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

[alsa-devel] Fwd: Re: Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again)
by Colomban Wendling 18 Sep '13

18 Sep '13

Here's my initial mail, but with gzipped outputs. Regards, Colomban -------- Message original -------- Sujet: Re: [alsa-devel] Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again) Date : Thu, 10 Nov 2011 02:11:17 +0100 De : Colomban Wendling <lists.ban(a)herbesfolles.org> Pour : Takashi Iwai <tiwai(a)suse.de> Copie à : alsa-devel(a)alsa-project.org Hi again, and sorry for the delay. Le 08/11/2011 07:32, Takashi Iwai a écrit : > At Mon, 07 Nov 2011 23:45:40 +0100, > Colomban Wendling wrote: >> >> Hi, >> >> Back to 3.0, the sound on my soundcard [1] started to be choppy, I >> reported it [2] and it got fixed (thanks to Takashi Iwai!). >> >> However, the story repeated with 3.1 (and probably 3.0.8 or before [3]): >> I've got similar choppy sound again. >> >> I bisected the few commits that happened on >> sound/pci/hda/patch_realtek.c, and finally found the villain: >> >> "8974bd51 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration" >> >> Reverting it from v3.1 fixes the problem. > > Does it? Very weird. This patch has nothing to do with the HD-audio > controller side but purely a codec change. Yep, I just check once again and it really fixes the problem here. >> master (94956ee) also suffers of the problem, but I couldn't test >> without that commit because reverting it fails, too much changes happened. > > Did you try to pass position_fix=1 (or 2) option? > Also, passing enable_msi=0 (or 1) may change anything? I just tried with the 3 kernels: * vanilla v3.1 * vanilla 3.2-rc1 (master at 1ea6b8f) * v3.1 with 8974bd51 reverted and none of the option changed anything: both vanilla always failed, the third always succeeded. Though, I haven't combined the options, just tried each at once. > In anyway, please give alsa-info.sh output again with the fresh > kernel. Run it with --no-upload option and attach the output. Here it is, for the 3 kernels cited above. Regards, Colomban

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[alsa-devel] [PATCH 0/4] add regmap_filelds and use it on Renesas Sound driver
by Kuninori Morimoto 17 Sep '13

17 Sep '13

Hi Mark These patches add new regmap_filelds API on kernel. ~ and use it on Renesas sound driver instead of original method. It can care about multi port register offset via regmap. 0xAAAA + 0x40 -- port 0 -- regX regY regZ 0xAAAA + 0x80 -- port 1 -- regX regY regZ This case, current API needs 2 (= port) x 3 (= regX/Y/Z) regmap_fileld, but this new API can care about all port via 3 regmap_filelds with port ID. I'm not sure that regmap_filelds is good naming or not. Please let me know if you have good naming idea. These are based on below branchs regmap/for-next + asoc/for-next Kuninori Morimoto (4): regmap: add regmap_field_update_bits() regmap: Add regmap_fields APIs ASoC: rsnd: gen: rsnd_gen_ops cares .probe and .remove ASoC: rsnd: use regmap instead of original register mapping method drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regmap.c | 117 +++++++++++++++++ include/linux/regmap.h | 14 +++ sound/soc/sh/rcar/core.c | 45 ------- sound/soc/sh/rcar/gen.c | 270 +++++++++++++++++++++++++--------------- 5 files changed, 307 insertions(+), 142 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: codecs: da9055: Update driver name to fix breakage due to pmic driver with same name
by Ashish Chavan 13 Sep '13

13 Sep '13

This patch updates i2c driver name and device id of da9055 codec driver. DA9055 is a PMIC + CODEC and currently, the corresponding PMIC driver also registers itself with the same name as codec, i.e. "da9055". Because of this the codec driver was broken. Now codec driver uses "da9055-codec" as driver name instead of "da9055". Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 2e1bf5d..e146f68 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1544,7 +1544,7 @@ static int da9055_remove(struct i2c_client *client) } static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); @@ -1552,7 +1552,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, .probe = da9055_i2c_probe, -- 1.7.1

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[alsa-devel] No sound from internal speaker on 2013 Macbook Air 6, 1
by Ian Munsie 13 Sep '13

