mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-29 11:40:38 +00:00
c0aa28ca5b
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Link to the right pads regardless of which one was created first in the ssrc demuxer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsource.c: (calculate_jitter): Improve debugging. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links): * gst/rtpmanager/gstrtpssrcdemux.h: Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
928 lines
26 KiB
C
928 lines
26 KiB
C
/* 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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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include "rtpsource.h"
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GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
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#define GST_CAT_DEFAULT rtp_source_debug
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#define RTP_MAX_PROBATION_LEN 32
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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_source_finalize (GObject * object);
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/* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
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G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
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static void
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rtp_source_class_init (RTPSourceClass * 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_source_finalize;
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GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
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}
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static void
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rtp_source_init (RTPSource * src)
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{
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/* sources are initialy on probation until we receive enough valid RTP
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* packets or a valid RTCP packet */
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src->validated = FALSE;
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src->probation = RTP_DEFAULT_PROBATION;
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src->payload = 0;
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src->clock_rate = -1;
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src->clock_base = -1;
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src->packets = g_queue_new ();
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src->seqnum_base = -1;
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src->last_rtptime = -1;
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src->stats.cycles = -1;
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src->stats.jitter = 0;
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src->stats.transit = -1;
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src->stats.curr_sr = 0;
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src->stats.curr_rr = 0;
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}
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static void
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rtp_source_finalize (GObject * object)
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{
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RTPSource *src;
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GstBuffer *buffer;
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src = RTP_SOURCE_CAST (object);
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while ((buffer = g_queue_pop_head (src->packets)))
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gst_buffer_unref (buffer);
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g_queue_free (src->packets);
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G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
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}
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/**
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* rtp_source_new:
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* @ssrc: an SSRC
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*
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* Create a #RTPSource with @ssrc.
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*
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* Returns: a new #RTPSource. Use g_object_unref() after usage.
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*/
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RTPSource *
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rtp_source_new (guint32 ssrc)
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{
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RTPSource *src;
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src = g_object_new (RTP_TYPE_SOURCE, NULL);
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src->ssrc = ssrc;
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return src;
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}
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/**
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* rtp_source_update_caps:
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* @src: an #RTPSource
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* @caps: a #GstCaps
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*
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* Parse @caps and store all relevant information in @source.
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*/
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void
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rtp_source_update_caps (RTPSource * src, GstCaps * caps)
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{
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GstStructure *s;
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guint val;
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gint ival;
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/* nothing changed, return */
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if (src->caps == caps)
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return;
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s = gst_caps_get_structure (caps, 0);
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if (gst_structure_get_int (s, "payload", &ival))
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src->payload = ival;
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GST_DEBUG ("got payload %d", src->payload);
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gst_structure_get_int (s, "clock-rate", &src->clock_rate);
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GST_DEBUG ("got clock-rate %d", src->clock_rate);
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if (gst_structure_get_uint (s, "clock-base", &val))
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src->clock_base = val;
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GST_DEBUG ("got clock-base %" G_GINT64_FORMAT, src->clock_base);
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if (gst_structure_get_uint (s, "seqnum-base", &val))
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src->seqnum_base = val;
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GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
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gst_caps_replace (&src->caps, caps);
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}
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/**
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* rtp_source_set_callbacks:
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* @src: an #RTPSource
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* @cb: callback functions
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* @user_data: user data
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*
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* Set the callbacks for the source.
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*/
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void
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rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
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gpointer user_data)
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{
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g_return_if_fail (RTP_IS_SOURCE (src));
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src->callbacks.push_rtp = cb->push_rtp;
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src->callbacks.clock_rate = cb->clock_rate;
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src->user_data = user_data;
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}
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/**
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* rtp_source_set_as_csrc:
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* @src: an #RTPSource
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*
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* Configure @src as a CSRC, this will validate the RTpSource.
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*/
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void
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rtp_source_set_as_csrc (RTPSource * src)
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{
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g_return_if_fail (RTP_IS_SOURCE (src));
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src->validated = TRUE;
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src->is_csrc = TRUE;
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}
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/**
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* rtp_source_set_rtp_from:
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* @src: an #RTPSource
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* @address: the RTP address to set
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*
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* Set that @src is receiving RTP packets from @address. This is used for
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* collistion checking.
