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

[alsa-devel] [PATCH] ASoC: TWL4030: Add supply for audio serial interface control
by Peter Ujfalusi 11 Feb '10

11 Feb '10

The serial interface (TDM/I2S) for the audio block have been constantly enabled. Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so the interface is only enabled, when there is a need for it. For example when only the analog loopback is enabled, there is no need to keep the serial interface active. I have added the persons who contributed to the Voice path of twl4030 codec driver, so they might have the ability to test this patch, and send an update for the Voice path, if it is necessary Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- This patch has been tested in OPTION 1 mode, where the Voice path is not in use. I have looked for dependencies in the Voice path regarding to the Audio interface enable bit, but I have not found any. If someone could test, if the Voice path also needs the AIF_EN bit set, than please send a patch, which adds this new supply to the DAPM rule for the Voice path. sound/soc/codecs/twl4030.c | 14 +++++++++++++- 1 files changed, 13 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b32aeb3..277862e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -55,7 +55,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x0c, /* REG_ATXR1PGA (0xB) */ 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ - 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_AUDIO_IF (0xE) */ 0x00, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ @@ -1203,6 +1203,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1337,6 +1339,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Digital R1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L1 Playback Mixer", NULL, "AIF Enable"}, + {"Digital R2 Playback Mixer", NULL, "AIF Enable"}, + {"Digital L2 Playback Mixer", NULL, "AIF Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1455,6 +1462,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "APLL Enable"}, {"ADC Virtual Right2", NULL, "APLL Enable"}, + {"ADC Virtual Left1", NULL, "AIF Enable"}, + {"ADC Virtual Right1", NULL, "AIF Enable"}, + {"ADC Virtual Left2", NULL, "AIF Enable"}, + {"ADC Virtual Right2", NULL, "AIF Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, -- 1.6.6.1

4 5

[alsa-devel] [RESEND][PATCHv2 3/4] Add ASoC support for Devkit8000
by Thomas Weber 11 Feb '10

11 Feb '10

This patch expands the omap3beagle sound soc for the beagle board clone DevKit8000. Signed-off-by: Thomas Weber <weber(a)corscience.de> --- sound/soc/omap/Kconfig | 8 +++++--- sound/soc/omap/omap3beagle.c | 6 +++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..18ebdc7 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -94,12 +94,14 @@ config SND_OMAP_SOC_OMAP3_PANDORA Say Y if you want to add support for SoC audio on the OMAP3 Pandora. config SND_OMAP_SOC_OMAP3_BEAGLE - tristate "SoC Audio support for OMAP3 Beagle" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle and Devkit8000" + depends on TWL4030_CORE && SND_OMAP_SOC + depends on (MACH_OMAP3_BEAGLE || MACH_DEVKIT8000) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Beagleboard. + Say Y if you want to add support for SoC audio on the Beagleboard or + the clone Devkit8000. config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index d88ad5c..240e097 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -117,11 +117,11 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!machine_is_omap3_beagle()) { - pr_debug("Not OMAP3 Beagle!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { + pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); return -ENODEV; } - pr_info("OMAP3 Beagle SoC init\n"); + pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); if (!omap3beagle_snd_device) { -- 1.6.6.1

4 3

[alsa-devel] Report : targa-dig and channels on GT725
by gens.guillaume＠free.fr 11 Feb '10

11 Feb '10

Hello, I have to report a problem about my GT725 laptop. I use the model : targa-dig. Connect a headphone to my laptop doesn't mute integrated speakers. I can do it manually by muting channels (Center, LFE) to have the sound only on my headphones. I use the alsa-driver (alsa-driver-20100211) from Takashi Iwai. I have the same symptomes that the stock kernel (from Archlinux). PS: and the microphone doesn't seem to work. See my alsa-info on attached file.

