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

[alsa-devel] [PATCH] ALSA: HDA: Fix microphone(s) on Lenovo Edge 13
by David Henningsson 04 Feb '11

04 Feb '11

This Edge 13 model has an internal mic at 0x1a and should therefore use the asus quirk. Since my backport of the asus model is now tested and working, would it be possible to apply that patch as well as this quirk to 2.6.38? Thanks Takashi for keeping up with all the patches coming in! :-) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 11

[alsa-devel] NVIDIA HDMI surround sound is broken starting with kernel 2.6.36
by VDR User 04 Feb '11

04 Feb '11

The behavior described below started with 2.6.36 and follows through up to the most recent stable kernel, 2.6.37. The last properly working driver is in kernel 2.6.35.9. I should also mention that while the tests below use the static_hdmi_pcm flag recently introduced to alsa git, I also tested with the in-kernel drivers for each stable kernel. The particular device used for testing is an Nvidia GT220 video card connected directly to my receiver via HDMI. Can someone please look into this? Tested: cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=1, receiver ON cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=1, receiver OFF cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=0, receiver ON cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=0, receiver OFF In all cases only the front-left and front-right speakers worked. LFE, rear-left, rear-right, and center were missing. ELD correctly reported channels corresponding to receiver ON/OFF. (receiver on=7.1, receiver off=tv stereo). Also, even though the eld file is removed, the eld info is _not_ updated when stopping alsa & unloading drivers, then reloading. A reboot is required to get the eld info to update. This is why I cold boot the pc for each test. With kernel 2.6.35.9 drivers, front-left & front-right worked with receiver OFF as expected. I then turn the receiver on and all channels worked. I didn't bother to stop alsa, reboot, or even look at eld info since it works. Thanks for any & all help, Derek

2 3

[alsa-devel] UCM list/set/get
by pl bossart 03 Feb '11

03 Feb '11

I've been trying to figure out what exactly needs to be filled in UCM configuration files, and so far it's been a pretty frustrating experience with alsa-utils/alsaucm :-( The interactive mode in alsa-utils/alsaucm is broken, I fixed a couple of segfaults and bad output (patches attached), but so far the only thing I can do is list the verbs...It looks like alsa-lib/include/use-case.h and alsa-utils/alsaucm are not aligned, whenever I query with predefined identifiers alsaucm seg faults or provides an error message that doesn't help much. This happens with the latest code. [ume@plb alsaucm]$ ./alsaucm listcards 0: AudioPC this is my AudioPCI card 1: SDP4430 this is my SDP4430 card 2: TestHDA A test HDA card [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _verbs 0: HiFi Play and record HiFi quality Music. 1: Voice Full duplex low-power voice call So far so good. Now the fun begins [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _devices ./alsaucm: error failed to get list _devices: No such file or directory [ume@plb alsaucm]$ ./alsaucm -c AudioPCI set _verb HiFi ALSA lib main.c:1250:(set_verb_user) error: failed to initialize new use case: HiFi ./alsaucm: error failed to set _verb=HiFi: No such device [ume@plb alsaucm]$ ./alsaucm -c AudioPCI get _verb ./alsaucm: error failed to get _verb: No such file or directory Likewise if the .conf file contains a cset in the default values, then the parser fails (I took the example from src/ucm/parser.c). Commenting out the cset is the only solution. What exactly is this index supposed to mean? SectionDefaults [ cdev "hw:AudioPCI" cset "name='Master Playback Switch',index=2 1,1" ] [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _verbs ALSA lib main.c:302:(import_master_config) Unable to execute default sequence ALSA lib main.c:637:(snd_use_case_mgr_open) error: failed to import AudioPCI use case configuration -19 ./alsaucm: error failed to open sound card AudioPCI: No such device Ideally I'd like to write an interface between PulseAudio and UCM, there are existing modules I can reuse to set voice or hifi modes, depending on what the client roles are. But if I can't make the examples work I don't think this is going to go anywhere. -Pierre

5 10

[alsa-devel] [PATCH v6 0/3] input: Add support for ECI (multimedia) accessories
by tapio.vihuri＠nokia.com 03 Feb '11

