gmane.linux.alsa.devel

Clearification needed about structure _snd_pcm

William Estrada — 2008-09-06T20:05:34

Hi group, I have found the documentation about ALSA very confusing. Where can I find the definitions about the fields in the structure_snd_pcm? Specifically Buffer, Rate, Period, Buffer Size, Period size, etc. Basically all of these: My program is not honoring the 'packet size'.

snd_pcm_hw_param_any returns bad data

William Estrada — 2008-09-06T17:52:31

Hi group, I'm using snd_pcm_hw_param_any to retrieve the parameters from an opened device and I'm getting garbage back in the structures. My code is here: http://64.124.13.3/_ALSA_/ This is a simple test to show the problem. The output is shown in show.txt. as you can see the before ( within the open function ) is much different than the after. I am assuming that it is my code that has the problem, any ideas??? Thanks for your time.

HP tx2000 and snd-hda-intel

Kory Prince — 2008-09-06T01:05:17

Hello, I have an HP Pavillion TX2000z. This laptop has uses the snd-hda-intel and has the ALC861VD realtek codec. The existing configuration for the codec does not support the tx2000 series laptops fully. There are several configurations that have various levels of support. If you specify for the model to use the existing hp model (made for the tx1000 series) then: 1.Headphones Work 2.Speakers Work 3.Front Mic Works (the jack mic by the headphone jack) 4.The External Mics do not work (There are two mics at the top of the laptop... I am guessing left and right for a stereo Mic) 5.It seems like there is something mapped to an input that does not belong there. The simplest way to explain this is when I open audacity and set it to monitor input that there is significant white noise coming in. The Ext. Mics are not working so it can't come from there (as far as I know) and I have nothing plugged into the front Mic. I cannot find any way to mute this noise. When using the 3-stack model (with the option posit

[PATCH] hda: multiple SPDIF outputs support

Matthew Ranostay — 2008-09-05T21:45:54

Added support for outputting a stream to multiple SPDIF outs on supporting codecs. --- Signed-off-by: Matthew Ranostay embeddedalley.com> diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index 4f32911..696d77e 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c < at >< at > -1454,12 +1454,22 < at >< at > static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, codec->spdif_ctls = val; if (change) { +hda_nid_t *d; snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); + +for (d = codec->slave_dig_outs; *d; d++) { +snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); +snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_2, + val >> 8); +} } mutex_unlock(&codec->spdif_mutex); < at >< at > -1491,10 +1501,16 < at >< at > static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kc

[PATCH] hda: removed unneeded hp_nid references.

Matthew Ranostay — 2008-09-05T21:06:19

Removed unneeded hp_nid references for 92hd73xx codec family. --- Signed-off-by: Matthew Ranostay embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 6ad6ef3..5b715a7 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c < at >< at > -3775,17 +3775,14 < at >< at > again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ -spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break; case 0x4: /* 8 Channel */ -spec->multiout.hp_nid = 0x18; spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; break; case 0x5: /* 10 Channel */ -spec->multiout.hp_nid = 0x19; spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; };

[PATCH] ALSA: ASoC V2: optimize init sequence ofFreescale MPC8610 sound drivers

Timur Tabi — 2008-09-05T20:51:28

In the Freescale MPC8610 sound drivers, relocate all code from the _prepare functions into the corresponding _hw_params functions. These drivers assumed that the sample size is known in the _prepare function and not in the _hw_params function, but this is not true. Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it. Turn off snooping for DMA operations to/from I/O registers, since that's not necessary. Some comment blocks were not near the code they reference, so they were moved. Signed-off-by: Timur Tabi freescale.com> --- This patch is for ASoC V2 only. sound/soc/fsl/fsl_dma.c | 283 ++++++++++++++++++++++------------------------- sound/soc/fsl/fsl_ssi.c | 20 ++-- 2 files changed, 141 insertions(+), 162 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 5da0069..8821289 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c < at >< at > -319,9 +319,66 < at >< at > error: } /*

