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When set this property will allow the jitterbuffer to start delivering packets as soon as N most recent packets have consecutive seqnum. A faststart-min-packets of zero disables this feature. This heuristic is also used in rtpsource which implements the probation mechanism and a similar heuristic is used to handle long gaps. https://bugzilla.gnome.org/show_bug.cgi?id=769536
1297 lines
38 KiB
C
1297 lines
38 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <string.h>
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#include <stdlib.h>
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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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#define MAX_WINDOW RTP_JITTER_BUFFER_MAX_WINDOW
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#define MAX_TIME (2 * GST_SECOND)
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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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GType
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rtp_jitter_buffer_mode_get_type (void)
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{
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static GType jitter_buffer_mode_type = 0;
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static const GEnumValue jitter_buffer_modes[] = {
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{RTP_JITTER_BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
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{RTP_JITTER_BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
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{RTP_JITTER_BUFFER_MODE_BUFFER, "Do low/high watermark buffering",
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"buffer"},
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{RTP_JITTER_BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks",
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"synced"},
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{0, NULL, NULL},
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};
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if (!jitter_buffer_mode_type) {
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jitter_buffer_mode_type =
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g_enum_register_static ("RTPJitterBufferMode", jitter_buffer_modes);
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}
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return jitter_buffer_mode_type;
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}
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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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g_mutex_init (&jbuf->clock_lock);
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jbuf->packets = g_queue_new ();
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jbuf->mode = RTP_JITTER_BUFFER_MODE_SLAVE;
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rtp_jitter_buffer_reset_skew (jbuf);
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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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if (jbuf->media_clock_synced_id)
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g_signal_handler_disconnect (jbuf->media_clock,
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jbuf->media_clock_synced_id);
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if (jbuf->media_clock)
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gst_object_unref (jbuf->media_clock);
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if (jbuf->pipeline_clock)
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gst_object_unref (jbuf->pipeline_clock);
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g_queue_free (jbuf->packets);
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g_mutex_clear (&jbuf->clock_lock);
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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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/**
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* rtp_jitter_buffer_get_mode:
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* @jbuf: an #RTPJitterBuffer
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*
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* Get the current jitterbuffer mode.
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*
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* Returns: the current jitterbuffer mode.
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*/
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RTPJitterBufferMode
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rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf)
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{
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return jbuf->mode;
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}
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/**
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* rtp_jitter_buffer_set_mode:
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* @jbuf: an #RTPJitterBuffer
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* @mode: a #RTPJitterBufferMode
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*
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* Set the buffering and clock slaving algorithm used in the @jbuf.
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*/
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void
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rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode)
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{
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jbuf->mode = mode;
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}
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GstClockTime
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rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf)
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{
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return jbuf->delay;
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}
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void
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rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay)
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{
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jbuf->delay = delay;
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jbuf->low_level = (delay * 15) / 100;
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/* the high level is at 90% in order to release packets before we fill up the
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* buffer up to the latency */
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jbuf->high_level = (delay * 90) / 100;
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GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %"
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GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay),
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GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
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}
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/**
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* rtp_jitter_buffer_set_clock_rate:
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* @jbuf: an #RTPJitterBuffer
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* @clock_rate: the new clock rate
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*
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* Set the clock rate in the jitterbuffer.
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*/
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void
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rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
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{
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if (jbuf->clock_rate != clock_rate) {
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GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
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G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
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jbuf->clock_rate = clock_rate;
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rtp_jitter_buffer_reset_skew (jbuf);
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}
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}
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/**
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* rtp_jitter_buffer_get_clock_rate:
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* @jbuf: an #RTPJitterBuffer
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*
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* Get the currently configure clock rate in @jbuf.
