]> rtime.felk.cvut.cz Git - frescor/ffmpeg.git/commitdiff
Fix compilation of beosaudio.cpp, not tested if it actually works though.
authorreimar <reimar@9553f0bf-9b14-0410-a0b8-cfaf0461ba5b>
Wed, 30 Sep 2009 13:01:48 +0000 (13:01 +0000)
committerreimar <reimar@9553f0bf-9b14-0410-a0b8-cfaf0461ba5b>
Wed, 30 Sep 2009 13:01:48 +0000 (13:01 +0000)
git-svn-id: file:///var/local/repositories/ffmpeg/trunk@20098 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

libavdevice/beosaudio.cpp

index c98a2398c54fb8dacf661e3c1fc10d26a182e781..c22cb9bcb49e5d82d0fee2d658b6c46732dd46ad 100644 (file)
@@ -297,11 +297,12 @@ static int audio_write_header(AVFormatContext *s1)
     return 0;
 }
 
-static int audio_write_packet(AVFormatContext *s1, int stream_index,
-                              const uint8_t *buf, int size, int64_t force_pts)
+static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     AudioData *s = (AudioData *)s1->priv_data;
     int len, ret;
+    const uint8_t *buf = pkt->data;
+    int size = pkt->size;
 #ifdef LATENCY_CHECK
 bigtime_t lat1, lat2;
 lat1 = s->player->Latency();
@@ -372,7 +373,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
     st->codec->sample_rate = s->sample_rate;
     st->codec->channels = s->channels;
     return 0;
-    av_set_pts_info(s1, 48, 1, 1000000);  /* 48 bits pts in us */
+    av_set_pts_info(st, 48, 1, 1000000);  /* 48 bits pts in us */
 }
 
 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
@@ -430,6 +431,7 @@ static AVInputFormat audio_beos_demuxer = {
     audio_read_packet,
     audio_read_close,
     NULL,
+    NULL,
     AVFMT_NOFILE,
 };