mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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bba6f097b1
It is compared to other extended RTP timestamps all over rtpjitterbuffer
and since 4df3da3bab the initial extended RTP timestamp is not equal
anymore to the plain RTP time.
Continue passing a non-extended RTP timestamp via the `sync` signal for
backwards compatibility. It will always be a timestamp inside the first
extended timestamp period anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6575>
5520 lines
180 KiB
C
5520 lines
180 KiB
C
/*
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* Farsight Voice+Video library
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*
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* Copyright 2007 Collabora Ltd,
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* Copyright 2007 Nokia Corporation
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* @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
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* Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
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* Copyright 2015 Kurento (http://kurento.org/)
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* @author: Miguel París <mparisdiaz@gmail.com>
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* Copyright 2016 Pexip AS
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* @author: Havard Graff <havard@pexip.com>
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* @author: Stian Selnes <stian@pexip.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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*/
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/**
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* SECTION:element-rtpjitterbuffer
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* @title: rtpjitterbuffer
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*
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* This element reorders and removes duplicate RTP packets as they are received
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* from a network source.
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*
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* The element needs the clock-rate of the RTP payload in order to estimate the
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* delay. This information is obtained either from the caps on the sink pad or,
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* when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
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* To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
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*
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* The rtpjitterbuffer will wait for missing packets up to a configurable time
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* limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
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* late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
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* property is set, lost packets will result in a custom serialized downstream
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* event of name GstRTPPacketLost. The lost packet events are usually used by a
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* depayloader or other element to create concealment data or some other logic
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* to gracefully handle the missing packets.
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*
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* The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incoming
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* buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
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* buffer.
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*
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* The jitterbuffer can also be configured to send early retransmission events
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* upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
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* this mode, the jitterbuffer tries to estimate when a packet should arrive and
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* sends a custom upstream event named GstRTPRetransmissionRequest when the
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* packet is considered late. The initial expected packet arrival time is
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* calculated as follows:
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*
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* - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
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* T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
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* calculated from the DTS (or PTS is no DTS) of two consecutive RTP
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* packets with different rtptime.
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*
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* - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
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* seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
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* previously scheduled timeout is overwritten.
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*
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* - If seqnum N arrived, all seqnum older than
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* N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
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* immediately. This is to request fast feedback for abnormally reorder
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* packets before any of the previous timeouts is triggered.
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*
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* A late packet triggers the GstRTPRetransmissionRequest custom upstream
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* event. After the initial timeout expires and the retransmission event is
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* sent, the timeout is scheduled for
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* T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
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* arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
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* GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
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* again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
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* #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
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* retransmission requests are sent and the regular logic is performed to
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* schedule a lost packet as discussed above.
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*
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* This element acts as a live element and so adds #GstRtpJitterBuffer:latency
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* to the pipeline.
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*
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* This element will automatically be used inside rtpbin.
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
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* ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
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* inserted into the pipeline to smooth out network jitter and to reorder the
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* out-of-order RTP packets.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <stdio.h>
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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 <gst/net/net.h>
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#include "gstrtpjitterbuffer.h"
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#include "rtpjitterbuffer.h"
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#include "rtpstats.h"
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#include "rtptimerqueue.h"
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#include "gstrtputils.h"
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#include <gst/glib-compat-private.h>
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GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
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#define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
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/* RTPJitterBuffer signals and args */
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enum
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{
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SIGNAL_REQUEST_PT_MAP,
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SIGNAL_CLEAR_PT_MAP,
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SIGNAL_HANDLE_SYNC,
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SIGNAL_ON_NPT_STOP,
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SIGNAL_SET_ACTIVE,
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LAST_SIGNAL
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};
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#define DEFAULT_LATENCY_MS 200
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#define DEFAULT_DROP_ON_LATENCY FALSE
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#define DEFAULT_TS_OFFSET 0
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#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT 0
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#define DEFAULT_DO_LOST FALSE
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#define DEFAULT_POST_DROP_MESSAGES FALSE
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#define DEFAULT_DROP_MESSAGES_INTERVAL_MS 200
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#define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
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#define DEFAULT_PERCENT 0
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#define DEFAULT_DO_RETRANSMISSION FALSE
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#define DEFAULT_RTX_NEXT_SEQNUM TRUE
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#define DEFAULT_RTX_DELAY -1
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#define DEFAULT_RTX_MIN_DELAY 0
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#define DEFAULT_RTX_DELAY_REORDER 3
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#define DEFAULT_RTX_RETRY_TIMEOUT -1
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#define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
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#define DEFAULT_RTX_RETRY_PERIOD -1
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#define DEFAULT_RTX_MAX_RETRIES -1
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#define DEFAULT_RTX_DEADLINE -1
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#define DEFAULT_RTX_STATS_TIMEOUT 1000
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#define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
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#define DEFAULT_MAX_DROPOUT_TIME 60000
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#define DEFAULT_MAX_MISORDER_TIME 2000
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#define DEFAULT_RFC7273_SYNC FALSE
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#define DEFAULT_ADD_REFERENCE_TIMESTAMP_META FALSE
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#define DEFAULT_FASTSTART_MIN_PACKETS 0
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#define DEFAULT_SYNC_INTERVAL 0
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#define DEFAULT_RFC7273_USE_SYSTEM_CLOCK FALSE
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#define DEFAULT_RFC7273_REFERENCE_TIMESTAMP_META_ONLY FALSE
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#define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
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#define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
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enum
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{
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PROP_0,
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PROP_LATENCY,
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PROP_DROP_ON_LATENCY,
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PROP_TS_OFFSET,
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PROP_MAX_TS_OFFSET_ADJUSTMENT,
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PROP_DO_LOST,
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PROP_POST_DROP_MESSAGES,
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PROP_DROP_MESSAGES_INTERVAL,
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PROP_MODE,
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PROP_PERCENT,
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PROP_DO_RETRANSMISSION,
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PROP_RTX_NEXT_SEQNUM,
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PROP_RTX_DELAY,
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PROP_RTX_MIN_DELAY,
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PROP_RTX_DELAY_REORDER,
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PROP_RTX_RETRY_TIMEOUT,
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PROP_RTX_MIN_RETRY_TIMEOUT,
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PROP_RTX_RETRY_PERIOD,
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PROP_RTX_MAX_RETRIES,
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PROP_RTX_DEADLINE,
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PROP_RTX_STATS_TIMEOUT,
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PROP_STATS,
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PROP_MAX_RTCP_RTP_TIME_DIFF,
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PROP_MAX_DROPOUT_TIME,
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PROP_MAX_MISORDER_TIME,
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PROP_RFC7273_SYNC,
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PROP_ADD_REFERENCE_TIMESTAMP_META,
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PROP_FASTSTART_MIN_PACKETS,
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PROP_SYNC_INTERVAL,
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PROP_RFC7273_USE_SYSTEM_CLOCK,
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PROP_RFC7273_REFERENCE_TIMESTAMP_META_ONLY,
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};
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#define JBUF_LOCK(priv) G_STMT_START { \
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GST_TRACE("Locking from thread %p", g_thread_self()); \
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(g_mutex_lock (&(priv)->jbuf_lock)); \
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GST_TRACE("Locked from thread %p", g_thread_self()); \
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} G_STMT_END
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#define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
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JBUF_LOCK (priv); \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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} G_STMT_END
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#define JBUF_UNLOCK(priv) G_STMT_START { \
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GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
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(g_mutex_unlock (&(priv)->jbuf_lock)); \
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} G_STMT_END
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#define JBUF_WAIT_QUEUE(priv) G_STMT_START { \
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GST_DEBUG ("waiting queue"); \
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(priv)->waiting_queue++; \
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g_cond_wait (&(priv)->jbuf_queue, &(priv)->jbuf_lock); \
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(priv)->waiting_queue--; \
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GST_DEBUG ("waiting queue done"); \
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} G_STMT_END
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#define JBUF_SIGNAL_QUEUE(priv) G_STMT_START { \
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if (G_UNLIKELY ((priv)->waiting_queue)) { \
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GST_DEBUG ("signal queue, %d waiters", (priv)->waiting_queue); \
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g_cond_signal (&(priv)->jbuf_queue); \
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} \
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} G_STMT_END
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#define JBUF_WAIT_TIMER(priv) G_STMT_START { \
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GST_DEBUG ("waiting timer"); \
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(priv)->waiting_timer++; \
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g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
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(priv)->waiting_timer--; \
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GST_DEBUG ("waiting timer done"); \
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} G_STMT_END
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#define JBUF_WAIT_TIMER_CHECK(priv, label) G_STMT_START { \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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JBUF_WAIT_TIMER (priv); \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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} G_STMT_END
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#define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
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if (G_UNLIKELY ((priv)->waiting_timer)) { \
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GST_DEBUG ("signal timer, %d waiters", (priv)->waiting_timer); \
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g_cond_signal (&(priv)->jbuf_timer); \
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} \
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} G_STMT_END
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#define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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GST_DEBUG ("waiting event"); \
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(priv)->waiting_event = TRUE; \
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g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
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(priv)->waiting_event = FALSE; \
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GST_DEBUG ("waiting event done"); \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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} G_STMT_END
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#define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
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if (G_UNLIKELY ((priv)->waiting_event)) { \
