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

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


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@22194 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

libavcodec/aac_ac3_parser.c

@@ -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;
 }