I'm having a lot of trouble writing random bytes to an ALSA playback device using libasound
. Eventually, my goal is to be able to route playback stream over the network and have it played on a remote device.
The code presented in this question reads a WAV file into memory and writes it to the driver via snd_pcm_writei
and it works. However, a crucial difference between what this code does and what I'm trying to do is the fact that I don't have all the data available to me right away. I'm looking to stream data as it becomes available.
Adapting the above sample code to fit my needs, I end up with this:
#include <stdio.h>
#include <unistd.h>
#include <alsa/asoundlib.h>
static snd_pcm_t *PlaybackHandle;
int init_playback(const char *device, int samplerate, int channels) {
int err;
printf("Init parameters: %s %d %d\n", device, samplerate, channels);
if((err = snd_pcm_open(&PlaybackHandle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
printf("Can't open audio %s: %s\n", device, snd_strerror(err));
return -1;
}
if ((err = snd_pcm_set_params(PlaybackHandle, SND_PCM_FORMAT_S16, SND_PCM_ACCESS_RW_INTERLEAVED, channels, samplerate, 1, 500000)) < 0) {
printf("Can't set sound parameters: %s\n", snd_strerror(err));
return -1;
}
return 0;
}
int play_bytes(const void *bytes, int len) {
snd_pcm_uframes_t frames, count;
snd_pcm_uframes_t bufsize, period_size;
frames = 0;
count = 0;
snd_pcm_prepare(PlaybackHandle);
snd_pcm_get_params(PlaybackHandle, &bufsize, &period_size);
printf("bufsize=%d\n", (int) bufsize);
do {
int remaining = len - count;
int buflen = remaining < bufsize ? remaining : bufsize;
frames = snd_pcm_writei(PlaybackHandle, bytes + count, buflen);
// If an error, try to recover from it
if (frames == -EPIPE) {
printf("EPIPE\n");
snd_pcm_prepare(PlaybackHandle);
}
if (frames < 0) {
printf("Recovering\n");
frames = snd_pcm_recover(PlaybackHandle, frames, 0);
}
if (frames < 0)
{
printf("Error playing wave: %s\n", snd_strerror(frames));
break;
}
// Update our pointer
count += frames;
//printf("count=%d len=%d\n", (int)count, len);
} while (count < len);
// Wait for playback to completely finish
if (count == len)
snd_pcm_drain(PlaybackHandle);
return 0;
}
int close_playback() {
snd_pcm_close(PlaybackHandle);
return 0;
}
int main(int argc, char **argv) {
if(argc < 1) {
printf("Usage: %s <WAV-file>\n", argv[0]);
return -1;
}
int fd;
unsigned long long len;
fd = open(argv[1], O_RDONLY);
// Find the length
len = lseek(fd, 0, SEEK_END);
// Skip the first 44 bytes (header)
lseek(fd, 44, SEEK_SET);
len -= 44;
char *data = malloc(len);
read(fd, data, len);
init_playback("default", 44100, 2);
play_bytes(data, len);
close_playback();
return 0;
}
This code can be compiled with gcc playback.c -o playback -lasound
. The WAV file I'm using can be found here.
When I run this code snippet, where I chunk the incoming data based on bufsize
there are fragments of audio that is repeated in the playback depending on the chunk size. A large chunk size yields fewer repetitions than a small chunk size. Combining this with how the audio sounds, I believe that a small fragment at the end of each chunk is being repeated.
The parameters I used were:
Why does sending the entire WAV file in one shot work whereas sending chunks of it not work? How do I send chunks of audio data to the driver and have it play properly?
frames = snd_pcm_writei(PlaybackHandle, bytes + count, buflen);
count += frames;
snd_pcm_writei()
measures the size in frames, but you treat it as bytes. So you skip over only one fourth of the data that was just played.