August 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] ASoC updates for v3.11
by Mark Brown 15 Aug '13

15 Aug '13

The following changes since commit d4e4ab86bcba5a72779c43dc1459f71fea3d89c8: Linux 3.11-rc5 (2013-08-11 18:04:20 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11-rc5 for you to fetch changes up to 44ffb69ec617f64cd83182e8b64d93b84758d636: Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus (2013-08-15 11:37:54 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.11 A few driver specific fixes here plus one core fix for a memory corruption issue in DAPM initialisation which could lead to crashes. ---------------------------------------------------------------- Brian Austin (2): ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume ASoC: cs42l52: Add new TLV for Beep Volume Lars-Peter Clausen (1): ASoC: dapm: Fix empty list check in dapm_new_mux() Lothar Waßmann (2): ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control Mark Brown (4): Merge remote-tracking branch 'asoc/fix/cs42l52' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/sgtl5000' into asoc-linus Merge remote-tracking branch 'asoc/fix/tegra' into asoc-linus Stephen Warren (1): ASoC: tegra: fix Tegra30 I2S capture parameter setup sound/soc/codecs/cs42l52.c | 5 ++++- sound/soc/codecs/sgtl5000.c | 18 ++++++++++++++---- sound/soc/soc-dapm.c | 7 ++++--- sound/soc/tegra/tegra30_i2s.c | 2 +- 4 files changed, 23 insertions(+), 9 deletions(-)

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[alsa-devel] [PATCH] ALSA: core: allow SND_DMAENGINE_PCM use from modules
by Daniel Mack 15 Aug '13

15 Aug '13

When users of SND_DMAENGINE_PCM are built as module, the config symbol SND_DMAENGINE_PCM must be tristate, otherwise the linker will fail. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sound/core/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 94ce1c4..313f22e 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -7,7 +7,7 @@ config SND_PCM select SND_TIMER config SND_DMAENGINE_PCM - bool + tristate config SND_HWDEP tristate -- 1.8.3.1

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Re: [alsa-devel] [PATCH 0/2] add sysfs for acpi interfaces of thinkpad hardware mute
by David Henningsson 15 Aug '13

15 Aug '13

(Added alsa-devel to CC) On 08/14/2013 09:16 AM, Alex Hung wrote: > Added in David and Tim who also has some ideas to share. > > On Wed, Aug 14, 2013 at 3:00 PM, Matthew Garrett > <matthew.garrett(a)nebula.com> wrote: >> On Wed, 2013-08-14 at 14:41 +0800, Alex Hung wrote: >>> Hi Matthew, >>> >>> The problem is that thinkpad-acpi does not aware of mute is changed by >>> either hotkey presses or in desktop. >> >> Ok, how about this. Register the LED with the kernel LED subsystem. Add >> a new LED trigger for volume-mute. Have the HDA driver call that trigger >> whenever the mute state changes. >> >> -- >> Matthew Garrett <matthew.garrett(a)nebula.com> > > > So, to take a wider grip on this issue. First, the hardware mute. Because this is directly connected to the internal speaker (and headphones?), it needs to turn into a ALSA kcontrol on the sound card that controls speaker and/or headphones. And it needs to named accordingly, e g "Speaker Playback Switch" if it controls the speaker only. Exactly how this is going to work out given the arbitrary module load order of snd-hda-intel and thinkpad-acpi, I'm not sure, but the way of creating another sound card (which I've seen on some thinkpads) just isn't working out for userspace. Second, about the mute LEDs and mic-mute LEDs, we currently have several interfaces to choose from: a) HP LEDs currently uses a ALSA kcontrol to control them. For Mute LED there is also a limited automatic control of this LED from kernel space (by default), so it follows the status of "Master Playback Switch" IIRC. b) thinkpad-acpi currently uses custom sysfs nodes to control the LEDs. c) and we also have the /sys/class/leds interface. There might be even more solutions. It's important we get a consistent interfaces towards userspace for the LEDs. And we should also make sure we don't have a permissions problem: if we want the mute/mic-mute LED to follow the status of the currently selected sound card in PulseAudio, which IMO should be our goal, writes cannot be root-only. To have the HDA driver call into the LED subsystem is better than nothing, but in the end we would want the LEDs to follow whatever sound input/output the user currently uses, whether that is bluetooth, USB, HDMI, or internal audio. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH 0/5] ASoC: pxa dmaengine preparation patches
by Daniel Mack 15 Aug '13

