Thread: Starting Linux Audio Programming OSS

Ah. So my first step is to not use OSS, and go with ALSA. I might even look into SDL Audio as well. Since I have programmed with SDL before.

I will go take a look at the ALSA libs then. Hopefully I won't get to confused at that.

Ok. I managed to compile the example program on ALSA. Been studying it for awhile. Looking at the ALSA API too.

Code:

#include <stdio.h>
	#include <stdlib.h>
	#include <alsa/asoundlib.h>
	      
	main (int argc, char *argv[])
	{
		int i;
		int err;
		short buf[128];
		snd_pcm_t *playback_handle;
		snd_pcm_hw_params_t *hw_params;
	
		if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
			fprintf (stderr, "cannot open audio device %s (%s)\n", 
				 argv[1],
				 snd_strerror (err));
			exit (1);
		}
		   
		if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
			fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
				 
		if ((err = snd_pcm_hw_params_any (playback_handle, hw_params)) < 0) {
			fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
			fprintf (stderr, "cannot set access type (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
			fprintf (stderr, "cannot set sample format (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_rate_resample (playback_handle, hw_params, 44100, 0)) < 0) {
			fprintf (stderr, "cannot set sample rate (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2)) < 0) {
			fprintf (stderr, "cannot set channel count (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) {
			fprintf (stderr, "cannot set parameters (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		snd_pcm_hw_params_free (hw_params);
	
		if ((err = snd_pcm_prepare (playback_handle)) < 0) {
			fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		for (i = 0; i < 10; ++i) {
			if ((err = snd_pcm_writei (playback_handle, buf, 128)) != 128) {
				fprintf (stderr, "write to audio interface failed (%s)\n",
					 snd_strerror (err));
				exit (1);
			}
		}
	
		snd_pcm_close (playback_handle);
		exit (0);
	}

So far I kinda understand what is going on here. Maby someone here can help me understand this next part. How exactly do I feed a decoded FLAC stream to ALSA. I take it when I want to write the decoded streams to ALSA I hae to use snd_pcm_writei function. Not sure.

I studied the FLAC C decode example code, and api for a few days.

SourceForge.net Repository - [flac] View of /flac/examples/c/decode/file/main.c

but still not sure how ALSA and libflac can be linked together to make it work.

Been searching on google for some help, but not finding much on programming ALSA.