diff options
| -rw-r--r-- | ChangeLog | 8 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpjitterbuffer.c | 36 |
2 files changed, 31 insertions, 13 deletions
@@ -1,3 +1,11 @@ +2007-08-31 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtpmanager/gstrtpjitterbuffer.c: + (gst_rtp_jitter_buffer_flush_stop), + (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop): + Use extended timestamp to release buffers from the jitterbuffer so that + we can handle the rtp wraparound correctly. + 2007-08-29 Wim Taymans <wim.taymans@gmail.com> * gst/rtpmanager/gstrtpjitterbuffer.c: diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index f4d59bb6..bd1c07ea 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -149,6 +149,7 @@ struct _GstRtpJitterBufferPrivate /* clock rate and rtp timestamp offset */ gint32 clock_rate; gint64 clock_base; + guint64 exttimestamp; /* when we are shutting down */ GstFlowReturn srcresult; @@ -531,6 +532,7 @@ gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer) priv->next_seqnum = -1; priv->clock_rate = -1; priv->eos = FALSE; + priv->exttimestamp = -1; JBUF_UNLOCK (priv); } @@ -587,6 +589,7 @@ gst_rtp_jitter_buffer_change_state (GstElement * element, priv->peer_latency = 0; /* block until we go to PLAYING */ priv->blocked = TRUE; + priv->exttimestamp = -1; JBUF_UNLOCK (priv); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: @@ -925,6 +928,7 @@ gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer) guint32 rtp_time; GstClockTime timestamp; gint64 running_time; + guint64 exttimestamp; priv = jitterbuffer->priv; @@ -951,8 +955,15 @@ again: seqnum = gst_rtp_buffer_get_seq (outbuf); - GST_DEBUG_OBJECT (jitterbuffer, "Popped buffer #%d, now %d left", - seqnum, rtp_jitter_buffer_num_packets (priv->jbuf)); + /* get the max deadline to wait for the missing packets, this is the time + * of the currently popped packet */ + rtp_time = gst_rtp_buffer_get_timestamp (outbuf); + exttimestamp = gst_rtp_buffer_ext_timestamp (&priv->exttimestamp, rtp_time); + + GST_DEBUG_OBJECT (jitterbuffer, + "Popped buffer #%d, rtptime %u, exttime %" G_GUINT64_FORMAT + ",now %d left", seqnum, rtp_time, exttimestamp, + rtp_jitter_buffer_num_packets (priv->jbuf)); /* If we don't know what the next seqnum should be (== -1) we have to wait * because it might be possible that we are not receiving this buffer in-order, @@ -980,26 +991,25 @@ again: GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum); } - /* get the max deadline to wait for the missing packets, this is the time - * of the currently popped packet */ - rtp_time = gst_rtp_buffer_get_timestamp (outbuf); - - GST_DEBUG_OBJECT (jitterbuffer, "rtp_time %u, base %" G_GINT64_FORMAT, - rtp_time, priv->clock_base); + GST_DEBUG_OBJECT (jitterbuffer, + "exttimestamp %" G_GUINT64_FORMAT ", base %" G_GINT64_FORMAT, + exttimestamp, priv->clock_base); /* if no clock_base was given, take first ts as base */ if (priv->clock_base == -1) - priv->clock_base = rtp_time; + priv->clock_base = exttimestamp; /* take rtp timestamp offset into account, this can wrap around */ - rtp_time -= priv->clock_base; + exttimestamp -= priv->clock_base; /* bring timestamp to gst time */ - timestamp = gst_util_uint64_scale (GST_SECOND, rtp_time, priv->clock_rate); + timestamp = + gst_util_uint64_scale_int (exttimestamp, GST_SECOND, priv->clock_rate); GST_DEBUG_OBJECT (jitterbuffer, - "rtptime %u, clock-rate %u, timestamp %" GST_TIME_FORMAT, rtp_time, - priv->clock_rate, GST_TIME_ARGS (timestamp)); + "exttimestamp %" G_GUINT64_FORMAT ", clock-rate %u, timestamp %" + GST_TIME_FORMAT, exttimestamp, priv->clock_rate, + GST_TIME_ARGS (timestamp)); /* bring to running time */ running_time = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, |