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*
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* Returns: the current clock-rate
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*/
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guint32
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rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
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{
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return jbuf->clock_rate;
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}
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static void
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media_clock_synced_cb (GstClock * clock, gboolean synced,
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RTPJitterBuffer * jbuf)
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{
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GstClockTime internal, external;
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g_mutex_lock (&jbuf->clock_lock);
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if (jbuf->pipeline_clock) {
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internal = gst_clock_get_internal_time (jbuf->media_clock);
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external = gst_clock_get_time (jbuf->pipeline_clock);
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gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
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}
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g_mutex_unlock (&jbuf->clock_lock);
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}
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/**
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* rtp_jitter_buffer_set_media_clock:
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* @jbuf: an #RTPJitterBuffer
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* @clock: (transfer full): media #GstClock
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* @clock_offset: RTP time at clock epoch or -1
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*
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* Sets the media clock for the media and the clock offset
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*
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*/
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void
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rtp_jitter_buffer_set_media_clock (RTPJitterBuffer * jbuf, GstClock * clock,
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guint64 clock_offset)
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{
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g_mutex_lock (&jbuf->clock_lock);
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if (jbuf->media_clock) {
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if (jbuf->media_clock_synced_id)
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g_signal_handler_disconnect (jbuf->media_clock,
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jbuf->media_clock_synced_id);
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jbuf->media_clock_synced_id = 0;
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gst_object_unref (jbuf->media_clock);
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}
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jbuf->media_clock = clock;
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jbuf->media_clock_offset = clock_offset;
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if (jbuf->pipeline_clock && jbuf->media_clock &&
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jbuf->pipeline_clock != jbuf->media_clock) {
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jbuf->media_clock_synced_id =
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g_signal_connect (jbuf->media_clock, "synced",
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G_CALLBACK (media_clock_synced_cb), jbuf);
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if (gst_clock_is_synced (jbuf->media_clock)) {
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GstClockTime internal, external;
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internal = gst_clock_get_internal_time (jbuf->media_clock);
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external = gst_clock_get_time (jbuf->pipeline_clock);
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gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
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}
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gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
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}
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g_mutex_unlock (&jbuf->clock_lock);
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}
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/**
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* rtp_jitter_buffer_set_pipeline_clock:
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* @jbuf: an #RTPJitterBuffer
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* @clock: pipeline #GstClock
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*
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* Sets the pipeline clock
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*
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*/
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void
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rtp_jitter_buffer_set_pipeline_clock (RTPJitterBuffer * jbuf, GstClock * clock)
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{
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g_mutex_lock (&jbuf->clock_lock);
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if (jbuf->pipeline_clock)
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gst_object_unref (jbuf->pipeline_clock);
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jbuf->pipeline_clock = clock ? gst_object_ref (clock) : NULL;
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if (jbuf->pipeline_clock && jbuf->media_clock &&
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jbuf->pipeline_clock != jbuf->media_clock) {
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if (gst_clock_is_synced (jbuf->media_clock)) {
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GstClockTime internal, external;
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internal = gst_clock_get_internal_time (jbuf->media_clock);
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external = gst_clock_get_time (jbuf->pipeline_clock);
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gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
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}
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gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
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}
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g_mutex_unlock (&jbuf->clock_lock);
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}
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gboolean
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rtp_jitter_buffer_get_rfc7273_sync (RTPJitterBuffer * jbuf)
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{
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return jbuf->rfc7273_sync;
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}
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void
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rtp_jitter_buffer_set_rfc7273_sync (RTPJitterBuffer * jbuf,
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gboolean rfc7273_sync)
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{
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jbuf->rfc7273_sync = rfc7273_sync;
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}
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/**
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* rtp_jitter_buffer_reset_skew:
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* @jbuf: an #RTPJitterBuffer
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*
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* Reset the skew calculations in @jbuf.
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*/
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void
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rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
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{
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jbuf->base_time = -1;
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jbuf->base_rtptime = -1;
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jbuf->base_extrtp = -1;
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jbuf->media_clock_base_time = -1;
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jbuf->ext_rtptime = -1;
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jbuf->last_rtptime = -1;
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jbuf->window_pos = 0;
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jbuf->window_filling = TRUE;
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jbuf->window_min = 0;
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jbuf->skew = 0;
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jbuf->prev_send_diff = -1;
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jbuf->prev_out_time = -1;
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jbuf->need_resync = TRUE;
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GST_DEBUG ("reset skew correction");
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}
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/**
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* rtp_jitter_buffer_disable_buffering:
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* @jbuf: an #RTPJitterBuffer
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* @disabled: the new state
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*
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* Enable or disable buffering on @jbuf.
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*/
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void
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rtp_jitter_buffer_disable_buffering (RTPJitterBuffer * jbuf, gboolean disabled)
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{
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jbuf->buffering_disabled = disabled;
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}
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static void
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rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
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GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
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{
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jbuf->base_time = time;
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jbuf->media_clock_base_time = -1;
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jbuf->base_rtptime = gstrtptime;
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jbuf->base_extrtp = ext_rtptime;
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jbuf->prev_out_time = -1;
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jbuf->prev_send_diff = -1;
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if (reset_skew) {
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jbuf->window_filling = TRUE;
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jbuf->window_pos = 0;
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jbuf->window_min = 0;
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jbuf->window_size = 0;
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jbuf->skew = 0;
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}
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jbuf->need_resync = FALSE;
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}
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static guint64
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get_buffer_level (RTPJitterBuffer * jbuf)
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{
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RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
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guint64 level;
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/* first buffer with timestamp */
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high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
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while (high_buf) {
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if (high_buf->dts != -1 || high_buf->pts != -1)
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break;
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high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
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}
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low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
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while (low_buf) {
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if (low_buf->dts != -1 || low_buf->pts != -1)
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break;
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low_buf = (RTPJitterBufferItem *) g_list_next (low_buf);
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}
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if (!high_buf || !low_buf || high_buf == low_buf) {
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level = 0;
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} else {
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guint64 high_ts, low_ts;
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high_ts = high_buf->dts != -1 ? high_buf->dts : high_buf->pts;
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low_ts = low_buf->dts != -1 ? low_buf->dts : low_buf->pts;
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if (high_ts > low_ts)
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level = high_ts - low_ts;
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else
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level = 0;
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GST_LOG_OBJECT (jbuf,
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"low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
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G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
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level);
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}
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return level;
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}
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static void
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update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
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{
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gboolean post = FALSE;
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guint64 level;
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level = get_buffer_level (jbuf);
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GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
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if (jbuf->buffering_disabled) {
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GST_DEBUG ("buffering is disabled");
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level = jbuf->high_level;
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}
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if (jbuf->buffering) {
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post = TRUE;
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if (level >= jbuf->high_level) {
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GST_DEBUG ("buffering finished");
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jbuf->buffering = FALSE;
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}
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} else {
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if (level < jbuf->low_level) {
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GST_DEBUG ("buffering started");
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jbuf->buffering = TRUE;
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post = TRUE;
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}
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}
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if (post) {
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gint perc;
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if (jbuf->buffering && (jbuf->high_level != 0)) {
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perc = (level * 100 / jbuf->high_level);
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perc = MIN (perc, 100);
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} else {
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perc = 100;
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}
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if (percent)
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*percent = perc;
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GST_DEBUG ("buffering %d", perc);
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}
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}
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/* For the clock skew we use a windowed low point averaging algorithm as can be
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* found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
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* over Network Delays":
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* http://www.grame.fr/Ressources/pub/TR-050601.pdf
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* http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
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*
|
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* The idea is that the jitter is composed of:
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*
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* J = N + n
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*
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* N : a constant network delay.
