Hi everyone :)

I'm currently assigned to a task that requires me to capture raw audio stream from a dummy sound card. Is this even possible?

Previously, I have been able to capture raw audio stream from my default sound card, by using the following example I got from http://www.linuxjournal.com/article/6735

/* This example reads from the default PCM device

and writes to standard output for 5 seconds of data.

*/

/* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API

#include <alsa/asoundlib.h>

int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer;

/* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); }

/* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params);

/* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params);

/* Set the desired hardware parameters. */

/* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);

/* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);

/* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2);

/* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);

/* Set period size to 32 frames. */ frames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir);

/* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); }

/* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size);

/* We want to loop for 5 seconds */ snd_pcm_hw_params_get_period_time(params, &val, &dir); loops = 5000000 / val;

while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); }

snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer);

return 0; }

The above example allows me to capture 5 seconds of raw audio. I can use aplay to play the recorded sound and it plays nicely. But right now, I'm working on a project that involves the usage of a small development board. Basically, this board does not have a sound card or an actual speaker. So I thought, I can use the dummy soundcard provided by the linux kernel by calling out modprobe snd-dummy. Right now, I'm still testing it in my PC so to simulate the board's environment. I have configured the .asoundrc file and create a new dummy pcm by putting the lines below onto my .asoundrc file.

pcm.dummy{ type hw card Dummy }

So, I change the above ecample code, so it listens to this new dummy pcm rather than the 'default'. Like this :

rc = snd_pcm_open(&handle, "dummy", SND_PCM_STREAM_CAPTURE, 0); Supposedly, the above modification would allow me to record from this dummy pcm, no? I do this by playing a song, using aplay -f cd -D dummy song.wav and at the same time, execute the above example. As expected, no sound was coming out from my speaker. When the program finished recording for 5 seconds, I play back the result, but all i heard was just noise.

Of curiousity, I try using file plugins. I modify my .asoundrc file like this:

pcm.dummy{ type plug slave{ pcm file format S16_LE channels 2 rate 44100 } }

pcm.file{ type file slave{ pcm d } file /home/mydir/out.raw }

pcm.d{ type hw card Dummy }

Then i called aplay -f cd -D dummy song.wav again, well.. still no sound coming out from the speaker, but it does output the raw audio file into my /home/mydir/out.raw file. I play the out.raw using aplay -f cd /home/mydir/out.raw it's flawless.

But I can't use this for my implementation. What I need is actually a way to read raw audio data (or stream) from the dummy sound card, to a buffer inside my program. I need this because I'm going to stream the buffer to my server, so I can listen the sound from my server. I can't afford to use the file plugin approach, basically because later on, in my development board, i won't have that much space.

So, the question is: capturing raw audio data from a dummy soundcard, is this possible? I'm pretty sure that if the file plugin works, means that the raw audio data is there. It's just that i'm probably doing a wrong approach to read it. Hence, needs explanation and help..

Please help me.. :(

Thank you so much for reading my long request.