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GST_DEBUG ("signal event"); \
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g_cond_signal (&(priv)->jbuf_event); \
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} \
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} G_STMT_END
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#define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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GST_DEBUG ("waiting query"); \
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(priv)->waiting_query = TRUE; \
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g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
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(priv)->waiting_query = FALSE; \
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GST_DEBUG ("waiting query done"); \
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if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
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goto label; \
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} G_STMT_END
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#define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
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(priv)->last_query = res; \
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if (G_UNLIKELY ((priv)->waiting_query)) { \
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GST_DEBUG ("signal query"); \
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g_cond_signal (&(priv)->jbuf_query); \
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} \
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} G_STMT_END
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#define GST_BUFFER_IS_RETRANSMISSION(buffer) \
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GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
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struct _GstRtpJitterBufferPrivate
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{
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GstPad *sinkpad, *srcpad;
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GstPad *rtcpsinkpad;
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RTPJitterBuffer *jbuf;
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GMutex jbuf_lock;
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guint waiting_queue;
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GCond jbuf_queue;
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guint waiting_timer;
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GCond jbuf_timer;
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gboolean waiting_event;
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GCond jbuf_event;
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gboolean waiting_query;
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GCond jbuf_query;
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gboolean last_query;
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gboolean discont;
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gboolean ts_discont;
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gboolean active;
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guint64 out_offset;
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guint32 segment_seqnum;
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gboolean timer_running;
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GThread *timer_thread;
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/* properties */
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guint latency_ms;
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guint64 latency_ns;
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gboolean drop_on_latency;
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gint64 ts_offset;
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guint64 max_ts_offset_adjustment;
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gboolean do_lost;
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gboolean post_drop_messages;
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guint drop_messages_interval_ms;
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gboolean do_retransmission;
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gboolean rtx_next_seqnum;
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gint rtx_delay;
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guint rtx_min_delay;
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gint rtx_delay_reorder;
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gint rtx_retry_timeout;
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gint rtx_min_retry_timeout;
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gint rtx_retry_period;
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gint rtx_max_retries;
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guint rtx_stats_timeout;
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gint rtx_deadline_ms;
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|
gint max_rtcp_rtp_time_diff;
|
|
guint32 max_dropout_time;
|
|
guint32 max_misorder_time;
|
|
guint faststart_min_packets;
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|
gboolean add_reference_timestamp_meta;
|
|
guint sync_interval;
|
|
gboolean rfc7273_use_system_clock;
|
|
gboolean rfc7273_reference_timestamp_meta_only;
|
|
|
|
/* Reference for GstReferenceTimestampMeta */
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|
GstCaps *reference_timestamp_caps;
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|
|
|
/* RTP header extension ID for RFC6051 64-bit NTP timestamps */
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|
guint8 ntp64_ext_id;
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|
|
|
/* Known CNAME / SSRC mappings */
|
|
GList *cname_ssrc_mappings;
|
|
|
|
/* the last seqnum we pushed out */
|
|
guint32 last_popped_seqnum;
|
|
/* the next expected seqnum we push */
|
|
guint32 next_seqnum;
|
|
/* seqnum-base, if known */
|
|
guint32 seqnum_base;
|
|
/* last output time */
|
|
GstClockTime last_out_time;
|
|
/* last valid input timestamp and rtptime pair */
|
|
GstClockTime ips_pts;
|
|
guint64 ips_rtptime;
|
|
GstClockTime packet_spacing;
|
|
gint equidistant;
|
|
|
|
GQueue gap_packets;
|
|
|
|
/* the next expected seqnum we receive */
|
|
GstClockTime last_in_pts;
|
|
guint32 next_in_seqnum;
|
|
|
|
/* "normal" timers */
|
|
RtpTimerQueue *timers;
|
|
/* timers used for RTX statistics backlog */
|
|
RtpTimerQueue *rtx_stats_timers;
|
|
|
|
/* start and stop ranges */
|
|
GstClockTime npt_start;
|
|
GstClockTime npt_stop;
|
|
guint64 ext_timestamp;
|
|
guint64 last_elapsed;
|
|
guint64 estimated_eos;
|
|
GstClockID eos_id;
|
|
|
|
/* state */
|
|
gboolean eos;
|
|
guint last_percent;
|
|
|
|
/* clock rate and rtp timestamp offset */
|
|
gint last_pt;
|
|
guint32 last_ssrc;
|
|
gint32 clock_rate;
|
|
gint64 clock_base;
|
|
gint64 ts_offset_remainder;
|
|
|
|
/* when we are shutting down */
|
|
GstFlowReturn srcresult;
|
|
gboolean blocked;
|
|
|
|
/* for sync */
|
|
GstSegment segment;
|
|
GstClockID clock_id;
|
|
GstClockTime timer_timeout;
|
|
guint16 timer_seqnum;
|
|
/* the latency of the upstream peer, we have to take this into account when
|
|
* synchronizing the buffers. */
|
|
GstClockTime peer_latency;
|
|
guint64 last_sr_ext_rtptime;
|
|
GstBuffer *last_sr;
|
|
guint32 last_sr_ssrc;
|
|
GstClockTime last_sr_ntpnstime;
|
|
|
|
GstClockTime last_known_ntpnstime;
|
|
guint64 last_known_ext_rtptime;
|
|
|
|
/* some accounting */
|
|
guint64 num_pushed;
|
|
guint64 num_lost;
|
|
guint64 num_late;
|
|
guint64 num_duplicates;
|
|
guint64 num_rtx_requests;
|
|
guint64 num_rtx_success;
|
|
guint64 num_rtx_failed;
|
|
gdouble avg_rtx_num;
|
|
guint64 avg_rtx_rtt;
|
|
RTPPacketRateCtx packet_rate_ctx;
|
|
|
|
/* for the jitter */
|
|
GstClockTime last_dts;
|
|
GstClockTime last_pts;
|
|
guint64 last_rtptime;
|
|
GstClockTime last_ntpnstime;
|
|
GstClockTime avg_jitter;
|
|
|
|
/* for dropped packet messages */
|
|
GstClockTime last_drop_msg_timestamp;
|
|
/* accumulators; reset every time a drop message is posted */
|
|
guint num_too_late;
|
|
guint num_drop_on_latency;
|
|
};
|
|
typedef enum
|
|
{
|
|
REASON_TOO_LATE,
|
|
REASON_DROP_ON_LATENCY
|
|
} DropMessageReason;
|
|
|
|
typedef struct
|
|
{
|
|
gchar *cname;
|
|
guint32 ssrc;
|
|
} CNameSSRCMapping;
|
|
|
|
static void
|
|
cname_ssrc_mapping_free (CNameSSRCMapping * mapping)
|
|
{
|
|
g_free (mapping->cname);
|
|
g_free (mapping);
|
|
}
|
|
|
|
static void
|
|
insert_cname_ssrc_mapping (GstRtpJitterBuffer * jbuf, const gchar * cname,
|
|
guint32 ssrc)
|
|
{
|
|
CNameSSRCMapping *map;
|
|
GList *l;
|
|
|
|
GST_DEBUG_OBJECT (jbuf, "Adding SSRC %08x to CNAME %s", ssrc, cname);
|
|
|
|
for (l = jbuf->priv->cname_ssrc_mappings; l; l = l->next) {
|
|
map = l->data;
|
|
|
|
if (map->ssrc == ssrc) {
|
|
if (strcmp (cname, map->cname) != 0) {
|
|
g_free (map->cname);
|
|
map->cname = g_strdup (cname);
|
|
}
|
|
return;
|
|
}
|
|
}
|
|
|
|
map = g_new0 (CNameSSRCMapping, 1);
|
|
map->cname = g_strdup (cname);
|
|
map->ssrc = ssrc;
|
|
jbuf->priv->cname_ssrc_mappings =
|
|
g_list_prepend (jbuf->priv->cname_ssrc_mappings, map);
|
|
}
|
|
|
|
static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("application/x-rtp"
|
|
/* "clock-rate = (int) [ 1, 2147483647 ], "
|
|
* "payload = (int) , "
|
|
* "encoding-name = (string) "
|
|
*/ )
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
|
|
GST_PAD_SINK,
|
|
GST_PAD_REQUEST,
|
|
GST_STATIC_CAPS ("application/x-rtcp")
|
|
);
|
|
|
|
static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
|
|
GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("application/x-rtp"
|
|
/* "payload = (int) , "
|
|
* "clock-rate = (int) , "
|
|
* "encoding-name = (string) "
|
|
*/ )
|
|
);
|
|
|
|
static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
|
|
|
|
#define gst_rtp_jitter_buffer_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_PRIVATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer,
|
|
GST_TYPE_ELEMENT);
|
|
GST_ELEMENT_REGISTER_DEFINE (rtpjitterbuffer, "rtpjitterbuffer", GST_RANK_NONE,
|
|
GST_TYPE_RTP_JITTER_BUFFER);
|
|
|
|
/* object overrides */
|
|
static void gst_rtp_jitter_buffer_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec);
|
|
static void gst_rtp_jitter_buffer_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec);
|
|
static void gst_rtp_jitter_buffer_finalize (GObject * object);
|
|
|
|
/* element overrides */
|
|
static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
|
|
* element, GstStateChange transition);
|
|
static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
|
|
static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
|
|
GstPad * pad);
|
|
static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
|
|
static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
|
|
GstClock * clock);
|
|
|
|
/* pad overrides */
|
|
static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
|
|
static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
|
|
GstObject * parent);
|
|
|
|
/* sinkpad overrides */
|
|
static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
|
|
GstObject * parent, GstEvent * event);
|
|
static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
|
|
GstObject * parent, GstBuffer * buffer);
|
|
static GstFlowReturn gst_rtp_jitter_buffer_chain_list (GstPad * pad,
|
|
GstObject * parent, GstBufferList * buffer_list);
|
|
|
|
static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
|
|
GstObject * parent, GstEvent * event);
|
|
static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
|
|
GstObject * parent, GstBuffer * buffer);
|
|
|
|
static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
|
|
GstObject * parent, GstQuery * query);
|
|
|
|
/* srcpad overrides */
|
|
static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
|
|
GstObject * parent, GstEvent * event);
|
|
static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
|
|
GstObject * parent, GstPadMode mode, gboolean active);
|
|
static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
|
|
static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
|
|
GstObject * parent, GstQuery * query);
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
|
|
static GstClockTime
|
|
gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
|
|
gboolean active, guint64 base_time);
|
|
static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
|
|
static void do_handle_sync_inband (GstRtpJitterBuffer * jitterbuffer,
|
|
guint64 ntpnstime);
|
|
|
|
static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
|
|
|
|
static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
|
|
|
|
static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
|
|
jitterbuffer);
|
|
|
|
static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
|
|
const RtpTimer * timer, GstClockTime dts, gboolean success);
|
|
|
|
static GstClockTime get_current_running_time (GstRtpJitterBuffer *
|
|
jitterbuffer);
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
|
|
gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
|
|
|
|
gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
|
|
gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:latency:
|
|
*
|
|
* The maximum latency of the jitterbuffer. Packets will be kept in the buffer
|
|
* for at most this time.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_LATENCY,
|
|
g_param_spec_uint ("latency", "Buffer latency in ms",
|
|
"Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:drop-on-latency:
|
|
*
|
|
* Drop oldest buffers when the queue is completely filled.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
|
|
g_param_spec_boolean ("drop-on-latency",
|
|
"Drop buffers when maximum latency is reached",
|
|
"Tells the jitterbuffer to never exceed the given latency in size",
|
|
DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:ts-offset:
|
|
*
|
|
* Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
|
|
* This is mainly used to ensure interstream synchronisation.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
|
|
g_param_spec_int64 ("ts-offset", "Timestamp Offset",
|
|
"Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
|
|
G_MAXINT64, DEFAULT_TS_OFFSET,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:max-ts-offset-adjustment:
|
|
*
|
|
* The maximum number of nanoseconds per frame that time offset may be
|
|
* adjusted with. This is used to avoid sudden large changes to time stamps.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
|
|
g_param_spec_uint64 ("max-ts-offset-adjustment",
|
|
"Max Timestamp Offset Adjustment",
|
|
"The maximum number of nanoseconds per frame that time stamp "
|
|
"offsets may be adjusted (0 = no limit).", 0, G_MAXUINT64,
|
|
DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
|
|
G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:do-lost:
|
|
*
|
|
* Send out a GstRTPPacketLost event downstream when a packet is considered
|
|
* lost.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_DO_LOST,
|
|
g_param_spec_boolean ("do-lost", "Do Lost",
|
|
"Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:post-drop-messages:
|
|
*
|
|
* Post custom messages to the bus when a packet is dropped by the
|
|
* jitterbuffer due to arriving too late, being already considered lost,
|
|
* or being dropped due to the drop-on-latency property being enabled.
|
|
* Message is of type GST_MESSAGE_ELEMENT and contains a GstStructure named
|
|
* "drop-msg" with the following fields:
|
|
*
|
|
* * #guint `seqnum`: Seqnum of dropped packet.
|
|
* * #guint64 `timestamp`: PTS timestamp of dropped packet.
|
|
* * #GString `reason`: Reason for dropping the packet.
|
|
* * #guint `num-too-late`: Number of packets arriving too late since
|
|
* last drop message.
|
|
* * #guint `num-drop-on-latency`: Number of packets dropped due to the
|
|
* drop-on-latency property since last drop message.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_POST_DROP_MESSAGES,
|
|
g_param_spec_boolean ("post-drop-messages", "Post drop messages",
|
|
"Post a custom message to the bus when a packet is dropped by the jitterbuffer",
|
|
DEFAULT_POST_DROP_MESSAGES,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:drop-messages-interval:
|
|
*
|
|
* Minimal time in milliseconds between posting dropped packet messages, if enabled
|
|
* by setting property by setting #GstRtpJitterBuffer:post-drop-messages to %TRUE.
|
|
* If interval is set to 0, every dropped packet will result in a drop message being posted.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_DROP_MESSAGES_INTERVAL,
|
|
g_param_spec_uint ("drop-messages-interval",
|
|
"Drop message interval",
|
|
"Minimal time between posting dropped packet messages", 0,
|
|
G_MAXUINT, DEFAULT_DROP_MESSAGES_INTERVAL_MS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:mode:
|
|
*
|
|
* Control the buffering and timestamping mode used by the jitterbuffer.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MODE,
|
|
g_param_spec_enum ("mode", "Mode",
|
|
"Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
|
|
DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:percent:
|
|
*
|
|
* The percent of the jitterbuffer that is filled.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_PERCENT,
|
|
g_param_spec_int ("percent", "percent",
|
|
"The buffer filled percent", 0, 100,
|
|
0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:do-retransmission:
|
|
*
|
|
* Send out a GstRTPRetransmission event upstream when a packet is considered
|
|
* late and should be retransmitted.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
|
|
g_param_spec_boolean ("do-retransmission", "Do Retransmission",
|
|
"Send retransmission events upstream when a packet is late",
|
|
DEFAULT_DO_RETRANSMISSION,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-next-seqnum
|
|
*
|
|
* Estimate when the next packet should arrive and schedule a retransmission
|
|
* request for it.
|
|
* This is, when packet N arrives, a GstRTPRetransmission event is schedule
|
|
* for packet N+1. So it will be requested if it does not arrive at the expected time.
|
|
* The expected time is calculated using the dts of N and the packet spacing.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
|
|
g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
|
|
"Estimate when the next packet should arrive and schedule a "
|
|
"retransmission request for it.",
|
|
DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-delay:
|
|
*
|
|
* When a packet did not arrive at the expected time, wait this extra amount
|
|
* of time before sending a retransmission event.
|
|
*
|
|
* When -1 is used, the max jitter will be used as extra delay.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
|
|
g_param_spec_int ("rtx-delay", "RTX Delay",
|
|
"Extra time in ms to wait before sending retransmission "
|
|
"event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-min-delay:
|
|
*
|
|
* When a packet did not arrive at the expected time, wait at least this extra amount
|
|
* of time before sending a retransmission event.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
|
|
g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
|
|
"Minimum time in ms to wait before sending retransmission "
|
|
"event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-delay-reorder:
|
|
*
|
|
* Assume that a retransmission event should be sent when we see
|
|
* this much packet reordering.
|
|
*
|
|
* When -1 is used, the value will be estimated based on observed packet
|
|
* reordering. When 0 is used packet reordering alone will not cause a
|
|
* retransmission event (Since 1.10).
