December 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [patch] alsa-utils fix for intl support
by Hans-Christian Egtvedt 08 Dec '08

08 Dec '08

Hi, Feel free to comment on the included patch. I had to add it because aseqnet would not link without -lintl. -- Best regards, Hans-Christian Egtvedt

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[alsa-devel] [patch] alsa-utils check if __USE_BSD is defined before compiling "BSD functions"
by Hans-Christian Egtvedt 08 Dec '08

08 Dec '08

Hi, Another bug/issue I tripped over when compiling alsa-utils in an environment using uClibc to supply the C library functions. Here I have enabled some old BSD style functions. The attached patch will honor them if they are enabled. Without this patch I get a redefined error during compile. -- Best regards, Hans-Christian Egtvedt

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[alsa-devel] [PATCH 0/7] ASoC updates
by Mark Brown 08 Dec '08

08 Dec '08

The following changes since commit 682d5874f3d654b5d13d9b8dd56b9e05cfadd01b: Takashi Iwai (1): ALSA: ASoC: Fix old-style trigger callback in s3c2443-ac97.c are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (4): ASoC: Rename snd_soc_register_card() to snd_soc_init_card() ASoC: Annotate core removal function ASoC: Remove in-code changelog from AD73311 driver ASoC: Don't free static data in WM9713 Peter Ujfalusi (3): ASoC: TWL4030: Change the capture volume control to TLV ASoC: TWL4030: Change the common playback volume controls ASoC: TWL4030: Add volume controls for outputs include/sound/soc.h | 2 +- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 5 +-- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/twl4030.c | 74 +++++++++++++++++++++++++++++++++------ sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 3 +- sound/soc/soc-core.c | 8 ++-- 27 files changed, 91 insertions(+), 45 deletions(-)

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[alsa-devel] HDA: CLFE DAC on ALC889 (Acer Aspire 8930G)
by Hector Martin 07 Dec '08

07 Dec '08

I've been trying to get ALSA working with this laptop. So far the attached patch adds rear support to the existing 4930g model, and tries to add CLFE support. However, I haven't been able to get CLFE to work. I don't seem to get any audio through DACs/mixers other than the first two (front and rear). I can assign the CLFE output (pin 0x16) to the front or rear DAC and hear the front audio through the CLFE speakers. I can also force the front DAC (0x02) or rear DAC (0x03) to use the CLFE stream and hear it through the front or rear speakers, so the audio is getting to the codec. But either DACs 0x04 and 0x05 aren't working, or their associated mixers (0x0e and 0x0f) aren't passing through the audio, or the OUT pins don't like selections above 1. I've checked that they're unmuted and that the volumes are turned up on both. I've also tried the 0x25->0x26 DAC/mixer auxiliary pathway, which doesn't work either. Here's what the relevant NIDs correspond to: 0x02 - (normally) Front DAC 0x03 - (normally) Rear DAC 0x04 - (normally) CLFE DAC 0x05 - (presumably) side DAC (no side speakers) 0x0c - (normally) Front mixer 0x0d - (normally) Rear mixer 0x0e - (normally) CLFE mixer 0x0f - (presumably) side mixer 0x14 - Front speakers pin 0x15 - Headphones pin (also EAPD for speaker amp) 0x16 - CLFE speakers pin 0x1b - Rear speakers pin Here's what the codec graph looks like: http://marcansoft.com/transf/codec.png Any ideas? I'm pretty stumped here. Are there any other magic bits to enable the remaining DACs or mixers? -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc

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Re: [alsa-devel] [Swfdec] Can't use Mozilla plugin or play video with 0.8.2 / 0.9.2
by Benjamin Otte 07 Dec '08

07 Dec '08

I think I know what this is. But I spent the best part of my breakfast wondering about it. I think using Pulse Audio might work around this problem, but I think it's a driver, configuration issue on the ALSA side or a misunderstanding of the API on my part, which is why I'm sending this to the ALSA list. So here's the problem: Swfdec uses ALSA in non-blocking mode and polls its file descriptor(s). Whenever the audio queue becomes empty enough that it's useful to render more audio, the file descriptor wakes up from the poll. This setting is apparently configurable using various hw params, however, Swfdec does not configure it and has been relying on the fact that the sound device picks proper defaults, thinking that a sound card with a small buffer will likely want to wake up more often than something like Pulse Audio's new design, which keeps a 2+ second buffer for the audio. It seems though that your sound card wakes up roughly whenever a new sample can be rendered, which is 44100 times per second (or more, as from your profile it looks like ALSA resamples the audio). This causes the whole rendering code inside Swfdec to run for every sample, which is so much overhead that it kills all your performance. My question to you ALSA people now is: Who's responsible for ensuring that the number of wakeups is sane? Should I do some configuring of the audio device and if so, what kind of configuration, or should ALSA be able to pick reasonable defaults for me? Cheers, Benjamin On Sun, Dec 7, 2008 at 2:46 AM, Jesse Zbikowski <embeddedlinuxguy(a)gmail.com> wrote: > Hi Riccardo, > > On Fri, Dec 5, 2008 at 2:32 PM, Riccardo Magliocchetti > <riccardo.magliocchetti(a)gmail.com> wrote: >> Give it a run with oprofile, this is my opreport > > Thanks for this data. It looks like I may have an audio bottleneck. > opreport -t3 shows > > 31.5811 no-vmlinux > 14.4701 libswfdec-0.9.so.3.0.0 > 8.3210 libasound_module_rate_speexrate.so > 6.0088 libc-2.8.90.so > 4.7060 libpixman-1.so.0.12.0 > 3.6615 python2.5 > 3.2831 libcairo.so.2.10800.0 > > If I just look at the libraries I see the top libswfdec routines are > all audio related: > > 14.1497 libswfdec-0.9.so.3.0.0 swfdec_sound_buffer_get_n_samples > 2.1666 libswfdec-0.9.so.3.0.0 swfdec_audio_format_get_channels > 1.9369 libswfdec-0.9.so.3.0.0 swfdec_audio_format_new > 1.7904 libswfdec-0.9.so.3.0.0 swfdec_audio_flv_render > 1.3205 libswfdec-0.9.so.3.0.0 __i686.get_pc_thunk.bx > 1.2073 libswfdec-0.9.so.3.0.0 swfdec_audio_format_get_granularity > > I could try using the PulseAudio backend instead of ALSA. > >> You can try to improve your performance by using latest stable version of >> cairo and pixman > > OK, I am currently using the latest Ubuntu versions which are a bit > behind latest stable. I will see if this makes a difference. > > Package Ibex Latest > xorg-xserver 1.5.2 1.5.3 > cairo 1.8.0 1.8.4 > pixman 0.12 0.13.2 > >> play a bit with your X video driver config ie. exa vs xaa. > > I did not see libexa.so in my oprofile output although it is > installed, so I will make sure this is enabled. > > Thanks again, > Jesse Z. > _______________________________________________ > Swfdec mailing list > Swfdec(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/swfdec >

