pdfr & 0x0f)) { /* rate is same? */ + if ((chip->image[AD1848_DATA_FORMAT] & 0x0f) == (pdfr & 0x0f)) { /* rate is same? */ snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x10); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); + snd_cs4231_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT] = pdfr); snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x10); udelay(100); /* Fixes audible clicks at least on GUS MAX */ full_calib = 0; @@ -570,12 +570,12 @@ static void snd_cs4231_playback_format(s snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); if (chip->hardware != CS4231_HW_INTERWAVE && !chip->single_dma) { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, - (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ? + snd_cs4231_out(chip, AD1848_DATA_FORMAT, + (chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE) ? (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) : pdfr); } else { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT] = pdfr); + snd_cs4231_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT] = pdfr); } spin_unlock_irqrestore(&chip->reg_lock, flags); if (chip->hardware == CS4231_HW_OPL3SA2) @@ -598,8 +598,8 @@ static void snd_cs4231_capture_format(st if (chip->hardware == CS4231_HW_CS4231A || (chip->hardware & CS4231_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); - if ((chip->image[CS4231_PLAYBK_FORMAT] & 0x0f) == (cdfr & 0x0f) || /* rate is same? */ - (chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { + if ((chip->image[AD1848_DATA_FORMAT] & 0x0f) == (cdfr & 0x0f) || /* rate is same? */ + (chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) { snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x20); snd_cs4231_out(chip, CS4231_REC_FORMAT, chip->image[CS4231_REC_FORMAT] = cdfr); snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] &= ~0x20); @@ -611,9 +611,9 @@ static void snd_cs4231_capture_format(st snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); if (chip->hardware != CS4231_HW_INTERWAVE) { - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, - ((chip->single_dma ? cdfr : chip->image[CS4231_PLAYBK_FORMAT]) & 0xf0) | + if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) { + snd_cs4231_out(chip, AD1848_DATA_FORMAT, + ((chip->single_dma ? cdfr : chip->image[AD1848_DATA_FORMAT]) & 0xf0) | (cdfr & 0x0f)); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_down(chip); @@ -639,7 +639,7 @@ static unsigned long snd_cs4231_timer_re if (chip->hardware & CS4231_HW_CS4236B_MASK) return 14467; else - return chip->image[CS4231_PLAYBK_FORMAT] & 1 ? 9969 : 9920; + return chip->image[AD1848_DATA_FORMAT] & 1 ? 9969 : 9920; } static int snd_cs4231_timer_start(struct snd_timer * timer) @@ -681,11 +681,11 @@ static void snd_cs4231_init(struct snd_c #endif snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO | - CS4231_CALIB_MODE); - chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO | + AD1848_CALIB_MODE); + chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; + snd_cs4231_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_down(chip); @@ -709,7 +709,7 @@ static void snd_cs4231_init(struct snd_c snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT]); + snd_cs4231_out(chip, AD1848_DATA_FORMAT, chip->image[AD1848_DATA_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_down(chip); @@ -751,8 +751,8 @@ static int snd_cs4231_open(struct snd_cs snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - chip->image[CS4231_PIN_CTRL] |= CS4231_IRQ_ENABLE; - snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); + chip->image[AD1848_PIN_CTRL] |= AD1848_IRQ_ENABLE; + snd_cs4231_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | CS4231_RECORD_IRQ | CS4231_TIMER_IRQ); @@ -781,19 +781,19 @@ static void snd_cs4231_close(struct snd_ snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ cs4231_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ - chip->image[CS4231_PIN_CTRL] &= ~CS4231_IRQ_ENABLE; - snd_cs4231_out(chip, CS4231_PIN_CTRL, chip->image[CS4231_PIN_CTRL]); + chip->image[AD1848_PIN_CTRL] &= ~AD1848_IRQ_ENABLE; + snd_cs4231_out(chip, AD1848_PIN_CTRL, chip->image[AD1848_PIN_CTRL]); /* now disable record & playback */ - if (chip->image[CS4231_IFACE_CTRL] & (CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO)) { + if (chip->image[AD1848_IFACE_CTRL] & (AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO)) { spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); + snd_cs4231_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -875,11 +875,11 @@ static int snd_cs4231_playback_prepare(s spin_lock_irqsave(&chip->reg_lock, flags); chip->p_dma_size = size; - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO); snd_dma_program(chip->dma1, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); - count = snd_cs4231_get_count(chip->image[CS4231_PLAYBK_FORMAT], count) - 1; - snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); + count = snd_cs4231_get_count(chip->image[AD1848_DATA_FORMAT], count) - 1; + snd_cs4231_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); + snd_cs4231_out(chip, AD1848_DATA_UPR_CNT, (unsigned char) (count >> 8)); spin_unlock_irqrestore(&chip->reg_lock, flags); #if 0 snd_cs4231_debug(chip); @@ -917,12 +917,12 @@ static int snd_cs4231_capture_prepare(st spin_lock_irqsave(&chip->reg_lock, flags); chip->c_dma_size = size; - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); snd_dma_program(chip->dma2, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); count = snd_cs4231_get_count(chip->image[CS4231_REC_FORMAT], count) - 1; if (chip->single_dma && chip->hardware != CS4231_HW_INTERWAVE) { - snd_cs4231_out(chip, CS4231_PLY_LWR_CNT, (unsigned char) count); - snd_cs4231_out(chip, CS4231_PLY_UPR_CNT, (unsigned char) (count >> 8)); + snd_cs4231_out(chip, AD1848_DATA_LWR_CNT, (unsigned char) count); + snd_cs4231_out(chip, AD1848_DATA_UPR_CNT, (unsigned char) (count >> 8)); } else { snd_cs4231_out(chip, CS4231_REC_LWR_CNT, (unsigned char) count); snd_cs4231_out(chip, CS4231_REC_UPR_CNT, (unsigned char) (count >> 8)); @@ -937,7 +937,7 @@ void snd_cs4231_overrange(struct snd_cs4 unsigned char res; spin_lock_irqsave(&chip->reg_lock, flags); - res = snd_cs4231_in(chip, CS4231_TEST_INIT); + res = snd_cs4231_in(chip, AD1848_TEST_INIT); spin_unlock_irqrestore(&chip->reg_lock, flags); if (res & (0x08 | 0x02)) /* detect overrange only above 0dB; may be user selectable? */ chip->capture_substream->runtime->overrange++; @@ -990,7 +990,7 @@ static snd_pcm_uframes_t snd_cs4231_play struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); size_t ptr; - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) + if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) return 0; ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); @@ -1001,7 +1001,7 @@ static snd_pcm_uframes_t snd_cs4231_capt struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); size_t ptr; - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE)) + if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE)) return 0; ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); @@ -1024,12 +1024,12 @@ static int snd_cs4231_probe(struct snd_c id = 0; for (i = 0; i < 50; i++) { mb(); - if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) + if (cs4231_inb(chip, CS4231P(REGSEL)) & AD1848_INIT) udelay(2000); else { spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - id = snd_cs4231_in(chip, CS4231_MISC_INFO) & 0x0f; + snd_cs4231_out(chip, AD1848_MISC_INFO, CS4231_MODE2); + id = snd_cs4231_in(chip, AD1848_MISC_INFO) & 0x0f; spin_unlock_irqrestore(&chip->reg_lock, flags); if (id == 0x0a) break; /* this is valid value */ @@ -1070,10 +1070,10 @@ static int snd_cs4231_probe(struct snd_c mb(); spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->image[CS4231_MISC_INFO] = CS4231_MODE2; + chip->image[AD1848_MISC_INFO] = CS4231_MODE2; switch (chip->hardware) { case CS4231_HW_INTERWAVE: - chip->image[CS4231_MISC_INFO] = CS4231_IW_MODE3; + chip->image[AD1848_MISC_INFO] = CS4231_IW_MODE3; break; case CS4231_HW_CS4235: case CS4231_HW_CS4236B: @@ -1081,15 +1081,15 @@ static int snd_cs4231_probe(struct snd_c case CS4231_HW_CS4238B: case CS4231_HW_CS4239: if (hw == CS4231_HW_DETECT3) - chip->image[CS4231_MISC_INFO] = CS4231_4236_MODE3; + chip->image[AD1848_MISC_INFO] = CS4231_4236_MODE3; else chip->hardware = CS4231_HW_CS4236; break; } - chip->image[CS4231_IFACE_CTRL] = - (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | - (chip->single_dma ? CS4231_SINGLE_DMA : 0); + chip->image[AD1848_IFACE_CTRL] = + (chip->image[AD1848_IFACE_CTRL] & ~AD1848_SINGLE_DMA) | + (chip->single_dma ? AD1848_SINGLE_DMA : 0); if (chip->hardware != CS4231_HW_OPTI93X) { chip->image[CS4231_ALT_FEATURE_1] = 0x80; chip->image[CS4231_ALT_FEATURE_2] = @@ -1355,13 +1355,13 @@ static void snd_cs4231_resume(struct snd */ snd_cs4231_busy_wait(chip); spin_lock_irqsave(&chip->reg_lock, flags); - chip->mce_bit &= ~CS4231_MCE; + chip->mce_bit &= ~AD1848_MCE; timeout = cs4231_inb(chip, CS4231P(REGSEL)); cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0 || + if ((timeout & AD1848_MCE) == 0 || !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { return; } @@ -1665,8 +1665,8 @@ static int snd_cs4231_get_mux(struct snd unsigned long flags; spin_lock_irqsave(&chip->reg_lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[CS4231_LEFT_INPUT] & CS4231_MIXS_ALL) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[CS4231_RIGHT_INPUT] & CS4231_MIXS_ALL) >> 6; + ucontrol->value.enumerated.item[0] = (chip->image[AD1848_LEFT_INPUT] & AD1848_MIXS_ALL) >> 6; + ucontrol->value.enumerated.item[1] = (chip->image[AD1848_RIGHT_INPUT] & AD1848_MIXS_ALL) >> 6; spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } @@ -1684,12 +1684,12 @@ static int snd_cs4231_put_mux(struct snd left = ucontrol->value.enumerated.item[0] << 6; right = ucontrol->value.enumerated.item[1] << 6; spin_lock_irqsave(&chip->reg_lock, flags); - left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left; - right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right; - change = left != chip->image[CS4231_LEFT_INPUT] || - right != chip->image[CS4231_RIGHT_INPUT]; - snd_cs4231_out(chip, CS4231_LEFT_INPUT, left); - snd_cs4231_out(chip, CS4231_RIGHT_INPUT, right); + left = (chip->image[AD1848_LEFT_INPUT] & ~AD1848_MIXS_ALL) | left; + right = (chip->image[AD1848_RIGHT_INPUT] & ~AD1848_MIXS_ALL) | right; + change = left != chip->image[AD1848_LEFT_INPUT] || + right != chip->image[AD1848_RIGHT_INPUT]; + snd_cs4231_out(chip, AD1848_LEFT_INPUT, left); + snd_cs4231_out(chip, AD1848_RIGHT_INPUT, right); spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } @@ -1810,19 +1810,19 @@ int snd_cs4231_put_double(struct snd_kco } static struct snd_kcontrol_new snd_cs4231_controls[] = { -CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1), CS4231_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), CS4231_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), CS4231_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +CS4231_DOUBLE("Capture Volume", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -1830,24 +1830,24 @@ CS4231_DOUBLE("Capture Volume", 0, CS423 .get = snd_cs4231_get_mux, .put = snd_cs4231_put_mux, }, -CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) +CS4231_DOUBLE("Mic Boost", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_SINGLE("Loopback Capture Switch", 0, AD1848_LOOPBACK, 0, 1, 0), +CS4231_SINGLE("Loopback Capture Volume", 0, AD1848_LOOPBACK, 2, 63, 1) }; - + static struct snd_kcontrol_new snd_opti93x_controls[] = { CS4231_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), CS4231_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), CS4231_DOUBLE("PCM Playback Switch", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), + AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), CS4231_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), + AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 31, 1), CS4231_DOUBLE("FM Playback Switch", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), + AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), CS4231_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), + AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 1, 1, 15, 1), CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), CS4231_DOUBLE("Line Playback Volume", 0, @@ -1857,17 +1857,17 @@ CS4231_DOUBLE("Mic Playback Switch", 0, CS4231_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), CS4231_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), + AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 5, 5, 1, 0), CS4231_DOUBLE("CD Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), + AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), CS4231_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), + AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 1, 1, 15, 1), CS4231_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), CS4231_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), CS4231_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), + AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", diff -urp linux-ref/sound/isa/cs423x/cs4236_lib.c linux-mm/sound/isa/cs423x/cs4236_lib.c --- linux-ref/sound/isa/cs423x/cs4236_lib.c 2008-06-15 12:16:32.492545497 +0200 +++ linux-mm/sound/isa/cs423x/cs4236_lib.c 2008-06-15 07:23:22.069571493 +0200 @@ -188,7 +188,7 @@ static void snd_cs4236_playback_format(s spin_lock_irqsave(&chip->reg_lock, flags); /* set fast playback format change and clean playback FIFO */ snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] | 0x10); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, pdfr & 0xf0); + snd_cs4231_out(chip, AD1848_DATA_FORMAT, pdfr & 0xf0); snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1] & ~0x10); snd_cs4236_ext_out(chip, CS4236_DAC_RATE, rate); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -330,12 +330,12 @@ int snd_cs4236_create(struct snd_card *c snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), snd_cs4236_ext_map[reg]); /* initialize compatible but more featured registers */ - snd_cs4231_out(chip, CS4231_LEFT_INPUT, 0x40); - snd_cs4231_out(chip, CS4231_RIGHT_INPUT, 0x40); - snd_cs4231_out(chip, CS4231_AUX1_LEFT_INPUT, 0xff); - snd_cs4231_out(chip, CS4231_AUX1_RIGHT_INPUT, 0xff); - snd_cs4231_out(chip, CS4231_AUX2_LEFT_INPUT, 0xdf); - snd_cs4231_out(chip, CS4231_AUX2_RIGHT_INPUT, 0xdf); + snd_cs4231_out(chip, AD1848_LEFT_INPUT, 0x40); + snd_cs4231_out(chip, AD1848_RIGHT_INPUT, 0x40); + snd_cs4231_out(chip, AD1848_AUX1_LEFT_INPUT, 0xff); + snd_cs4231_out(chip, AD1848_AUX1_RIGHT_INPUT, 0xff); + snd_cs4231_out(chip, AD1848_AUX2_LEFT_INPUT, 0xdf); + snd_cs4231_out(chip, AD1848_AUX2_RIGHT_INPUT, 0xdf); snd_cs4231_out(chip, CS4231_RIGHT_LINE_IN, 0xff); snd_cs4231_out(chip, CS4231_LEFT_LINE_IN, 0xff); snd_cs4231_out(chip, CS4231_RIGHT_LINE_IN, 0xff); @@ -715,8 +715,8 @@ CS4236_MASTER_DIGITAL("Master Digital Vo CS4236_DOUBLE("Capture Boost Volume", 0, CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), -CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), CS4236_DOUBLE("DSP Playback Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), CS4236_DOUBLE("DSP Playback Volume", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1), @@ -737,25 +737,25 @@ CS4236_DOUBLE("Mic Capture Switch", 0, C CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), CS4236_DOUBLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), -CS4231_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Line Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -CS4231_DOUBLE("Line Capture Bypass", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 5, 5, 1, 1), - -CS4231_DOUBLE("CD Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("CD Volume", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("CD Capture Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), +CS4231_DOUBLE("Line Playback Switch", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Line Volume", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Line Capture Switch", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 6, 6, 1, 1), +CS4231_DOUBLE("Line Capture Bypass", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 5, 5, 1, 1), + +CS4231_DOUBLE("CD Playback Switch", 0, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Volume", 0, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("CD Capture Switch", 0, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 6, 6, 1, 1), CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), CS4236_DOUBLE1("Mono Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), CS4231_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -CS4231_DOUBLE("Analog Loopback Capture Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), +CS4231_DOUBLE("Capture Volume", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0), +CS4231_DOUBLE("Analog Loopback Capture Switch", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 7, 7, 1, 0), -CS4231_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) +CS4231_SINGLE("Digital Loopback Playback Switch", 0, AD1848_LOOPBACK, 0, 1, 0), +CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, AD1848_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) }; static struct snd_kcontrol_new snd_cs4235_controls[] = { @@ -775,8 +775,8 @@ CS4231_DOUBLE("Master Digital Volume", 1 CS4236_DOUBLE("Capture Volume", 0, CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), -CS4231_DOUBLE("PCM Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +CS4231_DOUBLE("PCM Switch", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Volume", 0, AD1848_LEFT_OUTPUT, AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), CS4236_DOUBLE("DSP Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), @@ -789,20 +789,20 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0), -CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -CS4231_DOUBLE("Aux Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), - -CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -CS4231_DOUBLE("Aux Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Capture Switch", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 6, 6, 1, 1), +CS4231_DOUBLE("Aux Volume", 0, AD1848_AUX1_LEFT_INPUT, AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1), + +CS4231_DOUBLE("Aux Playback Switch", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Capture Switch", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 6, 6, 1, 1), +CS4231_DOUBLE("Aux Volume", 1, AD1848_AUX2_LEFT_INPUT, AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1), CS4236_DOUBLE1("Master Mono Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), CS4236_DOUBLE1("Mono Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), CS4231_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -CS4231_DOUBLE("Analog Loopback Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), +CS4231_DOUBLE("Analog Loopback Switch", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 7, 7, 1, 0), }; #define CS4236_IEC958_ENABLE(xname, xindex) \ diff -urp linux-ref/sound/isa/sscape.c linux-mm/sound/isa/sscape.c --- linux-ref/sound/isa/sscape.c 2008-06-15 12:16:32.504618272 +0200 +++ linux-mm/sound/isa/sscape.c 2008-06-15 07:23:22.077521453 +0200 @@ -983,7 +983,7 @@ static void ad1845_playback_format(struc * NOTE: We seem to need to write to the MSB before the LSB * to get the correct sample frequency. */ - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0)); + snd_cs4231_out(chip, AD1848_DATA_FORMAT, (format & 0xf0)); snd_cs4231_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); snd_cs4231_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); @@ -1089,7 +1089,7 @@ static int __devinit create_ad1845(struc * b) enable frequency selection (for capture/playback) */ spin_lock_irqsave(&chip->reg_lock, flags); - snd_cs4231_out(chip, CS4231_MISC_INFO, + snd_cs4231_out(chip, AD1848_MISC_INFO, CS4231_MODE2 | 0x10); val = snd_cs4231_in(chip, AD1845_PWR_DOWN_CTRL); snd_cs4231_out(chip, AD1845_PWR_DOWN_CTRL, diff -urp linux-ref/sound/sparc/cs4231.c linux-mm/sound/sparc/cs4231.c --- linux-ref/sound/sparc/cs4231.c 2008-06-14 00:05:03.000000000 +0200 +++ linux-mm/sound/sparc/cs4231.c 2008-06-15 12:18:45.786928299 +0200 @@ -137,7 +137,7 @@ static struct snd_cs4231 *cs4231_list; #include <sound/cs4231-regs.h> /* XXX offsets are different than PC ISA chips... */ -#define CS4231U(chip, x) ((chip)->port + ((c_d_c_CS4231##x) << 2)) +#define CS4231U(chip, x) ((chip)->port + ((c_d_c_AD1848##x) << 2)) /* SBUS DMA register defines. */ @@ -196,20 +196,20 @@ static struct snd_cs4231 *cs4231_list; */ static unsigned char freq_bits[14] = { - /* 5510 */ 0x00 | CS4231_XTAL2, - /* 6620 */ 0x0E | CS4231_XTAL2, - /* 8000 */ 0x00 | CS4231_XTAL1, - /* 9600 */ 0x0E | CS4231_XTAL1, - /* 11025 */ 0x02 | CS4231_XTAL2, - /* 16000 */ 0x02 | CS4231_XTAL1, - /* 18900 */ 0x04 | CS4231_XTAL2, - /* 22050 */ 0x06 | CS4231_XTAL2, - /* 27042 */ 0x04 | CS4231_XTAL1, - /* 32000 */ 0x06 | CS4231_XTAL1, - /* 33075 */ 0x0C | CS4231_XTAL2, - /* 37800 */ 0x08 | CS4231_XTAL2, - /* 44100 */ 0x0A | CS4231_XTAL2, - /* 48000 */ 0x0C | CS4231_XTAL1 + /* 5510 */ 0x00 | AD1848_XTAL2, + /* 6620 */ 0x0E | AD1848_XTAL2, + /* 8000 */ 0x00 | AD1848_XTAL1, + /* 9600 */ 0x0E | AD1848_XTAL1, + /* 11025 */ 0x02 | AD1848_XTAL2, + /* 16000 */ 0x02 | AD1848_XTAL1, + /* 18900 */ 0x04 | AD1848_XTAL2, + /* 22050 */ 0x06 | AD1848_XTAL2, + /* 27042 */ 0x04 | AD1848_XTAL1, + /* 32000 */ 0x06 | AD1848_XTAL1, + /* 33075 */ 0x0C | AD1848_XTAL2, + /* 37800 */ 0x08 | AD1848_XTAL2, + /* 44100 */ 0x0A | AD1848_XTAL2, + /* 48000 */ 0x0C | AD1848_XTAL1 }; static unsigned int rates[14] = { @@ -240,7 +240,7 @@ static unsigned char snd_cs4231_original 0xbf, /* 06/06 - loc */ 0xbf, /* 07/07 - roc */ 0x20, /* 08/08 - pdfr */ - CS4231_AUTOCALIB, /* 09/09 - ic */ + AD1848_AUTOCALIB, /* 09/09 - ic */ 0x00, /* 0a/10 - pc */ 0x00, /* 0b/11 - ti */ CS4231_MODE2, /* 0c/12 - mi */ @@ -300,7 +300,7 @@ static void snd_cs4231_ready(struct snd_ for (timeout = 250; timeout > 0; timeout--) { int val = __cs4231_readb(chip, CS4231U(chip, REGSEL)); - if ((val & CS4231_INIT) == 0) + if ((val & AD1848_INIT) == 0) break; udelay(100); } @@ -311,7 +311,7 @@ static void snd_cs4231_dout(struct snd_c { snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & AD1848_INIT) snd_printdd("out: auto calibration time out - reg = 0x%x, " "value = 0x%x\n", reg, value); @@ -344,7 +344,7 @@ static unsigned char snd_cs4231_in(struc { snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & AD1848_INIT) snd_printdd("in: auto calibration time out - reg = 0x%x\n", reg); #endif @@ -368,7 +368,7 @@ static void snd_cs4231_busy_wait(struct /* end of cleanup sequence */ for (timeout = 500; timeout > 0; timeout--) { int val = __cs4231_readb(chip, CS4231U(chip, REGSEL)); - if ((val & CS4231_INIT) == 0) + if ((val & AD1848_INIT) == 0) break; msleep(1); } @@ -382,16 +382,16 @@ static void snd_cs4231_mce_up(struct snd spin_lock_irqsave(&chip->lock, flags); snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & AD1848_INIT) snd_printdd("mce_up - auto calibration time out (0)\n"); #endif - chip->mce_bit |= CS4231_MCE; + chip->mce_bit |= AD1848_MCE; timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL)); if (timeout == 0x80) snd_printdd("mce_up [%p]: serious init problem - " "codec still busy\n", chip->port); - if (!(timeout & CS4231_MCE)) + if (!(timeout & AD1848_MCE)) __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), CS4231U(chip, REGSEL)); spin_unlock_irqrestore(&chip->lock, flags); @@ -405,18 +405,18 @@ static void snd_cs4231_mce_down(struct s snd_cs4231_busy_wait(chip); spin_lock_irqsave(&chip->lock, flags); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & AD1848_INIT) snd_printdd("mce_down [%p] - auto calibration time out (0)\n", CS4231U(chip, REGSEL)); #endif - chip->mce_bit &= ~CS4231_MCE; + chip->mce_bit &= ~AD1848_MCE; reg = __cs4231_readb(chip, CS4231U(chip, REGSEL)); __cs4231_writeb(chip, chip->mce_bit | (reg & 0x1f), CS4231U(chip, REGSEL)); if (reg == 0x80) snd_printdd("mce_down [%p]: serious init problem " "- codec still busy\n", chip->port); - if ((reg & CS4231_MCE) == 0) { + if ((reg & AD1848_MCE) == 0) { spin_unlock_irqrestore(&chip->lock, flags); return; } @@ -429,8 +429,8 @@ static void snd_cs4231_mce_down(struct s spin_unlock_irqrestore(&chip->lock, flags); msleep(1); spin_lock_irqsave(&chip->lock, flags); - reg = snd_cs4231_in(chip, CS4231_TEST_INIT); - reg &= CS4231_CALIB_IN_PROGRESS; + reg = snd_cs4231_in(chip, AD1848_TEST_INIT); + reg &= AD1848_CALIB_IN_PROGRESS; } while (reg && time_before(jiffies, timeout)); spin_unlock_irqrestore(&chip->lock, flags); @@ -464,7 +464,7 @@ static void cs4231_dma_trigger(struct sn struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); struct cs4231_dma_control *dma_cont; - if (what & CS4231_PLAYBACK_ENABLE) { + if (what & AD1848_PLAYBACK_ENABLE) { dma_cont = &chip->p_dma; if (on) { dma_cont->prepare(dma_cont, 0); @@ -476,7 +476,7 @@ static void cs4231_dma_trigger(struct sn dma_cont->enable(dma_cont, 0); } } - if (what & CS4231_RECORD_ENABLE) { + if (what & AD1848_CAPTURE_ENABLE) { dma_cont = &chip->c_dma; if (on) { dma_cont->prepare(dma_cont, 1); @@ -505,10 +505,10 @@ static int snd_cs4231_trigger(struct snd snd_pcm_group_for_each_entry(s, substream) { if (s == chip->playback_substream) { - what |= CS4231_PLAYBACK_ENABLE; + what |= AD1848_PLAYBACK_ENABLE; snd_pcm_trigger_done(s, substream); } else if (s == chip->capture_substream) { - what |= CS4231_RECORD_ENABLE; + what |= AD1848_CAPTURE_ENABLE; snd_pcm_trigger_done(s, substream); } } @@ -516,13 +516,13 @@ static int snd_cs4231_trigger(struct snd spin_lock_irqsave(&chip->lock, flags); if (cmd == SNDRV_PCM_TRIGGER_START) { cs4231_dma_trigger(substream, what, 1); - chip->image[CS4231_IFACE_CTRL] |= what; + chip->image[AD1848_IFACE_CTRL] |= what; } else { cs4231_dma_trigger(substream, what, 0); - chip->image[CS4231_IFACE_CTRL] &= ~what; + chip->image[AD1848_IFACE_CTRL] &= ~what; } - snd_cs4231_out(chip, CS4231_IFACE_CTRL, - chip->image[CS4231_IFACE_CTRL]); + snd_cs4231_out(chip, AD1848_IFACE_CTRL, + chip->image[AD1848_IFACE_CTRL]); spin_unlock_irqrestore(&chip->lock, flags); break; } @@ -554,16 +554,16 @@ static unsigned char snd_cs4231_get_form { unsigned char rformat; - rformat = CS4231_LINEAR_8; + rformat = AD1848_LINEAR_8; switch (format) { case SNDRV_PCM_FORMAT_MU_LAW: - rformat = CS4231_ULAW_8; + rformat = AD1848_ULAW_8; break; case SNDRV_PCM_FORMAT_A_LAW: - rformat = CS4231_ALAW_8; + rformat = AD1848_ALAW_8; break; case SNDRV_PCM_FORMAT_S16_LE: - rformat = CS4231_LINEAR_16; + rformat = AD1848_LINEAR_16; break; case SNDRV_PCM_FORMAT_S16_BE: rformat = CS4231_LINEAR_16_BIG; @@ -573,7 +573,7 @@ static unsigned char snd_cs4231_get_form break; } if (channels > 1) - rformat |= CS4231_STEREO; + rformat |= AD1848_STEREO; return rformat; } @@ -588,25 +588,25 @@ static void snd_cs4231_calibrate_mute(st return; } if (!mute) { - snd_cs4231_dout(chip, CS4231_LEFT_INPUT, - chip->image[CS4231_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_INPUT, - chip->image[CS4231_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LOOPBACK, - chip->image[CS4231_LOOPBACK]); - } - snd_cs4231_dout(chip, CS4231_AUX1_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX1_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX1_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_LEFT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_LEFT_INPUT]); - snd_cs4231_dout(chip, CS4231_AUX2_RIGHT_INPUT, - mute ? 0x80 : chip->image[CS4231_AUX2_RIGHT_INPUT]); - snd_cs4231_dout(chip, CS4231_LEFT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_LEFT_OUTPUT]); - snd_cs4231_dout(chip, CS4231_RIGHT_OUTPUT, - mute ? 0x80 : chip->image[CS4231_RIGHT_OUTPUT]); + snd_cs4231_dout(chip, AD1848_LEFT_INPUT, + chip->image[AD1848_LEFT_INPUT]); + snd_cs4231_dout(chip, AD1848_RIGHT_INPUT, + chip->image[AD1848_RIGHT_INPUT]); + snd_cs4231_dout(chip, AD1848_LOOPBACK, + chip->image[AD1848_LOOPBACK]); + } + snd_cs4231_dout(chip, AD1848_AUX1_LEFT_INPUT, + mute ? 0x80 : chip->image[AD1848_AUX1_LEFT_INPUT]); + snd_cs4231_dout(chip, AD1848_AUX1_RIGHT_INPUT, + mute ? 0x80 : chip->image[AD1848_AUX1_RIGHT_INPUT]); + snd_cs4231_dout(chip, AD1848_AUX2_LEFT_INPUT, + mute ? 0x80 : chip->image[AD1848_AUX2_LEFT_INPUT]); + snd_cs4231_dout(chip, AD1848_AUX2_RIGHT_INPUT, + mute ? 0x80 : chip->image[AD1848_AUX2_RIGHT_INPUT]); + snd_cs4231_dout(chip, AD1848_LEFT_OUTPUT, + mute ? 0x80 : chip->image[AD1848_LEFT_OUTPUT]); + snd_cs4231_dout(chip, AD1848_RIGHT_OUTPUT, + mute ? 0x80 : chip->image[AD1848_RIGHT_OUTPUT]); snd_cs4231_dout(chip, CS4231_LEFT_LINE_IN, mute ? 0x80 : chip->image[CS4231_LEFT_LINE_IN]); snd_cs4231_dout(chip, CS4231_RIGHT_LINE_IN, @@ -629,8 +629,8 @@ static void snd_cs4231_playback_format(s snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, - (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) ? + snd_cs4231_out(chip, AD1848_DATA_FORMAT, + (chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE) ? (pdfr & 0xf0) | (chip->image[CS4231_REC_FORMAT] & 0x0f) : pdfr); spin_unlock_irqrestore(&chip->lock, flags); @@ -653,9 +653,9 @@ static void snd_cs4231_capture_format(st snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) { - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, - ((chip->image[CS4231_PLAYBK_FORMAT]) & 0xf0) | + if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) { + snd_cs4231_out(chip, AD1848_DATA_FORMAT, + ((chip->image[AD1848_DATA_FORMAT]) & 0xf0) | (cdfr & 0x0f)); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); @@ -679,7 +679,7 @@ static unsigned long snd_cs4231_timer_re { struct snd_cs4231 *chip = snd_timer_chip(timer); - return chip->image[CS4231_PLAYBK_FORMAT] & 1 ? 9969 : 9920; + return chip->image[AD1848_DATA_FORMAT] & 1 ? 9969 : 9920; } static int snd_cs4231_timer_start(struct snd_timer *timer) @@ -733,13 +733,13 @@ static void __init snd_cs4231_init(struc #endif snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | - CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | - CS4231_RECORD_PIO | - CS4231_CALIB_MODE); - chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | + AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | + AD1848_CAPTURE_PIO | + AD1848_CALIB_MODE); + chip->image[AD1848_IFACE_CTRL] |= AD1848_AUTOCALIB; + snd_cs4231_out(chip, AD1848_IFACE_CTRL, chip->image[AD1848_IFACE_CTRL]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); @@ -766,8 +766,8 @@ static void __init snd_cs4231_init(struc snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, - chip->image[CS4231_PLAYBK_FORMAT]); + snd_cs4231_out(chip, AD1848_DATA_FORMAT, + chip->image[AD1848_DATA_FORMAT]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); @@ -841,17 +841,17 @@ static void snd_cs4231_close(struct snd_ /* now disable record & playback */ - if (chip->image[CS4231_IFACE_CTRL] & - (CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO)) { + if (chip->image[AD1848_IFACE_CTRL] & + (AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO)) { spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - chip->image[CS4231_IFACE_CTRL] &= - ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); - snd_cs4231_out(chip, CS4231_IFACE_CTRL, - chip->image[CS4231_IFACE_CTRL]); + chip->image[AD1848_IFACE_CTRL] &= + ~(AD1848_PLAYBACK_ENABLE | AD1848_PLAYBACK_PIO | + AD1848_CAPTURE_ENABLE | AD1848_CAPTURE_PIO); + snd_cs4231_out(chip, AD1848_IFACE_CTRL, + chip->image[AD1848_IFACE_CTRL]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->lock, flags); @@ -929,8 +929,8 @@ static int snd_cs4231_playback_prepare(s spin_lock_irqsave(&chip->lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | - CS4231_PLAYBACK_PIO); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_PLAYBACK_ENABLE | + AD1848_PLAYBACK_PIO); BUG_ON(runtime->period_size > 0xffff + 1); @@ -965,8 +965,8 @@ static int snd_cs4231_capture_prepare(st unsigned long flags; spin_lock_irqsave(&chip->lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_RECORD_ENABLE | - CS4231_RECORD_PIO); + chip->image[AD1848_IFACE_CTRL] &= ~(AD1848_CAPTURE_ENABLE | + AD1848_CAPTURE_PIO); chip->c_periods_sent = 0; @@ -981,7 +981,7 @@ static void snd_cs4231_overrange(struct unsigned char res; spin_lock_irqsave(&chip->lock, flags); - res = snd_cs4231_in(chip, CS4231_TEST_INIT); + res = snd_cs4231_in(chip, AD1848_TEST_INIT); spin_unlock_irqrestore(&chip->lock, flags); /* detect overrange only above 0dB; may be user selectable? */ @@ -991,7 +991,7 @@ static void snd_cs4231_overrange(struct static void snd_cs4231_play_callback(struct snd_cs4231 *chip) { - if (chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE) { + if (chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE) { snd_pcm_period_elapsed(chip->playback_substream); snd_cs4231_advance_dma(&chip->p_dma, chip->playback_substream, &chip->p_periods_sent); @@ -1000,7 +1000,7 @@ static void snd_cs4231_play_callback(str static void snd_cs4231_capture_callback(struct snd_cs4231 *chip) { - if (chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE) { + if (chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE) { snd_pcm_period_elapsed(chip->capture_substream); snd_cs4231_advance_dma(&chip->c_dma, chip->capture_substream, &chip->c_periods_sent); @@ -1014,7 +1014,7 @@ static snd_pcm_uframes_t snd_cs4231_play struct cs4231_dma_control *dma_cont = &chip->p_dma; size_t ptr; - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) + if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_PLAYBACK_ENABLE)) return 0; ptr = dma_cont->address(dma_cont); if (ptr != 0) @@ -1030,7 +1030,7 @@ static snd_pcm_uframes_t snd_cs4231_capt struct cs4231_dma_control *dma_cont = &chip->c_dma; size_t ptr; - if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE)) + if (!(chip->image[AD1848_IFACE_CTRL] & AD1848_CAPTURE_ENABLE)) return 0; ptr = dma_cont->address(dma_cont); if (ptr != 0) @@ -1049,12 +1049,12 @@ static int __init snd_cs4231_probe(struc for (i = 0; i < 50; i++) { mb(); - if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & AD1848_INIT) msleep(2); else { spin_lock_irqsave(&chip->lock, flags); - snd_cs4231_out(chip, CS4231_MISC_INFO, CS4231_MODE2); - id = snd_cs4231_in(chip, CS4231_MISC_INFO) & 0x0f; + snd_cs4231_out(chip, AD1848_MISC_INFO, CS4231_MODE2); + id = snd_cs4231_in(chip, AD1848_MISC_INFO) & 0x0f; vers = snd_cs4231_in(chip, CS4231_VERSION); spin_unlock_irqrestore(&chip->lock, flags); if (id == 0x0a) @@ -1074,9 +1074,9 @@ static int __init snd_cs4231_probe(struc spin_unlock_irqrestore(&chip->lock, flags); - chip->image[CS4231_MISC_INFO] = CS4231_MODE2; - chip->image[CS4231_IFACE_CTRL] = - chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA; + chip->image[AD1848_MISC_INFO] = CS4231_MODE2; + chip->image[AD1848_IFACE_CTRL] = + chip->image[AD1848_IFACE_CTRL] & ~AD1848_SINGLE_DMA; chip->image[CS4231_ALT_FEATURE_1] = 0x80; chip->image[CS4231_ALT_FEATURE_2] = 0x01; if (vers & 0x20) @@ -1319,9 +1319,9 @@ static int snd_cs4231_get_mux(struct snd spin_lock_irqsave(&chip->lock, flags); ucontrol->value.enumerated.item[0] = - (chip->image[CS4231_LEFT_INPUT] & CS4231_MIXS_ALL) >> 6; + (chip->image[AD1848_LEFT_INPUT] & AD1848_MIXS_ALL) >> 6; ucontrol->value.enumerated.item[1] = - (chip->image[CS4231_RIGHT_INPUT] & CS4231_MIXS_ALL) >> 6; + (chip->image[AD1848_RIGHT_INPUT] & AD1848_MIXS_ALL) >> 6; spin_unlock_irqrestore(&chip->lock, flags); return 0; @@ -1343,12 +1343,12 @@ static int snd_cs4231_put_mux(struct snd spin_lock_irqsave(&chip->lock, flags); - left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left; - right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right; - change = left != chip->image[CS4231_LEFT_INPUT] || - right != chip->image[CS4231_RIGHT_INPUT]; - snd_cs4231_out(chip, CS4231_LEFT_INPUT, left); - snd_cs4231_out(chip, CS4231_RIGHT_INPUT, right); + left = (chip->image[AD1848_LEFT_INPUT] & ~AD1848_MIXS_ALL) | left; + right = (chip->image[AD1848_RIGHT_INPUT] & ~AD1848_MIXS_ALL) | right; + change = left != chip->image[AD1848_LEFT_INPUT] || + right != chip->image[AD1848_RIGHT_INPUT]; + snd_cs4231_out(chip, AD1848_LEFT_INPUT, left); + snd_cs4231_out(chip, AD1848_RIGHT_INPUT, right); spin_unlock_irqrestore(&chip->lock, flags); @@ -1517,27 +1517,27 @@ static int snd_cs4231_put_double(struct ((shift_right) << 19) | ((mask) << 24) | ((invert) << 22) } static struct snd_kcontrol_new snd_cs4231_controls[] __initdata = { -CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, - CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, - CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, AD1848_LEFT_OUTPUT, + AD1848_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, AD1848_LEFT_OUTPUT, + AD1848_RIGHT_OUTPUT, 0, 0, 63, 1), CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, - CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, - CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, - CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, - CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, AD1848_AUX1_LEFT_INPUT, + AD1848_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, AD1848_AUX1_LEFT_INPUT, + AD1848_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 1, AD1848_AUX2_LEFT_INPUT, + AD1848_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 1, AD1848_AUX2_LEFT_INPUT, + AD1848_AUX2_RIGHT_INPUT, 0, 0, 31, 1), CS4231_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), CS4231_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), CS4231_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, +CS4231_DOUBLE("Capture Volume", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 0, 0, 15, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1546,13 +1546,13 @@ CS4231_DOUBLE("Capture Volume", 0, CS423 .get = snd_cs4231_get_mux, .put = snd_cs4231_put_mux, }, -CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, +CS4231_DOUBLE("Mic Boost", 0, AD1848_LEFT_INPUT, AD1848_RIGHT_INPUT, 5, 5, 1, 0), -CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1), +CS4231_SINGLE("Loopback Capture Switch", 0, AD1848_LOOPBACK, 0, 1, 0), +CS4231_SINGLE("Loopback Capture Volume", 0, AD1848_LOOPBACK, 2, 63, 1), /* SPARC specific uses of XCTL{0,1} general purpose outputs. */ -CS4231_SINGLE("Line Out Switch", 0, CS4231_PIN_CTRL, 6, 1, 1), -CS4231_SINGLE("Headphone Out Switch", 0, CS4231_PIN_CTRL, 7, 1, 1) +CS4231_SINGLE("Line Out Switch", 0, AD1848_PIN_CTRL, 6, 1, 1), +CS4231_SINGLE("Headphone Out Switch", 0, AD1848_PIN_CTRL, 7, 1, 1) }; static int __init snd_cs4231_mixer(struct snd_card *card) @@ -1647,7 +1647,7 @@ static irqreturn_t snd_cs4231_sbus_inter struct snd_cs4231 *chip = dev_id; /*This is IRQ is not raised by the cs4231*/ - if (!(__cs4231_readb(chip, CS4231U(chip, STATUS)) & CS4231_GLOBALIRQ)) + if (!(__cs4231_readb(chip, CS4231U(chip, STATUS)) & AD1848_GLOBALIRQ)) return IRQ_NONE; /* ACK the APC interrupt. */ ---------------------------------------------------------------------- Z Map24 Twoja droga jest prosta! Sprawdz! kliknij >>> http://link.interia.pl/f1e3b

