WMAL: Shift output samples by the specified number of padding zeroes.

[ffmpeg] / libavcodec / aac_ac3_parser.c

diff --git a/libavcodec/aac_ac3_parser.c b/libavcodec/aac_ac3_parser.c
index f28f3f26782b382d15338878ce9759299a0354eb..8132ce899a811726f35bdbd1e32654acc5324215 100644 (file)
--- a/libavcodec/aac_ac3_parser.c
+++ b/libavcodec/aac_ac3_parser.c
@@ -1,22 +1,22 @@
 /*
- * Common AAC and AC3 parser
- * Copyright (c) 2003 Fabrice Bellard.
- * Copyright (c) 2003 Michael Niedermayer.
+ * Common AAC and AC-3 parser
+ * Copyright (c) 2003 Fabrice Bellard
+ * Copyright (c) 2003 Michael Niedermayer
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -49,9 +49,10 @@ get_next:
             if(len<=0){
                 i=END_NOT_FOUND;
             }else{
+                s->state=0;
                 i-= s->header_size -1;
                 s->remaining_size = len;
-                if(!new_frame_start){
+                if(!new_frame_start || pc->index+i<=0){
                     s->remaining_size += i;
                     goto get_next;
                 }
@@ -70,19 +71,33 @@ get_next:
     *poutbuf_size = buf_size;
 
     /* update codec info */
-    avctx->sample_rate = s->sample_rate;
-    /* allow downmixing to stereo (or mono for AC3) */
-    if(avctx->request_channels > 0 &&
-            avctx->request_channels < s->channels &&
-            (avctx->request_channels <= 2 ||
-            (avctx->request_channels == 1 &&
-            avctx->codec_id == CODEC_ID_AC3))) {
-        avctx->channels = avctx->request_channels;
-    } else {
-        avctx->channels = s->channels;
+    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 &&
+                (avctx->request_channels <= 2 ||
+                (avctx->request_channels == 1 &&
+                (avctx->codec_id == CODEC_ID_AC3 ||
+                 avctx->codec_id == CODEC_ID_EAC3)))) {
+            avctx->channels = avctx->request_channels;
+        } else {
+            avctx->channels = s->channels;
+            avctx->channel_layout = s->channel_layout;
+        }
+        s1->duration = s->samples;
+        avctx->audio_service_type = s->service_type;
     }
+
     avctx->bit_rate = s->bit_rate;
-    avctx->frame_size = s->samples;
 
     return i;
 }