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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 19 Jul '08

19 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: Fri, 18 Jul 2008 23:47:45 +0400 <snip> > > Were you in the correct subdirectory when running 'patch' ? > > --Sergei. Yes, I was. Thanks for confirming this. best regards, Landis _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

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[alsa-devel] ASOC, master clock direction
by Jon Smirl 19 Jul '08

19 Jul '08

The MPC5200 has three clock configurations 1) Clock out, normal master mode 2) Clock in, normal slave mode 3) Something Freescale calls cellphone slave but it is more like clock distribution. In cellslave mode one i2s port is in normal slave mode. The clock from this port is used to drive up to three other i2s ports in master mode. I have two choices for set_sysclk: #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 When the clock frequency changes I need to notify the i2s drivers for the dependent channels. Do we need a third choice, SND_SOC_CLOCK_SLAVE? Alternatively I can use another clock id value. -- Jon Smirl jonsmirl(a)gmail.com

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Re: [alsa-devel] [Alsa-user] is this card supported by ALSA?
by Landis McGauhey 19 Jul '08

19 Jul '08

> Date: Wed, 16 Jul 2008 16:52:39 +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; ALSA(a)sc8-sf-spam2.sourceforge.net > Subject: Re: [Alsa-user] is this card supported by ALSA? > > On 16-07-08 16:44, Rene Herman wrote: > > > 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. > > Oh, I so love myself. Try this instead: > > # patch -p1 -r < ens1371-ac97_reset_hack.diff > # patch -p1 -r < ens1371-ac97.diff > > I'll make sure I get some sleep before I post again. > > Rene. OK, this is interesting: # patch -p1 -r < ens1371-ac97_reset_hack.diff= bash: ens1371-ac97_reset_hack.diff: No such file or directory # patch -p1 -r < ens1371-ac97.diff bash: ens1371-ac97.diff: No such file or directory Go figure. Attached are two high-res pix of the card. The smallest chips were just too darned small for me to read the printing; maybe you can magnify the pix and make something of them. Thanks again for all your help. I think I'm just going to be happy with OSS for awhile and see if I can't get it to support Skype. If you decide to keep working on this and you want to test something, you're welcome to contact me. best regards as always, Landis _________________________________________________________________ Keep your kids safer online with Windows Live Family Safety. http://www.windowslive.com/family_safety/overview.html?ocid=TXT_TAGLM_WL_fa…

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[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

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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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