4 7

[alsa-devel] [PATCH] Added model selection for iMac 24"
by Travis Place 19 Jun '08

19 Jun '08

Changelog: Added the SSID of a known iMac 24" to automatically use ALC885_IMAC24 quirk. --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -6196,6 +6196,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), + SND_PCI_QUIRK(0x106b, 0x00a0, "Apple iMac 24''", ALC885_IMAC24), {} }; Signed-off-by: Travis Place <wishie(a)wishie.net>

2 2

[alsa-devel] What does snd_pcm_delay() actually return?
by Lennart Poettering 19 Jun '08

19 Jun '08

Takashi, Jaroslav, could you please explain what exactly snd_pcm_delay() returns? Some applications (such as WINE) assume it is the time that would pass until we reach an underrun if we would stop writing any further data to the PCM device. Other applications (such as most media players) use it for time synchronisation. i.e. assume that it is the time that passes until a sample I write to a PCM device now would take to be played. Why does this make a difference? For all drivers/ioplug backends where audio is not directly written to the hardware buffer of the audio device the underrun might happen much earlier than the last sample written is heard. This seems to be the case of USB audio for example. If I fill up the hw buffer completely, I will be notified about a buffer underrun much earlier than the buffer size would suggest. Another example is the PulseAudio backend: for each client stream we maintain a private playback buffer. After that buffer there is the hardware buffer. If the client buffer runs empty we'd like to signal an underrun. However, in that situation it still will take some time until the underrun is really hearable, because the audio first has to pass the second buffer in line, the hardware buffer. Or in other words: in some backends (PCI/DMA) after a sample is read from the ALSA playback buffer and the read index is increased it is immediately heard. In others it will still take substantial time until it is heard. The current API doesn't expose any information about how long that additional time can be, and it is not clear if snd_pcm_delay() includes or does not include this extra time in its result. The ALSA documentation claims that snd_pcm_delay() returns the "distance between current application frame position and sound frame position". That would point that WINE's interpretation is correct. (i.e. ignore the extra time) However, the USB audio driver actually seems to return the total time delay as it is useful for synchronization, so follows what media players expect. (i.e. take the extra time into account) I personally believe that the USB audio driver does it right because the most important use for snd_pcm_delay() is to achieve synchronization between audio and video. However, the other value is very important too. Not just for WINE, but also for my own work, PulseAudio. Most of the time the small difference between those two values doesn't really matter, since the difference is very small and for classic PCI/DMI devices zero. However, with USB I am now encountering this problem, because I cannot reliably estimate from the data ALSA supplies me with when the next underrun would happen. Also, the WINE people are pretty vocal that I am not right with my interepretation of the sitaution. Takashi, Jaroslav, could you please eleborate on this, and explain which interpretation is the correct one? Also, regardless which one is the correct one, could we please add a second API function the allows clients to query the other value? I have already raised this issue in differents words two times before [1]. I never got a comment from you on this. So, please please with cream on top, respond! Thank you very much, Lennart [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-April/007354.html -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

8 52

[alsa-devel] Linux Plumbers Conference CFP
by Lennart Poettering 19 Jun '08

19 Jun '08

Heya! Anyone doing (audio) infrastructure work on Linux? If so, please make sure to join us at the Linux Plumbers Conference in Portland/Oregon in September. I will be doing an Audio track there and would like to make sure that everyone who does audio infrastructure work will submit a paper, or at least attend, or maybe just know about this conference! It's a conference about the kernel-userspace interfacing, about the lower levels in our software stack. For further details, checkout the conference website: http://www.linuxplumbersconf.org/ Or this blog story of mine: http://0pointer.de/blog/projects/plumbersconf.html Spread the word! Thanks, Lennart (Sorry for this Spam!) -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] [PATCH] Support for MSI K8N v1.0 onboard CA0106 based soundcard
by Travis Place 19 Jun '08

19 Jun '08

NOTE: The Exact model number of the card onboard is unknown to me (SB0438). Not sure how I am to find this information either. For the time being, i have just used the same model as seems to be on the other (updated?) version of the motherboard. Changelog: Add support for the MSI K8N Diamond Motherboard (version 1.0) onboard CA0106 soundcard. --- ca0106_main.c 2008-06-18 04:24:05.000000000 +1000 +++ ca0106_main.c.wishie 2008-06-18 04:39:26.000000000 +1000 @@ -249,6 +249,10 @@ .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , + { .serial = 0x10091102, + .name = "MSI K8N Diamond MB version 1.0 [SB0438]", + .gpio_type = 2, + .i2c_adc = 1 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". Signed-off-by: Travis Place <wishie(a)wishie.net>

2 1

[alsa-devel] Nvida HDA MCP61 - Realtek ALC622 question
by randall 19 Jun '08

19 Jun '08

Andy: Who works on the snd-hda-intel sound module? I have been unable to get the codec working on my Athlon 64 3200+ processor (64 bit libs) despite gradually working my way up from Alsa 1.0.14 to 1.0.17.git and fighting YaST2 issues with Suse 10.3 Since I do software development, I'd like to add my $0.02 worth of coding to get the Alsa code working for my ECS EliteGroup motherboard as I have seen others posting to the web frustrated by the same problem (indeed openSuse has a web site devoted to soundcard issues, but no one has yet fixed this problem) I did a dump using the alsa-info.sh to pastebin, but probably have to run that again. http://pastebin.ca/1048637 is the latest dump of my board. Randall

2 1

[alsa-devel] snd-hda-intel module now working
by randall 19 Jun '08

19 Jun '08

After rebuilding the kernel about 5 times and double-checking the KMixer controls, I have finally successfully been able to get the onboard sound working. Took 2 days tweaking and rebuilding the kernel. I used the "3stack-6ch-dig" setting for the model and turned on a SND debug variable. - Randall

2 2

[alsa-devel] [PATCH - asihpi 2/2] Disable S24_3LE incompatible with 2^N buffer size.
by linux＠audioscience.com 19 Jun '08