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
|
|
g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
|
|
"Sending retransmission event when this much reordering "
|
|
"(0 disable)",
|
|
-1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-retry-timeout:
|
|
*
|
|
* When no packet has been received after sending a retransmission event
|
|
* for this time, retry sending a retransmission event.
|
|
*
|
|
* When -1 is used, the value will be estimated based on observed round
|
|
* trip time.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
|
|
g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
|
|
"Retry sending a transmission event after this timeout in "
|
|
"ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-min-retry-timeout:
|
|
*
|
|
* The minimum amount of time between retry timeouts. When
|
|
* GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
|
|
* minimum interval between retry timeouts.
|
|
*
|
|
* When -1 is used, the value will be estimated based on the
|
|
* packet spacing.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
|
|
g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
|
|
"Minimum timeout between sending a transmission event in "
|
|
"ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-retry-period:
|
|
*
|
|
* The amount of time to try to get a retransmission.
|
|
*
|
|
* When -1 is used, the value will be estimated based on the jitterbuffer
|
|
* latency and the observed round trip time.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
|
|
g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
|
|
"Try to get a retransmission for this many ms "
|
|
"(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-max-retries:
|
|
*
|
|
* The maximum number of retries to request a retransmission.
|
|
*
|
|
* This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
|
|
* When -1 is used, the number of retransmission request will not be limited.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
|
|
g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
|
|
"The maximum number of retries to request a retransmission. "
|
|
"(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-deadline:
|
|
*
|
|
* The deadline for a valid RTX request in ms.
|
|
*
|
|
* How long the RTX RTCP will be valid for.
|
|
* When -1 is used, the size of the jitterbuffer will be used.
|
|
*
|
|
* Since: 1.10
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
|
|
g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
|
|
"The deadline for a valid RTX request in milliseconds. "
|
|
"(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:rtx-stats-timeout:
|
|
*
|
|
* The time to wait for a retransmitted packet after it has been
|
|
* considered lost in order to collect RTX statistics.
|
|
*
|
|
* Since: 1.10
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
|
|
g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
|
|
"The time to wait for a retransmitted packet after it has been "
|
|
"considered lost in order to collect statistics (ms)",
|
|
0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
|
|
g_param_spec_uint ("max-dropout-time", "Max dropout time",
|
|
"The maximum time (milliseconds) of missing packets tolerated.",
|
|
0, G_MAXINT32, DEFAULT_MAX_DROPOUT_TIME,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
|
|
g_param_spec_uint ("max-misorder-time", "Max misorder time",
|
|
"The maximum time (milliseconds) of misordered packets tolerated.",
|
|
0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpJitterBuffer:stats:
|
|
*
|
|
* Various jitterbuffer statistics. This property returns a GstStructure
|
|
* with name application/x-rtp-jitterbuffer-stats with the following fields:
|
|
*
|
|
* * #guint64 `num-pushed`: the number of packets pushed out.
|
|
* * #guint64 `num-lost`: the number of packets considered lost.
|
|
* * #guint64 `num-late`: the number of packets arriving too late.
|
|
* * #guint64 `num-duplicates`: the number of duplicate packets.
|
|
* * #guint64 `avg-jitter`: the average jitter in nanoseconds.
|
|
* * #guint64 `rtx-count`: the number of retransmissions requested.
|
|
* * #guint64 `rtx-success-count`: the number of successful retransmissions.
|
|
* * #gdouble `rtx-per-packet`: average number of RTX per packet.
|
|
* * #guint64 `rtx-rtt`: average round trip time per RTX.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_STATS,
|
|
g_param_spec_boxed ("stats", "Statistics",
|
|
"Various statistics", GST_TYPE_STRUCTURE,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:max-rtcp-rtp-time-diff
|
|
*
|
|
* The maximum amount of time in ms that the RTP time in the RTCP SRs
|
|
* is allowed to be ahead of the last RTP packet we received. Use
|
|
* -1 to disable ignoring of RTCP packets.
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
|
|
g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
|
|
"Maximum amount of time in ms that the RTP time in RTCP SRs "
|
|
"is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
|
|
DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
|
|
g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
|
|
"Synchronize received streams to the RFC7273 clock "
|
|
"(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:add-reference-timestamp-meta:
|
|
*
|
|
* When syncing to a RFC7273 clock or after clock synchronization via RTCP or
|
|
* inband NTP-64 header extensions has happened, add #GstReferenceTimestampMeta
|
|
* to buffers with the original reconstructed reference clock timestamp.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_ADD_REFERENCE_TIMESTAMP_META,
|
|
g_param_spec_boolean ("add-reference-timestamp-meta",
|
|
"Add Reference Timestamp Meta",
|
|
"Add Reference Timestamp Meta to buffers with the original clock timestamp "
|
|
"before any adjustments when syncing to an RFC7273 clock or after clock "
|
|
"synchronization via RTCP or inband NTP-64 header extensions has happened.",
|
|
DEFAULT_ADD_REFERENCE_TIMESTAMP_META,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:faststart-min-packets
|
|
*
|
|
* The number of consecutive packets needed to start (set to 0 to
|
|
* disable faststart. The jitterbuffer will by default start after the
|
|
* latency has elapsed)
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_FASTSTART_MIN_PACKETS,
|
|
g_param_spec_uint ("faststart-min-packets", "Faststart minimum packets",
|
|
"The number of consecutive packets needed to start (set to 0 to "
|
|
"disable faststart. The jitterbuffer will by default start after "
|
|
"the latency has elapsed)",
|
|
0, G_MAXUINT, DEFAULT_FASTSTART_MIN_PACKETS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:sync-interval:
|
|
*
|
|
* Determines how often to sync streams using RTCP data or inband NTP-64
|
|
* header extensions.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_SYNC_INTERVAL,
|
|
g_param_spec_uint ("sync-interval", "Sync Interval",
|
|
"RTCP SR / NTP-64 interval synchronization (ms) (0 = always)",
|
|
0, G_MAXUINT, DEFAULT_SYNC_INTERVAL,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:rfc7273-use-system-clock:
|
|
*
|
|
* Uses the system clock as media clock in RFC7273 mode instead of
|
|
* instantiating an NTP or PTP clock.
|
|
*
|
|
* This will always provide the correct sender timestamps in the
|
|
* `GstReferenceTimestampMeta` as long as the system clock is synced to the
|
|
* actual media clock with at most a few seconds difference.
|
|
*
|
|
* PTS on outgoing buffers would be as accurate as the synchronization
|
|
* between the actual media clock and the system clock.
|
|
*
|
|
* This can be useful if only recovery of the original sender timestamps is
|
|
* needed and syncing to a PTP/NTP clock would be unnecessarily complex, or
|
|
* if the system clock already is synchronized to the correct clock and
|
|
* doing it another time inside GStreamer would be unnecessary.
|
|
*
|
|
* Since: 1.24
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RFC7273_USE_SYSTEM_CLOCK,
|
|
g_param_spec_boolean ("rfc7273-use-system-clock",
|
|
"Use system clock as RFC7273 clock",
|
|
"Use system clock as RFC7273 media clock (requires system clock "
|
|
"to be synced externally)", DEFAULT_RFC7273_USE_SYSTEM_CLOCK,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpJitterBuffer:rfc7273-reference-timestamp-meta-only:
|
|
*
|
|
* When enabled, the jitterbuffer calculates the PTS of the outgoing buffers
|
|
* according to the configured mode as if not RFC7273 mode is enabled.
|
|
*
|
|
* The timestamps calculated from the RFC7273 clock are only put into the
|
|
* reference timestamp meta, if enabled via the corresponding property.
|
|
*
|
|
* This is useful in combination with the `rfc7273-use-system-clock`, or
|
|
* generally if synchronization should not be affected by the original
|
|
* sender timestamps but the original sender timestamps should nonetheless
|
|
* be available as metadata.
|
|
*
|
|
* Since: 1.24
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_RFC7273_REFERENCE_TIMESTAMP_META_ONLY,
|
|
g_param_spec_boolean ("rfc7273-reference-timestamp-meta-only",
|
|
"Use RFC7273 clock only for reference timestamp meta",
|
|
"When enabled the PTS on the buffers are calculated normally and the "
|
|
"RFC7273 clock is only used for the reference timestamp meta",
|
|
DEFAULT_RFC7273_REFERENCE_TIMESTAMP_META_ONLY,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
|
|
/**
|
|
* GstRtpJitterBuffer::request-pt-map:
|
|
* @buffer: the object which received the signal
|
|
* @pt: the pt
|
|
*
|
|
* Request the payload type as #GstCaps for @pt.
|
|
*/
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
|
|
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
|
|
request_pt_map), NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpJitterBuffer::handle-sync:
|
|
* @buffer: the object which received the signal
|
|
* @struct: a GstStructure containing sync values.
|
|
*
|
|
* Be notified of new sync values.
|
|
*/
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
|
|
g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
|
|
handle_sync), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
|
|
|
|
/**
|
|
* GstRtpJitterBuffer::on-npt-stop:
|
|
* @buffer: the object which received the signal
|
|
*
|
|
* Signal that the jitterbuffer has pushed the RTP packet that corresponds to
|
|
* the npt-stop position.
|
|
*/
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
|
|
g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
|
|
on_npt_stop), NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* GstRtpJitterBuffer::clear-pt-map:
|
|
* @buffer: the object which received the signal
|
|
*
|
|
* Invalidate the clock-rate as obtained with the
|
|
* #GstRtpJitterBuffer::request-pt-map signal.
|
|
*/
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
|
|
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
|
|
NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* GstRtpJitterBuffer::set-active:
|
|
* @buffer: the object which received the signal
|
|
*
|
|
* Start pushing out packets with the given base time. This signal is only
|
|
* useful in buffering mode.
|
|
*
|
|
* Returns: the time of the last pushed packet.
|
|
*/
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
|
|
g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
|
|
NULL, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN, G_TYPE_UINT64);
|
|
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
|
|
gstelement_class->request_new_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
|
|
gstelement_class->release_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
|
|
gstelement_class->provide_clock =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
|
|
gstelement_class->set_clock =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
|
|
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_jitter_buffer_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_jitter_buffer_sink_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&gst_rtp_jitter_buffer_sink_rtcp_template);
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"RTP packet jitter-buffer", "Filter/Network/RTP",
|
|
"A buffer that deals with network jitter and other transmission faults",
|
|
"Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
|
|
"Wim Taymans <wim.taymans@gmail.com>");
|
|
|
|
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
|
|
klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
|
|
|
|
GST_DEBUG_CATEGORY_INIT
|
|
(rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_jitter_buffer_chain_rtcp);
|
|
|
|
gst_type_mark_as_plugin_api (RTP_TYPE_JITTER_BUFFER_MODE, 0);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = gst_rtp_jitter_buffer_get_instance_private (jitterbuffer);
|
|
jitterbuffer->priv = priv;
|
|
|
|
priv->latency_ms = DEFAULT_LATENCY_MS;
|
|
priv->latency_ns = priv->latency_ms * GST_MSECOND;
|
|
priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
|
|
priv->ts_offset = DEFAULT_TS_OFFSET;
|
|
priv->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
|
|
priv->do_lost = DEFAULT_DO_LOST;
|
|
priv->post_drop_messages = DEFAULT_POST_DROP_MESSAGES;
