comparison dec_audio.c @ 5190:59df6b778d78

Beta AAC decoding support, seeking totally broken yet, add philipps mpeg4 video in qt to ffmpeg4 although it's still buggy in decoding

5189:c663455448e8

			  settings */
  vorbis_comment   vc; /* struct that stores all the bitstream user comments */
  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
  vorbis_block     vb; /* local working space for packet->PCM decode */
} ov_struct_t;
#endif

#ifdef USE_LIBAVCODEC
#ifdef USE_LIBAVCODEC_SO
#include <libffmpeg/avcodec.h>

  /* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */
  // Is there always 1024 samples/frame ? ***** Albeu
  sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
#endif
  break;
case AFM_PCM:
case AFM_DVDPCM:
case AFM_ALAW:
  // PCM, aLaw
  sh_audio->audio_out_minsize=2048;

  vorbis_synthesis_init(&ov->vd,&ov->vi);
  vorbis_block_init(&ov->vd,&ov->vb);
  mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
} break;
#endif
#ifdef USE_LIBMAD
 case AFM_MAD:
   {
     printf("%s %s %s (%s)\n", mad_version, mad_copyright, mad_author, mad_build);

						    actually consumed */
	  }
	}
      } break;
#endif
      case AFM_PCM: // AVI PCM
        len=demux_read_data(sh_audio->ds,buf,minlen);
        break;
      case AFM_DVDPCM: // DVD PCM
      { int j;

    //      if(verbose){ printf("Resyncing AC3 audio...");fflush(stdout);}
          sh_audio->ac3_frame=ac3_decode_frame(); // resync
    //      if(verbose) printf(" OK!\n");
          break;
#endif
        case AFM_A52:
        case AFM_ACM:
        case AFM_DSHOW:
	case AFM_HWAC3:
          sh_audio->a_in_buffer_len=0;        // reset ACM/DShow audio buffer

5190:59df6b778d78

			  settings */
  vorbis_comment   vc; /* struct that stores all the bitstream user comments */
  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
  vorbis_block     vb; /* local working space for packet->PCM decode */
} ov_struct_t;
#endif

#ifdef HAVE_FAAD
#include <faad.h>
static faacDecHandle faac_hdec;
static faacDecFrameInfo faac_finfo;
static int faac_bytesconsumed = 0;
static unsigned char *faac_buffer;
/* configure maximum supported channels, *
 * this is theoretically max. 64 chans   */
#define FAAD_MAX_CHANNELS 6
#define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS)		       
#endif

#ifdef USE_LIBAVCODEC
#ifdef USE_LIBAVCODEC_SO
#include <libffmpeg/avcodec.h>

  /* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */
  // Is there always 1024 samples/frame ? ***** Albeu
  sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
#endif
  break;
case AFM_AAC:
  // AAC (MPEG2 Audio, MPEG4 Audio)
#ifndef HAVE_FAAD
  mp_msg(MSGT_DECAUDIO,MSGL_ERR,"Error: Cannot decode AAC data, because MPlayer was compiled without FAAD support\n"/*MSGTR_NoFAAD*/);
  driver=0;
#else  
  mp_msg(MSGT_DECAUDIO,MSGL_V,"Using FAAD to decode AAC content!\n"/*MSGTR_UseFAAD*/);
  // Samples per frame * channels per frame, this might not work with >2 chan AAC, need test samples! ::atmos
  sh_audio->audio_out_minsize=2048*2;
#endif  
  break;
case AFM_PCM:
case AFM_DVDPCM:
case AFM_ALAW:
  // PCM, aLaw
  sh_audio->audio_out_minsize=2048;

  vorbis_synthesis_init(&ov->vd,&ov->vi);
  vorbis_block_init(&ov->vd,&ov->vb);
  mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
} break;
#endif

#ifdef HAVE_FAAD
case AFM_AAC: {
  unsigned long faac_samplerate, faac_channels;
  faacDecConfigurationPtr faac_conf;
  faac_hdec = faacDecOpen();

