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

[alsa-devel] Query about regmap SPI
by Ashish Chavan 06 Mar '12

06 Mar '12

Hi, I am planning to add SPI support for da7210 codec driver. SPI for this codec has slightly different format than others. It has 8 bit register address and 8 bit data. Out of 8 bits of register address, one bit (i.e. LSB) is used to indicate read or write. Value of 1 at LSB denotes READ while value of 0 at LSB denotes write operation. Thus in order for SPI READ to work correctly, the register address need to be left shifted by one and then ORed with 0x1. In the same way for SPI WRITE to work correctly, it need to be left shifted by one. I have gone though the regmap sources and found that ORing can be achieved by setting correct read_flag_mask. But I couldn't find anything by which I can tell regmap core to shift the register address before ORing it with read_flag_mask. Am I missing anything here? Is there any other way to achieve this using regmap? The only other way I can think of is providing a wrapper function to read and write in codec driver and doing the shifting job inside that wrapper. TIA!

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Re: [alsa-devel] [PATCH v6] sound/soc/codecs: add LAPIS Semiconductor ML26124
by Mark Brown 06 Mar '12

06 Mar '12

On Tue, Mar 06, 2012 at 12:03:20PM +0900, Tomoya MORINAGA wrote: > 2012年3月2日21:58 Mark Brown <broonie(a)opensource.wolfsonmicro.com>: > >> Let me know why I should use SND_SOC_REGMAP not _I2C. > > Have you checked what _I2C and _REGMAP do? > I've just checked it. > If SND_SOC_REGMAP is used, it seems there is no processing in it. This is not the case. > So, can I delete snd_soc_codec_set_cache_io ? No. > > Use module_i2c_driver(). > I understand. > As a matter of interest, why can I use module_i2c_driver ? It is best practice. > There is no ASoC driver uses module_i2c_driver. This is not the case. As I have *REPEATEDLY* told you you should be sumbitting code against current development versions of the subsystem. > >> +static int ml26124_update_bits(struct snd_soc_codec *codec, unsigned short reg, > >> + unsigned int mask, unsigned int value) > > Why are you open coding this in your driver? There is no point in > > replicating subsystem functionality. > For ML26124 spec, it's necessary. > Because the codec's register read-address and write-address is not the same. > E.G. > Sampling Rate Register > R: offset=0 > W: offset=1 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.

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[alsa-devel] [PATCH] snd-hda-intel: add quirk to detect CD input on Gigabyte EP45-DS3
by Marton Balint 06 Mar '12

06 Mar '12

My CD input got lost in commit 68ef0561efe494143516df38c03a16b837b8e79c. Raymond helped me to add the necessary pin fixup to make it appear again. In fact, this is basically his patch. It fixes alsa bug #5541. Signed-off-by: Marton Balint <cus(a)passwd.hu> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f286bb8..5e53020 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4367,6 +4367,7 @@ enum { ALC882_FIXUP_PB_M5210, ALC882_FIXUP_ACER_ASPIRE_7736, ALC882_FIXUP_ASUS_W90V, + ALC889_FIXUP_CD, ALC889_FIXUP_VAIO_TT, ALC888_FIXUP_EEE1601, ALC882_FIXUP_EAPD, @@ -4494,6 +4495,13 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_CD] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1c, 0x993301f0 }, /* CD */ + { } + } + }, [ALC889_FIXUP_VAIO_TT] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { @@ -4650,6 +4658,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 1.7.3.4

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[alsa-devel] [PATCH - aplay 1/1] aplay - option to dump HW parameters
by pavel.hofman＠ivitera.com 06 Mar '12

