March 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Question about sound channel
by Kuninori Morimoto 21 Mar '12

21 Mar '12

Hi ALSA ML I have question about sound channel. If input sound was MONO (= 1ch), but codec supports I2S (= 2ch) only, How to output sound ? What should the driver do in this case ? Best regards --- Kuninori Morimoto

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[alsa-devel] Last minute ASoC updates for 3.4
by Mark Brown 21 Mar '12

21 Mar '12

The following changes since commit 5ec65ee589fdaca7298b6303fd74ad6c121a8f38: ASoC: mx27vis-aic32x4: Convert it to platform driver (2012-03-13 23:09:59 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.4 for you to fetch changes up to 22f8d055350066b4a87de4adea8c5213cac54534: ASoC: wm8994: Provide VMID mode control and fix default sequence (2012-03-19 19:08:17 +0000) ---------------------------------------------------------------- Last minute ASoC updates for 3.4 There's a couple of small features here that were added late on but have been in -next in my tree and some bug fixes. The wm_hubs stuff is actually bug fixes - the stuff that's currently in 3.4 is a half way house between the two solutions that the latest change allows the machine to select between. ---------------------------------------------------------------- Daniel Mack (1): ASoC: pxa-ssp: atomically set stream active masks Mark Brown (5): ASoC: wm8996: Add 44.1kHz support ASoC: core: Fix obscure leak of runtime array ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF ASoC: wm8994: Add missing break in resume ASoC: wm8994: Provide VMID mode control and fix default sequence Sangsu Park (1): ASoC: Samsung: Added to support mono recording Timur Tabi (1): ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master sound/soc/codecs/wm8994.c | 187 ++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8994.h | 8 ++ sound/soc/codecs/wm8996.c | 19 +++-- sound/soc/codecs/wm_hubs.c | 16 +--- sound/soc/fsl/p1022_ds.c | 24 ++++-- sound/soc/pxa/pxa-ssp.c | 61 +++++++++------ sound/soc/samsung/i2s.c | 13 +++- sound/soc/soc-core.c | 8 +- 8 files changed, 239 insertions(+), 97 deletions(-)

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Re: [alsa-devel] [PATCH v6] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 21 Mar '12

21 Mar '12

On Tue, Mar 06, 2012 at 07:49:16PM +0900, Tomoya MORINAGA wrote: > On Tue, Mar 06, 2012　19:00 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > This is not what the above prototype says, the above prototype has only > > one register in it. You need to make this *much* clearer for the > > registers it's affecting. > Sorry, I couldn't understand your saying / concern. > Do you mean I shouldn't use ml26124_update_bits() but snd_soc_update_bits() ? Ideally you should be using the framework features. If there is some reason why you can't you need to make it much clearer what the reason is so people reading the code can understand.

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Re: [alsa-devel] [PATCH v5] sound/soc/lapis: add machine driver for ML7213 Carrier Board
by Mark Brown 21 Mar '12

21 Mar '12

On Tue, Mar 06, 2012 at 02:49:24PM +0900, Tomoya MORINAGA wrote: > 2012年3月2日22:05 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > > You're also missing a blank line between this and the previous stanza. > Sorry, I can't understand your saying. Your file should look like other similar files in the kernel. > > Use snd_soc_register_card() and a real device. > OK. > As a matter of interest, why do you recommend me to use > snd_soc_register_card() ? It's been the standard since 2.6.38.

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[alsa-devel] [PATCH 1/3] ASoC: tegra: fix comment indentation in ALC5632 machine
by Stephen Warren 21 Mar '12

21 Mar '12

Fix comment indentation to clear checkpatch errors in a later patch. Signed-off-by: Stephen Warren <swarren(a)wwwdotorg.org> --- This series is for 3.5, and based on a merge of next-20120320 plus ASoC's for-3.5 branch, plus a couple minor patches to get recent next- compiling for Tegra, that don't affect the context of this series. Applying all the patches to the ASoC tree should be fine; I don't anticpate any conflicts with the likely Tegra tree for 3.5. sound/soc/tegra/tegra_alc5632.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e45ccd8..03ae095 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,16 +1,16 @@ /* -* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver -* -* Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> -* -* Authors: Leon Romanovsky <leon(a)leon.nu> -* Andrey Danin <danindrey(a)mail.ru> -* Marc Dietrich <marvin24(a)gmx.de> -* -* 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. -*/ + * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver + * + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * + * Authors: Leon Romanovsky <leon(a)leon.nu> + * Andrey Danin <danindrey(a)mail.ru> + * Marc Dietrich <marvin24(a)gmx.de> + * + * 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 <asm/mach-types.h> -- 1.7.0.4

