Merge remote-tracking branch 'qatar/master'

* qatar/master: (27 commits)
  avconv: free packet in write_frame() when discarding due to frame number limit
  FATE: use +/- flag option syntax for vp8 emu-edge tests
  lavf: make av_interleave_packet_per_dts() private.
  lavf: deprecate av_read_packet().
  oggdec: output correct timestamps for Vorbis
  avconv: pass input stream timestamps to audio encoders
  lavc: shrink encoded audio packet size after encoding.
  xa: set correct bit rate
  xa: do not set bit_rate, block_align, or bits_per_coded_sample
  xa: fix end-of-file handling
  xa: fix timestamp calculation
  bink: fix typo in FFALIGN() argument
  bink: align plane width to 8 when calculating bundle sizes
  doc: pass -Idoc texi2html and texi2pod
  doc: texi2pod: add -I flag
  movenc: Add a min_frag_duration option
  rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
  libavformat: Set the default for the max_delay option to -1
  Generate manpages for AV{Format,Codec}Context AVOptions.
  doc/avconv: remove entries for AVOptions.
  ...

Conflicts:
	doc/Makefile
	doc/ffmpeg.texi
	doc/muxers.texi
	ffmpeg.c
	libavcodec/Makefile
	libavcodec/options.c
	libavcodec/vp8.c
	libavformat/options.c
	tests/fate/demux.mak
	tests/ref/fate/truemotion1-15
	tests/ref/fate/truemotion1-24

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 10 insertions, 10 deletions

diff --git a/libavformat/xa.c b/libavformat/xa.c
index 1e51eec934..c54d7f8b9c 100644
--- a/libavformat/xa.c
+++ b/libavformat/xa.c
@@ -38,7 +38,6 @@
 typedef struct MaxisXADemuxContext {
     uint32_t out_size;
     uint32_t sent_bytes;
-    uint32_t audio_frame_counter;
 } MaxisXADemuxContext;
 
 static int xa_probe(AVProbeData *p)
@@ -81,12 +80,15 @@ static int xa_read_header(AVFormatContext *s)
     avio_skip(pb, 2);       /* Skip the tag */
     st->codec->channels     = avio_rl16(pb);
     st->codec->sample_rate  = avio_rl32(pb);
-    /* Value in file is average byte rate*/
-    st->codec->bit_rate     = avio_rl32(pb) * 8;
-    st->codec->block_align  = avio_rl16(pb);
-    st->codec->bits_per_coded_sample = avio_rl16(pb);
+    avio_skip(pb, 4);       /* Skip average byte rate */
+    avio_skip(pb, 2);       /* Skip block align */
+    avio_skip(pb, 2);       /* Skip bits-per-sample */
+
+    st->codec->bit_rate = av_clip(15LL * st->codec->channels * 8 *
+                                  st->codec->sample_rate / 28, 0, INT_MAX);
 
     avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+    st->start_time = 0;
 
     return 0;
 }
@@ -100,8 +102,8 @@ static int xa_read_packet(AVFormatContext *s,
     unsigned int packet_size;
     int ret;
 
-    if(xa->sent_bytes > xa->out_size)
-        return AVERROR(EIO);
+    if (xa->sent_bytes >= xa->out_size)
+        return AVERROR_EOF;
     /* 1 byte header and 14 bytes worth of samples * number channels per block */
     packet_size = 15*st->codec->channels;
 
@@ -111,9 +113,7 @@ static int xa_read_packet(AVFormatContext *s,
 
     pkt->stream_index = st->index;
     xa->sent_bytes += packet_size;
-    pkt->pts = xa->audio_frame_counter;
-    /* 14 bytes Samples per channel with 2 samples per byte */
-    xa->audio_frame_counter += 28 * st->codec->channels;
+    pkt->duration = 28;
 
     return ret;
 }