06 Mar '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Dumping HW parameters of the current device with snd_pcm_hw_params_dump, a new option --dump-hw-params added. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/aplay/aplay.1 b/aplay/aplay.1 index 2acaed5..0195322 100644 --- a/aplay/aplay.1 +++ b/aplay/aplay.1 @@ -184,6 +184,13 @@ the month, %H hour, %M minute and %S second. In addition, %v is the file number, starting at 1. When this option is specified, intermediate directories for the output file are created automatically. This option has no effect if \-\-separate\-channels is specified. +.TP +\fI\-\-dump\-hw\-params\fP +Dump hw_params of the device preconfigured status to stderr. The dump +lists capabilities of the selected device such as supported formats, +sampling rates, numbers of channels, period and buffer bytes/sizes/times. +For raw device hw:X this option basically lists hardware capabilities of +the soundcard. .SH SIGNALS When recording, SIGINT, SIGTERM and SIGABRT will close the output diff --git a/aplay/aplay.c b/aplay/aplay.c index 73a9544..0633cbd 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -129,6 +129,7 @@ static int max_file_time = 0; static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; +static int dump_hw_params = 0; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -223,7 +224,8 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --max-file-time=# start another output file when the old file has recorded\n" " for this many seconds\n" " --process-id-file write the process ID here\n" -" --use-strftime apply the strftime facility to the output file name\n") +" --use-strftime apply the strftime facility to the output file name\n" +" --dump-hw-params dump hw_params of the device\n") , command); printf(_("Recognized sample formats are:")); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { @@ -416,7 +418,8 @@ enum { OPT_TEST_NOWAIT, OPT_MAX_FILE_TIME, OPT_PROCESS_ID_FILE, - OPT_USE_STRFTIME + OPT_USE_STRFTIME, + OPT_DUMP_HWPARAMS }; int main(int argc, char *argv[]) @@ -461,6 +464,7 @@ int main(int argc, char *argv[]) {"process-id-file", 1, 0, OPT_PROCESS_ID_FILE}, {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, + {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {0, 0, 0, 0} }; char *pcm_name = "default"; @@ -662,6 +666,9 @@ int main(int argc, char *argv[]) case OPT_USE_STRFTIME: use_strftime = 1; break; + case OPT_DUMP_HWPARAMS: + dump_hw_params = 1; + break; default: fprintf(stderr, _("Try `%s --help' for more information.\n"), command); return 1; @@ -1059,6 +1066,13 @@ static void set_params(void) error(_("Broken configuration for this PCM: no configurations available")); prg_exit(EXIT_FAILURE); } + if (dump_hw_params) { + fprintf(stderr, _("HW Params of device \"%s\":\n"), + snd_pcm_name(handle)); + fprintf(stderr, "--------------------\n"); + snd_pcm_hw_params_dump(params, log); + fprintf(stderr, "--------------------\n"); + } if (mmap_flag) { snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof()); snd_pcm_access_mask_none(mask); -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: wm8962: Remove defaults for volatile registers
by Mark Brown 06 Mar '12