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Re: [alsa-devel] Status of the aloop loopback driver
by Takashi Iwai 05 Dec '08

05 Dec '08

At Fri, 5 Dec 2008 17:11:07 +0000, Andy Whitcroft wrote: > > I am interesting in finding out the status of, and plans for the aloop > loopback driver. I note that this has been about for a while and has > not been pushed up to the mainline kernel. Are you intending on > pushing that up or is it planning on being in the alsa-drivers tarball > for the forseeable future? It's still buggy, unfortunately, due to the current design. The timing of capture and playback isn't synchronized, so you'll get often noisy outputs. I've worked on this driver yesterday, just did some clean up works. Hopefully I'll be able to fix the issue sometime later... thanks, Takashi

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[alsa-devel] [PATCH] tlv320aic3x: control additions and cleanups
by Daniel Mack 05 Dec '08

05 Dec '08

Some minor updates for sound/soc/codecs/tlv320aic3x.c: - removed "DAC" from name for "Mono playback switch", "HP playback switch" and "HPCOM playback switch". The registers used in the macro don't have anything to do with the DAC but mute the whole output, hence the names were misleading. - split "Line Playback Switch" into "LineL Playback Switch" and "LineR Playback Switch" - split "Line PGA Bypass Playback Volume" into "LineL Left PGA Bypass Playback Volume" and "LineR Right PGA Bypass Playback Volume" - split "Line Line2 Bypass Playback Volume" into "LineL Line2 Bypass Playback Volume" and "LineR Line2 Bypass Playback Volume" - Added "HP Right PGA Bypass Playback Volume" Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Signed-off-by: Uli Luckas <u.luckas(a)road.de> --- sound/soc/codecs/tlv320aic3x.c | 24 ++++++++++++++++-------- 1 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 537bbbf..7cdaea3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -253,16 +253,22 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, - 0x01, 0), - SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - PGAR_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, + DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, + 0, 0x7f, 1), + SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), - SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, @@ -270,8 +276,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, + PGAR_2_HPROUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, @@ -281,7 +289,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, 0, 0x7f, 1), -- 1.5.6.5

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[alsa-devel] [PATCH] tlv320aic3x.c: controller naming fixes
by Daniel Mack 05 Dec '08

05 Dec '08

- removed "DAC" from name for "Mono playback switch", "HP playback switch" and "HPCOM playback switch". The registers used in the macro don't have anything to do with the DAC but mute the whole output, hence the names were misleading. - renamed "HPL PGA Bypass Playback Volume" to "HPL Left PGA Bypass Playback Volume" - renamed "HPR PGA Bypass Playback Volume" to "HPR Left PGA Bypass Playback Volume" Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- This is the second part of the patch I sent yesterday, plus the naming fixes recently discussed on the list. sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a12dc3..081f7c4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -268,7 +268,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), - SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, @@ -276,20 +276,20 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + SOC_SINGLE("HPL Left PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, 0, 0x7f, 1), - SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, + SOC_SINGLE("HPR Left PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, 0, 0x7f, 1), -- 1.5.6.5

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[alsa-devel] [PATCH] ASoC: Add support for OMAP3 Pandora
by Grazvydas Ignotas 05 Dec '08

