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

[alsa-devel] [PATCH 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3 -- - Joonyoung Shim

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[alsa-devel] [PATCH RESEND 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3

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[alsa-devel] [PATCH RESEND 1/1] ASoC: twl4030 - Fix build error
by Joonyoung Shim 16 Mar '09

16 Mar '09

CC sound/soc/codecs/twl4030.o sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops') sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops') Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ce4b43b..5ed5f1b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1383,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops twl4030_dai_ops = { + .hw_params = twl4030_hw_params, + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1397,11 +1403,7 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); -- 1.5.6.3

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[alsa-devel] [PATCH 8/9] ARM: DaVinci: ASoC: Adds header file for DM646X DIT/DIR driver
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds header file for DM646X DIT/DIR driver Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/codecs/codec_stubs.h | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/codec_stubs.h diff --git a/sound/soc/codecs/codec_stubs.h b/sound/soc/codecs/codec_stubs.h new file mode 100644 index 0000000..693797b --- /dev/null +++ b/sound/soc/codecs/codec_stubs.h @@ -0,0 +1,23 @@ +/* + * ALSA SoC DIT/DIR driver header + * + * Author: Steve Chen, <schen(a)mvista.com> + * Copyright: (C) 2008 MontaVista Software, Inc., <source(a)mvista.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. + */ + +#ifndef CODEC_STUBS_H +#define CODEC_STUBS_H + +#ifdef CONFIG_SND_SOC_CODEC_STUBS +extern struct snd_soc_dai dit_stub_dai[]; +extern struct snd_soc_dai dir_stub_dai[]; +#else +#define dit_stub_dai NULL +#define dir_stub_dai NULL +#endif + +#endif /* CODEC_STUBS_H */ -- 1.5.6

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[alsa-devel] [PATCH 5/9] ARM: DaVinci: ASoC: Adds Audio Serial Port support for TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds Audio Serial Port support for TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies to the DaVinci git tree available at http://git.kernel.org/?p=linux/kernel/git/khilman/linux-davinci.git;a=commi…. arch/arm/mach-davinci/include/mach/asp.h | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index e0abc43..670afe9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -6,16 +6,26 @@ #include <mach/irqs.h> -/* Bases of register banks */ +/* Bases of dm644x and dm355 register banks */ #define DAVINCI_ASP0_BASE 0x01E02000 #define DAVINCI_ASP1_BASE 0x01E04000 -/* EDMA channels */ + +/* Bases of dm646x register banks */ +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 + +/* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 #define DAVINCI_DMA_ASP1_TX 8 #define DAVINCI_DMA_ASP1_RX 9 + +/* EDMA channels of dm646x */ +#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6 +#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 +#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT -- 1.5.6

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[alsa-devel] [PATCH 4/9] ARM: DaVinci: ASoC: Adds ALSA PCM interface for the TI DM646X processor
by Naresh Medisetty 16 Mar '09

16 Mar '09

Adds ALSA PCM interface for the TI DM646X processor Signed-off-by: Naresh Medisetty <naresh(a)ti.com> --- This patch applies on the ASoC tree available at http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;…. sound/soc/davinci/davinci-pcm.h | 22 ++++++++++++++++++---- 1 files changed, 18 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb..c7d8711 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,16 +12,30 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ + enum dma_event_q eventq_no; /* event queue number */ }; struct evm_snd_platform_data { + char *clk_name; int tx_dma_ch; int rx_dma_ch; + u32 tx_dma_offset; + u32 rx_dma_offset; + enum dma_event_q eventq_no; /* event queue number */ + unsigned int codec_fmt; + + /* McASP specific fields */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; }; extern struct snd_soc_platform davinci_soc_platform; -- 1.5.6

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[alsa-devel] Using snd-aloop as default soundcard (need urgent help T.T)
by Santo Chow 16 Mar '09

