June 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Takashi Iwai 13 Jun '09

13 Jun '09

At Sat, 18 Apr 2009 21:36:35 -0300, Gustavo Vieira wrote: > > Em Sáb, 2009-04-18 às 10:57 +0200, Takashi Iwai escreveu: > > > The jack sense is still broken, though. I'm attaching the alsa-info > > with > > > the snapshot loaded. > > > > Could you attach alsa-info.sh output at both HP plugged / unplugged > > states? > > > > Now things got really interesting. :) > > When trying to get the output you requested I discovered that running > alsa-info with the headphones plugged made them work and silenced the > speakers. Then, removing the HP do not make the sound come back to the > speakers, but running alsa-info again does. That is, I can force the > jack sense by running alsa-info. > > I'm attaching the following files: > * alsa-info-snapshot.txt: The output when the system is turned on, > without HP. When I plug the HP and run alsa-info, the HP work > and the out is the same as this file. > * alsa-info-snapshot-hp-1.txt: The output when HP work. > * alsa-info-snapshot-hp-out.txt: The output of the execution of > alsa-info that makes the speakers work when HP are removed. Hm, what happens if you pass model=hp-dv5 option? Takashi

3 13

[alsa-devel] [PATCH] ASoC: Zoom2: Add machine driver for Zoom2 board
by Lopez Cruz, Misael 12 Jun '09

12 Jun '09

Add support for Zoom2 board. Zoom2 machine driver connects both codec DAIs (audio and voice) to omap3 McBSP ports in the following way: HiFi <-> McBSP2 Voice <-> McBSP3 The zoom2 driver has the following DAPM widgets: * Ext Mic: MAINMIC, SUBMIC (with bias) * Ext Spk: HFL, HFR * Headset Stereophone: HSOL, HSOR * Headset Mic: HSMIC (with bias) * Aux In: AUXL, AUXR Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/omap/Kconfig | 7 + sound/soc/omap/Makefile | 2 + sound/soc/omap/zoom2.c | 301 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 310 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/zoom2.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 675732e..397f3d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -64,4 +64,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE help Say Y if you want to add support for SoC audio on the Beagleboard. +config SND_OMAP_SOC_ZOOM2 + tristate "SoC Audio support for Zoom2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on Zoom2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 0c9e4ac..5016840 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -13,6 +13,7 @@ snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -21,3 +22,4 @@ obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c new file mode 100644 index 0000000..be2e307 --- /dev/null +++ b/sound/soc/omap/zoom2.c @@ -0,0 +1,301 @@ +/* + * zoom2.c -- SoC audio for Zoom2 + * + * Author: Misael Lopez Cruz <x0052729(a)ti.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" + +#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) + +static int zoom2_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 zoom2_ops = { + .hw_params = zoom2_hw_params, +}; + +static int zoom2_hw_voice_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_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFM); + if (ret) { + 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_DSP_A | + SND_SOC_DAIFMT_IB_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 zoom2_voice_ops = { + .hw_params = zoom2_hw_voice_params, +}; + +/* Zoom2 machine DAPM */ +static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Aux In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 1", NULL, "Ext Mic"}, + {"Mic Bias 2", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Stereophone: HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Aux In: AUXL, AUXR */ + {"Aux In", NULL, "AUXL"}, + {"Aux In", NULL, "AUXR"}, +}; + +static int zoom2_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add Zoom2 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets, + ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up Zoom2 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Zoom2 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(codec, "Aux In"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Enable voice interface */ + reg = codec->read(codec, TWL4030_REG_VOICE_IF); + reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; + codec->write(codec, TWL4030_REG_VOICE_IF, reg); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link zoom2_dai[] = { + { + .name = "TWL4030 I2S", + .stream_name = "TWL4030 Audio", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .init = zoom2_twl4030_init, + .ops = &zoom2_ops, + }, + { + .name = "TWL4030 PCM", + .stream_name = "TWL4030 Voice", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE], + .init = zoom2_twl4030_voice_init, + .ops = &zoom2_voice_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_zoom2 = { + .name = "Zoom2", + .platform = &omap_soc_platform, + .dai_link = zoom2_dai, + .num_links = ARRAY_SIZE(zoom2_dai), +}; + +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 2, /* 81 ms */ + .sysclk = 26000, +}; + +/* Audio subsystem */ +static struct snd_soc_device zoom2_snd_devdata = { + .card = &snd_soc_zoom2, + .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, +}; + +static struct platform_device *zoom2_snd_device; + +static int __init zoom2_soc_init(void) +{ + int ret; + + if (!machine_is_omap_zoom2()) { + pr_debug("Not Zoom2!\n"); + return -ENODEV; + } + printk(KERN_INFO "Zoom2 SoC init\n"); + + zoom2_snd_device = platform_device_alloc("soc-audio", -1); + if (!zoom2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata); + zoom2_snd_devdata.dev = &zoom2_snd_device->dev; + *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + + ret = platform_device_add(zoom2_snd_device); + if (ret) + goto err1; + + BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); + gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(zoom2_snd_device); + + return ret; +} +module_init(zoom2_soc_init); + +static void __exit zoom2_soc_exit(void) +{ + gpio_free(ZOOM2_HEADSET_MUX_GPIO); + + platform_device_unregister(zoom2_snd_device); +} +module_exit(zoom2_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC Zoom2"); +MODULE_LICENSE("GPL"); + -- 1.5.4.3