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*/
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void
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rtp_source_set_rtp_from (RTPSource * src, GstNetAddress * address)
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{
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g_return_if_fail (RTP_IS_SOURCE (src));
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src->have_rtp_from = TRUE;
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memcpy (&src->rtp_from, address, sizeof (GstNetAddress));
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}
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/**
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* rtp_source_set_rtcp_from:
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* @src: an #RTPSource
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* @address: the RTCP address to set
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*
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* Set that @src is receiving RTCP packets from @address. This is used for
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* collistion checking.
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*/
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void
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rtp_source_set_rtcp_from (RTPSource * src, GstNetAddress * address)
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{
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g_return_if_fail (RTP_IS_SOURCE (src));
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src->have_rtcp_from = TRUE;
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memcpy (&src->rtcp_from, address, sizeof (GstNetAddress));
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}
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static GstFlowReturn
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push_packet (RTPSource * src, GstBuffer * buffer)
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{
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GstFlowReturn ret = GST_FLOW_OK;
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/* push queued packets first if any */
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while (!g_queue_is_empty (src->packets)) {
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GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
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GST_DEBUG ("pushing queued packet");
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if (src->callbacks.push_rtp)
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src->callbacks.push_rtp (src, buffer, src->user_data);
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else
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gst_buffer_unref (buffer);
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}
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GST_DEBUG ("pushing new packet");
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/* push packet */
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if (src->callbacks.push_rtp)
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ret = src->callbacks.push_rtp (src, buffer, src->user_data);
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else
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gst_buffer_unref (buffer);
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return ret;
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}
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static gint
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get_clock_rate (RTPSource * src, guint8 payload)
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{
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if (src->clock_rate == -1) {
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gint clock_rate = -1;
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if (src->callbacks.clock_rate)
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clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
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GST_DEBUG ("new payload %d, got clock-rate %d", payload, clock_rate);
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src->clock_rate = clock_rate;
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}
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src->payload = payload;
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return src->clock_rate;
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}
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/* Jitter is the variation in the delay of received packets in a flow. It is
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* measured by comparing the interval when RTP packets were sent to the interval
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* at which they were received. For instance, if packet #1 and packet #2 leave
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* 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
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* milliseconds. */
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static void
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calculate_jitter (RTPSource * src, GstBuffer * buffer,
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RTPArrivalStats * arrival)
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{
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guint64 ntpnstime;
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guint32 rtparrival, transit, rtptime;
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gint32 diff;
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gint clock_rate;
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guint8 pt;
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/* get arrival time */
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if ((ntpnstime = arrival->ntpnstime) == GST_CLOCK_TIME_NONE)
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goto no_time;
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pt = gst_rtp_buffer_get_payload_type (buffer);
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GST_DEBUG ("SSRC %08x got payload %d", src->ssrc, pt);
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/* get clockrate */
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if ((clock_rate = get_clock_rate (src, pt)) == -1)
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goto no_clock_rate;
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rtptime = gst_rtp_buffer_get_timestamp (buffer);
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/* no clock-base, take first rtptime as base */
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if (src->clock_base == -1) {
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GST_DEBUG ("using clock-base of %" G_GUINT32_FORMAT, rtptime);
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src->clock_base = rtptime;
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}
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/* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
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* care about the absolute value, just the difference. */
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rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND);
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/* transit time is difference with RTP timestamp */
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transit = rtparrival - rtptime;
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/* get ABS diff with previous transit time */
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if (src->stats.transit != -1) {
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if (transit > src->stats.transit)
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diff = transit - src->stats.transit;
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else
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diff = src->stats.transit - transit;
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} else
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diff = 0;
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src->stats.transit = transit;
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/* update jitter, the value we store is scaled up so we can keep precision. */
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src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
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src->stats.prev_rtptime = src->stats.last_rtptime;
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src->stats.last_rtptime = rtparrival;
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GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
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rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
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return;
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/* ERRORS */
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no_time:
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{
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GST_WARNING ("cannot get current time");
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return;
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}
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no_clock_rate:
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{
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GST_WARNING ("cannot get clock-rate for pt %d", pt);
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return;
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}
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}
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static void
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init_seq (RTPSource * src, guint16 seq)
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{
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src->stats.base_seq = seq;
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src->stats.max_seq = seq;
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src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
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src->stats.cycles = 0;
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src->stats.packets_received = 0;
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src->stats.octets_received = 0;
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src->stats.bytes_received = 0;
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src->stats.prev_received = 0;
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src->stats.prev_expected = 0;
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GST_DEBUG ("base_seq %d", seq);
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}
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/**
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* rtp_source_process_rtp:
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* @src: an #RTPSource
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* @buffer: an RTP buffer
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*
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* Let @src handle the incomming RTP @buffer.