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* n : random added noise. The noise is concentrated around 0
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*
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* In the receiver we can track the elapsed time at the sender with:
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*
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* send_diff(i) = (Tsi - Ts0);
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*
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* Tsi : The time at the sender at packet i
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* Ts0 : The time at the sender at the first packet
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*
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* This is the difference between the RTP timestamp in the first received packet
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* and the current packet.
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*
|
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* At the receiver we have to deal with the jitter introduced by the network.
|
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*
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* recv_diff(i) = (Tri - Tr0)
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*
|
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* Tri : The time at the receiver at packet i
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* Tr0 : The time at the receiver at the first packet
|
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*
|
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* Both of these values contain a jitter Ji, a jitter for packet i, so we can
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* write:
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*
|
|
* recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
|
|
*
|
|
* Cri : The time of the clock at the receiver for packet i
|
|
* D + ni : The jitter when receiving packet i
|
|
*
|
|
* We see that the network delay is irrelevant here as we can elliminate D:
|
|
*
|
|
* recv_diff(i) = (Cri + ni) - (Cr0 + n0))
|
|
*
|
|
* The drift is now expressed as:
|
|
*
|
|
* Drift(i) = recv_diff(i) - send_diff(i);
|
|
*
|
|
* We now keep the W latest values of Drift and find the minimum (this is the
|
|
* one with the lowest network jitter and thus the one which is least affected
|
|
* by it). We average this lowest value to smooth out the resulting network skew.
|
|
*
|
|
* Both the window and the weighting used for averaging influence the accuracy
|
|
* of the drift estimation. Finding the correct parameters turns out to be a
|
|
* compromise between accuracy and inertia.
|
|
*
|
|
* We use a 2 second window or up to 512 data points, which is statistically big
|
|
* enough to catch spikes (FIXME, detect spikes).
|
|
* We also use a rather large weighting factor (125) to smoothly adapt. During
|
|
* startup, when filling the window, we use a parabolic weighting factor, the
|
|
* more the window is filled, the faster we move to the detected possible skew.
|
|
*
|
|
* Returns: @time adjusted with the clock skew.
|
|
*/
|
|
static GstClockTime
|
|
calculate_skew (RTPJitterBuffer * jbuf, guint64 ext_rtptime,
|
|
GstClockTime gstrtptime, GstClockTime time)
|
|
{
|
|
guint64 send_diff, recv_diff;
|
|
gint64 delta;
|
|
gint64 old;
|
|
gint pos, i;
|
|
GstClockTime out_time;
|
|
guint64 slope;
|
|
|
|
/* elapsed time at sender */
|
|
send_diff = gstrtptime - jbuf->base_rtptime;
|
|
|
|
/* we don't have an arrival timestamp so we can't do skew detection. we
|
|
* should still apply a timestamp based on RTP timestamp and base_time */
|
|
if (time == -1 || jbuf->base_time == -1)
|
|
goto no_skew;
|
|
|
|
/* elapsed time at receiver, includes the jitter */
|
|
recv_diff = time - jbuf->base_time;
|
|
|
|
/* measure the diff */
|
|
delta = ((gint64) recv_diff) - ((gint64) send_diff);
|
|
|
|
/* measure the slope, this gives a rought estimate between the sender speed
|
|
* and the receiver speed. This should be approximately 8, higher values
|
|
* indicate a burst (especially when the connection starts) */
|
|
if (recv_diff > 0)
|
|
slope = (send_diff * 8) / recv_diff;
|
|
else
|
|
slope = 8;
|
|
|
|
GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
|
|
GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
|
|
GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
|
|
|
|
