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

[alsa-devel] [PATCH] ASoC: Use late enable handling for direct voice, speaker and headphone
by Mark Brown 11 Jul '11

11 Jul '11

This ensures appropriate clocking for bypass paths to speaker and headphone and direct voice paths on affected revisions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 30 +++++++++++++++++++++--------- sound/soc/codecs/wm_hubs.c | 3 --- 3 files changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 03af7ad..6e85b88 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -876,7 +876,7 @@ SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8d1258c..96ea69b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1146,13 +1146,33 @@ SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), +SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), }; static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { @@ -1283,14 +1303,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), - -SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, - left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), -SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, - right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - SND_SOC_DAPM_POST("Debug log", post_ev), }; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2d6c88b..7e60e22 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -598,9 +598,6 @@ SND_SOC_DAPM_MIXER("IN2L PGA", WM8993_POWER_MANAGEMENT_2, 7, 0, SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0, in2r_pga, ARRAY_SIZE(in2r_pga)), -/* Dummy widgets to represent differential paths */ -SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0, mixinl, ARRAY_SIZE(mixinl)), SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0, -- 1.7.5.4

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[alsa-devel] [PATCH 0/5] treewide: Add missing breaks in switch/case
by Joe Perches 10 Jul '11

10 Jul '11

Found by grep. Uncompiled/untested. Joe Perches (5): gpio-tps65910.c: Add missing breaks in switch/case tps65910-irq: Add missing breaks in switch/case ath5k: Add missing breaks in switch/case fusb300_udc: Add missing breaks in switch/case dbri: Add missing breaks in switch/case drivers/gpio/gpio-tps65910.c | 2 ++ drivers/mfd/tps65910-irq.c | 2 ++ drivers/net/wireless/ath/ath5k/desc.c | 3 +++ drivers/usb/gadget/fusb300_udc.c | 3 +++ sound/sparc/dbri.c | 5 +++++ 5 files changed, 15 insertions(+), 0 deletions(-) -- 1.7.6.131.g99019

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[alsa-devel] max channels of 3 jacks 6 channels motherboard using model=auto realtek codec
by Raymond Yau 10 Jul '11

10 Jul '11

Refer to commit ce764ab22e40a046065c9417ee5f04ad2a816ac1 ALSA: hda - Add channel-mode support to Realtek auto-parser It seem that max channel is six instead of two when the channel mode is 2ch for those 3 jacks motherboard using hda-emu and alsa-info from http://thread.gmane.org/gmane.linux.alsa.devel/86816 > PCM 0 p 44100 2 16 Open PCM ALC887 Analog for play Available PCM parameters: channels: 2/6 formats: S16_LE S32_LE rates: 44100 48000 96000 192000 Prepare PCM, rate=44100, channels=2, format=16 bits because spec->multiout.max_channels = spec->multiout.num_dacs * 2; is assigned after alc_auto_add_multi_channel_mode() increase spec->multiout.num_dacs from 1 to 3 spec->multiout.num_dacs = num_pins + 1;

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[alsa-devel] [PATCH] ASoC: OMAP4: HDMI: Make stricter dependency on OMAP2_DSS
by Ricardo Neri 09 Jul '11

09 Jul '11

When building DSS HDMI driver with audio support, there is dependency on SND_SOC to register the ASoC audio codec. Also, SND_OMAP_SOC_OMAP4_HDMI has dependency on OMAP2_DSS. Then, to avoid build breakages, SND_OMAP_SOC_OMAP4_HDMI, is restricted to be available only when OMAP2_DSS AND SND_SOC have the same build type. Also, dependency on ARCH_OMAP4 is removed, as it is checked indirectly by OMAP4_DSS_HDMI. Reported-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Ricardo Neri <ricardo.neri(a)ti.com> --- sound/soc/omap/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e8ca5be..ffbd83f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -113,7 +113,7 @@ config SND_OMAP_SOC_SDP4430 config SND_OMAP_SOC_OMAP4_HDMI tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" - depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 + depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && (OMAP2_DSS = SND_SOC) select SND_OMAP_SOC_HDMI help Say Y if you want to add support for SoC HDMI audio on Texas Instruments -- 1.7.4.1

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[alsa-devel] ALSA become "Segmentation fault" on current linus tree
by kuninori.morimoto.gx＠renesas.com 09 Jul '11

09 Jul '11

Dear Liam, Mark Currect linus tree become "Segmentation fault" when I use aplay. # aplay xxx.wav Segmentation fault I used "git bisect" and 1st bad commit was below. I'm using sh7372 mackerel board. commit 22de71ba03311cdc1063757c50a1488cb90a1fca Author: Liam Girdwood <lrg(a)ti.com> Date: Thu May 12 16:14:04 2011 +0100 ASoC: core - allow ASoC more flexible machine name Allow ASoC machine drivers to register a driver name and a longname. This allows user space to determine the flavour of machine driver. Signed-off-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Best regards --- Kuninori Morimoto

