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gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the jitterbuffer if the gap is too big, we need ...
Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): When checking the seqnum, reset the jitterbuffer if the gap is too big, we need to do this so that we can better handle a restarted source. Fix some comments. * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew), (rtp_jitter_buffer_insert): Tweak the skew resync diff. Use our working seqnum compare function in -base. Rework the jitterbuffer insert code to make it clearer and more performant by only retrieving the seqnum of the input buffer once and by adding some G_LIKELY compiler hints. Improve debugging for duplicate packets. * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp): Fix a comment, we don't do skew correction here..
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4 changed files with 81 additions and 38 deletions
20
ChangeLog
20
ChangeLog
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@ -1,3 +1,23 @@
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2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/rtpmanager/gstrtpjitterbuffer.c:
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(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
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When checking the seqnum, reset the jitterbuffer if the gap is too big,
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we need to do this so that we can better handle a restarted source.
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Fix some comments.
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* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
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(rtp_jitter_buffer_insert):
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Tweak the skew resync diff.
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Use our working seqnum compare function in -base.
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Rework the jitterbuffer insert code to make it clearer and more
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performant by only retrieving the seqnum of the input buffer once and by
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adding some G_LIKELY compiler hints.
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Improve debugging for duplicate packets.
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* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
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Fix a comment, we don't do skew correction here..
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2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
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2008-05-26 Wim Taymans <wim.taymans@collabora.co.uk>
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Patch by: Håvard Graff <havard dot graff at tandberg dot com>
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Patch by: Håvard Graff <havard dot graff at tandberg dot com>
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@ -885,13 +885,33 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
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if (priv->eos)
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if (priv->eos)
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goto have_eos;
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goto have_eos;
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/* let's check if this buffer is too late, we cannot accept packets with
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/* let's check if this buffer is too late, we can only accept packets with
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* bigger seqnum than the one we already pushed. */
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* bigger seqnum than the one we last pushed. */
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if (priv->last_popped_seqnum != -1) {
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if (priv->last_popped_seqnum != -1) {
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/* FIXME. isn't this supposed to be <= ? */
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gint gap;
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if (gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum) < 0)
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gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
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if (gap <= 0) {
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/* priv->last_popped_seqnum >= seqnum, this packet is too late or the
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* sender might have been restarted with different seqnum. */
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if (gap < -100) {
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GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
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priv->last_popped_seqnum = -1;
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priv->next_seqnum = -1;
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} else {
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goto too_late;
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goto too_late;
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}
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}
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} else {
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/* priv->last_popped_seqnum < seqnum, this is a new packet */
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if (gap > 3000) {
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GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
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gap);
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priv->last_popped_seqnum = -1;
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priv->next_seqnum = -1;
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}
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}
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}
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/* let's drop oldest packet if the queue is already full and drop-on-latency
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/* let's drop oldest packet if the queue is already full and drop-on-latency
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* is set. We can only do this when there actually is a latency. When no
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* is set. We can only do this when there actually is a latency. When no
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@ -1041,7 +1061,7 @@ again:
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if (priv->eos)
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if (priv->eos)
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goto do_eos;
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goto do_eos;
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}
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}
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/* wait for packets or flushing now */
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/* underrun, wait for packets or flushing now */
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priv->waiting = TRUE;
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priv->waiting = TRUE;
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JBUF_WAIT_CHECK (priv, flushing);
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JBUF_WAIT_CHECK (priv, flushing);
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priv->waiting = FALSE;
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priv->waiting = FALSE;
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@ -1187,7 +1207,7 @@ again:
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if (gap > 0) {
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if (gap > 0) {
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GstEvent *event;
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GstEvent *event;
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/* we had a gap and thus we lost a packet. Creat an event for this. */
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/* we had a gap and thus we lost a packet. Create an event for this. */
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GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
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GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
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priv->num_late++;
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priv->num_late++;
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discont = TRUE;
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discont = TRUE;
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@ -217,8 +217,10 @@ again:
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delta_diff = jbuf->prev_send_diff - send_diff;
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delta_diff = jbuf->prev_send_diff - send_diff;
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/* server changed rtp timestamps too quickly, reset skew detection and start
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/* server changed rtp timestamps too quickly, reset skew detection and start
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* again. */
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* again. This value is sortof arbitrary and can be a bad measurement up if