2 1

[alsa-devel] snd-ctxfi surround volume
by The Source 12 Jun '09

12 Jun '09

Hello again. I have another minor problem with this driver. When I use hw:0,1 (surround) device for output, the volume even with slider on maximum is much lower than through front (hw:0,0) device. Can you do something about it? Thank you.

2 1

Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 12 Jun '09

12 Jun '09

Looking at the board, it is model SB1040, with CA0110-IBG LF chip. Is that an hda type chip? > -------- Original Message -------- > Subject: Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted: > From: Takashi Iwai <tiwai(a)suse.de> > Date: Thu, June 11, 2009 6:54 pm > To: bill(a)tellytopia.com > Cc: "Gernot Kohlhaas" <gernot.kohlhaas(a)googlemail.com>, > alsa-devel(a)alsa-project.org > At Thu, 11 Jun 2009 14:28:02 -0700, > bill(a)tellytopia.com wrote: > > > > Yes, I have loaded the module snd-ctfxi many times, > > and unloaded snd-hda-intl. > > Also blacklisted snd-hda-intel, and put snd-ctxfi into modprobe.conf. > Are you sure that your board has really emu20k1 or emu20k2 chip? > X-Fi isn't always with these chips. Some are HD-audio only (thus > snd-hda-intel is the right driver) and some are with CA0106 chip. > Takashi

3 2

[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 12 Jun '09

12 Jun '09

Hi, $SUBJECT is now on my sound-unstable git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git together with other experimental patches. If you're using 2.6.27-rc* git tree, pull the master branch of the tree above into yours, and run make oldconfig. That is, % cd /your/git-tree % git pull git://...../sound-unstable-2.6.git master If you are not using 2.6.27-rc*, or not familiar with git, you can try alsa-driver-unstable snapshot tarball available at ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ Run configure, make and make install as usual ALSA-driver tarball. (I didn't check whether the tarball correctly includes the sbxfi stuff. It should have been generated automatically. If not, wait for a while. If it still doesn't include sbxfi code, please report. I'll fix it tomorrow morning.) The driver is built only for 2.6.26 or later. If you have an older kernel, edit alsa-driver*/kconfig-vers and change the version of CONFIG_SND_SBXFI to 2.6.24 or whatever you want. Then run ./gitcompile, instead of configure in this case to update the configure script. **NOTE** The driver is totally untested. It's just compiled without errors, but not reviewed after a quick writing. So, don't expect it ever runs at the first try. A crash is highly possible. There are some build conditions found in sound/pci/sbxfi/sbxfi.c, starting with XXX_*. You can change it if you want. As default, it's for non-fullduplex but accept different rates. Not sure whether this works at all. Any test- (and better debugging-) reports are appreciated. thanks, Takashi