1 0

[alsa-devel] [PATCHv3 3/4] Add ASoC support for DevKit8000
by Thomas Weber 11 Feb '10

11 Feb '10

This patch expands the omap3beagle sound soc for the beagle board clone Devkit8000. Change log V2 Expand omap3beagle sound soc for DevKit8000 support. V3 Separate Kconfig for Devkit8000. Signed-off-by: Thomas Weber <swirl(a)gmx.li> --- sound/soc/omap/Kconfig | 8 ++++++++ sound/soc/omap/Makefile | 1 + sound/soc/omap/omap3beagle.c | 6 +++--- 3 files changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..8520b75 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -101,6 +101,14 @@ config SND_OMAP_SOC_OMAP3_BEAGLE help Say Y if you want to add support for SoC audio on the Beagleboard. +config SND_OMAP_SOC_DEVKIT8000 + tristate "SoC Audio support for DevKit8000" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_DEVKIT8000 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the DevKit8000 + config SND_OMAP_SOC_ZOOM2 tristate "SoC Audio support for Zoom2" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 19283e5..c8195c9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -29,5 +29,6 @@ obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_DEVKIT8000) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index d88ad5c..240e097 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -117,11 +117,11 @@ static int __init omap3beagle_soc_init(void) { int ret; - if (!machine_is_omap3_beagle()) { - pr_debug("Not OMAP3 Beagle!\n"); + if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) { + pr_debug("Not OMAP3 Beagle or Devkit8000!\n"); return -ENODEV; } - pr_info("OMAP3 Beagle SoC init\n"); + pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); if (!omap3beagle_snd_device) { -- 1.6.4.4

5 7

[alsa-devel] Sound distortion with Realtek ALC889A
by Damien Brossard 11 Feb '10

11 Feb '10

Hi. I recently bought a Gigabyte GA-MA78GM-S2H motherboard. This board uses the (in)famous (from what i've read for now) alc889a chip. While playing 5.1 content, I can hear noise or distortion coming from the center channel. I can not test my configuration with another OS, so I can't fully dismiss a HW problem. However, i made several tests with 5.1 and 2.0 speakers which points me to soundcard problem. I've made some tests with speaker-test (speaker-test -D surround51 -c 6) : 1) Each time a sound is played either on front left channel or rear left channel, a noise can be heared on the center channel (and subwoofer, this is confirmed pluging a stereo headphone on the center/lfe jack). No noise when a sound is played on right channels (front or rear). 2) In alsamixer, if I set the volume to 0 for the front or rear left channels, i can hear the noise on center/lfe when a sound is "played" on this channels 3) In alsamixer, if I mute the front or rear left channels, i can hear no more noise on center/lfe Complementary tests have been made with alsaplayer and using a simple mp3 file. In this configuration (stereo output), center and rear channels play the same sound as front channel. 1) If all channels are unmuted, I hear no noise. 2) If either the front channel or the rear channel are muted, I can hear the noise on center channel. I tried this with alsa-driver 1.0.20 and 1.0.21 with the same result, I recently went back to 1.0.20 due to kernel upgrade. Things that seem strange to me (maybe not so strange to alsa dev :)) : 1) alsamixer shows volumes in percentage, however when I set a channel volume to 50% and the master volume to 50%, I hope to find a resulting volume of 25%, not 0. 2) lspci reports my soundcard subsystem id as 1458:a022, but in /proc/asound/card0/codec#0 the subsystem id is 1458:a002 Googling around made me find few references (in alsa-devel list) to a similar problem : http://mailman.alsa-project.org/pipermail/alsa-devel/2008-July/008942.html http://mailman.alsa-project.org/pipermail/alsa-devel/2008-August/009754.html The distortion / noise described here is exactly what i can hear with my system. Apparently the user solved the issue buying another soundcard. I'd like to be sure it is the only way to go. Attached is the file produced by alsa-info.sh Thanks in advance. Best regards Damien Brossard

2 3

[alsa-devel] Wierdness in alsa-driver configure script
by Eliot Blennerhassett 11 Feb '10

11 Feb '10

checking for kernel version... 2.6.28-17-generic checking for GCC version... ./configure: eval: line 5540: syntax error near unexpected token `)' ./configure: eval: line 5540: `my_compiler_version=4.3.3-5ubuntu4)' Kernel compiler: Used compiler: gcc (Ubuntu 4.3.3-5ubuntu4) 4.3.3 gcc --version gives (yes, it is ugly!): gcc (Ubuntu 4.3.3-5ubuntu4) 4.3.3 Copyright (C) 2008 Free Software Foundation, Inc. The following change works for this particular case, but perhaps there is a better answer? --- a/configure.in +++ b/configure.in @@ -363,7 +363,7 @@ AC_DEFUN([CHECK_COMPILER], [ ;; [[0-9]].[[0-9]]*) if test -z "$ac_compiler_version"; then - ac_compiler_version="$i" + ac_compiler_version=`echo $i | sed -e 's|[[^0-9]]*$[[0-9.]]\+$.*$|\1|'` fi ;; esac