19 Jun '08

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> diff --git a/pci/asihpi/asihpi.c b/pci/asihpi/asihpi.c index ab90b08..c715ad2 100644 --- a/pci/asihpi/asihpi.c +++ b/pci/asihpi/asihpi.c @@ -328,21 +328,28 @@ static void print_hwparams(struct snd_pcm_hw_params *p) static snd_pcm_format_t hpi_to_alsa_formats[] = { -1, /* INVALID */ - SNDRV_PCM_FORMAT_U8, /* { HPI_FORMAT_PCM8_UNSIGNED 1 */ - SNDRV_PCM_FORMAT_S16, /* { HPI_FORMAT_PCM16_SIGNED 2 */ - -1, /* { HPI_FORMAT_MPEG_L1 3 */ - SNDRV_PCM_FORMAT_MPEG, /* { HPI_FORMAT_MPEG_L2 4 */ - SNDRV_PCM_FORMAT_MPEG, /* { HPI_FORMAT_MPEG_L3 5 */ - -1, /* { HPI_FORMAT_DOLBY_AC2 6 */ - -1, /* { HPI_FORMAT_DOLBY_AC3 7 */ - SNDRV_PCM_FORMAT_S16_BE,/* { HPI_FORMAT_PCM16_BIGENDIAN 8 */ - -1, /* { HPI_FORMAT_AA_TAGIT1_HITS 9 */ - -1, /* { HPI_FORMAT_AA_TAGIT1_INSERTS 10 */ - SNDRV_PCM_FORMAT_S32, /* { HPI_FORMAT_PCM32_SIGNED 11 */ - -1, /* { HPI_FORMAT_RAW_BITSTREAM 12 */ - -1, /* { HPI_FORMAT_AA_TAGIT1_HITS_EX1 13 */ - SNDRV_PCM_FORMAT_FLOAT, /* { HPI_FORMAT_PCM32_FLOAT 14 */ - SNDRV_PCM_FORMAT_S24 /* { HPI_FORMAT_PCM24_SIGNED 15 */ + SNDRV_PCM_FORMAT_U8, /* HPI_FORMAT_PCM8_UNSIGNED 1 */ + SNDRV_PCM_FORMAT_S16, /* HPI_FORMAT_PCM16_SIGNED 2 */ + -1, /* HPI_FORMAT_MPEG_L1 3 */ + SNDRV_PCM_FORMAT_MPEG, /* HPI_FORMAT_MPEG_L2 4 */ + SNDRV_PCM_FORMAT_MPEG, /* HPI_FORMAT_MPEG_L3 5 */ + -1, /* HPI_FORMAT_DOLBY_AC2 6 */ + -1, /* HPI_FORMAT_DOLBY_AC3 7 */ + SNDRV_PCM_FORMAT_S16_BE,/* HPI_FORMAT_PCM16_BIGENDIAN 8 */ + -1, /* HPI_FORMAT_AA_TAGIT1_HITS 9 */ + -1, /* HPI_FORMAT_AA_TAGIT1_INSERTS 10 */ + SNDRV_PCM_FORMAT_S32, /* HPI_FORMAT_PCM32_SIGNED 11 */ + -1, /* HPI_FORMAT_RAW_BITSTREAM 12 */ + -1, /* HPI_FORMAT_AA_TAGIT1_HITS_EX1 13 */ + SNDRV_PCM_FORMAT_FLOAT, /* HPI_FORMAT_PCM32_FLOAT 14 */ +#if 1 + /* ALSA can't handle 3 byte sample size together with power-of-2 + * constraint on buffer_bytes, so disable this format + */ + -1 +#else + /* SNDRV_PCM_FORMAT_S24_3LE */ /* { HPI_FORMAT_PCM24_SIGNED 15 */ +#endif }; @@ -978,7 +985,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) runtime->hw = snd_card_asihpi_playback; /* Strictly only necessary for HPI6205 adapters */ err = snd_pcm_hw_constraint_pow2(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE); + SNDRV_PCM_HW_PARAM_BUFFER_BYTES); if (err < 0) return err; @@ -1195,7 +1202,7 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) /* Strictly only necessary for HPI6205 adapters */ err = snd_pcm_hw_constraint_pow2(runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE); + SNDRV_PCM_HW_PARAM_BUFFER_BYTES); if (err < 0) return err; @@ -2025,6 +2032,7 @@ static int snd_asihpi_meter_get(struct snd_kcontrol *kcontrol, short anGain0_01dB[HPI_MAX_CHANNELS], i; u16 err; + //err = HPI_MeterGetRms(phSubSys, hControl, anGain0_01dB); err = HPI_MeterGetPeak(phSubSys, hControl, anGain0_01dB); for (i = 0; i < HPI_MAX_CHANNELS; i++) { -- 1.5.4.3

3 2

[alsa-devel] [PATCH][asihpi] Meter control return peak.
by Eliot Blennerhassett 19 Jun '08

19 Jun '08

Use Peak meter instead of Rms meter because it is supported by all card families. Minor checkpatch cleanups. Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> --- diff --git a/pci/asihpi/asihpi.c b/pci/asihpi/asihpi.c index 8cbefab..ab90b08 100644 --- a/pci/asihpi/asihpi.c +++ b/pci/asihpi/asihpi.c @@ -383,15 +383,15 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, rate_max = 100000; } else { /* on cards without SRC, - valid rates are determined by sampleclock */ + valid rates are determined by sampleclock */ err = HPI_MixerGetControl(phSubSys, asihpi->hMixer, HPI_SOURCENODE_CLOCK_SOURCE, 0, 0, 0, HPI_CONTROL_SAMPLECLOCK, &hControl); for (idx = 0; idx < 100; idx++) { if (HPI_ControlQuery(phSubSys, hControl, - HPI_SAMPLECLOCK_SAMPLERATE, idx, 0, - &sampleRate)) + HPI_SAMPLECLOCK_SAMPLERATE, + idx, 0, &sampleRate)) break; rate_min = min(rate_min, sampleRate); @@ -443,7 +443,7 @@ static void snd_card_asihpi_pcm_samplerates(struct snd_card_asihpi *asihpi, } } - /*printk(KERN_INFO "Supported rates %X %d %d\n", + /* printk(KERN_INFO "Supported rates %X %d %d\n", rates, rate_min, rate_max); */ pcmhw->rates = rates; pcmhw->rate_min = rate_min; @@ -885,7 +885,7 @@ static void snd_card_asihpi_playback_format(struct snd_card_asihpi *asihpi, u32 hStream, struct snd_pcm_hardware *pcmhw) { - struct hpi_format hpi_format; + struct hpi_format hpi_format; u16 wFormat; u16 err; u32 hControl; @@ -1126,10 +1126,10 @@ static void snd_card_asihpi_capture_format(struct snd_card_asihpi *asihpi, for (wFormat = HPI_FORMAT_PCM8_UNSIGNED; wFormat <= HPI_FORMAT_PCM24_SIGNED; wFormat++) { - HPI_FormatCreate( - &hpi_format, 2, wFormat, dwSampleRate, 128000, 0); - - err = HPI_InStreamQueryFormat(phSubSys, hStream, + HPI_FormatCreate(&hpi_format, 2, wFormat, dwSampleRate, + 128000, 0); + err = + HPI_InStreamQueryFormat(phSubSys, hStream, &hpi_format); if (!err) pcmhw->formats |= @@ -2025,7 +2025,7 @@ static int snd_asihpi_meter_get(struct snd_kcontrol *kcontrol, short anGain0_01dB[HPI_MAX_CHANNELS], i; u16 err; - err = HPI_MeterGetRms(phSubSys, hControl, anGain0_01dB); + err = HPI_MeterGetPeak(phSubSys, hControl, anGain0_01dB); for (i = 0; i < HPI_MAX_CHANNELS; i++) { #if ASIHPI_LINEAR_METERS @@ -2796,7 +2796,7 @@ int __devinit snd_asihpi_probe(struct pci_dev *pci_dev, asihpi->support_mmap = (!err); asihpi->support_mrx = (((asihpi->wType & 0xFF00) == 0x8900) || - ((asihpi->wType & 0xF000) == 0x6000)); + ((asihpi->wType & 0xF000) == 0x6000)); printk(KERN_INFO "Supports mmap:%d grouping:%d\n", -- -- Eliot Blennerhassett www.audioscience.com

2 1

[alsa-devel] question on hda_codec.c behavior
by randall 18 Jun '08

18 Jun '08

I've not yet been able to get the Nvidia MCP61 (RealTek ALC662) onboard sound working (EliteGroup Computer Systems) on the Athlon 64 3200+ motherboard. A hwinfo is enclosed: 21: PCI 05.0: 0403 Audio device [Created at pci.301] UDI: /org/freedesktop/Hal/devices/pci_10de_3f0 Unique ID: CvwD.9hpqWdunz16 SysFS ID: /devices/pci0000:00/0000:00:05.0 SysFS BusID: 0000:00:05.0 Hardware Class: sound Model: "Elitegroup MCP61 High Definition Audio" Vendor: pci 0x10de "nVidia Corporation" Device: pci 0x03f0 "MCP61 High Definition Audio" SubVendor: pci 0x1019 "Elitegroup Computer Systems" SubDevice: pci 0x2151 Revision: 0xa2 Driver: "HDA Intel" Driver Modules: "snd_hda_intel" Memory Range: 0xfe028000-0xfe02bfff (rw,non-prefetchable) IRQ: 22 (409 events) Module Alias: "pci:v000010DEd000003F0sv00001019sd00002151bc04sc03i00" Driver Info #0: Driver Status: snd_hda_intel is active Driver Activation Cmd: "modprobe snd_hda_intel" Config Status: cfg=no, avail=yes, need=no, active=unknown and the pastebin is http://pastebin.ca/1048637 for all the information on the sound setup. YaST2 does the typical sound card editing, and you can hear the speakers click when the card is (de)activated. But no sound is available, even using the speaker-test command or running alsaconf. I've gradually worked my way up from v1.0.14 to the current v1.0.17rl1 but have no success yet in getting the sound modules to work for the Nvidia MCP61 setup. After posting in the install newsgroup one of the persons trying to help me out suggested that I send a note to the developer's group for the following reason, the /var/log/message file contains the following upon booting up > hda_codec: Unknown model for ALC662, trying auto-probe from BIOS... > ALSA sound/pci/hda/hda_codec.c:2333: autoconfig: line_outs=1 > (0x14/0x0/0x0/0x0/0x0) > ALSA sound/pci/hda/hda_codec.c:2337: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > ALSA sound/pci/hda/hda_codec.c:2341: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) > ALSA sound/pci/hda/hda_codec.c:2349: inputs: mic=0x18, fmic=0x19, > line=0x1a, fline=0x0, cd=0x0, aux=0x0 > hda_codec: num_steps = 0 for NID=0xc And the thought was that "Unknown model" should not be occurring. The /etc/modprobe.d/sound file has one of the allowed models for this codec. Any ideas? Does hda_codec.c need a patch? Randall

1 0

Re: [alsa-devel] Listing raw midi devices "The Right Way"?
by Clemens Ladisch 18 Jun '08

18 Jun '08

Jacob Ole Juul Kolding wrote: > I already found a code example that reads raw midi packages form a > specified device but I need to have a complete list of ALL raw midi > capable devices and ports on the system. > > What is the "Right Way" for an app to build such a list? See the source code of amidi. You probably should be using the sequencer instead of the raw MIDI interface, in which case look at the source code of arecordmidi. HTH Clemens

2 2

[alsa-devel] Listing raw midi devices "The Right Way"?
by Jacob Ole Juul Kolding 18 Jun '08

18 Jun '08

Hello all I'm working on an app that used to run on Linux but then moved to OS X and gained MIDI capabilities. Now it's moving back to Linux so I need to be able to read midi events from devices. I already found a code example that reads raw midi packages form a specified device but I need to have a complete list of ALL raw midi capable devices and ports on the system. What is the "Right Way" for an app to build such a list? Sincerely Jacob Kolding Juvul Tech

1 0

[alsa-devel] [PATCH] ASoC: fix PM=n build
by Mark Brown 17 Jun '08

17 Jun '08

From: Randy Dunlap <randy.dunlap(a)oracle.com> Fix sound/soc build failure when CONFIG_PM=n: linux-next-20080617/sound/soc/soc-core.c:829: error: 'soc_resume_deferred' undeclared (first use in this function) soc3.out:make[3]: *** [sound/soc/soc-core.o] Error 1 Signed-off-by: Randy Dunlap <randy.dunlap(a)oracle.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b931039..bdbbc6a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -825,8 +825,10 @@ static int soc_probe(struct platform_device *pdev) /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); +#ifdef CONFIG_PM /* deferred resume work */ INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); +#endif return 0; -- 1.5.2.5

1 0

[alsa-devel] hda-intel on Asus M2N-X Plus NVIDIA GeForce 6100/nForce 430 mobo
by Nigel Henry 17 Jun '08

17 Jun '08

I've just built a new machine, and expected problems with the onboard soundcard, and also have problems with the onboard NIC, but that's another problem, which is resolved by installing my last remaining realtek 8139 PCI card. Anyway the mobo is an Asus M2N-X Plus NVIDIA GeForce 6100/nForce 430, and the Southbridge (which appears to handle allsorts of stuff) is MCP61. the specs for the soundcard from the Asus site are below. ALC662 6-Channel High Definition Audio Azalia CODEC Support Jack-Sensing, Enumeration, Multi-streaming S/PDIF out on back I/O port This is using the Fedora 8 distro, which has alsa driver 1.0.15, and is probably a bit old. The card is not detected at bootup, but I did a modprobe snd-hda-intel, using various model options. the modules were loaded, but a cat /proc/asound/cards showed "no soundcards detected" I don't want to waste too much time on this onboard card, and will probably buy a cheap well supported card, just to get some sounds on this new machine. All the same I've run the alsa-info.sh script, and the link is below. http://pastebin.ca/1048647 Any comments, as always welcome. Btw: My other machines use Audigy2 soundblaster (emu10k1) on the first machine, and Ensoniq (ens1371) on the other machine, and no problems with these cards at all. I cannot repeat my opinions/thoughts about hda-intel cards. Thanks for all your work you folks. Nigel.

