gst/rtpmanager/: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some...
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c:
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
(rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
(rtp_jitter_buffer_new), (compare_seqnum),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
(rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove complicated async queue and replace with more simple jitterbuffer
code while also fixing some bugs.
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
(create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
(create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
* gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
(on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
(gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
Use new jitterbuffer code.
Expose some new signals in preparation for handling EOS.
2007-08-10 17:16:53 +00:00
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/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <string.h>
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2007-08-13 06:16:40 +00:00
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#include <stdlib.h>
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gst/rtpmanager/: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some...
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c:
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
(rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
(rtp_jitter_buffer_new), (compare_seqnum),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
(rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove complicated async queue and replace with more simple jitterbuffer
code while also fixing some bugs.
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
(create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
(create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
* gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
(on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
(gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
Use new jitterbuffer code.
Expose some new signals in preparation for handling EOS.
2007-08-10 17:16:53 +00:00
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include "rtpjitterbuffer.h"
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GST_DEBUG_CATEGORY_STATIC (rtp_jitter_buffer_debug);
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#define GST_CAT_DEFAULT rtp_jitter_buffer_debug
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/* signals and args */
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enum
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{
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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/* GObject vmethods */
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static void rtp_jitter_buffer_finalize (GObject * object);
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/* static guint rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; */
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G_DEFINE_TYPE (RTPJitterBuffer, rtp_jitter_buffer, G_TYPE_OBJECT);
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static void
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rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = (GObjectClass *) klass;
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gobject_class->finalize = rtp_jitter_buffer_finalize;
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GST_DEBUG_CATEGORY_INIT (rtp_jitter_buffer_debug, "rtpjitterbuffer", 0,
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"RTP Jitter Buffer");
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}
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static void
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rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
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{
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jbuf->packets = g_queue_new ();
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}
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static void
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rtp_jitter_buffer_finalize (GObject * object)
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{
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RTPJitterBuffer *jbuf;
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jbuf = RTP_JITTER_BUFFER_CAST (object);
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rtp_jitter_buffer_flush (jbuf);
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g_queue_free (jbuf->packets);
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G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
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}
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/**
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* rtp_jitter_buffer_new:
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*
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* Create an #RTPJitterBuffer.
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*
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* Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
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*/
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RTPJitterBuffer *
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rtp_jitter_buffer_new (void)
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{
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RTPJitterBuffer *jbuf;
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jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
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return jbuf;
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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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* rtp_jitter_buffer_insert:
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* @jbuf: an #RTPJitterBuffer
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* @buf: a buffer
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*
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* Inserts @buf into the packet queue of @jbuf. The sequence number of the
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* packet will be used to sort the packets. This function takes ownerhip of
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* @buf when the function returns %TRUE.
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*
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* Returns: %FALSE if a packet with the same number already existed.
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*/
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gboolean
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rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf)
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{
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GList *list;
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gint func_ret = 1;
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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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/* loop the list to skip strictly smaller seqnum buffers */
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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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if (func_ret == 0)
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return FALSE;
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if (list)
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g_queue_insert_before (jbuf->packets, list, buf);
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else
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g_queue_push_tail (jbuf->packets, buf);
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return TRUE;
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}
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/**
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* rtp_jitter_buffer_pop:
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* @jbuf: an #RTPJitterBuffer
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*
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* Pops the oldest buffer from the packet queue of @jbuf.
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*
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* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
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*/
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GstBuffer *
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rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf)
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{
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GstBuffer *buf;
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g_return_val_if_fail (jbuf != NULL, FALSE);
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buf = g_queue_pop_tail (jbuf->packets);
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return buf;
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}
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/**
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* rtp_jitter_buffer_flush:
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* @jbuf: an #RTPJitterBuffer
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*
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* Flush all packets from the jitterbuffer.
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*/
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void
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rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf)
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{
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GstBuffer *buffer;
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g_return_if_fail (jbuf != NULL);
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while ((buffer = g_queue_pop_head (jbuf->packets)))
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gst_buffer_unref (buffer);
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}
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/**
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* rtp_jitter_buffer_num_packets:
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* @jbuf: an #RTPJitterBuffer
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*
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* Get the number of packets currently in "jbuf.
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*
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* Returns: The number of packets in @jbuf.
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*/
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guint
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rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
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{
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g_return_val_if_fail (jbuf != NULL, 0);
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return jbuf->packets->length;
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}
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/**
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* rtp_jitter_buffer_get_ts_diff:
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* @jbuf: an #RTPJitterBuffer
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*
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* Get the difference between the timestamps of first and last packet in the
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* jitterbuffer.
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*
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* Returns: The difference expressed in the timestamp units of the packets.
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*/
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guint32
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rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
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{
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guint32 high_ts, low_ts;
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GstBuffer *high_buf, *low_buf;
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g_return_val_if_fail (jbuf != NULL, 0);
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high_buf = g_queue_peek_head (jbuf->packets);
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low_buf = g_queue_peek_tail (jbuf->packets);
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if (!high_buf || !low_buf || high_buf == low_buf)
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return 0;
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high_ts = gst_rtp_buffer_get_timestamp (high_buf);
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low_ts = gst_rtp_buffer_get_timestamp (low_buf);
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/* it needs to work if ts wraps */
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if (high_ts >= low_ts) {
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return high_ts - low_ts;
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} else {
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return high_ts + G_MAXUINT32 + 1 - low_ts;
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}
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}
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