/* if the difference between the sender timeline and the receiver timeline
|
|
* changed too quickly we have to resync because the server likely restarted
|
|
* its timestamps. */
|
|
if (ABS (delta - jbuf->skew) > GST_SECOND) {
|
|
GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
|
|
GST_TIME_ARGS (ABS (delta - jbuf->skew)));
|
|
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
|
|
send_diff = 0;
|
|
delta = 0;
|
|
}
|
|
|
|
pos = jbuf->window_pos;
|
|
|
|
if (G_UNLIKELY (jbuf->window_filling)) {
|
|
/* we are filling the window */
|
|
GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
|
|
jbuf->window[pos++] = delta;
|
|
/* calc the min delta we observed */
|
|
if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
|
|
jbuf->window_min = delta;
|
|
|
|
if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
|
|
jbuf->window_size = pos;
|
|
|
|
/* window filled */
|
|
GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
|
|
|
|
/* the skew is now the min */
|
|
jbuf->skew = jbuf->window_min;
|
|
jbuf->window_filling = FALSE;
|
|
} else {
|
|
gint perc_time, perc_window, perc;
|
|
|
|
/* figure out how much we filled the window, this depends on the amount of
|
|
* time we have or the max number of points we keep. */
|
|
perc_time = send_diff * 100 / MAX_TIME;
|
|
perc_window = pos * 100 / MAX_WINDOW;
|
|
perc = MAX (perc_time, perc_window);
|
|
|
|
/* make a parabolic function, the closer we get to the MAX, the more value
|
|
* we give to the scaling factor of the new value */
|
|
perc = perc * perc;
|
|
|
|
/* quickly go to the min value when we are filling up, slowly when we are
|
|
* just starting because we're not sure it's a good value yet. */
|
|
jbuf->skew =
|
|
(perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
|
|
jbuf->window_size = pos + 1;
|
|
}
|
|
} else {
|
|
/* pick old value and store new value. We keep the previous value in order
|
|
* to quickly check if the min of the window changed */
|
|
old = jbuf->window[pos];
|
|
jbuf->window[pos++] = delta;
|
|
|
|
if (G_UNLIKELY (delta <= jbuf->window_min)) {
|
|
/* if the new value we inserted is smaller or equal to the current min,
|
|
* it becomes the new min */
|
|
jbuf->window_min = delta;
|
|
} else if (G_UNLIKELY (old == jbuf->window_min)) {
|
|
gint64 min = G_MAXINT64;
|
|
|
|
/* if we removed the old min, we have to find a new min */
|
|
for (i = 0; i < jbuf->window_size; i++) {
|
|
/* we found another value equal to the old min, we can stop searching now */
|
|
if (jbuf->window[i] == old) {
|
|
min = old;
|
|
break;
|
|
}
|
|
if (jbuf->window[i] < min)
|
|
min = jbuf->window[i];
|
|
}
|
|
jbuf->window_min = min;
|
|
}
|
|
/* average the min values */
|
|
jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
|
|
GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
|
|
delta, jbuf->window_min);
|
|
}
|
|
/* wrap around in the window */
|
|
if (G_UNLIKELY (pos >= jbuf->window_size))
|
|
pos = 0;
|
|
jbuf->window_pos = pos;
|
|
|
|
no_skew:
|
|
/* the output time is defined as the base timestamp plus the RTP time
|
|
* adjusted for the clock skew .*/
|
|
if (jbuf->base_time != -1) {
|
|
out_time = jbuf->base_time + send_diff;
|
|
/* skew can be negative and we don't want to make invalid timestamps */
|
|
if (jbuf->skew < 0 && out_time < -jbuf->skew) {
|
|
out_time = 0;
|
|
} else {
|
|
out_time += jbuf->skew;
|
|
}
|
|
} else
|
|
out_time = -1;
|
|
|
|
GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
|
|
jbuf->skew, GST_TIME_ARGS (out_time));
|
|
|
|
return out_time;
|
|
}
|
|
|
|
static void
|
|
queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
|
|
{
|
|
GQueue *queue = jbuf->packets;
|
|
|
|
/* It's more likely that the packet was inserted at the tail of the queue */
|
|
if (G_LIKELY (list)) {
|
|
item->prev = list;
|
|
item->next = list->next;
|
|
list->next = item;
|
|
} else {
|
|
item->prev = NULL;
|
|
item->next = queue->head;
|
|
queue->head = item;
|
|
}
|
|
if (item->next)
|
|
item->next->prev = item;
|
|
else
|
|
queue->tail = item;
|
|
queue->length++;
|
|
}
|
|
|
|
GstClockTime
|
|
rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
|
|
gboolean estimated_dts, guint32 rtptime, GstClockTime base_time)
|
|
{
|
|
guint64 ext_rtptime;
|
|
GstClockTime gstrtptime, pts;
|
|
GstClock *media_clock, *pipeline_clock;
|
|
guint64 media_clock_offset;
|
|
gboolean rfc7273_mode;
|
|
|
|
/* rtp time jumps are checked for during skew calculation, but bypassed
|
|
* in other mode, so mind those here and reset jb if needed.