16 Mar '09

Hi every one.. I'm currently on a project that requires me to capture a raw audio data from a dummy soundcard. I've so far manage to do that, by using snd-aloop. Now the problem i have is.. when i'm running several audio program at the same time, i can only hear the sound from the first program that i ran. i've followed the example os dmix, but this doesn't help either. So, right now, i have a Loopback soundcard as my default. My .asoundrc looks like this: pcm.!default{ type hw card 1 } Card 0 is my actual soundcard (that can actually produce sound). Card 1, when i checked on the /proc/asound/cards actually pointing to Loopback card. So right now, the way i test the card 1, is to call arecord from card 1, and play it to card 0. Right now, i'm still developing this in my ubuntu PC. so i ran this arecord -f cd -D hw:1,1 | aplay -f cd -D hw:0 in one terminal, and in the other terminal i can play mplayer, watching youtube, etc.. and the sound would come out nicely. (Thanks to all of your efforts, nevertheless) But I notice that, when i play mplayer, then i open a web-browser, and surf youtube, the sound from youtube is gone. But if i close the browser, close the mplayer, then reopen the browser, go to youtube and play a video, i can hear the sound nicely. but then, if during the time i start mplayer, it will complain about 'no sound'. Turns out, I have to use dmix. So, i modify my .asoundrc like this: pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:0,0" rate 44100 } bindings { 0 0 1 1 } } i put 'pcm "hw:0,0" ' over there, because i want to make sure that this setting works with my actual sound card first. If it works with my card 0, it should work with my card 1. with the above .asoundrc file, i can play multiple sound from different application at the same time. I even tried to remove the "bindings" tag, and it still able to play multiple sound very well. so i guess, this should work on my card 1. then i change the .asoundrc become like this again: pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1025 slave { pcm "hw:1,1" period_time 0 period_size 881 buffer_size 3524 rate 44100 } }the value for perio_time, period_size, and buffer_size have been adjusted by me to suits my hardware requirement. This is the setting that works on my machine. It's just that, in the end, I'm still ended up with one sound at a time. I can't run mplayer + youtube at the same time. I can only run one of them. And furthermore, if i just paused the mplayer, the audio device is still locks. It will be good enough if someone can hint me a way to unlock the audio device. So, what i want to ask is.. can this Loopback card handle more than 1 sound at a time? Have anyone tried this before? I have tried changing the value of .asoundrc, switching the slave pcm to pcm "hw:1,0" (not working) pcm "hw:1" (works, but only 1 sound too). Any idea guys? -santo-

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[alsa-devel] Problem with Pulse Audio plugin?
by Tom Watson 16 Mar '09

16 Mar '09

I'm doing some experimenting and found that a simple sequential play of a small sound file caused an assertion in the pulse audio plugin. The base system is Fedora 8, and it uses alsa 1.0.16 versions. I looked at the 1.0.19 versions and they still have assertions, so the problem may still be there. It is pretty easy to trip up: $ aplay blah.wav blah.wav Playing WAVE 'blah.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Playing WAVE 'blah.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_pulse.c:524: pulse_hw_params: Assertion `!pcm->stream' failed. Aborted by signal Aborted... (where "blah.wav" is a simple short wav file) The first time through, it plays find, but the second time it doesn't. If this has been solved in 1.0.19, (I haven't installed it), please let me know and I'll go away. If not, it may be a problem in the pulse audio thing. Thanks. -- Tom Watson tsw(a)johana.com

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Re: [alsa-devel] [PATCH 1/2] Echoaudio: Add support for Indigo express cards
by Takashi Iwai 16 Mar '09

16 Mar '09

At Mon, 16 Mar 2009 09:14:20 +0100 (CET), Giuliano Pochini wrote: > > > > On Mon, 16 Mar 2009, Takashi Iwai wrote: > > >> This patch adds support for IndigoIOx and IndigoDJx. > >> > >> Signed-off-by Giuliano Pochini <pochini(a)shiny.it> > > > > The code looks almost good, but could you do the following? > > > > - split to two patches, one for kernel tree and one for alsa-driver > > tree > > There is a little problem: since about a month ago the git server is > SLOWWW. It takes ages to download the alsa-kernel repository at 5KB/s :( > I'll try from some server at work which stay on 24/7. Better to try my sound git tree on kernel.org. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git The bare alsa-driver tree is: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build.git The corresponding alsa-driver snapshot is: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Takashi