[PATCH] hda: SPDIF mux controls

Matthew Ranostay — 2008-09-05T18:55:02

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dynamically create mux controls for SPDIF outs on certain IDT/Sigmatel codecs. - --- Signed-off-by: Matthew Ranostay embeddedalley.com> diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 9968ee4..4637d76 100644 - --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c < at >< at > -173,6 +173,9 < at >< at > struct sigmatel_spec { unsigned int num_dmics; hda_nid_t *dmux_nids; unsigned int num_dmuxes; +hda_nid_t *smux_nids; +unsigned int num_smuxes; + hda_nid_t dig_in_nid; hda_nid_t mono_nid; hda_nid_t anabeep_nid; < at >< at > -193,6 +196,8 < at >< at > struct sigmatel_spec { unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; +struct hda_input_mux *sinput_mux; +unsigned int cur_smux[2]; unsigned int powerdown_adcs; /* i/o switches */ < at >< at > -209,6 +214,7 < at >< at > struct sigmatel_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_dimux; struct hda_input_mux private_imux; +struct hda_inpu

[PATCH 0/14] ASoC: Blackfin and I2C updats

Mark Brown — 2008-09-05T14:35:05

The following changes since commit 44e2c3045f77c69d18ba4afda888a4cdec4a33fd: Cliff Cai (1): ALSA: ASoC codec: fix compiling error in ad1980 driver after ASoC API changed are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Cliff Cai (10): ASoC codec: SSM2602 audio codec driver ASoC: Blackfin: SPORT peripheral interface driver ASoC: Blackfin: DMA Driver for AC97 sound chip ASoC: Blackfin: AC97 Blackfin CPU DAI driver ASoC: Blackfin: DMA Driver for I2S sound chip ASoC: Blackfin: I2S CPU DAI driver ASoC: Blackfin: board driver for AD1980/1 audio codec ASoC: Blackfin: board driver for SSM2602 sound chip ASoC: Blackfin: add Blackfin arch ASoC Kconfig and Makefile ASoC: Blackfin: Include Blackfin architecture support in build Jean Delvare (4): ASoC: Fix an error path in neo1973_wm8753 ASoC: Convert wm8753 to a new-style i2c driver ASoC: Convert neo1973/lm4857 to a new-style i2

[PATCH 0/9] ASoC Blackfin supporting (v2)

Bryan Wu — 2008-09-05T10:21:33

Hi Mark, With Cliff's effort, we update this patch series quickly according to your review. v1-v2: - move ASoC Blackfin Kconfig and Makefile patch to be the last one in the series - fix coding style issues - The SND_SOC_DAFIMT_LEFT_J: ought to be default - fix other minor issues v0-v1: - fix coding style issues - use latest ASoC API - split the whole patch into this 9 patches in a patchset Thanks a lot -Bryan

[PATCH 1/1] ASoC codec: SSM2602 audio codec driver (v3)

Bryan Wu — 2008-09-05T10:09:57

From: Cliff Cai analog.com> v2-v3: - add the codec to SND_SOC_ALL_CODECS - coding style fixing - rename registers' name - fix an issue with DAPM and the bias configuration. v1-v2: - coding style fixing - use pr_xxx macros to replace printk(KERN_XXX...) - use new-style i2c interface - update to use latest ASoC API Signed-off-by: Cliff Cai analog.com> Signed-off-by: Bryan Wu kernel.org> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ssm2602.c | 775 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ssm2602.h | 130 ++++++++ 4 files changed, 911 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ssm2602.c create mode 100644 sound/soc/codecs/ssm2602.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5d77dc3..2223993 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig < at >< at > -15,6 +15,7 < at >< at > config SND_SOC_ALL_CODECS select SND_SOC_CS4270 select SND_SOC_T