|
|
* Only reset if valid input time, which is likely for UDP input
|
|
* where we expect this might happen due to async thread effects
|
|
* (in seek and state change cycles), but not so much for TCP input */
|
|
if (GST_CLOCK_TIME_IS_VALID (dts) && !estimated_dts &&
|
|
jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
|
|
jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
|
|
GstClockTime ext_rtptime = jbuf->ext_rtptime;
|
|
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
|
|
if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
|
|
ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
|
|
/* reset even if we don't have valid incoming time;
|
|
* still better than producing possibly very bogus output timestamp */
|
|
GST_WARNING ("rtp delta too big, reset skew");
|
|
rtp_jitter_buffer_reset_skew (jbuf);
|
|
}
|
|
}
|
|
|
|
/* Return the last time if we got the same RTP timestamp again */
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
|
|
if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime) {
|
|
return jbuf->prev_out_time;
|
|
}
|
|
|
|
/* keep track of the last extended rtptime */
|
|
jbuf->last_rtptime = ext_rtptime;
|
|
|
|
g_mutex_lock (&jbuf->clock_lock);
|
|
media_clock = jbuf->media_clock ? gst_object_ref (jbuf->media_clock) : NULL;
|
|
pipeline_clock =
|
|
jbuf->pipeline_clock ? gst_object_ref (jbuf->pipeline_clock) : NULL;
|
|
media_clock_offset = jbuf->media_clock_offset;
|
|
g_mutex_unlock (&jbuf->clock_lock);
|
|
|
|
gstrtptime =
|
|
gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
|
|
|
|
if (G_LIKELY (jbuf->base_rtptime != -1)) {
|
|
/* check elapsed time in RTP units */
|
|
if (gstrtptime < jbuf->base_rtptime) {
|
|
/* elapsed time at sender, timestamps can go backwards and thus be
|
|
* smaller than our base time, schedule to take a new base time in
|
|
* that case. */
|
|
GST_WARNING ("backward timestamps at server, schedule resync");
|
|
jbuf->need_resync = TRUE;
|
|
}
|
|
}
|
|
|
|
switch (jbuf->mode) {
|
|
case RTP_JITTER_BUFFER_MODE_NONE:
|
|
case RTP_JITTER_BUFFER_MODE_BUFFER:
|
|
/* send 0 as the first timestamp and -1 for the other ones. This will
|
|
* interpolate them from the RTP timestamps with a 0 origin. In buffering
|
|
* mode we will adjust the outgoing timestamps according to the amount of
|
|
* time we spent buffering. */
|
|
if (jbuf->base_time == -1)
|
|
dts = 0;
|
|
else
|
|
dts = -1;
|
|
break;
|
|
case RTP_JITTER_BUFFER_MODE_SYNCED:
|
|
/* synchronized clocks, take first timestamp as base, use RTP timestamps
|
|
* to interpolate */
|
|
if (jbuf->base_time != -1 && !jbuf->need_resync)
|
|
dts = -1;
|
|
break;
|
|
case RTP_JITTER_BUFFER_MODE_SLAVE:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* need resync, lock on to time and gstrtptime if we can, otherwise we
|
|
* do with the previous values */
|
|
if (G_UNLIKELY (jbuf->need_resync && dts != -1)) {
|
|
GST_INFO ("resync to time %" GST_TIME_FORMAT ", rtptime %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (dts), GST_TIME_ARGS (gstrtptime));
|
|
rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, FALSE);
|
|
}
|
|
|
|
GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
|
|
GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
|
|
GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
|
|
GST_TIME_ARGS (gstrtptime - jbuf->base_rtptime));
|
|
|
|
rfc7273_mode = media_clock && pipeline_clock
|
|
&& gst_clock_is_synced (media_clock);
|
|
|
|
if (rfc7273_mode && jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
|
|
&& (media_clock_offset == -1 || !jbuf->rfc7273_sync)) {
|
|
GstClockTime internal, external;
|
|
GstClockTime rate_num, rate_denom;
|
|
GstClockTime nsrtptimediff, rtpntptime, rtpsystime;
|
|
|
|
gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
|
|
&rate_denom);
|
|
|
|
/* Slave to the RFC7273 media clock instead of trying to estimate it
|
|
* based on receive times and RTP timestamps */
|
|
|
|
if (jbuf->media_clock_base_time == -1) {
|
|
if (jbuf->base_time != -1) {
|
|
jbuf->media_clock_base_time =
|
|
gst_clock_unadjust_with_calibration (media_clock,
|
|
jbuf->base_time + base_time, internal, external, rate_num,
|
|
rate_denom);
|
|
} else {
|
|
if (dts != -1)
|
|
jbuf->media_clock_base_time =
|
|
gst_clock_unadjust_with_calibration (media_clock, dts + base_time,
|
|
internal, external, rate_num, rate_denom);
|
|
else
|
|
jbuf->media_clock_base_time =
|
|
gst_clock_get_internal_time (media_clock);
|
|
jbuf->base_rtptime = gstrtptime;
|
|
}
|
|
}
|
|
|
|
if (gstrtptime > jbuf->base_rtptime)
|
|