16 112

Re: [alsa-devel] plug:dmix slows playback
by Jaroslav Kysela 12 Jun '09

12 Jun '09

On Fri, 12 Jun 2009, Steve Langstaff wrote: >> From: alsa-devel-bounces(a)alsa-project.org [mailto:alsa-devel- >> bounces(a)alsa-project.org] On Behalf Of Jaroslav Kysela > >> On Fri, 12 Jun 2009, Steve Langstaff wrote: > >>> Slave: Hardware PCM card 0 'LM4549' device 0 subdevice 0 >> >>> msbits : 16 >> >> If I remember correctly, this value was 32 in your dmix setup. >> Could you try command: >> >> aplay -D"plug:'dmix:FORMAT=S16_LE'" <filename> > > Yes, that works fine - is there something wrong with my configuration do you > think? It seems that 32-bit samples are not handled correctly in driver. What's your hardware? Output from alsa-info.sh script might be helpful for identification. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

Re: [alsa-devel] plug:dmix slows playback
by Jaroslav Kysela 12 Jun '09

12 Jun '09

On Fri, 12 Jun 2009, Steve Langstaff wrote: >> From: alsa-devel-bounces(a)alsa-project.org [mailto:alsa-devel- >> bounces(a)alsa-project.org] On Behalf Of Jaroslav Kysela > >> On Fri, 12 Jun 2009, Steve Langstaff wrote: >> >>> Hi All. >>> >>> I've got an installation of ALSA (1.0.19 on an embedded device) where >> using >>> plug and dmix as my output device slows down playback - voices and >> tones are >>> lower in pitch and the playback takes longer than when not using >> plug:dmix, >>> e.g. >> >> Does 48000Hz 2 channel source work OK with direct device (without dmix) >> for your hardware? > > Yes, I converted the original sample to 48000 Hz, Stereo, and it sounds OK > when I don't use dmix. > > /mnt # time aplay -v ignition48000s.wav .... > Slave: Hardware PCM card 0 'LM4549' device 0 subdevice 0 > msbits : 16 If I remember correctly, this value was 32 in your dmix setup. Could you try command: aplay -D"plug:'dmix:FORMAT=S16_LE'" <filename> Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

Re: [alsa-devel] plug:dmix slows playback
by Jaroslav Kysela 12 Jun '09

12 Jun '09

On Fri, 12 Jun 2009, Steve Langstaff wrote: > Hi All. > > I've got an installation of ALSA (1.0.19 on an embedded device) where using > plug and dmix as my output device slows down playback - voices and tones are > lower in pitch and the playback takes longer than when not using plug:dmix, > e.g. Does 48000Hz 2 channel source work OK with direct device (without dmix) for your hardware? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