05 Dec '08

This patch adds basic support for OMAP3 Pandora. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 311 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 321 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index da39f27..a7b1d77 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -46,3 +46,11 @@ config SND_OMAP_SOC_SDP3430 help Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 29cf3a8..76fedd9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -11,9 +11,11 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c new file mode 100644 index 0000000..bd91594 --- /dev/null +++ b/sound/soc/omap/omap3pandora.c @@ -0,0 +1,311 @@ +/* + * omap3pandora.c -- SoC audio for Pandora Handheld Console + * + * Author: Gražvydas Ignotas <notasas(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define OMAP3_PANDORA_DAC_POWER_GPIO 118 +#define OMAP3_PANDORA_AMP_POWER_GPIO 14 + +#define PREFIX "ASoC omap3pandora: " + +static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + pr_err(PREFIX "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + } else { + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + mdelay(1); + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + } + + return 0; +} + +/* + * Audio paths on Pandora board: + * + * |O| ---> PCM DAC +-> AMP -> Headphone Jack + * |M| A +--------> Line Out + * |A| <~~clk~~+ + * |P| <--- TWL4030 <--------- Line In and MICs + */ +static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { + SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, + 0, 0, NULL, 0, omap3pandora_hp_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), +}; + +static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (external)", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"Headphone Amplifier", NULL, "PCM DAC"}, + {"Line Out", NULL, "PCM DAC"}, + {"Headphone Jack", NULL, "Headphone Amplifier"}, +}; + +static const struct snd_soc_dapm_route omap3pandora_in_map[] = { + {"INL", NULL, "Line In"}, + {"INR", NULL, "Line In"}, + {"INL", NULL, "Mic (Internal)"}, + {"INR", NULL, "Mic (external)"}, +}; + +static int omap3pandora_out_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + ARRAY_SIZE(omap3pandora_out_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + ARRAY_SIZE(omap3pandora_out_map)); + + return snd_soc_dapm_sync(codec); +} + +static int omap3pandora_in_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ARRAY_SIZE(omap3pandora_in_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + ARRAY_SIZE(omap3pandora_in_map)); + + return snd_soc_dapm_sync(codec); +} + +static struct snd_soc_ops omap3pandora_out_ops = { + .hw_params = omap3pandora_out_hw_params, +}; + +static struct snd_soc_ops omap3pandora_in_ops = { + .hw_params = omap3pandora_in_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3pandora_dai[] = { + { + .name = "PCM1773", + .stream_name = "HiFi Out", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_out_ops, + .init = omap3pandora_out_init, + }, { + .name = "TWL4030", + .stream_name = "Line/Mic In", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_in_ops, + .init = omap3pandora_in_init, + } +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_omap3pandora = { + .name = "omap3pandora", + .platform = &omap_soc_platform, + .dai_link = omap3pandora_dai, + .num_links = ARRAY_SIZE(omap3pandora_dai), +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3pandora_snd_data = { + .card = &snd_soc_card_omap3pandora, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3pandora_snd_device; + +static int __init omap3pandora_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_pandora()) { + pr_debug(PREFIX "Not OMAP3 Pandora\n"); + return -ENODEV; + } + pr_info("OMAP3 Pandora SoC init\n"); + + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); + if (ret) { + pr_err(PREFIX "Failed to get DAC power GPIO\n"); + return ret; + } + + ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set DAC power GPIO direction\n"); + goto fail0; + } + + ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power"); + if (ret) { + pr_err(PREFIX "Failed to get amp power GPIO\n"); + goto fail0; + } + + ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set amp power GPIO direction\n"); + goto fail1; + } + + omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); + if (omap3pandora_snd_device == NULL) { + pr_err(PREFIX "Platform device allocation failed\n"); + ret = -ENOMEM; + goto fail1; + } + + platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data); + omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev; + *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ + *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */ + + ret = platform_device_add(omap3pandora_snd_device); + if (ret) { + pr_err(PREFIX "Unable to add platform device\n"); + goto fail2; + } + + return 0; + +fail2: + platform_device_put(omap3pandora_snd_device); +fail1: + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); +fail0: + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + return ret; +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_device_unregister(omap3pandora_snd_device); + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); +} +module_exit(omap3pandora_soc_exit); + +MODULE_AUTHOR("Grazvydas Ignotas <notasas(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 3

[alsa-devel] [PATCH] ASoC: Add support for OMAP3 Pandora
by Grazvydas Ignotas 04 Dec '08