|
|
priv->drop_messages_interval_ms = DEFAULT_DROP_MESSAGES_INTERVAL_MS;
|
|
priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
|
|
priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
|
|
priv->rtx_delay = DEFAULT_RTX_DELAY;
|
|
priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
|
|
priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
|
|
priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
|
|
priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
|
|
priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
|
|
priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
|
|
priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
|
|
priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
|
|
priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
|
|
priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
|
|
priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
|
|
priv->faststart_min_packets = DEFAULT_FASTSTART_MIN_PACKETS;
|
|
priv->add_reference_timestamp_meta = DEFAULT_ADD_REFERENCE_TIMESTAMP_META;
|
|
priv->sync_interval = DEFAULT_SYNC_INTERVAL;
|
|
priv->rfc7273_use_system_clock = DEFAULT_RFC7273_USE_SYSTEM_CLOCK;
|
|
priv->rfc7273_reference_timestamp_meta_only =
|
|
DEFAULT_RFC7273_REFERENCE_TIMESTAMP_META_ONLY;
|
|
|
|
priv->ts_offset_remainder = 0;
|
|
priv->last_dts = -1;
|
|
priv->last_pts = -1;
|
|
priv->last_rtptime = -1;
|
|
priv->last_ntpnstime = -1;
|
|
priv->last_known_ext_rtptime = -1;
|
|
priv->last_known_ntpnstime = -1;
|
|
priv->avg_jitter = 0;
|
|
priv->last_drop_msg_timestamp = GST_CLOCK_TIME_NONE;
|
|
priv->num_too_late = 0;
|
|
priv->num_drop_on_latency = 0;
|
|
priv->segment_seqnum = GST_SEQNUM_INVALID;
|
|
priv->timers = rtp_timer_queue_new ();
|
|
priv->rtx_stats_timers = rtp_timer_queue_new ();
|
|
priv->jbuf = rtp_jitter_buffer_new ();
|
|
g_mutex_init (&priv->jbuf_lock);
|
|
g_cond_init (&priv->jbuf_queue);
|
|
g_cond_init (&priv->jbuf_timer);
|
|
g_cond_init (&priv->jbuf_event);
|
|
g_cond_init (&priv->jbuf_query);
|
|
g_queue_init (&priv->gap_packets);
|
|
gst_segment_init (&priv->segment, GST_FORMAT_TIME);
|
|
|
|
/* reset skew detection initially */
|
|
rtp_jitter_buffer_reset_skew (priv->jbuf);
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
|
|
rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
|
|
priv->active = TRUE;
|
|
|
|
priv->srcpad =
|
|
gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
|
|
"src");
|
|
|
|
gst_pad_set_activatemode_function (priv->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
|
|
gst_pad_set_query_function (priv->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
|
|
gst_pad_set_event_function (priv->srcpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
|
|
|
|
priv->sinkpad =
|
|
gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
|
|
"sink");
|
|
|
|
gst_pad_set_chain_function (priv->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
|
|
gst_pad_set_chain_list_function (priv->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain_list));
|
|
gst_pad_set_event_function (priv->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
|
|
gst_pad_set_query_function (priv->sinkpad,
|
|
GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
|
|
|
|
gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
|
|
gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
|
|
|
|
GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
|
|
}
|
|
|
|
static void
|
|
free_item_and_retain_sticky_events (RTPJitterBufferItem * item,
|
|
gpointer user_data)
|
|
{
|
|
GList **l = user_data;
|
|
|
|
if (item->data && item->type == ITEM_TYPE_EVENT
|
|
&& GST_EVENT_IS_STICKY (item->data)) {
|
|
*l = g_list_prepend (*l, item->data);
|
|
item->data = NULL;
|
|
}
|
|
|
|
rtp_jitter_buffer_free_item (item);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_finalize (GObject * object)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
|
|
priv = jitterbuffer->priv;
|
|
|
|
g_object_unref (priv->timers);
|
|
g_object_unref (priv->rtx_stats_timers);
|
|
g_mutex_clear (&priv->jbuf_lock);
|
|
g_cond_clear (&priv->jbuf_queue);
|
|
g_cond_clear (&priv->jbuf_timer);
|
|
g_cond_clear (&priv->jbuf_event);
|
|
g_cond_clear (&priv->jbuf_query);
|
|
|
|
rtp_jitter_buffer_flush (priv->jbuf, NULL, NULL);
|
|
g_list_free_full (priv->cname_ssrc_mappings,
|
|
(GDestroyNotify) cname_ssrc_mapping_free);
|
|
priv->cname_ssrc_mappings = NULL;
|
|
g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
|
|
g_queue_clear (&priv->gap_packets);
|
|
g_object_unref (priv->jbuf);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static GstIterator *
|
|
gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstPad *otherpad = NULL;
|
|
GstIterator *it = NULL;
|
|
GValue val = { 0, };
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
|
|
|
|
if (pad == jitterbuffer->priv->sinkpad) {
|
|
otherpad = jitterbuffer->priv->srcpad;
|
|
} else if (pad == jitterbuffer->priv->srcpad) {
|
|
otherpad = jitterbuffer->priv->sinkpad;
|
|
} else if (pad == jitterbuffer->priv->rtcpsinkpad) {
|
|
it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
|
|
}
|
|
|
|
if (it == NULL) {
|
|
g_value_init (&val, GST_TYPE_PAD);
|
|
g_value_set_object (&val, otherpad);
|
|
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
|
|
g_value_unset (&val);
|
|
}
|
|
|
|
return it;
|
|
}
|
|
|
|
static GstPad *
|
|
create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
|
|
|
|
priv->rtcpsinkpad =
|
|
gst_pad_new_from_static_template
|
|
(&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
|
|
gst_pad_set_chain_function (priv->rtcpsinkpad,
|
|
gst_rtp_jitter_buffer_chain_rtcp);
|
|
gst_pad_set_event_function (priv->rtcpsinkpad,
|
|
(GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
|
|
gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
|
|
gst_rtp_jitter_buffer_iterate_internal_links);
|
|
gst_pad_set_active (priv->rtcpsinkpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
|
|
|
|
return priv->rtcpsinkpad;
|
|
}
|
|
|
|
static void
|
|
remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
|
|
|
|
gst_pad_set_active (priv->rtcpsinkpad, FALSE);
|
|
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
|
|
priv->rtcpsinkpad = NULL;
|
|
}
|
|
|
|
static GstPad *
|
|
gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
|
|
priv = jitterbuffer->priv;
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
|
|
if (priv->rtcpsinkpad != NULL)
|
|
goto exists;
|
|
|
|
result = create_rtcp_sink (jitterbuffer);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_warning ("rtpjitterbuffer: this is not our template");
|
|
return NULL;
|
|
}
|
|
exists:
|
|
{
|
|
g_warning ("rtpjitterbuffer: pad already requested");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
|
|
g_return_if_fail (GST_IS_PAD (pad));
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
|
|
|
|
if (priv->rtcpsinkpad == pad) {
|
|
remove_rtcp_sink (jitterbuffer);
|
|
} else
|
|
goto wrong_pad;
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
wrong_pad:
|
|
{
|
|
g_warning ("gstjitterbuffer: asked to release an unknown pad");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstClock *
|
|
gst_rtp_jitter_buffer_provide_clock (GstElement * element)
|
|
{
|
|
return gst_system_clock_obtain ();
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
|
|
|
|
rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
|
|
|
|
return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
/* this will trigger a new pt-map request signal, FIXME, do something better. */
|
|
|
|
JBUF_LOCK (priv);
|
|
priv->clock_rate = -1;
|
|
/* do not clear current content, but refresh state for new arrival */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
|
|
rtp_jitter_buffer_reset_skew (priv->jbuf);
|
|
JBUF_UNLOCK (priv);
|
|
}
|
|
|
|
static GstClockTime
|
|
gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
|
|
guint64 offset)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstClockTime last_out;
|
|
RTPJitterBufferItem *item;
|
|
|
|
priv = jbuf->priv;
|
|
|
|
JBUF_LOCK (priv);
|
|
GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
|
|
active, GST_TIME_ARGS (offset));
|
|
|
|
if (active != priv->active) {
|
|
/* add the amount of time spent in paused to the output offset. All
|
|
* outgoing buffers will have this offset applied to their timestamps in
|
|
* order to make them arrive in time in the sink. */
|
|
priv->out_offset = offset;
|
|
GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->out_offset));
|
|
priv->active = active;
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
if (!active) {
|
|
rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
|
|
}
|
|
if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
|
|
/* head buffer timestamp and offset gives our output time */
|
|
last_out = item->pts + priv->ts_offset;
|
|
} else {
|
|
/* use last known time when the buffer is empty */
|
|
last_out = priv->last_out_time;
|
|
}
|
|
JBUF_UNLOCK (priv);
|
|
|
|
return last_out;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstPad *other;
|
|
GstCaps *caps;
|
|
GstCaps *templ;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
|
|
priv = jitterbuffer->priv;
|
|
|
|
other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
|
|
|
|
caps = gst_pad_peer_query_caps (other, filter);
|
|
|
|
templ = gst_pad_get_pad_template_caps (pad);
|
|
if (caps == NULL) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "use template");
|
|
caps = templ;
|
|
} else {
|
|
GstCaps *intersect;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
|
|
|
|
intersect = gst_caps_intersect (caps, templ);
|
|
gst_caps_unref (caps);
|
|
gst_caps_unref (templ);
|
|
|
|
caps = intersect;
|
|
}
|
|
gst_object_unref (jitterbuffer);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static void
|
|
_get_cname_ssrc_mappings (GstRtpJitterBuffer * jitterbuffer,
|
|
const GstStructure * s)
|
|
{
|
|
guint i;
|
|
guint n_fields = gst_structure_n_fields (s);
|
|
|
|
for (i = 0; i < n_fields; i++) {
|
|
const gchar *field_name = gst_structure_nth_field_name (s, i);
|
|
if (g_str_has_prefix (field_name, "ssrc-")
|
|
&& g_str_has_suffix (field_name, "-cname")) {
|
|
const gchar *str = gst_structure_get_string (s, field_name);
|
|
gchar *endptr;
|
|
guint32 ssrc = g_ascii_strtoll (field_name + 5, &endptr, 10);
|
|
|
|
if (!endptr || *endptr != '-')
|
|
continue;
|
|
|
|
insert_cname_ssrc_mapping (jitterbuffer, str, ssrc);
|
|
}
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Must be called with JBUF_LOCK held
|
|
*/
|
|
|
|
static gboolean
|
|
gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
|
|
GstCaps * caps, gint pt)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstStructure *caps_struct;
|
|
guint val;
|
|
gint payload = -1;
|
|
GstClockTime tval;
|
|
const gchar *ts_refclk, *mediaclk;
|
|
GstCaps *ts_meta_ref = NULL;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
/* first parse the caps */
|
|
caps_struct = gst_caps_get_structure (caps, 0);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
|
|
|
|
if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
|
|
&& payload != pt) {
|
|
GST_ERROR_OBJECT (jitterbuffer,
|
|
"Got caps with wrong payload type (got %d, expected %d)", pt, payload);
|
|
return FALSE;
|
|
}
|
|
|
|
if (payload != -1) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
|
|
priv->last_pt = payload;
|
|
}
|
|
|
|
/* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
|
|
* measure the amount of data in the buffer */
|
|
if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
|
|
goto error;
|
|
|
|
if (priv->clock_rate <= 0)
|
|
goto wrong_rate;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
|
|
|
|
rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
|
|
|
|
gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
|
|
|
|
/* The clock base is the RTP timestamp corresponding to the npt-start value. We
|
|
* can use this to track the amount of time elapsed on the sender. */
|
|
priv->ext_timestamp = -1;
|
|
if (gst_structure_get_uint (caps_struct, "clock-base", &val)) {
|
|
priv->clock_base = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, val);
|
|
priv->ext_timestamp = priv->clock_base;
|
|
} else {
|
|
priv->clock_base = -1;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
|
|
priv->clock_base);
|
|
|
|
if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
|
|
/* first expected seqnum, only update when we didn't have a previous base. */
|
|
if (priv->next_in_seqnum == -1)
|
|
priv->next_in_seqnum = val;
|
|
if (priv->next_seqnum == -1) {
|
|
priv->next_seqnum = val;
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
priv->seqnum_base = val;
|
|
} else {
|
|
priv->seqnum_base = -1;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
|
|
|
|
/* the start and stop times. The seqnum-base corresponds to the start time. We
|
|
* will keep track of the seqnums on the output and when we reach the one
|
|
* corresponding to npt-stop, we emit the npt-stop-reached signal */
|
|
if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
|
|
priv->npt_start = tval;
|
|
else
