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, 23 insertions, 5 deletions

diff --git a/libavformat/riff.c b/libavformat/riff.c
index 70134276f0..774a02f82d 100644
--- a/libavformat/riff.c
+++ b/libavformat/riff.c
@@ -414,7 +414,7 @@ void ff_end_tag(AVIOContext *pb, int64_t start)
 /* returns the size or -1 on error */
 int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
 {
-    int bps, blkalign, bytespersec;
+    int bps, blkalign, bytespersec, frame_size;
     int hdrsize = 18;
     int waveformatextensible;
     uint8_t temp[256];
@@ -423,6 +423,14 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
 
     if(!enc->codec_tag || enc->codec_tag > 0xffff)
         return -1;
+
+    /* We use the known constant frame size for the codec if known, otherwise
+       fallback to using AVCodecContext.frame_size, which is not as reliable
+       for indicating packet duration */
+    frame_size = av_get_audio_frame_duration(enc, 0);
+    if (!frame_size)
+        frame_size = enc->frame_size;
+
     waveformatextensible =   (enc->channels > 2 && enc->channel_layout)
                           || enc->sample_rate > 48000
                           || av_get_bits_per_sample(enc->codec_id) > 16;
@@ -449,7 +457,9 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
     }
 
     if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
-        blkalign = enc->frame_size; //this is wrong, but it seems many demuxers do not work if this is set correctly
+        /* this is wrong, but it seems many demuxers do not work if this is set
+           correctly */
+        blkalign = frame_size;
         //blkalign = 144 * enc->bit_rate/enc->sample_rate;
     } else if (enc->codec_id == CODEC_ID_AC3) {
             blkalign = 3840; //maximum bytes per frame
@@ -489,7 +499,7 @@ int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
         bytestream_put_le32(&riff_extradata, 0);                          /* dwPTSHigh */
     } else if (enc->codec_id == CODEC_ID_GSM_MS || enc->codec_id == CODEC_ID_ADPCM_IMA_WAV) {
         hdrsize += 2;
-        bytestream_put_le16(&riff_extradata, enc->frame_size); /* wSamplesPerBlock */
+        bytestream_put_le16(&riff_extradata, frame_size); /* wSamplesPerBlock */
     } else if(enc->extradata_size){
         riff_extradata_start= enc->extradata;
         riff_extradata= enc->extradata + enc->extradata_size;
@@ -657,10 +667,18 @@ int ff_get_bmp_header(AVIOContext *pb, AVStream *st)
 void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale)
 {
     int gcd;
+    int audio_frame_size;
+
+    /* We use the known constant frame size for the codec if known, otherwise
+       fallback to using AVCodecContext.frame_size, which is not as reliable
+       for indicating packet duration */
+    audio_frame_size = av_get_audio_frame_duration(stream, 0);
+    if (!audio_frame_size)
+        audio_frame_size = stream->frame_size;
 
     *au_ssize= stream->block_align;
-    if(stream->frame_size && stream->sample_rate){
-        *au_scale=stream->frame_size;
+    if (audio_frame_size && stream->sample_rate) {
+        *au_scale = audio_frame_size;
         *au_rate= stream->sample_rate;
     }else if(stream->codec_type == AVMEDIA_TYPE_VIDEO ||
              stream->codec_type == AVMEDIA_TYPE_DATA ||