04 Dec '08

This patch adds basic support for OMAP3 Pandora. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- I'm not sure if splitting DAPM and widgets was good idea, I can adjust if needed. The map is also not quite right, it says headphone jack is connected to TWL4030 codec OUTL and OUTR, but that is not really the case. It is connected to external DAC, and DAC is connected to OMAP. However DAC uses clock from TWL4030, so TWL needs to be active. I've basically done that connection to get event callback, where I can switch DAC and headphone amp on/off. sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 293 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 303 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index da39f27..a7b1d77 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -46,3 +46,11 @@ config SND_OMAP_SOC_SDP3430 help Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 29cf3a8..76fedd9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -11,9 +11,11 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c new file mode 100644 index 0000000..4f92877 --- /dev/null +++ b/sound/soc/omap/omap3pandora.c @@ -0,0 +1,293 @@ +/* + * omap3pandora.c -- SoC audio for Pandora Handheld Console + * + * Author: Gražvydas Ignotas <notasas(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define OMAP3_PANDORA_DAC_POWER_GPIO 118 +#define OMAP3_PANDORA_AMP_POWER_GPIO 14 + +#define PREFIX "ASoC omap3pandora: " + +static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + pr_err(PREFIX "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, + SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, + SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); + + return 0; +} + +/* Machine DAPM widgets */ +static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", omap3pandora_hp_event), +}; + +static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (external)", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"Headphone Jack", NULL, "OUTL"}, + {"Headphone Jack", NULL, "OUTR"}, +}; + +static const struct snd_soc_dapm_route omap3pandora_in_map[] = { + {"INL", NULL, "Line In"}, + {"INR", NULL, "Line In"}, + {"INL", NULL, "Mic (Internal)"}, + {"INR", NULL, "Mic (external)"}, +}; + +static int omap3pandora_out_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + ARRAY_SIZE(omap3pandora_out_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + ARRAY_SIZE(omap3pandora_out_map)); + + return snd_soc_dapm_sync(codec); +} + +static int omap3pandora_in_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ARRAY_SIZE(omap3pandora_in_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + ARRAY_SIZE(omap3pandora_in_map)); + + return snd_soc_dapm_sync(codec); +} + +static struct snd_soc_ops omap3pandora_out_ops = { + .hw_params = omap3pandora_out_hw_params, +}; + +static struct snd_soc_ops omap3pandora_in_ops = { + .hw_params = omap3pandora_in_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3pandora_dai[] = { + { + .name = "PCM1773", + .stream_name = "HiFi Out", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_out_ops, + .init = omap3pandora_out_init, + }, { + .name = "TWL4030", + .stream_name = "Line/Mic In", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_in_ops, + .init = omap3pandora_in_init, + } +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_omap3pandora = { + .name = "omap3pandora", + .dai_link = omap3pandora_dai, + .num_links = ARRAY_SIZE(omap3pandora_dai), +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3pandora_snd_data = { + .card = &snd_soc_card_omap3pandora, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3pandora_snd_device; + +static int __init omap3pandora_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_pandora()) { + pr_debug(PREFIX "Not OMAP3 Pandora\n"); + return -ENODEV; + } + pr_info("OMAP3 Pandora SoC init\n"); + + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); + if (ret) { + pr_err(PREFIX "Failed to get DAC power GPIO\n"); + return ret; + } + + ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set DAC power GPIO direction\n"); + goto fail0; + } + + ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power"); + if (ret) { + pr_err(PREFIX "Failed to get amp power GPIO\n"); + goto fail0; + } + + ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set amp power GPIO direction\n"); + goto fail1; + } + + omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); + if (omap3pandora_snd_device == NULL) { + pr_err(PREFIX "Platform device allocation failed\n"); + ret = -ENOMEM; + goto fail1; + } + + platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data); + omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev; + *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ + *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */ + + ret = platform_device_add(omap3pandora_snd_device); + if (ret) { + printk(KERN_ERR "Unable to add platform device\n"); + goto fail2; + } + + return 0; + +fail2: + platform_device_put(omap3pandora_snd_device); +fail1: + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); +fail0: + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + return ret; +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_device_unregister(omap3pandora_snd_device); + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); +} +module_exit(omap3pandora_soc_exit); + +MODULE_AUTHOR("Grazvydas Ignotas <notasas(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 1

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 04 Dec '08

04 Dec '08

The following changes since commit 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2: Daniel Mack (1): ASoC: tlv320aic3x: headset/button press support are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (4): ASoC: Push debugfs files out of the snd_soc_device structure ASoC: Remove device from platform suspend and resume operations ASoC: Remove platform device from DAI suspend and resume operations ASoC: Remove obsolete declaration of struct snd_soc_clock_info include/sound/soc-dai.h | 6 +-- include/sound/soc.h | 15 +++---- sound/soc/atmel/atmel-pcm.c | 6 +-- sound/soc/atmel/atmel_ssc_dai.c | 6 +-- sound/soc/au1x/psc-ac97.c | 6 +-- sound/soc/au1x/psc-i2s.c | 6 +-- sound/soc/blackfin/bf5xx-ac97.c | 6 +-- sound/soc/blackfin/bf5xx-i2s.c | 6 +-- sound/soc/pxa/pxa-ssp.c | 6 +-- sound/soc/pxa/pxa2xx-ac97.c | 6 +-- sound/soc/pxa/pxa2xx-i2s.c | 6 +-- sound/soc/s3c24xx/s3c2412-i2s.c | 16 +++---- sound/soc/s3c24xx/s3c24xx-i2s.c | 6 +-- sound/soc/soc-core.c | 85 +++++++++++++++++++++----------------- 14 files changed, 83 insertions(+), 99 deletions(-)

2 5

[alsa-devel] question for the all-knowing - firewire audio hardware
by S. Aguinaga 04 Dec '08

04 Dec '08

Hello! I have a couple of machines connected to FireWire Motu devices (running OS X). We want to develop some new apps on linux and the only info I've found is: http://www.olafchrist.de/ieee1394/index.php, but its rather old and single device specific. Are there any other solutions available? Or are there other devices (firewire) that are supported by the ALSA project? Any info on this will be great!! // Sal

2 3

Re: [alsa-devel] [PATCH - patch_nvhdmi.c 1/1] Resubmiited with full name. Added id for MCP67. Please note that I'm not really sure what I'm doing but this works for me :-)
by Takashi Iwai 04 Dec '08

04 Dec '08

At Wed, 3 Dec 2008 21:11:22 +0000, Scott Waye wrote: > > > Signed-off-by: Scott Waye <scott(a)waye.co.uk> Applied now. Thanks. Takashi > > diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c > index 1360d54..b78c8f2 100644 > --- a/pci/hda/patch_nvhdmi.c > +++ b/pci/hda/patch_nvhdmi.c > @@ -162,5 +162,6 @@ static int patch_nvhdmi(struct hda_codec *codec) > struct hda_codec_preset snd_hda_preset_nvhdmi[] = { > { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, > { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, > + { .id = 0x10de0067, .name = "NVIDIA MCP67 HDMI", .patch = patch_nvhdmi }, > {} /* terminator */ > }; > -- > 1.5.6 >

