Using ALSA for Custom Real Time Audio Effects

Another resource I found

http://www.linuxjournal.com/article/6735?page=0,2

for recording

and

Introduction to Sound Programming with ALSA | Linux Journal

for playback

Mike

That looks like a great find. I'm having trouble with compiling though. I think I need to use the -lasound flag, but whenever I try that I get:

/usr/bin/ld: cannot find -lasound

collect2: error: ld returned 1 exit status

and I cannot figure out how to fix this.

Ryan

Hey Ryan, hopefully I'll get some time to play with this this evening. If so I'll post some pointers. I have no reason other than it piqued my interest. I was close on my initial assumptions, but I guess you need to use the alsa API open and not the unix open (or fopen). If you find anything out, post back so we can all learn.

Mike

I was able to get the examples to compile, but first I had to run

sudo apt-get install libasound2 libasound2-dev

to make sure I had the libraries. Then I was able to compile on the pi without error.

I had some runtim errors on the pi, but I don't have any sound hardware.

Worked ok on my desktop.

Mike

Thanks! That worked right away, now onto figuring out how to grab one sample at a time and put it straight back out...

Ryan

Here is the code I made that would hopefully pass audio straight through the raspberry pi, without any reverb yet. It doesn't work but I think I'm on the right track. I was kind of hoping that the Raspberry Pi Cirrus Logic interface was good enough that the cirrus board would be the default board after running a script that sets the default input and output to be the line in and line out on the board, but after running my code, I don't think that is what happened.

I got this showing up over and over in terminal:

short write: wrote -25 bytes

error from read: Inappropriate ioctl for device

short write: wrote -25 bytes

error from read: Inappropriate ioctl for device

underrun occurred

short write: wrote -25 bytes

I'm not sure if there is any easy fix, but if you have time or see anything obvious here is my code (specifically I'm confused about whether or not I need separate handle and params varaibles for the input and output, and I'm guessing that I didn't need two rc variables now that I think about it:

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

#include <alsa/asoundlib.h>
#include <stdio.h>

int main() {
  long loops;
  int rc1, rc2;
  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). */
  rc1 = snd_pcm_open(&handle, "default",
  SND_PCM_STREAM_CAPTURE, 0);
  if (rc1 < 0) {
  fprintf(stderr,
  "unable to open pcm device: %s\n",
  snd_strerror(rc1));
  exit(1);
  }

  /* Open PCM device for playback. */
  rc2 = snd_pcm_open(&handle, "default",
  SND_PCM_STREAM_PLAYBACK, 0);
  if (rc2 < 0) {
  fprintf(stderr,
  "unable to open pcm device: %s\n",
  snd_strerror(rc2));
  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);

  /* one channel (mono) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 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 */
  rc1 = snd_pcm_hw_params(handle, params);
  if (rc1 < 0) {
  fprintf(stderr,
  "unable to set hw parameters: %s\n",
  snd_strerror(rc1));
  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);
  /* 5 seconds in microseconds divided by
  * period time */
  loops = 5000000 / val;

  while (loops > 0) {
  loops--;
  rc1 = snd_pcm_readi(handle, buffer, frames);
  if (rc1 == -EPIPE) {
  /* EPIPE means overrun */
  fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle);
  } else if (rc1 < 0) {
  fprintf(stderr, "error from read: %s\n", snd_strerror(rc1));
  } else if (rc1 != (int)frames) {
  fprintf(stderr, "short read, read %d frames\n", rc1);
  }
  rc2 = snd_pcm_writei(handle, buffer, frames);
  if (rc2 == -EPIPE) {
  /* EPIPE means underrun */
  fprintf(stderr, "underrun occurred\n");
  snd_pcm_prepare(handle);
  } else if (rc2 < 0) {
  fprintf(stderr,
  "error from writei: %s\n",
  snd_strerror(rc2));
  } else if (rc2 != (int)frames) {
  fprintf(stderr,
  "short write, write %d frames\n", rc2);
  }
  if (rc1 != size){
  fprintf(stderr, "short write: wrote %d bytes\n", rc1);
  }
  }

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

  return 0;
}

Nothing jumps out at me as wrong, but I won't be able to look at this in depth until later this evening.

Mike

Nothing jumps out at me as wrong, but I won't be able to look at this in depth until later this evening.

Mike