Merge remote-tracking branch 'qatar/master'

* qatar/master: (31 commits)
  cdxl demux: do not create packets with uninitialized data at EOF.
  Replace computations of remaining bits with calls to get_bits_left().
  amrnb/amrwb: Remove get_bits usage.
  cosmetics: reindent
  avformat: do not require a pixel/sample format if there is no decoder
  avformat: do not fill-in audio packet duration in compute_pkt_fields()
  lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
  dca_parser: parse the sample rate and frame durations
  libspeexdec: do not set AVCodecContext.frame_size
  libopencore-amr: do not set AVCodecContext.frame_size
  alsdec: do not set AVCodecContext.frame_size
  siff: do not set AVCodecContext.frame_size
  amr demuxer: do not set AVCodecContext.frame_size.
  aiffdec: do not set AVCodecContext.frame_size
  mov: do not set AVCodecContext.frame_size
  ape: do not set AVCodecContext.frame_size.
  rdt: remove workaround for infinite loop with aac
  avformat: do not require frame_size in avformat_find_stream_info() for CELT
  avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
  avformat: do not require frame_size in avformat_find_stream_info() for AAC
  ...

Conflicts:
	doc/APIchanges
	libavcodec/Makefile
	libavcodec/avcodec.h
	libavcodec/h264.c
	libavcodec/h264_ps.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/x86/dsputil_mmx.c
	libavformat/utils.c

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

1 files changed, 0 insertions, 13 deletions

diff --git a/libavformat/mov.c b/libavformat/mov.c
index 4ad2051efc..3ca1f97658 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -1464,20 +1464,16 @@ int ff_mov_read_stsd_entries(MOVContext *c, AVIOContext *pb, int entries)
         // force sample rate for qcelp when not stored in mov
         if (st->codec->codec_tag != MKTAG('Q','c','l','p'))
             st->codec->sample_rate = 8000;
-        st->codec->frame_size= 160;
         st->codec->channels= 1; /* really needed */
         break;
     case CODEC_ID_AMR_NB:
         st->codec->channels= 1; /* really needed */
         /* force sample rate for amr, stsd in 3gp does not store sample rate */
         st->codec->sample_rate = 8000;
-        /* force frame_size, too, samples_per_frame isn't always set properly */
-        st->codec->frame_size  = 160;
         break;
     case CODEC_ID_AMR_WB:
         st->codec->channels    = 1;
         st->codec->sample_rate = 16000;
-        st->codec->frame_size  = 320;
         break;
     case CODEC_ID_MP2:
     case CODEC_ID_MP3:
@@ -1487,12 +1483,10 @@ int ff_mov_read_stsd_entries(MOVContext *c, AVIOContext *pb, int entries)
     case CODEC_ID_GSM:
     case CODEC_ID_ADPCM_MS:
     case CODEC_ID_ADPCM_IMA_WAV:
-        st->codec->frame_size = sc->samples_per_frame;
         st->codec->block_align = sc->bytes_per_frame;
         break;
     case CODEC_ID_ALAC:
         if (st->codec->extradata_size == 36) {
-            st->codec->frame_size = AV_RB32(st->codec->extradata+12);
             st->codec->channels   = AV_RB8 (st->codec->extradata+21);
             st->codec->sample_rate = AV_RB32(st->codec->extradata+32);
         }
@@ -2058,13 +2052,6 @@ static int mov_read_trak(MOVContext *c, AVIOContext *pb, MOVAtom atom)
 
     avpriv_set_pts_info(st, 64, 1, sc->time_scale);
 
-    if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
-        !st->codec->frame_size && sc->stts_count == 1) {
-        st->codec->frame_size = av_rescale(sc->stts_data[0].duration,
-                                           st->codec->sample_rate, sc->time_scale);
-        av_dlog(c->fc, "frame size %d\n", st->codec->frame_size);
-    }
-
     mov_build_index(c, st);
 
     if (sc->dref_id-1 < sc->drefs_count && sc->drefs[sc->dref_id-1].path) {