1 0

[alsa-devel] [PATCH] ALSA: Reduce stall detection timeout in riptide.c
by nokos＠gmx.net 03 Dec '08

03 Dec '08

Reduce the command timeout to 0.5sec. Should be enough to allow a working command interface but removes a RCU stall and slow resume on some revisions where the AC97 revision detection stalls in resume. Signed-off-by: Peter Gruber <nokos(a)gmx.net> --- diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e9f0706..d0ccfc6 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -172,7 +172,7 @@ #define MAX_WRITE_RETRY 10 /* cmd interface limits */ #define MAX_ERROR_COUNT 10 -#define CMDIF_TIMEOUT 500000 +#define CMDIF_TIMEOUT 50000 #define RESET_TRIES 5 #define READ_PORT_ULONG(p) inl((unsigned long)&(p))

2 1

[alsa-devel] [PATCH 0/5] ASoC updates
by Mark Brown 03 Dec '08

03 Dec '08

The following changes since commit 0ecfe7987855d21c2a89ffe003ddf0ee11b42d47: Mark Brown (1): ASoC: Don't free static data in WM9713 are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Daniel Mack (1): ASoC: tlv320aic3x: headset/button press support Grazvydas Ignotas (1): ASoC: TWL4030: Add input selection and gain controls Mark Brown (3): ASoC: Push workqueue data into snd_soc_card ASoC: Push platform registration down into the card ASoC: Fix WM8903 right mixer bypass path include/sound/soc.h | 9 +- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 +- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 31 ++++++- sound/soc/codecs/tlv320aic3x.h | 46 +++++++++- sound/soc/codecs/twl4030.c | 177 +++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 16 +++ sound/soc/codecs/wm8903.c | 8 +- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/soc-of-simple.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/e800_wm9712.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/palm27x.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/soc-core.c | 77 +++++++++------ 36 files changed, 348 insertions(+), 74 deletions(-)

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Re: [alsa-devel] Gateway MX3042 Laptop. No sound
by Takashi Iwai 03 Dec '08

03 Dec '08

At Wed, 3 Dec 2008 06:39:24 -0600, Michael Satterwhite wrote: > > On Wednesday 03 December 2008 01:02:07 am you wrote: > > At Wed, 3 Dec 2008 00:12:54 +0100, > > > > Nigel Henry wrote: > > > I'm trying to help someone on the Kubuntu list, using Hardy Heron 8.04, > > > which uses alsa driver 1.0.16. This is with a Gateway MX3042 Laptop, but > > > no sounds. > > > > > > The output from running alsa-info.sh is below. > > > > Try my sound.git tree and sound-unstable.git tree. > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git > > > > The snapshot tarballs are found in > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > > I'll be happy to (and *DO* thank you for your help!). I'm not sure what this > means, however. Are these drivers, a kernel ... how do I try them? It's a linux kernel git tree. Ask other Ubuntu guys how to build your kernel. Or, use the alsa-driver snapshot tarball in the ftp directory above. You can install it just like the normal alsa-driver release tarball. Takashi

1 0

[alsa-devel] [PATCH] tlv320aic3x: headset/button press support
by Daniel Mack 03 Dec '08

03 Dec '08

Hi, This patch adds a function to setup headset and button press detection modes. Also, aic3x_headset_detected() returns the value from the current state's register rather than the interrupt cause flag so it works without an interrupt as well. Best regards, Daniel

3 5

[alsa-devel] [PATCH] ASoC: TWL4030: Add input selection and gain controls
by Grazvydas Ignotas 03 Dec '08