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if (delta_diff > GST_SECOND / 4) {
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* there are many packets missing because then we get a big gap that is
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* unrelated to a timestamp switch. */
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if (delta_diff > GST_SECOND) {
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GST_DEBUG ("delta changed too quickly %" GST_TIME_FORMAT " reset skew",
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GST_DEBUG ("delta changed too quickly %" GST_TIME_FORMAT " reset skew",
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GST_TIME_ARGS (delta_diff));
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GST_TIME_ARGS (delta_diff));
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rtp_jitter_buffer_reset_skew (jbuf);
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rtp_jitter_buffer_reset_skew (jbuf);
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@ -326,23 +328,6 @@ no_skew:
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return out_time;
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return out_time;
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}
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}
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static gint
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compare_seqnum (GstBuffer * a, GstBuffer * b, RTPJitterBuffer * jbuf)
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{
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guint16 seq1, seq2;
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seq1 = gst_rtp_buffer_get_seq (a);
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seq2 = gst_rtp_buffer_get_seq (b);
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/* check if diff more than half of the 16bit range */
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if (abs (seq2 - seq1) > (1 << 15)) {
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/* one of a/b has wrapped */
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return seq1 - seq2;
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} else {
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return seq2 - seq1;
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}
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}
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/**
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/**
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* rtp_jitter_buffer_insert:
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* rtp_jitter_buffer_insert:
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* @jbuf: an #RTPJitterBuffer
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* @jbuf: an #RTPJitterBuffer
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@ -362,22 +347,32 @@ rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
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GstClockTime time, guint32 clock_rate, gboolean * tail)
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GstClockTime time, guint32 clock_rate, gboolean * tail)
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{
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{
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GList *list;
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GList *list;
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gint func_ret = 1;
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guint32 rtptime;
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guint32 rtptime;
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guint16 seqnum;
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g_return_val_if_fail (jbuf != NULL, FALSE);
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g_return_val_if_fail (jbuf != NULL, FALSE);
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g_return_val_if_fail (buf != NULL, FALSE);
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g_return_val_if_fail (buf != NULL, FALSE);
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/* loop the list to skip strictly smaller seqnum buffers */
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seqnum = gst_rtp_buffer_get_seq (buf);
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list = jbuf->packets->head;
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while (list
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&& (func_ret =
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compare_seqnum (GST_BUFFER_CAST (list->data), buf, jbuf)) < 0)
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list = list->next;
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/* we hit a packet with the same seqnum, return FALSE to notify a duplicate */
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/* loop the list to skip strictly smaller seqnum buffers */
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if (func_ret == 0)
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for (list = jbuf->packets->head; list; list = g_list_next (list)) {
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return FALSE;
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guint16 qseq;
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gint gap;
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qseq = gst_rtp_buffer_get_seq (GST_BUFFER_CAST (list->data));
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/* compare the new seqnum to the one in the buffer */
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gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
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/* we hit a packet with the same seqnum, notify a duplicate */
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if (G_UNLIKELY (gap == 0))
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goto duplicate;
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/* seqnum > qseq, we can stop looking */
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if (G_LIKELY (gap < 0))
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break;
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}
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/* do skew calculation by measuring the difference between rtptime and the
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/* do skew calculation by measuring the difference between rtptime and the
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* receive time, this function will retimestamp @buf with the skew corrected
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* receive time, this function will retimestamp @buf with the skew corrected
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@ -391,11 +386,19 @@ rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
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else
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else
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g_queue_push_tail (jbuf->packets, buf);
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g_queue_push_tail (jbuf->packets, buf);
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/* tail was changed when we did not find a previous packet */
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/* tail was changed when we did not find a previous packet, we set the return
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* flag when requested. */
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if (tail)
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if (tail)
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*tail = (list == NULL);
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*tail = (list == NULL);
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return TRUE;
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return TRUE;
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/* ERRORS */
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duplicate:
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{
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GST_WARNING ("duplicate packet %d found", (gint) seqnum);
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return FALSE;
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}
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}
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}
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/**
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/**
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@ -940,7 +940,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
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GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
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GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
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seqnr, src->stats.packets_received, src->stats.octets_received);
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seqnr, src->stats.packets_received, src->stats.octets_received);
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/* calculate jitter and perform skew correction */
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/* calculate jitter for the stats */
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calculate_jitter (src, buffer, arrival);
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calculate_jitter (src, buffer, arrival);
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/* we're ready to push the RTP packet now */
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/* we're ready to push the RTP packet now */
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