13 Sep '13

Hey all, I'm trying to get sound working on my new mid 2013 Macbook Air 6,1. Sound works via the headphone jack, but I haven't managed to get the internal speaker to work. I've checked all the controls in the mixer, tried a few different mac related model= options to snd-hda-intel and even tried tweaking things with hda_analyzer to no avail. I've uninstalled pulseaudio (for now) to get rid of it's ALSA hook, but had the same result both with and without it. I'm running on a 3.11-rc6+ built today (had a few other fixes to make for this machine) - I didn't bother including the hda i915 driver as I don't need sound via the display port, but the hda driver does print a warning about that missing module. The speaker wasn't working on the 3.10 kernel from Debian either which included that module. Any help would be appreciated - I've attached my alsa-info output taken after a reboot. Cheers, -Ian -- http://sites.google.com/site/DarkStarJunkSpace -- http://darkstarshout.blogspot.com/ -- Please avoid sending me Word or PowerPoint attachments. See http://www.gnu.org/philosophy/no-word-attachments.html

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[alsa-devel] where to get snd_pcm_hardware info when usb audio device plug in?
by loody 09 Sep '13

09 Sep '13

hi all: is it possible to get snd_pcm_hardware information in /proc, /sys or anywhere when usb audio device plug in? if no, is there api I can take in usermode to get snd_pcm_hardware of this usb audio device? -- Thanks for your great help,

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[alsa-devel] US-1641 drivers involvment
by Luca Novarini 08 Sep '13

08 Sep '13

Hello, I would participate, if possible, in the ALSA development to add support to the USB card Tascam US-1641. Any reply or help from who tried in this task before is really really appreciated. Thank you! Bye. LL

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[alsa-devel] [PATCH 00/14] SPDIF support
by Russell King - ARM Linux 04 Sep '13

04 Sep '13

This is my latest set of patches to get SPDIF working on the Dove Cubox platform. As per the last time, I've left out the "card" part of this, because that is only useful for non-DT setups, which aren't supported in mainline. The first seven patches have already been applied by Mark, and are included here for completeness by anyone who wishes to test this series. There is very little change between this and the previous set of patches. The HACK patch is gone, as I've found out through much digging and experimentation with DAPM (inspite of the complete lack of assistance from ASoC developers) how to use the stream widgets without the widget overwriting bug getting in the way. The CPU DAI widgets are attached to the CPU DAI context, and their access to the driver private data updated accordingly. Why attaching them to the CPU DAI context previously didn't work, I'm not entirely sure, but that certainly previously failed in testing (it may have been due to the duplicated DAI routes and/or the lack of output pin on the SPDIF tranceiver.) I've added three patches: one to add IEC958 channel status setting, another to prefer the external clock over the internal clock, and removal of the IEC958 subframe support (see the commit message for that patch why this can not be supported.) This is a non-DPCM driver, and is the absolute minimum solution which can provide correct hardware behaviour while still being conformant with single zImage kernels and not causing any known regressions. The same conditions apply as per my previous posting - the DAI link needs to be setup and the associated DAPM routes to tell the CPU DAI which outputs are in use, like this: DAI link: .name = "S/PDIF1", .stream_name = "IEC958 Playback", .platform_name = "mvebu-audio.1", .cpu_dai_name = "mvebu-audio.1", .codec_dai_name = "dit-hifi", .codec_name = "spdif-dit", static const struct snd_soc_dapm_route routes[] = { { "Playback", NULL, "spdifdo" }, }; When DPCM is eventually implemented, the DAPM route remains, the DAI link becomes two, as previously illustrated in other discussions. Full diffstat for this series: arch/arm/mach-dove/common.c | 4 +- arch/arm/mach-kirkwood/common.c | 24 ++--- sound/soc/codecs/spdif_transciever.c | 15 +++- sound/soc/kirkwood/Kconfig | 5 - sound/soc/kirkwood/Makefile | 4 +- sound/soc/kirkwood/kirkwood-dma.c | 114 ++++-------------- sound/soc/kirkwood/kirkwood-i2s.c | 213 +++++++++++++++++++++++++++++----- sound/soc/kirkwood/kirkwood-openrd.c | 12 ++- sound/soc/kirkwood/kirkwood-t5325.c | 8 +- sound/soc/kirkwood/kirkwood.h | 45 +++++++- sound/soc/soc-dapm.c | 2 +- 11 files changed, 295 insertions(+), 151 deletions(-) Diffstat for the patches remaining for mainline: sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/kirkwood/kirkwood-i2s.c | 170 +++++++++++++++++++++++++++++++--- sound/soc/kirkwood/kirkwood-openrd.c | 8 ++ sound/soc/kirkwood/kirkwood-t5325.c | 4 + sound/soc/kirkwood/kirkwood.h | 34 +++++++ 5 files changed, 205 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 0/3] ASoC: generic simple sound card DT bindings
by Markus Pargmann 04 Sep '13