2 4

[alsa-devel] Thoughts on ASOC v2 driver architecture
by Jon Smirl 17 Jun '08

17 Jun '08

I'm trying to sort out an architecture that will let all of the code used in my ASOC drivers to be automatically loaded by the kernel. So if I'm confused and have some of this wrong please explain where I've gone astray. First, what are the physical hardware devices present? From the device tree it says I have two i2s controllers each with an i2c controlled codec. It also says I have a system wide DMA controller. And finally there is the "fabric" which represent the specific pcb (populated jacks, etc) used by the system (the root of the device tree). Upon booting the kernel it will find the five open firmware devices (two i2s, two codecs, one DMA) and load the appropriate of drivers for them. Note that this is the system wide DMA driver, not the ASOC one. Three drivers will be loaded and five devices will be created. The codec devices will each register one codec and multiple codec DAIs. Each i2c channel will register a platform_dai. The next part, the platform device, is where I'm having trouble. Why is the DMA subsystem in it's own ASOC device? I already have a separate system wide device for my DMA hardware. Looking at the implementations of a couple of the ASOC DMA drivers, there is nothing global about the driver, everything is done with per channel instances. Should the DMA code instead simply be a library that is linked to by the i2s drivers? And then move the entry points registered by the DMA driver up into the platform_dai? It seems to me like implementation of the DMA code should be a private matter for the platform_dai/i2s driver. Note that I'm not saying that there shouldn't be modularized ASOC DMA code, instead, why should ASOC be aware of this shared code? To see if this model would work I tried to visualize a system where one i2s controller is using DMA and the second is doing programmed IO. So if the DMA API is pushed into the platform_dai that would free up the platform device. It could then be used to support the fabric code (fabric is the motherboard specific bits of code). It should probably be renamed to the machine device instead of platform device. To handle the case of HDA not needing a fabric driver, ASOC would have a default internal machine driver that a fabric driver could override. This also implies that the fabric driver can't be the last driver registered with ASOC. There would be one global fabric driver, but it would get called on a per instance basis. You could implement (made up scenarios) two amps on the motherboard and only enough power to turn one of them on. Or maybe a front panel switch that switches which channel the front panel controls act on. I'd end up with this object model generic machine (optional machine specific fabric override) platform i2s (platform_dai, soc_snd_card) generic codec (choice of codec dai) platform i2s (platform_dai, soc_snd_card) generic codec (choice of codec dai) The platform specific i2s drivers are linked to a shared library of platform DMA code. -- Jon Smirl jonsmirl(a)gmail.com

4 22

[alsa-devel] git-http interface to ALSA git repositories?
by Timur Tabi 17 Jun '08

17 Jun '08

My company's firewall blocks the git protocol, so I need to use http:// to clone git repositories. What is the http path for the ALSA git repository? -- Timur Tabi Linux kernel developer at Freescale

3 2

[alsa-devel] Mute sound
by Jorge Peixoto de Morais Neto 17 Jun '08

17 Jun '08

Hi. Under kernel 2.6.23, I could get sound from the alsa device created by the module snd_bt87x. In kernel 2.6.24, there is no sound. Also, in kernel 2.6.23 the device would only appear if I modprobed the module with the option load_all=1. In kernel 2.6.24, the option load_all=1 is unnecessary for the device to appear. This is probably because I have in 2007 sent an email to this list with information about my board (it was responded by Takashi). If you need any addition other than what I provide below, ask at any time. jorge@jorge ~ $ uname -a Linux jorge 2.6.24-gentoo-r8d #4 Mon Jun 16 17:58:17 BRT 2008 i686 AMD Athlon(tm) XP 2600+ AuthenticAMD GNU/Linux jorge@jorge ~ $ sudo lspci -v 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host Bridge Subsystem: VIA Technologies, Inc. Unknown device 0000 Flags: bus master, 66MHz, medium devsel, latency 8 Memory at e0000000 (32-bit, prefetchable) [size=128M] Capabilities: [a0] AGP version 2.0 Capabilities: [c0] Power Management version 2 00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge (prog-if 00 [Normal decode]) Flags: bus master, 66MHz, medium devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: dde00000-dfefffff Prefetchable memory behind bridge: cdc00000-ddcfffff Capabilities: [80] Power Management version 2 00:07.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) Subsystem: Realtek Semiconductor Co., Ltd. RT8139 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ec00 [size=256] Memory at dfffff00 (32-bit, non-prefetchable) [size=256] Capabilities: [50] Power Management version 2 Kernel driver in use: 8139too 00:0b.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) Subsystem: PROLINK Microsystems Corp Unknown device 4011 Flags: bus master, medium devsel, latency 32, IRQ 5 Memory at dddfe000 (32-bit, prefetchable) [size=4K] Capabilities: [44] Vital Product Data <?> Capabilities: [4c] Power Management version 2 Kernel driver in use: bttv Kernel modules: bttv 00:0b.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) Subsystem: PROLINK Microsystems Corp Unknown device 4011 Flags: bus master, medium devsel, latency 32, IRQ 5 Memory at dddff000 (32-bit, prefetchable) [size=4K] Capabilities: [44] Vital Product Data <?> Capabilities: [4c] Power Management version 2 Kernel driver in use: Bt87x Kernel modules: snd-bt87x 00:0c.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10) Subsystem: Micro-Star International Co., Ltd. Unknown device 5900 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at e800 [size=256] Capabilities: [c0] Power Management version 2 Kernel driver in use: C-Media PCI Kernel modules: snd-cmipci 00:10.0 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at dc00 [size=32] Capabilities: [80] Power Management version 2 Kernel driver in use: uhci_hcd 00:10.1 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at e000 [size=32] Capabilities: [80] Power Management version 2 Kernel driver in use: uhci_hcd 00:10.2 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at e400 [size=32] Capabilities: [80] Power Management version 2 Kernel driver in use: uhci_hcd 00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82) (prog-if 20 [EHCI]) Subsystem: VIA Technologies, Inc. USB 2.0 Flags: bus master, medium devsel, latency 32, IRQ 10 Memory at dffffe00 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 Kernel driver in use: ehci_hcd 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge Subsystem: VIA Technologies, Inc. Unknown device 0000 Flags: bus master, stepping, medium devsel, latency 0 Capabilities: [c0] Power Management version 2 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc. VT82C586/B/VT82C686/A/B/VT8233/A/C/VT8235 PIPC Bus Master IDE Flags: bus master, medium devsel, latency 32, IRQ 255 [virtual] Memory at 000001f0 (32-bit, non-prefetchable) [size=8] [virtual] Memory at 000003f0 (type 3, non-prefetchable) [size=1] [virtual] Memory at 00000170 (32-bit, non-prefetchable) [size=8] [virtual] Memory at 00000370 (type 3, non-prefetchable) [size=1] I/O ports at fc00 [size=16] Capabilities: [c0] Power Management version 2 01:00.0 VGA compatible controller: nVidia Corporation NV17 [GeForce4 MX 440] (rev a3) (prog-if 00 [VGA controller]) Flags: bus master, 66MHz, medium devsel, latency 32, IRQ 11 Memory at de000000 (32-bit, non-prefetchable) [size=16M] Memory at d0000000 (32-bit, prefetchable) [size=128M] Memory at ddc80000 (32-bit, prefetchable) [size=512K] Expansion ROM at dfee0000 [disabled] [size=128K] Capabilities: [60] Power Management version 2 Capabilities: [44] AGP version 2.0 jorge@jorge ~ $ cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 6 model : 8 model name : AMD Athlon(tm) XP 2600+ stepping : 1 cpu MHz : 2133.528 cache size : 256 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 mmx fxsr sse syscall mmxext 3dnowext 3dnow ts bogomips : 4271.41 clflush size : 32 jorge@jorge /proc/asound $ ls Bt878 card0 card1 cards CMI8738 devices hwdep oss pcm timers version jorge@jorge /proc/asound $ cat cards 0 [CMI8738 ]: CMI8738-MC6 - C-Media CMI8738 C-Media CMI8738 (model 55) at 0xe800, irq 11 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xdddff000, irq 5 jorge@jorge /proc/asound $ cat devices 0: [ 0] : control 4: [ 0- 0]: hardware dependent 8: [ 0- 0]: raw midi 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 18: [ 0- 2]: digital audio playback 24: [ 0- 0]: digital audio capture 26: [ 0- 2]: digital audio capture 32: [ 1] : control 33: : timer 56: [ 1- 0]: digital audio capture 57: [ 1- 1]: digital audio capture jorge@jorge /proc/asound $ cat hwdep 00-00: OPL3 FM jorge@jorge /proc/asound $ cat pcm 00-02: CMI8738-MC6 : C-Media PCI IEC958 : playback 1 : capture 1 00-01: CMI8738-MC6 : C-Media PCI 2nd DAC : playback 1 00-00: CMI8738-MC6 : C-Media PCI DAC/ADC : playback 1 : capture 1 01-01: Bt87x Analog : Bt87x Analog : capture 1 01-00: Bt87x Digital : Bt87x Digital : capture 1 jorge@jorge /proc/asound $ cat version Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Nov 20 19:16:42 2007 UTC). -- Software is like sex: it is better when it is free - Linus Torvalds

1 1

Re: [alsa-devel] Problem regarding CA0106
by Travis Place 16 Jun '08

16 Jun '08

On Mon, 2008-06-16 at 18:00 +0200, Takashi Iwai wrote: > At Tue, 17 Jun 2008 01:52:08 +1000, > Travis Place wrote: > > > > On Mon, 2008-06-16 at 12:38 +0200, Takashi Iwai wrote: > > > At Sat, 14 Jun 2008 03:18:33 +1000, > > > Travis Place wrote: > > > > > > > > Just been speaking with a guy in #alsa about the problems with his "MSI > > > > K8N Diamond MB" which has an onboard "SB Live 24bit" or "Audigy". > > > > > > > > While looking through ca0106_main.c in GIT, i saw the following section: > > > > > > > > /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 > > > > */ > > > > /* SB0438 > > > > * CTRL:CA0106-DAT > > > > * ADC: WM8775SEDS > > > > * DAC: CS4382-KQZ > > > > */ > > > > { .serial = 0x10091462, > > > > .name = "MSI K8N Diamond MB [SB0438]", > > > > .gpio_type = 2, > > > > .i2c_adc = 1 } , > > > > > > > > For this guys sound to work, we had to change the serial line to: > > > > > > > > { .serial = 0x10091102, > > > > > > Does it mean that this number 1009:1102 is missing, or a wrong > > > 1009:1462 entry matches with his hardware? > > > > > Well, it appears to me (due to another report filed previously) that > > the motherboard is a 'newer revision' and has a different ID for the > > soundcard. > > Hm, so you didn't receive alsa-info.sh or "lspci -nv" output? > Of course I got alsa-info.sh output :) (asks himself why he didnt include it though) http://pastebin.ca/1048644 > > > In the former case, it happens often. We just need to add a new entry > > > for the newer model. In the latter case, it's odd. > > > > > > > I think that adding the new ID to the driver wont cause any immediate > > problems. The old value (1009:1462) should remain, until we can confirm > > it against another K8N Diamond motherboard (although im not sure how > > common these boards are). > > Yeah, I guess so, too. But just to be sure... > Well, we'll just have to see what happens, I guess. For the record, in both older ALSA versions, and 1.0.16, changing only the serial seems to be sufficient. So adding another serial should be an easy fix. Thanks, Travis > > Takashi

1 0

[alsa-devel] [Fwd: Re: Problem regarding CA0106]
by Travis Place 16 Jun '08

16 Jun '08

1 0

[alsa-devel] alsa and 2.6.25.4 kernel
by linuxconsole＠free.fr 16 Jun '08

16 Jun '08

Hi, I can't play sound or use mixer with alsa on 2.6.25.4 kernel. I tried all alsa-libs (1.16, 1.17rc1, hg) without success. the file /proc/asound/cards is : cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xf0100000 irq 18 strace -e open amixer : ... open("/dev/snd/controlC0", O_RDONLY) = -1 ENODEV (No such device) open("/dev/aloadC0", O_RDONLY) = -1 ENODEV (No such device) ls -l /dev/snd/controlC0 crw-rwxrw- 1 root users 116, 0 Aug 23 2004 /dev/snd/controlC0 what happens ??? Regards. Yann.

2 2

[alsa-devel] Function to update the PCM status in alsa lib?
by anuj aggarwal 16 Jun '08

16 Jun '08

Hi, I am calling snd_pcm_period_elapsed() in my DMA callback routine. After doing that, I check for more data in the 'runtime' struct and pass a new request to xfer data to the codec driver. snd_pcm_period_elapsed() is used to update the pcm status for the next period, by the codec driver. I want to know which function is used by the ALSA library to do the same when it receives a new request from the user to transfer data (i.e. when user calls snd_pcm_writei()). Thanks, Anuj

1 0

[alsa-devel] [PATCH 1/2] ad1848: separate header for registers definition
by Krzysztof Helt 16 Jun '08

16 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Separate AD1848 registers definition into a new header. The new header can be used by compatible CS4231 chip driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch is a start combining ad1848_lib and cs4231_lib functionality. My approach is to make the cs4231_lib module use the ad1848_lib module. This should allow removing most of identical functions from the cs4231_lib and use the functions from the ad1848_lib directly. If this approach is not a good idea, please drop the patch. My current plan is to reuse the ad1848 register constants in the cs423x code and remove duplicated cs4231 constants. This should make the ad1848_lib and cs4231_lib easier to compare for differences. The next step is to use the snd_cs4231 structure instead of the snd_ad1848 structure inside the ad1848_lib. This will make functions from the ad1848_lib taking the same arguments as the functions in the cs4231_lib. The last step would be to export needed functions from the ad1848_lib and use them directly in the cs4231_lib. This should reduce the size of the cs4231_lib while preserving smaller the ad1848_lib for driver which uses only ad1848 functionality. Kind regards, Krzysztof diff -urpN linux-ref/include/sound/ad1848-regs.h linux-mm/include/sound/ad1848-regs.h --- linux-ref/include/sound/ad1848-regs.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-mm/include/sound/ad1848-regs.h 2008-06-15 07:00:08.041351281 +0200 @@ -0,0 +1,104 @@ +#ifndef __SOUND_AD1848_REGS_H +#define __SOUND_AD1848_REGS_H + +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Definitions for AD1847/AD1848/CS4248 chips registers + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* IO ports */ + +#define AD1848P(chip, x) ((chip)->port + c_d_c_AD1848##x) + +#define c_d_c_AD1848REGSEL 0 +#define c_d_c_AD1848REG 1 +#define c_d_c_AD1848STATUS 2 +#define c_d_c_AD1848PIO 3 + +/* codec registers */ + +#define AD1848_LEFT_INPUT 0x00 /* left input control */ +#define AD1848_RIGHT_INPUT 0x01 /* right input control */ +#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ +#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ +#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ +#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ +#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ +#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ +#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ +#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ +#define AD1848_PIN_CTRL 0x0a /* pin control */ +#define AD1848_TEST_INIT 0x0b /* test and initialization */ +#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ +#define AD1848_LOOPBACK 0x0d /* loopback control */ +#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ +#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ + +/* definitions for codec register select port - CODECP( REGSEL ) */ + +#define AD1848_INIT 0x80 /* CODEC is initializing */ +#define AD1848_MCE 0x40 /* mode change enable */ +#define AD1848_TRD 0x20 /* transfer request disable */ + +/* definitions for codec status register - CODECP( STATUS ) */ + +#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ + +/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ + +#define AD1848_ENABLE_MIC_GAIN 0x20 + +#define AD1848_MIXS_LINE1 0x00 +#define AD1848_MIXS_AUX1 0x40 +#define AD1848_MIXS_LINE2 0x80 +#define AD1848_MIXS_ALL 0xc0 + +/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ + +#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ +#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ +#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ +#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ +#define AD1848_STEREO 0x10 /* stereo mode */ +/* bits 3-1 define frequency divisor */ +#define AD1848_XTAL1 0x00 /* 24.576 crystal */ +#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ + +/* definitions for interface control register - AD1848_IFACE_CTRL */ + +#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ +#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ +#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ +#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ +#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ +#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ +#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ + +/* definitions for pin control register - AD1848_PIN_CTRL */ + +#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ +#define AD1848_XCTL1 0x40 /* external control #1 */ +#define AD1848_XCTL0 0x80 /* external control #0 */ + +/* definitions for test and init register - AD1848_TEST_INIT */ + +#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ +#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ + +#endif /* __SOUND_AD1848_REGS_H */ diff -urpN linux-ref/include/sound/ad1848.h linux-mm/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-06-15 06:58:15.293013657 +0200 +++ linux-mm/include/sound/ad1848.h 2008-06-15 00:14:36.728628406 +0200 @@ -25,84 +25,7 @@ #include "pcm.h" #include <linux/interrupt.h> -/* IO ports */ - -#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) - -#define c_d_c_AD1848REGSEL 0 -#define c_d_c_AD1848REG 1 -#define c_d_c_AD1848STATUS 2 -#define c_d_c_AD1848PIO 3 - -/* codec registers */ - -#define AD1848_LEFT_INPUT 0x00 /* left input control */ -#define AD1848_RIGHT_INPUT 0x01 /* right input control */ -#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ -#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ -#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ -#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define AD1848_PIN_CTRL 0x0a /* pin control */ -#define AD1848_TEST_INIT 0x0b /* test and initialization */ -#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ -#define AD1848_LOOPBACK 0x0d /* loopback control */ -#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ -#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define AD1848_INIT 0x80 /* CODEC is initializing */ -#define AD1848_MCE 0x40 /* mode change enable */ -#define AD1848_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ - -#define AD1848_ENABLE_MIC_GAIN 0x20 - -#define AD1848_MIXS_LINE1 0x00 -#define AD1848_MIXS_AUX1 0x40 -#define AD1848_MIXS_LINE2 0x80 -#define AD1848_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ - -#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ -#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ -#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define AD1848_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define AD1848_XTAL1 0x00 /* 24.576 crystal */ -#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - AD1848_IFACE_CTRL */ - -#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ -#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ -#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ -#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ -#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ -#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - AD1848_PIN_CTRL */ - -#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ -#define AD1848_XCTL1 0x40 /* external control #1 */ -#define AD1848_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - AD1848_TEST_INIT */ - -#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ +#include "ad1848-regs.h" /* defines for codec.mode */ ---------------------------------------------------------------------- Tania telefonia internetowa! Sprawdz >>> http://link.interia.pl/f1e2e

