diff audio.c @ 468:60f897e8dd2d libavformat

pass AVPacket into av_write_frame() fixes the random dts/pts during encoding asf preroll fix no more initial zero frames for b frame encoding mpeg-es dts during demuxing fixed .ffm timestamp scale fixed, ffm is still broken though

--- a/audio.c	Tue May 25 23:06:00 2004 +0000
+++ b/audio.c	Sat May 29 02:06:32 2004 +0000
@@ -164,11 +164,12 @@
     }
 }
 
-static int audio_write_packet(AVFormatContext *s1, int stream_index,
-                              const uint8_t *buf, int size, int64_t pts)
+static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     AudioData *s = s1->priv_data;
     int len, ret;
+    int size= pkt->size;
+    uint8_t *buf= pkt->data;
 
     while (size > 0) {
         len = AUDIO_BLOCK_SIZE - s->buffer_ptr;