03 Dec '08

The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- Compared to my previous patch, I've changed "Microphone Boost" name to "Input Boost Volume", as it affects line inputs too. I thought that hiding this control for line-in was not a good idea, as it might still be useful for weak sources. "Capture" was also changed to "Input", as inputs can be also routed to outputs, not only recorded. sound/soc/codecs/twl4030.c | 177 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 16 ++++ 2 files changed, 190 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ffd5120..3c9fdf2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -46,9 +46,9 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ - 0x0a, /* REG_AVADC_CTL (0x7) */ + 0x20, /* REG_ANAMICL (0x5) */ + 0x00, /* REG_ANAMICR (0x6) */ + 0x00, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ 0x0c, /* REG_ATXL1PGA (0xA) */ @@ -347,6 +347,162 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +static int twl4030_get_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + int result = 0; + + /* one bit must be set a time */ + reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN; + if (reg != 0) { + result++; + while ((reg & 1) == 0) { + result++; + reg >>= 1; + } + } + + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int twl4030_put_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicl, micbias, avadc_ctl; + + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN); + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicl |= TWL4030_MAINMIC_EN; + micbias |= TWL4030_MICBIAS1_EN; + break; + case 2: + anamicl |= TWL4030_HSMIC_EN; + micbias |= TWL4030_HSMICBIAS_EN; + break; + case 3: + anamicl |= TWL4030_AUXL_EN; + break; + case 4: + anamicl |= TWL4030_CKMIC_EN; + break; + default: + break; + } + + /* If some input is selected, enable amp and ADC */ + if (value != 0) { + anamicl |= TWL4030_MICAMPL_EN; + avadc_ctl |= TWL4030_ADCL_EN; + } else { + anamicl &= ~TWL4030_MICAMPL_EN; + avadc_ctl &= ~TWL4030_ADCL_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static int twl4030_get_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + int value = 0; + + reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; + switch (reg) { + case TWL4030_SUBMIC_EN: + value = 1; + break; + case TWL4030_AUXR_EN: + value = 2; + break; + default: + break; + } + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int twl4030_put_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicr, micbias, avadc_ctl; + + anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~TWL4030_MICBIAS2_EN; + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicr |= TWL4030_SUBMIC_EN; + micbias |= TWL4030_MICBIAS2_EN; + break; + case 2: + anamicr |= TWL4030_AUXR_EN; + break; + default: + break; + } + + if (value != 0) { + anamicr |= TWL4030_MICAMPR_EN; + avadc_ctl |= TWL4030_ADCR_EN; + } else { + anamicr &= ~TWL4030_MICAMPR_EN; + avadc_ctl &= ~TWL4030_ADCR_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static const char *twl4030_left_in_sel[] = { + "None", + "Main Mic", + "Headset Mic", + "Line In", + "Carkit Mic", +}; + +static const char *twl4030_right_in_sel[] = { + "None", + "Sub Mic", + "Line In", +}; + +static const struct soc_enum twl4030_left_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel), + twl4030_left_in_sel); + +static const struct soc_enum twl4030_right_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel), + twl4030_right_in_sel); + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -378,6 +534,12 @@ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); */ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); +/* + * Gain control for input amplifiers + * 0 dB to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -420,6 +582,15 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0, digital_capture_tlv), + + SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN, + 0, 3, 5, 0, input_gain_tlv), + + /* Input source controls */ + SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux, + twl4030_get_left_input, twl4030_put_left_input), + SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux, + twl4030_get_right_input, twl4030_put_right_input), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 09865d9..a2065d4 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,7 +113,16 @@ #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ + +#define TWL4030_MICBIAS2_CTL 0x40 +#define TWL4030_MICBIAS1_CTL 0x20 +#define TWL4030_HSMICBIAS_EN 0x04 +#define TWL4030_MICBIAS2_EN 0x02 +#define TWL4030_MICBIAS1_EN 0x01 + /* ANAMICL (0x05) Fields */ + #define TWL4030_CNCL_OFFSET_START 0x80 #define TWL4030_OFFSET_CNCL_SEL 0x60 #define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 @@ -127,10 +136,17 @@ #define TWL4030_MAINMIC_EN 0x01 /* ANAMICR (0x06) Fields */ + #define TWL4030_MICAMPR_EN 0x10 #define TWL4030_AUXR_EN 0x04 #define TWL4030_SUBMIC_EN 0x01 +/* AVADC_CTL (0x07) Fields */ + +#define TWL4030_ADCL_EN 0x08 +#define TWL4030_AVADC_CLK_PRIORITY 0x04 +#define TWL4030_ADCR_EN 0x02 + /* AUDIO_IF (0x0E) Fields */ #define TWL4030_AIF_SLAVE_EN 0x80 -- 1.5.4.3