nsrtptimediff = gstrtptime - jbuf->base_rtptime;
|
|
else
|
|
nsrtptimediff = 0;
|
|
|
|
rtpntptime = nsrtptimediff + jbuf->media_clock_base_time;
|
|
|
|
rtpsystime =
|
|
gst_clock_adjust_with_calibration (media_clock, rtpntptime, internal,
|
|
external, rate_num, rate_denom);
|
|
|
|
if (rtpsystime > base_time)
|
|
pts = rtpsystime - base_time;
|
|
else
|
|
pts = 0;
|
|
|
|
GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
|
|
} else if (rfc7273_mode && (jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
|
|
|| jbuf->mode == RTP_JITTER_BUFFER_MODE_SYNCED)
|
|
&& media_clock_offset != -1 && jbuf->rfc7273_sync) {
|
|
GstClockTime ntptime, rtptime_tmp;
|
|
GstClockTime ntprtptime, rtpsystime;
|
|
GstClockTime internal, external;
|
|
GstClockTime rate_num, rate_denom;
|
|
|
|
/* Don't do any of the dts related adjustments further down */
|
|
dts = -1;
|
|
|
|
/* Calculate the actual clock time on the sender side based on the
|
|
* RFC7273 clock and convert it to our pipeline clock
|
|
*/
|
|
|
|
gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
|
|
&rate_denom);
|
|
|
|
ntptime = gst_clock_get_internal_time (media_clock);
|
|
|
|
ntprtptime = gst_util_uint64_scale (ntptime, jbuf->clock_rate, GST_SECOND);
|
|
ntprtptime += media_clock_offset;
|
|
ntprtptime &= 0xffffffff;
|
|
|
|
rtptime_tmp = rtptime;
|
|
/* Check for wraparounds, we assume that the diff between current RTP
|
|
* timestamp and current media clock time can't be bigger than
|
|
* 2**31 clock units */
|
|
if (ntprtptime > rtptime_tmp && ntprtptime - rtptime_tmp >= 0x80000000)
|
|
rtptime_tmp += G_GUINT64_CONSTANT (0x100000000);
|
|
else if (rtptime_tmp > ntprtptime && rtptime_tmp - ntprtptime >= 0x80000000)
|
|
ntprtptime += G_GUINT64_CONSTANT (0x100000000);
|
|
|
|
if (ntprtptime > rtptime_tmp)
|
|
ntptime -=
|
|
gst_util_uint64_scale (ntprtptime - rtptime_tmp, jbuf->clock_rate,
|
|
GST_SECOND);
|
|
else
|
|
ntptime +=
|
|
gst_util_uint64_scale (rtptime_tmp - ntprtptime, jbuf->clock_rate,
|
|
GST_SECOND);
|
|
|
|
rtpsystime =
|
|
gst_clock_adjust_with_calibration (media_clock, ntptime, internal,
|
|
external, rate_num, rate_denom);
|
|
/* All this assumes that the pipeline has enough additional
|
|
* latency to cover for the network delay */
|
|
if (rtpsystime > base_time)
|
|
pts = rtpsystime - base_time;
|
|
else
|
|
pts = 0;
|
|
|
|
GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
|
|
} else {
|
|
/* If we used the RFC7273 clock before and not anymore,
|
|
* we need to resync it later again */
|
|
jbuf->media_clock_base_time = -1;
|
|
|
|
/* do skew calculation by measuring the difference between rtptime and the
|
|
* receive dts, this function will return the skew corrected rtptime. */
|
|
pts = calculate_skew (jbuf, ext_rtptime, gstrtptime, dts);
|
|
}
|
|
|
|
/* check if timestamps are not going backwards, we can only check this if we
|
|
* have a previous out time and a previous send_diff */
|
|
if (G_LIKELY (pts != -1 && jbuf->prev_out_time != -1
|
|
&& jbuf->prev_send_diff != -1)) {
|
|
/* now check for backwards timestamps */
|
|
if (G_UNLIKELY (
|
|
/* if the server timestamps went up and the out_time backwards */
|
|
(gstrtptime - jbuf->base_rtptime > jbuf->prev_send_diff
|
|
&& pts < jbuf->prev_out_time) ||
|
|
/* if the server timestamps went backwards and the out_time forwards */
|
|
(gstrtptime - jbuf->base_rtptime < jbuf->prev_send_diff
|
|
&& pts > jbuf->prev_out_time) ||
|
|
/* if the server timestamps did not change */
|
|
gstrtptime - jbuf->base_rtptime == jbuf->prev_send_diff)) {
|
|
GST_DEBUG ("backwards timestamps, using previous time");
|
|
pts = jbuf->prev_out_time;
|
|
}
|
|
}
|
|
|
|
if (dts != -1 && pts + jbuf->delay < dts) {
|
|
/* if we are going to produce a timestamp that is later than the input
|
|
* timestamp, we need to reset the jitterbuffer. Likely the server paused
|
|
* temporarily */
|
|
GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
|
|
GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (pts),
|
|
jbuf->delay, GST_TIME_ARGS (dts));
|
|
rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, TRUE);
|
|
pts = dts;
|
|
}
|
|
|
|
jbuf->prev_out_time = pts;
|
|
jbuf->prev_send_diff = gstrtptime - jbuf->base_rtptime;
|
|
|
|
if (media_clock)
|
|
gst_object_unref (media_clock);
|
|
if (pipeline_clock)
|
|
gst_object_unref (pipeline_clock);
|
|
|
|
return pts;
|
|
}
|
|
|
|
|
|
/**
|
|
* rtp_jitter_buffer_insert:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
* @item: an #RTPJitterBufferItem to insert
|
|
* @head: TRUE when the head element changed.
