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

[alsa-devel] salsa with noninterleaved needs irqpoll?
by maarten van es 26 Apr '09

26 Apr '09

>>> >>> And what about the sample format? I looked at the drivers; the two >>> sound cards do not have a common sample format. >>> >>> >>> Best regards, >>> Clemens >>> >> > > Thanks, the problem was indeed that I was setting hwparams that are > not supported by the card. I do think salsa lib could return a more > relevant error than "operation not permitted" and I'll see if I can > come up with something for that. > Right now though I am trying to figure out what snd_pcm_writen needs > for its second parameter, void** bufs. I get the error "bad address" > when passing in an array of pointers to a malloc-ed space big enough > to hold a channels' worth of data. Sorry if I am missing obvious > documentation about this, but what should I pass for this parameter? > > Regards > Maarten > Hi again, Ok, now I finally got everything working and am asking the following mostly out of curiosity. The problem I had with snd_pcm_writen by the way has to do somehow with using a chroot environment; everything works when directly booting into the uclibc / salsa system. For it to work I need to add "irqpoll" to the kernel boot options. Without this, the card does not start playing (though no error messages from salsa) and after a few seconds the kernel tells me to add irqpoll to the boot options. On a glibc / alsa based system everything works fine without this option as it does with a c-media card with salsa. What does it do and why is it needed with the hdsp 9632 on a system with salsa? Thanks Maarten

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[alsa-devel] Getting the correct capture mixer
by Ludovico Cavedon 26 Apr '09

26 Apr '09

Hi, I am working on fixing some audio related bugs of a VoIP application (qutecom) which, like most VoIP applications, allows the user to change the playback and recording volume. For playback volume I am opening the "Master" mixer element. However I am not able to figure out how the application should get the correct mixer for playback. Is there any rule? For example, my sound card has two capture volumes, one "Int Mic" and one "Ext Mic". I guess I should choose on the basis of what is the user-selected mic. In fact there are also two mutually exclusive switches "IntMic" and "ExtMic". But how should the application understand that? Here part of the output of amixer, my sound card is "Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]" ---------------------------- Simple mixer control 'Ext Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'ExtMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Int Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'IntMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] ------------------------ As an aside, it looks like that the capture volume has no effect, as well all as the cswitches and pswitches of "Int Mic" and "Ext Mic". Thanks, Ludovico

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[alsa-devel] .pointer return value
by Harsha, Priya 25 Apr '09

25 Apr '09

Hi, Can anyone educate me on what value should be returned to .pointer function? Should I be passing the number of frames played till that point from the time stream started or thnumber of frames played such that the bytes is within the buf_size? And if I am using, snd_pcm_indirect_playback_pointer, what should be my input to it? Should I be passing the number of bytes played till that point from the time stream started or the number of bytes within buf_size? Thanks, Harsha

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Re: [alsa-devel] salsa error message setting hwparams
by maarten van es 24 Apr '09

24 Apr '09

On Fri, Apr 24, 2009 at 10:30 AM, maarten van es <maarten.h.van.es(a)gmail.com> wrote: > Hi > > Right now I don't have the machine at hand as I'm at work, but I don't > think the sample format is the problem. Like I said, my program works > fine with the hdsp on glibc / alsa. On uclibc / salsa I get the error > setting any hw param, starting with setting the access method to > rw_interleaved, and including setting the sample rate while I switched > the card already to the correct rate with amixer. > Furthermore, if the problem is an unsupported format I wouldn't expect > the error to be "operation not permitted". > I'll try that program anyway when I'm back home to see what it tells me. > > Thanks > Maarten > > On Fri, Apr 24, 2009 at 10:03 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> (please don't top-post) >> >> maarten van es wrote: >>> Thanks, but that's not it. I make sure I set the correct sample rate >>> with amixer first. >> >> And what about the sample format? I looked at the drivers; the two >> sound cards do not have a common sample format. >> >> >> Best regards, >> Clemens >> > Thanks, the problem was indeed that I was setting hwparams that are not supported by the card. I do think salsa lib could return a more relevant error than "operation not permitted" and I'll see if I can come up with something for that. Right now though I am trying to figure out what snd_pcm_writen needs for its second parameter, void** bufs. I get the error "bad address" when passing in an array of pointers to a malloc-ed space big enough to hold a channels' worth of data. Sorry if I am missing obvious documentation about this, but what should I pass for this parameter? Regards Maarten

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[alsa-devel] Help
by Sudeept Prusti 24 Apr '09

