move zero size hack from ogg.c to utils.c

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2959 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

libavformat/ogg.c

@@ -73,10 +73,6 @@ static int ogg_write_packet(AVFormatContext *avfcontext,
 
     pts= av_rescale(pts, avctx->sample_rate, AV_TIME_BASE);
 
-    if(!size){
-//        av_log(avfcontext, AV_LOG_DEBUG, "zero packet\n");
-        return 0;
-    }
 //    av_log(avfcontext, AV_LOG_DEBUG, "M%d\n", size);
 
     /* flush header packets so audio starts on a new page */

libavformat/utils.c

@@ -1698,8 +1698,14 @@ int av_write_frame(AVFormatContext *s, int stream_index, const uint8_t *buf,
 
     st = s->streams[stream_index];
     pts_mask = (1LL << s->pts_wrap_bits) - 1;
-    ret = s->oformat->write_packet(s, stream_index, buf, size, 
-                                   st->pts.val & pts_mask);
+
+    /* HACK/FIXME we skip all zero size audio packets so a encoder can pass pts by outputing zero size packets */
+    if(st->codec.codec_type==CODEC_TYPE_AUDIO && size==0)
+        ret = 0;
+    else
+        ret = s->oformat->write_packet(s, stream_index, buf, size, 
+                                       st->pts.val & pts_mask);
+    
     if (ret < 0)
         return ret;
 
@@ -1708,9 +1714,8 @@ int av_write_frame(AVFormatContext *s, int stream_index, const uint8_t *buf,
     case CODEC_TYPE_AUDIO:
         frame_size = get_audio_frame_size(&st->codec, size);
 
-        /* note, we skip the initial 0-size packets as they are most likely equal to the encoder delay,
+        /* HACK/FIXME, we skip the initial 0-size packets as they are most likely equal to the encoder delay,
            but it would be better if we had the real timestamps from the encoder */
-//        av_log(s, AV_LOG_DEBUG, "%d %lld %lld\n", size, st->pts.num, st->pts.val);
         if (frame_size >= 0 && (size || st->pts.num!=st->pts.den>>1 || st->pts.val)) {
             av_frac_add(&st->pts, 
                         (int64_t)s->pts_den * frame_size);