dont set the sampling rate just because 1 mp3 packet header says so (fixes playback speed on some old mencoder generated avis which where then dumped to mp3)

Originally committed as revision 6837 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 2 insertions, 2 deletions

diff --git a/libavformat/mp3.c b/libavformat/mp3.c
index 582e7e8547..0e39d25ba7 100644
--- a/libavformat/mp3.c
+++ b/libavformat/mp3.c
@@ -247,7 +247,7 @@ static void id3_create_tag(AVFormatContext *s, uint8_t *buf)
 static int mp3_read_probe(AVProbeData *p)
 {
     int max_frames, first_frames;
-    int fsize, frames;
+    int fsize, frames, sample_rate;
     uint32_t header;
     uint8_t *buf, *buf2, *end;
     AVCodecContext avctx;
@@ -267,7 +267,7 @@ static int mp3_read_probe(AVProbeData *p)
 
         for(frames = 0; buf2 < end; frames++) {
             header = (buf2[0] << 24) | (buf2[1] << 16) | (buf2[2] << 8) | buf2[3];
-            fsize = mpa_decode_header(&avctx, header);
+            fsize = mpa_decode_header(&avctx, header, &sample_rate);
             if(fsize < 0)
                 break;
             buf2 += fsize;