January 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [hdspm: Add RME RayDAT/AIO 0/2] Reworked support for RayDAT/AIO
by Adrian Knoth 26 Jan '11

26 Jan '11

Hi! Finally, here's the patch that adds RME RayDAT and AIO support to hdpsm. Code was taken from Florian Faber's http://wiki.linuxproaudio.org/index.php/Driver:hdspe revision (2010-12-25) and heavily modified to (almost) comply with the coding style. I also enabled S/PDIF input and augmented the code to allow for all period sizes between 64 and 4096 samples on RayDATs/AIOs. As pointed out by Takashi earlier this month, this version now doesn't use typedefs or enums in ioctl structs. I hope that I caught all upstream changes that happened in the last three years. ;) Obviously, the patch is extremely long, it adds MMAP access, support for the additional TCO hardware module, more MIDI ports and probably a lot more. It's more like a new generation than an evolution, and if you think it's too much, you could still call it hdspe and make it a new driver instead, however, given that OSX and Win32 only have one hdsp driver for all cards, it feels wrong to have three on Linux. In theory, all this can be merged with the hdsp driver, but that's for some other time. ;) Cheers Adrian Knoth (2): Add support for RME RayDAT and AIO Add RayDAT and AIO strings to Kconfig include/hdspm.h | 347 ++++- pci/Kconfig | 6 +- pci/rme9652/hdspm.c | 4227 +++++++++++++++++++++++++++++++++++++-------------- 3 files changed, 3424 insertions(+), 1156 deletions(-) -- 1.7.2.3

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[alsa-devel] [RFC][PATCH 0/3] ASoC: sh: fsi: sound format selection
by Kuninori Morimoto 26 Jan '11

26 Jan '11

Dear Mark, Liam, Paul These are FSI-codecs sound format selection patches. Kuninori Morimoto(3) ASoC: sh: fsi: move chan_num from fsi_stream to fsi_priv ASoC: sh: fsi: free from NULL pointer of struct sh_fsi_platform_info ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF format FSI-codecs sound format should be selected on fsi-xxx.c for ALSA. But FSI sound format is still selected from FSI platform data for now. 3rd patch modify it. (1st and 2nd are needed from 3rd patch) But current FSI can use I2S/PCM and SPDIF today. I2S/PCM can be selected by snd_soc_dai_set_fmt from fsi-xxx, but can not select SPDIF today. So, 3rd patch add 2 modes to FSI. DAI mode and SPDIF mode. If DAI mode (default), I2S/PCM format can be selected from fsi-xxx by snd_soc_dai_set_fmt. If SPDIF mode, FSI become SPDIF format. But I'm not sure it is OK for SPDIF. So, I set these patches as [RFC] Best regards -- Kuninori Morimoto

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[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 26 Jan '11

26 Jan '11

The following changes since commit 20a4e7fc7e213365ea3771d7bf1e10a6bab853be: ASoC: Handle low measured DC offsets for wm_hubs devices (2011-01-21 18:20:16 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Dmitry Eremin-Solenikov (1): ASoC: correct link specifications for corgi, poodle and spitz Lars-Peter Clausen (2): ASoC: Fix codec device id format used by some dai_links ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2s sound/soc/atmel/snd-soc-afeb9260.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/pxa/corgi.c | 4 ++-- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 4 ++-- sound/soc/samsung/neo1973_gta02_wm8753.c | 6 +++--- sound/soc/samsung/neo1973_wm8753.c | 6 +++--- sound/soc/samsung/s3c24xx_simtec_hermes.c | 4 ++-- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 ++-- sound/soc/samsung/s3c24xx_uda134x.c | 2 +- 10 files changed, 18 insertions(+), 18 deletions(-)

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[alsa-devel] [PATCH] ALSA: HDA: Fix automute on Thinkpad L412/L512
by David Henningsson 26 Jan '11

26 Jan '11

More Thinkpad machines with invalid SKU found, that disables automute between speakers and headphones on these machines. I'm considering putting in a quirk mask instead, but I'm not exactly sure on how wide I should make it...any suggestion? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[PATCH] ALSA: HDA: Add a new model "asus" for Conexant 5066/205xx
by David Henningsson 26 Jan '11

26 Jan '11

Backport of commit a1d6906e2d2b4655e248f490ab088c27876a600a (currently in linux/kernel/git/tiwai/sound-2.6.git tree), meant for kernel 2.6.38. BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 41 ++++++++++++++++++++++++- 2 files changed, 40 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 16ae430..0caf77e 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5066 ============= laptop Basic Laptop config (default) hp-laptop HP laptops, e g G60 + asus Asus K52JU, Lenovo G560 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 9bb030a..32bdd23 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -127,6 +127,7 @@ struct conexant_spec { unsigned int ideapad:1; unsigned int thinkpad:1; unsigned int hp_laptop:1; + unsigned int asus:1; unsigned int ext_mic_present; unsigned int recording; @@ -2312,6 +2313,19 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } + +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_asus_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 0); +} + + /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_hp_laptop_automic(struct hda_codec *codec) { @@ -2433,6 +2447,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } /* unsolicited event for jack sensing */ +static void cxt5066_asus_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_asus: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_asus_automic(codec); + break; + } +} + +/* unsolicited event for jack sensing */ static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); @@ -3053,6 +3081,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); + else if (spec->asus) + cxt5066_asus_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3080,6 +3110,7 @@ enum { CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3091,6 +3122,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTRO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", }; @@ -3103,6 +3135,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_HP_LAPTOP), + SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), @@ -3112,6 +3145,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ {} }; @@ -3167,13 +3201,16 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_ASUS: case CXT5066_HP_LAPTOP: codec->patch_ops.init = cxt5066_init; - codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + codec->patch_ops.unsol_event = board_config == CXT5066_ASUS ? + cxt5066_asus_event : cxt5066_hp_laptop_event; spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_hp_laptop; spec->num_init_verbs++; - spec->hp_laptop = 1; + spec->hp_laptop = board_config == CXT5066_HP_LAPTOP; + spec->asus = board_config == CXT5066_ASUS; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; spec->mixers[spec->num_mixers++] = cxt5066_mixers; /* no S/PDIF out */ -- 1.7.1 --------------080403010602060706040708--