04 Sep '13

Hi, The patches were already in the fsl-ssi ac97 series. I moved the seperated simple sound card DT driver into the existing simple card driver. Regards, Markus Markus Pargmann (3): ASoC: generic: simple card, use private data ASoC: generic simple sound card DT bindings ASoC: fsl: Kconfig, visible config items for simple card .../bindings/sound/generic,simple-dt-card.txt | 28 +++ include/sound/simple_card.h | 4 - sound/soc/fsl/Kconfig | 23 ++- sound/soc/generic/Kconfig | 1 + sound/soc/generic/simple-card.c | 225 ++++++++++++++++++--- 5 files changed, 246 insertions(+), 35 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/generic,simple-dt-card.txt

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[alsa-devel] [PATCH 1/5] ASoC: Remove support for reg_access_defaults
by Lars-Peter Clausen 02 Sep '13

02 Sep '13

No users of reg_access_defaults are left and new drivers are going to use regmap for this, so support for it can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 24 -------------------- sound/soc/soc-cache.c | 63 --------------------------------------------------- sound/soc/soc-core.c | 9 -------- 3 files changed, 96 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7184805..ae6efc6d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -403,12 +403,6 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg); int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -542,22 +536,6 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** - * struct snd_soc_reg_access - Describes whether a given register is - * readable, writable or volatile. - * - * @reg: the register number - * @read: whether this register is readable - * @write: whether this register is writable - * @vol: whether this register is volatile - */ -struct snd_soc_reg_access { - u16 reg; - u16 read; - u16 write; - u16 vol; -}; - -/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -760,8 +738,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - short reg_access_size; - const struct snd_soc_reg_access *reg_access_default; enum snd_soc_compress_type compress_type; /* codec bias level */ diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index e72f554..eaa898f 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -275,66 +275,3 @@ int snd_soc_cache_sync(struct snd_soc_codec *codec) return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); - -static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - unsigned int min, max, index; - - codec_drv = codec->driver; - min = 0; - max = codec_drv->reg_access_size - 1; - do { - index = (min + max) / 2; - if (codec_drv->reg_access_default[index].reg == reg) - return index; - if (codec_drv->reg_access_default[index].reg < reg) - min = index + 1; - else - max = index; - } while (min <= max); - return -1; -} - -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].vol; -} -EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); - -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].read; -} -EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); - -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].write; -} -EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7d13c4f..b78fa93 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4222,15 +4222,6 @@ int snd_soc_register_codec(struct device *dev, } } - if (codec_drv->reg_access_size && codec_drv->reg_access_default) { - if (!codec->volatile_register) - codec->volatile_register = snd_soc_default_volatile_register; - if (!codec->readable_register) - codec->readable_register = snd_soc_default_readable_register; - if (!codec->writable_register) - codec->writable_register = snd_soc_default_writable_register; - } - for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); -- 1.8.0

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[alsa-devel] [PATCH 0/5] Remove ASoC-level I/O functions from cq93vc
by Mark Brown 02 Sep '13

02 Sep '13

This patch series removes the use of the ASoC-level read and write functions from the cq93vc driver as part of a wider push to remove them completely and just use regmap for all register I/O. Since the driver is essentially doing what regmap-mmio is doing this is done by adding a MMIO regmap to the core device and using that. This is compile tested only, I don't have any hardware to run on. Since I anticipate doing more cleanups over the CODEC drivers during the next release cycle it'd be good to merge via ASoC, though there shouldn't be any overlap with the first patch. Mark Brown (5): mfd: davinci_voicecodec: Remove unused read and write functions mfd: davinci_voicecodec: Provide a regmap for register I/O ASoC: cq93vc: Use core I/O functions ASoC: cq93vc: Don't use control data for core driver data ASoC: cq93vc: Use regmap for I/O drivers/mfd/davinci_voicecodec.c | 23 +++++++++++-------- include/linux/mfd/davinci_voicecodec.h | 3 +++ sound/soc/codecs/cq93vc.c | 42 +++++++++++----------------------- 3 files changed, 29 insertions(+), 39 deletions(-)

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