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[alsa-devel] CVS, git, or something else?
by Keith A. Milner 26 Nov '08

26 Nov '08

I've been trying to check out the latest alsa-driver source to do some work on it. I followed the instructions at: http://www.alsa-project.org/main/index.php/GIT_Server#Occasional_Developers But when I cd alsa-drivers and gitcompile, it fails: make[1]: Entering directory `/homedirs/kamilner/Development/Workspace/alsa/alsa-driver/acore' copying file alsa-kernel/core/init.c patching file init.c Hunk #2 succeeded at 280 (offset 3 lines). Hunk #3 succeeded at 306 (offset 3 lines). Hunk #4 FAILED at 568. 1 out of 4 hunks FAILED -- saving rejects to file init.c.rej make[1]: *** [init.c] Error 1 make[1]: Leaving directory `/homedirs/kamilner/Development/Workspace/alsa/alsa-driver/acore' make: *** [dep] Error 1 Do I have the right version? Am I going about this the right way? I see a lot of posts on this list talking about CVS but none mentioning git. Maybe I should be using CVS instead? -- Keith A. Milner

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[alsa-devel] Intel HDA and FEDORA 9 with 2.6.27.4 kernel
by Pilipovic, Josip 25 Nov '08

25 Nov '08

Hi All ... I am running Fedora 9 with 2.6.27.4 kernel. I have download and instaled ALSA 1.0.18. Card is EUROTECH INC CATALYST with Intel ATOM processor and Intel System Controler Hub US15W. Attached is output from alsa-info.sh and /proc/asound/card0. I have run bunch of different voice test but no sound on either output on this card. I have compile alsa library pcm test and run on this board no sound. I have activated and added some dignostic to hda controler code Here is print out from /var/log/messages Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:822: hda_codec_setup_stream: NID=0x11, stream=0x3, channel=0, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1434: azx_pcm_prepare: bufsize=0x4400, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1443: JP:azx_pcm_prepare: fifosize=0xc0 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:822: hda_codec_setup_stream: NID=0x10, stream=0x3, channel=0, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:822: hda_codec_setup_stream: NID=0x11, stream=0x3, channel=0, format=0x4011 Nov 24 18:58:55 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1457: JP:azx_pcm_trigger: fifosize=0x1 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1457: JP:azx_pcm_trigger: fifosize=0x0 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x10 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x11 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x10 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x11 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x10 Nov 24 18:58:57 localhost kernel: ALSA /home/saic/alsa-driver-1.0.18/pci/hda/hda_codec.c:834: hda_codec_cleanup_stream: NID=0x11

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[alsa-devel] via82xx: unreliable audio, strange mixer behavior
by Forest Bond 25 Nov '08

25 Nov '08

Hi, I'm using snd-via82xx on an Ubuntu 8.04 system. alsa-info.sh output is here: http://www.alsa-project.org/db/?f=1c3d5570e7b57d4d2f79b964a323dd7b4efd7e78 Here are the symptoms: Audio goes away after a semi-random interval. Sometimes it returns after `modprobe -r snd-via82xx' followed by `modprobe snd-via82xx', but sometimes it seems to take a full reboot. This looks similar to: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3D3194 Some folks think that this kind of issue is triggered by audio volume being above a certain level. I'm not entirely sure if this is true, but I think it is. One time, I apparently triggered the issue by unplugging and replugging the audio cable from the mainboard during playback. Audio can be heard even when the PCM volume level is 0 (as told by alsamixer). The volume does get louder when I turn up that mixer control, but it never goes away entirely when I turn the volume to zero. Mute works as expected. This issue is impacting a significant project for me and my company. I'm willing to spend as much time as it takes to debug, and I'm fairly technical. I can also offer a bounty of some sort, if that interests anyone. Can anyone offer any input? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.pytagsfs.org

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[alsa-devel] [PATCH] ASoC: Fix TW4030 Kconfig dependency
by arunks＠mistralsolutions.com 25 Nov '08

