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| author | rfelker <rfelker@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2005-03-29 17:59:08 +0000 |
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| committer | rfelker <rfelker@b3059339-0415-0410-9bf9-f77b7e298cf2> | 2005-03-29 17:59:08 +0000 |
| commit | 9a873224f674d879546eae9febf5c4292b8de61f (patch) | |
| tree | 18c4257c648710b6acd8f84f0a778423b26a423b /libmpcodecs | |
| parent | d8c4e2dbe8b2677e3f2d0c6671927d2c044d389e (diff) | |
| download | mpv-9a873224f674d879546eae9febf5c4292b8de61f.tar.bz2 mpv-9a873224f674d879546eae9febf5c4292b8de61f.tar.xz | |
step 1 of fixing ad_faad:
use internal downmixing just like liba52 does if the output is <= 2 channels
actually this is broken since it makes it impossible to manually use
af_pan; however liba52 already has that limitation, and without this
patch, aac audio comes out TOTALLY wrong on 2-channel systems.
hopefully someone will find a better solution later.
next up: making ad_faad reorder the channels according to what mplayer
expects, so they won't all come out the wrong speakers...
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@15020 b3059339-0415-0410-9bf9-f77b7e298cf2
Diffstat (limited to 'libmpcodecs')
| -rw-r--r-- | libmpcodecs/ad_faad.c | 11 |
1 files changed, 10 insertions, 1 deletions
diff --git a/libmpcodecs/ad_faad.c b/libmpcodecs/ad_faad.c index 91253aba6c..efc91ea7e3 100644 --- a/libmpcodecs/ad_faad.c +++ b/libmpcodecs/ad_faad.c @@ -66,6 +66,7 @@ static int aac_probe(unsigned char *buffer, int len) return pos; } +extern int audio_output_channels; static int init(sh_audio_t *sh) { unsigned long faac_samplerate; @@ -90,6 +91,7 @@ static int init(sh_audio_t *sh) /* XXX: FAAD support FLOAT output, how do we handle * that (FAAD_FMT_FLOAT)? ::atmos */ + if (audio_output_channels <= 2) faac_conf->downMatrix = 1; switch(sh->samplesize){ case 1: // 8Bit mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); @@ -129,6 +131,12 @@ static int init(sh_audio_t *sh) // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi } else { // We have ES DS in codecdata + faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec); + if (audio_output_channels <= 2) { + faac_conf->downMatrix = 1; + faacDecSetConfiguration(faac_hdec, faac_conf); + } + /*int i; for(i = 0; i < sh_audio->codecdata_len; i++) printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/ @@ -145,6 +153,7 @@ static int init(sh_audio_t *sh) mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug! mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels); sh->channels = faac_channels; + if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1; sh->samplerate = faac_samplerate; sh->samplesize=2; //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; @@ -245,8 +254,8 @@ static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) int buflen=ds_get_packet(sh->ds, &bufptr); if(buflen<=0) break; faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen); -// printf("FAAC decoded %d of %d (err: %d) \n",faac_finfo.bytesconsumed,buflen,faac_finfo.error); } + //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]); if(faac_finfo.error > 0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n", |