|
|
priv->npt_start = 0;
|
|
|
|
if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
|
|
priv->npt_stop = tval;
|
|
else
|
|
priv->npt_stop = -1;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
|
|
|
|
if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
|
|
gboolean use_system_clock;
|
|
gboolean reference_timestamp_meta_only;
|
|
GstClock *clock = NULL;
|
|
guint64 clock_offset = -1;
|
|
gint64 clock_correction = 0;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
|
|
ts_refclk);
|
|
|
|
use_system_clock = priv->rfc7273_use_system_clock;
|
|
reference_timestamp_meta_only = priv->rfc7273_reference_timestamp_meta_only;
|
|
|
|
if (g_str_has_prefix (ts_refclk, "ntp=")) {
|
|
if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
|
|
GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
|
|
} else {
|
|
const gchar *host, *portstr;
|
|
gchar *hostname;
|
|
guint port;
|
|
|
|
host = ts_refclk + sizeof ("ntp=") - 1;
|
|
if (host[0] == '[') {
|
|
/* IPv6 */
|
|
portstr = strchr (host, ']');
|
|
if (portstr && portstr[1] == ':')
|
|
portstr = portstr + 1;
|
|
else
|
|
portstr = NULL;
|
|
} else {
|
|
portstr = strrchr (host, ':');
|
|
}
|
|
|
|
|
|
if (!portstr || sscanf (portstr, ":%u", &port) != 1)
|
|
port = 123;
|
|
|
|
if (portstr)
|
|
hostname = g_strndup (host, (portstr - host));
|
|
else
|
|
hostname = g_strdup (host);
|
|
|
|
if (use_system_clock) {
|
|
clock =
|
|
g_object_new (GST_TYPE_SYSTEM_CLOCK, "clock-type",
|
|
GST_CLOCK_TYPE_REALTIME, NULL);
|
|
/* difference between UNIX epoch and NTP epoch */
|
|
clock_correction = GST_RTP_NTP_UNIX_OFFSET * GST_SECOND;
|
|
} else {
|
|
clock = gst_ntp_clock_new (NULL, hostname, port, 0);
|
|
}
|
|
|
|
ts_meta_ref = gst_caps_new_simple ("timestamp/x-ntp",
|
|
"host", G_TYPE_STRING, hostname, "port", G_TYPE_INT, port, NULL);
|
|
|
|
g_free (hostname);
|
|
}
|
|
} else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
|
|
const gchar *domainstr =
|
|
ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
|
|
guint domain;
|
|
|
|
if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
|
|
domain = 0;
|
|
|
|
if (use_system_clock) {
|
|
clock =
|
|
g_object_new (GST_TYPE_SYSTEM_CLOCK, "clock-type",
|
|
GST_CLOCK_TYPE_REALTIME, NULL);
|
|
/* difference between UNIX and PTP/TAI (37 leap seconds as of October 2023) */
|
|
clock_correction = 37 * GST_SECOND;
|
|
} else {
|
|
clock = gst_ptp_clock_new (NULL, domain);
|
|
}
|
|
|
|
ts_meta_ref = gst_caps_new_simple ("timestamp/x-ptp",
|
|
"version", G_TYPE_STRING, "IEEE1588-2008",
|
|
"domain", G_TYPE_INT, domain, NULL);
|
|
} else if (!g_strcmp0 (ts_refclk, "local")) {
|
|
ts_meta_ref = gst_caps_new_empty_simple ("timestamp/x-ntp");
|
|
} else {
|
|
if (use_system_clock) {
|
|
clock =
|
|
g_object_new (GST_TYPE_SYSTEM_CLOCK, "clock-type",
|
|
GST_CLOCK_TYPE_REALTIME, NULL);
|
|
} else {
|
|
GST_FIXME_OBJECT (jitterbuffer,
|
|
"Unsupported timestamp reference clock");
|
|
}
|
|
}
|
|
|
|
if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
|
|
|
|
if (!g_str_has_prefix (mediaclk, "direct=") ||
|
|
!g_ascii_string_to_unsigned (&mediaclk[7], 10, 0, G_MAXUINT64,
|
|
&clock_offset, NULL))
|
|
GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
|
|
if (strstr (mediaclk, "rate=") != NULL) {
|
|
GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
|
|
clock_offset = -1;
|
|
}
|
|
}
|
|
|
|
rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset,
|
|
clock_correction, reference_timestamp_meta_only);
|
|
} else {
|
|
rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1, 0, FALSE);
|
|
ts_meta_ref = gst_caps_new_empty_simple ("timestamp/x-ntp");
|
|
}
|
|
|
|
gst_caps_take (&priv->reference_timestamp_caps, ts_meta_ref);
|
|
|
|
_get_cname_ssrc_mappings (jitterbuffer, caps_struct);
|
|
priv->ntp64_ext_id =
|
|
gst_rtp_get_extmap_id_for_attribute (caps_struct,
|
|
GST_RTP_HDREXT_BASE GST_RTP_HDREXT_NTP_64);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
|
|
return FALSE;
|
|
}
|
|
wrong_rate:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
JBUF_LOCK (priv);
|
|
/* mark ourselves as flushing */
|
|
priv->srcresult = GST_FLOW_FLUSHING;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
|
|
/* this unblocks any waiting pops on the src pad task */
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
JBUF_SIGNAL_QUERY (priv, FALSE);
|
|
JBUF_SIGNAL_QUEUE (priv);
|
|
JBUF_SIGNAL_TIMER (priv);
|
|
JBUF_UNLOCK (priv);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
JBUF_LOCK (priv);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
|
|
/* Mark as non flushing */
|
|
priv->srcresult = GST_FLOW_OK;
|
|
gst_segment_init (&priv->segment, GST_FORMAT_TIME);
|
|
priv->last_popped_seqnum = -1;
|
|
priv->last_out_time = GST_CLOCK_TIME_NONE;
|
|
priv->next_seqnum = -1;
|
|
priv->seqnum_base = -1;
|
|
priv->ips_rtptime = -1;
|
|
priv->ips_pts = GST_CLOCK_TIME_NONE;
|
|
priv->packet_spacing = 0;
|
|
priv->next_in_seqnum = -1;
|
|
priv->clock_rate = -1;
|
|
priv->ntp64_ext_id = 0;
|
|
priv->last_pt = -1;
|
|
priv->last_ssrc = -1;
|
|
priv->eos = FALSE;
|
|
priv->estimated_eos = -1;
|
|
priv->last_elapsed = 0;
|
|
priv->ext_timestamp = -1;
|
|
priv->avg_jitter = 0;
|
|
priv->last_dts = -1;
|
|
priv->last_rtptime = -1;
|
|
priv->last_ntpnstime = -1;
|
|
priv->last_known_ext_rtptime = -1;
|
|
priv->last_known_ntpnstime = -1;
|
|
priv->last_in_pts = 0;
|
|
priv->equidistant = 0;
|
|
priv->segment_seqnum = GST_SEQNUM_INVALID;
|
|
priv->last_drop_msg_timestamp = GST_CLOCK_TIME_NONE;
|
|
priv->num_too_late = 0;
|
|
priv->num_drop_on_latency = 0;
|
|
g_list_free_full (priv->cname_ssrc_mappings,
|
|
(GDestroyNotify) cname_ssrc_mapping_free);
|
|
priv->cname_ssrc_mappings = NULL;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
|
|
rtp_jitter_buffer_flush (priv->jbuf, NULL, NULL);
|
|
rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
|
|
rtp_jitter_buffer_reset_skew (priv->jbuf);
|
|
rtp_timer_queue_remove_all (priv->timers);
|
|
g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
|
|
g_queue_clear (&priv->gap_packets);
|
|
JBUF_UNLOCK (priv);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
|
|
GstPadMode mode, gboolean active)
|
|
{
|
|
gboolean result;
|
|
GstRtpJitterBuffer *jitterbuffer = NULL;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
switch (mode) {
|
|
case GST_PAD_MODE_PUSH:
|
|
if (active) {
|
|
/* allow data processing */
|
|
gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
|
|
|
|
/* start pushing out buffers */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
|
|
result = gst_pad_start_task (jitterbuffer->priv->srcpad,
|
|
(GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
|
|
} else {
|
|
/* make sure all data processing stops ASAP */
|
|
gst_rtp_jitter_buffer_flush_start (jitterbuffer);
|
|
|
|
/* NOTE this will hardlock if the state change is called from the src pad
|
|
* task thread because we will _join() the thread. */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
|
|
result = gst_pad_stop_task (pad);
|
|
}
|
|
break;
|
|
default:
|
|
result = FALSE;
|
|
break;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_jitter_buffer_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (element);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
JBUF_LOCK (priv);
|
|
/* reset negotiated values */
|
|
priv->clock_rate = -1;
|
|
priv->clock_base = -1;
|
|
priv->peer_latency = 0;
|
|
priv->last_pt = -1;
|
|
priv->last_ssrc = -1;
|
|
priv->ntp64_ext_id = 0;
|
|
g_list_free_full (priv->cname_ssrc_mappings,
|
|
(GDestroyNotify) cname_ssrc_mapping_free);
|
|
priv->cname_ssrc_mappings = NULL;
|
|
/* block until we go to PLAYING */
|
|
priv->blocked = TRUE;
|
|
priv->timer_running = TRUE;
|
|
priv->srcresult = GST_FLOW_OK;
|
|
priv->timer_thread =
|
|
g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
JBUF_LOCK (priv);
|
|
/* unblock to allow streaming in PLAYING */
|
|
priv->blocked = FALSE;
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
JBUF_SIGNAL_TIMER (priv);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* we are a live element because we sync to the clock, which we can only
|
|
* do in the PLAYING state */
|
|
if (ret != GST_STATE_CHANGE_FAILURE)
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
JBUF_LOCK (priv);
|
|
/* block to stop streaming when PAUSED */
|
|
priv->blocked = TRUE;
|
|
unschedule_current_timer (jitterbuffer);
|
|
JBUF_UNLOCK (priv);
|
|
if (ret != GST_STATE_CHANGE_FAILURE)
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
JBUF_LOCK (priv);
|
|
gst_buffer_replace (&priv->last_sr, NULL);
|
|
priv->timer_running = FALSE;
|
|
priv->srcresult = GST_FLOW_FLUSHING;
|
|
unschedule_current_timer (jitterbuffer);
|
|
JBUF_SIGNAL_TIMER (priv);
|
|
JBUF_SIGNAL_QUERY (priv, FALSE);
|
|
JBUF_SIGNAL_QUEUE (priv);
|
|
JBUF_UNLOCK (priv);
|
|
g_thread_join (priv->timer_thread);
|
|
priv->timer_thread = NULL;
|
|
gst_clear_caps (&priv->reference_timestamp_caps);
|
|
g_list_free_full (priv->cname_ssrc_mappings,
|
|
(GDestroyNotify) cname_ssrc_mapping_free);
|
|
priv->cname_ssrc_mappings = NULL;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_LATENCY:
|
|
{
|
|
GstClockTime latency;
|
|
|
|
gst_event_parse_latency (event, &latency);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
|
|
|
|
JBUF_LOCK (priv);
|
|
/* adjust the overall buffer delay to the total pipeline latency in
|
|
* buffering mode because if downstream consumes too fast (because of
|
|
* large latency or queues, we would start rebuffering again. */
|
|
if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
|
|
RTP_JITTER_BUFFER_MODE_BUFFER) {
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, latency);
|
|
}
|
|
JBUF_UNLOCK (priv);
|
|
|
|
ret = gst_pad_push_event (priv->sinkpad, event);
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_push_event (priv->sinkpad, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* handles and stores the event in the jitterbuffer, must be called with
|
|
* LOCK */
|
|
static gboolean
|
|
queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
gboolean head;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
gst_event_parse_caps (event, &caps);
|
|
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
{
|
|
GstSegment segment;
|
|
gst_event_copy_segment (event, &segment);
|
|
|
|
priv->segment_seqnum = gst_event_get_seqnum (event);
|
|
|
|
/* we need time for now */
|
|
if (segment.format != GST_FORMAT_TIME) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
|
|
gst_event_unref (event);
|
|
|
|
gst_segment_init (&segment, GST_FORMAT_TIME);
|
|
event = gst_event_new_segment (&segment);
|
|
gst_event_set_seqnum (event, priv->segment_seqnum);
|
|
}
|
|
|
|
priv->segment = segment;
|
|
break;
|
|
}
|
|
case GST_EVENT_EOS:
|
|
priv->eos = TRUE;
|
|
rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "adding event");
|
|
head = rtp_jitter_buffer_append_event (priv->jbuf, event);
|
|
if (head || priv->eos)
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
ret = gst_pad_push_event (priv->srcpad, event);
|
|
gst_rtp_jitter_buffer_flush_start (jitterbuffer);
|
|
/* wait for the loop to go into PAUSED */
|
|
gst_pad_pause_task (priv->srcpad);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
ret = gst_pad_push_event (priv->srcpad, event);
|
|
ret =
|
|
gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
|
|
GST_PAD_MODE_PUSH, TRUE);
|
|
break;
|
|
default:
|
|
if (GST_EVENT_IS_SERIALIZED (event)) {
|
|
/* serialized events go in the queue */
|
|
JBUF_LOCK (priv);
|
|
if (priv->srcresult != GST_FLOW_OK) {
|
|
/* Errors in sticky event pushing are no problem and ignored here
|
|
* as they will cause more meaningful errors during data flow.
|
|
* For EOS events, that are not followed by data flow, we still
|
|
* return FALSE here though.
|
|
*/
|
|
if (!GST_EVENT_IS_STICKY (event) ||
|
|
GST_EVENT_TYPE (event) == GST_EVENT_EOS)
|
|
goto out_flow_error;
|
|
}
|
|
/* refuse more events on EOS */
|
|
if (priv->eos)
|
|
goto out_eos;
|
|
ret = queue_event (jitterbuffer, event);
|
|
JBUF_UNLOCK (priv);
|
|
} else {
|
|
/* non-serialized events are forwarded downstream immediately */
|
|
ret = gst_pad_push_event (priv->srcpad, event);
|
|
}
|
|
break;
|
|
}
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
out_flow_error:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"refusing event, we have a downstream flow error: %s",
|
|
gst_flow_get_name (priv->srcresult));
|
|
JBUF_UNLOCK (priv);
|
|
gst_event_unref (event);
|
|
return FALSE;
|
|
}
|
|
out_eos:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
|
|
JBUF_UNLOCK (priv);
|
|
gst_event_unref (event);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_START:
|
|
gst_event_unref (event);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_event_unref (event);
|
|
break;
|
|
default:
|
|
ret = gst_pad_event_default (pad, parent, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/*
|
|
* Must be called with JBUF_LOCK held, will release the LOCK when emitting the
|
|
* signal. The function returns GST_FLOW_ERROR when a parsing error happened and
|
|
