annotate stream/audio_in.c @ 29888:5c39c41f38e8

Deobfuscate the special hack to disable cache for live555. Cache can not be used for it, since it does not provide any data stream, the data is provided to the demuxer "behind MPlayer's back".

#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) {
#ifdef CONFIG_ALSA
    case AUDIO_IN_ALSA:
	ai->alsa.handle = NULL;
	ai->alsa.log = NULL;
	ai->alsa.device = strdup("default");
	return 0;
#endif
#ifdef CONFIG_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 CONFIG_ALSA
    case AUDIO_IN_ALSA:
	if (ai_alsa_init(ai) < 0) return -1;
	ai->setup = 1;
	return 0;
#endif
#ifdef CONFIG_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 CONFIG_ALSA
    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 CONFIG_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 CONFIG_ALSA
    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 CONFIG_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 CONFIG_ALSA
    int i;
#endif
    if (ai->setup) return -1;
    switch (ai->type) {
#ifdef CONFIG_ALSA
    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 CONFIG_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 CONFIG_ALSA
	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 CONFIG_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 CONFIG_ALSA
    case AUDIO_IN_ALSA:
	return snd_pcm_start(ai->alsa.handle);
#endif
#ifdef CONFIG_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 CONFIG_ALSA
    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 CONFIG_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;
    }
}