3 3

[alsa-devel] Problem regarding CA0106
by Travis Place 16 Jun '08

16 Jun '08

Just been speaking with a guy in #alsa about the problems with his "MSI K8N Diamond MB" which has an onboard "SB Live 24bit" or "Audigy". While looking through ca0106_main.c in GIT, i saw the following section: /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ /* SB0438 * CTRL:CA0106-DAT * ADC: WM8775SEDS * DAC: CS4382-KQZ */ { .serial = 0x10091462, .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , For this guys sound to work, we had to change the serial line to: { .serial = 0x10091102, Seems to be a bit hackish, and maybe some more investigation is required, but, it got sound to work, and the card is recognised. Perhaps there is (for some odd reason) 2 variants of the onboard sound on that motherboard (Perhaps a rev 1.0 and rev 1.1 motherboard difference?) Let me know if any more information is required, or if I can just make and submit a patch (ill add a new section for the new card, not the hackish way). Thanks, Travis Place

2 1

[alsa-devel] [PATCH 1/2] opti93x: add support for Opti93x codec in cs4231-lib
by Krzysztof Helt 15 Jun '08

15 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch adds support for WSS compatible Opti93x codec to the cs4231-lib. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch was sent some time ago but was not accepted due to regression of the driver on the opti 930 chip. The regression is fixed with already accepted patch, so it should work now. The patch was prepared against the 2.6.26-rc5 kernel. Tested on Opti 930, 931 and 933. The previous repost: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-January/005679.ht… diff -urp linux-rc2/include/sound/cs4231-regs.h linux-rc5/include/sound/cs4231-regs.h --- linux-rc2/include/sound/cs4231-regs.h 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/include/sound/cs4231-regs.h 2008-06-08 00:33:55.041903713 +0200 @@ -177,4 +177,12 @@ #define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ #define CS4236_VERSION 0x9c /* chip version and ID */ +/* definitions for extended registers - OPTI93X */ +#define OPTi931_AUX_LEFT_INPUT 0x10 +#define OPTi931_AUX_RIGHT_INPUT 0x11 +#define OPTi93X_MIC_LEFT_INPUT 0x14 +#define OPTi93X_MIC_RIGHT_INPUT 0x15 +#define OPTi93X_OUT_LEFT 0x16 +#define OPTi93X_OUT_RIGHT 0x17 + #endif /* __SOUND_CS4231_REGS_H */ diff -urp linux-rc2/include/sound/cs4231.h linux-rc5/include/sound/cs4231.h --- linux-rc2/include/sound/cs4231.h 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/include/sound/cs4231.h 2008-06-08 00:35:54.808064369 +0200 @@ -58,6 +58,7 @@ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ #define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) @@ -120,6 +121,8 @@ unsigned char snd_cs4236_ext_in(struct s void snd_cs4231_mce_up(struct snd_cs4231 *chip); void snd_cs4231_mce_down(struct snd_cs4231 *chip); +void snd_cs4231_overrange(struct snd_cs4231 *chip); + irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); diff -urp linux-rc2/sound/isa/cs423x/cs4231_lib.c linux-rc5/sound/isa/cs423x/cs4231_lib.c --- linux-rc2/sound/isa/cs423x/cs4231_lib.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/sound/isa/cs423x/cs4231_lib.c 2008-06-08 00:38:32.130878968 +0200 @@ -119,6 +119,42 @@ static unsigned char snd_cs4231_original 0x00, /* 1f/31 - cbrl */ }; +static unsigned char snd_opti93x_original_image[32] = +{ + 0x00, /* 00/00 - l_mixout_outctrl */ + 0x00, /* 01/01 - r_mixout_outctrl */ + 0x88, /* 02/02 - l_cd_inctrl */ + 0x88, /* 03/03 - r_cd_inctrl */ + 0x88, /* 04/04 - l_a1/fm_inctrl */ + 0x88, /* 05/05 - r_a1/fm_inctrl */ + 0x80, /* 06/06 - l_dac_inctrl */ + 0x80, /* 07/07 - r_dac_inctrl */ + 0x00, /* 08/08 - ply_dataform_reg */ + 0x00, /* 09/09 - if_conf */ + 0x00, /* 0a/10 - pin_ctrl */ + 0x00, /* 0b/11 - err_init_reg */ + 0x0a, /* 0c/12 - id_reg */ + 0x00, /* 0d/13 - reserved */ + 0x00, /* 0e/14 - ply_upcount_reg */ + 0x00, /* 0f/15 - ply_lowcount_reg */ + 0x88, /* 10/16 - reserved/l_a1_inctrl */ + 0x88, /* 11/17 - reserved/r_a1_inctrl */ + 0x88, /* 12/18 - l_line_inctrl */ + 0x88, /* 13/19 - r_line_inctrl */ + 0x88, /* 14/20 - l_mic_inctrl */ + 0x88, /* 15/21 - r_mic_inctrl */ + 0x80, /* 16/22 - l_out_outctrl */ + 0x80, /* 17/23 - r_out_outctrl */ + 0x00, /* 18/24 - reserved */ + 0x00, /* 19/25 - reserved */ + 0x00, /* 1a/26 - reserved */ + 0x00, /* 1b/27 - reserved */ + 0x00, /* 1c/28 - cap_dataform_reg */ + 0x00, /* 1d/29 - reserved */ + 0x00, /* 1e/30 - cap_upcount_reg */ + 0x00 /* 1f/31 - cap_lowcount_reg */ +}; + /* * Basic I/O functions */ @@ -895,7 +931,7 @@ static int snd_cs4231_capture_prepare(st return 0; } -static void snd_cs4231_overrange(struct snd_cs4231 *chip) +void snd_cs4231_overrange(struct snd_cs4231 *chip) { unsigned long flags; unsigned char res; @@ -1054,8 +1090,11 @@ static int snd_cs4231_probe(struct snd_c chip->image[CS4231_IFACE_CTRL] = (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | (chip->single_dma ? CS4231_SINGLE_DMA : 0); - chip->image[CS4231_ALT_FEATURE_1] = 0x80; - chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + if (chip->hardware != CS4231_HW_OPTI93X) { + chip->image[CS4231_ALT_FEATURE_1] = 0x80; + chip->image[CS4231_ALT_FEATURE_2] = + chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + } ptr = (unsigned char *) &chip->image; snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1376,6 +1415,7 @@ const char *snd_cs4231_chip_id(struct sn case CS4231_HW_INTERWAVE: return "AMD InterWave"; case CS4231_HW_OPL3SA2: return chip->card->shortname; case CS4231_HW_AD1845: return "AD1845"; + case CS4231_HW_OPTI93X: return "OPTi 93x"; default: return "???"; } } @@ -1401,8 +1441,13 @@ static int snd_cs4231_new(struct snd_car chip->rate_constraint = snd_cs4231_xrate; chip->set_playback_format = snd_cs4231_playback_format; chip->set_capture_format = snd_cs4231_capture_format; - memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); - + if (chip->hardware == CS4231_HW_OPTI93X) + memcpy(&chip->image, &snd_opti93x_original_image, + sizeof(snd_opti93x_original_image)); + else + memcpy(&chip->image, &snd_cs4231_original_image, + sizeof(snd_cs4231_original_image)); + *rchip = chip; return 0; } @@ -1790,6 +1835,48 @@ CS4231_SINGLE("Loopback Capture Switch", CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) }; +static struct snd_kcontrol_new snd_opti93x_controls[] = { +CS4231_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +CS4231_DOUBLE("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +CS4231_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +CS4231_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("CD Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Capture Volume", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +} +}; + int snd_cs4231_mixer(struct snd_cs4231 *chip) { struct snd_card *card; @@ -1802,10 +1889,22 @@ int snd_cs4231_mixer(struct snd_cs4231 * strcpy(card->mixername, chip->pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4231_controls[idx], chip))) < 0) - return err; - } + if (chip->hardware == CS4231_HW_OPTI93X) + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], + chip)); + if (err < 0) + return err; + } + else + for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], + chip)); + if (err < 0) + return err; + } return 0; } @@ -1815,6 +1914,7 @@ EXPORT_SYMBOL(snd_cs4236_ext_out); EXPORT_SYMBOL(snd_cs4236_ext_in); EXPORT_SYMBOL(snd_cs4231_mce_up); EXPORT_SYMBOL(snd_cs4231_mce_down); +EXPORT_SYMBOL(snd_cs4231_overrange); EXPORT_SYMBOL(snd_cs4231_interrupt); EXPORT_SYMBOL(snd_cs4231_chip_id); EXPORT_SYMBOL(snd_cs4231_create); --------------------------------------------------------------- Sprawdz jak zdobyc zdrowy usmiech! Kliknij >> http://link.interia.pl/f1e26

2 4

[alsa-devel] Vendor support in Alsa Drivers
by Emmanuel Kasper 15 Jun '08

15 Jun '08

Hello With a couple of other french folks, we are building a kind of webshop for Linux/FreeBSD compatible hardware. We will reference only devices which work with a free driver, so no nvidia binary thingie or atheros HAL. Based on the information found on the alsa wiki we plan to reference the following sound devices: M-Audio Audiophile 24/96 ( snd-ice1712 driver ) Terratec AUREON 5.1 M-Audio Fast Track usb Hercules Muse Pocket LT Ion TTUSB usb turnable ( these 4 working with snd-usb-driver ) We would be also interested to indicate on each product description, if the vendor of the product, or the chipset maker for that matters had been helpful with developers trying to write a free driver. So for the devices mentionned above, was any of the vendor helpful by providing programming documentation or sample hardware ? If yes we will mention it on the product description with an extra tag like "programmer friendly" Regards -- Emmanuel Kasper System Administrator www.libera.cc Find, compare, buy Linux & FreeBSD compatible hardware

6 7

[alsa-devel] input source selection not working (snd-hda-intel)
by Bruno Schneider 14 Jun '08

14 Jun '08

Hi all, I'm using a computer with a ASUS P5LD2-X motherboard. The audio controller, is a Realtek ALC662. I'm using an updated Debian Lenny (testing) system. No matter what I set as the input source, sound is always recorded from the back microphone jack. Before posting here, I tried to get help from my Debian list and from the alsa-user list [1]. I'm starting to think this is a bug on the snd-hda-intel module. Please help. [1] http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg22667.html -- Bruno Schneider http://www.dcc.ufla.br/~bruno/

1 0

[alsa-devel] [PATCH][ASoC v2] Update Freescale MPC8610HPCD fabric driver to support multiple codecs
by Timur Tabi 14 Jun '08

14 Jun '08

Update the Freescale MPC8610 HPCD fabric driver to create a new platform device for each SSI in the system, instead of registering multiple PCMs under one card. This is necessary for alsamixer to control the individual codecs. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 236 +++++++++++++++++++++--------------------- 1 files changed, 117 insertions(+), 119 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 8813291..2cf7d95 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -40,13 +40,10 @@ struct mpc8610_hpcd_data { struct snd_soc_card soc_card; struct ccsr_guts __iomem *guts; - unsigned int num_configs; - struct snd_soc_pcm_config configs[MAX_SSI]; - struct mpc8610_hpcd_names { - char pcm_name[32]; - char ssi_name[32]; - char codec_name[32]; - } names[MAX_SSI]; + struct snd_soc_pcm_config pcm_config; + char pcm_name[32]; + char ssi_name[32]; + char codec_name[32]; }; /** @@ -91,29 +88,25 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) { struct snd_soc_codec *codec; struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - unsigned int i; int ret = 0; - for (i = 0; i < soc_card_data->num_configs; i++) { - codec = snd_soc_card_get_codec(soc_card, - soc_card_data->names[i].codec_name, - soc_card_data->configs[i].codec_num); + codec = snd_soc_card_get_codec(soc_card, soc_card_data->codec_name, + soc_card_data->pcm_config.codec_num); - if (!codec) { - dev_err(soc_card->dev, "could not find codec\n"); - return -ENODEV; - } + if (!codec) { + dev_err(soc_card->dev, "could not find codec\n"); + return -ENODEV; + } - /* The codec driver should have called - * snd_soc_card_config_codec() by now. - */ - ret = snd_soc_card_init_codec(codec, soc_card); - if (ret < 0) { - dev_err(soc_card->dev, - "could not initialize codec %s-%u\n", - codec->name, codec->num); - continue; - } + /* The codec driver should have called + * snd_soc_card_config_codec() by now. + */ + ret = snd_soc_card_init_codec(codec, soc_card); + if (ret < 0) { + dev_err(soc_card->dev, + "could not initialize codec %s-%u\n", + codec->name, codec->num); + return -ENODEV; } return 0; @@ -316,10 +309,16 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) struct snd_soc_card *soc_card = NULL; struct mpc8610_hpcd_data *soc_card_data; struct device_node *guts_np = NULL; - struct device_node *ssi_np; - unsigned int ssi = 0; + struct device_node *ssi_np = + (struct device_node *)platform_get_resource(pdev, 0, 0)->start; int ret = -ENODEV; + struct device_node *codec_np; + const char *compat; + const char *p; + unsigned int bus; + unsigned int address; + soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); if (!soc_card_data) { dev_err(&pdev->dev, "could not allocate card structure\n"); @@ -353,100 +352,68 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) soc_card->private_data = soc_card_data; soc_card->dev = &pdev->dev; - /* Scan the device tree for SSI nodes. Each one we find that has a - * codec is registered. Remember, we only support MAX_SSI of these. - */ - for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { - struct snd_soc_pcm_config *pcm_config; - struct mpc8610_hpcd_names *name; - struct device_node *codec_np; - const char *compat; - const char *p; - unsigned int bus; - unsigned int address; - int i; - - name = &soc_card_data->names[ssi]; - pcm_config = &soc_card_data->configs[ssi]; - - pcm_config->name = name->pcm_name; - pcm_config->codec = name->codec_name; - pcm_config->codec_dai = name->codec_name; - pcm_config->platform = "fsl-elo"; - pcm_config->cpu_dai = name->ssi_name; - pcm_config->ops = &mpc8610_hpcd_ops; - pcm_config->playback = 1; - pcm_config->capture = 1; - - sprintf(name->pcm_name, "MPC8610HPCD.%u", ssi); - - /* Get the SSI name */ - i = of_get_integer(ssi_np, "cell-index"); - if (i < 0) { - dev_err(soc_card->dev, "no cell-index for %s\n", - ssi_np->full_name); - continue; - } - sprintf(name->ssi_name, "ssi%u", i); - - /* Get the codec name and ID */ - codec_np = find_codec(ssi_np); - if (!codec_np) { - dev_err(soc_card->dev, "missing codec node for %s\n", - ssi_np->full_name); - continue; - } - - /* We assume that the first (or only) string in the compatible - * field is the one that counts. - */ - compat = of_get_property(codec_np, "compatible", NULL); - if (!compat) { - dev_err(soc_card->dev, - "missing compatible property for %s\n", - ssi_np->full_name); - continue; - } - - /* We only care about the part after the comma */ - p = strchr(compat, ','); - strcpy(name->codec_name, p ? p + 1 : compat); - - /* Now determine the I2C bus and address of the codec */ - bus = of_get_integer(of_get_parent(codec_np), "cell-index"); - if (bus < 0) { - dev_err(soc_card->dev, - "cannot determine I2C bus number for %s\n", - codec_np->full_name); - continue; - } + soc_card_data->pcm_config.name = soc_card_data->pcm_name; + soc_card_data->pcm_config.codec = soc_card_data->codec_name; + soc_card_data->pcm_config.codec_dai = soc_card_data->codec_name; + soc_card_data->pcm_config.platform = "fsl-elo"; + soc_card_data->pcm_config.cpu_dai = soc_card_data->ssi_name; + soc_card_data->pcm_config.ops = &mpc8610_hpcd_ops; + soc_card_data->pcm_config.playback = 1; + soc_card_data->pcm_config.capture = 1; + + sprintf(soc_card_data->pcm_name, "MPC8610HPCD.%u", pdev->id); + + /* Get the SSI name */ + sprintf(soc_card_data->ssi_name, "ssi%u", pdev->id); + + /* Get the codec name and ID */ + codec_np = find_codec(ssi_np); + if (!codec_np) { + dev_err(soc_card->dev, "missing codec node for %s\n", + ssi_np->full_name); + goto error; + } - address = of_get_integer(codec_np, "reg"); - if (address < 0) { - dev_err(soc_card->dev, - "cannot determine I2C address for %s\n", - codec_np->full_name); - continue; - } + /* We assume that the first (or only) string in the compatible + * field is the one that counts. + */ + compat = of_get_property(codec_np, "compatible", NULL); + if (!compat) { + dev_err(soc_card->dev, + "missing compatible property for %s\n", + ssi_np->full_name); + goto error; + } - pcm_config->codec_num = bus << 16 | address; + /* We only care about the part after the comma */ + p = strchr(compat, ','); + strcpy(soc_card_data->codec_name, p ? p + 1 : compat); - dev_dbg(soc_card->dev, "registering cpu %s with codec %s-%x\n", - pcm_config->cpu_dai, pcm_config->codec, - pcm_config->codec_num); + /* Now determine the I2C bus and address of the codec */ + bus = of_get_integer(of_get_parent(codec_np), "cell-index"); + if (bus < 0) { + dev_err(soc_card->dev, + "cannot determine I2C bus number for %s\n", + codec_np->full_name); + goto error; + } - if (++ssi == MAX_SSI) - break; + address = of_get_integer(codec_np, "reg"); + if (address < 0) { + dev_err(soc_card->dev, + "cannot determine I2C address for %s\n", + codec_np->full_name); + goto error; } - ret = snd_soc_card_create_pcms(soc_card, soc_card_data->configs, ssi); + soc_card_data->pcm_config.codec_num = bus << 16 | address; + + ret = snd_soc_card_create_pcms(soc_card, &soc_card_data->pcm_config, 1); if (ret) { dev_err(soc_card->dev, "could not create PCMs\n"); return ret; } - soc_card_data->num_configs = ssi; - platform_set_drvdata(pdev, soc_card); ret = snd_soc_card_register(soc_card); @@ -493,7 +460,8 @@ static struct platform_driver mpc8610_hpcd_driver = { .remove = __devexit_p(mpc8610_hpcd_remove), }; -static struct platform_device *pdev; +/* Platform device pointers for each SSI */ +static struct platform_device *pdev[MAX_SSI]; /** * mpc8610_hpcd_init: fabric driver initialization. @@ -502,6 +470,9 @@ static struct platform_device *pdev; */ static int __init mpc8610_hpcd_init(void) { + struct resource res; + struct device_node *ssi_np; + unsigned int i = 0; int ret; pr_info("Freescale MPC8610 HPCD ASoC fabric driver\n"); @@ -512,12 +483,35 @@ static int __init mpc8610_hpcd_init(void) return ret; } - pdev = platform_device_register_simple(mpc8610_hpcd_driver.driver.name, - 0, NULL, 0); - if (!pdev) { - pr_err("mpc8610-hpcd: could not register device\n"); - platform_driver_unregister(&mpc8610_hpcd_driver); - return ret; + memset(pdev, 0, sizeof(pdev)); + + memset(&res, 0, sizeof(res)); + res.name = "ssi"; + + for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { + int id; + + id = of_get_integer(ssi_np, "cell-index"); + if (id < 0) { + pr_err("mpc8610-hpcd: no cell-index for %s\n", + ssi_np->full_name); + continue; + } + if (!find_codec(ssi_np)) + /* No codec node? Skip it */ + continue; + + res.start = (resource_size_t) ssi_np; + + pdev[i] = platform_device_register_simple( + mpc8610_hpcd_driver.driver.name, id, &res, 1); + if (!pdev[i]) { + pr_err("mpc8610-hpcd: could not register %s\n", + ssi_np->full_name); + continue; + } + + i++; } return 0; @@ -530,7 +524,11 @@ static int __init mpc8610_hpcd_init(void) */ static void __exit mpc8610_hpcd_exit(void) { - platform_device_unregister(pdev); + unsigned int i; + + for (i = 0; i < MAX_SSI; i++) + platform_device_unregister(pdev[i]); + platform_driver_unregister(&mpc8610_hpcd_driver); } -- 1.5.5