03 Feb '11

From: Tapio Vihuri <tapio.vihuri(a)nokia.com> Thank you for comments. ECI input driver changes: - Add debuging config option with adequate "help" in Kconfig - Clean out debug messages - Clean out error messages - Force all buttons up event when unplugging accessory ECI controller driver changes: - Remove unused debuging config option Created against linux-2.6.38-rc2 at: 70d1f36 Merge branch 'for-linus' of git://git.infradead.org/ubi-2.6 Please review. --- v5 message --- Thank you for comments. ECI input driver changes: - Use sparse_keymap_setup() properly, remove unneeded input set_bits ECI controller driver changes: - Added "button" debugfs entry for input driver buttons handling testing --- v4 message --- Thank you for comments. ECI input driver changes: - Use driver name in debugfs - reject accessory stating more than maximum 31 buttons - clean out couple comments ECI controller driver changes: - name change, from ecibus to eci_xxx_ctrl - location change, from drivers/ecibus to drivers/input/misc - major rework from platform driver to i2c-driver - removed mrst platform dependency Checkpatch warn about Kconfig, but I think it is false positive, and I will report it to the maintainer. Platform code changes: - medfield_ prefix changed to mrst_ - added ECI controller needed power control - remover ECI controller platform driver stuff - added i2c stuff for ECI controller driver Created against linux-2.6.37-rc8 at: a25cecc Merge branch 'next' of git://git.monstr.eu/linux-2.6-microblaze --- v3 message --- Thank you for comments. ECI input driver changes: - Makefile and Kconfig sorted alphabetically - using sparse keymap library, much cleaner solution - correct string aligment with 80 column - debugfs initialization fix - sysfs cable entry using just 0/1 - using cheaper __set_bit - insane comments fixed - remove unneeded misc device Created against linux-2.6.37-rc8 at: 989d873 Merge master.kernel.org:/home/rmk/linux-2.6-arm --- v2 message --- Thank you for comments. In series two there Kconfig dependency problem fixed, preventing compilation if X86_MRST (Intel Moorestown) or INPUT_MISC is not selected. Also created against linux-2.6.37-rc8 --- v1 message --- Hi all This patch set introduce Multimedia Headset Accessory support for Nokia phones. Technically those are known as ECI (Enhancement Control Interface) If headset has many buttons, like play, vol+, vol- etc. then it is propably ECI accessory. Among several buttons ECI accessories contains memory for storing several parameters. This ECI input driver provides the following features: - reading ECI configuration memory - ECI buttons as input events Drive is constructed as follows: - ECI accessory input driver deals with headset accessory - ECI bus control driver deals the HW transfering data to/from headset - platform data match used HW In the future accessory detection logic will be added using ALSA jack reporting. Created against linux-2.6.37-rc6 Please review. Tapio Vihuri (3): ECI: input: introduce ECI accessory input driver ECI: introducing ECI controller driver ECI: adding platform data for ECI driver arch/x86/platform/mrst/mrst.c | 48 ++ drivers/input/misc/Kconfig | 65 +++ drivers/input/misc/Makefile | 3 +- drivers/input/misc/eci.c | 949 ++++++++++++++++++++++++++++++++++++ drivers/input/misc/eci_at20_ctrl.c | 566 +++++++++++++++++++++ include/linux/input/eci.h | 179 +++++++ 6 files changed, 1809 insertions(+), 1 deletions(-) create mode 100644 drivers/input/misc/eci.c create mode 100644 drivers/input/misc/eci_at20_ctrl.c create mode 100644 include/linux/input/eci.h

3 7

[alsa-devel] WL1273 FM Radio driver...
by Matti J. Aaltonen 03 Feb '11

03 Feb '11

Hello I have been trying to get the WL1273 FM radio driver into the kernel for some time. It has been kind of difficult, one of the reasons is that I didn't realize I should have tried to involve all relevant maintainers to the discussion form the beginning (AsoC, Media and MFD). At Mark's suggestion I'm trying to reopen the discussion now. The driver consists of an MFD core and two child drivers (the audio codec and the V4L2 driver). And the question is mainly about the role of the MFD driver: the original design had the IO functions in the core. Currently the core is practically empty mainly because Mauro very strongly wanted to have “everything” in the V4L2 driver. I liked the original design because it didn't have the bug that the current MFD has: the codec can be initialized before the V4L2 part sets the IO function pointers. Having in principle equally capable interface drivers is symmetrical and esthetically pleasing:-) Also somebody could easily leave out the existing interfaces and create a completely new one based for example to sysfs or something... Having the IO in the core would also conveniently hide the physical communication layer and make it easy to change I2C to UART, which is also supported by the chip. Thanks, Matti