WM8750 mixer controls

Harald Radke — 2008-09-05T08:25:36

Hi there! I noticed that using "alsamixer" several controls that are connected to sound input (ALC) are listed in the "playback" tab as well as in "capture", as well as certain playback controls as "bass" "trebel" also can be found under "capture". However, changing those controls' values only seem to take effect in the "playback" tab. I was wondering if this is intentional or unintentional done by the WM8750 codec driver or an alsamixer "feature". Thx Greez, Harry

Making envy24control/kenvy24 work on Envy24 HT/HT-Scards

Vedran Miletić — 2008-09-05T07:17:39

Hi, any ideas if it would be possible to make envy24control or kenvy24 work on HT/HT-S cards? I'm referring to VU-meters / mixer / settings / profiles part, not the routing since those chipsets don't support it, according to VIA.

ALSA SOC VI API's for 2 PCM DEVICE ON SAME CARD

dinesh — 2008-09-05T06:48:57

Hi, Actually i am stuck with a problem i have two register 2 channels of my SOC device as 2 different pcm devices on same card. But my driver uses ALSA V1 api's and i am not able to find out any example code for this using v1 api's. (SIMILAR TO MPC5200_PSC.C) If anyone can help me please. Thanks, Dinesh

proper use of snd_mixer_selem_set_capture_switch ?

Aaron J. Grier — 2008-09-05T00:57:08

does anyone have a good example or walkthrough how to change capture sources? I'm having a heck of a time. my calls go like this: /* setup */ snd_mixer_selem_id_alloca snd_mixer_open snd_mixer_attach snd_mixer_selem_register snd_mixer_load /* set up PCM element, unused in this example */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* set up Line element (linecapture) */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* set up Mic element (miccapture) */ snd_mixer_selem_id_set_index snd_mixer_selem_id_set_name snd_mixer_find_selem /* here's that part that throws me... I'm trying to sit in a loop and * trade capture back and forth between Line and Mic. code reproduced * here verbatim... */ do { snd_mixer_selem_channel_id_t chn; for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { int junk; if (snd_mixer_selem_has_capture_channel(linecapture, chn)) {

Problems with pulseaudio / asla on PS3

Kevin Gilbert — 2008-09-04T22:42:08

Hi, I'm trying to get an jukebox application going on a Sony PlayStation3 using Qt-Embedded, Phonon and PulseAudio. The application is working well on my KDE- based laptop where it appears that the application is using Phonon -> ALSA -> PulseAudio. But there is no sound from the application when run on the PS3. I initially reported this problem to the PulseAudio mailing list. The first post on this topic is at https://tango.0pointer.de/pipermail/pulseaudio- discuss/2008-August/002249.html . In pacticular, see these posts: 1) https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008- September/002346.html where I detail the results of a final test using the latest development version of PulseAudio; and 2) https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008- September/002347.html where a PulseAudio developer diagnoses the problem as being within ALSA. For convenience, the following is from the latter post: "So, what happens here is this: your device can only do non-interleaved audio and i

are freebob firewire cards likely to be supported by alsa in the future?

andy baxter — 2008-09-04T21:46:35

Hello, Just wondering if there are any plans to develop alsa support for firewire freebob cards like the M-Audio firewire solo (which I have). I have this working fine under jackd using the freebob driver, but as far as I know there isn't an alsa driver for it. I assume that since there is already open source code to drive the chip, it should be possible to write an alsa driver? This isn't an urgent thing for me - for the moment I'm quite happy to use my onboard card for playing music / movies etc, and keep the firewire card for recording music etc, but it might be useful sometimes to have an alsa driver for this card as well. andy

Problems recording with HP DV2839TX laptop.

