The AAC demuxer now depends on id3v1.o.

[ffmpeg] / libavformat / oggparsespeex.c

diff --git a/libavformat/oggparsespeex.c b/libavformat/oggparsespeex.c
index 19789e093030cc7c66a69c4184981ce92492a4c9..140a58a9eb4b0c04eef32cd3886149d96d423b1f 100644 (file)
--- a/libavformat/oggparsespeex.c
+++ b/libavformat/oggparsespeex.c
@@ -25,7 +25,7 @@
 #include <stdlib.h>
 #include "libavutil/bswap.h"
 #include "libavutil/avstring.h"
-#include "libavcodec/bitstream.h"
+#include "libavcodec/get_bits.h"
 #include "libavcodec/bytestream.h"
 #include "avformat.h"
 #include "oggdec.h"
@@ -40,17 +40,19 @@ static int speex_header(AVFormatContext *s, int idx) {
         return 0;
 
     if (os->seq == 0) {
-    st->codec->codec_type = CODEC_TYPE_AUDIO;
-    st->codec->codec_id = CODEC_ID_SPEEX;
+        st->codec->codec_type = CODEC_TYPE_AUDIO;
+        st->codec->codec_id = CODEC_ID_SPEEX;
 
-    st->codec->sample_rate = AV_RL32(p + 36);
-    st->codec->channels = AV_RL32(p + 48);
-    st->codec->extradata_size = os->psize;
-    st->codec->extradata = av_malloc(st->codec->extradata_size);
-    memcpy(st->codec->extradata, p, st->codec->extradata_size);
+        st->codec->sample_rate = AV_RL32(p + 36);
+        st->codec->channels = AV_RL32(p + 48);
+        st->codec->frame_size = AV_RL32(p + 56);
+        st->codec->extradata_size = os->psize;
+        st->codec->extradata = av_malloc(st->codec->extradata_size
+                                         + FF_INPUT_BUFFER_PADDING_SIZE);
+        memcpy(st->codec->extradata, p, st->codec->extradata_size);
 
-    st->time_base.num = 1;
-    st->time_base.den = st->codec->sample_rate;
+        st->time_base.num = 1;
+        st->time_base.den = st->codec->sample_rate;
     } else
         vorbis_comment(s, p, os->psize);