[alsa-devel] plug:dmix slows playback
by Steve Langstaff 12 Jun '09

12 Jun '09

Hi All. I've got an installation of ALSA (1.0.19 on an embedded device) where using plug and dmix as my output device slows down playback - voices and tones are lower in pitch and the playback takes longer than when not using plug:dmix, e.g. /usr/share/sounds/alsa # time aplay ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono real 0m 3.08s user 0m 0.04s sys 0m 0.07s /usr/share/sounds/alsa # time aplay -D plug:dmix ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono real 0m 6.00s user 0m 0.09s sys 0m 0.08s See how the 'real' time for playing the file goes from 3.08s to 6.00s I have also noticed that the audio is played out of both (L+R) channels when not using -D plug:dmix, but is only played out of the right channel when using -D plug:dmix. I am an ALSA beginner - any ideas where I should start looking? Using the -v option on aplay we get more output... /usr/share/sounds/alsa # aplay -v ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'LM4549' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 /usr/share/sounds/alsa # aplay -v -D plug:dmix ignition.wav Playing WAVE 'ignition.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 2720 period_size : 170 period_time : 21250 tstamp_mode : NONE period_step : 1 avail_min : 170 period_event : 0 start_threshold : 2720 stop_threshold : 2720 silence_threshold: 0 silence_size : 0 boundary : 356515840 Slave: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 16320 period_size : 1020 period_time : 21250 tstamp_mode : NONE period_step : 1 avail_min : 1020 period_event : 0 start_threshold : 16320 stop_threshold : 16320 silence_threshold: 0 silence_size : 0 boundary : 2139095040 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16320 period_size : 1020 period_time : 21250 tstamp_mode : NONE period_step : 1 avail_min : 1020 period_event : 0 start_threshold : 16320 stop_threshold : 16320 silence_threshold: 0 silence_size : 0 boundary : 2139095040 Hardware PCM card 0 'LM4549' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16320 period_size : 1020 period_time : 21250 tstamp_mode : ENABLE period_step : 1 avail_min : 1020 period_event : 0 start_threshold : 1 stop_threshold : 2139095040 silence_threshold: 0 silence_size : 2139095040 boundary : 2139095040 appl_ptr : 0 hw_ptr : 0

1 0

Re: [alsa-devel] no sound in ubuntu 8.10 on AD1986 sound card
by Daniel Chen 12 Jun '09

12 Jun '09

Your 'PCM' is muted - is that intentional? Also, unless you're explicitly using digital out, you should mute 'IEC958'. Lastly, if you're using headphones, unmute that element, too. On Jun 10, 2009 12:18 PM, "sandip gangakhedkar" <sandipfloyd(a)gmail.com> wrote: Hi all, I am facing a very perplexing issue with sound in Ubuntu 8.10 which I have never encountered before with linux. I do not hear any sound from my speakers. The sound card is AD 1986 (Analog Devices) and it worked wonderfully without any hiccups before on 8.10 and other Ubuntu versions. My alsa-info script is here: http://www.alsa-project.org/db/?f=7f34b4e28d97178d7f709740f91ff66ea08a9468 I suspect pulseaudio is the culprit: Output of : /var/log/syslog | grep pulseaudio is here: http://paste.ubuntu.com/192629/ Running : speaker-test -c2 -D hw:0,0 -t wav -l1 gives no sound. My motherboard is ASUS A8N-VM on a AMD 64 Athlon 3200 Desktop PC.The sound card is detected so no problems there. I have reinstalled ALSA, pulseaudio and all the related packages but to no avail. I am pretty sure this is not a problem of inadeuate support because sound was working before without any issues. Any ideas/suggestions on how to fix this problem will be greatly appreciated. I am also raising an email bug report at alsa-devel so anyone can canter around there and provide expert help :) Thanks, Sandip _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

