Merge remote-tracking branch 'qatar/master'

* qatar/master: (51 commits)
  cin audio: use sign_extend() instead of casting to int16_t
  cin audio: restructure decoding loop to avoid a separate counter variable
  cin audio: use local variable for delta value
  cin audio: remove unneeded cast from void*
  cin audio: validate the channel count
  cin audio: remove unneeded AVCodecContext pointer from CinAudioContext
  dsicin: fix several audio-related fields in the CIN demuxer
  flacdec: use av_get_bytes_per_sample() to get sample size
  dca: handle errors from dca_decode_block()
  dca: return error if the frame header is invalid
  dca: return proper error codes instead of -1
  utvideo: handle empty Huffman trees
  binkaudio: change short to int16_t
  binkaudio: only decode one block at a time.
  binkaudio: store interleaved overlap samples in BinkAudioContext.
  binkaudio: pre-calculate quantization factors
  binkaudio: add some buffer overread checks.
  atrac3: support float or int16 output using request_sample_fmt
  atrac3: add CODEC_CAP_SUBFRAMES capability
  atrac3: return appropriate error codes instead of -1
  ...

Conflicts:
	libavcodec/atrac1.c
	libavcodec/dca.c
	libavformat/mov.c

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

3 files changed, 7 insertions, 4 deletions

diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 60ff6dcf03..90cf3ad44d 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -676,6 +676,7 @@ typedef struct AVStream {
         int duration_count;
         double duration_error[2][2][MAX_STD_TIMEBASES];
         int64_t codec_info_duration;
+        int nb_decoded_frames;
     } *info;
 
     /**
diff --git a/libavformat/dsicin.c b/libavformat/dsicin.c
index 09e80e944f..ab098dc017 100644
--- a/libavformat/dsicin.c
+++ b/libavformat/dsicin.c
@@ -131,9 +131,8 @@ static int cin_read_header(AVFormatContext *s, AVFormatParameters *ap)
     st->codec->codec_tag = 0;  /* no tag */
     st->codec->channels = 1;
     st->codec->sample_rate = 22050;
-    st->codec->bits_per_coded_sample = 16;
+    st->codec->bits_per_coded_sample = 8;
     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;
 
     return 0;
 }
@@ -211,7 +210,8 @@ static int cin_read_packet(AVFormatContext *s, AVPacket *pkt)
 
     pkt->stream_index = cin->audio_stream_index;
     pkt->pts = cin->audio_stream_pts;
-    cin->audio_stream_pts += cin->audio_buffer_size * 2 / cin->file_header.audio_frame_size;
+    pkt->duration = cin->audio_buffer_size - (pkt->pts == 0);
+    cin->audio_stream_pts += pkt->duration;
     cin->audio_buffer_size = 0;
     return 0;
 }
diff --git a/libavformat/utils.c b/libavformat/utils.c
index 8a78308447..2cf42d3c43 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -2200,7 +2200,7 @@ static int has_codec_parameters(AVCodecContext *avctx)
 static int has_decode_delay_been_guessed(AVStream *st)
 {
     return st->codec->codec_id != CODEC_ID_H264 ||
-        st->codec_info_nb_frames >= 6 + st->codec->has_b_frames;
+        st->info->nb_decoded_frames >= 6;
 }
 
 static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **options)
@@ -2226,6 +2226,8 @@ static int try_decode_frame(AVStream *st, AVPacket *avpkt, AVDictionary **option
             avcodec_get_frame_defaults(&picture);
             ret = avcodec_decode_video2(st->codec, &picture,
                                         &got_picture, avpkt);
+            if (got_picture)
+                st->info->nb_decoded_frames++;
             break;
         case AVMEDIA_TYPE_AUDIO:
             data_size = FFMAX(avpkt->size, AVCODEC_MAX_AUDIO_FRAME_SIZE);