[FFmpeg-cvslog] Make AAC in Ogg (ogm) work.

[Reimar Döffinger]

ffmpeg | branch: master | Reimar Döffinger <Reimar.Doeffinger at gmx.de> | Mon Feb  6 22:04:46 2012 +0100| [7c8d477299c9b5e89fc30ed22f9e42b50761342c] | committer: Reimar Döffinger

Make AAC in Ogg (ogm) work.

This needs the extradata to be extracted.
The approach used is the one MPlayer uses, though it is
unclear whether the 4 bytes extradata that are skipped
should be skipped always or only for AAC.
The AAC parser must be disabled, too, otherwise playback
still does not work.
Fixes trac issue #547.

Signed-off-by: Reimar Döffinger <Reimar.Doeffinger at gmx.de>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=7c8d477299c9b5e89fc30ed22f9e42b50761342c
---

 libavformat/oggparseogm.c |   21 ++++++++++++++++++---
 1 files changed, 18 insertions(+), 3 deletions(-)

diff --git a/libavformat/oggparseogm.c b/libavformat/oggparseogm.c
index 0a8a7c6..36f61c7 100644
--- a/libavformat/oggparseogm.c
+++ b/libavformat/oggparseogm.c
@@ -23,6 +23,7 @@
 **/
 
 #include <stdlib.h>
+#include "libavutil/avassert.h"
 #include "libavutil/intreadwrite.h"
 #include "libavcodec/get_bits.h"
 #include "libavcodec/bytestream.h"
@@ -40,6 +41,7 @@ ogm_header(AVFormatContext *s, int idx)
     const uint8_t *p = os->buf + os->pstart;
     uint64_t time_unit;
     uint64_t spu;
+    uint32_t size;
 
     if(!(*p & 1))
         return 0;
@@ -67,11 +69,13 @@ ogm_header(AVFormatContext *s, int idx)
             acid[4] = 0;
             cid = strtol(acid, NULL, 16);
             st->codec->codec_id = ff_codec_get_id(ff_codec_wav_tags, cid);
-            st->need_parsing = AVSTREAM_PARSE_FULL;
+            // our parser completely breaks AAC in Ogg
+            if (st->codec->codec_id != CODEC_ID_AAC)
+                st->need_parsing = AVSTREAM_PARSE_FULL;
         }
 
-        p += 4;                     /* useless size field */
-
+        size        = bytestream_get_le32(&p);
+        size        = FFMIN(size, os->psize);
         time_unit   = bytestream_get_le64(&p);
         spu         = bytestream_get_le64(&p);
         p += 4;                     /* default_len */
@@ -89,6 +93,17 @@ ogm_header(AVFormatContext *s, int idx)
             st->codec->bit_rate = bytestream_get_le32(&p) * 8;
             st->codec->sample_rate = spu * 10000000 / time_unit;
             avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+            if (size >= 56 && st->codec->codec_id == CODEC_ID_AAC) {
+                p += 4;
+                size -= 4;
+            }
+            if (size > 52) {
+                av_assert0(FF_INPUT_BUFFER_PADDING_SIZE <= 52);
+                size -= 52;
+                st->codec->extradata_size = size;
+                st->codec->extradata = av_malloc(size + FF_INPUT_BUFFER_PADDING_SIZE);
+                bytestream_get_buffer(&p, st->codec->extradata, size);
+            }
         }
     } else if (*p == 3) {
         if (os->psize > 8)