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AAC parser: Don't write channels, sample rate, and frame size each frame. 

Thanks to backwards compatible HE-AAC signalling these values are unreliable.

Originally committed as revision 22194 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Alex Converse 2010-03-04 02:30:51 +00:00
parent 38768cb70a
commit ca1fa4144f
1 changed files with 11 additions and 3 deletions
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14
libavcodec/aac_ac3_parser.c
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@ -71,10 +71,16 @@ get_next:
*poutbuf_size = buf_size;
/* update codec info */
avctx->sample_rate = s->sample_rate;
if(s->codec_id)
avctx->codec_id = s->codec_id;
/* Due to backwards compatible HE-AAC the sample rate, channel count,
and total number of samples found in an AAC ADTS header are not
reliable. Bit rate is still accurate because the total frame duration in
seconds is still correct (as is the number of bits in the frame). */
if (avctx->codec_id != CODEC_ID_AAC) {
avctx->sample_rate = s->sample_rate;
/* allow downmixing to stereo (or mono for AC-3) */
if(avctx->request_channels > 0 &&
avctx->request_channels < s->channels &&
@ -83,12 +89,14 @@ get_next:
(avctx->codec_id == CODEC_ID_AC3 ||
avctx->codec_id == CODEC_ID_EAC3)))) {
avctx->channels = avctx->request_channels;
} else if (avctx->codec_id != CODEC_ID_AAC || s->channels) {
} else {
avctx->channels = s->channels;
avctx->channel_layout = s->channel_layout;
}
avctx->bit_rate = s->bit_rate;
avctx->frame_size = s->samples;
}
avctx->bit_rate = s->bit_rate;
return i;
}