1 0

[alsa-devel] snd-ctxfi oops
by The Source 12 Jun '09

12 Jun '09

Hello. I use snd-ctxfi from 8 June snapshot. I hadn't oopses since I installed this driver but recently I got one again: Kernel failure message 1: BUG: sleeping function called from invalid context at mm/slub.c:1599 in_atomic(): 0, irqs_disabled(): 1, pid: 2537, name: gstreamer-prope Pid: 2537, comm: gstreamer-prope Tainted: P 2.6.29.4-167.fc11.x86_64 #1 Call Trace: [<ffffffff8103ff0f>] __might_sleep+0x10b/0x110 [<ffffffff810cd734>] __kmalloc+0x73/0x130 [<ffffffffa0b4b142>] ? daio_rsc_init+0xaa/0x125 [snd_ctxfi] [<ffffffffa0b4b212>] dao_rsc_init+0x55/0x1c0 [snd_ctxfi] [<ffffffffa0b4b3d2>] dao_rsc_reinit+0x55/0x5d [snd_ctxfi] [<ffffffff813abd6c>] ? _spin_lock_irqsave+0x32/0x3b [<ffffffffa0b454fe>] atc_spdif_out_passthru+0x92/0x136 [snd_ctxfi] [<ffffffffa0b4744d>] ct_pcm_playback_open+0x8b/0x11a [snd_ctxfi] [<ffffffffa0b34498>] snd_pcm_open_substream+0x46/0x83 [snd_pcm] [<ffffffffa0b345a8>] snd_pcm_open+0xd3/0x1db [snd_pcm] [<ffffffff810406fd>] ? default_wake_function+0x0/0x14 [<ffffffffa0b3471a>] snd_pcm_playback_open+0x31/0x36 [snd_pcm] [<ffffffff813abe16>] ? lock_kernel+0x21/0x31 [<ffffffffa00ec57d>] snd_open+0x105/0x184 [snd] [<ffffffff810d7da3>] chrdev_open+0x152/0x171 [<ffffffff810d7c51>] ? chrdev_open+0x0/0x171 [<ffffffff810d3922>] __dentry_open+0x143/0x273 [<ffffffff810d3b22>] nameidata_to_filp+0x42/0x53 [<ffffffff810df1bd>] do_filp_open+0x3fd/0x7b8 [<ffffffff810e0f0b>] ? vfs_ioctl+0x22/0x87 [<ffffffff811b95b2>] ? might_fault+0x1f/0x21 [<ffffffff811b96b1>] ? __strncpy_from_user+0x1e/0x49 [<ffffffff810e859c>] ? alloc_fd+0x11b/0x12d [<ffffffff810d3712>] do_sys_open+0x59/0xda [<ffffffff810d37c6>] sys_open+0x20/0x22 [<ffffffff8101133a>] system_call_fastpath+0x16/0x1b

2 1

Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 12 Jun '09

12 Jun '09

Yes, I have loaded the module snd-ctfxi many times, and unloaded snd-hda-intl. Also blacklisted snd-hda-intel, and put snd-ctxfi into modprobe.conf. I get more warnings on compiling the driver than you do - see below. (the messages below are from today not from Tuesday. The driver source was downloaded this morning.) Fedora Core 8: [root@localhost alsa-driver-unstable]# uname -a Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:18:33 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux (the messages below are from today not from Tuesday. The driver source was downloaded this morning.) --------------------------- configure: creating ./config.status config.status: creating version config.status: creating Makefile.conf A couple of days ago I got this: config.status: WARNING: 'Makefile.conf.in' seems to ignore the --datarootdir setting but now this: $as_echo "$as_me: WARNING: $ac_file_inputs seems to ignore the --datarootdir setting" >&2;} config.status: creating snddevices -------------------------------------------------------------------------- These messages are from the "make": CC [M] /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctatc.o /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctatc.c: In function â€˜spdif_passthru_playback_prepareâ€™: /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctatc.c:872: warning: â€˜errâ€™ may be used uninitialized in this function CC [M] /root/Desktop/alsa-driver-unstable/pci/ctxfi/ctvmem.o .......... CC [M] /root/Desktop/alsa-driver-unstable/pci/hda/hda_eld.o In file included from /root/Desktop/alsa-driver-unstable/pci/hda/hda_eld.c:4: /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c: In function â€˜hdmi_update_eldâ€™: /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c:281: warning: passing argument 1 of â€˜get_unaligned_le64â€™ discards qualifiers from pointer target type /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c:284: warning: passing argument 1 of â€˜get_unaligned_le16â€™ discards qualifiers from pointer target type /root/Desktop/alsa-driver-unstable/pci/hda/../../alsa-kernel/pci/hda/hda_eld.c:285: warning: passing argument 1 of â€˜get_unaligned_le16â€™ discards qualifiers from pointer target type CC [M] /root/Desktop/alsa-driver-unstable/pci/hda/patch_nvhdmi.o ......... CC [M] /root/Desktop/alsa-driver-unstable/pci/riptide/riptide.o /root/Desktop/alsa-driver-unstable/pci/riptide/riptide.c: In function â€˜alsa_card_riptide_initâ€™: /root/Desktop/alsa-driver-unstable/pci/riptide/riptide.c:2206: warning: ignoring return value of â€˜pci_register_driverâ€™, declared with attribute warn_unused_result LD [M] /root/Desktop/alsa-driver-unstable/pci/riptide/snd-riptide.o ........... No errors at end: LD [M] /root/Desktop/alsa-driver-unstable/usb/usx2y/snd-usb-usx2y.ko make[1]: Leaving directory `/usr/src/kernels/2.6.23.1-42.fc8-x86_64' utils/link-modules /root/Desktop/alsa-driver-unstable ALSA modules were successfully compiled. [root@localhost alsa-driver-unstable]#