06 Mar '12

Save a little RAM. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 17 +++-------------- 1 files changed, 3 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d5ee3ec..e5aaf24 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -116,11 +116,11 @@ static struct reg_default wm8962_reg[] = { { 1, 0x049F }, /* R1 - Right Input volume */ { 2, 0x0000 }, /* R2 - HPOUTL volume */ { 3, 0x0000 }, /* R3 - HPOUTR volume */ - { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0018 }, /* R5 - ADC & DAC Control 1 */ { 6, 0x2008 }, /* R6 - ADC & DAC Control 2 */ { 7, 0x000A }, /* R7 - Audio Interface 0 */ - { 8, 0x01E4 }, /* R8 - Clocking2 */ + { 9, 0x0300 }, /* R9 - Audio Interface 1 */ { 10, 0x00C0 }, /* R10 - Left DAC volume */ { 11, 0x00C0 }, /* R11 - Right DAC volume */ @@ -129,7 +129,7 @@ static struct reg_default wm8962_reg[] = { { 15, 0x6243 }, /* R15 - Software Reset */ { 17, 0x007B }, /* R17 - ALC1 */ - { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x1C32 }, /* R19 - ALC3 */ { 20, 0x3200 }, /* R20 - Noise Gate */ { 21, 0x00C0 }, /* R21 - Left ADC volume */ @@ -153,10 +153,6 @@ static struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ - { 47, 0x0000 }, /* R47 - Thermal Shutdown Status */ - { 48, 0x8027 }, /* R48 - Additional Control (4) */ - { 49, 0x0010 }, /* R49 - Class D Control 1 */ - { 51, 0x0003 }, /* R51 - Class D Control 2 */ { 56, 0x0506 }, /* R56 - Clocking 4 */ @@ -168,8 +164,6 @@ static struct reg_default wm8962_reg[] = { { 64, 0x0810 }, /* R64 - DC Servo 4 */ - { 66, 0x0000 }, /* R66 - DC Servo 6 */ - { 68, 0x001B }, /* R68 - Analogue PGA Bias */ { 69, 0x0000 }, /* R69 - Analogue HP 0 */ @@ -302,9 +296,6 @@ static struct reg_default wm8962_reg[] = { { 516, 0x8100 }, /* R516 - GPIO 5 */ { 517, 0x8100 }, /* R517 - GPIO 6 */ - { 560, 0x0000 }, /* R560 - Interrupt Status 1 */ - { 561, 0x0000 }, /* R561 - Interrupt Status 2 */ - { 568, 0x0030 }, /* R568 - Interrupt Status 1 Mask */ { 569, 0xFFED }, /* R569 - Interrupt Status 2 Mask */ @@ -316,8 +307,6 @@ static struct reg_default wm8962_reg[] = { { 768, 0x1C00 }, /* R768 - DSP2 Power Management */ - { 1037, 0x0000 }, /* R1037 - DSP2_ExecControl */ - { 8192, 0x0000 }, /* R8192 - DSP2 Instruction RAM 0 */ { 9216, 0x0030 }, /* R9216 - DSP2 Address RAM 2 */ -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: wm8994: Use fast VMID ramp
by Mark Brown 06 Mar '12

06 Mar '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 581049a..a6df001 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -808,7 +808,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x0 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, @@ -816,7 +816,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_SEL_MASK, WM8994_BIAS_ENA | 0x2); - msleep(100); + msleep(50); snd_soc_update_bits(codec, WM8994_ANTIPOP_2, WM8994_VMID_RAMP_MASK | WM8994_BIAS_SRC, -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: wm8994: Use audio mode for jack detection when system is active
by Mark Brown 06 Mar '12

06 Mar '12

When we are out of system sleep always use audio mode for jack detection in order to avoid potential performance issues handing off between modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 77 ++++++++++++++++++++++++++++----------------- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 49 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2417ef9..bc12d09 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -685,8 +685,6 @@ SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0, static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - u16 old = snd_soc_read(codec, WM8994_ANTIPOP_2) - & WM1811_JACKDET_MODE_MASK; if (!wm8994->jackdet || !wm8994->jack_cb) return; @@ -694,28 +692,17 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; - if (mode == old) + if (mode == wm8994->jackdet_mode) return; - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, mode); - - switch (mode) { - case WM1811_JACKDET_MODE_MIC: - case WM1811_JACKDET_MODE_AUDIO: - switch (old) { - case WM1811_JACKDET_MODE_MIC: - case WM1811_JACKDET_MODE_AUDIO: - break; - default: - msleep(2); - break; - } + wm8994->jackdet_mode = mode; - default: - break; - } + /* Always use audio mode to detect while the system is active */ + if (mode != WM1811_JACKDET_MODE_NONE) + mode = WM1811_JACKDET_MODE_AUDIO; + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, mode); } static void active_reference(struct snd_soc_codec *codec) @@ -2749,7 +2736,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { }; #ifdef CONFIG_PM -static int wm8994_suspend(struct snd_soc_codec *codec) +static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -2783,7 +2770,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec) return 0; } -static int wm8994_resume(struct snd_soc_codec *codec) +static int wm8994_codec_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -2842,8 +2829,8 @@ static int wm8994_resume(struct snd_soc_codec *codec) return 0; } #else -#define wm8994_suspend NULL -#define wm8994_resume NULL +#define wm8994_codec_suspend NULL +#define wm8994_codec_resume NULL #endif static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) @@ -3955,8 +3942,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .probe = wm8994_codec_probe, .remove = wm8994_codec_remove, - .suspend = wm8994_suspend, - .resume = wm8994_resume, + .suspend = wm8994_codec_suspend, + .resume = wm8994_codec_resume, .set_bias_level = wm8994_set_bias_level, }; @@ -3983,11 +3970,43 @@ static int __devexit wm8994_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int wm8994_suspend(struct device *dev) +{ + struct wm8994_priv *wm8994 = dev_get_drvdata(dev); + + /* Drop down to power saving mode when system is suspended */ + if (wm8994->jackdet && !wm8994->active_refcount) + regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + wm8994->jackdet_mode); + + return 0; +} + +static int wm8994_resume(struct device *dev) +{ + struct wm8994_priv *wm8994 = dev_get_drvdata(dev); + + if (wm8994->jackdet && wm8994->jack_cb) + regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + WM1811_JACKDET_MODE_AUDIO); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8994_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(wm8994_suspend, wm8994_resume) +}; + static struct platform_driver wm8994_codec_driver = { .driver = { - .name = "wm8994-codec", - .owner = THIS_MODULE, - }, + .name = "wm8994-codec", + .owner = THIS_MODULE, + .pm = &wm8994_pm_ops, + }, .probe = wm8994_probe, .remove = __devexit_p(wm8994_remove), }; diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f996d14..2f4d2d1 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -122,6 +122,7 @@ struct wm8994_priv { bool jack_mic; int btn_mask; bool jackdet; + int jackdet_mode; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- 1.7.9.1

