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[alsa-devel] [PATCH] ALSA: hda: Add HDA vendor ID for Wolfson Microelectronics
by Mark Brown 28 Oct '08

28 Oct '08

Add Wolfson Microelectronics to the HDA vendor ID table. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/pci/hda/hda_codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6447754..ba1ab73 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -64,6 +64,7 @@ static struct hda_vendor_id hda_vendor_ids[] = { { 0x14f1, "Conexant" }, { 0x17e8, "Chrontel" }, { 0x1854, "LG" }, + { 0x1aec, "Wolfson Microelectronics" }, { 0x434d, "C-Media" }, { 0x8384, "SigmaTel" }, {} /* terminator */ -- 1.5.6.5

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Re: [alsa-devel] [PATCH] Remove unused variable 'codec' from sound/soc/codecs/tlv320aic23.c
by Arun KS 28 Oct '08

28 Oct '08

Manish Katiyar wrote: > [ Apologies if this is not the proper list for this patch, but I > couldn't find any list more appropriate for this] > > Below patch removes the following warning. > sound/soc/codecs/tlv320aic23.c:424: warning: unused variable 'codec' CCing Troy Troy has got some patch which adds support for more mclk frequencies. So that variable will be required. Thanks, Arun KS > > > Signed-off-by: Manish Katiyar <mkatiyar(a)gmail.com> > --- > sound/soc/codecs/tlv320aic23.c | 2 -- > 1 files changed, 0 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c > index 44308da..45395d0 100644 > --- a/sound/soc/codecs/tlv320aic23.c > +++ b/sound/soc/codecs/tlv320aic23.c > @@ -421,8 +421,6 @@ static int tlv320aic23_set_dai_fmt(struct > snd_soc_dai *codec_dai, > static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, > int clk_id, unsigned int freq, int dir) > { > - struct snd_soc_codec *codec = codec_dai->codec; > - > switch (freq) { > case 12000000: > return 0;

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[alsa-devel] [PATCH] Remove unused variable 'codec' from sound/soc/codecs/tlv320aic23.c
by Manish Katiyar 28 Oct '08

28 Oct '08

[ Apologies if this is not the proper list for this patch, but I couldn't find any list more appropriate for this] Below patch removes the following warning. sound/soc/codecs/tlv320aic23.c:424: warning: unused variable 'codec' Signed-off-by: Manish Katiyar <mkatiyar(a)gmail.com> --- sound/soc/codecs/tlv320aic23.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308da..45395d0 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -421,8 +421,6 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - switch (freq) { case 12000000: return 0; -- 1.5.4.3 Thanks - Manish

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[alsa-devel] [PATCH] ASoC/pxa-ssp.c: fix build on pxa2xx
by Daniel Ribeiro 27 Oct '08

27 Oct '08

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[alsa-devel] [PATCH] ASoC/pxa-ssp.c: keep up with pxa api changes
by Daniel Ribeiro 27 Oct '08

27 Oct '08

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[alsa-devel] problem persist on Mac Pro (quad pro) using fedora 9
by S. Aguinaga 27 Oct '08

27 Oct '08

Hello Team, I'm hoping that some of you folks might know the work-around and or fix to my Mac Pro (quad core) installation of alsa ( on fedora 9 ). I wanted to get the record to work copied the same thing I did on my laptop MacBook Pro, which is working ( to some extent, such as with audacity ), but on the Mac Pro tower it has gotten worse. I used to have audio on the front speaker, and now I have none. And I'm not sure what else to try. aplay 1.wav does not generate sound, but alsamixer seems to have all ports on and looking ok. 1. /etc/modprobe.d/sound reads: options snd-hda-intel model=macpro 2. ~/.asound reads ( but my gui volume controls tools don't work ) ... I'm wondering if there anyway to not rely on pulseaudio or if I need to have pulseaudio working on my fedora 9 installation. pcm.pulse{ type pulse } ctl.pulse { type pulse } pcm.phononpulse { type plug slave.pcm { type pulse } hint { show on description "PulseAudio" } } 3. the file /etc/asound.conf reads the following ... so I wonder if this is enough and should I erase .asoundrc ? If so, what else do I need to get audio working. pcm.!default { type hw card Intel } ctl.!default { type hw card Intel } I also have asound.state and where should I check to see if it is ok? // Sal // Northwestern University - HARP

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[alsa-devel] 'multi' plugin signals POLLOUT but snd_pcm_avail_update() returns 0
by Lennart Poettering 27 Oct '08

27 Oct '08

Heya! There's a apparently a bug in the 'multi' plugin. When it is used it will pretty often signal POLLOUT when used with snd_pcm_poll_descriptors_revents() and the subsequent call to snd_pcm_update_avail() will still return 0. Doesn't look right to me. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Recording problem with pulse plugin
by Takashi Iwai 27 Oct '08

27 Oct '08

Hi Lennart, I got a bug report about the recording with pulse alsa-plugin. It gets just -EIO error. I checked it, and can reproduce it actually. The problem appears to happen since 1.0.18rc1, and the affecting commit is: commit 90c32999189cf6f5c63f40dadb8076eca379713b Author: Lennart Poettering <mznyfn(a)0pointer.de> Date: Fri Jul 18 21:45:47 2008 +0200 send both an uncork and a trigger in _start() The problem is that pa_stream_trigger() returns NULL for capture. I made a quick fix for this (the patch attached below), but not entirely sure whether it's the right thing. Please check. thanks, Takashi diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index a5f794f..3d15ff0 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -145,18 +145,15 @@ static int pulse_start(snd_pcm_ioplug_t * io) u = pa_stream_trigger(pcm->stream, pulse_stream_success_cb, pcm->p); - if (!u) { - pa_operation_unref(o); - err = -EIO; - goto finish; - } pcm->underrun = 0; err_o = pulse_wait_operation(pcm->p, o); - err_u = pulse_wait_operation(pcm->p, u); + if (u) + err_u = pulse_wait_operation(pcm->p, u); pa_operation_unref(o); - pa_operation_unref(u); + if (u) + pa_operation_unref(u); if (err_o < 0 || err_u < 0) { err = -EIO;

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[alsa-devel] NVIDIA HDMI Support
by Robert Vincent Krakora 27 Oct '08

27 Oct '08

Hello, I see that NVIDIA HDMI support is in the tip but not in RC3. Will it be in RC4? Thanks in advance. Best Regards, -- Rob Krakora Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

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[alsa-devel] ASOC: For SND_SOC_DAIFMT_IB_IF what is invert baseed on?
by Richard Zhao 27 Oct '08

27 Oct '08

Hi, include/sound/soc.h /* * DAI hardware signal inversions */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */ #define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ #define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ #define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ What are frame cock and bit clock invert based on? I2S, PCM or some else bus protocols? Or just high level voltage or low level voltage? Thanks Richard

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