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*
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* Returns: a #GstFlowReturn.
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*/
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GstFlowReturn
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rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
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RTPArrivalStats * arrival)
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{
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GstFlowReturn result = GST_FLOW_OK;
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guint16 seqnr, udelta;
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RTPSourceStats *stats;
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g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
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g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
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stats = &src->stats;
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seqnr = gst_rtp_buffer_get_seq (buffer);
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rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
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if (stats->cycles == -1) {
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GST_DEBUG ("received first buffer");
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/* first time we heard of this source */
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init_seq (src, seqnr);
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src->stats.max_seq = seqnr - 1;
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src->probation = RTP_DEFAULT_PROBATION;
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}
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udelta = seqnr - stats->max_seq;
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/* if we are still on probation, check seqnum */
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if (src->probation) {
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guint16 expected;
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expected = src->stats.max_seq + 1;
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/* when in probation, we require consecutive seqnums */
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if (seqnr == expected) {
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/* expected packet */
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GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
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src->probation--;
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src->stats.max_seq = seqnr;
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if (src->probation == 0) {
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GST_DEBUG ("probation done!");
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init_seq (src, seqnr);
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} else {
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GstBuffer *q;
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GST_DEBUG ("probation %d: queue buffer", src->probation);
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/* when still in probation, keep packets in a list. */
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g_queue_push_tail (src->packets, buffer);
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/* remove packets from queue if there are too many */
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while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
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q = g_queue_pop_head (src->packets);
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gst_object_unref (q);
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}
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goto done;
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}
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} else {
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GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
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src->probation = RTP_DEFAULT_PROBATION;
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src->stats.max_seq = seqnr;
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goto done;
|
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}
|
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} else if (udelta < RTP_MAX_DROPOUT) {
|
|
/* in order, with permissible gap */
|
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if (seqnr < stats->max_seq) {
|
|
/* sequence number wrapped - count another 64K cycle. */
|
|
stats->cycles += RTP_SEQ_MOD;
|
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}
|
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stats->max_seq = seqnr;
|
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} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
|
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/* the sequence number made a very large jump */
|
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if (seqnr == stats->bad_seq) {
|
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/* two sequential packets -- assume that the other side
|
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* restarted without telling us so just re-sync
|
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* (i.e., pretend this was the first packet). */
|
|
init_seq (src, seqnr);
|
|
} else {
|
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/* unacceptable jump */
|
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stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
|
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goto bad_sequence;
|
|
}
|
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} else {
|
|
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
|
|
GST_WARNING ("duplicate or reordered packet");
|
|
}
|
|
|
|
src->stats.octets_received += arrival->payload_len;
|
|
src->stats.bytes_received += arrival->bytes;
|
|
src->stats.packets_received++;
|
|
/* the source that sent the packet must be a sender */
|
|
src->is_sender = TRUE;
|
|
src->validated = TRUE;
|
|
|
|
GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
|
|
seqnr, src->stats.packets_received, src->stats.octets_received);
|
|
|
|
/* calculate jitter and perform skew correction */
|
|
calculate_jitter (src, buffer, arrival);
|
|
|
|
/* we're ready to push the RTP packet now */
|
|
result = push_packet (src, buffer);
|
|
|
|
done:
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
bad_sequence:
|
|
{
|
|
GST_WARNING ("unacceptable seqnum received");
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* rtp_source_process_bye:
|
|
* @src: an #RTPSource
|
|
* @reason: the reason for leaving
|
|
*
|
|
* Notify @src that a BYE packet has been received. This will make the source
|
|
* inactive.
|
|
*/
|
|
void
|
|
rtp_source_process_bye (RTPSource * src, const gchar * reason)
|
|
{
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
|
|
GST_STR_NULL (reason));
|
|
|
|
/* copy the reason and mark as received_bye */
|
|
g_free (src->bye_reason);
|
|
src->bye_reason = g_strdup (reason);
|
|
src->received_bye = TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_send_rtp:
|
|
* @src: an #RTPSource
|
|
* @buffer: an RTP buffer
|
|
* @ntpnstime: the NTP time when this buffer was captured in nanoseconds
|
|
*
|
|
* Send an RTP @buffer originating from @src. This will make @src a sender.