* GST_FLOW_FLUSHING when the element is shutting down. On success
|
|
* GST_FLOW_OK is returned.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
|
|
guint8 pt)
|
|
{
|
|
GValue ret = { 0 };
|
|
GValue args[2] = { {0}, {0} };
|
|
GstCaps *caps;
|
|
gboolean res;
|
|
|
|
g_value_init (&args[0], GST_TYPE_ELEMENT);
|
|
g_value_set_object (&args[0], jitterbuffer);
|
|
g_value_init (&args[1], G_TYPE_UINT);
|
|
g_value_set_uint (&args[1], pt);
|
|
|
|
g_value_init (&ret, GST_TYPE_CAPS);
|
|
g_value_set_boxed (&ret, NULL);
|
|
|
|
JBUF_UNLOCK (jitterbuffer->priv);
|
|
g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
|
|
&ret);
|
|
JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
|
|
|
|
g_value_unset (&args[0]);
|
|
g_value_unset (&args[1]);
|
|
caps = (GstCaps *) g_value_dup_boxed (&ret);
|
|
g_value_unset (&ret);
|
|
if (!caps)
|
|
goto no_caps;
|
|
|
|
res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
|
|
gst_caps_unref (caps);
|
|
|
|
if (G_UNLIKELY (!res))
|
|
goto parse_failed;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
no_caps:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
out_flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
parse_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/* call with jbuf lock held */
|
|
static GstMessage *
|
|
check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstMessage *message = NULL;
|
|
|
|
if (percent == -1)
|
|
return NULL;
|
|
|
|
/* Post a buffering message */
|
|
if (priv->last_percent != percent) {
|
|
priv->last_percent = percent;
|
|
message =
|
|
gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
|
|
gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
|
|
}
|
|
|
|
return message;
|
|
}
|
|
|
|
/* call with jbuf lock held */
|
|
static GstMessage *
|
|
new_drop_message (GstRtpJitterBuffer * jitterbuffer, guint seqnum,
|
|
GstClockTime timestamp, DropMessageReason reason)
|
|
{
|
|
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstMessage *drop_msg = NULL;
|
|
GstStructure *s;
|
|
GstClockTime current_time;
|
|
GstClockTime time_diff;
|
|
const gchar *reason_str;
|
|
|
|
current_time = get_current_running_time (jitterbuffer);
|
|
time_diff = current_time - priv->last_drop_msg_timestamp;
|
|
|
|
if (reason == REASON_TOO_LATE) {
|
|
priv->num_too_late++;
|
|
reason_str = "too-late";
|
|
} else if (reason == REASON_DROP_ON_LATENCY) {
|
|
priv->num_drop_on_latency++;
|
|
reason_str = "drop-on-latency";
|
|
} else {
|
|
GST_WARNING_OBJECT (jitterbuffer, "Invalid reason for drop message");
|
|
return drop_msg;
|
|
}
|
|
|
|
/* Only create new drop_msg if time since last drop_msg is larger that
|
|
* that the set interval, or if it is the first drop message posted */
|
|
if ((time_diff >= priv->drop_messages_interval_ms * GST_MSECOND) ||
|
|
(priv->last_drop_msg_timestamp == GST_CLOCK_TIME_NONE)) {
|
|
|
|
s = gst_structure_new ("drop-msg",
|
|
"seqnum", G_TYPE_UINT, seqnum,
|
|
"timestamp", GST_TYPE_CLOCK_TIME, timestamp,
|
|
"reason", G_TYPE_STRING, reason_str,
|
|
"num-too-late", G_TYPE_UINT, priv->num_too_late,
|
|
"num-drop-on-latency", G_TYPE_UINT, priv->num_drop_on_latency, NULL);
|
|
|
|
priv->last_drop_msg_timestamp = current_time;
|
|
priv->num_too_late = 0;
|
|
priv->num_drop_on_latency = 0;
|
|
drop_msg = gst_message_new_element (GST_OBJECT (jitterbuffer), s);
|
|
}
|
|
return drop_msg;
|
|
}
|
|
|
|
|
|
static inline GstClockTimeDiff
|
|
timeout_offset (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
return priv->ts_offset + priv->out_offset + priv->latency_ns;
|
|
}
|
|
|
|
static inline GstClockTime
|
|
get_pts_timeout (const RtpTimer * timer)
|
|
{
|
|
if (timer->timeout == -1)
|
|
return -1;
|
|
|
|
return timer->timeout - timer->offset;
|
|
}
|
|
|
|
static inline gboolean
|
|
safe_add (guint64 * res, guint64 val, gint64 offset)
|
|
{
|
|
if (val <= G_MAXINT64) {
|
|
gint64 tmp = (gint64) val + offset;
|
|
if (tmp >= 0) {
|
|
*res = tmp;
|
|
return TRUE;
|
|
}
|
|
return FALSE;
|
|
}
|
|
/* From here, val > G_MAXINT64 */
|
|
|
|
/* Negative value */
|
|
if (offset < 0 && val < -offset)
|
|
return FALSE;
|
|
|
|
*res = val + offset;
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
update_timer_offsets (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
RtpTimer *test = rtp_timer_queue_peek_earliest (priv->timers);
|
|
GstClockTimeDiff new_offset = timeout_offset (jitterbuffer);
|
|
|
|
while (test) {
|
|
if (test->type != RTP_TIMER_EXPECTED) {
|
|
GstClockTime pts = get_pts_timeout (test);
|
|
if (safe_add (&test->timeout, pts, new_offset)) {
|
|
test->offset = new_offset;
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Invalidating timeout (pts lower than new offset)");
|
|
test->timeout = GST_CLOCK_TIME_NONE;
|
|
test->offset = 0;
|
|
}
|
|
}
|
|
|
|
rtp_timer_queue_reschedule (priv->timers, test);
|
|
test = rtp_timer_get_next (test);
|
|
}
|
|
}
|
|
|
|
static void
|
|
update_offset (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (priv->ts_offset_remainder != 0) {
|
|
GST_DEBUG ("adjustment %" G_GUINT64_FORMAT " remain %" G_GINT64_FORMAT
|
|
" off %" G_GINT64_FORMAT, priv->max_ts_offset_adjustment,
|
|
priv->ts_offset_remainder, priv->ts_offset);
|
|
if (ABS (priv->ts_offset_remainder) > priv->max_ts_offset_adjustment) {
|
|
if (priv->ts_offset_remainder > 0) {
|
|
priv->ts_offset += priv->max_ts_offset_adjustment;
|
|
priv->ts_offset_remainder -= priv->max_ts_offset_adjustment;
|
|
} else {
|
|
priv->ts_offset -= priv->max_ts_offset_adjustment;
|
|
priv->ts_offset_remainder += priv->max_ts_offset_adjustment;
|
|
}
|
|
} else {
|
|
priv->ts_offset += priv->ts_offset_remainder;
|
|
priv->ts_offset_remainder = 0;
|
|
}
|
|
|
|
update_timer_offsets (jitterbuffer);
|
|
}
|
|
}
|
|
|
|
static GstClockTime
|
|
apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (timestamp == -1)
|
|
return -1;
|
|
|
|
/* apply the timestamp offset, this is used for inter stream sync */
|
|
if (!safe_add (×tamp, timestamp, priv->ts_offset))
|
|
timestamp = 0;
|
|
/* add the offset, this is used when buffering */
|
|
timestamp += priv->out_offset;
|
|
|
|
return timestamp;
|
|
}
|
|
|
|
static void
|
|
unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
if (priv->clock_id) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
|
|
gst_clock_id_unschedule (priv->clock_id);
|
|
priv->clock_id = NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
update_current_timer (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
RtpTimer *timer;
|
|
|
|
timer = rtp_timer_queue_peek_earliest (priv->timers);
|
|
|
|
/* we never need to wakeup the timer thread when there is no more timers, if
|
|
* it was waiting on a clock id, it will simply do later and then wait on
|
|
* the conditions */
|
|
if (timer == NULL) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "no more timers");
|
|
return;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "waiting till %" GST_TIME_FORMAT
|
|
" and earliest timeout is at %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (priv->timer_timeout), GST_TIME_ARGS (timer->timeout));
|
|
|
|
/* wakeup the timer thread in case the timer queue was empty */
|
|
JBUF_SIGNAL_TIMER (priv);
|
|
|
|
/* no need to wait if the current wait is earlier or later */
|
|
if (timer->timeout != -1 && timer->timeout >= priv->timer_timeout)
|
|
return;
|
|
|
|
/* for other cases, force a reschedule of the timer thread */
|
|
unschedule_current_timer (jitterbuffer);
|
|
}
|
|
|
|
/* get the extra delay to wait before sending RTX */
|
|
static GstClockTime
|
|
get_rtx_delay (GstRtpJitterBufferPrivate * priv)
|
|
{
|
|
GstClockTime delay;
|
|
|
|
if (priv->rtx_delay == -1) {
|
|
/* the maximum delay for any RTX-packet is given by the latency, since
|
|
anything after that is considered lost. For various calulcations,
|
|
(given large avg_jitter and/or packet_spacing), the resulting delay
|
|
could exceed the configured latency, ending up issuing an RTX-request
|
|
that would never arrive in time. To help this we cap the delay
|
|
for any RTX with the last possible time it could still arrive in time. */
|
|
GstClockTime delay_max = (priv->latency_ns > priv->avg_rtx_rtt) ?
|
|
priv->latency_ns - priv->avg_rtx_rtt : priv->latency_ns;
|
|
|
|
if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
|
|
delay = DEFAULT_AUTO_RTX_DELAY;
|
|
} else {
|
|
/* jitter is in nanoseconds, maximum of 2x jitter and half the
|
|
* packet spacing is a good margin */
|
|
delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
|
|
}
|
|
|
|
delay = MIN (delay_max, delay);
|
|
} else {
|
|
delay = priv->rtx_delay * GST_MSECOND;
|
|
}
|
|
if (priv->rtx_min_delay > 0)
|
|
delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
|
|
|
|
return delay;
|
|
}
|
|
|
|
/* we just received a packet with seqnum and dts.
|
|
*
|
|
* First check for old seqnum that we are still expecting. If the gap with the
|
|
* current seqnum is too big, unschedule the timeouts.
|
|
*
|
|
* If we have a valid packet spacing estimate we can set a timer for when we
|
|
* should receive the next packet.
|
|
* If we don't have a valid estimate, we remove any timer we might have
|
|
* had for this packet.
|
|
*/
|
|
static void
|
|
update_rtx_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
|
|
GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum,
|
|
gboolean is_rtx, RtpTimer * timer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
gboolean is_stats_timer = FALSE;
|
|
|
|
if (timer && rtp_timer_queue_find (priv->rtx_stats_timers, timer->seqnum))
|
|
is_stats_timer = TRUE;
|
|
|
|
/* schedule immediatly expected timer which exceed the maximum RTX delay
|
|
* reorder configuration */
|
|
if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
|
|
RtpTimer *test = rtp_timer_queue_peek_earliest (priv->timers);
|
|
while (test) {
|
|
gint gap;
|
|
|
|
/* filter the timer type to speed up this loop */
|
|
if (test->type != RTP_TIMER_EXPECTED) {
|
|
test = rtp_timer_get_next (test);
|
|
continue;
|
|
}
|
|
|
|
gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
|
|
test->type, test->seqnum, seqnum, gap);
|
|
|
|
/* if this expected packet have a smaller gap then the configured one,
|
|
* then earlier timer are not expected to have bigger gap as the timer
|
|
* queue is ordered */
|
|
if (gap <= priv->rtx_delay_reorder)
|
|
break;
|
|
|
|
/* max gap, we exceeded the max reorder distance and we don't expect the
|
|
* missing packet to be this reordered */
|
|
if (test->num_rtx_retry == 0 && test->type == RTP_TIMER_EXPECTED)
|
|
rtp_timer_queue_update_timer (priv->timers, test, test->seqnum,
|
|
-1, 0, 0, FALSE);
|
|
|
|
test = rtp_timer_get_next (test);
|
|
}
|
|
}
|
|
|
|
do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
|
|
&& priv->rtx_next_seqnum;
|
|
|
|
if (timer && timer->type != RTP_TIMER_DEADLINE) {
|
|
if (timer->num_rtx_retry > 0) {
|
|
if (is_rtx) {
|
|
update_rtx_stats (jitterbuffer, timer, dts, TRUE);
|
|
/* don't try to estimate the next seqnum because this is a retransmitted
|
|
* packet and it probably did not arrive with the expected packet
|
|
* spacing. */
|
|
do_next_seqnum = FALSE;
|
|
}
|
|
|
|
if (!is_stats_timer && (!is_rtx || timer->num_rtx_retry > 1)) {
|
|
RtpTimer *stats_timer = rtp_timer_dup (timer);
|
|
/* Store timer in order to record stats when/if the retransmitted
|
|
* packet arrives. We should also store timer information if we've
|
|
* requested retransmission more than once since we may receive
|
|
* several retransmitted packets. For accuracy we should update the
|
|
* stats also when the redundant retransmitted packets arrives. */
|
|
stats_timer->timeout = pts + priv->rtx_stats_timeout * GST_MSECOND;
|
|
stats_timer->type = RTP_TIMER_EXPECTED;
|
|
rtp_timer_queue_insert (priv->rtx_stats_timers, stats_timer);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) {
|
|
GstClockTime next_expected_pts, delay;
|
|
|
|
/* calculate expected arrival time of the next seqnum */
|
|
next_expected_pts = pts + priv->packet_spacing;
|
|
|
|
delay = get_rtx_delay (priv);
|
|
|
|
/* and update/install timer for next seqnum */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, next_expected_pts %"
|
|
GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", est packet duration %"
|
|
GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
|
|
GST_TIME_ARGS (next_expected_pts), GST_TIME_ARGS (delay),
|
|
GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
|
|
|
|
if (timer && !is_stats_timer) {
|
|
timer->type = RTP_TIMER_EXPECTED;
|
|
rtp_timer_queue_update_timer (priv->timers, timer, priv->next_in_seqnum,
|
|
next_expected_pts, delay, 0, TRUE);
|
|
} else {
|
|
rtp_timer_queue_set_expected (priv->timers, priv->next_in_seqnum,
|
|
next_expected_pts, delay, priv->packet_spacing);
|
|
}
|
|
} else if (timer && timer->type != RTP_TIMER_DEADLINE && !is_stats_timer) {
|
|
/* if we had a timer, remove it, we don't know when to expect the next
|
|
* packet. */
|
|
rtp_timer_queue_unschedule (priv->timers, timer);
|
|
rtp_timer_free (timer);
|
|
}
|
|
}
|
|
|
|
static void
|
|
calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
|
|
