alsa-record-example.cc

/* 
  A Minimal Capture Program

  This program opens an audio interface for capture, configures it for
  stereo, 16 bit, 44.1kHz, interleaved conventional read/write
  access. Then its reads a chunk of random data from it, and exits. It
  isn't meant to be a real program.

  From on Paul David's tutorial : http://equalarea.com/paul/alsa-audio.html

  Fixes rate and buffer problems

  sudo apt-get install libasound2-dev
  gcc -o alsa-record-example -lasound alsa-record-example.c && ./alsa-record-example hw:0
  
  [X] /usr/bin/aarch64-linux-gnu-gcc -o alsa-record-example -std=c11 -I/usr/include/ -L/usr/lib/aarch64-linux-gnu/libasound.so.2.0.0 alsa-record-example.c 
  
  g++ alsa-record-example.cc -I/usr/include/  -o alsa-record-example -lm -ldl -lasound 
  gcc alsa-record-example.c -I/usr/include/  -o alsa-record-example -lm -ldl -lasound   
  ./alsa-record-example hw:2,0

  - More information: https://vovkos.github.io/doxyrest/samples/alsa/page_pcm.html#doxid-pcm
*/
  
  
#include <signal.h>
#include <chrono>
#include <stdio.h>
#include <stdlib.h>
#include <stdbool.h>
#include <alsa/asoundlib.h>

static bool running = true;
static snd_pcm_t* capture_handle = NULL;
/* Signals handling */
static void handle_sigterm(int signo) { running = false; }

void init_signal() {
  struct sigaction sa;
  sa.sa_flags = 0;

  sigemptyset(&sa.sa_mask);
  sa.sa_handler = handle_sigterm;
  sigaction(SIGTERM, &sa, NULL);
  sigaction(SIGINT, &sa, NULL);
}

void snd_param_init(snd_pcm_t **capture_handle,
                    const char* name, 
                    int buffer_frames, 
                    unsigned int rate, 
                    snd_pcm_hw_params_t *hw_params, 
                    snd_pcm_format_t format) {
  int err;

  if ((err = snd_pcm_open (capture_handle, name, SND_PCM_STREAM_CAPTURE, 0)) < 0) {
    fprintf (stderr, "cannot open audio device %s (%s)\n", 
             name,
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "audio interface opened\n");
		   
  if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
    fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params allocated\n");
				 
  if ((err = snd_pcm_hw_params_any (*capture_handle, hw_params)) < 0) {
    fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params initialized\n");
	
  if ((err = snd_pcm_hw_params_set_access (*capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
    fprintf (stderr, "cannot set access type (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params access setted\n");
	
  if ((err = snd_pcm_hw_params_set_format (*capture_handle, hw_params, format)) < 0) {
    fprintf (stderr, "cannot set sample format (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params format setted\n");
	
  if ((err = snd_pcm_hw_params_set_rate_near (*capture_handle, hw_params, &rate, 0)) < 0) {
    fprintf (stderr, "cannot set sample rate (%s)\n",
             snd_strerror (err));
    exit (1);
  }
	
  fprintf(stdout, "hw_params rate setted\n");

  if ((err = snd_pcm_hw_params_set_channels (*capture_handle, hw_params, 1)) < 0) {
    fprintf (stderr, "cannot set channel count (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params channels setted\n");
	
  if ((err = snd_pcm_hw_params (*capture_handle, hw_params)) < 0) {
    fprintf (stderr, "cannot set parameters (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params setted\n");
	
  snd_pcm_hw_params_free (hw_params);

  fprintf(stdout, "hw_params freed\n");
	
  if ((err = snd_pcm_prepare (*capture_handle)) < 0) {
    fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
             snd_strerror (err));
    exit (1);
  }  
}

int main (int argc, char *argv[])
{
  int i;
  int err;
  char *buffer;
  int buffer_frames = 22050;
  unsigned int rate = 44100;
  snd_pcm_hw_params_t *hw_params;
  snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;

  init_signal();
  snd_param_init(&capture_handle, argv[1], buffer_frames, rate, hw_params, format);

  fprintf(stdout, "audio interface prepared\n");

  buffer = (char*) malloc(128 * snd_pcm_format_width(format) / 8 * 2);

  fprintf(stdout, "buffer allocated\n");

  while(running){
    auto start = std::chrono::high_resolution_clock::now();
    if ((err = snd_pcm_readi (capture_handle, buffer, buffer_frames)) != buffer_frames) {
      fprintf (stderr, "read from audio interface failed (%s)\n",
               err, snd_strerror (err));
    }
    
    auto end = std::chrono::high_resolution_clock::now();
    auto duration = std::chrono::duration_cast<std::chrono::milliseconds>(end - start).count();
    fprintf(stdout, "read %d done %ld ms \n", buffer[0], duration);
  }

  free(buffer);
  fprintf(stdout, "buffer freed\n");
	
  snd_pcm_close (capture_handle);
  fprintf(stdout, "audio interface closed\n");
  return 0;
}