October 2008 - Alsa-devel - mailman.alsa-project.org

[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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[alsa-devel] ASoC: Merging at91 and avr32 dirs - test report
by ext-jukka.hynninen＠vaisala.com 27 Oct '08

27 Oct '08

I tested the patches and got everything working with minor changes. However, there seems to be some problem with PM, occasionally ASoC is stuck with Standby mode (looking at dapm_widgets) when I start the playback. Restarting the player brings the sound back. I'll carry out some further investigation and get back to this. My kernel is 2.6.27-rc3, and machine driver is adapted from playpaq_wm8510.c to run on at91rm9200 board. The SSC peripheral clock was disabled more often than it was enabled, and that catched the bug trap in arch/arm/mach-at91/clock.c. I'm not sure if this was the real problem, but it's the way I managed to solve it. -Jukka diff -uprN a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c --- a/sound/soc/atmel/atmel_ssc_dai.c 2008-10-10 17:23:46.000000000 +0300 +++ b/sound/soc/atmel/atmel_ssc_dai.c 2008-10-10 17:26:06.000000000 +0300 @@ -261,11 +261,11 @@ static void atmel_ssc_shutdown(struct sn spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { - /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dau: Stopping clock\n"); - clk_disable(ssc_p->ssc->clk); - if (ssc_p->initialized) { + /* Shutdown the SSC clock. */ + pr_debug("atmel_ssc_dai: Stopping clock\n"); + clk_disable(ssc_p->ssc->clk); + free_irq(ssc_p->ssc->irq, ssc_p); ssc_p->initialized = 0; }

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[alsa-devel] [PATCH 1/1] ASoC: Blackfin: update SPORT0 port selector (v2)
by Bryan Wu 27 Oct '08

27 Oct '08

From: Cliff Cai <cliff.cai(a)analog.com> - Setting the TFS pin selector for SPORT 0 based on whether the selected port id F or G. If the port is F then no conflict should exist for the TFS. When Port G is selected and EMAC then there is a conflict between the PHY interrupt line and TFS. Current settings prevent the conflict by ignoring the TFS pin when Port G is selected. This allows both ssm2602 using Port G and EMAC concurrently. - some code cleanup Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Bryan Wu <cooloney(a)kernel.org> --- sound/soc/blackfin/bf5xx-i2s.c | 34 ++++++++++++++++++++++------------ 1 files changed, 22 insertions(+), 12 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 827587f..e020c16 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -70,12 +70,24 @@ static struct sport_param sport_params[2] = { } }; -static u16 sport_req[][7] = { - { P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}, - { P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}, -}; +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) @@ -98,23 +110,21 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); ret = -EINVAL; break; } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - ret = -EINVAL; - break; - case SND_SOC_DAIFMT_CBM_CFS: - ret = -EINVAL; - break; case SND_SOC_DAIFMT_CBM_CFM: break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: case SND_SOC_DAIFMT_CBS_CFM: ret = -EINVAL; break; default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); ret = -EINVAL; break; } -- 1.5.6

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[alsa-devel] Only one element for multichannel mixer controls please!
by Lennart Poettering 27 Oct '08

27 Oct '08

Heya! It seems that quite a few ALSA drivers for multichannel sound cards export three seperate stereo mixer controls to control front, rear and center/lfe independantly. That sucks because a client (such as PulseAudio) cannot make use of this. Instead they should export a single 6ch mixer control! The "amixer scontents" output looks very strange for these drivers since it lists a "Rear" control that as a "Front Left" and a "Front Right" channel. So, what is it now: front or rear? Is there any reason for this current driver behaviour? I mean, the mixer knows channel ids like SND_MIXER_SCHN_REAR_LEFT perfectly well. Or is it that those devices have some devices joined and this cannot be expressd with a single boolean "joined" for each control? Please enlighten me! Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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