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[alsa-devel] [PATCH 1/1] echomixer: Fix echomixer to work with the new drivers
by Giuliano Pochini 16 Mar '09

16 Mar '09

There is a long standing bug in the drivers for cards with a vmixer because I overlooked a detail in the c++ generic driver by echoaudio. Those cards do not have a line-out volume control. It is a virtual control provided by the generic driver. The bug is harmless because the DSP just ignores the command to change the volume. Since that control has been removed, echomixer must be updated. With this patch it uses the vmixer to fake the line-out volume. This patch makes echomixer work with the new drivers. Signed-off-by Giuliano Pochini <pochini(a)shiny.it> --- alsa-tools-1.0.19/echomixer/echomixer.c__orig2 2009-03-15 19:03:24.000000000 +0100 +++ alsa-tools-1.0.19/echomixer/echomixer.c 2009-03-15 19:36:23.000000000 +0100 @@ -1178,6 +1178,29 @@ void UpdateMixerVolume(int outchannel) { +// Changes the vmixer volume according to the current Line-out volume for vmixer cards. +void UpdateVMixerVolume(int outchannel) { + snd_ctl_elem_id_t *id; + snd_ctl_elem_value_t *control; + int err, val, ch; + + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_value_alloca(&control); + snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER); + + for (ch=0; ch<nPOut; ch++) { + val=Add_dB(vmixerControl.mixer[outchannel][ch].Gain, lineoutControl.Gain[outchannel]); + ClampOutputVolume(&val); + snd_ctl_elem_id_set_numid(id, vmixerControl.mixer[outchannel][ch].id); + snd_ctl_elem_value_set_id(control, id); + snd_ctl_elem_value_set_integer(control, 0, val); + if ((err = snd_ctl_elem_write(ctlhandle, control)) < 0) + printf("Control %s element write error: %s\n", card, snd_strerror(err)); + } +} + + + void LineOut_volume_changed(GtkWidget *widget, gpointer ch) { char str[16]; int err, channel, val; @@ -1204,6 +1227,9 @@ void LineOut_volume_changed(GtkWidget *w snd_ctl_elem_value_set_integer(control, channel, val); if ((err=snd_ctl_elem_write(ctlhandle, control))<0) printf("Control %s element write error: %s\n", card, snd_strerror(err)); + } else if (vmixerId) { + UpdateVMixerVolume(channel); + UpdateMixerVolume(channel); } else { // Otherwise we have to emulate it. UpdatePCMVolume(channel); UpdateMixerVolume(channel); @@ -1217,10 +1243,11 @@ void LineOut_volume_changed(GtkWidget *w void Vmixer_volume_changed(GtkWidget *widget, gpointer ch) { char str[16]; - int val, channel; + int val, rval, channel; int o, v; channel=(int)(long)ch; + val=INVERT((int)GTK_ADJUSTMENT(widget)->value); #ifdef REVERSE v=channel; @@ -1230,13 +1257,17 @@ void Vmixer_volume_changed(GtkWidget *wi o=channel; #endif - val=INVERT((int)GTK_ADJUSTMENT(widget)->value); + // Emulate the line-out volume if this card can't do it in hw. + if (!lineoutId) { + rval=Add_dB(val, lineoutControl.Gain[o]); + ClampOutputVolume(&rval); + } - SetMixerGain(&vmixerControl.mixer[o][v], val); + SetMixerGain(&vmixerControl.mixer[o][v], rval); vmixerControl.mixer[o][v].Gain=val; if (Gang) { - SetMixerGain(&vmixerControl.mixer[o^1][v^1], val); + SetMixerGain(&vmixerControl.mixer[o^1][v^1], rval); vmixerControl.mixer[o^1][v^1].Gain=val; } -- Giuliano.

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