2 1

[alsa-devel] [PATCH] ASoC: remove duplicated code on pxa-ssp.c
by Daniel Ribeiro 11 Jun '09

11 Jun '09

* We don't need to write the registers twice, remove the first write. * DAIFMT_INV switch is duplicated inside DAIFMT_FORMAT switch, move it out. (This patch is for Mark's for-2.6.32 branch, I have not checked if the code is duplicated on current 2.6.30) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 50 ++++++++++++++-------------------------------- 1 files changed, 15 insertions(+), 35 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c4540..e22c5ce 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -457,31 +457,27 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; case SND_SOC_DAIFMT_DSP_A: @@ -489,22 +485,6 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; default: -- tg: (a3ba169..) asoc/remove-code-duplication (depends on: asoc/upstream-for-2.6.32) -- Daniel Ribeiro

2 1

[alsa-devel] HDA Intel IDT 92HD71B7X - HP Pavilion DV7-1299EF - Subwoofer not Working
by Wael Nasreddine (a.k.a eMxyzptlk) 11 Jun '09

11 Jun '09

Hello, Last night I bought myself a new laptop, the HP Pavilion DV7-1299EF. This laptop has 2 frontal speakers and a subwoofer in the back, on Windows, when I play a sound the 3 speakers works perfectly, and since it has a subwoofer the spearkers hardly has any bass capabilities, On linux the subwoofer is not working, only the 2 speakers are, hence the sound doesn't have any bass and it's loud and annoying, very annoying... I'm using alsa-drivers compiled from today's git checkout.. Here's the snd-hda-intel module options I tried options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=hp-m4 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=hp-dv5 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=hp-hdx options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=dell-m4-1 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=dell-m4-2 options snd-hda-intel enable=1 probe_mask=1 enable_msi=1 single_cmd=0 power_save_controller=0 power_save=0 model=dell-m4-3 Whatever the model option I try, I always end up with only 2 speakers... Here's the alsa-info.sh output -> http://www.alsa-project.org/db/?f=0fd7cb5749b12dd4322879bb3828f9391bfee487 I also attached some more informative files.. Thanks for any help, it's really appreciated -- Wael Nasreddine Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)gmail.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :.

3 2

[alsa-devel] x-fi drivers and ati hdmi sound
by Matthias Diehl 11 Jun '09

11 Jun '09

hey Takashi, i'm really looking forward to you newest x-fi "hack". i loaded the file >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… << and tried to install it but i think there are problems with my graphics adapter (ati radeon hd 4870) which has a hdmi-soundchip onboard (hda intel). i dont know how to deactivate the hdmi-soundchip on my graphics adapter. lspci | grep audio shows me when installing your driver next to the hdmi driver: 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 05:00.0 Multimedia audio controller: Creative Labs SB X-Fi but there's no device to select in the audio options. do you have any idea or help for me? kind regards from germany, matthias ____________________________________________________________ Text: GRATIS für alle WEB.DE-Nutzer: Die maxdome Movie-FLAT! Jetzt freischalten unter http://movieflat.web.de

1 0

[alsa-devel] Want to test a patch
by Chris Schumann 11 Jun '09

11 Jun '09

Hello all, I was hoping to get involved to fix als bug #305 and related ones, but I see Takashi Iwai has beaten me to it. Awesome. I'd like to test his patch of 5-27 for snd_cs46xx but I haven't done driver development before. How do I test out this patch on my hardware (ThinkPad 600X)? Thanks, Chris

2 2

[alsa-devel] [PATCH] alsa: lx6464es - support index module parameter
by Tim Blechmann 11 Jun '09

