Sound Recording By Using ALSA-lib Pls Help

[gokseld7]

Hello guys,
I am new in Linux programming,but i need to write a c program which uses the ALSA library and records sound to a file.In fact I found a code does this,but the recorded sound file is in .pcm format and i can't play it. The code is below:

   /*


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 thing i did not understand,when i use arecord >sample.pcm command in terminal,i can play the pcm file well.But in this program i can't.Is that because the frequency and mono/stereo stuff? How can i fix that problem?

[Nominal Animal]

when i use arecord >sample.pcm command in terminal,i can play the pcm file well.

That is because by default, arecord outputs WAV format audio (uncompressed), not raw data.

But in this program i can't.

You can, if you tell aplay the exact format:

aplay -t raw -f S16_LE -c2 -r44100 sample.pcm

A better solution is to make your program output WAV data, too. All you need to do is construct a 44-byte header chunk (which does have to include the total number of samples in the file -- but you already know that beforehand, so it's not a problem), and save it before the sample data.

The format is described in various places on the internet, for example here and here. To get you started, I recommend you begin with a helper function:

static void set16le(void *const ptr, const unsigned int value)
{
    unsigned char *const data = ptr;
    data[0] = value & 255U;
    data[1] = (value >> 8) & 255U;
}

static void set32le(void *const ptr, const unsigned int value)
{
    unsigned char *const data = ptr;
    data[0] = value & 255U;
    data[1] = (value >> 8) & 255U;
    data[2] = (value >> 16) & 255U;
    data[3] = (value >> 24) & 255U;
}

int write_wav_header_s16le(FILE *const out, const int samplesperchannel, const int channels, const int rate)
{
    unsigned char header[44];

    header[0] = 'R';
    header[1] = 'I';
    header[2] = 'F';
    header[3] = 'F';
    set32le(header + 4, 36U + 2U*channels*samplesperchannel);
    header[8] = 'W';
    header[9] = 'A';
    header[10] = 'V';
    header[11] = 'E';
    header[12] = 'f';
    header[13] = 'm';
    header[14] = 't';
    header[15] = ' ';
    set32le(header + 16, 16U);
    set16le(header + 20, 1U); /* PCM format */
    set16le(header + 22, channels);
    set32le(header + 24, rate);
    set32le(header + 28, rate * 2U * channels);
    set16le(header + 32, 2U * channels);
    set16le(header + 34, 16U); /* Bits per sample */
    header[36] = 'd';
    header[37] = 'a';
    header[38] = 't';
    header[39] = 'a';
    set32le(header + 40, 2U*channels*samplesperchannel);

    if (fwrite(header, 44, 1, out) != 1)
        return -1;
    else
        return 0;
}

This is untested; I suggest you read the aforementioned links, and compare the header part to what e.g. arecord generates, using for example dd if=sound.pcm bs=1 count=44 status=none | hexdump -C or similar.

To use, you just need to call write_wav_header_s16le() before writing out the data.

Also, instead of the low-level write() call on lines 115-118, you should use

if (fwrite(buffer, size, 1, stdout) != 1) {
    fflush(stdout);
    fprintf(stderr, "Error writing to standard output.\n");
    break;
}

No need to prematurely optimize your code. Make sure it works first, then consider if avoiding the overhead (very small for binary data) with standard I/O is worth it.

[gokseld7]

Thank you so much for all advises,i played that pcm file,now i'm gonna try to make the output waw