April 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirk for a Dell laptop
by Hui Wang 28 Apr '14

28 Apr '14

When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255, SID: 0x10280674), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c643dfc..c1952c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4621,6 +4621,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0674, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.1.2

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[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 28 Apr '14

28 Apr '14

The following changes since commit a798c10faf62a505d24e5f6213fbaf904a39623f: Linux 3.15-rc2 (2014-04-20 11:08:50 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-rc2 for you to fetch changes up to 98810a6dcf8af768d7b0b71e5075ff9c199f5196: Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/jz4740', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/tlv320aic3x' into asoc-linus (2014-04-22 22:01:07 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.15 A smattering of driver-specific fixes here, nothing generic. The Cirrus CODEC conversions to devm_ are leak fixes - the conversion adds missing error handling code. ---------------------------------------------------------------- Alexandre Belloni (1): ASoC: tlv320aic31xx: document that the regulators are mandatory Andrew Lunn (1): ASoC: alc5623: Fix regmap endianness Axel Lin (2): ASoC: cs42l52: Convert to use devm_gpio_request_one ASoC: cs42l73: Convert to use devm_gpio_request_one Ben Dooks (1): ASoC: rsnd: fix clock prepare/unprepare Christian Engelmayer (1): ASoC: Intel: Fix incorrect sizeof() in sst_hsw_stream_get_volume() Dan Carpenter (1): ASoC: Intel: some incorrect sizeof() usages Jarkko Nikula (1): ASoC: dapm: Fix widget double free with auto-disable DAPM kcontrol Lars-Peter Clausen (1): ASoC: jz4740: Remove Makefile entry for removed file Mark Brown (3): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/alc5623', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73' and 'asoc/fix/fsl-spdif' into asoc-linus Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/jz4740', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/tlv320aic3x' into asoc-linus Nicolin Chen (1): ASoC: fsl_spdif: Fix wrong OFFSET of STC_SYSCLK_DIV Sebastian Reichel (1): ASoC: tlv320aic3x: fix shared reset pin for DT Wenkai Du (1): ASoC: Intel: Fix audio crash due to negative address offset Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 6 +++--- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/cs42l52.c | 6 ++++-- sound/soc/codecs/cs42l73.c | 6 ++++-- sound/soc/codecs/tlv320aic3x.c | 9 +++++++-- sound/soc/fsl/fsl_spdif.h | 4 ++-- sound/soc/intel/sst-dsp-priv.h | 2 +- sound/soc/intel/sst-haswell-ipc.c | 7 ++++--- sound/soc/jz4740/Makefile | 2 -- sound/soc/sh/rcar/src.c | 4 ++-- sound/soc/sh/rcar/ssi.c | 4 ++-- sound/soc/soc-dapm.c | 1 - 12 files changed, 30 insertions(+), 23 deletions(-)

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[alsa-devel] [PATCH] SoC: rt5640: Add the rt5639 support to the OF match table
by Oder Chiou 28 Apr '14

28 Apr '14

The patch adds the rt5639 support to the OF match table. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 79635ee..2133fa0 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2157,6 +2157,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #if defined(CONFIG_OF) static const struct of_device_id rt5640_of_match[] = { + { .compatible = "realtek,rt5639", }, { .compatible = "realtek,rt5640", }, {}, }; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2] ASoC: rt5640: Add minimal support for RT5642
by bardliao＠realtek.com 28 Apr '14

28 Apr '14

From: Bard Liao <bardliao(a)realtek.com> We have been using rt5640.c codec driver with RT5642 codec chip before commit 022d21f004c1 ("ASoC: rt5640: add rt5639 support"). That commits starts using device ID reading in reset register for adding device specific controls and routes runtime. Now since device ID appears to be different between RT5640 and RT5642 the driver doesn't add those controls and routes that are valid also on RT5642. Fix this by adding a device ID found by debugging and minimal code for supporting RT5642. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This patch add a mask for reading the device ID. This can remove the affect of other bits which is not related to device ID. --- sound/soc/codecs/rt5640.c | 8 +++++--- sound/soc/codecs/rt5640.h | 10 +++++++--- 2 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6674372..79635ee 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1997,8 +1997,9 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); - switch (snd_soc_read(codec, RT5640_RESET)) { - case RT5640_RESET_ID: + switch (snd_soc_read(codec, RT5640_RESET) & RT5640_ID_MASK) { + case RT5640_ID_5640: + case RT5640_ID_5642: snd_soc_add_codec_controls(codec, rt5640_specific_snd_controls, ARRAY_SIZE(rt5640_specific_snd_controls)); @@ -2009,7 +2010,7 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_specific_dapm_routes, ARRAY_SIZE(rt5640_specific_dapm_routes)); break; - case RT5639_RESET_ID: + case RT5640_ID_5639: snd_soc_dapm_new_controls(&codec->dapm, rt5639_specific_dapm_widgets, ARRAY_SIZE(rt5639_specific_dapm_widgets)); @@ -2149,6 +2150,7 @@ static const struct regmap_config rt5640_regmap = { static const struct i2c_device_id rt5640_i2c_id[] = { { "rt5640", 0 }, { "rt5639", 0 }, + { "rt5642", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 3b50459..ded2059 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,9 +14,6 @@ #include <sound/rt5640.h> -#define RT5639_RESET_ID 0x0008 -#define RT5640_RESET_ID 0x000c - /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd @@ -195,6 +192,13 @@ #define RT5640_R_VOL_MASK (0x3f) #define RT5640_R_VOL_SFT 0 +/* SW Reset & Device ID (0x00) */ +#define RT5640_ID_MASK (0x3 << 1) +#define RT5640_ID_5639 (0x0 << 1) +#define RT5640_ID_5640 (0x1 << 1) +#define RT5640_ID_5642 (0x3 << 1) + + /* IN1 and IN2 Control (0x0d) */ /* IN3 and IN4 Control (0x0e) */ #define RT5640_BST_SFT1 12 -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] SoC: rt5640: Add the rt5639 support to the OF match table
by Oder Chiou 28 Apr '14