4 5

[alsa-devel] [PATCH] ASoC: PXA: formatting
by Sven Neumann 02 Feb '11

02 Feb '11

Signed-off-by: Sven Neumann <s.neumann(a)raumfeld.com> --- sound/soc/pxa/raumfeld.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 0fd60f4..4a4b73e 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -229,19 +229,19 @@ static struct snd_soc_dai_link raumfeld_dai[] = { { .name = "ak4104", .stream_name = "Playback", - .cpu_dai_name = "pxa-ssp-dai.1", - .codec_dai_name = "ak4104-hifi", - .platform_name = "pxa-pcm-audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "ak4104-hifi", + .platform_name = "pxa-pcm-audio", .ops = &raumfeld_ak4104_ops, - .codec_name = "ak4104-codec.0", + .codec_name = "ak4104-codec.0", }, { .name = "CS4270", .stream_name = "CS4270", - .cpu_dai_name = "pxa-ssp-dai.0", - .platform_name = "pxa-pcm-audio", - .codec_dai_name = "cs4270-hifi", - .codec_name = "cs4270-codec.0-0048", + .cpu_dai_name = "pxa-ssp-dai.0", + .platform_name = "pxa-pcm-audio", + .codec_dai_name = "cs4270-hifi", + .codec_name = "cs4270-codec.0-0048", .ops = &raumfeld_cs4270_ops, },}; -- 1.7.1

3 3

[alsa-devel] [PATCH] ARM: tegra: Add to struct harmony_audio_platform_data
by Stephen Warren 01 Feb '11

01 Feb '11

Add fields to describe all the audio-related GPIOs on Harmony, except for the codec's GPIO IRQ, which will be passed in its i2c_board_info. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Mark, can you please apply this to your tegra-arch branch, and merge that. Thanks! arch/arm/mach-tegra/include/mach/harmony_audio.h | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-tegra/include/mach/harmony_audio.h b/arch/arm/mach-tegra/include/mach/harmony_audio.h index 5c46391..af08650 100644 --- a/arch/arm/mach-tegra/include/mach/harmony_audio.h +++ b/arch/arm/mach-tegra/include/mach/harmony_audio.h @@ -16,4 +16,7 @@ struct harmony_audio_platform_data { int gpio_spkr_en; + int gpio_hp_det; + int gpio_int_mic_en; + int gpio_ext_mic_en; }; -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: Tegra: Harmony: Remove redundant !!
by Stephen Warren 01 Feb '11

01 Feb '11

gpio_set_value* should accept logic values not just 0 or 1. The WM8903 GPIO driver has been fixed to work this way, so remove the redundant !! previously required when it didn't accept values >1. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/harmony.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index d1faa63..11e2cb8 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -132,7 +132,7 @@ static int harmony_event_int_spk(struct snd_soc_dapm_widget *w, struct harmony_audio_platform_data *pdata = harmony->pdata; gpio_set_value_cansleep(pdata->gpio_spkr_en, - !!SND_SOC_DAPM_EVENT_ON(event)); + SND_SOC_DAPM_EVENT_ON(event)); return 0; } -- 1.7.1

3 2

[alsa-devel] [PATCH 1/2] ASoC: omap: rx51: Add stereo output support to audio jack
by Jarkko Nikula 01 Feb '11

