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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

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[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

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[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(-)

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[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

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