ALSA

I ported an old Linux audio program, a simple Voice-over-IP & stereo demuxing application, from OSS to ALSA.

First impression of ALSA libasound : the documentation sucks, well, no, the documentation really sucks monkey rods ! I wonder why the wannabe 1st Linux audio API have such a pathetic API doc, example : snd_pcm_hw_params_set_period_time_near. And this is not an exception, the whole API go such poor doxygen comments. Very frustating for someone who is maintaining a library, and made a huge effort on javadocing lazy coder code.

About the API itself, well.. it's looking a bit over-complex near OSS, but it's no rocket science once you found the good examples for your poll based full duplex program : speexclient. But once it was ported the perfs are awesome, far of OSS and far of Windows sound system. I was able to achieve 5ms latency with a 8 inputs/8 outputs audio card, without using a rt patched kernel, nor huge optimization.

Well it's a pity you got a such technically impressive audio system ruined by poor doc & API.. no wonder why we have Arts, ESD, Pulseaudio, Jackd (the guys who thinks everybody have floating point CPU).

My version of speexclient, transformed for being a simple fullduplex passthru audio program (I removed all the speex encoding/decoding part and the network code for simplicity).

You still need speex for it's very good jitter buffer and alsa_device.c (can be found of speex GIT repository)

/***************************************************************************
 Copyright (C) 2004-2006 by Jean-Marc Valin
 Copyright (C) 2006 Commonwealth Scientific and Industrial Research
 Organisation (CSIRO) Australia

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#ifdef HAVE_CONFIG_H
#include 
#endif

#include 
#include 
#include 
#include 
#include 
#include 
#include 
#include  /* close() */
#include  /* memset() */

#include "alsa_device.h"

#include 

#include 

#define MAX_MSG 1500

#define SAMPLING_RATE 16000
#define FRAME_SIZE 320

int main(int argc, char *argv[]) {


 int nfds;
 struct pollfd *pfds;

 AlsaDevice *audio_dev;

 JitterBuffer * jitterBuffer = jitter_buffer_init(FRAME_SIZE * 2);
 int jitterTime = 0;

 /* Setup audio device, forced to pughw for avoiding pulseaudio */
 audio_dev = alsa_device_open("plughw:0,0", SAMPLING_RATE, 1, FRAME_SIZE);

 struct sched_param param;
 /*param.sched_priority = 40; */
 param.sched_priority = sched_get_priority_min(SCHED_FIFO);
 if (sched_setscheduler(0, SCHED_FIFO, ¶m))
  perror("sched_setscheduler");

 /* Setup all file descriptors for poll()ing */
 nfds = alsa_device_nfds(audio_dev);
 pfds = malloc(sizeof(*pfds) * (nfds));
 alsa_device_getfds(audio_dev, pfds, nfds);

 alsa_device_start(audio_dev);

 /* Infinite loop on capture, playback */
 while (1) {
  /* Wait for either 1) capture 2) playback */
  poll(pfds, nfds, -1);

  /* Ready to play a frame (playback) */
  if (alsa_device_playback_ready(audio_dev, pfds, nfds)) {
   short pcm[FRAME_SIZE];

   /* Get audio from the jitter buffer */
   JitterBufferPacket p;
   p.data = (char *)pcm;
   p.len = FRAME_SIZE * 2;

   // eat a buffer
   int res=jitter_buffer_get(jitterBuffer,&p,FRAME_SIZE * 2,NULL);
   if(res != JITTER_BUFFER_OK ) {
    memset(pcm,0,FRAME_SIZE*2);
   }

   jitter_buffer_tick(jitterBuffer);


   /* Playback the audio and reset the echo canceller if we got an underrun */
   if (alsa_device_write(audio_dev, pcm, FRAME_SIZE)) {
    printf("underrun\n");
   }
  }
  /* Audio available from the soundcard (capture) */
  if (alsa_device_capture_ready(audio_dev, pfds, nfds)) {
   short pcm[FRAME_SIZE];

   /* Get audio from the soundcard */
   alsa_device_read(audio_dev, pcm, FRAME_SIZE);

   JitterBufferPacket p;
   p.data = (char*)(pcm);
   p.len = FRAME_SIZE*2;
   jitterTime += FRAME_SIZE*2;
   p.timestamp = jitterTime;
   p.span = FRAME_SIZE*2;
   jitter_buffer_put(jitterBuffer, &p);
  }
 }

 return 0;
}