01 Feb '11

Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier. This patch adds support for it and stereo output can be active when "Jack Function" == "TV-OUT" || "Headphone". As the TPA6130 can output very high volume levels the output is limited with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Just repost with Mark's ack added. Previous thread here: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-January/035312.ht… --- sound/soc/omap/Kconfig | 1 + sound/soc/omap/rx51.c | 43 +++++++++++++++++++++++++++++++++++++++---- 2 files changed, 40 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a088db6..b592298 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,6 +24,7 @@ config SND_OMAP_SOC_RX51 select OMAP_MCBSP select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X + select SND_SOC_TPA6130A2 help Say Y if you want to add support for SoC audio on Nokia RX-51 hardware. This is also known as Nokia N900 product. diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 09fb0df..251afbe 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -31,6 +31,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <plat/mcbsp.h> +#include "../codecs/tpa6130a2.h" #include <asm/mach-types.h> @@ -47,7 +48,8 @@ enum { RX51_JACK_DISABLED, - RX51_JACK_TVOUT, /* tv-out */ + RX51_JACK_TVOUT, /* tv-out with stereo output */ + RX51_JACK_HP, /* headphone: stereo output, no mic */ }; static int rx51_spk_func; @@ -57,6 +59,15 @@ static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; + int hp = 0, tvout = 0; + + switch (rx51_jack_func) { + case RX51_JACK_TVOUT: + tvout = 1; + case RX51_JACK_HP: + hp = 1; + break; + } if (rx51_spk_func) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -66,9 +77,12 @@ static void rx51_ext_control(struct snd_soc_codec *codec) snd_soc_dapm_enable_pin(dapm, "DMic"); else snd_soc_dapm_disable_pin(dapm, "DMic"); + if (hp) + snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + else + snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - gpio_set_value(RX51_TVOUT_SEL_GPIO, - rx51_jack_func == RX51_JACK_TVOUT); + gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout); snd_soc_dapm_sync(dapm); } @@ -153,6 +167,19 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, return 0; } +static int rx51_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_codec *codec = w->dapm->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + tpa6130a2_stereo_enable(codec, 1); + else + tpa6130a2_stereo_enable(codec, 0); + + return 0; +} + static int rx51_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -212,11 +239,14 @@ static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = { static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), + SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event), }; static const struct snd_soc_dapm_route audio_map[] = { {"Ext Spk", NULL, "HPLOUT"}, {"Ext Spk", NULL, "HPROUT"}, + {"Headphone Jack", NULL, "LLOUT"}, + {"Headphone Jack", NULL, "RLOUT"}, {"DMic Rate 64", NULL, "Mic Bias 2V"}, {"Mic Bias 2V", NULL, "DMic"}, @@ -224,7 +254,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; -static const char *jack_function[] = {"Off", "TV-OUT"}; +static const char *jack_function[] = {"Off", "TV-OUT", "Headphone"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), @@ -265,6 +295,11 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) /* Set up RX-51 specific audio path audio_map */ snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + err = tpa6130a2_add_controls(codec); + if (err < 0) + return err; + snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- 1.7.2.3

3 6

[alsa-devel] [PATCH 1/3] ASoC: soc_jack - add function to determine jack type
by Harsha Priya 01 Feb '11

01 Feb '11

This patch adds a helper function that determines the jack type given the mic bias value. For any codec that would give the mic bias value on a jack insertion, this function would return the type of jack based on mic bias range Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc.h | 15 +++++++++++++++ sound/soc/soc-jack.c | 18 ++++++++++++++++++ 2 files changed, 33 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ecdaef..afe98eb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -246,6 +246,19 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +enum soc_mic_bias_zones { + /* mic bias volutage range for Headphones*/ + SND_SOC_JACK_HP_START = 0, + SND_SOC_JACK_HP_END = 400, + /* mic bias volutage range for American Headset*/ + SND_SOC_JACK_AM_HS_START = 400, + SND_SOC_JACK_AM_HS_END = 650, + /* mic bias volutage range for Headset*/ + SND_SOC_JACK_HS_START = 650, + SND_SOC_JACK_HS_END = 2000, + SND_SOC_UNDEFINED, +}; + enum snd_soc_control_type { SND_SOC_CUSTOM, SND_SOC_I2C, @@ -307,6 +320,8 @@ void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb); void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, struct notifier_block *nb); +int snd_soc_jack_get_type(int mic_bias); + #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc..462965a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -186,6 +186,24 @@ void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack + * + * @mic_bias: mic bias value detected + * + * Based on the mic bias value passed, this function helps identify + * the type of jack as either SND_JACK_HEADSET or SND_JACK_HEADPHONE. + */ +int snd_soc_jack_get_type(int mic_bias) +{ + if (mic_bias >= SND_SOC_JACK_HP_START && mic_bias < SND_SOC_JACK_HS_END) + return SND_JACK_HEADSET; + else + return SND_JACK_HEADPHONE; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); + #ifdef CONFIG_GPIOLIB /* gpio detect */ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) -- 1.7.3.4

4 6