May 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] MFD: twl4030-codec: Update e-mail address
by Peter Ujfalusi 09 May '11

09 May '11

Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- drivers/mfd/twl4030-codec.c | 3 ++- include/linux/mfd/twl4030-codec.h | 1 + 2 files changed, 3 insertions(+), 1 deletions(-) diff --git a/drivers/mfd/twl4030-codec.c b/drivers/mfd/twl4030-codec.c index c02fded..3641a6f 100644 --- a/drivers/mfd/twl4030-codec.c +++ b/drivers/mfd/twl4030-codec.c @@ -2,6 +2,7 @@ * MFD driver for twl4030 codec submodule * * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * Contact: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * Copyright: (C) 2009 Nokia Corporation * @@ -270,6 +271,6 @@ static void __devexit twl4030_codec_exit(void) } module_exit(twl4030_codec_exit); -MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi(a)nokia.com>"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi(a)ti.com>"); MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/twl4030-codec.h b/include/linux/mfd/twl4030-codec.h index 2ec317c..89a6820 100644 --- a/include/linux/mfd/twl4030-codec.h +++ b/include/linux/mfd/twl4030-codec.h @@ -2,6 +2,7 @@ * MFD driver for twl4030 codec submodule * * Author: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> + * Contact: Peter Ujfalusi <peter.ujfalusi(a)ti.com> * * Copyright: (C) 2009 Nokia Corporation * -- 1.7.5.rc3

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[alsa-devel] [PATCH] ASoC: factor out intel_scu_ipc related read/write
by Lu Guanqun 09 May '11

09 May '11

A new 'enum snd_soc_control_type' is added to indicate this operation. Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- include/sound/soc.h | 1 + sound/soc/codecs/sn95031.c | 33 ++++++-------------------------- sound/soc/soc-cache.c | 46 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 53 insertions(+), 27 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6ce3e57..802b9a7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -251,6 +251,7 @@ enum snd_soc_control_type { SND_SOC_CUSTOM = 1, SND_SOC_I2C, SND_SOC_SPI, + SND_SOC_INTEL_SCU, }; enum snd_soc_compress_type { diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 84ffdeb..a11e324 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -29,7 +29,6 @@ #include <linux/delay.h> #include <linux/slab.h> -#include <asm/intel_scu_ipc.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -166,30 +165,6 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) EXPORT_SYMBOL_GPL(sn95031_get_mic_bias); /*end - adc helper functions */ -static inline unsigned int sn95031_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - u8 value = 0; - int ret; - - ret = intel_scu_ipc_ioread8(reg, &value); - if (ret) - pr_err("read of %x failed, err %d\n", reg, ret); - return value; - -} - -static inline int sn95031_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - int ret; - - ret = intel_scu_ipc_iowrite8(reg, value); - if (ret) - pr_err("write of %x failed, err %d\n", reg, ret); - return ret; -} - static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -827,8 +802,14 @@ EXPORT_SYMBOL_GPL(sn95031_jack_detection); /* codec registration */ static int sn95031_codec_probe(struct snd_soc_codec *codec) { + int ret; + pr_debug("codec_probe called\n"); + ret = snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_INTEL_SCU); + if (ret < 0) + return ret; + codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.idle_bias_off = 1; @@ -891,8 +872,6 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, - .read = sn95031_read, - .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a217db2..65986a9 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -20,6 +20,42 @@ #include <trace/events/asoc.h> +#if defined(CONFIG_INTEL_SCU_IPC) +#include <asm/intel_scu_ipc.h> +static unsigned int snd_soc_16_8_intel_scu_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value = 0; + int ret; + + ret = intel_scu_ipc_ioread8(reg, &value); + if (ret < 0) + dev_err(codec->dev, "read of %x failed, err %d\n", reg, ret); + return value; +} + +static int snd_soc_16_8_intel_scu_write(void *control_data, + const char *data, int len) +{ + struct snd_soc_codec *codec = control_data; + unsigned int reg, value; + int ret; + + reg = (data[0] << 8) | data[1]; + value = data[2]; + + ret = intel_scu_ipc_iowrite8(reg, value); + if (ret < 0) { + dev_err(codec->dev, "write of %x failed, err %d\n", reg, ret); + return ret; + } + return len; +} +#else +#define snd_soc_16_8_intel_scu_read NULL +#define snd_soc_16_8_intel_scu_write NULL +#endif + #if defined(CONFIG_SPI_MASTER) static int do_spi_write(void *control_data, const void *msg, int len) @@ -544,6 +580,16 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, struct spi_device, dev); break; + + case SND_SOC_INTEL_SCU: + if (io_types[i].addr_bits != 16 || io_types[i].data_bits != 8) + return -EINVAL; + + codec->hw_read = snd_soc_16_8_intel_scu_read; + codec->hw_write = snd_soc_16_8_intel_scu_write; + codec->control_data = codec; + + break; } return 0;

