May 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v12 0/6] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 08 May '15

08 May '15

Based on 4.1-rc1 As Russell King's DRM ELD helper is not yet in any repository, it is included here with a little patch. v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (6): drm/edid: add function to help find SADs sound/core: add DRM ELD helper sound/core: remove the trace from the DRM ELD helper drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 178 +++++++++++++++++++-- include/drm/drm_edid.h | 19 +++ include/sound/pcm_drm_eld.h | 6 + include/sound/tda998x.h | 23 +++ sound/core/Kconfig | 3 + sound/core/Makefile | 1 + sound/core/pcm_drm_eld.c | 91 +++++++++++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 132 +++++++++++++++ 11 files changed, 501 insertions(+), 11 deletions(-) create mode 100644 include/sound/pcm_drm_eld.h create mode 100644 include/sound/tda998x.h create mode 100644 sound/core/pcm_drm_eld.c create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

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[alsa-devel] [PATCH] ASoC: fsl_ssi: No need call of_device_is_available()
by Fabio Estevam 08 May '15

08 May '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> The comment and the call to of_device_is_available() are not really needed. It is the expected behaviour to probe only the ssi nodes that are enabled in the device tree. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0d48804..c7647e0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1292,13 +1292,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) void __iomem *iomem; char name[64]; - /* SSIs that are not connected on the board should have a - * status = "disabled" - * property in their device tree nodes. - */ - if (!of_device_is_available(np)) - return -ENODEV; - of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) return -EINVAL; -- 1.9.1

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[alsa-devel] [PATCH 1/4] ASoC: brownstone: Automatically disconnect non-connected pins
by Lars-Peter Clausen 08 May '15

08 May '15

Most DAPM input and output pins of the wm8994 are either used in the card's DAPM routing table or are marked as not connected. The only exception is DMIC2DAT input. Given that DMIC1DAT is explicitly mentioned in the DAPM routes lets assume that DMIC2DAT simply has been overlooked and should be marked as not connected. Set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/pxa/brownstone.c | 25 +------------------------ 1 file changed, 1 insertion(+), 24 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 79936e3..2b26318 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -45,29 +45,6 @@ static const struct snd_soc_dapm_route brownstone_audio_map[] = { {"MICBIAS1", NULL, "Main Mic"}, }; -static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* set endpoints to not connected */ - snd_soc_dapm_nc_pin(dapm, "HPOUT2P"); - snd_soc_dapm_nc_pin(dapm, "HPOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); - snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); - snd_soc_dapm_nc_pin(dapm, "IN1LN"); - snd_soc_dapm_nc_pin(dapm, "IN1LP"); - snd_soc_dapm_nc_pin(dapm, "IN1RP"); - snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); - snd_soc_dapm_nc_pin(dapm, "IN2RN"); - snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); - snd_soc_dapm_nc_pin(dapm, "IN2LN"); - - return 0; -} - static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -115,7 +92,6 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &brownstone_ops, - .init = brownstone_wm8994_init, }, }; @@ -132,6 +108,7 @@ static struct snd_soc_card brownstone = { .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), .dapm_routes = brownstone_audio_map, .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), + .fully_routed = true, }; static int brownstone_probe(struct platform_device *pdev) -- 1.8.0

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[alsa-devel] [TINYCOMPRESS][PATCH v3 0/7] Convert build system to autotools
by Qais Yousef 08 May '15

08 May '15

This series converts the build system to use autotools and splits the files into lib and utils. Also patch 5 fixes error handling code. I can't test Android builds but both cross and native compiling worked for me. I left AUTHORS, ChangeLog and NEWS files empty. I used the license header to fill COPYING. Changes in v3: - fix a typo in PATCH 1 where we compile cplay.c instead of crec.c for crecord app. - Add mising SOB lines. Qais Yousef (7): Convert the build system to autotools add .gitignore file makefile.linux: delete as no longer necessary/used Android.mk: Update to use the new location of source files compress.c: fix check for errors from poll(), read() and write() src/lib/utils.c: remove this empty file tinycompress.h: explicitly include required headers .gitignore | 39 ++++ AUTHORS | 0 Android.mk | 6 +- COPYING | 51 +++++ ChangeLog | 0 INSTALL | 365 ++++++++++++++++++++++++++++++++++++ Makefile.am | 3 + NEWS | 0 autogen.sh | 6 + configure.ac | 32 ++++ include/Makefile.am | 6 + include/tinycompress/tinycompress.h | 3 + m4/.place_holder | 0 makefile.linux | 41 ---- src/Makefile.am | 1 + src/lib/Makefile.am | 5 + compress.c => src/lib/compress.c | 22 ++- src/utils/Makefile.am | 11 ++ cplay.c => src/utils/cplay.c | 0 crec.c => src/utils/crec.c | 0 utils.c | 62 ------ 21 files changed, 537 insertions(+), 116 deletions(-) create mode 100644 .gitignore create mode 100644 AUTHORS create mode 100644 COPYING create mode 100644 ChangeLog create mode 100644 INSTALL create mode 100644 Makefile.am create mode 100644 NEWS create mode 100755 autogen.sh create mode 100644 configure.ac create mode 100644 include/Makefile.am create mode 100644 m4/.place_holder delete mode 100644 makefile.linux create mode 100644 src/Makefile.am create mode 100644 src/lib/Makefile.am rename compress.c => src/lib/compress.c (98%) create mode 100644 src/utils/Makefile.am rename cplay.c => src/utils/cplay.c (100%) rename crec.c => src/utils/crec.c (100%) delete mode 100644 utils.c -- 2.1.0

