Mercurial > mplayer.hg
changeset 5304:6175c9cfab11
Support for decoder specific config from mp4 header for AAC decoder.
author | atmos4 |
---|---|
date | Sun, 24 Mar 2002 03:08:20 +0000 |
parents | 534f16f50c17 |
children | 77ac28af44ec |
files | dec_audio.c |
diffstat | 1 files changed, 45 insertions(+), 33 deletions(-) [+] |
line wrap: on
line diff
--- a/dec_audio.c Sun Mar 24 03:07:18 2002 +0000 +++ b/dec_audio.c Sun Mar 24 03:08:20 2002 +0000 @@ -842,42 +842,54 @@ faacDecConfigurationPtr faac_conf; faac_hdec = faacDecOpen(); + if(faac_buffer == NULL) + faac_buffer = (unsigned char*)calloc(1,FAAD_BUFFLEN); + demux_read_data(sh_audio->ds, faac_buffer, FAAD_BUFFLEN); + + // 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. + /* 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); + 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 */ + faac_bytesconsumed = faacDecInit(faac_hdec, faac_buffer, + &faac_samplerate, &faac_channels); - /* init the codec */ - if((faac_bytesconsumed = faacDecInit(faac_hdec, faac_buffer, &faac_samplerate, &faac_channels)) < 0) { + } 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 @@ 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;