gstreamer/gst/rtpmanager/rtpjitterbuffer.c

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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
/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include <stdlib.h>
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
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "rtpjitterbuffer.h"
GST_DEBUG_CATEGORY_STATIC (rtp_jitter_buffer_debug);
#define GST_CAT_DEFAULT rtp_jitter_buffer_debug
/* signals and args */
enum
{
LAST_SIGNAL
};
enum
{
PROP_0
};
/* GObject vmethods */
static void rtp_jitter_buffer_finalize (GObject * object);
/* static guint rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; */
G_DEFINE_TYPE (RTPJitterBuffer, rtp_jitter_buffer, G_TYPE_OBJECT);
static void
rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = rtp_jitter_buffer_finalize;
GST_DEBUG_CATEGORY_INIT (rtp_jitter_buffer_debug, "rtpjitterbuffer", 0,
"RTP Jitter Buffer");
}
static void
rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
{
jbuf->packets = g_queue_new ();
}
static void
rtp_jitter_buffer_finalize (GObject * object)
{
RTPJitterBuffer *jbuf;
jbuf = RTP_JITTER_BUFFER_CAST (object);
rtp_jitter_buffer_flush (jbuf);
g_queue_free (jbuf->packets);
G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
}
/**
* rtp_jitter_buffer_new:
*
* Create an #RTPJitterBuffer.
*
* Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
*/
RTPJitterBuffer *
rtp_jitter_buffer_new (void)
{
RTPJitterBuffer *jbuf;
jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
return jbuf;
}
static gint
compare_seqnum (GstBuffer * a, GstBuffer * b, RTPJitterBuffer * jbuf)
{
guint16 seq1, seq2;
seq1 = gst_rtp_buffer_get_seq (a);
seq2 = gst_rtp_buffer_get_seq (b);
/* check if diff more than half of the 16bit range */
if (abs (seq2 - seq1) > (1 << 15)) {
/* one of a/b has wrapped */
return seq1 - seq2;
} else {
return seq2 - seq1;
}
}
/**
* rtp_jitter_buffer_insert:
* @jbuf: an #RTPJitterBuffer
* @buf: a buffer
*
* Inserts @buf into the packet queue of @jbuf. The sequence number of the
* packet will be used to sort the packets. This function takes ownerhip of
* @buf when the function returns %TRUE.
*
* Returns: %FALSE if a packet with the same number already existed.
*/
gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf)
{
GList *list;
gint func_ret = 1;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (buf != NULL, FALSE);
/* loop the list to skip strictly smaller seqnum buffers */
list = jbuf->packets->head;
while (list
&& (func_ret =
compare_seqnum (GST_BUFFER_CAST (list->data), buf, jbuf)) < 0)
list = list->next;
/* we hit a packet with the same seqnum, return FALSE to notify a duplicate */
if (func_ret == 0)
return FALSE;
if (list)
g_queue_insert_before (jbuf->packets, list, buf);
else
g_queue_push_tail (jbuf->packets, buf);
return TRUE;
}
/**
* rtp_jitter_buffer_pop:
* @jbuf: an #RTPJitterBuffer
*
* Pops the oldest buffer from the packet queue of @jbuf.
*
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
*/
GstBuffer *
rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf)
{
GstBuffer *buf;
g_return_val_if_fail (jbuf != NULL, FALSE);
buf = g_queue_pop_tail (jbuf->packets);
return buf;
}
/**
* rtp_jitter_buffer_flush:
* @jbuf: an #RTPJitterBuffer
*
* Flush all packets from the jitterbuffer.
*/
void
rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf)
{
GstBuffer *buffer;
g_return_if_fail (jbuf != NULL);
while ((buffer = g_queue_pop_head (jbuf->packets)))
gst_buffer_unref (buffer);
}
/**
* rtp_jitter_buffer_num_packets:
* @jbuf: an #RTPJitterBuffer
*
* Get the number of packets currently in "jbuf.
*
* Returns: The number of packets in @jbuf.
*/
guint
rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
{
g_return_val_if_fail (jbuf != NULL, 0);
return jbuf->packets->length;
}
/**
* rtp_jitter_buffer_get_ts_diff:
* @jbuf: an #RTPJitterBuffer
*
* Get the difference between the timestamps of first and last packet in the
* jitterbuffer.
*
* Returns: The difference expressed in the timestamp units of the packets.
*/
guint32
rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
{
guint32 high_ts, low_ts;
GstBuffer *high_buf, *low_buf;
g_return_val_if_fail (jbuf != NULL, 0);
high_buf = g_queue_peek_head (jbuf->packets);
low_buf = g_queue_peek_tail (jbuf->packets);
if (!high_buf || !low_buf || high_buf == low_buf)
return 0;
high_ts = gst_rtp_buffer_get_timestamp (high_buf);
low_ts = gst_rtp_buffer_get_timestamp (low_buf);
/* it needs to work if ts wraps */
if (high_ts >= low_ts) {
return high_ts - low_ts;
} else {
return high_ts + G_MAXUINT32 + 1 - low_ts;
}
}