11 Jun '09

trivial patch to put the device to a specific index Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 870bfc5..dd0f12b 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -43,6 +43,9 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Digigram LX6464ES interface."); + static const char card_name[] = "LX6464ES"; -- 1.6.2.1

2 4

[alsa-devel] AC97 atmel
by Sedji Gaouaou 11 Jun '09

11 Jun '09

Hi, I have a patch which adds support for ATMEL AT91. I would like to know on which branch should I base my patch (linux ketchup or http://www.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git). Regards, Sedji

4 11

Re: [alsa-devel] SB X-Fi driver fixes (Re: Testers wanted:
by bill＠tellytopia.com 11 Jun '09

11 Jun '09

I am using the snd-ctxfi driver on a 64-bit version of fedora code 7. Is there anyone else working on a 64-bit system? There are some compile warnings during the compilation of the driver, but no errors. But no sound, either, so far. I have also taken the 1.0.20 library, utils and tools sources from the Alsa main downloads. In order to compile the utils I had to link the 64-bit sound libraries to /usr/lib/libasound.2.0.0, in other words: (cd /lib64; ln -s /usr/lib/libasound.2.0.0 libasound.2.0.0; ln -s /usr/lib/libasound.2.0.0 libasound.2) There is no /lib64/libasound.la Also had to install xmlto (yum install xmlto). So far I have added the snd-hda-intel driver to the blacklist but still the Soundcard Detection shows snd-hda-intel. Any ideas, anyone? Regards Bill

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: change set_tdm_slot api to allow slot_size override.
by Daniel Ribeiro 11 Jun '09

11 Jun '09

Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override, I will submit this as a separate patch after I figure how to handle the I2S case. (this series is meant for Mark's for-2.6.32 branch + "remove duplicated code" patch I sent yesterday) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..6692410 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1220,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..613cccb 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(7) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

2 2

[alsa-devel] [PATCH v2 2/2] ASoC: pxa-ssp.c, Automatically set TDM when needed
by Daniel Ribeiro 11 Jun '09

11 Jun '09

* Automatically sets TDM mode for frame_width larger than 32 bits, if the user doesn't setup the TDM slots with set_tdm_slot(). * Reset SSCR0_EDSS and SSCR0_DSS on pxa_ssp_set_dai_fmt. * Makes SSCR0_MOD optional. * Clears SSCR1_RWOT case SSCR0_MOD is set. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 88 ++++++++++++++++++++++++++++------------------ 1 files changed, 53 insertions(+), 35 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 613cccb..7e72c41 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -452,8 +452,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr0 = ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | + SSCR0_ACS | SSCR0_DSS | SSCR0_EDSS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -496,7 +496,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; break; @@ -532,10 +532,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; int chn = params_channels(params); - u32 sscr0; - u32 sspsp; + u32 sscr0, sscr1, sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int slot_width, frame_width = 0; + + /* check if the user explicitly set a slot_width */ + sscr0 = ssp_read_reg(ssp, SSCR0); + + if (sscr0 & (SSCR0_EDSS | SSCR0_DSS)) + slot_width = (sscr0 & SSCR0_DSS) + + (sscr0 & SSCR0_EDSS ? 17 : 1); + else + frame_width = slot_width = width * chn; /* generate correct DMA params */ if (cpu_dai->dma_data) @@ -545,35 +553,53 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), + cpu_dai->dma_data = ssp_get_dma_params(ssp, slot_width > 16, substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (sscr0 & SSCR0_SSE) return 0; - /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; + if (width == 16 && cpu_is_pxa3xx()) + sscr0 |= SSCR0_FPCKE; #endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; + + if (frame_width > 0) { + /* Not using network mode */ + if (frame_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(frame_width - 16); + else + sscr0 |= SSCR0_DataSize(frame_width); + + if (frame_width > 32) { + /* + * Network mode is needed to support this frame width. + * We assume the wire is not networked and setup a + * fake network mode here. Use as many slots as needed + * each with 32 bits. + */ + int slots = frame_width / 32; + + sscr0 |= SSCR0_MOD; + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* + * Set active slots. Only set an active TX slot + * if we are going to use it. + */ + ssp_write_reg(ssp, SSRSA, slots - 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ssp_write_reg(ssp, SSTSA, slots - 1); + } } + + /* If SSCR0_MOD is set we can't use SSCR1_RWOT */ + if (sscr0 & SSCR0_MOD) { + sscr1 = ssp_read_reg(ssp, SSCR1); + ssp_write_reg(ssp, SSCR1, sscr1 & ~SSCR1_RWOT); + } + ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -620,14 +646,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, break; } - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - dump_registers(ssp); return 0; -- tg: (5231b18..) asoc/ssp-internals (depends on: asoc/set_tdm_slot) -- Daniel Ribeiro

