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

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

One thing to check in the meantime, look for listing 2 in the examples I linked, it will dump the capabilities of the card, ensure it is capable of full duplex, you may have to open for read, read, close, open for write, write, close, etc...

Mike

Hey Ryan, one other thing maybe to look at is Jack instead of ALSA.

JACK Audio Connection Kit|Home

It looks interesting at the very least.

Mike

Hi Mike,

I did have a couple of errors in my code, I needed two handle variables. Unfortunately that was only enough to get rid of some clear errors, it still doesn't really work. I have heard of JACK, maybe that is the way to go since it did seem like ALSA might have some inherent problems. My code kept having underrun errors and would output garbage, although I probably am using the buffers wrong .

Ryan

Yeah, I was looking at real time audio processing using ALSA and there were several articles about kernel tweaks to get the performance right. Jack didn't seem to have those same limitations. I looked over their website briefly, looks like there are some examples.

Mike

Just for fun, I ran

sudo apt-get install jack jackd1 libjack-dev

dl'd https://github.com/jackaudio/jack2/blob/master/example-clients/simple_client.c

which is supposed to pipe the input to the output

and compiled using

gcc -o simple_client simple_client.c -ljack -lm

that may be a good starting point for you.

http://jackaudio.github.io/api/

search for simple_client.c in the above link

Mike