Merge remote-tracking branch 'qatar/master'

* qatar/master:
  tiertexseq: set correct block_align for audio
  tiertexseq: set audio stream start time to 0
  voc/avs: Do not change the sample rate mid-stream.
  segafilm: use the sample rate as the time base for audio streams
  ea: fix audio pts
  psx-str: fix audio pts
  vqf: set packet duration
  tta demuxer: set packet duration
  mpegaudio_parser: do not ignore information from the first parsed frame
  mpegaudio_parser: be less picky about the start position
  thp: set audio packet durations
  avcodec: add a Vorbis parser to get packet duration
  vorbisdec: read the previous window flag for long windows
  lavc: free the output packet when encoding failed or produced no output.
  lavc: preserve avpkt->destruct in ff_alloc_packet().
  lavc: clarify the meaning of AVCodecContext.frame_number.
  mpegts: Pad the packet buffer in handle_packet().
  mpegts: Do not call read_sl_header() when no bytes remain in the buffer.

Conflicts:
	libavcodec/mpegaudio_parser.c
	libavcodec/version.h
	libavformat/mpegts.c
	tests/ref/fate/pva-demux

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

1 files changed, 12 insertions, 4 deletions

diff --git a/libavformat/vocdec.c b/libavformat/vocdec.c
index ea06b53ecf..6df4d8d01f 100644
--- a/libavformat/vocdec.c
+++ b/libavformat/vocdec.c
@@ -86,9 +86,13 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
 
         switch (type) {
         case VOC_TYPE_VOICE_DATA:
-            dec->sample_rate = 1000000 / (256 - avio_r8(pb));
-            if (sample_rate)
-                dec->sample_rate = sample_rate;
+            if (!dec->sample_rate) {
+                dec->sample_rate = 1000000 / (256 - avio_r8(pb));
+                if (sample_rate)
+                    dec->sample_rate = sample_rate;
+                avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
+            } else
+                avio_skip(pb, 1);
             dec->channels = channels;
             tmp_codec = avio_r8(pb);
             dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
@@ -110,7 +114,11 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
             break;
 
         case VOC_TYPE_NEW_VOICE_DATA:
-            dec->sample_rate = avio_rl32(pb);
+            if (!dec->sample_rate) {
+                dec->sample_rate = avio_rl32(pb);
+                avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
+            } else
+                avio_skip(pb, 4);
             dec->bits_per_coded_sample = avio_r8(pb);
             dec->channels = avio_r8(pb);
             tmp_codec = avio_rl16(pb);