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[alsa-devel] HDA intel, wired headset w/mic, jack mic detect
by Kevin Hilman 20 Mar '12

20 Mar '12

Hello, I'm trying to get a wired headset with a mic working on MacBookAir running v3.0 (3.0.0-13-generic from Ubuntu to be precise.) It has Intel HDA audio[1] and everything works fine, including the built-in mic. The only think I can't seem to get working is a headset w/mic. I know the HW for dynamic detect works because the mic is detected dynamically under OSX. Is there some setting I need to tweak in order to enable the automatic mic detection? Thanks for any pointers, Kevin [1] $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xa0600000 irq 46

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[alsa-devel] [PATCH 1/3] ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
by Mark Brown 19 Mar '12

19 Mar '12

It can just be enabled all the time with no impact. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 16 +++++----------- 1 files changed, 5 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c08d1c2..f13f288 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -643,8 +643,6 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0), - SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0, @@ -869,11 +867,9 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { - { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, @@ -890,11 +886,9 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { - { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, @@ -996,6 +990,11 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, WM8993_LINEOUT2_MODE, WM8993_LINEOUT2_MODE); + if (!lineout1_diff && !lineout2_diff) + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, + WM8993_LINEOUT_VMID_BUF_ENA); + if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); @@ -1068,11 +1067,6 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, WM8993_LINEOUT2P_ENA, val); - if (!hubs->lineout1n_ena && !hubs->lineout1p_ena && - !hubs->lineout2n_ena && !hubs->lineout2p_ena) - snd_soc_update_bits(codec, WM8993_ANTIPOP1, - WM8993_LINEOUT_VMID_BUF_ENA, 0); - /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, WM8993_INPUTS_CLAMP, 0); -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: cs4270: Check that we can enable regulators on resume
by Mark Brown 19 Mar '12

19 Mar '12

It's possible that the regulator enable will fail and if it does we may as well just give up with trying to bring the rest of the device up and report the original error. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 1d672f5..15df950 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -602,8 +602,10 @@ static int cs4270_soc_resume(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int reg; - regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), - cs4270->supplies); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies), + cs4270->supplies); + if (ret != 0) + return ret; /* In case the device was put to hard reset during sleep, we need to * wait 500ns here before any I2C communication. */ -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: dapm: Remove bodges for no-widget CODECs
by Mark Brown 19 Mar '12

19 Mar '12

Now that we're creating widgets for all DAIs there should be no more need for the bodges we've been carrying for non-DAPM CODEC drivers so remove them. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- I don't actually have any no widget CODECs so can't really test - Timur, would it be possible for you to give it a spin? sound/soc/soc-dapm.c | 36 ++++-------------------------------- 1 files changed, 4 insertions(+), 32 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8ce1e9c..6e5869d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1441,12 +1441,10 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { - if (d->n_widgets || d->codec == NULL) { - if (d->idle_bias_off) - d->target_bias_level = SND_SOC_BIAS_OFF; - else - d->target_bias_level = SND_SOC_BIAS_STANDBY; - } + if (d->idle_bias_off) + d->target_bias_level = SND_SOC_BIAS_OFF; + else + d->target_bias_level = SND_SOC_BIAS_STANDBY; } dapm_reset(card); @@ -1491,32 +1489,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } - /* If there are no DAPM widgets then try to figure out power from the - * event type. - */ - if (!dapm->n_widgets) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - case SND_SOC_DAPM_STREAM_RESUME: - dapm->target_bias_level = SND_SOC_BIAS_ON; - break; - case SND_SOC_DAPM_STREAM_STOP: - if (dapm->codec && dapm->codec->active) - dapm->target_bias_level = SND_SOC_BIAS_ON; - else - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - dapm->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - case SND_SOC_DAPM_STREAM_NOP: - dapm->target_bias_level = dapm->bias_level; - break; - default: - break; - } - } - /* Force all contexts in the card to the same bias state if * they're not ground referenced. */ -- 1.7.9.1

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[alsa-devel] ASoC:Question rate constraint between the dais
by Ramesh Babu 19 Mar '12

19 Mar '12

Hi I would like to understand how to handle the different sampling rates between dais. I need to make sure that all simultaneous playback and capture (across the all dai_links) needs to be of same frequency family. If the sampling frequency of a new request is different from ongoing playback/capture, then it needs to be rejected. How do I handle it in machine driver? Thanks Ramesh

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