GstClockTime pts)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
/* we need consecutive seqnums with a different
|
|
* rtptime to estimate the packet spacing. */
|
|
if (priv->ips_rtptime != rtptime) {
|
|
/* rtptime changed, check pts diff */
|
|
if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) {
|
|
GstClockTime new_packet_spacing = pts - priv->ips_pts;
|
|
GstClockTime old_packet_spacing = priv->packet_spacing;
|
|
|
|
/* Biased towards bigger packet spacings to prevent
|
|
* too many unneeded retransmission requests for next
|
|
* packets that just arrive a little later than we would
|
|
* expect */
|
|
if (old_packet_spacing > new_packet_spacing)
|
|
priv->packet_spacing =
|
|
(new_packet_spacing + 3 * old_packet_spacing) / 4;
|
|
else if (old_packet_spacing > 0)
|
|
priv->packet_spacing =
|
|
(3 * new_packet_spacing + old_packet_spacing) / 4;
|
|
else
|
|
priv->packet_spacing = new_packet_spacing;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"new packet spacing %" GST_TIME_FORMAT
|
|
" old packet spacing %" GST_TIME_FORMAT
|
|
" combined to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (new_packet_spacing),
|
|
GST_TIME_ARGS (old_packet_spacing),
|
|
GST_TIME_ARGS (priv->packet_spacing));
|
|
}
|
|
priv->ips_rtptime = rtptime;
|
|
priv->ips_pts = pts;
|
|
}
|
|
}
|
|
|
|
static void
|
|
insert_lost_event (GstRtpJitterBuffer * jitterbuffer,
|
|
guint16 seqnum, guint lost_packets, GstClockTime timestamp,
|
|
GstClockTime duration, guint num_rtx_retry)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstEvent *event = NULL;
|
|
guint next_in_seqnum;
|
|
|
|
/* we had a gap and thus we lost some packets. Create an event for this. */
|
|
if (lost_packets > 1)
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
|
|
seqnum + lost_packets - 1);
|
|
else
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
|
|
|
|
priv->num_lost += lost_packets;
|
|
priv->num_rtx_failed += num_rtx_retry;
|
|
|
|
next_in_seqnum = (seqnum + lost_packets) & 0xffff;
|
|
|
|
/* we now only accept seqnum bigger than this */
|
|
if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) {
|
|
priv->next_in_seqnum = next_in_seqnum;
|
|
priv->last_in_pts = timestamp;
|
|
}
|
|
|
|
/* Avoid creating events if we don't need it. Note that we still need to create
|
|
* the lost *ITEM* since it will be used to notify the outgoing thread of
|
|
* lost items (so that we can set discont flags and such) */
|
|
if (priv->do_lost) {
|
|
/* create packet lost event */
|
|
if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
|
|
duration = priv->packet_spacing;
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
|
|
gst_structure_new ("GstRTPPacketLost",
|
|
"seqnum", G_TYPE_UINT, (guint) seqnum,
|
|
"timestamp", G_TYPE_UINT64, timestamp,
|
|
"duration", G_TYPE_UINT64, duration,
|
|
"retry", G_TYPE_UINT, num_rtx_retry, NULL));
|
|
}
|
|
if (rtp_jitter_buffer_append_lost_event (priv->jbuf,
|
|
event, seqnum, lost_packets))
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_handle_missing_packets (GstRtpJitterBuffer * jitterbuffer,
|
|
guint32 missing_seqnum, guint16 current_seqnum, GstClockTime pts, gint gap,
|
|
GstClockTime now)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstClockTime est_pkt_duration, est_pts;
|
|
gboolean equidistant = priv->equidistant > 0;
|
|
GstClockTime last_in_pts = priv->last_in_pts;
|
|
GstClockTimeDiff offset = timeout_offset (jitterbuffer);
|
|
GstClockTime rtx_delay = get_rtx_delay (priv);
|
|
guint16 remaining_gap;
|
|
GstClockTimeDiff remaining_duration;
|
|
GstClockTimeDiff remainder_duration;
|
|
guint i;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Missing packets: (#%u->#%u), gap %d, pts %" GST_TIME_FORMAT
|
|
", last-pts %" GST_TIME_FORMAT,
|
|
missing_seqnum, current_seqnum - 1, gap, GST_TIME_ARGS (pts),
|
|
GST_TIME_ARGS (last_in_pts));
|
|
|
|
if (equidistant) {
|
|
GstClockTimeDiff total_duration;
|
|
gboolean too_late;
|
|
|
|
/* the total duration spanned by the missing packets */
|
|
total_duration = MAX (0, GST_CLOCK_DIFF (last_in_pts, pts));
|
|
|
|
/* interpolate between the current time and the last time based on
|
|
* number of packets we are missing, this is the estimated duration
|
|
* for the missing packet based on equidistant packet spacing. */
|
|
est_pkt_duration = total_duration / (gap + 1);
|
|
|
|
/* if we have valid packet-spacing, use that */
|
|
if (total_duration > 0 && priv->packet_spacing) {
|
|
est_pkt_duration = priv->packet_spacing;
|
|
}
|
|
|
|
est_pts = last_in_pts + est_pkt_duration;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "estimated missing packet pts %"
|
|
GST_TIME_FORMAT " and duration %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (est_pts), GST_TIME_ARGS (est_pkt_duration));
|
|
|
|
/* a packet is considered too late if our estimated pts plus all
|
|
applicable offsets are in the past */
|
|
too_late = now > (est_pts + offset);
|
|
|
|
/* Here we optimistically try to save any packets that could potentially
|
|
be saved by making sure we create lost/rtx timers for them, and for
|
|
the rest that could not possibly be saved, we create a "multi-lost"
|
|
event immediately containing the missing duration and sequence numbers */
|
|
if (too_late) {
|
|
guint lost_packets;
|
|
GstClockTime lost_duration;
|
|
GstClockTimeDiff gap_time;
|
|
guint max_saveable_packets = 0;
|
|
GstClockTime max_saveable_duration;
|
|
GstClockTime saveable_duration;
|
|
|
|
/* gap time represents the total duration of all missing packets */
|
|
gap_time = MAX (0, GST_CLOCK_DIFF (est_pts, pts));
|
|
|
|
/* based on the estimated packet duration, we
|
|
can figure out how many packets we could possibly save */
|
|
if (est_pkt_duration && offset > 0)
|
|
max_saveable_packets = offset / est_pkt_duration;
|
|
|
|
/* and say that the amount of lost packet is the sequence-number
|
|
gap minus these saveable packets, but at least 1 */
|
|
lost_packets = MAX (1, (gint) gap - (gint) max_saveable_packets);
|
|
|
|
/* now we know how many packets we can possibly save */
|
|
max_saveable_packets = gap - lost_packets;
|
|
|
|
/* we convert that to time */
|
|
max_saveable_duration = max_saveable_packets * est_pkt_duration;
|
|
|
|
/* determine the actual amount of time we can save */
|
|
saveable_duration = MIN (max_saveable_duration, gap_time);
|
|
|
|
/* and we now have the duration we need to fill */
|
|
lost_duration = GST_CLOCK_DIFF (saveable_duration, gap_time);
|
|
|
|
/* this multi-lost-packet event will be inserted directly into the packet-queue
|
|
for immediate processing */
|
|
if (lost_packets > 0) {
|
|
RtpTimer *timer;
|
|
GstClockTime timestamp = apply_offset (jitterbuffer, est_pts);
|
|
|
|
GST_INFO_OBJECT (jitterbuffer, "lost event for %d packet(s) (#%d->#%d) "
|
|
"for duration %" GST_TIME_FORMAT, lost_packets, missing_seqnum,
|
|
missing_seqnum + lost_packets - 1, GST_TIME_ARGS (lost_duration));
|
|
|
|
insert_lost_event (jitterbuffer, missing_seqnum, lost_packets,
|
|
timestamp, lost_duration, 0);
|
|
|
|
timer = rtp_timer_queue_find (priv->timers, missing_seqnum);
|
|
if (timer && timer->type != RTP_TIMER_DEADLINE) {
|
|
if (timer->queued)
|
|
rtp_timer_queue_unschedule (priv->timers, timer);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "removing timer for seqnum #%u",
|
|
missing_seqnum);
|
|
rtp_timer_free (timer);
|
|
}
|
|
|
|
missing_seqnum += lost_packets;
|
|
est_pts += lost_duration;
|
|
}
|
|
}
|
|
|
|
} else {
|
|
/* If we cannot assume equidistant packet spacing, the only thing we now
|
|
* for sure is that the missing packets have expected pts not later than
|
|
* the last received pts. */
|
|
est_pkt_duration = 0;
|
|
est_pts = pts;
|
|
}
|
|
|
|
/* Figure out how many more packets we are missing. */
|
|
remaining_gap = current_seqnum - missing_seqnum;
|
|
/* and how much time these packets represent */
|
|
remaining_duration = MAX (0, GST_CLOCK_DIFF (est_pts, pts));
|
|
/* Given the calculated packet-duration (packet spacing when equidistant),
|
|
the remainder is what we are left with after subtracting the ideal time
|
|
for the gap */
|
|
remainder_duration =
|
|
MAX (0, GST_CLOCK_DIFF (est_pkt_duration * remaining_gap,
|
|
remaining_duration));
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "remaining gap of %u, with "
|
|
"duration %" GST_TIME_FORMAT " gives remainder duration %"
|
|
GST_STIME_FORMAT, remaining_gap, GST_TIME_ARGS (remaining_duration),
|
|
GST_STIME_ARGS (remainder_duration));
|
|
|
|
for (i = 0; i < remaining_gap; i++) {
|
|
GstClockTime duration = est_pkt_duration;
|
|
/* we add the remainder on the first packet */
|
|
if (i == 0)
|
|
duration += remainder_duration;
|
|
|
|
/* clip duration to what is actually left */
|
|
remaining_duration = MAX (0, GST_CLOCK_DIFF (est_pts, pts));
|
|
duration = MIN (duration, remaining_duration);
|
|
|
|
if (priv->do_retransmission) {
|
|
RtpTimer *timer = rtp_timer_queue_find (priv->timers, missing_seqnum);
|
|
|
|
/* if we had a timer for the missing packet, update it. */
|
|
if (timer && timer->type == RTP_TIMER_EXPECTED) {
|
|
timer->duration = duration;
|
|
if (timer->timeout > (est_pts + rtx_delay) && timer->num_rtx_retry == 0) {
|
|
rtp_timer_queue_update_timer (priv->timers, timer, timer->seqnum,
|
|
est_pts, rtx_delay, 0, TRUE);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Update RTX timer(s) #%u, "
|
|
"pts %" GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT,
|
|
missing_seqnum, GST_TIME_ARGS (est_pts),
|
|
GST_TIME_ARGS (rtx_delay), GST_TIME_ARGS (duration));
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer(s) #%u, "
|
|
"pts %" GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT,
|
|
missing_seqnum, GST_TIME_ARGS (est_pts),
|
|
GST_TIME_ARGS (rtx_delay), GST_TIME_ARGS (duration));
|
|
rtp_timer_queue_set_expected (priv->timers, missing_seqnum, est_pts,
|
|
rtx_delay, duration);
|
|
}
|
|
} else {
|
|
GST_INFO_OBJECT (jitterbuffer,
|
|
"Add Lost timer for #%u, pts %" GST_TIME_FORMAT
|
|
", duration %" GST_TIME_FORMAT ", offset %" GST_STIME_FORMAT,
|
|
missing_seqnum, GST_TIME_ARGS (est_pts),
|
|
GST_TIME_ARGS (duration), GST_STIME_ARGS (offset));
|
|
rtp_timer_queue_set_lost (priv->timers, missing_seqnum, est_pts,
|
|
duration, offset);
|
|
}
|
|
|
|
missing_seqnum++;
|
|
est_pts += duration;
|
|
}
|
|
}
|
|
|
|
static void
|
|
calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
|
|
guint32 rtptime)
|
|
{
|
|
gint32 rtpdiff;
|
|
GstClockTimeDiff dtsdiff, rtpdiffns, diff;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
|
|
goto no_time;
|
|
|
|
if (priv->last_dts != -1)
|
|
dtsdiff = dts - priv->last_dts;
|
|
else
|
|
dtsdiff = 0;
|
|
|
|
if (priv->last_rtptime != -1)
|
|
rtpdiff = rtptime - (guint32) priv->last_rtptime;
|
|
else
|
|
rtpdiff = 0;
|
|
|
|
/* Guess whether stream currently uses equidistant packet spacing. If we
|
|
* often see identical timestamps it means the packets are not
|
|
* equidistant. */
|
|
if (rtptime == priv->last_rtptime)
|
|
priv->equidistant -= 2;
|
|
else
|
|
priv->equidistant += 1;
|
|
priv->equidistant = CLAMP (priv->equidistant, -7, 7);
|
|
|
|
priv->last_dts = dts;
|
|
priv->last_rtptime = rtptime;
|
|
|
|
if (rtpdiff > 0)
|
|
rtpdiffns =
|
|
gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
|
|
else
|
|
rtpdiffns =
|
|
-gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
|
|
|
|
diff = ABS (dtsdiff - rtpdiffns);
|
|
|
|
/* jitter is stored in nanoseconds */
|
|
priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
|
|
|
|
GST_LOG_OBJECT (jitterbuffer,
|
|
"dtsdiff %" GST_STIME_FORMAT " rtptime %" GST_STIME_FORMAT
|
|
", clock-rate %d, diff %" GST_STIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
|
|
GST_STIME_ARGS (dtsdiff), GST_STIME_ARGS (rtpdiffns), priv->clock_rate,
|
|
GST_STIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_time:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"no dts or no clock-rate, can't calculate jitter");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static gint
|
|
compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
|
|
{
|
|
GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
|
|
GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
|
|
guint seq_a, seq_b;
|
|
|
|
gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
|
|
seq_a = gst_rtp_buffer_get_seq (&rtp_a);
|
|
gst_rtp_buffer_unmap (&rtp_a);
|
|
|
|
gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
|
|
seq_b = gst_rtp_buffer_get_seq (&rtp_b);
|
|
gst_rtp_buffer_unmap (&rtp_b);
|
|
|
|
return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
|
|
}
|
|
|
|
static gboolean
|
|
handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer,
|
|
guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
guint gap_packets_length;
|
|
gboolean reset = FALSE;
|
|
gboolean future = gap > 0;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
|
|
GList *l;
|
|
guint32 prev_gap_seq = -1;
|
|
gboolean all_consecutive = TRUE;
|
|
|
|
g_queue_insert_sorted (&priv->gap_packets, buffer,
|
|
(GCompareDataFunc) compare_buffer_seqnum, NULL);
|
|
|
|
for (l = priv->gap_packets.head; l; l = l->next) {