4 26

[alsa-devel] Specs available for Sweex External 5.1 with digital in/out USB 2.0 / C-Media CM6207
by Erik Slagter 14 Jun '08

14 Jun '08

Hi there, Today I bought this sound device, I was hoping it had the C-Media CM650? USB chip which is already supported by alsa (according to the list). Instead it has a CM6207 inside. The good news is that it supports USB audio "class" and it's more or less usable, some mixer outputs are available (although they're all called "speaker") and with some trickery the various outputs can be used. The other good news is that Sweex supplies the "official" datasheet from C-media which I think is quite interesting and may very well be useful to resolve the last few issues. These are the actual issues (so far, I didn't yet test it exhaustively): - using 2 channel output the sound is garbled; it looks like the "wrong" 2 channel alt setting is used (output of /proc/asound/card?/stream0 on request), 4/6 channel output is OK though. - some high squeeking noise although C-Media claims the IC to be high-end (or maybe their definition of high-end differs from mine ;-)) - all outputs are labelled "speaker 1" - only two outputs have a mute knob, the others cannot be silenced in any way, only turned down - one of the two outputs with the mute knob doesn't appear to do anything (or maybe it's the sp/dif output?) - the other (with mute) controls the front output - the others (without mute) don't seem to do anything at all Is this a typical case of (a) bug(s) in the usb audio class driver or should I think of "vendor extensions" here? Would any of the developers have some interest in the datasheet of the chip and or any data of this thing? Of course I'd like it very much if the device would be fully supported but I can only ask nicely ;-)

4 9

[alsa-devel] [PATCH] hda: Add support for 92HD73xxx codecs
by Matthew Ranostay 13 Jun '08

13 Jun '08

Added support for new family of IDT codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a0..115cee6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -349,7 +349,7 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { static hda_nid_t stac92hd71bxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x14, 0x18, 0x19, 0x1e, 0x0f }; static hda_nid_t stac927x_pin_nids[14] = { @@ -641,15 +641,21 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect ports 0d and 0f to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* connect port 0d to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x02}, /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + /* unmute right and left channels for nodes 0x0a, 0x0d */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} +}; + +static struct hda_verb stac92hd71bxx_enable_portf[] = { + /* connect port 0f to audio mixer */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for nodes 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -1313,20 +1319,20 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x0181302e, 0x01019020, 0x90a000f0, 0x90a000f0, + 0x01452050, 0x01114010, }; -static unsigned int dell_m4_1_pin_configs[13] = { +static unsigned int dell_m4_1_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x23a1902e, 0x40f000f0, 0x90a000f0, 0x40f000f0, + 0x4f0000f0, 0x23014250, }; -static unsigned int dell_m4_2_pin_configs[13] = { +static unsigned int dell_m4_2_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x23a1902e, 0x40f000f0, 0x40f000f0, 0x40f000f0, + 0x044413b0, 0x23014250, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -2633,7 +2639,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) + if (def_conf && (def_conf == AC_JACK_PORT_NONE)) continue; nid = spec->dmic_nids[i]; @@ -3103,13 +3109,14 @@ static int stac92xx_init(struct hda_codec *codec) 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + if (def_conf && def_conf != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); @@ -3614,12 +3621,39 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: + switch (codec->vendor_id) { + case 0x111d76b6: /* 4 Port without Analog Mixer */ + case 0x111d76b7: + case 0x111d76b4: /* 6 Port without Analog Mixer */ + case 0x111d76b5: + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71bxx_core_init; + break; + case 0x111d7603: /* 6 Port with Analog Mixer */ + case 0x111d7608: + /* no output amps */ + spec->num_pwrs = 0; + /* fallthru */ + default: + spec->mixer = stac92hd71bxx_analog_mixer; + spec->init = stac92hd71bxx_analog_core_init; + } + + if (codec->vendor_id != 0x111d7608) + snd_hda_sequence_write(codec, stac92hd71bxx_enable_portf); + else { + /* skip the VSW on port f */ + spec->num_pins--; + stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + } + if (spec->board_config < 0) { snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); @@ -3634,19 +3668,6 @@ again: stac92xx_set_config_regs(codec); } - switch (codec->vendor_id) { - case 0x111d76b6: /* 4 Port without Analog Mixer */ - case 0x111d76b7: - case 0x111d76b4: /* 6 Port without Analog Mixer */ - case 0x111d76b5: - spec->mixer = stac92hd71bxx_mixer; - spec->init = stac92hd71bxx_core_init; - break; - default: - spec->mixer = stac92hd71bxx_analog_mixer; - spec->init = stac92hd71bxx_analog_core_init; - } - spec->aloopback_mask = 0x20; spec->aloopback_shift = 0; @@ -3666,7 +3687,6 @@ again: spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->multiout.num_dacs = 1; @@ -4306,10 +4326,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },

2 3

[alsa-devel] [PATCH 1/2] ASoC: Check for exact register match in wm97xx_reset()
by Mark Brown 13 Jun '08

13 Jun '08

To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f..4739011 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba3301..0db9637 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; } -- 1.5.5.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: Check for exact register match in wm97xx_reset()
by Mark Brown 13 Jun '08

13 Jun '08

To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f..09e2985 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == 0x6174) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != 0x6174) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba3301..131bb5b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == 0x6174) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != 0x6174) return -EIO; return 0; } -- 1.5.5.4

2 3

[alsa-devel] ALSA 1.0.17rc1 release
by Jaroslav Kysela 13 Jun '08

13 Jun '08

Hi all, ALSA 1.0.17rc1 packages (except alsa-plugins & alsa-oss - no changes) are available for download on http://www.alsa-project.org . Only difference between testing release is a change in alsa-driver for better support of RedHat kernels. http://www.alsa-project.org/main/index.php/Changes_v1.0.16_v1.0.17rc1 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

3 5

[alsa-devel] Names for ALSA controls
by Juergen Beisert 13 Jun '08

13 Jun '08

Hi, when I register these mixer controls: Line Playback Switch Line Playback Route Mono Playback Switch Mono Playback Route Phone Playback Switch Phone Playback Route Headphone Playback Switch Headphone Playback Route PCM Playback Switch PCM Playback Volume Line Bypass Capture Switch Line Bypass Capture Volume ADC Capture Route Running "alsamixer" fails with: ALSA lib simple_none.c:1218:(simple_add1) helem (MIXER,'Headphone Playback Switch',0,0,0) appears twice or more It seems there are restrictions what kind of control name I can use for each control, but what restrictions? Are there better names for my output amplifiers? Each of it I can enable/disable (-> Switch) and select a source (-> Route) independently. Regards, Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

2 2

[alsa-devel] [PATCH][ASoC v2] Add checks for codec DAI number during match-up
by Timur Tabi 13 Jun '08

13 Jun '08

Update the Codec DAI matching scheme to take the DAI number into account, and not just the name. This is necessary for systems with multiple codecs that use the Codec ID number to differentiate them. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a88e3a0..cc543e6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -801,6 +801,7 @@ codec_dai: /* yes, then check the codec_dai list */ list_for_each_entry(codec_dai, &codec_dai_list, list) { if (!strcmp(config->codec_dai, codec_dai->name) && + (codec_dai->id == config->codec_num) && try_module_get(codec_dai->dev->driver->owner)) { dbg("ASoC %s %s: Match for %s %s\n", soc_card->name, config->name, @@ -1767,6 +1768,7 @@ struct snd_soc_dai *snd_soc_register_codec_dai( dai->playback = template->playback; dai->capture = template->capture; dai->name = template->name; + dai->id = template->id; dai->ac97_control = template->ac97_control; mutex_lock(&client_mutex); list_add(&dai->list, &codec_dai_list); -- 1.5.5

2 1

[alsa-devel] [PATCH] maestro3: Fix hw volume on HP OmniBook
by Ville Syrjälä 13 Jun '08

13 Jun '08

Make the hw volume buttons work correctly on some HP OmniBook laptops. The original quirk was apparently applied a bit too early and it was also lacking some critial register writes. This improved sequence was discovered by trial and error (like the original sequence). Tested and found working on OB500 and OB6000 laptops. Signed-off-by: Ville Syrjala <syrjala(a)sci.fi> --- sound/pci/maestro3.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) Index: linux-2.6.25.4/sound/pci/maestro3.c =================================================================== --- linux-2.6.25.4.orig/sound/pci/maestro3.c +++ linux-2.6.25.4/sound/pci/maestro3.c @@ -2427,6 +2427,29 @@ snd_m3_amp_enable(struct snd_m3 *chip, i outw(0xffff, io + GPIO_MASK); } +static void +snd_m3_hv_init(struct snd_m3 *chip) +{ + unsigned long io = chip->iobase; + u16 val = GPI_VOL_DOWN | GPI_VOL_UP; + + if (!chip->is_omnibook) + return; + + /* + * Volume buttons on some HP OmniBook laptops + * require some GPIO magic to work correctly. + */ + outw(0xffff, io + GPIO_MASK); + outw(0x0000, io + GPIO_DATA); + + outw(~val, io + GPIO_MASK); + outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION); + outw(val, io + GPIO_MASK); + + outw(0xffff, io + GPIO_MASK); +} + static int snd_m3_chip_init(struct snd_m3 *chip) { @@ -2442,21 +2465,6 @@ snd_m3_chip_init(struct snd_m3 *chip) DISABLE_LEGACY); pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w); - if (chip->is_omnibook) { - /* - * Volume buttons on some HP OmniBook laptops don't work - * correctly. This makes them work for the most part. - * - * Volume up and down buttons on the laptop side work. - * Fn+cursor_up (volme up) works. - * Fn+cursor_down (volume down) doesn't work. - * Fn+F7 (mute) works acts as volume up. - */ - outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK); - outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION); - outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA); - outw(0xffff, io + GPIO_MASK); - } pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n); n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD); n |= chip->hv_config; @@ -2644,6 +2652,8 @@ static int m3_resume(struct pci_dev *pci snd_m3_enable_ints(chip); snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -2783,6 +2793,8 @@ snd_m3_create(struct snd_card *card, str snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, -- Ville Syrjälä syrjala(a)sci.fi http://www.sci.fi/~syrjala/

1 0

[alsa-devel] alsa midi tutorial
by andy baxter 12 Jun '08

12 Jun '08

hello, Can anyone recommend a good, reasonably simple, tutorial on programming for the alsa midi system? What I want to do is feed drum trigger signals from an arduino board (a usb development board that lets you interface electronics to the computer) to a midi sequencer. I had a look at the following page from the alsa api reference: http://www.alsa-project.org/alsa-doc/alsa-lib/seq.html but I'm finding it a bit hard to follow - I'm looking for something a bit simpler with code examples to work through that will give me a way in to alsa midi programming. Does anyone know of anything like this? thanks, andy baxter.