28 Apr '14

The patch adds the rt5639 support to the OF match table. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 79635ee..f2cc3ed 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2158,6 +2158,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #if defined(CONFIG_OF) static const struct of_device_id rt5640_of_match[] = { { .compatible = "realtek,rt5640", }, + { .compatible = "realtek,rt5639", }, {}, }; MODULE_DEVICE_TABLE(of, rt5640_of_match); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/5] Simple-audio-card support for HP t5325
by Andrew Lunn 28 Apr '14

28 Apr '14

This patchset allows the Kirkwood based HP t5325 device make use of simple-card to instantiate its audio hardware. Patch 1 Adds a DT binding to the ALC5623 codec Patch 2 Makes use of this new binding Patch 3 Add fixes up the kirkwood sound node in DT Patch 4 Extends simple-card to support the hw_params() needed when using MVEBU sound hardware. Patch 5 Swaps t5323 to using DT, removing all need for C code. Andrew Lunn (5): ASoC: alc5623: Add device tree binding ARM: Kirkwood: Use DT to instansiate codec ARM: Kirkwood: DT: Add missing #sound-dai-cells property ASoC: simple-card: Support setting mclk via a fixed factor ARM: Kirkwood: t5325: Use simple-card to instantiate audio .../devicetree/bindings/sound/alc5623.txt | 23 ++++++++++++ .../devicetree/bindings/sound/simple-card.txt | 2 ++ arch/arm/boot/dts/kirkwood-t5325.dts | 33 +++++++++++++++++ arch/arm/boot/dts/kirkwood.dtsi | 1 + arch/arm/mach-mvebu/board-t5325.c | 41 ---------------------- arch/arm/mach-mvebu/board.h | 6 ---- arch/arm/mach-mvebu/kirkwood.c | 3 -- sound/soc/codecs/alc5623.c | 13 +++++++ sound/soc/generic/simple-card.c | 28 +++++++++++++++ 9 files changed, 100 insertions(+), 50 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5623.txt delete mode 100644 arch/arm/mach-mvebu/board-t5325.c -- 1.9.2

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[alsa-devel] [PATCH 0/2] Fix rxclk rate measurement
by Nicolin Chen 28 Apr '14

28 Apr '14

Add sysclk clock so as to calculate the rxclk rate correctly Nicolin Chen (2): ASoC: fsl_spdif: Fix clock source for rxclk rate measurement ASoC: fsl_spdif: Drop hard code in clk_get() for rxclk sound/soc/fsl/fsl_spdif.c | 17 ++++++++++++++--- sound/soc/fsl/fsl_spdif.h | 2 ++ 2 files changed, 16 insertions(+), 3 deletions(-) -- 1.8.4

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[alsa-devel] Bug? poll() timeout in alsa-jack plugin
by Sergey 27 Apr '14

