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[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

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[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

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[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

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[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

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[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);

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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"

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[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);

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[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"

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[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;

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[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--

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