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit f187700c2d769bd3160b52f99cc4e747d5c5014b: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-05 08:20:19 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: pxa2xx-pcm: remove unused variable 'dai' Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Liam Girdwood (4): ASoC: core - Add API call to register platform kcontrols. ASoC: core - Add platform widget IO ASoC: core - Add convenience register for platform kcontrol and DAPM ASoC: core - Add platform IO tracing Mark Brown (3): ASoC: Manage WM8731 ACTIVE bit as a supply widget ASoC: Manage WM8731 ACTIVE bit as a supply widget Merge branch 'for-3.0' into for-3.1 Stephen Warren (1): ASoC: Tegra: Implement SPDIF CPU DAI include/sound/soc-dapm.h | 2 + include/sound/soc.h | 12 + include/trace/events/asoc.h | 45 ++++ sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +--- sound/soc/pxa/pxa2xx-pcm.c | 1 - sound/soc/soc-core.c | 47 ++++ sound/soc/soc-dapm.c | 13 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 12 files changed, 976 insertions(+), 30 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h

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[alsa-devel] Jack event API - decision needed
by David Henningsson 09 Jul '11

09 Jul '11

Hi, I'm still new in the community in that sense that I'm not sure how decisions are made. But I could use the outcome of such a decision. Background: I'm trying to pull together the missing pieces of jack detection on both kernel / plumbing / application layers, so that when you plug something in (headset, microphone etc), userspace is notified and can take appropriate actions (e g routing decisions). As part of that I wrote a udev patch a few days ago, which nobody commented on in alsa-devel [1], but was somewhat disliked by at least Kay Sievers who maintains udev [2], who preferred we would rewrite our input layer to do something else within ALSA. So before I proceed further I'd like to know if 1) We're continuing the path with /dev/input devices 2a) We'll rewrite these devices to be read-only ALSA mixer controls 2b) We'll rewrite these devices to be something within ALSA, but not exactly mixer controls. For options 2a) and 2b) I guess the existing /dev/input thing should be deprecated and/or removed. So part of decision should maybe be based on information about how widespread the usage of these devices are currently...? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/040916.html [2] e g http://www.spinics.net/lists/hotplug/msg04949.html

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Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Mark Brown 09 Jul '11

09 Jul '11

On Fri, Jul 08, 2011 at 12:36:34PM +0200, Takashi Iwai wrote: > just wondering how about holding a sound-BoF in LinuxCon Europe in > October? > I know ELCE will be held at the very same time, so it may need the > adjustment of schedule. But in general what do you think? I have no current plans to attend these conferences, I'm planning to go to LPC instead. Not sure if I'd be able to arrange it unless I get a KS invite. > An alternative is a workshop in KS2011 day 1, but I think KS suits > better for more broader topics, e.g. the discussions on the media- > notification support or the jack implementation like a thread in the > last month. Yes, I don't think it's good for covering at KS as it's far too detailed. If we're going to do something probably LinuxCon/ELC Europe make most sense.

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit 873bd4cb4fbba6a3e99f750e17ef2ba6ef96e9d3: ASoC: Don't set invalid name string to snd_card->driver field (2011-07-05 14:39:27 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 The WM8731 patch fixes resume when we suspend with an active playback or record. Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Mark Brown (1): ASoC: Manage WM8731 ACTIVE bit as a supply widget sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 2 files changed, 4 insertions(+), 27 deletions(-)

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 09 Jul '11

09 Jul '11

At Thu, 7 Jul 2011 21:35:17 +0900, Toshiharu Okada wrote: > > Date: Thu, 07 Jul 2011 11:00:47 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > >> > >> Therefore, I am not considering about ASoC. > >> Do I need to fit this driver to an ASoC framework for an upstream? > > > >Yes, I guess this would be a better choice in your case. > > Thank you for your comment. > >From now on, We study about ASoC. > If there are a document, a website, a driver, etc. which are referred, > please tell us. Check Documentation/sound/alsa/soc/* files. > Why is choosing ASoS better? > Is it better to correspond to ASoC, when Audio Codec is unfixed? It's not clear what you mean exactly as "unfixed", but in general, the decision rather depends on the usage of the device. If it's designed for a use as a desktop PC component (such as a PCI card or onboard-audio on PC), it'd make sense to implement as a stand-alone, self-contained driver. OTOH, if it's targeted mainly for embedded area, ASoC is the right answer. It's more modularized, and can be more flexibly configured in the end. Takashi

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