27 Apr '14

Hello. I'm trying to debug mozilla code, which uses poll() when playing sound and works fine with regular alsa device, but plays no sound over alsa-jack plugin. I narrowed it down to a small sample. In that sample poll() returns 0 even when pcm can accept data. Please, help me to fix either a bug in that sample, or a bug in alsa-plugins. The problem is hardware-independent. To reproduce it you need: 1. install and start jack2 daemon: jackd -r -dalsa -dhw:0 -r44100 -p1024 -n2 2. add to .asoundrc: pcm.jackplug { type plug slave { pcm "jack" } } pcm.jack { type jack playback_ports { 0 system:playback_1 1 system:playback_2 } capture_ports { 0 system:capture_1 1 system:capture_2 } } 3. build and run the poll() code example gcc -o poll-example poll-example.c -lasound ./poll-example According to the common sense as long as PCM is ready to receive data it's expected that poll() returns 1 and writei() is called. Instead poll() returns 0 and the code plays nothing. ---------- poll() code example --------- #include <assert.h> #include <poll.h> #include <alsa/asoundlib.h> short buffer[96000]; int main() { snd_pcm_t *pcm; assert( snd_pcm_open(&pcm, "jackplug", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) == 0 ); assert( snd_pcm_nonblock(pcm, 1) == 0 ); assert( snd_pcm_set_params(pcm, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 1/*channels*/, 48000/*rate*/, 1, 100000/*latency*/) == 0 ); nfds_t nfds = snd_pcm_poll_descriptors_count(pcm); assert( nfds > 0 ); struct pollfd *saved_fds = calloc(nfds, sizeof(struct pollfd)); assert( snd_pcm_poll_descriptors(pcm, saved_fds, nfds) == nfds ); // poll() and write(20) int timeout = 1000; int r = poll(saved_fds, nfds, timeout); fprintf(stderr, "poll=%d\n", r); if (r > 0) { unsigned short revents; r = snd_pcm_poll_descriptors_revents(pcm, saved_fds, nfds, &revents); fprintf(stderr, "snd_pcm_poll_descriptors_revents=%d\n", r); if (r == 0) fprintf(stderr, "snd_pcm_writei(20)=%d\n", snd_pcm_writei(pcm, buffer, 20)); } // poll() and write(50) r = poll(saved_fds, nfds, timeout); fprintf(stderr, "poll=%d\n", r); if (r > 0) { unsigned short revents; r = snd_pcm_poll_descriptors_revents(pcm, saved_fds, nfds, &revents); fprintf(stderr, "snd_pcm_poll_descriptors_revents=%d\n", r); if (r == 0) fprintf(stderr, "snd_pcm_writei(50)=%d\n", snd_pcm_writei(pcm, buffer, 50)); } // poll() and write(100) r = poll(saved_fds, nfds, timeout); fprintf(stderr, "poll=%d\n", r); if (r > 0) { unsigned short revents; r = snd_pcm_poll_descriptors_revents(pcm, saved_fds, nfds, &revents); fprintf(stderr, "snd_pcm_poll_descriptors_revents=%d\n", r); if (r == 0) fprintf(stderr, "snd_pcm_writei(100)=%d\n", snd_pcm_writei(pcm, buffer, 100)); } assert( snd_pcm_close(pcm) == 0 ); return 0; } ---------- end of code example --------- I tried to fix that in jack_pcm.c downloaded from http://git.alsa-project.org/?p=alsa-plugins.git;a=blob;f=jack/pcm_jack.c Making the first poll() return 1 was easy, I just added write(jack->fd, &i, 1); to the end of snd_pcm_jack_prepare() function. But that haven't solved the problem, because next call to snd_pcm_poll_descriptors_revents() exhausts the socket and next poll() still returns 0. Another guess was to define .transfer in snd_pcm_ioplug_callback_t and handle read()s and write()s to polling socket there, but it seems that .transfer is not called on writei() at all. What's the correct approach? Is there any official documentation about using poll() for ALSA? Or maybe there're some manuals with meaning of snd_pcm_ioplug_callback_t fields? --

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Re: [alsa-devel] [PATCH] ASoC: fsl-ssi: Allow to use driver with generic audio card
by Mark Brown 26 Apr '14

26 Apr '14

On Sat, Apr 26, 2014 at 09:02:06PM +0400, Alexander Shiyan wrote: > Sat, 26 Apr 2014 17:55:19 +0100 от Mark Brown <broonie(a)kernel.org>: > > On Sat, Apr 26, 2014 at 06:49:16PM +0400, Alexander Shiyan wrote: > > These are cases where we can do configuration but an invalid > > configuration has been provided and we have spotted an error. Ignoring > > that should cause runtime problems - I guess it isn't for you though so > > there may be something else going on? > > Especially for cases like DAI format we should probably enhance the > > framework so that these devices can set a variable in their struct > > saying what they do support but for now we don't do that. > Other solution to this problem is to check ret != EPROBE_DEFER > in the generic sound card. All other cases should be treated as > good for setup parameters. Is this better? I wouldn't expect to be getting -EPROBE_DEFER outside of the probe path? Can we take a step back here, please - what's the problem you're seeing?

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[alsa-devel] [PATCH] ASoC: Remove name_prefix unset during DAI link init hack again
by Lars-Peter Clausen 26 Apr '14

26 Apr '14

This was initially removed in commit 6479f15ad ("ASoC: Remove name_prefix unset during DAI link init hack"), but was brought back in commit 503ae5e0 ("ASoC: core: Add helpers for dai link and aux dev init") by accident. This patch removes it again. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f18112a..94c1668 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1272,14 +1272,10 @@ static int soc_aux_dev_init(struct snd_soc_card *card, { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - const char *temp; int ret; rtd->card = card; - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ if (aux_dev->init) { ret = aux_dev->init(&codec->dapm); @@ -1287,8 +1283,6 @@ static int soc_aux_dev_init(struct snd_soc_card *card, return ret; } - codec->name_prefix = temp; - rtd->codec = codec; return 0; @@ -1300,15 +1294,10 @@ static int soc_dai_link_init(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - const char *temp; int ret; rtd->card = card; - /* machine controls, routes and widgets are not prefixed */ - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ if (dai_link->init) { ret = dai_link->init(rtd); @@ -1316,8 +1305,6 @@ static int soc_dai_link_init(struct snd_soc_card *card, return ret; } - codec->name_prefix = temp; - rtd->codec = codec; return 0; -- 1.8.0

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