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mirror of https://github.com/mpv-player/mpv synced 2024-11-18 21:16:10 +01:00

Support for decoder specific config from mp4 header for AAC decoder.

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5305 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
atmos4 2002-03-24 03:08:20 +00:00
parent 582d209217
commit a699ffad78

View File

@ -842,42 +842,54 @@ case AFM_AAC: {
faacDecConfigurationPtr faac_conf;
faac_hdec = faacDecOpen();
#if 1
/* 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: 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);
faac_buffer = (unsigned char*)calloc(1,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) {
// If we don't get the ES descriptor, try manual config
if(!sh_audio->codecdata_len) {
#if 1
/* 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: 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
/* init the codec */
faac_bytesconsumed = faacDecInit(faac_hdec, faac_buffer,
&faac_samplerate, &faac_channels);
} else { // We have ES DS in codecdata
/*int i;
for(i = 0; i < sh_audio->codecdata_len; i++)
printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
faac_bytesconsumed = faacDecInit2(faac_hdec, sh_audio->codecdata,
sh_audio->codecdata_len, &faac_samplerate, &faac_channels);
}
if(faac_bytesconsumed < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
faacDecClose(faac_hdec);
free(faac_buffer);
@ -890,9 +902,9 @@ case AFM_AAC: {
sh_audio->samplerate = faac_samplerate;
if(!sh_audio->i_bps) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
sh_audio->i_bps = 128*1000/8; // XXX: HACK!!! There's currently no way to get bitrate from libfaad2! ::atmos
sh_audio->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
} else
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s rate from MP4 header!\n",sh_audio->i_bps*8/1000);
mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh_audio->i_bps*8/1000);
}
} break;