Travis Place — 2008-09-04T16:46:50

Hello everyone, Im Travis (wishie) once of the many helpers who man the unofficial #alsa on freenode irc network.. I usually spend my time fixing other peoples sound issues, but now i have my own. I have a HP DV2839TX laptop, and i get ALOT of static when recording via the inbuilt mic. Also, the mic controls are very strange.. I have (in the Capture section of alsamixer).. Digital, Docking Mic, External Mic, Internal Mic.. only the "Digital" control seems to effect my recording volume at all. Strangely, though, the '* Mic' controls in the PLAYBACK section of alsamixer do change the recording properties somewhat (volume being the main factor). In any event, i get a rather large amount of static, no matter what mixer settings i try. This isnt a problem in the 'other' OS. My alsa-info.sh output for alsa-driver 1.0.16 is at : http://www.alsa-project.org/db/?f=0d1df6446ff1c71ad305e7b7b367f0a6b22b80c8 My alsa-info.sh output for alsa-driver 1.0.18rc1 is at : http://www.alsa-project.org/db/?f=9e5cd99f

Playing silence instead of pausing during underrun

Timur Tabi — 2008-09-04T15:04:32

If I understand it correctly, when ALSA detects an underrun during playback, it throttles the driver by sending pause and resume commands. It turns out that our hardware doesn't really like this. We're looking at various solutions, but one proposal would be to keep the hardware running during a pause, but just have it play silence instead. That is, ALSA keeps feeding silence data during an underrun recovery. Is there an easy way to do this in ALSA? Is there another driver that does this?

DaVinci ASoC DMA stalls

Mark Lokowich — 2008-09-04T13:39:26

I've been using the DaVinci ASoC for a few months and have recently upgraded to the 2.6.26 based DaVinci git kernel. Now my audio DMA stalls more readily after stopping an active stream. I can manually trigger the event by poking the ESR to reactivate the stalled stream, suggesting the problem is in the ASP-to-DMA XEVT interface. This problem is less prevalent in the 2.6.25 based kernel. Any help? Regards, Mark Lokowich Advanced Communication Design

Audio buffer mmap-ing on sh arch vs. cache aliasingproblem

Pawel MOLL — 2008-09-04T11:23:54

Hi Guys, Recently I was investigating an issue with capturing audio using USB Audio Class device on a sh4-based board. "Bad voice quality" was reported... Finally I have traced the problem to something which is (unfortunately) well known to sh developers as a D-cache aliasing (or synonym) problem. Briefly speaking: due to some MMU design decisions, one can have two different virtual address pointing to the same physical location, which is fine, but going via different cache slots! So if there was a value of "0" in the memory and user "A" will write "1" there, user "B" will still read "0"... The solution is to ensure all TLB entries (so virtual memory areas) are beginning from a 16kB-aligned virtual address. Otherwise it is necessary to flush the cache between accesses from "A" and "B" sides. And now. The USB Audio Class driver (sound/usb/usbaudio.c) is allocating the sound buffer like this... static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { [...] runtime->dma_are

[PATCH 1/1] ASoC codec: fix compiling error in ad1980driver after ASoC API changed

Bryan Wu — 2008-09-04T08:25:54

From: Cliff Cai analog.com> Signed-off-by: Cliff Cai analog.com> Signed-off-by: Bryan Wu kernel.org> --- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad1980.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index bfbab3d..4e09c1f 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c < at >< at > -141,7 +141,7 < at >< at > static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -struct snd_soc_codec_dai ad1980_dai = { +struct snd_soc_dai ad1980_dai = { .name = "AC97", .playback = { .stream_name = "Playback", diff --git a/sound/soc/codecs/ad1980.h b/sound/soc/codecs/ad1980.h index 5d4710d..db6c850 100644 --- a/sound/soc/codecs/ad1980.h +++ b/sound/soc/codecs/ad1980.h < at >< at > -17,7 +17,7 < at >< at > #define PR50x2000 #define PR60x4000 -extern struct snd_soc_codec_dai ad1980_dai; +extern struct snd_soc_dai ad1980_dai; extern struct snd_soc_codec_device soc_codec_