#if 0
  /* Set the default object type and samplerate */
  /* This is useful for RAW AAC files */
  faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
  if(sh_audio->samplerate)
    faac_conf->defSampleRate = sh_audio->samplerate;
  /* XXX: is outputFormat samplesize of compressed data or samplesize of
   * decoded data, maybe upsampled? Also, FAAD support FLOAT output,
   * how do we handle that (FAAD_FMT_FLOAT)? ::atmos
   */
  if(sh_audio->samplesize)
    switch(sh_audio->samplesize){
      case 1: // 8Bit
	mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
      default:
      case 2: // 16Bit
	faac_conf->outputFormat = FAAD_FMT_16BIT;
	break;
      case 3: // 24Bit
	faac_conf->outputFormat = FAAD_FMT_24BIT;
	break;
      case 4: // 32Bit
	faac_conf->outputFormat = FAAD_FMT_32BIT;
	break;
    }
  faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.

  faacDecSetConfiguration(faac_hdec, faac_conf);
#endif

  if(faac_buffer == NULL)
    faac_buffer = (unsigned char*)malloc(FAAD_BUFFLEN);
  memset(faac_buffer, 0, FAAD_BUFFLEN);
  demux_read_data(sh_audio->ds, faac_buffer, FAAD_BUFFLEN);

  /* init the codec */
  if((faac_bytesconsumed = faacDecInit(faac_hdec, faac_buffer, &faac_samplerate, &faac_channels)) < 0) {
    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
    faacDecClose(faac_hdec);
    free(faac_buffer);
    faac_buffer = NULL;
    driver = 0;
  } else {
    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", faac_bytesconsumed); // XXX: remove or move to debug!
    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz  channels: %d\n", faac_samplerate, faac_channels);
    sh_audio->channels = faac_channels;
    sh_audio->samplerate = faac_samplerate;
    sh_audio->i_bps = 128*1000/8; // XXX: HACK!!! There's currently no way to get bitrate from libfaad2! ::atmos
  }  
	    
} break;		
#endif

#ifdef USE_LIBMAD
 case AFM_MAD:
   {
     printf("%s %s %s (%s)\n", mad_version, mad_copyright, mad_author, mad_build);

						    actually consumed */
	  }
	}
      } break;
#endif
		       
#ifdef HAVE_FAAD		       
      case AFM_AAC: {
	int /*i,*/ k, j = 0;	      
	void *faac_sample_buffer;
	
	len = 0;
	while(len < minlen) {
	  /* update buffer */
    	  if (faac_bytesconsumed > 0) {
	    for (k = 0; k < (FAAD_BUFFLEN - faac_bytesconsumed); k++)
	      faac_buffer[k] = faac_buffer[k + faac_bytesconsumed];
	    demux_read_data(sh_audio->ds, faac_buffer + (FAAD_BUFFLEN) - faac_bytesconsumed, faac_bytesconsumed);
	    faac_bytesconsumed = 0;
	  }
	  /*for (i = 0; i < 16; i++)
	    printf ("%02X ", faac_buffer[i]);
	  printf ("\n");*/
	  do {
	    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, faac_buffer+j);
	    /* update buffer index after faacDecDecode */
	    faac_bytesconsumed += faac_finfo.bytesconsumed;
	    if(faac_finfo.error > 0) {
	      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Trying to resync!\n");
	      j++;
	    } else
	      break;
	  } while(j < FAAD_BUFFLEN);	  


	  if(faac_finfo.error > 0) {
	    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
	      faacDecGetErrorMessage(faac_finfo.error));
	  } else if (faac_finfo.samples == 0)
	    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
	  else {
	    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%dBytes)!\n", faac_finfo.samples*faac_finfo.channels);
	    memcpy(buf+len,faac_sample_buffer, faac_finfo.samples*faac_finfo.channels);
	    len += faac_finfo.samples*faac_finfo.channels;
	  }
	}

      } break;
#endif		    
      case AFM_PCM: // AVI PCM
        len=demux_read_data(sh_audio->ds,buf,minlen);
        break;
      case AFM_DVDPCM: // DVD PCM
      { int j;

    //      if(verbose){ printf("Resyncing AC3 audio...");fflush(stdout);}
          sh_audio->ac3_frame=ac3_decode_frame(); // resync
    //      if(verbose) printf(" OK!\n");
          break;
#endif
#ifdef HAVE_FAAD
	case AFM_AAC:
	  //if(faac_buffer != NULL)
	  faac_bytesconsumed = 0;
	  memset(faac_buffer, 0, FAAD_BUFFLEN);
  	  //demux_read_data(sh_audio->ds, faac_buffer, FAAD_BUFFLEN);
	  break;
#endif
        case AFM_A52:
        case AFM_ACM:
        case AFM_DSHOW:
	case AFM_HWAC3:
          sh_audio->a_in_buffer_len=0;        // reset ACM/DShow audio buffer