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Re: [alsa-devel] [PATCH 3/3] ASoC: rt5645: add device tree support
by Mark Brown 08 May '15

08 May '15

On Fri, May 08, 2015 at 11:13:39AM +0300, Mika Westerberg wrote: > Looks like some of the context is lost but if I decode the above > right, you want to know if there is anything that corresponds to > of_get_named_gpio() in the generic property API? > You can use fwnode_get_named_gpiod() if you need to extract a GPIO > directly from firmware node. Otherwise you should be using > devm_gpiod_get() and friends. Right, almost certainly devm_gpiod_get() in this case as I've said several times now :(

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[alsa-devel] [PATCH] ASoC: dapm: Add missing mutex unlock
by Charles Keepax 08 May '15

08 May '15

The is a missing mutex unlock on the error path in snd_soc_dapm_get_enum_double. This was introduced in commit 561ed680b764 ("ASoC: dapm: Add support for autodisable mux controls"). This patch adds the missing unlock. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ea3348e..6374852 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3075,8 +3075,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg != SND_SOC_NOPM && dapm_kcontrol_is_powered(kcontrol)) { int ret = soc_dapm_read(dapm, e->reg, &reg_val); - if (ret) + if (ret) { + mutex_unlock(&card->dapm_mutex); return ret; + } } else { reg_val = dapm_kcontrol_get_value(kcontrol); } -- 1.7.2.5

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Re: [alsa-devel] [PATCH v3 3/8] mfd: arizona: Add support for WM8998 and WM1814
by Richard Fitzgerald 08 May '15

08 May '15

On Thu, May 07, 2015 at 09:52:38AM +0100, Lee Jones wrote: > On Fri, 01 May 2015, Richard Fitzgerald wrote: > > > Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> > > --- > > drivers/mfd/Kconfig | 6 + > > drivers/mfd/Makefile | 3 + > > drivers/mfd/arizona-core.c | 105 ++- > > drivers/mfd/arizona-i2c.c | 8 + > > drivers/mfd/arizona-irq.c | 9 + > > drivers/mfd/arizona.h | 5 + > > drivers/mfd/wm8998-tables.c | 1592 +++++++++++++++++++++++++++++++++ > > include/linux/mfd/arizona/core.h | 3 + > > include/linux/mfd/arizona/pdata.h | 2 + > > include/linux/mfd/arizona/registers.h | 220 +++++ > > 10 files changed, 1945 insertions(+), 8 deletions(-) > > create mode 100644 drivers/mfd/wm8998-tables.c > > How orthogonal is this patch? Can I just take it? > > For my own reference: > Acked-by: Lee Jones <lee.jones(a)linaro.org> This patches code added by patch 1 "mfd: arizona: Split INx_MODE into two fields" so you'll need to take that one as well. But if you are ok with that change too then yes it's ok to take both of them into mfd.

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Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 08 May '15

08 May '15

> > > My laptop has built-in front left, front right, rear left, rear right and > > > subwoofer and something is strange. > > > When I set channel output to 5.1 or 4.1 and external speakers are > > > disconnected, then front left and front right work as expected but > > > > built-in > > > > > subwoofer is heard only if I test rear left channel (wrong channel > > > > mapping?) > > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pc > > i/hda/hda_codec.c?id=cd4035e814b04c009ece9939396d06370c53ed44 > > > > Seem only two pairs of stereo speakers and internal subwoofer at left > > channel before this patch > > > > ALSA: hda - Enable surround speakers (when line out is also present) > > On my laptop case "Virtual SurroundSound" is written . When I test front left > and rear left, I hear distinct sound, so it would mean it's from two different > sources of sound. The driver use more than two channels only after David 's patch But all internal speakers of the notebook are at front of the user, The 3D sound effect is quite different from your external 5.1 speakers Do you mean your 5930g need champ {FL FR RL RR LFE }? This can be easily implemented by using "swap center/lfe" switch for those codec support R/L swap http://git.alsa-project.org/?p=alsa-lib.git;a=blob_plain;f=test/chmap.c;hb=… http://git.alsa-project.org/?p=alsa-lib.git;a=tree;f=src/conf/pcm;hb=HEAD If codec does not support R/L swap, your 4.1 swap chmap need 5.0 routing Do you expect pulseaudio show 4.1 swap(5.0) profile for internal speakers and 4.1 and 5.1 profiles for external speakers ?

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[alsa-devel] Focusrite 6i6 invalid PCM device "front"
by Ramon Marco Navarro 08 May '15

08 May '15

# Background I can't seem to use any of the PCM device list defined automatically for my [Focusrite 6i6 device][1]. I tested this using `speaker-test`. # Tests: ## `speaker-test` output comparing between using [`sysdefault`][2] and [`front:USB`][3]. The `sysdefault` device emits sound for all of the formats, except for FLOAT_LE 44100Hz and 48000Hz. While front:USB doesn't emit anything at all. [1]: http://www.alsa-project.org/db/?f=b1c10accc89cd34dfdeb0d61b087a3cf413ea1a0 [2]: https://gist.githubusercontent.com/ramonmaruko/deb1918efa27eb860aa1/raw/f7a… [3]: https://gist.githubusercontent.com/ramonmaruko/1377846e39a7a6acd72f/raw/4d6…

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Re: [alsa-devel] [PATCH v2 1/4] ASoC: Add gtm601 codec driver
by Mark Brown 07 May '15

07 May '15

On Wed, May 06, 2015 at 10:38:47PM +0200, Marek Belisko wrote: > This driver add PCM interface to a GTM601 UMTS modem chip. > There is no configuration interface. Applied, thanks. Please remember to CC maintainers on patches, you've missed Liam for some reason.

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