25 Nov '08

From: Arun KS <arunks(a)mistralsolutions.com> Fixes Kconfig dependency of TWL4030 audio codec driver with TWL4030 core driver on both overo and omap2evm boards Signed-off-by: Arun KS <arunks(a)mistralsolutions.com> --- sound/soc/omap/Kconfig | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 6c56277..0900437 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -24,7 +24,7 @@ config SND_OMAP_SOC_OSK5912 config SND_OMAP_SOC_OVERO tristate "SoC Audio support for Gumstix Overo" - depends on SND_OMAP_SOC && MACH_OVERO + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help @@ -32,7 +32,7 @@ config SND_OMAP_SOC_OVERO config SND_OMAP_SOC_OMAP2EVM tristate "SoC Audio support for OMAP2EVM board" - depends on SND_OMAP_SOC && MACH_OMAP2EVM + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP2EVM select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help -- 1.5.6.3

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[alsa-devel] [PATCH 00/13] ASoC updates
by Mark Brown 25 Nov '08

25 Nov '08

The following changes since commit 4f904735c809e44c11f57cd4f82446aac1243e0e: Mark Brown (1): ALSA: ASoC: Fix typo in snd_soc_card update documentation are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Arun KS (1): ASoC: Fix TWL4030 Kconfig dependency Dmitry Baryshkov (1): ASoC: tosa: move gpio probing to machine callbacks Jarkko Nikula (2): ASoC: OMAP: Apply channel constrains to N810 machine driver ASoC: OMAP: Add support for mono audio links in McBSP DAI Mark Brown (4): ASoC: Use supplied DAI for WM9713 rather than substream ASoC: Remove DAI type information ASoC: Flag AD1980 as an AC97 interface ASoC: Lower priority of resume work logging Misael Lopez Cruz (1): ASoC: Add support for TI SDP3430 Peter Ujfalusi (4): ASoC: TWL4030: Disable soft-volume ASoC: TWL4030: Change the Master volume control to TLV ASoC: TWL4030: Add CGAIN volume control ASoC: TWL4030: Add helper function for output gain controls include/sound/soc-dai.h | 2 +- include/sound/soc.h | 8 -- sound/soc/atmel/atmel_ssc_dai.c | 3 - sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 1 - sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/pcm3008.c | 1 - sound/soc/codecs/twl4030.c | 180 +++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 18 +--- sound/soc/davinci/davinci-i2s.c | 1 - sound/soc/fsl/mpc5200_psc_i2s.c | 1 - sound/soc/omap/Kconfig | 11 ++- sound/soc/omap/Makefile | 3 +- sound/soc/omap/n810.c | 4 + sound/soc/omap/omap-mcbsp.c | 24 +++-- sound/soc/omap/sdp3430.c | 152 ++++++++++++++++++++++++++++++ sound/soc/pxa/pxa-ssp.c | 4 - sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/pxa/pxa2xx-i2s.c | 1 - sound/soc/pxa/tosa.c | 30 +++++-- sound/soc/s3c24xx/neo1973_wm8753.c | 1 - sound/soc/s3c24xx/s3c2412-i2s.c | 1 - sound/soc/s3c24xx/s3c2443-ac97.c | 4 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 1 - sound/soc/sh/hac.c | 4 +- sound/soc/sh/ssi.c | 2 - sound/soc/soc-core.c | 39 +++------ 31 files changed, 410 insertions(+), 102 deletions(-) create mode 100644 sound/soc/omap/sdp3430.c

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[alsa-devel] [PATCH 2/2] sound/soc/soc-core.c: cleanup kernel-doc
by Qinghuang Feng 25 Nov '08

25 Nov '08

There is no argument named @clk_id in snd_soc_dai_set_fmt, remove its' comment. Signed-off-by: Qinghuang Feng <qhfeng.kernel(a)gmail.com> --- diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16c7453..81ee4ba 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1805,7 +1805,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @clk_id: DAI specific clock ID * @fmt: SND_SOC_DAIFMT_ format value. * * Configures the DAI hardware format and clocking.

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[alsa-devel] [PATCH] [ASOC] tosa: move gpio probing to machine callbacks
by Dmitry Baryshkov 25 Nov '08

