Merge remote-tracking branch 'qatar/master'

* qatar/master: (36 commits)
  adpcmenc: Use correct frame_size for Yamaha ADPCM.
  avcodec: add ff_samples_to_time_base() convenience function to internal.h
  adx parser: set duration
  mlp parser: set duration instead of frame_size
  gsm parser: set duration
  mpegaudio parser: set duration instead of frame_size
  (e)ac3 parser: set duration instead of frame_size
  flac parser: set duration instead of frame_size
  avcodec: add duration field to AVCodecParserContext
  avutil: add av_rescale_q_rnd() to allow different rounding
  pnmdec: remove useless .pix_fmts
  libmp3lame: support float and s32 sample formats
  libmp3lame: renaming, rearrangement, alignment, and comments
  libmp3lame: use the LAME default bit rate
  libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
  libmp3lame: cosmetics: remove some pointless comments
  libmp3lame: convert some debugging code to av_dlog()
  libmp3lame: remove outdated comment.
  libmp3lame: do not set coded_frame->key_frame.
  libmp3lame: improve error handling in MP3lame_encode_init()
  ...

Conflicts:
	doc/APIchanges
	libavcodec/libmp3lame.c
	libavcodec/pcxenc.c
	libavcodec/pnmdec.c
	libavcodec/pnmenc.c
	libavcodec/sgienc.c
	libavcodec/utils.c
	libavformat/hls.c
	libavutil/avutil.h
	libswscale/x86/swscale_mmx.c

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

1 files changed, 16 insertions, 3 deletions

diff --git a/libavformat/utils.c b/libavformat/utils.c
index 39cf766bdf..155ba67757 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -942,11 +942,10 @@ static void compute_pkt_fields(AVFormatContext *s, AVStream *st,
         compute_frame_duration(&num, &den, st, pc, pkt);
         if (den && num) {
             pkt->duration = av_rescale_rnd(1, num * (int64_t)st->time_base.den, den * (int64_t)st->time_base.num, AV_ROUND_DOWN);
-
-            if(pkt->duration != 0 && s->packet_buffer)
-                update_initial_durations(s, st, pkt);
         }
     }
+    if(pkt->duration != 0 && s->packet_buffer)
+        update_initial_durations(s, st, pkt);
 
     /* correct timestamps with byte offset if demuxers only have timestamps
        on packet boundaries */
@@ -1099,6 +1098,20 @@ static int read_frame_internal(AVFormatContext *s, AVPacket *pkt)
                 if (pkt->size) {
                 got_packet:
                     pkt->duration = 0;
+                    if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+                        if (st->codec->sample_rate > 0) {
+                            pkt->duration = av_rescale_q_rnd(st->parser->duration,
+                                                             (AVRational){ 1, st->codec->sample_rate },
+                                                             st->time_base,
+                                                             AV_ROUND_DOWN);
+                        }
+                    } else if (st->codec->time_base.num != 0 &&
+                               st->codec->time_base.den != 0) {
+                        pkt->duration = av_rescale_q_rnd(st->parser->duration,
+                                                         st->codec->time_base,
+                                                         st->time_base,
+                                                         AV_ROUND_DOWN);
+                    }
                     pkt->stream_index = st->index;
                     pkt->pts = st->parser->pts;
                     pkt->dts = st->parser->dts;