3 3

[alsa-devel] Gateway MX3042 Laptop. No sound
by Nigel Henry 03 Dec '08

03 Dec '08

I'm trying to help someone on the Kubuntu list, using Hardy Heron 8.04, which uses alsa driver 1.0.16. This is with a Gateway MX3042 Laptop, but no sounds. The output from running alsa-info.sh is below. !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Tue Dec 2 16:30:06 CST 2008 !!Linux Distribution !!------------------ Ubuntu 8.04.1 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 8.04.1" !!Kernel Information !!------------------ Kernel release: 2.6.24-22-generic Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.16 Library version: Utilities version: 1.0.15 !!Loaded ALSA modules !!------------------- snd_intel8x0 !!Soundcards recognised by ALSA !!----------------------------- 0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4 Intel 82801DB-ICH4 with AD1981B at irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1f.5 0401: 8086:24c5 (rev 03) Subsystem: 161f:203e !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 !!Loaded sound module options !!-------------------------- !!Module: snd_intel8x0 ac97_clock : 0 ac97_quirk : <NULL> buggy_irq : N buggy_semaphore : N enable : N id : <NULL> index : -1 joystick : 0 spdif_aclink : 0 xbox : N !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: Analog Devices AD1981B PCI Subsys Vendor: 0x161f PCI Subsys Device: 0x203e Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 16-bit 3D enhancement : No 3D Stereo Enhancement Current setup Mic gain : 20dB [ 20dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Extended ID : codec=0 rev=1 AMAP DSA=0 SPDIF VRA Extended status : SPCV SPDIF=10/11 SPDIF VRA PCM front DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Category=0x2 Generation=1 Rate=48kHz AD18XX configuration Unchained : 0x1000,0x0000,0x0000 Chained : 0x0000,0x0000,0x0000 0:00 = 0090 0:02 = 0000 0:04 = 0000 0:06 = 0000 0:08 = 0000 0:0a = 0000 0:0c = 001f 0:0e = 805f 0:10 = 0808 0:12 = 0606 0:14 = 0000 0:16 = 1f1f 0:18 = 0000 0:1a = 0505 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 800f 0:28 = 0605 0:2a = 0435 0:2c = bb80 0:2e = 0000 0:30 = 0000 0:32 = bb80 0:34 = 0000 0:36 = 0000 0:38 = 0000 0:3a = 2824 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 4000 0:5e = 0000 0:60 = 8080 0:62 = 0000 0:64 = 8000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0000 0:6e = 0000 0:70 = 0000 0:72 = 1800 0:74 = 1001 0:76 = 2050 0:78 = 0000 0:7a = 0000 0:7c = 4144 0:7e = 5374 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 2008-12-02 11:09 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 24 2008-12-02 11:09 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 2008-12-02 14:27 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 25 2008-12-02 11:09 /dev/snd/pcmC0D1c crw-rw---- 1 root audio 116, 26 2008-12-02 11:09 /dev/snd/pcmC0D2c crw-rw---- 1 root audio 116, 27 2008-12-02 11:09 /dev/snd/pcmC0D3c crw-rw---- 1 root audio 116, 20 2008-12-02 11:09 /dev/snd/pcmC0D4p crw-rw---- 1 root audio 116, 1 2008-12-02 11:09 /dev/snd/seq crw-rw---- 1 root audio 116, 33 2008-12-02 11:09 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 1: Intel ICH - MIC ADC [Intel 82801DB-ICH4 - MIC ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 2: Intel ICH - MIC2 ADC [Intel 82801DB-ICH4 - MIC2 ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 3: Intel ICH - ADC2 [Intel 82801DB-ICH4 - ADC2] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [I82801DBICH4] Card hw:0 'I82801DBICH4'/'Intel 82801DB-ICH4 with AD1981B at irq 17' Mixer name : 'Analog Devices AD1981B' Components : 'AC97a:41445374' Controls : 34 Simple ctrls : 23 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Headphone Jack Sense',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 23 [74%] [0.00dB] [on] Capture [off] Front Right: Playback 23 [74%] [0.00dB] [on] Capture [off] Simple mixer control 'Line Jack Sense',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 25 [81%] [3.00dB] [on] Capture [off] Front Right: Playback 25 [81%] [3.00dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mic Boost ( 20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Playback AC97-SPSA',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 3 [100%] Simple mixer control 'IEC958 Playback Source',0 Capabilities: enum Items: 'AC-Link' 'A/D Converter' Item0: 'AC-Link' Simple mixer control 'Aux',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [on] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [on] Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 0 [0%] [0.00dB] [on] Front Right: Capture 0 [0%] [0.00dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output !!------------- --startcollapse-- state.I82801DBICH4 { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Master Playback Switch' value.0 true value.1 true } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Master Playback Volume' value.0 31 value.1 31 } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Headphone Playback Volume' value.0 31 value.1 31 } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Mono Playback Switch' value true } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Master Mono Playback Volume' value 31 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Playback Switch' value true } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Phone Playback Volume' value 0 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Playback Switch' value false } control.10 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Mic Playback Volume' value 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost ( 20dB)' value true } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 true value.1 true } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Line Playback Volume' value.0 23 value.1 23 } control.14 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'CD Playback Switch' value.0 true value.1 true } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'CD Playback Volume' value.0 25 value.1 25 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Aux Playback Switch' value.0 true value.1 true } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Aux Playback Volume' value.0 0 value.1 0 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'PCM Playback Switch' value.0 true value.1 true } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'PCM Playback Volume' value.0 31 value.1 31 } control.20 { comment.access 'read write' comment.type ENUMERATED comment.count 2 comment.item.0 Mic comment.item.1 CD comment.item.2 Video comment.item.3 Aux comment.item.4 Line comment.item.5 Mix comment.item.6 'Mix Mono' comment.item.7 Phone iface MIXER name 'Capture Source' value.0 Mix value.1 Mix } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.22 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.23 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mix comment.item.1 Mic iface MIXER name 'Mono Output Select' value Mix } control.24 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic1 } control.25 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.26 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value cf00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.27 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.28 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.29 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 3' iface MIXER name 'IEC958 Playback AC97-SPSA' value 3 } control.30 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 AC-Link comment.item.1 'A/D Converter' iface MIXER name 'IEC958 Playback Source' value AC-Link } control.31 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Stereo Mic' value true } control.32 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Headphone Jack Sense' value true } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Jack Sense' value true } control.34 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module nls_iso8859_1 nls_cp437 vfat fat usb_storage libusual ipv6 af_packet rfcomm l2cap bluetooth rfkill_input i915 drm ppdev cpufreq_userspace cpufreq_stats cpufreq_powersave cpufreq_conservative cpufreq_ondemand freq_table dock container sbs sbshc iptable_filter ip_tables x_tables aes_i586 dm_crypt dm_mod sbp2 parport_pc lp parport joydev tifm_sd mmc_core pcmcia arc4 ecb blkcipher b43 rfkill mac80211 cfg80211 led_class input_polldev snd_intel8x0 snd_ac97_codec ac97_bus snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy video output evdev psmouse tifm_7xx1 serio_raw snd_seq_oss tifm_core yenta_socket rsrc_nonstatic snd_seq_midi pcmcia_core snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device battery ac button pcspkr snd soundcore iTCO_wdt iTCO_vendor_support snd_page_alloc intel_agp shpchp pci_hotplug agpgart ext3 jbd mbcache sg sr_mod cdrom sd_mod ata_generic usbhid hid b44 ata_piix ohci1394 pata_acpi ssb mii ieee1394 libata scsi_mod ehci_hcd uhci_hcd usbcore thermal processor fan fbcon tileblit font bitblit softcursor fuse Any help in getting Michael's sounds working, greatly appreciated. Nigel.

