view libmpdemux/audio_in.c @ 13568:1cb0e1833515

Currently vbeGetProtModeInfo call the 0x4f0a function of int 10h the get a simple 32 bits protected mode interface to some VESA functions. This protected mode interface is interesting because it's quicker than the raw int 10h interface. Unfortunatly, begining with VBE 3.0, the 0x4f0a function is optional, and some video cards don't implement it (3dfx, intel 845/855/865...). This protected mode interface is then only used in vbeSetWindow and vbeSetDisplayStart : ?- vbeSetWindow already implement an alternative methode if protected mode interface is not available. ?- vbeSetDisplayStart also contain an alternative implementation, but this one is disabled with a #if 0. I don't exactly know why because it works well ! So currently, cards which don't have the 0x4f0a function are not supported. This patch correct this. ?- vbeGetProtModeInfo failure is not fatal. ?- vbeSetDisplayStart has it's alternative implementation reenabled. ? ?it's used only with cards which don't have the 0x4f0a function ? ?so this won't make any difference for cards which were already ? ?working. This patch also make the failure of vbeGetModeInfo not fatal. The VBE 3.0 standard state that GetModeInfo can fail with some mode which are listed as supported if the mode can't be used in the current situation (not enough video memory for example). So a failure of vbeGetModeInfo don't mean that other modes won't work and should really not be fatal. patch by Aurelien Jacobs <[e-mail]>

#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