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[alsa-devel] [PATCH 01/12] ASoC: Create codec DAPM widgets before calling the codecs probe function
by Lars-Peter Clausen 09 May '11

09 May '11

This allows to create DAPM routes depending on those widgets in the codecs probe function. This is helpful when supporting similar codecs with minor differences in the DAPM routing with the same driver. Something similar has already been done for cards in commit a841ebb9(ASoC: Create card DAPM widgets early so they can be used in callbacks) Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 133edeb..a477e21 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1495,6 +1495,10 @@ static int soc_probe_codec(struct snd_soc_card *card, soc_init_codec_debugfs(codec); + if (driver->dapm_widgets) + snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, + driver->num_dapm_widgets); + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { @@ -1508,9 +1512,6 @@ static int soc_probe_codec(struct snd_soc_card *card, if (driver->controls) snd_soc_add_controls(codec, driver->controls, driver->num_controls); - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, - driver->num_dapm_widgets); if (driver->dapm_routes) snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, driver->num_dapm_routes); -- 1.7.2.5

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Re: [alsa-devel] [PATCH] SOC: UDA134x: Remove POWER_OFF_ON_STANDBY define.
by Mark Brown 08 May '11

08 May '11

On Tue, May 03, 2011 at 02:46:32PM +0200, Marek Belisko wrote: > Define POWER_OFF_ON_STANDBY cause trobles when trying to get some > sound from codec because code for bias setup was not compiled > (define wasn't defined). This define was removed in commit: > cc3202f5 but again introduced by commit: f0fba2ad1 which then > completely break codec functionality so remove it again. Applied, thanks.

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Re: [alsa-devel] [PATCH] SOC: UDA134x: Remove POWER_OFF_ON_STANDBY define.
by Liam Girdwood 08 May '11

08 May '11

On Tue, 2011-05-03 at 14:46 +0200, Marek Belisko wrote: > Define POWER_OFF_ON_STANDBY cause trobles when trying to get some > sound from codec because code for bias setup was not compiled > (define wasn't defined). This define was removed in commit: > cc3202f5 but again introduced by commit: f0fba2ad1 which then > completely break codec functionality so remove it again. > > Signed-off-by: Marek Belisko <marek.belisko(a)open-nandra.com> > --- > sound/soc/codecs/uda134x.c | 2 -- > 1 files changed, 0 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c > index 48ffd40..a7b8f30 100644 > --- a/sound/soc/codecs/uda134x.c > +++ b/sound/soc/codecs/uda134x.c > @@ -601,9 +601,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { > .reg_cache_step = 1, > .read = uda134x_read_reg_cache, > .write = uda134x_write, > -#ifdef POWER_OFF_ON_STANDBY > .set_bias_level = uda134x_set_bias_level, > -#endif > }; > > static int __devinit uda134x_codec_probe(struct platform_device *pdev) Acked-by: Liam Girdwood <lrg(a)ti.com>

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[alsa-devel] buffer_size return by snd_pcm_hw_params_get_buffer_size_max
by Raymond Yau 08 May '11

08 May '11

it seem that the value return by snd_pcm_hw_params_get_buffer_size_max() depend on /proc/asound/Intel/pcm0p/sub0/prealloc Is it a bug in alsa-lib or snd-hda-intel driver refer to pulseaudio/src/modules/alsa/alsa-util.c if ((ret = snd_pcm_hw_params_get_buffer_size_max(hwparams, &max_frames)) < 0) pa_log_warn("snd_pcm_hw_params_get_buffer_size_max() failed: %s", pa_alsa_strerror(ret)); else pa_log_debug("Maximum hw buffer size is %lu ms", (long unsigned) (max_frames * PA_MSEC_PER_SEC / _ss.rate)); cat /proc/asound/Intel/pcm0p/sub0/prealloc 64 D: alsa-util.c: Checking for superset analog-stereo (front:%f) D: alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... D: alsa-util.c: Managed to open front:1 I: alsa-util.c: Trying to disable ALSA period wakeups, using timers only D: alsa-util.c: Maximum hw buffer size is 371 ms D: alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples). I: alsa-util.c: ALSA period wakeups disabled I: alsa-source.c: Successfully opened device front:1. I: alsa-source.c: Selected mapping 'Analog Stereo' (analog-stereo). I: alsa-source.c: Successfully enabled mmap() mode. I: alsa-source.c: Successfully enabled timer-based scheduling mode. echo 32768 > /proc/asound/Intel/pcm0p/sub0/prealloc D: alsa-util.c: Checking for superset analog-stereo (front:%f) D: alsa-util.c: Trying front:1 with SND_PCM_NO_AUTO_FORMAT ... D: alsa-util.c: Managed to open front:1 I: alsa-util.c: Trying to disable ALSA period wakeups, using timers only D: alsa-util.c: Maximum hw buffer size is 190217 ms D: alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples). I: alsa-util.c: ALSA period wakeups disabled I: alsa-sink.c: Successfully opened device front:1. I: alsa-sink.c: Selected mapping 'Analog Stereo' (analog-stereo). I: alsa-sink.c: Successfully enabled mmap() mode. I: alsa-sink.c: Successfully enabled timer-based scheduling mode.

