view stream/audio_in.c @ 24590:2c238fa777ff

ao_alsa: Fix get_space() return values larger than buffersize After a buffer underrun the ALSA get_space() function sometimes returned values larger than the ao had set in ao_data.buffersize. Fix this by replacing the old check against MAX_OUTBURST by one against ao_data.buffersize. There should be no need for the MAX_OUTBURST check; the current MPlayer side should no longer have any constant limit on the amount of data an ao can buffer or request at once. The get_space() values larger than ao_data.buffersize triggered errors in audio decoding causing the current attempt to fill audio buffers to be aborted. I'm not sure how often that caused behavior noticeably worse then an underrun already is.

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

#include "config.h"

#include "audio_in.h"
#include "mp_msg.h"
#include "help_mp.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, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret));
		if (ret == -EPIPE) {
		    if (ai_alsa_xrun(ai) == 0) {
			mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut);
		    } else {
			mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover);
		    }
		}
	    } else {
		mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples);
	    }
	    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, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno));
	    } else {
		mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples);
	    }
	    return -1;
	}
	return ret;
#endif
    default:
	return -1;
    }
}