2 1

[alsa-devel] [PATCH - patch_nvhdmi.c 1/1] Added id for MCP67. Please note that I'm not really sure what I'm doing but this works for me :-)
by scott＠waye.co.uk 03 Dec '08

03 Dec '08

From: root <scott(a)waye.co.uk> Signed-off-by: root <scott(a)waye.co.uk> diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c index 1360d54..b78c8f2 100644 --- a/pci/hda/patch_nvhdmi.c +++ b/pci/hda/patch_nvhdmi.c @@ -162,5 +162,6 @@ static int patch_nvhdmi(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0067, .name = "NVIDIA MCP67 HDMI", .patch = patch_nvhdmi }, {} /* terminator */ }; -- 1.5.6

2 1

[PATCH - snd-intel-hda 1/1] Add partial 8930G support
by Hector Martin 03 Dec '08

03 Dec '08

Signed-off-by: Hector Martin <hector(a)marcansoft.com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..6d6e06a 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -1302,7 +1302,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, /* Enable unsolicited event for HP jack */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal HP to front */ +/* Connect Front speakers to front */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1310,6 +1310,16 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Rear speakers to rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect CLFE speakers to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Set EAPD to enable speakers */ + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } }; @@ -1365,8 +1375,15 @@ static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; + /* Front */ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + /* CLFE */ + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + /* Rear */ + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, -- 1.5.6.4 --------------060204020609000202040409--

1 0

[PATCH - snd-intel-hda 1/1] Add partial 8930G support
by Hector Martin 03 Dec '08

03 Dec '08

Signed-off-by: Hector Martin <hector(a)marcansoft.com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..6d6e06a 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -1302,7 +1302,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, /* Enable unsolicited event for HP jack */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal HP to front */ +/* Connect Front speakers to front */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1310,6 +1310,16 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Rear speakers to rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect CLFE speakers to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Set EAPD to enable speakers */ + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } }; @@ -1365,8 +1375,15 @@ static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; + /* Front */ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + /* CLFE */ + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + /* Rear */ + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, -- 1.5.6.4 --------------070404090902050101040901--

1 0

[alsa-devel] alsa-driver-1.0.18rc3 fails to compile against linux-2.6.27.7/arch/powerpc
by kordex - 03 Dec '08

03 Dec '08

make -C /lib/modules/2.6.27.7/sourceSUBDIRS=/root/alsa/alsa-driver-1.0.18rc3 CPP="gcc -E" CC="gcc" modules make[1]: Siirrytään hakemistoon "/usr/src/linux-2.6.27.7" CC [M] /root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.o In file included from /usr/src/linux-2.6.27.7/arch/powerpc/include/asm/system.h:528, from include/linux/list.h:7, from include/linux/module.h:9, from /root/alsa/alsa-driver-1.0.18rc3/include/adriver.h:26, from /root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.c:1: include/asm-generic/cmpxchg-local.h: In function '__cmpxchg_local_generic': include/asm-generic/cmpxchg-local.h:23: error: implicit declaration of function 'local_irq_save' include/asm-generic/cmpxchg-local.h:44: error: implicit declaration of function 'local_irq_restore' make[3]: *** [/root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.o] Virhe 1 make[2]: *** [/root/alsa/alsa-driver-1.0.18rc3/acore] Virhe 2 make[1]: *** [_module_/root/alsa/alsa-driver-1.0.18rc3] Virhe 2 make[1]: Poistutaan hakemistosta "/usr/src/linux-2.6.27.7" make: *** [compile] Virhe 2 root@powerline:/root/alsa/alsa-driver-1.0.18rc3# root@powerline:/root/alsa/alsa-driver-1.0.18rc3# I used this configure line (compiling for emu 1212m): ./configure --with-cards=emu10k1 --with-sequencer=yes --with-oss=yes Gcc version: root@powerline:/root/alsa/alsa-driver-1.0.18rc3# gcc -v Using built-in specs. Target: powerpc-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 4.3.2-1' --with-bugurl=file:///usr/share/doc/gcc-4.3/README.Bugs --enable-languages=c,c++,fortran,objc,obj-c++ --prefix=/usr --enable-shared --with-system-zlib --libexecdir=/usr/lib --without-included-gettext --enable-threads=posix --enable-nls --with-gxx-include-dir=/usr/include/c++/4.3 --program-suffix=-4.3 --enable-clocale=gnu --enable-libstdcxx-debug --enable-objc-gc --enable-mpfr --disable-softfloat --enable-secureplt --enable-targets=powerpc-linux,powerpc64-linux --with-cpu=default32 --with-long-double-128 --enable-checking=release --build=powerpc-linux-gnu --host=powerpc-linux-gnu --target=powerpc-linux-gnu Thread model: posix gcc version 4.3.2 (Debian 4.3.2-1)

1 0

[alsa-devel] Where is xfi driver now?
by Nickolay Hodyunya 02 Dec '08

02 Dec '08

Hello, I have updated to the latest git tree of sound-usntable kernel 1 day ago and now I can not find xfi drivers. Where are they go? -- Regards, Nickolay Hodyunya.

2 1

[alsa-devel] [PATCH] ALSA: ac97 - Include ac97_codec.h for ac97_bus_type declaration
by Mark Brown 02 Dec '08

02 Dec '08

This fixes a sparse warning caused by the lack of a connection with the prototype for ac97_bus_type. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/ac97_bus.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index 7fa37e1..a351dd0 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/device.h> #include <linux/string.h> +#include <sound/ac97_codec.h> /* * Let drivers decide whether they want to support given codec from their -- 1.5.6.5

2 1