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[alsa-devel] [PATCH v2 00/17] ASoC: add imx-sgtl5000 machine driver working with fsl_ssi
by Shawn Guo 05 Mar '12

05 Mar '12

Changes since v1: * Address comments from Mark, Timur, and Sascha (Thanks for review). * Drop patches that have been applied by Mark. Richard Zhao (2): ARM: imx: convert audmux to a platform driver ASoC: fsl: add dt support for imx-audmux Shawn Guo (15): ASoC: imx: move eukrea audmux call into ASoC machine driver ASoC: imx: move phycore audmux call into ASoC machine driver ARM: imx: merge audmux-v1 and audmux-v2 ASoC: imx: move audmux driver into sound/soc/imx ASoC: imx: rename audmux prefix mxc to imx ASoC: imx: separate imx-pcm bits from imx-ssi driver ASoC: imx: add an explicit Kconfig option for imx-ssi driver ASoC: fsl: separate SSI and DMA Kconfig options ASoC: imx: merge sound/soc/imx into sound/soc/fsl ASoC: fsl: create fsl_utils to accommodate the common functions ASoC: fsl: check property 'compatible' for the machine name ASoC: fsl: make fsl_ssi driver compilable on ARM/IMX ASoC: fsl: remove the fatal error checking on codec-handle ASoC: fsl: let fsl_ssi work with imx pcm and machine drivers ASoC: fsl: add imx-sgtl5000 machine driver .../bindings/sound/imx-audio-sgtl5000.txt | 24 +++ .../devicetree/bindings/sound/imx-audmux.txt | 13 ++ arch/arm/mach-imx/Kconfig | 6 - arch/arm/mach-imx/eukrea_mbimx27-baseboard.c | 20 -- arch/arm/mach-imx/eukrea_mbimxsd-baseboard.c | 1 - arch/arm/mach-imx/eukrea_mbimxsd25-baseboard.c | 17 -- arch/arm/mach-imx/eukrea_mbimxsd35-baseboard.c | 17 -- arch/arm/mach-imx/mach-pca100.c | 13 -- arch/arm/mach-imx/mach-pcm043.c | 13 -- arch/arm/mach-imx/mm-imx21.c | 6 + arch/arm/mach-imx/mm-imx25.c | 7 + arch/arm/mach-imx/mm-imx27.c | 7 + arch/arm/mach-imx/mm-imx3.c | 13 ++ arch/arm/mach-imx/mm-imx5.c | 22 +++ arch/arm/plat-mxc/Kconfig | 6 - arch/arm/plat-mxc/Makefile | 2 - arch/arm/plat-mxc/audmux-v1.c | 64 ------- arch/arm/plat-mxc/include/mach/audmux.h | 60 ------- sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/fsl/Kconfig | 124 +++++++++++++- sound/soc/fsl/Makefile | 29 +++- sound/soc/{imx => fsl}/eukrea-tlv320.c | 42 ++++- sound/soc/fsl/fsl_ssi.c | 139 ++++++++++++---- sound/soc/fsl/fsl_utils.c | 135 ++++++++++++++ sound/soc/fsl/fsl_utils.h | 27 +++ .../audmux-v2.c => sound/soc/fsl/imx-audmux.c | 185 +++++++++++++++----- sound/soc/fsl/imx-audmux.h | 60 +++++++ sound/soc/{imx => fsl}/imx-pcm-dma-mx2.c | 2 +- sound/soc/{imx => fsl}/imx-pcm-fiq.c | 0 sound/soc/fsl/imx-pcm.c | 105 +++++++++++ sound/soc/fsl/imx-pcm.h | 32 ++++ sound/soc/fsl/imx-sgtl5000.c | 181 +++++++++++++++++++ sound/soc/{imx => fsl}/imx-ssi.c | 90 +---------- sound/soc/{imx => fsl}/imx-ssi.h | 16 +-- sound/soc/fsl/mpc8610_hpcd.c | 159 ++---------------- sound/soc/{imx => fsl}/mx27vis-aic32x4.c | 20 +- sound/soc/fsl/p1022_ds.c | 181 ++------------------ sound/soc/{imx => fsl}/phycore-ac97.c | 27 +++- sound/soc/{imx => fsl}/wm1133-ev1.c | 25 ++-- sound/soc/imx/Kconfig | 60 ------- sound/soc/imx/Makefile | 19 -- 42 files changed, 1140 insertions(+), 831 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: core - Add card mutex locking subclasses
by Liam Girdwood 05 Mar '12