25 Nov '08

Signed-off-by: Dmitry Baryshkov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/tosa.c | 30 ++++++++++++++++++++++-------- 1 files changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index afefe41..69ca79b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -230,10 +230,32 @@ static struct snd_soc_dai_link tosa_dai[] = { }, }; +static int tosa_probe(struct platform_device *dev) +{ + int ret; + + ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); + if (ret) + return ret; + ret = gpio_direction_output(TOSA_GPIO_L_MUTE, 0); + if (ret) + gpio_free(TOSA_GPIO_L_MUTE); + + return ret; +} + +static int tosa_remove(struct platform_device *dev) +{ + gpio_free(TOSA_GPIO_L_MUTE); + return 0; +} + static struct snd_soc_machine tosa = { .name = "Tosa", .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), + .probe = tosa_probe, + .remove = tosa_remove, }; static struct snd_soc_device tosa_snd_devdata = { @@ -251,11 +273,6 @@ static int __init tosa_init(void) if (!machine_is_tosa()) return -ENODEV; - ret = gpio_request(TOSA_GPIO_L_MUTE, "Headphone Jack"); - if (ret) - return ret; - gpio_direction_output(TOSA_GPIO_L_MUTE, 0); - tosa_snd_device = platform_device_alloc("soc-audio", -1); if (!tosa_snd_device) { ret = -ENOMEM; @@ -272,15 +289,12 @@ static int __init tosa_init(void) platform_device_put(tosa_snd_device); err_alloc: - gpio_free(TOSA_GPIO_L_MUTE); - return ret; } static void __exit tosa_exit(void) { platform_device_unregister(tosa_snd_device); - gpio_free(TOSA_GPIO_L_MUTE); } module_init(tosa_init); -- 1.5.6.5

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[alsa-devel] [PATCH] ALSA: SoC: Add support for TI SDP3430
by Misael Lopez Cruz 25 Nov '08

25 Nov '08

This patch add ASoC support for TI SDP3430. It's based on Gumstix Overo SoC code by Steve Sakoman. Signed-off-by: Misael Lopez Cruz <mesak82(a)gmail.com> --- sound/soc/omap/Kconfig | 7 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp3430.c | 152 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/sdp3430.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 0daeee4..e816fa4 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -38,4 +38,11 @@ config SND_OMAP_SOC_OVERO help Say Y if you want to add support for SoC audio on the Gumstix Overo. +config SND_OMAP_SOC_SDP3430 + tristate "SoC Audio support for Texas Instruments SDP3430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 4bae404..64e560b 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -10,9 +10,11 @@ snd-soc-n810-objs := n810.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o +snd-soc-sdp3430-objs := sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o +obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c new file mode 100644 index 0000000..85fd160 --- /dev/null +++ b/sound/soc/omap/sdp3430.c @@ -0,0 +1,152 @@ +/* + * sdp3430.c -- SoC audio for TI OMAP3430 SDP + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * Based on: + * Author: Steve Sakoman <steve(a)sakoman.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 <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int sdp3430_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; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "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) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops sdp3430_ops = { + .hw_params = sdp3430_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp3430_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &sdp3430_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_machine snd_soc_machine_sdp3430 = { + .name = "SDP3430", + .dai_link = &sdp3430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp3430_snd_devdata = { + .machine = &snd_soc_machine_sdp3430, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *sdp3430_snd_device; + +static int __init sdp3430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_3430sdp()) { + pr_debug("Not SDP3430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP3430 SoC init\n"); + + sdp3430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp3430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp3430_snd_device, &sdp3430_snd_devdata); + sdp3430_snd_devdata.dev = &sdp3430_snd_device->dev; + *(unsigned int *)sdp3430_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(sdp3430_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp3430_snd_device); + + return ret; +} +module_init(sdp3430_soc_init); + +static void __exit sdp3430_soc_exit(void) +{ + platform_device_unregister(sdp3430_snd_device); +} +module_exit(sdp3430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP3430"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

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[alsa-devel] [topic/asoc][PATCH 0/2] ASoC: OMAP: Add support for mono audio links in McBSP DAI (take #2)
by Jarkko Nikula 25 Nov '08

25 Nov '08

Hi Second version adding support for mono audio links in OMAP McBSP DAI. At this time, the channels_min is defined to 1 in omap-mcbsp.c and no hacks in machine drivers wanting to use mono link are required. If channel restrictions are required, then machine driver can use snd_pcm_hw_constraint_minmax. Now only N810 machine driver is affected since all other OMAP machine drivers are using TWL4030 or TLV320AIC23 codecs which all claim to support only 2 channels so the link remains always stereo even playing mono audio. -- Jarkko

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[alsa-devel] pcm_prepare, and pcm_trigger
by John Kacur 25 Nov '08

25 Nov '08

More ASoC fundamentals questions. when doing cat 8k8bitpcm.wav > /dev/dsp I'm seeing neither myplatform_pcm_prepare, nor myplatform_pcm_trigger called. Is this normal? Who calls these functions? Thanks

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