15 Aug '13

Mark, This series is an excerpt of my pxa-dma rework that can be merged independently in the next merge window. The rest of the series will take some time to get sorted out, and given that -rc5 is released already, I don't think the entire series will make it in time. Thanks, Daniel Daniel Mack (5): ALSA: move dmaengine implementation from ASoC to ALSA core ASoC: pxa: pxa-ssp: add DT bindings ASoC: pxa: use snd_dmaengine_dai_dma_data ASoC: pxa: pxa-ssp: set dma filter data from startup hook ASoC: pxa: add DT bindings for pxa2xx-pcm .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 28 ++ .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 + include/sound/pxa2xx-lib.h | 7 - sound/arm/pxa2xx-ac97.c | 26 +- sound/arm/pxa2xx-pcm-lib.c | 52 ++- sound/arm/pxa2xx-pcm.c | 5 +- sound/arm/pxa2xx-pcm.h | 6 +- sound/core/Kconfig | 3 + sound/core/Makefile | 3 + sound/core/pcm_dmaengine.c | 367 +++++++++++++++++++++ sound/soc/Kconfig | 5 +- sound/soc/Makefile | 4 - sound/soc/omap/Kconfig | 2 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/mmp-pcm.c | 8 +- sound/soc/pxa/mmp-sspa.c | 12 +- sound/soc/pxa/pxa-ssp.c | 76 +++-- sound/soc/pxa/pxa2xx-ac97.c | 67 ++-- sound/soc/pxa/pxa2xx-i2s.c | 28 +- sound/soc/pxa/pxa2xx-pcm.c | 21 +- sound/soc/soc-dmaengine-pcm.c | 367 --------------------- sound/soc/spear/Kconfig | 2 +- 22 files changed, 607 insertions(+), 499 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa-ssp.txt create mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 sound/core/pcm_dmaengine.c delete mode 100644 sound/soc/soc-dmaengine-pcm.c -- 1.8.3.1

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[alsa-devel] [PATCH 1/2] ASoC: pcm3008: Add DAPM support
by Mark Brown 15 Aug '13

15 Aug '13

From: Mark Brown <broonie(a)linaro.org> Make it possible to connect external devices to the CODEC and ensure continued operation with non-DAPM support removed from the core. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm3008.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8b9b378..19f5028 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,6 +28,22 @@ #include "pcm3008.h" +static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINL"), +SND_SOC_DAPM_INPUT("VINR"), + +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { + { "PCM3008 Capture", NULL, "VINL" }, + { "PCM3008 Capture", NULL, "VINR" }, + + { "VOUTL", NULL, "PCM3008 Playback" }, + { "VOUTR", NULL, "PCM3008 Playback" }, +}; + #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -77,6 +93,10 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { .suspend = pcm3008_soc_suspend, .resume = pcm3008_soc_resume, + .dapm_widgets = pcm3008_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), + .dapm_routes = pcm3008_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes), }; static int pcm3008_codec_probe(struct platform_device *pdev) -- 1.8.4.rc2

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[alsa-devel] [PATCH] ASoC: tegra: support a Mic Jack in the Tegra+RT5640 machine driver
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> Add a Mic Jack widget to the Tegra+RT5640 machine driver, and document this in the DT binding. This enables the DT to include the Mic Jack in the audio routing table, and hence enables capture of audio, in addition to the previously-working playback. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt | 1 + sound/soc/tegra/tegra_rt5640.c | 1 + 2 files changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt index cba4f88..dc62249 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt @@ -16,6 +16,7 @@ Required properties: * Headphones * Speakers + * Mic Jack - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index 08794f9..4511c5a 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -99,6 +99,7 @@ static struct snd_soc_jack_gpio tegra_rt5640_hp_jack_gpio = { static const struct snd_soc_dapm_widget tegra_rt5640_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), }; static const struct snd_kcontrol_new tegra_rt5640_controls[] = { -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC: tegra: fix Tegra30 I2S capture parameter setup
by Stephen Warren 15 Aug '13

