Merge remote-tracking branch 'qatar/master'

* qatar/master:
  Add LATM demuxer
  avplay: flush audio decoder with empty packets at EOF if the decoder has CODEC_CAP_DELAY set.
  8svx/iff: fix decoding of compressed stereo 8svx files.
  8svx: log an error message if output buffer is too small
  8svx: check packet size before reading the initial sample value.
  8svx: output 8-bit samples instead of 16-bit.
  8svx: split delta decoding into a separate function.
  mp4: Don't read an empty Decoder Config Descriptor
  fate.sh: Ignore errors from rm command during cleanup.
  fate.sh: Run git-pull in quiet mode to avoid console spam.
  Apple ProRes decoder
  rtmp: Make the input FLV parser handle data cut at any point
  rv34: Check for invalid slices offsets
  eval: test isnan(sqrt(-1)) instead of just sqrt(-1)

Conflicts:
	Changelog
	libavcodec/8svx.c
	libavcodec/proresdec.c
	libavcodec/version.h
	libavformat/iff.c
	libavformat/version.h
	tests/ref/fate/eval

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

1 files changed, 23 insertions, 6 deletions

diff --git a/ffplay.c b/ffplay.c
index 29e1120d88..1a2eb1b08a 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -1994,10 +1994,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
     AVCodecContext *dec= is->audio_st->codec;
     int n, len1, data_size;
     double pts;
+    int new_packet = 0;
+    int flush_complete = 0;
 
     for(;;) {
         /* NOTE: the audio packet can contain several frames */
-        while (pkt_temp->size > 0) {
+        while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) {
+            if (flush_complete)
+                break;
+            new_packet = 0;
             data_size = sizeof(is->audio_buf1);
             len1 = avcodec_decode_audio3(dec,
                                         (int16_t *)is->audio_buf1, &data_size,
@@ -2010,8 +2015,13 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
 
             pkt_temp->data += len1;
             pkt_temp->size -= len1;
-            if (data_size <= 0)
+
+            if (data_size <= 0) {
+                /* stop sending empty packets if the decoder is finished */
+                if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
+                    flush_complete = 1;
                 continue;
+            }
 
             if (dec->sample_fmt != is->audio_src_fmt) {
                 if (is->reformat_ctx)
@@ -2072,12 +2082,11 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
         }
 
         /* read next packet */
-        if (packet_queue_get(&is->audioq, pkt, 1) < 0)
+        if ((new_packet = packet_queue_get(&is->audioq, pkt, 1)) < 0)
             return -1;
-        if(pkt->data == flush_pkt.data){
+
+        if (pkt->data == flush_pkt.data)
             avcodec_flush_buffers(dec);
-            continue;
-        }
 
         pkt_temp->data = pkt->data;
         pkt_temp->size = pkt->size;
@@ -2508,6 +2517,14 @@ static int read_thread(void *arg)
                 pkt->stream_index= is->video_stream;
                 packet_queue_put(&is->videoq, pkt);
             }
+            if (is->audio_stream >= 0 &&
+                is->audio_st->codec->codec->capabilities & CODEC_CAP_DELAY) {
+                av_init_packet(pkt);
+                pkt->data = NULL;
+                pkt->size = 0;
+                pkt->stream_index = is->audio_stream;
+                packet_queue_put(&is->audioq, pkt);
+            }
             SDL_Delay(10);
             if(is->audioq.size + is->videoq.size + is->subtitleq.size ==0){
                 if(loop!=1 && (!loop || --loop)){