1 0

[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: set mixername
by Daniel Mack 11 Jun '09

11 Jun '09

alsamixer and friends want the mixername to be set. Even though the driver does not exports a real mixer device, export the name doesn't harm. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/device.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index b9a2b31..2240624 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.15"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -426,6 +426,7 @@ static int __devinit init_card(struct snd_usb_caiaqdev *dev) strlcpy(card->driver, MODNAME, sizeof(card->driver)); strlcpy(card->shortname, dev->product_name, sizeof(card->shortname)); + strlcpy(card->mixername, dev->product_name, sizeof(card->mixername)); /* if the id was not passed as module option, fill it with a shortened * version of the product string which does not contain any -- 1.6.3.1

2 1

[alsa-devel] no sound in ubuntu 8.10 on AD1986 sound card
by sandip gangakhedkar 10 Jun '09

10 Jun '09

Hi all, I am facing a very perplexing issue with sound in Ubuntu 8.10 which I have never encountered before with linux. I do not hear any sound from my speakers. The sound card is AD 1986 (Analog Devices) and it worked wonderfully without any hiccups before on 8.10 and other Ubuntu versions. My alsa-info script is here: http://www.alsa-project.org/db/?f=7f34b4e28d97178d7f709740f91ff66ea08a9468 I suspect pulseaudio is the culprit: Output of : /var/log/syslog | grep pulseaudio is here: http://paste.ubuntu.com/192629/ Running : speaker-test -c2 -D hw:0,0 -t wav -l1 gives no sound. My motherboard is ASUS A8N-VM on a AMD 64 Athlon 3200 Desktop PC.The sound card is detected so no problems there. I have reinstalled ALSA, pulseaudio and all the related packages but to no avail. I am pretty sure this is not a problem of inadeuate support because sound was working before without any issues. Any ideas/suggestions on how to fix this problem will be greatly appreciated. I am also raising an email bug report at alsa-devel so anyone can canter around there and provide expert help :) Thanks, Sandip

3 3

[alsa-devel] [PATCH] ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
by Simos Xenitellis 10 Jun '09

10 Jun '09

A quirk is required for 8086:284b (rev 03) [Subsystem: 161f:2073]. The following has been tested with Alsa 1.0.20 (git master). Background details can be found at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4561 http://forum.ubuntu-gr.org/viewtopic.php?f=38&t=5290 Tested-by: Theodora Iliopoulou <th30dr(a)gmail.com> Signed-off-by: Simos Xenitellis <simos(a)gnome.org> -- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 42f944b..104d739 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -5939,6 +5941,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 }, { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 }, { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 }, + { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 }, { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 }, { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 }, { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },

2 1

Re: [alsa-devel] Testers wanted: New SB X-Fi driver
by Takashi Iwai 10 Jun '09

10 Jun '09

At Wed, 10 Jun 2009 18:21:55 +0200, MonsieurWeller wrote: > > Thanks for reply. > (Very) last little thing: i've tried to use the x-fi external rack with > my audigy2ZS (alsa driver 1.0.20 with emu10k1). And it seems to work (i > have only tried with the headphone jack). Maybe it's the same thing? Hard to tell, but it's anyway good to know. Thanks for the information! Takashi

1 0