24 Apr '09

Hi all, As i am working on alsa for minimal PCM playback (8khz,mono,16 bit pcm file). When i tried to allocate 1k buffer and send through snd_pcm_writei() call it doesn't play perfect and skips are there in between. If we keep on increasing in allocating the buffer for 31 kb then it could play the buffers randomly. Code is attached for ur reference below. ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #define k (31*1024) static void playback(FILE *fd); int main(int argc, char *argv[]) { FILE *fd; fd = fopen(argv[1],"r"); playback(fd); return 0; } static void playback(FILE *fd) { int rc,err; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; snd_pcm_uframes_t frames; unsigned char *buffer; int seek_size=0; int nonblock = 0; printf("Entered Playback Module \n"); /* Open PCM device for playback. */ rc = snd_pcm_open(&handle, "default",SND_PCM_STREAM_PLAYBACK, 0); if(rc >= 0) { printf("Open PCM Success: %d \n",rc); } if (rc < 0) { fprintf(stderr,"unable to open pcm device: %s\n",snd_strerror(rc)); exit(1); } /* PCM Hardware Parameters Allocation. */ err = snd_pcm_hw_params_malloc (&params); if(err>=0) { printf("Hardware Params Allocation Success: %d \n",err); } if (err < 0) { fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)); exit (1); } /* Intialize PCM Hardware Parameters */ err = snd_pcm_hw_params_any(handle, params); if(err >=0) { printf("PCM HW params Initialization Success : %d \n",err); } if (err < 0) { printf("Cannot initialize hardware parameter structure \n"); exit(EXIT_FAILURE); } /* Set the PCM Hardware parameters. */ /* Interleaved mode */ err = snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if (err >= 0) { printf("Access Type Available: %d \n",err); } if (err < 0) { printf("Access Type Not-available: %d \n",err); exit(EXIT_FAILURE); } /* Signed 16-bit little-endian format */ err = snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_S16_LE); if (err >= 0) { printf("Sample format Available: %d \n",err); } if (err < 0) { printf("Sample format Non-Available \n"); exit(EXIT_FAILURE); } /* One Channel (Mono) */ err = snd_pcm_hw_params_set_channels(handle, params, 1); if (err >= 0) { printf("Channels Available: %d \n",err); } if (err < 0) { printf("Channels Non-Available \n"); exit(EXIT_FAILURE); } /* 8khz sampling rate */ err = snd_pcm_hw_params_set_rate (handle, params, 8000, 0); if (err >= 0) { printf ("Sample Rate Set Success: %d \n",err); } if (err < 0) { fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)); exit (EXIT_FAILURE); } /* Write the parameters to the alsa-driver */ rc = snd_pcm_hw_params(handle, params); if(rc >=0) { printf("Set HW Parameters Success: %d \n",rc); } if (rc < 0) { fprintf(stderr,"Unable to Set HW Parameters: %s\n",snd_strerror(rc)); exit(1); } /* Allocate buffer*/ buffer = (unsigned char *) malloc(k); if (buffer == NULL) { printf("Not Enough Memory: %c \n",buffer); exit(1); } while (!feof(fd)) { if(fd != NULL) { printf("File pointer is not NULL \n"); } printf("Current Position of File Pointer is: %d\n",ftell(fd)); rc = fread(buffer,k,1,fd); printf("Data Read from File is: %d \n",rc); if(rc > 0) { printf("File Read Success \n"); } seek_size = seek_size + k; fseek(fd,seek_size,SEEK_SET); if (rc == 0) { printf("End of File Reached: %d \n",rc); break; } rc = snd_pcm_writei(handle, buffer, k); printf("Write: %d bytes \n",rc); if (rc == -EPIPE) { printf("underrun occurred \n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr,"Error From Writei Call: %s\n",snd_strerror(rc)); exit(1); } }//End of While fclose(fd); snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); }//End of Playback -------------------------------------------------------------------------------------------------------------- Can anyone help me out where i am doing wrong.Ur's valuable suggestions are welcomed. Best Regards, Sudeept

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[alsa-devel] [PATCH 0/1] ASoC: 4 channel mode for beagle board
by Peter Ujfalusi 24 Apr '09

24 Apr '09

Hello, Resending the last part of the previous series, which adds 4 channel mode support for the Beagle board. --- Peter Ujfalusi (1): ASoC: Beagle: Add support for 4 channel sound/soc/omap/omap3beagle.c | 26 ++++++++++++++++++-------- 1 files changed, 18 insertions(+), 8 deletions(-)

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[alsa-devel] ASoC updates for 2.6.31
by Mark Brown 24 Apr '09