2 2

[alsa-devel] audiowerk8 alsa driver
by matthias.nyffenegger＠bluewin.ch 12 Jun '08

12 Jun '08

Hi, I have written an alsa driver for the audiowerk8 soundcard, source code is available here: http://sourceforge. net/projects/aw8-alsa/. It also supports audiowerk2 but i see that in the meantime there's already a driver available. Anyway, i'd like to ask for integration into the alsa source tree but i don't know exactly how this is done. Can somebody give me some help with that? Regards, Matthias

2 1

[alsa-devel] [PATCH - ASIHPI 1/8] Common init files for HPI
by linux＠audioscience.com 12 Jun '08

12 Jun '08

From: Eliot Blennerhassett <linux(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> diff --git a/pci/asihpi/hpioctl.c b/pci/asihpi/hpioctl.c new file mode 100644 index 0000000..5192db3 --- /dev/null +++ b/pci/asihpi/hpioctl.c @@ -0,0 +1,440 @@ +/******************************************************************************* + + AudioScience HPI driver + Copyright (C) 1997-2003 AudioScience Inc. <support(a)audioscience.com> + + This program is free software; you can redistribute it and/or modify + it under the terms of version 2 of the GNU General Public License as + published by the Free Software Foundation; + + This program is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +Common Linux HPI ioctl and module probe/remove functions +*******************************************************************************/ +#define SOURCEFILE_NAME "hpioctl.c" + +#include "hpi.h" +#include "hpidebug.h" +#include "hpimsgx.h" +#include "hpioctl.h" + +#include <linux/fs.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <asm/uaccess.h> +#include <linux/stringify.h> + +#ifndef HPIMOD_DEFAULT_BUF_SIZE +# define HPIMOD_DEFAULT_BUF_SIZE 192000 +#endif + +#ifdef MODULE_FIRMWARE +MODULE_FIRMWARE("asihpi/dsp5000.bin"); +MODULE_FIRMWARE("asihpi/dsp6200.bin"); +MODULE_FIRMWARE("asihpi/dsp6205.bin"); +MODULE_FIRMWARE("asihpi/dsp6400.bin"); +MODULE_FIRMWARE("asihpi/dsp6600.bin"); +MODULE_FIRMWARE("asihpi/dsp8700.bin"); +MODULE_FIRMWARE("asihpi/dsp8900.bin"); +#endif + +static int bufsize = HPIMOD_DEFAULT_BUF_SIZE; +module_param(bufsize, int, + S_IRUGO +); +/* Allow the debug level to be changed after module load. + E.g. echo 2 > /sys/module/asihpi/parameters/hpiDebugLevel +*/ +module_param(hpiDebugLevel, int, + S_IRUGO | S_IWUSR +); + +/* List of adapters found */ +static struct hpi_adapter adapters[HPI_MAX_ADAPTERS]; + +/* Wrapper function to HPI_Message to enable dumping of the + message and response types. +*/ +static void HPI_MessageF( + struct hpi_message *phm, + struct hpi_response *phr, + struct file *file +) +{ + int nAdapter = phm->wAdapterIndex; + + if ((nAdapter >= HPI_MAX_ADAPTERS || nAdapter < 0) && + (phm->wObject != HPI_OBJ_SUBSYSTEM)) + phr->wError = HPI_ERROR_INVALID_OBJ_INDEX; + else + HPI_MessageEx(phm, phr, file); +} + +/* This is called from hpifunc.c functions, called by ALSA + * (or other kernel process) In this case there is no file descriptor + * available for the message cache code + */ +void HPI_Message( + struct hpi_message *phm, + struct hpi_response *phr +) +{ + HPI_MessageF(phm, phr, HOWNER_KERNEL); +} + +EXPORT_SYMBOL(HPI_Message); +/* export HPI_Message for radio-asihpi */ + +int asihpi_hpi_release( + struct file *file +) +{ + struct hpi_message hm; + struct hpi_response hr; + +/* HPI_DEBUG_LOG(INFO,"hpi_release file %p, pid %d\n", file, current->pid); */ + /* close the subsystem just in case the application forgot to. */ + HPI_InitMessage(&hm, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_CLOSE); + HPI_MessageEx(&hm, &hr, file); + return 0; +} + +#if LINUX_VERSION_CODE > KERNEL_VERSION(2, 6, 11) +long asihpi_hpi_ioctl( + struct file *file, + unsigned int cmd, + unsigned long arg +) +#else +int asihpi_hpi_ioctl( + struct inode *inode, + struct file *file, + unsigned int cmd, + unsigned long arg +) +#endif +{ + struct hpi_ioctl_linux __user *phpi_ioctl_data; + void __user *phm; + void __user *phr; + struct hpi_message hm; + struct hpi_response hr; + u32 uncopied_bytes; + struct hpi_adapter *pa; + + if (cmd != HPI_IOCTL_LINUX) + return -EINVAL; + + phpi_ioctl_data = (struct hpi_ioctl_linux __user *)arg; + + /* Read the message and response pointers from user space. */ + get_user(phm, &phpi_ioctl_data->phm); + get_user(phr, &phpi_ioctl_data->phr); + + /* Now read the message size and data from user space. */ + /* get_user(hm.wSize, (u16 __user *)phm); */ + uncopied_bytes = copy_from_user(&hm, phm, sizeof(hm)); + if (uncopied_bytes) + return -EFAULT; + + pa = &adapters[hm.wAdapterIndex]; + + if ((hm.wAdapterIndex > HPI_MAX_ADAPTERS) || (!pa->type)) { + HPI_InitResponse(&hr, HPI_OBJ_ADAPTER, HPI_ADAPTER_OPEN, + HPI_ERROR_BAD_ADAPTER_NUMBER); + + uncopied_bytes = copy_to_user(phr, &hr, sizeof(hr)); + if (uncopied_bytes) + return -EFAULT; + return 0; + } + + hr.wSize = 0; + /* Response filled either copy from cache, or by HPI_Message() */ + { + /* Dig out any pointers embedded in the message. */ + u16 __user *ptr = NULL; + u32 size = 0; + + /* -1=no data 0=read from user mem, 1=write to user mem */ + int wrflag = -1; + u32 nAdapter = hm.wAdapterIndex; + switch (hm.wFunction) { + case HPI_SUBSYS_CREATE_ADAPTER: + case HPI_SUBSYS_DELETE_ADAPTER: + /* Application must not use these functions! */ + hr.wSize = sizeof(struct hpi_response_header); + hr.wError = HPI_ERROR_INVALID_OPERATION; + hr.wFunction = hm.wFunction; + uncopied_bytes = copy_to_user(phr, &hr, hr.wSize); + if (uncopied_bytes) + return -EFAULT; + return 0; + case HPI_OSTREAM_WRITE: + case HPI_ISTREAM_READ: + /* Yes, sparse, this is correct. */ + ptr = (u16 __user *)hm.u.d.u.Data.pbData; + size = hm.u.d.u.Data.dwDataSize; + + hm.u.d.u.Data.pbData = pa->pBuffer; + /* + if (size > bufsize) { + size = bufsize; + hm.u.d.u.Data.dwDataSize = size; + } + */ + + if (hm.wFunction == HPI_ISTREAM_READ) + /* from card, WRITE to user mem */ + wrflag = 1; + else + wrflag = 0; + break; + + default: + break; + } + + if (mutex_lock_interruptible(&adapters[nAdapter].mutex)) + return -EINTR; + + if (wrflag == 0) { + if (size > bufsize) { + mutex_unlock(&adapters[nAdapter].mutex); + HPI_DEBUG_LOG(ERROR, + "Requested transfer of %d " + "bytes, max buffer size " + "is %d bytes.\n", size, bufsize); + return -EINVAL; + } + + uncopied_bytes = + copy_from_user(pa->pBuffer, ptr, size); + if (uncopied_bytes) + HPI_DEBUG_LOG(WARNING, + "Missed %d of %d " + "bytes from user\n", + uncopied_bytes, size); + } + + HPI_MessageF(&hm, &hr, file); + + if (wrflag == 1) { + if (size > bufsize) { + mutex_unlock(&adapters[nAdapter].mutex); + HPI_DEBUG_LOG(ERROR, + "Requested transfer of %d " + "bytes, max buffer size is " + "%d bytes.\n", size, bufsize); + return -EINVAL; + } + + uncopied_bytes = copy_to_user(ptr, pa->pBuffer, size); + if (uncopied_bytes) + HPI_DEBUG_LOG(WARNING, + "Missed %d of %d " + "bytes to user\n", + uncopied_bytes, size); + } + + mutex_unlock(&adapters[nAdapter].mutex); + } + + /* on return response size must be set */ + if (!hr.wSize) + return -EFAULT; + + /* Copy the response back to user space. */ + uncopied_bytes = copy_to_user(phr, &hr, sizeof(hr)); + if (uncopied_bytes) + return -EFAULT; + return 0; +} + +int __devinit asihpi_adapter_probe( + struct pci_dev *pci_dev, + const struct pci_device_id *pci_id +) +{ + int err, idx; + unsigned int memlen; + struct hpi_message hm; + struct hpi_response hr; + struct hpi_adapter adapter; + struct hpi_pci Pci; + + memset(&adapter, 0, sizeof(adapter)); + + printk(KERN_DEBUG "Probe PCI device (%04x:%04x,%04x:%04x,%04x)\n", + pci_dev->vendor, pci_dev->device, pci_dev->subsystem_vendor, + pci_dev->subsystem_device, pci_dev->devfn); + + HPI_InitMessage(&hm, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_CREATE_ADAPTER); + HPI_InitResponse(&hr, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_CREATE_ADAPTER, + HPI_ERROR_PROCESSING_MESSAGE); + + hm.wAdapterIndex = -1; /* an invalid index */ + + /* fill in HPI_PCI information from kernel provided information */ + adapter.pci = pci_dev; + + for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { + HPI_DEBUG_LOG(DEBUG, "Resource %d %s %llx-%llx\n", + idx, pci_dev->resource[idx].name, + (unsigned long long)pci_resource_start(pci_dev, idx), + (unsigned long long)pci_resource_end(pci_dev, idx)); + + memlen = pci_resource_len(pci_dev, idx); + if (memlen) { + adapter.apRemappedMemBase[idx] = + ioremap(pci_resource_start(pci_dev, idx), + memlen); + if (!adapter.apRemappedMemBase[idx]) { + HPI_DEBUG_LOG(ERROR, + "ioremap failed, aborting\n"); + /* unmap previously mapped pci mem space */ + goto err; + } + } else + adapter.apRemappedMemBase[idx] = NULL; + + Pci.apMemBase[idx] = adapter.apRemappedMemBase[idx]; + } + + Pci.wBusNumber = pci_dev->bus->number; + Pci.wVendorId = (u16)pci_dev->vendor; + Pci.wDeviceId = (u16)pci_dev->device; + Pci.wSubSysVendorId = (u16)(pci_dev->subsystem_vendor & 0xffff); + Pci.wSubSysDeviceId = (u16)(pci_dev->subsystem_device & 0xffff); + Pci.wDeviceNumber = pci_dev->devfn; + Pci.wInterrupt = pci_dev->irq; + Pci.pOsData = pci_dev; + + hm.u.s.Resource.wBusType = HPI_BUS_PCI; + hm.u.s.Resource.r.Pci = &Pci; + + /* call CreateAdapterObject on the relevant hpi module */ + HPI_MessageEx(&hm, &hr, HOWNER_KERNEL); + if (hr.wError) + goto err; + + adapter.pBuffer = vmalloc(bufsize); + if (adapter.pBuffer == NULL) { + HPI_DEBUG_LOG(ERROR, + "HPI could not allocate kernel buffer size %d\n", + bufsize); + goto err; + } + + adapter.index = hr.u.s.wAdapterIndex; + adapter.type = hr.u.s.awAdapterList[adapter.index]; + hm.wAdapterIndex = adapter.index; + + err = HPI_AdapterOpen(NULL, adapter.index); + if (err) + goto err; + + adapter.snd_card_asihpi = NULL; + /* WARNING can't init mutex in 'adapter' + * and then copy it to adapters[] ?!?! + */ + adapters[hr.u.s.wAdapterIndex] = adapter; + mutex_init(&adapters[adapter.index].mutex); + pci_set_drvdata(pci_dev, &adapters[adapter.index]); + + /* printk(KERN_INFO + "Probe found adapter ASI%04X HPI index #%d.\n", + adapter.type, + adapter.index); + */ + return 0; + +err: + for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { + if (adapter.apRemappedMemBase[idx]) { + iounmap(adapter.apRemappedMemBase[idx]); + adapter.apRemappedMemBase[idx] = NULL; + } + } + + if (adapter.pBuffer) + vfree(adapter.pBuffer); + + HPI_DEBUG_LOG(ERROR, "adapter_probe failed\n"); + return -ENODEV; +} + +void __devexit asihpi_adapter_remove( + struct pci_dev *pci_dev +) +{ + int idx; + struct hpi_message hm; + struct hpi_response hr; + struct hpi_adapter *pa; + pa = (struct hpi_adapter *)pci_get_drvdata(pci_dev); + + HPI_InitMessage(&hm, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_DELETE_ADAPTER); + hm.wAdapterIndex = pa->index; + HPI_MessageEx(&hm, &hr, HOWNER_KERNEL); + + /* unmap PCI memory space, mapped during device init. */ + for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { + if (pa->apRemappedMemBase[idx]) { + iounmap(pa->apRemappedMemBase[idx]); + pa->apRemappedMemBase[idx] = NULL; + } + } + + if (pa->pBuffer) + vfree(pa->pBuffer); + + pci_set_drvdata(pci_dev, NULL); + /* + printk(KERN_INFO "PCI device (%04x:%04x,%04x:%04x,%04x)," + " HPI index # %d, removed.\n", + pci_dev->vendor, pci_dev->device, + pci_dev->subsystem_vendor, + pci_dev->subsystem_device, pci_dev->devfn, + pa->index); + */ +} + +void __init asihpi_init( + void +) +{ + struct hpi_message hm; + struct hpi_response hr; + + memset(adapters, 0, sizeof(adapters)); + + /* HPI_DebugLevelSet(debug); now set directly as module param */ + printk(KERN_INFO "ASIHPI driver %d.%02d.%02d\n", + HPI_VER_MAJOR(HPI_VER), + HPI_VER_MINOR(HPI_VER), HPI_VER_RELEASE(HPI_VER)); + + HPI_InitMessage(&hm, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_DRIVER_LOAD); + HPI_MessageEx(&hm, &hr, HOWNER_KERNEL); +} + +void asihpi_exit( + void +) +{ + struct hpi_message hm; + struct hpi_response hr; + + HPI_InitMessage(&hm, HPI_OBJ_SUBSYSTEM, HPI_SUBSYS_DRIVER_UNLOAD); + HPI_MessageEx(&hm, &hr, HOWNER_KERNEL); +} + +/*********************************************************** +*/ diff --git a/pci/asihpi/hpioctl.h b/pci/asihpi/hpioctl.h new file mode 100644 index 0000000..5363472 --- /dev/null +++ b/pci/asihpi/hpioctl.h @@ -0,0 +1,58 @@ +/******************************************************************************* + + AudioScience HPI driver + Copyright (C) 1997-2003 AudioScience Inc. <support(a)audioscience.com> + + This program is free software; you can redistribute it and/or modify + it under the terms of version 2 of the GNU General Public License as + published by the Free Software Foundation; + + This program is distributed in the hope that it will be useful, + but WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + GNU General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +Linux HPI ioctl, and shared module init functions +*******************************************************************************/ + +int __devinit asihpi_adapter_probe( + struct pci_dev *pci_dev, + const struct pci_device_id *pci_id +); +void __devexit asihpi_adapter_remove( + struct pci_dev *pci_dev +); +void __init asihpi_init( + void +); +void __exit asihpi_exit( + void +); + +int asihpi_hpi_release( + struct file *file +); + +long asihpi_hpi_ioctl( + struct file *file, + unsigned int cmd, + unsigned long arg +); + +/* This is called from hpifunc.c functions, called by ALSA + * (or other kernel process) In this case there is no file descriptor + * available for the message cache code + */ +void HPI_Message( + struct hpi_message *phm, + struct hpi_response *phr +); + +#define HOWNER_KERNEL ((void *)-1) + + /*********************************************************** +*/ -- 1.5.4.3

3 10

[alsa-devel] Impact of SW on USB Audio?
by Pavel Hofman 12 Jun '08

12 Jun '08

Hi, Please excuse my non-devel question but the most knowledgeable people read this mailing list. Can user-space software (regular players) directly influence quality of USB adaptive audio output? IOW, how sensitive is adaptive USB to the rhythm of the data infeed to the USB controller? I understand for standard PCI soundcards it is all only about timely delivery of enough data to DMA memory,the card takes care of the rest itself. Do USB controllers shield applications from the rest in the same manner? The adaptive mode is sensitive to jitter of the data stream from the USB controller to the receiver. Can SW players influence the jitter, e.g. by moving the wav files from HDD to RAM disk etc.? I assume a lightly loaded system with no major data delays. Thanks a lot for any insight, I could not google-out any relevant information. Regards, Pavel.

2 2