view libmpdemux/audio_in.c @ 8763:19e96e60a3d0

Speed optimizations (runs twise as fast) and bugfix (wrong cutoff frequency buffer over run noise and garbeled output when wrong input format)
author anders
date Sat, 04 Jan 2003 06:19:25 +0000
parents 772d6d27fd66
children 31f12f99118b
line wrap: on
line source

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

#include "config.h"

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

#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) {
#ifdef HAVE_ALSA9	  
    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) {
#ifdef HAVE_ALSA9	  
    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) {
#ifdef HAVE_ALSA9	  
    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) {
#ifdef HAVE_ALSA9	  
    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)
{
#ifdef HAVE_ALSA9	  
    int i;
#endif
    if (ai->setup) return -1;
    switch (ai->type) {
#ifdef HAVE_ALSA9	  
    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) {
#ifdef HAVE_ALSA9	  
	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) {
#ifdef HAVE_ALSA9	  
    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) {
#ifdef HAVE_ALSA9	  
    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