15 Aug '13

From: Stephen Warren <swarren(a)nvidia.com> The Tegra30 I2S driver was writing the AHUB interface parameters to the playback path register rather than the capture path register. This caused the capture parameters not to be configured at all, so if capturing using non-HW-default parameters (e.g. 16-bit stereo rather than 8-bit mono) the audio would be corrupted. With this fixed, audio capture from an analog microphone works correctly on the Cardhu board. Cc: stable(a)vger.kernel.org Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d04146c..47565fd04 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); -- 1.8.1.5

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[alsa-devel] [PATCH v4 resent 0/2] Add freescale S/PDIF CPU DAI and machine drivers
by Nicolin Chen 15 Aug '13

15 Aug '13

Changelog: v3->v4: * Use regmap for CPU DAI driver. * Use individual clock source for 32KHz, 44KHz, 48KHz playback. * Determine clock source configuration from 'clocks' entry. * Added imx53 to compatible list, merged imx6q and imx6dl in the list. * Improve the algorism of reverse_bits(). * Dropped the unneeded clk_put(). v2->v3: * Removed a wrong tag from the commit of patch-1. v1->v2: * Dropped one applied patch for spdif dummy codec drivers. * Use generic DMA DT binding. * Let spdif controller driver calculate the clock div. * Added one optional clock source for spdif tx. * Reivsed documentation accordingly. Nicolin Chen (2): ASoC: fsl: Add S/PDIF CPU DAI driver ASoC: fsl: Add S/PDIF machine driver .../devicetree/bindings/sound/fsl,spdif.txt | 100 ++ .../devicetree/bindings/sound/imx-audio-spdif.txt | 29 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 4 + sound/soc/fsl/fsl_spdif.c | 1350 ++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 223 ++++ sound/soc/fsl/imx-spdif.c | 134 ++ 7 files changed, 1854 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-spdif.txt create mode 100644 sound/soc/fsl/fsl_spdif.c create mode 100644 sound/soc/fsl/fsl_spdif.h create mode 100644 sound/soc/fsl/imx-spdif.c

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[alsa-devel] [PATCH 1/16 v2] sound/soc/pxa/mioa701_wm9713.c: Avoid using ARRAY_AND_SIZE(e) as a function argument
by Julia Lawall 14 Aug '13

14 Aug '13

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Replace ARRAY_AND_SIZE(e) in function argument position to avoid hiding the arity of the called function. At the suggestion of Robert Jarzmik, this additionally removes the definition of ARRAY_AND_SIZE, which is not used elsewhere in the file. The semantic match that makes this change is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e,f; @@ f(..., - ARRAY_AND_SIZE(e) + e,ARRAY_SIZE(e) ,...) // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- Not compiled. v2 removes the definition of ARRAY_AND_SIZE. diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 97b711e..bbea778 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -56,8 +56,6 @@ #include "pxa2xx-ac97.h" #include "../codecs/wm9713.h" -#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) - #define AC97_GPIO_PULL 0x58 /* Use GPIO8 for rear speaker amplifier */ @@ -133,10 +131,11 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) unsigned short reg; /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, + ARRAY_SIZE(mioa701_dapm_widgets)); /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG);

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[alsa-devel] [PATCH 2/2] ASoC: wm8960: Fix shared LRCLK support
by Ma Haijun 14 Aug '13

14 Aug '13

Shared LRCLK initialization does not survive wm8960_reset, so place it after the reset. Signed-off-by: Ma Haijun <mahaijuns(a)gmail.com> --- sound/soc/codecs/wm8960.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 368d39f..b606dd3 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -990,6 +990,9 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pdata && pdata->shared_lrclk) + snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + /* Latch the update bits */ snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); @@ -1041,7 +1044,6 @@ static const struct regmap_config wm8960_regmap = { static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct wm8960_data *pdata = dev_get_platdata(&i2c->dev); struct wm8960_priv *wm8960; int ret; @@ -1054,16 +1056,6 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } - i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.8.1.2

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