view libmpdemux/audio_in.c @ 15205:19243f85e164

nico partially fixed the bug i reported; here's the rest of the fix. basically demux_audio was mixing data in its header buffer in a bogus manner, whereby it could sometimes "make up" valid mpeg headers where no such header actually occurred in the file. it should be correct now. btw these changes also fix the bug where mplayer reports huge initial cpu usage for sound when playing mp3 files.

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

#include "config.h"

#if defined(USE_TV) && (defined(HAVE_TV_V4L) || defined(HAVE_TV_V4L2))

#include "audio_in.h"
#include "mp_msg.h"
#include <string.h>
#include <errno.h>

// sanitizes ai structure before calling other functions
int audio_in_init(audio_in_t *ai, int type)
{
    ai->type = type;
    ai->setup = 0;

    ai->channels = -1;
    ai->samplerate = -1;
    ai->blocksize = -1;
    ai->bytes_per_sample = -1;
    ai->samplesize = -1;

    switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    case AUDIO_IN_ALSA:
	ai->alsa.handle = NULL;
	ai->alsa.log = NULL;
	ai->alsa.device = strdup("default");
	return 0;
#endif
#ifdef USE_OSS_AUDIO
    case AUDIO_IN_OSS:
	ai->oss.audio_fd = -1;
	ai->oss.device = strdup("/dev/dsp");
	return 0;
#endif
    default:
	return -1;
    }
}

int audio_in_setup(audio_in_t *ai)
{
    
    switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    case AUDIO_IN_ALSA:
	if (ai_alsa_init(ai) < 0) return -1;
	ai->setup = 1;
	return 0;
#endif
#ifdef USE_OSS_AUDIO
    case AUDIO_IN_OSS:
	if (ai_oss_init(ai) < 0) return -1;
	ai->setup = 1;
	return 0;
#endif
    default:
	return -1;
    }
}

int audio_in_set_samplerate(audio_in_t *ai, int rate)
{
    switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    case AUDIO_IN_ALSA:
	ai->req_samplerate = rate;
	if (!ai->setup) return 0;
	if (ai_alsa_setup(ai) < 0) return -1;
	return ai->samplerate;
#endif
#ifdef USE_OSS_AUDIO
    case AUDIO_IN_OSS:
	ai->req_samplerate = rate;
	if (!ai->setup) return 0;
	if (ai_oss_set_samplerate(ai) < 0) return -1;
	return ai->samplerate;
#endif
    default:
	return -1;
    }
}

int audio_in_set_channels(audio_in_t *ai, int channels)
{
    switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    case AUDIO_IN_ALSA:
	ai->req_channels = channels;
	if (!ai->setup) return 0;
	if (ai_alsa_setup(ai) < 0) return -1;
	return ai->channels;
#endif
#ifdef USE_OSS_AUDIO
    case AUDIO_IN_OSS:
	ai->req_channels = channels;
	if (!ai->setup) return 0;
	if (ai_oss_set_channels(ai) < 0) return -1;
	return ai->channels;
#endif
    default:
	return -1;
    }
}

int audio_in_set_device(audio_in_t *ai, char *device)
{
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    int i;
#endif
    if (ai->setup) return -1;
    switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    case AUDIO_IN_ALSA:
	if (ai->alsa.device) free(ai->alsa.device);
	ai->alsa.device = strdup(device);
	/* mplayer cannot handle colons in arguments */
	for (i = 0; i < (int)strlen(ai->alsa.device); i++) {
	    if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':';
	}
	return 0;
#endif
#ifdef USE_OSS_AUDIO
    case AUDIO_IN_OSS:
	if (ai->oss.device) free(ai->oss.device);
	ai->oss.device = strdup(device);
	return 0;
#endif
    default:
	return -1;
    }
}

int audio_in_uninit(audio_in_t *ai)
{
    if (ai->setup) {
	switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
	case AUDIO_IN_ALSA:
	    if (ai->alsa.log)
		snd_output_close(ai->alsa.log);
	    if (ai->alsa.handle) {
		snd_pcm_close(ai->alsa.handle);
	    }
	    ai->setup = 0;
	    return 0;
#endif
#ifdef USE_OSS_AUDIO
	case AUDIO_IN_OSS:
	    close(ai->oss.audio_fd);
	    ai->setup = 0;
	    return 0;
#endif
	}
    }
    return -1;
}

int audio_in_start_capture(audio_in_t *ai)
{
    switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    case AUDIO_IN_ALSA:
	return snd_pcm_start(ai->alsa.handle);
#endif
#ifdef USE_OSS_AUDIO
    case AUDIO_IN_OSS:
	return 0;
#endif
    default:
	return -1;
    }
}

int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer)
{
    int ret;
    
    switch (ai->type) {
#if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
    case AUDIO_IN_ALSA:
	ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size);
	if (ret != ai->alsa.chunk_size) {
	    if (ret < 0) {
		mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret));
		if (ret == -EPIPE) {
		    if (ai_alsa_xrun(ai) == 0) {
			mp_msg(MSGT_TV, MSGL_ERR, "Recovered from cross-run, some frames may be left out!\n");
		    } else {
			mp_msg(MSGT_TV, MSGL_ERR, "Fatal error, cannot recover!\n");
		    }
		}
	    } else {
		mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
	    }
	    return -1;
	}
	return ret;
#endif
#ifdef USE_OSS_AUDIO
    case AUDIO_IN_OSS:
	ret = read(ai->oss.audio_fd, buffer, ai->blocksize);
	if (ret != ai->blocksize) {
	    if (ret < 0) {
		mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno));
	    } else {
		mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
	    }
	    return -1;
	}
	return ret;
#endif
    default:
	return -1;
    }
}

#endif