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[alsa-devel] [PATCH] ASoC: Staticise non-exported symbols in cs4271
by Mark Brown 26 Jan '11

26 Jan '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4271.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 5357ec5..9c5b7db 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -386,7 +386,7 @@ static struct snd_soc_dai_ops cs4271_dai_ops = { .digital_mute = cs4271_digital_mute, }; -struct snd_soc_dai_driver cs4271_dai = { +static struct snd_soc_dai_driver cs4271_dai = { .name = "cs4271-hifi", .playback = { .stream_name = "Playback", @@ -526,7 +526,7 @@ static int cs4271_remove(struct snd_soc_codec *codec) return 0; }; -struct snd_soc_codec_driver soc_codec_dev_cs4271 = { +static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .probe = cs4271_probe, .remove = cs4271_remove, .suspend = cs4271_soc_suspend, -- 1.7.2.3

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[alsa-devel] [QUERY] Download calibration data to co-processor
by Patrick Lai 26 Jan '11

26 Jan '11

Hi, I am looking for ways to download audio calibration data to a co-processor under ALSA framework. I would like to know if there is already a precedent which handles this use case. I considered request_firmware() approach but there are few drawbacks to this approach. The audio calibration data size can be substantial as number of audio use cases grows. It also requires co-processor perform extensive parsing of data. Furthermore, host processor and co-processor need to establish common use case definitions so co-processor know what chunk of calibration data to apply in certain scenario such as device switch. If there is no precedent, my idea is to apply calibration data through sysfs. After introduction of ASoC multi-component architecture, PCM stream created from dailink is registered as device. So, there is syfs entry for each dailink. If ALSA ASoC can provide API for drivers(platform/CPU) to register additional attributes to the PCM device, this approach gives user-space application the ability to push down new calibration as audio situation changes. I believe it's a viable solution to my problem and perhaps can be useful to other systems as well. Your feedback is appreciated. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH v2 0/4] Tegra: Harmony: Add internal speaker support
by Stephen Warren 26 Jan '11

26 Jan '11

Harmony has headers on the board for an internal speaker and internal mic. This patch series adds various infra-structure, and enables usage of the internal speaker. The patch series includes both changes to code within the ASoC subsystem and Tegra subsystem, the latter provided only for context when reading the ASoC changes. Would you prefer I separated these so that the ASoC maintainers can grab a whole series and apply it, rather than just a subset of the patches? While patch 2 is physically located in the Tegra subsystem, I'd consider it part of the ASoC driver. Can such a change be checked into the ASoC tree? This new header file will be required to build to ASoC driver once the following patches are in. I was considering placing this in include/sound, following the example of various codecs, but it doesn't seem a good idea to pollute that directory with platform-specific files. The current location for the file follows the example of s3c24xx/simtec. Perhaps it should be in sound/soc/tegra? Stephen Warren (4): ASoC: WM8903: Expose GPIOs through gpiolib ARM: tegra: Add Harmony sound platform data type ARM: tegra: Platform data fixes for ASoC driver updates ASoC: tegra: Harmony: Support the internal speaker arch/arm/mach-tegra/board-harmony.c | 31 ++++++ arch/arm/mach-tegra/gpio-names.h | 2 + arch/arm/mach-tegra/include/mach/harmony_audio.h | 19 ++++ include/sound/wm8903.h | 7 +- sound/soc/codecs/wm8903.c | 117 +++++++++++++++++++++- sound/soc/tegra/harmony.c | 89 ++++++++++++++-- 6 files changed, 251 insertions(+), 14 deletions(-) create mode 100644 arch/arm/mach-tegra/include/mach/harmony_audio.h

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[PATCH] ALSA: HDA: Fix automute on Thinkpad L412/L512
by David Henningsson 26 Jan '11

26 Jan '11

BugLink: http://bugs.launchpad.net/bugs/707902 More Thinkpad machines with invalid SKU found, that disables automute between speakers and headphones on these machines. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index be4df4c..2fa9ed9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14954,9 +14954,11 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), - SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), - SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), + SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), {} -- 1.7.1 --------------050307010508090001070500--

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[alsa-devel] hdspmixer
by Jaroslav Kysela 26 Jan '11

26 Jan '11

Hi, working on new ALSA release, but it seems that alsa-tools/hdspmixer is out of sync with current linux/sound/hdsp.h / hdspm.h after this commit: commit e24e56795ea57e3dd7da45063ab71f04e231192d Author: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> Date: Sun Jan 9 19:42:59 2011 +0100 Add support for RME MADI, RayDAT and AIO Merged the work by Florian Faber that's distributed separately as hdspmixer64. Code taken from http://wiki.linuxproaudio.org/index.php/App:hdspmixer_64 Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Any clue what to do to get things fixed? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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