|
|
* @percent: the buffering percent after insertion
|
|
*
|
|
* Inserts @item 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.
|
|
*
|
|
* When @head is %TRUE, the new packet was added at the head of the queue and
|
|
* will be available with the next call to rtp_jitter_buffer_pop() and
|
|
* rtp_jitter_buffer_peek().
|
|
*
|
|
* Returns: %FALSE if a packet with the same number already existed.
|
|
*/
|
|
gboolean
|
|
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
|
|
gboolean * head, gint * percent)
|
|
{
|
|
GList *list, *event = NULL;
|
|
guint16 seqnum;
|
|
|
|
g_return_val_if_fail (jbuf != NULL, FALSE);
|
|
g_return_val_if_fail (item != NULL, FALSE);
|
|
|
|
list = jbuf->packets->tail;
|
|
|
|
/* no seqnum, simply append then */
|
|
if (item->seqnum == -1)
|
|
goto append;
|
|
|
|
seqnum = item->seqnum;
|
|
|
|
/* loop the list to skip strictly larger seqnum buffers */
|
|
for (; list; list = g_list_previous (list)) {
|
|
guint16 qseq;
|
|
gint gap;
|
|
RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
|
|
|
|
if (qitem->seqnum == -1) {
|
|
/* keep a pointer to the first consecutive event if not already
|
|
* set. we will insert the packet after the event if we can't find
|
|
* a packet with lower sequence number before the event. */
|
|
if (event == NULL)
|
|
event = list;
|
|
continue;
|
|
}
|
|
|
|
qseq = qitem->seqnum;
|
|
|
|
/* compare the new seqnum to the one in the buffer */
|
|
gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
|
|
|
|
/* we hit a packet with the same seqnum, notify a duplicate */
|
|
if (G_UNLIKELY (gap == 0))
|
|
goto duplicate;
|
|
|
|
/* seqnum > qseq, we can stop looking */
|
|
if (G_LIKELY (gap < 0))
|
|
break;
|
|
|
|
/* if we've found a packet with greater sequence number, cleanup the
|
|
* event pointer as the packet will be inserted before the event */
|
|
event = NULL;
|
|
}
|
|
|
|
/* if event is set it means that packets before the event had smaller
|
|
* sequence number, so we will insert our packet after the event */
|
|
if (event)
|
|
list = event;
|
|
|
|
append:
|
|
queue_do_insert (jbuf, list, (GList *) item);
|
|
|
|
/* buffering mode, update buffer stats */
|
|
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
|
|
update_buffer_level (jbuf, percent);
|
|
else if (percent)
|
|
*percent = -1;
|
|
|
|
/* head was changed when we did not find a previous packet, we set the return
|
|
* flag when requested. */
|
|
if (G_LIKELY (head))
|
|
*head = (list == NULL);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
duplicate:
|
|
{
|
|
GST_DEBUG ("duplicate packet %d found", (gint) seqnum);
|
|
if (G_LIKELY (head))
|
|
*head = FALSE;
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_pop:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
* @percent: the buffering percent
|
|
*
|
|
* Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
|
|
* have its timestamp adjusted with the incomming running_time and the detected
|
|
* clock skew.
|
|
*
|
|
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
|
|
*/
|
|
RTPJitterBufferItem *
|
|
rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
|
|
{
|
|
GList *item = NULL;
|
|
GQueue *queue;
|
|
|
|
g_return_val_if_fail (jbuf != NULL, NULL);
|
|
|
|
queue = jbuf->packets;
|
|
|
|
item = queue->head;
|
|
if (item) {
|
|
queue->head = item->next;
|
|
if (queue->head)
|
|
queue->head->prev = NULL;
|
|
else
|
|
queue->tail = NULL;
|
|
queue->length--;
|
|
}
|
|
|
|
/* buffering mode, update buffer stats */
|
|
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
|
|
update_buffer_level (jbuf, percent);
|
|
else if (percent)
|
|
*percent = -1;
|
|
|
|
return (RTPJitterBufferItem *) item;
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_peek:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
*
|
|
* Peek the oldest buffer from the packet queue of @jbuf.
|
|
*
|
|
* See rtp_jitter_buffer_insert() to check when an older packet was
|
|
* added.
|
|
*
|
|
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
|
|
*/
|
|
RTPJitterBufferItem *
|
|
rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
|
|
{
|
|
g_return_val_if_fail (jbuf != NULL, NULL);
|
|
|
|
return (RTPJitterBufferItem *) jbuf->packets->head;
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_flush:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
* @free_func: function to free each item
|
|
* @user_data: user data passed to @free_func
|
|
*
|
|
* Flush all packets from the jitterbuffer.