2 1

[alsa-devel] unpadded USB MIDI commands
by Willem Jan Palenstijn 11 Feb '10

11 Feb '10

Hi, I recently investigated some trouble with sending MIDI via a USB MIDI cable (15ca:1806) to a Roland MT-32. In some cases, commands were being dropped. (This is what is reported on https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752 . I'm the colleague mentioned by druon in note 0021737 there.) After we experimented a bit, it turned out that packets of 2 MIDI commands (8 bytes total) were being sent to the USB MIDI cable, and that if the first of these was a 2-byte MIDI command (such as a 0xC0 or 0xD0 command), the second command in the packet would get dropped. A workaround is to limit the packet size to 4 bytes, as is already happening on at least one other device (the ESI M4U referred to in usbmidi.c). I played with it some more this week, and it turns out that if the zero-padding from the USB MIDI commands is removed (violating the specs, as far as I understand), the second command does actually not get dropped. Example: When sending the two commands 0xD1 0x10 and 0x91 0x40 0x40 normally, it would produce: sending packet: [ 0d d1 10 00 09 91 40 40 ] When routing the MIDI output through the MIDI THRU port on the MT-32 to a MIDI port on an ES1371, the captured stream is: d1 10 and no note is played on the MT-32. (The 0x91 command is a note-on.) After hacking out the zero padding, it looks like this: sending packet: [ 0d d1 10 09 91 40 40 ] The re-captured stream is: d1 10 91 40 40, and a note is properly played. What do you think would be the right way to handle this problem? Don't zero-pad the outgoing packets for this specific device (and maybe a few others mentioned on bug #3752, if they end up suffering from the same bug)? Limit the outgoing packets to 4 bytes like with the ESI M4U for these devices? Add an option to the snd-usb-audio module to toggle this behaviour? Something else? Best regards, Willem Jan

2 1

[alsa-devel] Realtek ALC275
by Jason A. Donenfeld 11 Feb '10

11 Feb '10

Hello All, The new Vaio F series has a Realtek ALC 275, which is currently setup in patch_realtek.c to use the 269 configuration. Unfortunately this does not correctly enable the internal microphone, and the line-in is not tied to any mixer control. The attached file will enable the internal microphone, but will do so at the expense of the line-in. According to Takashi, "The problem is that the digital-mic of pin 0x12 has the exclusive connection to the selector 0x22, followed to ADC 0x11, while other analog input pins are connected to 0x23/0x24, to ADC 0x08/0x09. Thus, the driver requires the dynamic switching of ADCs, which isn't fully guaranteed to work. A similar situation is found for Conexant codecs, and a same style hack would be needed for the realtek codec." This bug is being tracked here: http://code.google.com/p/vaio-f11-linux/issues/detail?id=2 and here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4909 . I'm wondering if anyone has any advice on how to implement the necessary dynamic ADC switching. Thanks, Jason Donenfeld

1 0

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 10 Feb '10

10 Feb '10

The following changes since commit c50749de02f272be6e09b9016e13a17307d29066: Grazvydas Ignotas (1): ASoC: pandora: Add DAC regulator support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (1): ASoC: cs4270: enable regulators at probe time Mark Brown (1): ASoC: add phycore-ac97 sound support Paul Menzel (1): ASoC: Typo. s/Freecale/Freescale/ Peter Ujfalusi (1): ASoC: TWL4030: Add supply for audio serial interface control sound/soc/codecs/cs4270.c | 10 +++++ sound/soc/codecs/twl4030.c | 14 ++++++- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/Makefile | 2 + sound/soc/imx/phycore-ac97.c | 90 ++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 116 insertions(+), 2 deletions(-) create mode 100644 sound/soc/imx/phycore-ac97.c

2 1

[alsa-devel] [PATCH] ASoC: Typo. s/Freecale/Freescale/
by Paul Menzel 10 Feb '10

10 Feb '10

Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 5f006f0..c7d0fd9 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,5 +1,5 @@ config SND_IMX_SOC - tristate "SoC Audio for Freecale i.MX CPUs" + tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC && BROKEN select SND_PCM select FIQ -- 1.6.6.1

2 2