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, 13 insertions, 12 deletions

diff --git a/libavformat/electronicarts.c b/libavformat/electronicarts.c
index 9719c6711a..968682af7e 100644
--- a/libavformat/electronicarts.c
+++ b/libavformat/electronicarts.c
@@ -70,7 +70,6 @@ typedef struct EaDemuxContext {
 
     enum CodecID audio_codec;
     int audio_stream_index;
-    int audio_frame_counter;
 
     int bytes;
     int sample_rate;
@@ -472,7 +471,7 @@ static int ea_read_header(AVFormatContext *s)
             st->codec->bits_per_coded_sample / 4;
         st->codec->block_align = st->codec->channels*st->codec->bits_per_coded_sample;
         ea->audio_stream_index = st->index;
-        ea->audio_frame_counter = 0;
+        st->start_time = 0;
     }
 
     return 1;
@@ -522,24 +521,26 @@ static int ea_read_packet(AVFormatContext *s,
             if (ret < 0)
                 return ret;
             pkt->stream_index = ea->audio_stream_index;
-            pkt->pts = 90000;
-            pkt->pts *= ea->audio_frame_counter;
-            pkt->pts /= ea->sample_rate;
 
             switch (ea->audio_codec) {
             case CODEC_ID_ADPCM_EA:
-                /* 2 samples/byte, 1 or 2 samples per frame depending
-                 * on stereo; chunk also has 12-byte header */
-                ea->audio_frame_counter += ((chunk_size - 12) * 2) /
-                    ea->num_channels;
+            case CODEC_ID_ADPCM_EA_R1:
+            case CODEC_ID_ADPCM_EA_R2:
+            case CODEC_ID_ADPCM_IMA_EA_EACS:
+                pkt->duration = AV_RL32(pkt->data);
+                break;
+            case CODEC_ID_ADPCM_EA_R3:
+                pkt->duration = AV_RB32(pkt->data);
+                break;
+            case CODEC_ID_ADPCM_IMA_EA_SEAD:
+                pkt->duration = ret * 2 / ea->num_channels;
                 break;
             case CODEC_ID_PCM_S16LE_PLANAR:
             case CODEC_ID_MP3:
-                ea->audio_frame_counter += num_samples;
+                pkt->duration = num_samples;
                 break;
             default:
-                ea->audio_frame_counter += chunk_size /
-                    (ea->bytes * ea->num_channels);
+                pkt->duration = chunk_size / (ea->bytes * ea->num_channels);
             }
 
             packet_read = 1;