|
|
* This function takes ownership of @buffer and modifies the SSRC in the RTP
|
|
* packet to that of @src when needed.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
|
|
{
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
guint len;
|
|
guint32 rtptime;
|
|
guint64 ext_rtptime;
|
|
guint64 ntp_diff, rtp_diff;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
|
|
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
|
|
|
len = gst_rtp_buffer_get_payload_len (buffer);
|
|
|
|
rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
|
|
|
|
/* we are a sender now */
|
|
src->is_sender = TRUE;
|
|
|
|
/* update stats for the SR */
|
|
src->stats.packets_sent++;
|
|
src->stats.octets_sent += len;
|
|
|
|
rtptime = gst_rtp_buffer_get_timestamp (buffer);
|
|
ext_rtptime = src->last_rtptime;
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
|
|
|
|
GST_DEBUG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
|
|
src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime));
|
|
|
|
if (ext_rtptime > src->last_rtptime) {
|
|
rtp_diff = ext_rtptime - src->last_rtptime;
|
|
ntp_diff = ntpnstime - src->last_ntpnstime;
|
|
|
|
/* calc the diff so we can detect drift at the sender. This can also be used
|
|
* to guestimate the clock rate if the NTP time is locked to the RTP
|
|
* timestamps (as is the case when the capture device is providing the clock). */
|
|
GST_DEBUG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
|
|
GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff));
|
|
}
|
|
|
|
/* we keep track of the last received RTP timestamp and the corresponding
|
|
* NTP timestamp so that we can use this info when constructing SR reports */
|
|
src->last_rtptime = ext_rtptime;
|
|
src->last_ntpnstime = ntpnstime;
|
|
|
|
/* push packet */
|
|
if (src->callbacks.push_rtp) {
|
|
guint32 ssrc;
|
|
|
|
ssrc = gst_rtp_buffer_get_ssrc (buffer);
|
|
if (ssrc != src->ssrc) {
|
|
/* the SSRC of the packet is not correct, make a writable buffer and
|
|
* update the SSRC. This could involve a complete copy of the packet when
|
|
* it is not writable. Usually the payloader will use caps negotiation to
|
|
* get the correct SSRC from the session manager before pushing anything. */
|
|
buffer = gst_buffer_make_writable (buffer);
|
|
|
|
GST_WARNING ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
|
|
src->ssrc);
|
|
gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
|
|
}
|
|
GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
|
|
src->stats.packets_sent);
|
|
result = src->callbacks.push_rtp (src, buffer, src->user_data);
|
|
} else {
|
|
GST_WARNING ("no callback installed, dropping packet");
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_process_sr:
|
|
* @src: an #RTPSource
|
|
* @time: time of packet arrival
|
|
* @ntptime: the NTP time
|
|
* @rtptime: the RTP time
|
|
* @packet_count: the packet count
|
|
* @octet_count: the octect count
|
|
*
|
|
* Update the sender report in @src.
|
|
*/
|
|
void
|
|
rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
|
|
guint32 rtptime, guint32 packet_count, guint32 octet_count)
|
|
{
|
|
RTPSenderReport *curr;
|
|
gint curridx;
|
|
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
|
|
", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
|
|
(guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
|
|
packet_count, octet_count);
|
|
|
|
curridx = src->stats.curr_sr ^ 1;
|
|
curr = &src->stats.sr[curridx];
|
|
|
|
/* this is a sender now */
|
|
src->is_sender = TRUE;
|
|
|
|
/* update current */
|
|
curr->is_valid = TRUE;
|
|
curr->ntptime = ntptime;
|
|
curr->rtptime = rtptime;
|
|
curr->packet_count = packet_count;
|
|
curr->octet_count = octet_count;
|
|
curr->time = time;
|
|
|
|
/* make current */
|
|
src->stats.curr_sr = curridx;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_process_rb:
|
|
* @src: an #RTPSource
|
|
* @time: the current time in nanoseconds since 1970
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
*
|
|
* Update the report block in @src.
