Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavformat / tiertexseq.c

diff --git a/libavformat/tiertexseq.c b/libavformat/tiertexseq.c
index 194c83abc109946ae5f23ada43306374a405d443..a0c06deab6d8bbd323b2ec212ee42e4a2d174002 100644 (file)
--- a/libavformat/tiertexseq.c
+++ b/libavformat/tiertexseq.c
@@ -2,20 +2,20 @@
  * Tiertex Limited SEQ File Demuxer
  * Copyright (c) 2006 Gregory Montoir (cyx@users.sourceforge.net)
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -224,6 +224,7 @@ static int seq_read_header(AVFormatContext *s)
     if (!st)
         return AVERROR(ENOMEM);
 
+    st->start_time = 0;
     avpriv_set_pts_info(st, 32, 1, SEQ_SAMPLE_RATE);
     seq->audio_stream_index = st->index;
     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
@@ -233,7 +234,7 @@ static int seq_read_header(AVFormatContext *s)
     st->codec->sample_rate = SEQ_SAMPLE_RATE;
     st->codec->bits_per_coded_sample = 16;
     st->codec->bit_rate = st->codec->sample_rate * st->codec->bits_per_coded_sample * st->codec->channels;
-    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
+    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample / 8;
 
     return 0;
 }