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[alsa-devel] [PATCH] ASoC: soc-cache: Allow codec->cache_bypass to be used with snd_soc_hw_bulk_write_raw()
by Dimitris Papastamos 08 May '11

08 May '11

If we specifically want to write a block of data to the hw bypassing the cache, then allow this to happen inside snd_soc_hw_bulk_write_raw(). Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 11 ++++++----- 1 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index a217db2..687beec 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -404,12 +404,13 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r { int ret; - /* Ensure that the base register is volatile. Subsequently - * any other register that is touched by this routine should be - * volatile as well to ensure that we don't get out of sync with - * the cache. + /* To ensure that we don't get out of sync with the cache, check + * whether the base register is volatile or if we've directly asked + * to bypass the cache. Out of bounds registers are considered + * volatile. */ - if (!snd_soc_codec_volatile_register(codec, reg) + if (!codec->cache_bypass + && !snd_soc_codec_volatile_register(codec, reg) && reg < codec->driver->reg_cache_size) return -EINVAL; -- 1.7.5

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[alsa-devel] [PATCH v2 00/10] ASoC Support For Moorestown
by Lu Guanqun 07 May '11

07 May '11

This series of patches provides the ASoC implementation of moorestown audio. This series adds a new codec called upd9976 and a new machine driver called mrst_machine. The platform driver is mostly re-used. With this patchset, it can do: - playback via headphone - playback via internal speaker - capture via DMIC - capture via headset microphone - jack detection v2: - change pr_err to dev_err - change DAPM widgets' name (adding correct suffix) - change to data based init of controls, widgets and route in machine driver - add Master Volume - change original "HP & Spker Mixer Volume" to "PCM Volume" - squash some patches into one, resulting in less number of patches - move jack handling from machine driver to codec driver - in codec probe, set the bias level to standby instead of off Thanks for all the reviewers: - Mark Brown - Dimitris Papastamos - Takashi Iwai - Koul Vinod Any more comments are welcome. :) --- Lu Guanqun (8): ASoC: upd9976: Add Renesas uPD9976 codec driver ASoC: mrst_machine: add moorestown machine driver ASoC: upd9976: add DMIC support ASoC: upd9976: add Analog MIC support ASoC: upd9976: add jack detection function ASoC: mrst_machine: add capture functionality ASoC: mrst_machine: add jack detection support ASoC: mrst_machine: add initial config to machine driver Wang Xingchao (2): ASoC: sst_platform: modify current cpu dai driver to suit the needs for moorestown platform ASoC: upd9976: add capture ability for dai driver sound/soc/codecs/Kconfig | 4 sound/soc/codecs/Makefile | 2 sound/soc/codecs/upd9976.c | 697 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/upd9976.h | 87 +++++ sound/soc/mid-x86/Kconfig | 12 + sound/soc/mid-x86/Makefile | 2 sound/soc/mid-x86/mrst_machine.c | 228 ++++++++++++ sound/soc/mid-x86/sst_platform.c | 13 - 8 files changed, 1041 insertions(+), 4 deletions(-) create mode 100644 sound/soc/codecs/upd9976.c create mode 100644 sound/soc/codecs/upd9976.h create mode 100644 sound/soc/mid-x86/mrst_machine.c -- guanqun

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[alsa-devel] [PATCH] Alsa-plugins: Pulse: Fix snd_pcm_avail returning 0 in some cases
by David Henningsson 07 May '11

07 May '11

Due to a round-off error, snd_pcm_avail could in some cases return 0 even though more data could be written to the stream. This was discovered by Maarten Lankhorst [1], and there is also a test program available that triggers this error [2]. [1] https://tango.0pointer.de/pipermail/pulseaudio-discuss/2011-April/009935.ht… [2] https://tango.0pointer.de/pipermail/pulseaudio-discuss/attachments/20110420… -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] multi channels input and single channel out
by hong zhang 06 May '11

06 May '11

List, I have 8 channels of hdmi input and one channel output on my board. When arecord -f dat hdmi-rx.wav to store 8 channels data, aplay -f dat hdmi-rx.wav releases noise. Ia there any way arecord stores each channel data to 8 different files? Or I have to do beam forming to sort 8 channels out by custom application? Thanks. ---henry

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