|
|
*/
|
|
void
|
|
rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost,
|
|
gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr,
|
|
guint32 dlsr)
|
|
{
|
|
RTPReceiverReport *curr;
|
|
gint curridx;
|
|
guint32 ntp, A;
|
|
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
|
|
", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
|
|
src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
|
|
lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
|
|
|
|
curridx = src->stats.curr_rr ^ 1;
|
|
curr = &src->stats.rr[curridx];
|
|
|
|
/* update current */
|
|
curr->is_valid = TRUE;
|
|
curr->fractionlost = fractionlost;
|
|
curr->packetslost = packetslost;
|
|
curr->exthighestseq = exthighestseq;
|
|
curr->jitter = jitter;
|
|
curr->lsr = lsr;
|
|
curr->dlsr = dlsr;
|
|
|
|
/* calculate round trip */
|
|
ntp = (gst_rtcp_unix_to_ntp (time) >> 16) & 0xffffffff;
|
|
A = ntp - dlsr;
|
|
A -= lsr;
|
|
curr->round_trip = A;
|
|
|
|
GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
|
|
A >> 16, A & 0xffff);
|
|
|
|
/* make current */
|
|
src->stats.curr_rr = curridx;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_new_sr:
|
|
* @src: an #RTPSource
|
|
* @time: the current time in nanoseconds since 1970
|
|
* @ntptime: the NTP time
|
|
* @rtptime: the RTP time
|
|
* @packet_count: the packet count
|
|
* @octet_count: the octect count
|
|
*
|
|
* Get new values to put into a new SR report from this source.
|
|
*
|
|
* Returns: %TRUE on success.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_new_sr (RTPSource * src, GstClockTime ntpnstime,
|
|
guint64 * ntptime, guint32 * rtptime, guint32 * packet_count,
|
|
guint32 * octet_count)
|
|
{
|
|
guint64 t_rtp;
|
|
guint64 t_current_ntp;
|
|
GstClockTimeDiff diff;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
/* use the sync params to interpollate the date->time member to rtptime. We
|
|
* use the last sent timestamp and rtptime as reference points. We assume
|
|
* that the slope of the rtptime vs timestamp curve is 1, which is certainly
|
|
* sufficient for the frequency at which we report SR and the rate we send
|
|
* out RTP packets. */
|
|
t_rtp = src->last_rtptime;
|
|
|
|
GST_DEBUG ("last_ntpnstime %" GST_TIME_FORMAT ", last_rtptime %"
|
|
G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_ntpnstime), t_rtp);
|
|
|
|
if (src->clock_rate != -1) {
|
|
/* get the diff with the SR time */
|
|
diff = GST_CLOCK_DIFF (src->last_ntpnstime, ntpnstime);
|
|
|
|
/* now translate the diff to RTP time, handle positive and negative cases.
|
|
* If there is no diff, we already set rtptime correctly above. */
|
|
if (diff > 0) {
|
|
GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
|
|
t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
|
|
} else {
|
|
diff = -diff;
|
|
GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
|
|
t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
|
|
}
|
|
} else {
|
|
GST_WARNING ("no clock-rate, cannot interpollate rtp time");
|
|
}
|
|
|
|
/* convert the NTP time in nanoseconds to 32.32 fixed point */
|
|
t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
|
|
|
|
GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
|
|
(guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
|
|
(guint32) t_rtp);
|
|
|
|
if (ntptime)
|
|
*ntptime = t_current_ntp;
|
|
if (rtptime)
|
|
*rtptime = t_rtp;
|
|
if (packet_count)
|
|
*packet_count = src->stats.packets_sent;
|
|
if (octet_count)
|
|
*octet_count = src->stats.octets_sent;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_new_rb:
|
|
* @src: an #RTPSource
|
|
* @time: the current time in nanoseconds since 1970
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
*
|
|
* Get the values of the last RB report set with rtp_source_process_rb().