24 Apr '09

The following changes since commit 9b5b0c01598f9782690b09ce6c49f4ba116dde44: Mark Brown (1): Merge branch 'for-2.6.30' into for-2.6.31 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.31 Eric Miao (2): ASoC: simplify the SSP DMA parameters settings by run-time generation ASoC: change stereo/mono to 32-bit/16-bit for pxa-ssp Joonyoung Shim (2): ASoC: OMAP: Add checking to detect bufferless pcms ASoC: TWL4030: Add VDL path support Mark Brown (16): ASoC: Factor out generic widget power checks ASoC: Factor out DAPM power checks for DACs and ADCs ASoC: Make the DAPM power check an operation on the widget ASoC: Fix offset of freqmode in WM8580 PLL configuration ASoC: Fix WM8580 volume update handling for large register changes ASoC: Add power supply widget to DAPM ASoC: Use DAPM supply widget for WM8903 charge pump ASoC: Support CLK_DSP in WM8903 ASoC: Optimise configuration of WM8903 DC servo ASoC: Actively manage the DC servo for WM8903 ASoC: Remove redundant rate constraint for WM8903 ASoC: Implement WM8903 digital sidetone support Merge commit 'takashi/fix/asoc' into for-2.6.30 Merge branch 'for-2.6.30' into for-2.6.31 ASoC: s3c-i2s-v2 needs to declare a license for modular builds Merge branch 'for-2.6.30' into for-2.6.31 Peter Ujfalusi (2): ASoC: OMAP: Add 4 channel support to mcbsp ASoC: TWL4030: Add 4 channel TDM support Russell King - ARM Linux (1): ASoC: Fix warning in wm9705 Takashi Iwai (1): ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991 Documentation/sound/alsa/soc/dapm.txt | 1 + include/sound/soc-dapm.h | 9 +- sound/soc/codecs/Makefile | 1 - sound/soc/codecs/twl4030.c | 308 +++++++++++++++++++-------------- sound/soc/codecs/twl4030.h | 11 ++ sound/soc/codecs/wm8580.c | 16 +- sound/soc/codecs/wm8903.c | 118 ++++++++----- sound/soc/codecs/wm9705.c | 2 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-pcm.c | 9 +- sound/soc/pxa/pxa-ssp.c | 203 +++++----------------- sound/soc/s3c24xx/s3c-i2s-v2.c | 3 +- sound/soc/soc-dapm.c | 139 +++++++++++----- 13 files changed, 438 insertions(+), 387 deletions(-)

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Re: [alsa-devel] salsa error message setting hwparams
by Clemens Ladisch 24 Apr '09

24 Apr '09

(please don't top-post) maarten van es wrote: > Thanks, but that's not it. I make sure I set the correct sample rate > with amixer first. And what about the sample format? I looked at the drivers; the two sound cards do not have a common sample format. Best regards, Clemens

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[alsa-devel] salsa error message setting hwparams
by maarten van es 24 Apr '09

24 Apr '09

Hi everyone, I am running out of ideas of why I am getting the error "operation not permitted" when I try setting hw parameters with one of my audio cards. I have a custom linux system based on uclibc with salsa for audio and use the default device for playback. This sytem works fine on a via C7 based pc with via audio. When I try an exact copy of the system on an athlon64 based system with a HDPS9632 I get this error message. The card works fine with other linux systems. I tested several (recent) kernels with my system, all the same result. Any ideas why I might be getting this message? Maarten

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[alsa-devel] [PATCH] sc6000: fix older card initialization
by Krzysztof Helt 24 Apr '09

24 Apr '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The last patch to handle newer cards like SC7000 broke initialization of the SC6000. Fix this. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sc6000.c | 22 ++++++++++++---------- 1 files changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 983ab7e..c803b2e 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -391,7 +391,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, int config = mss_config | sc6000_mpu_irq_to_softcfg(mpu_irq[dev]); int err; - int cfg[2]; int old = 0; err = sc6000_dsp_reset(vport); @@ -421,11 +420,18 @@ static int __devinit sc6000_init_board(char __iomem *vport, answer, version[0], version[1]); /* set configuration */ - sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], - mss_port[dev]); - if (sc6000_hw_cfg_write(vport, cfg) < 0) { - snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); - return -EIO; + sc6000_write(vport, COMMAND_5C); + if (sc6000_read(vport) < 0) + old = 1; + + if (!old) { + int cfg[2]; + sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], + mss_port[dev]); + if (sc6000_hw_cfg_write(vport, cfg) < 0) { + snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); + return -EIO; + } } err = sc6000_setup_board(vport, config); if (err < 0) { @@ -434,10 +440,6 @@ static int __devinit sc6000_init_board(char __iomem *vport, } sc6000_dsp_reset(vport); - sc6000_write(vport, COMMAND_5C); - if (sc6000_read(vport) < 0) - old = 1; - sc6000_dsp_reset(vport); if (!old) { sc6000_write(vport, COMMAND_60); -- 1.6.0.3 ---------------------------------------------------------------------- Dzwonki na komork�! Sprawdz >> http://link.interia.pl/f2127

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