|
|
*/
|
|
void
|
|
rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
|
|
gpointer user_data)
|
|
{
|
|
GList *item;
|
|
|
|
g_return_if_fail (jbuf != NULL);
|
|
g_return_if_fail (free_func != NULL);
|
|
|
|
while ((item = g_queue_pop_head_link (jbuf->packets)))
|
|
free_func ((RTPJitterBufferItem *) item, user_data);
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_is_buffering:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
*
|
|
* Check if @jbuf is buffering currently. Users of the jitterbuffer should not
|
|
* pop packets while in buffering mode.
|
|
*
|
|
* Returns: the buffering state of @jbuf
|
|
*/
|
|
gboolean
|
|
rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
|
|
{
|
|
return jbuf->buffering && !jbuf->buffering_disabled;
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_set_buffering:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
* @buffering: the new buffering state
|
|
*
|
|
* Forces @jbuf to go into the buffering state.
|
|
*/
|
|
void
|
|
rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
|
|
{
|
|
jbuf->buffering = buffering;
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_get_percent:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
*
|
|
* Get the buffering percent of the jitterbuffer.
|
|
*
|
|
* Returns: the buffering percent
|
|
*/
|
|
gint
|
|
rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
|
|
{
|
|
gint percent;
|
|
guint64 level;
|
|
|
|
if (G_UNLIKELY (jbuf->high_level == 0))
|
|
return 100;
|
|
|
|
if (G_UNLIKELY (jbuf->buffering_disabled))
|
|
return 100;
|
|
|
|
level = get_buffer_level (jbuf);
|
|
percent = (level * 100 / jbuf->high_level);
|
|
percent = MIN (percent, 100);
|
|
|
|
return percent;
|
|
}
|
|
|
|
/**
|
|
* 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)
|
|
{
|
|
guint64 high_ts, low_ts;
|
|
RTPJitterBufferItem *high_buf, *low_buf;
|
|
guint32 result;
|
|
|
|
g_return_val_if_fail (jbuf != NULL, 0);
|
|
|
|
high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
|
|
low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
|
|
|
|
if (!high_buf || !low_buf || high_buf == low_buf)
|
|
return 0;
|
|
|
|
high_ts = high_buf->rtptime;
|
|
low_ts = low_buf->rtptime;
|
|
|
|
/* it needs to work if ts wraps */
|
|
if (high_ts >= low_ts) {
|
|
result = (guint32) (high_ts - low_ts);
|
|
} else {
|
|
result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_get_sync:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
* @rtptime: result RTP time
|
|
* @timestamp: result GStreamer timestamp
|
|
* @clock_rate: clock-rate of @rtptime
|
|
* @last_rtptime: last seen rtptime.
|
|
*
|
|
* Calculates the relation between the RTP timestamp and the GStreamer timestamp
|
|
* used for constructing timestamps.
|
|
*
|
|
* For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
|
|
* the GStreamer timestamp is currently @timestamp.
|
|
*
|
|
* The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
|
|
* @last_rtptime.
|
|
*/
|
|
void
|
|
rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
|
|
guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
|
|
{
|
|
if (rtptime)
|
|
*rtptime = jbuf->base_extrtp;
|
|
if (timestamp)
|
|
*timestamp = jbuf->base_time + jbuf->skew;
|
|
if (clock_rate)
|
|
*clock_rate = jbuf->clock_rate;
|
|
if (last_rtptime)
|
|
*last_rtptime = jbuf->last_rtptime;
|
|
}
|
|
|
|
/**
|
|
* rtp_jitter_buffer_can_fast_start:
|
|
* @jbuf: an #RTPJitterBuffer
|
|
* @num_packets: Number of consecutive packets needed
|
|
*
|
|
* Check if in the queue if there is enough packets with consecutive seqnum in
|
|
* order to start delivering them.
|
|
*
|
|
* Returns: %TRUE if the required number of consecutive packets was found.
|
|
*/
|
|
gboolean
|
|
rtp_jitter_buffer_can_fast_start (RTPJitterBuffer * jbuf, gint num_packet)
|
|
{
|
|
gboolean ret = TRUE;
|
|
RTPJitterBufferItem *last_item = NULL, *item;
|
|
gint i;
|
|
|
|
if (rtp_jitter_buffer_num_packets (jbuf) < num_packet)
|
|
return FALSE;
|
|
|
|
item = rtp_jitter_buffer_peek (jbuf);
|
|
for (i = 0; i < num_packet; i++) {
|
|
if (G_LIKELY (last_item)) {
|
|
guint16 expected_seqnum = last_item->seqnum + 1;
|
|
|
|
if (expected_seqnum != item->seqnum) {
|
|
ret = FALSE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
last_item = item;
|
|
item = (RTPJitterBufferItem *) last_item->next;
|
|
}
|
|
|
|
return ret;
|
|
}
|