|
|
*
|
|
* Returns: %TRUE on success.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
|
|
guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
|
|
guint32 * jitter, guint32 * lsr, guint32 * dlsr)
|
|
{
|
|
RTPSourceStats *stats;
|
|
guint64 extended_max, expected;
|
|
guint64 expected_interval, received_interval, ntptime;
|
|
gint64 lost, lost_interval;
|
|
guint32 fraction, LSR, DLSR;
|
|
GstClockTime sr_time;
|
|
|
|
stats = &src->stats;
|
|
|
|
extended_max = stats->cycles + stats->max_seq;
|
|
expected = extended_max - stats->base_seq + 1;
|
|
|
|
GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
|
|
", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
|
|
extended_max, expected, stats->packets_received, stats->base_seq);
|
|
|
|
lost = expected - stats->packets_received;
|
|
lost = CLAMP (lost, -0x800000, 0x7fffff);
|
|
|
|
expected_interval = expected - stats->prev_expected;
|
|
stats->prev_expected = expected;
|
|
received_interval = stats->packets_received - stats->prev_received;
|
|
stats->prev_received = stats->packets_received;
|
|
|
|
lost_interval = expected_interval - received_interval;
|
|
|
|
if (expected_interval == 0 || lost_interval <= 0)
|
|
fraction = 0;
|
|
else
|
|
fraction = (lost_interval << 8) / expected_interval;
|
|
|
|
GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
|
|
/* we scaled the jitter up for additional precision */
|
|
GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
|
|
", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
|
|
extended_max, stats->jitter >> 4);
|
|
|
|
if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
|
|
GstClockTime diff;
|
|
|
|
/* LSR is middle 32 bits of the last ntptime */
|
|
LSR = (ntptime >> 16) & 0xffffffff;
|
|
diff = time - sr_time;
|
|
GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
|
|
/* DLSR, delay since last SR is expressed in 1/65536 second units */
|
|
DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
|
|
} else {
|
|
/* No valid SR received, LSR/DLSR are set to 0 then */
|
|
GST_DEBUG ("no valid SR received");
|
|
LSR = 0;
|
|
DLSR = 0;
|
|
}
|
|
GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
|
|
DLSR >> 16, DLSR & 0xffff);
|
|
|
|
if (fractionlost)
|
|
*fractionlost = fraction;
|
|
if (packetslost)
|
|
*packetslost = lost;
|
|
if (exthighestseq)
|
|
*exthighestseq = extended_max;
|
|
if (jitter)
|
|
*jitter = stats->jitter >> 4;
|
|
if (lsr)
|
|
*lsr = LSR;
|
|
if (dlsr)
|
|
*dlsr = DLSR;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_last_sr:
|
|
* @src: an #RTPSource
|
|
* @time: time of packet arrival
|
|
* @ntptime: the NTP time
|
|
* @rtptime: the RTP time
|
|
* @packet_count: the packet count
|
|
* @octet_count: the octect count
|
|
*
|
|
* Get the values of the last sender report as set with rtp_source_process_sr().
|
|
*
|
|
* Returns: %TRUE if there was a valid SR report.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
|
|
guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
|
|
{
|
|
RTPSenderReport *curr;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
curr = &src->stats.sr[src->stats.curr_sr];
|
|
if (!curr->is_valid)
|
|
return FALSE;
|
|
|
|
if (ntptime)
|
|
*ntptime = curr->ntptime;
|
|
if (rtptime)
|
|
*rtptime = curr->rtptime;
|
|
if (packet_count)
|
|
*packet_count = curr->packet_count;
|
|
if (octet_count)
|
|
*octet_count = curr->octet_count;
|
|
if (time)
|
|
*time = curr->time;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_last_rb:
|
|
* @src: an #RTPSource
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
*
|
|
* Get the values of the last RB report set with rtp_source_process_rb().
|
|
*
|
|
* Returns: %TRUE if there was a valid SB report.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
|
|
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
|
|
guint32 * lsr, guint32 * dlsr)
|
|
{
|
|
RTPReceiverReport *curr;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
curr = &src->stats.rr[src->stats.curr_rr];
|
|
if (!curr->is_valid)
|
|
return FALSE;
|
|
|
|
if (fractionlost)
|
|
*fractionlost = curr->fractionlost;
|
|
if (packetslost)
|
|
*packetslost = curr->packetslost;
|
|
if (exthighestseq)
|
|
*exthighestseq = curr->exthighestseq;
|
|
if (jitter)
|
|
*jitter = curr->jitter;
|
|
if (lsr)
|
|
*lsr = curr->lsr;
|
|
if (dlsr)
|
|
*dlsr = curr->dlsr;
|
|
|
|
return TRUE;
|
|
}
|