05 Mar '12

This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. We currently use the codec->mutex for locking some DAPM operations. This can't guarantee correctness now with the recent multi-component and forth coming Dynamic PCM updates. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for all the DAPM and forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the DAPM operations (something that would never deadlock in real life). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 2 +- 2 files changed, 6 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 82bd773..f2b5ad1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -288,6 +288,11 @@ enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_BE = 1, }; +enum snd_soc_card_subclass { + SND_SOC_CARD_CLASS_INIT = 0, + SND_SOC_CARD_CLASS_PCM = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 29dbbd7..2038522 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1416,7 +1416,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_dai_link *dai_link; int ret, i, order; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); if (card->instantiated) { mutex_unlock(&card->mutex); -- 1.7.5.4

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[alsa-devel] [bug?, 3.1 -> 3.2.7 regression] ALC269 (ThinkPad L412): auto-mute mode disabled by default
by Jonathan Nieder 05 Mar '12

05 Mar '12

Hi Takashi et al, Pavel Yakunin wrote[1]: > There is a problem with a headphones jack detection in the 3.2 kernel for the ALC269 codec: > sound comes from both speaker and headphones simultaneously and never switches between them. > > The problem arised when I upgraded my system to the 3.2 kernel, and when I rolled back to the 3.1 > kernel, everything was ok. It turned out to be due to the auto-mute mode control, which is present on this model in 3.2 and not 3.1, being disabled by default. > On 2012-03-05 05:16, Jonathan Nieder wrote: >> One more test: if you run "alsactl init" as root, does the auto-mute >> mode control end up enabled or disabled? [...] > It ends up disabled. Versions tested: 3.1.y - auto-mute works, no auto-mute control present 3.2.y - auto-mute disabled by default but control works sound/for-next as of 2012-03-03 - likewise Is this behavior intended? More details, including alsa-info.sh logs, at [1]. Thanks, Jonathan [1] http://bugs.debian.org/661830

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