gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release buffers from the jitterbuffer so that we can h...

Original commit message from CVS:
* 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.
This commit is contained in:
Wim Taymans 2007-08-31 15:26:14 +00:00
parent 696bf74212
commit 33fd595e04
2 changed files with 31 additions and 13 deletions

View file

@ -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:

View file

@ -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,