|
|
GstBuffer *gap_buffer = l->data;
|
|
GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
|
|
guint32 gap_seq;
|
|
|
|
gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
|
|
|
|
all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
|
|
|
|
gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
|
|
if (prev_gap_seq == -1)
|
|
prev_gap_seq = gap_seq;
|
|
else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
|
|
all_consecutive = FALSE;
|
|
else
|
|
prev_gap_seq = gap_seq;
|
|
|
|
gst_rtp_buffer_unmap (&gap_rtp);
|
|
if (!all_consecutive)
|
|
break;
|
|
}
|
|
|
|
if (all_consecutive && gap_packets_length > 3) {
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"buffer too %s %d < %d, got 5 consecutive ones - reset",
|
|
(future ? "new" : "old"), gap,
|
|
(future ? max_dropout : -max_misorder));
|
|
reset = TRUE;
|
|
} else if (!all_consecutive) {
|
|
g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
|
|
g_queue_clear (&priv->gap_packets);
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"buffer too %s %d < %d, got no 5 consecutive ones - dropping",
|
|
(future ? "new" : "old"), gap,
|
|
(future ? max_dropout : -max_misorder));
|
|
buffer = NULL;
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"buffer too %s %d < %d, got %u consecutive ones - waiting",
|
|
(future ? "new" : "old"), gap,
|
|
(future ? max_dropout : -max_misorder), gap_packets_length + 1);
|
|
buffer = NULL;
|
|
}
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
|
|
gap, -max_misorder);
|
|
g_queue_push_tail (&priv->gap_packets, buffer);
|
|
buffer = NULL;
|
|
}
|
|
|
|
return reset;
|
|
}
|
|
|
|
static GstClockTime
|
|
get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
|
|
GstClockTime running_time = GST_CLOCK_TIME_NONE;
|
|
|
|
if (clock) {
|
|
GstClockTime base_time =
|
|
gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
|
|
GstClockTime clock_time = gst_clock_get_time (clock);
|
|
|
|
if (clock_time > base_time)
|
|
running_time = clock_time - base_time;
|
|
else
|
|
running_time = 0;
|
|
|
|
gst_object_unref (clock);
|
|
}
|
|
|
|
return running_time;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer,
|
|
GstPad * pad, GstObject * parent, guint16 seqnum)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GList *events = NULL, *l;
|
|
GList *buffers;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
|
|
rtp_jitter_buffer_flush (priv->jbuf,
|
|
(GFunc) free_item_and_retain_sticky_events, &events);
|
|
rtp_jitter_buffer_reset_skew (priv->jbuf);
|
|
rtp_timer_queue_remove_all (priv->timers);
|
|
priv->discont = TRUE;
|
|
priv->last_popped_seqnum = -1;
|
|
|
|
if (priv->gap_packets.head) {
|
|
GstBuffer *gap_buffer = priv->gap_packets.head->data;
|
|
GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
|
|
|
|
gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
|
|
priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
|
|
gst_rtp_buffer_unmap (&gap_rtp);
|
|
} else {
|
|
priv->next_seqnum = seqnum;
|
|
}
|
|
|
|
priv->last_in_pts = -1;
|
|
priv->next_in_seqnum = -1;
|
|
|
|
/* Insert all sticky events again in order, otherwise we would
|
|
* potentially loose STREAM_START, CAPS or SEGMENT events
|
|
*/
|
|
events = g_list_reverse (events);
|
|
for (l = events; l; l = l->next) {
|
|
rtp_jitter_buffer_append_event (priv->jbuf, l->data);
|
|
}
|
|
g_list_free (events);
|
|
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
|
|
/* reset spacing estimation when gap */
|
|
priv->ips_rtptime = -1;
|
|
priv->ips_pts = GST_CLOCK_TIME_NONE;
|
|
|
|
buffers = g_list_copy (priv->gap_packets.head);
|
|
g_queue_clear (&priv->gap_packets);
|
|
|
|
priv->ips_rtptime = -1;
|
|
priv->ips_pts = GST_CLOCK_TIME_NONE;
|
|
JBUF_UNLOCK (jitterbuffer->priv);
|
|
|
|
for (l = buffers; l; l = l->next) {
|
|
ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
|
|
l->data = NULL;
|
|
if (ret != GST_FLOW_OK) {
|
|
l = l->next;
|
|
break;
|
|
}
|
|
}
|
|
for (; l; l = l->next)
|
|
gst_buffer_unref (l->data);
|
|
g_list_free (buffers);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_fast_start (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
RTPJitterBufferItem *item;
|
|
RtpTimer *timer;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (priv->faststart_min_packets == 0)
|
|
return FALSE;
|
|
|
|
item = rtp_jitter_buffer_peek (priv->jbuf);
|
|
if (!item)
|
|
return FALSE;
|
|
|
|
timer = rtp_timer_queue_find (priv->timers, item->seqnum);
|
|
if (!timer || timer->type != RTP_TIMER_DEADLINE)
|
|
return FALSE;
|
|
|
|
if (rtp_jitter_buffer_can_fast_start (priv->jbuf,
|
|
priv->faststart_min_packets)) {
|
|
GST_INFO_OBJECT (jitterbuffer, "We found %i consecutive packet, start now",
|
|
priv->faststart_min_packets);
|
|
timer->timeout = -1;
|
|
rtp_timer_queue_reschedule (priv->timers, timer);
|
|
return TRUE;
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static GstClockTime
|
|
_get_inband_ntp_time (GstRtpJitterBuffer * jitterbuffer, GstRTPBuffer * rtp)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
guint8 *data;
|
|
guint size;
|
|
guint64 ntptime;
|
|
GstClockTime ntpnstime;
|
|
|
|
if (priv->ntp64_ext_id == 0)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
if (!gst_rtp_buffer_get_extension_onebyte_header (rtp, priv->ntp64_ext_id, 0,
|
|
(gpointer *) & data, &size)
|
|
&& !gst_rtp_buffer_get_extension_twobytes_header (rtp, NULL,
|
|
priv->ntp64_ext_id, 0, (gpointer *) & data, &size))
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
if (size != 8)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
ntptime = GST_READ_UINT64_BE (data);
|
|
ntpnstime =
|
|
gst_util_uint64_scale (ntptime, GST_SECOND, G_GUINT64_CONSTANT (1) << 32);
|
|
|
|
return ntpnstime;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
guint16 seqnum;
|
|
guint32 expected, rtptime;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstClockTime now;
|
|
GstClockTime dts, pts;
|
|
GstClockTime ntp_time;
|
|
GstClockTime inband_ntp_time;
|
|
guint64 latency_ts;
|
|
gboolean head;
|
|
gboolean duplicate;
|
|
gint percent = -1;
|
|
guint8 pt;
|
|
guint32 ssrc;
|
|
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
|
|
gboolean do_next_seqnum = FALSE;
|
|
GstMessage *msg = NULL;
|
|
GstMessage *drop_msg = NULL;
|
|
gboolean estimated_dts = FALSE;
|
|
gint32 packet_rate, max_dropout, max_misorder;
|
|
RtpTimer *timer = NULL;
|
|
gboolean is_rtx;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
|
|
goto invalid_buffer;
|
|
|
|
pt = gst_rtp_buffer_get_payload_type (&rtp);
|
|
seqnum = gst_rtp_buffer_get_seq (&rtp);
|
|
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
|
|
inband_ntp_time = _get_inband_ntp_time (jitterbuffer, &rtp);
|
|
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
is_rtx = GST_BUFFER_IS_RETRANSMISSION (buffer);
|
|
now = get_current_running_time (jitterbuffer);
|
|
|
|
/* make sure we have PTS and DTS set */
|
|
pts = GST_BUFFER_PTS (buffer);
|
|
dts = GST_BUFFER_DTS (buffer);
|
|
if (dts == -1)
|
|
dts = pts;
|
|
else if (pts == -1)
|
|
pts = dts;
|
|
|
|
if (dts == -1) {
|
|
/* If we have no DTS here, i.e. no capture time, get one from the
|
|
* clock now to have something to calculate with in the future. */
|
|
dts = now;
|
|
pts = dts;
|
|
|
|
/* Remember that we estimated the DTS if we are running already
|
|
* and this is not our first packet (or first packet after a reset).
|
|
* If it's the first packet, we somehow must generate a timestamp for
|
|
* everything, otherwise we can't calculate any times
|
|
*/
|
|
estimated_dts = (priv->next_in_seqnum != -1);
|
|
} else {
|
|
/* take the DTS of the buffer. This is the time when the packet was
|
|
* received and is used to calculate jitter and clock skew. We will adjust
|
|
* this DTS with the smoothed value after processing it in the
|
|
* jitterbuffer and assign it as the PTS. */
|
|
/* bring to running time */
|
|
dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Received packet #%d at time %" GST_TIME_FORMAT
|
|
", discont %d, rtx %d, inband NTP time %" GST_TIME_FORMAT, seqnum,
|
|
GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer), is_rtx,
|
|
GST_TIME_ARGS (inband_ntp_time));
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
|
|
if (G_UNLIKELY (priv->last_pt != pt)) {
|
|
GstCaps *caps;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
|
|
pt);
|
|
|
|
priv->last_pt = pt;
|
|
/* reset clock-rate so that we get a new one */
|
|
priv->clock_rate = -1;
|
|
|
|
priv->last_known_ext_rtptime = -1;
|
|
priv->last_known_ntpnstime = -1;
|
|
|
|
/* Try to get the clock-rate from the caps first if we can. If there are no
|
|
* caps we must fire the signal to get the clock-rate. */
|
|
if ((caps = gst_pad_get_current_caps (pad))) {
|
|
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
|
|
gst_caps_unref (caps);
|
|
}
|
|
}
|
|
|
|
if (G_UNLIKELY (priv->clock_rate == -1)) {
|
|
/* no clock rate given on the caps, try to get one with the signal */
|
|
if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
|
|
pt) == GST_FLOW_FLUSHING)
|
|
goto out_flushing;
|
|
|
|
if (G_UNLIKELY (priv->clock_rate == -1))
|
|
goto no_clock_rate;
|
|
|
|
gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
|
|
priv->last_known_ext_rtptime = -1;
|
|
priv->last_known_ntpnstime = -1;
|
|
}
|
|
|
|
if (G_UNLIKELY (priv->last_ssrc != ssrc)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "SSRC changed from %u to %u",
|
|
priv->last_ssrc, ssrc);
|
|
priv->last_ssrc = ssrc;
|
|
priv->last_known_ext_rtptime = -1;
|
|
priv->last_known_ntpnstime = -1;
|
|
}
|
|
|
|
/* don't accept more data on EOS */
|
|
if (G_UNLIKELY (priv->eos))
|
|
goto have_eos;
|
|
|
|
if (!is_rtx)
|
|
calculate_jitter (jitterbuffer, dts, rtptime);
|
|
|
|
if (priv->seqnum_base != -1) {
|
|
gint gap;
|
|
|
|
gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
|
|
|
|
if (gap < 0) {
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"packet seqnum #%d before seqnum-base #%d", seqnum,
|
|
priv->seqnum_base);
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
} else if (gap > 16384) {
|
|
/* From now on don't compare against the seqnum base anymore as
|
|
* at some point in the future we will wrap around and also that
|
|
* much reordering is very unlikely */
|
|
priv->seqnum_base = -1;
|
|
}
|
|
}
|
|
|
|
expected = priv->next_in_seqnum;
|
|
|
|
/* don't update packet-rate based on RTX, as those arrive highly unregularly */
|
|
if (!is_rtx) {
|
|
packet_rate = gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx,
|
|
seqnum, rtptime);
|
|
GST_TRACE_OBJECT (jitterbuffer, "updated packet_rate: %d", packet_rate);
|
|
}
|
|
max_dropout =
|
|
gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
|
|
priv->max_dropout_time);
|
|
max_misorder =
|
|
gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
|
|
priv->max_misorder_time);
|
|
GST_TRACE_OBJECT (jitterbuffer, "max_dropout: %d, max_misorder: %d",
|
|
max_dropout, max_misorder);
|
|
|
|
timer = rtp_timer_queue_find (priv->timers, seqnum);
|
|
if (is_rtx) {
|
|
if (G_UNLIKELY (!priv->do_retransmission))
|
|
goto unsolicited_rtx;
|
|
|
|
if (!timer)
|
|
timer = rtp_timer_queue_find (priv->rtx_stats_timers, seqnum);
|
|
|
|
/* If the first buffer is an (old) rtx, e.g. from before a reset, or
|
|
* already lost, ignore it */
|
|
if (!timer || expected == -1)
|
|
goto unsolicited_rtx;
|
|
}
|
|
|
|
/* now check against our expected seqnum */
|
|
if (G_UNLIKELY (expected == -1)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
|
|
|
|
/* calculate a pts based on rtptime and arrival time (dts) */
|
|
pts =
|
|
rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
|
|
rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)),
|
|
0, FALSE, &ntp_time);
|
|
|
|
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (pts))) {
|
|
/* A valid timestamp cannot be calculated, discard packet */
|
|
goto discard_invalid;
|
|
}
|
|
|
|
/* we don't know what the next_in_seqnum should be, wait for the last
|
|
* possible moment to push this buffer, maybe we get an earlier seqnum
|
|
* while we wait */
|
|
rtp_timer_queue_set_deadline (priv->timers, seqnum, pts,
|
|
timeout_offset (jitterbuffer));
|
|
|
|
do_next_seqnum = TRUE;
|
|
/* take rtptime and pts to calculate packet spacing */
|
|
priv->ips_rtptime = rtptime;
|
|
priv->ips_pts = pts;
|
|
|
|
} else {
|
|
gint gap;
|
|
/* now calculate gap */
|
|
gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
|
|
expected, seqnum, gap);
|
|
|
|
if (G_UNLIKELY (gap > 0 &&
|
|
rtp_timer_queue_length (priv->timers) >= max_dropout)) {
|
|
/* If we have timers for more than RTP_MAX_DROPOUT packets
|
|
* pending this means that we have a huge gap overall. We can
|
|
* reset the jitterbuffer at this point because there's
|
|
* just too much data missing to be able to do anything
|
|
* sensible with the past data. Just try again from the
|
|
* next packet */
|
|
GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting",
|
|
rtp_timer_queue_length (priv->timers), max_dropout);
|
|
g_queue_insert_sorted (&priv->gap_packets, buffer,
|
|
(GCompareDataFunc) compare_buffer_seqnum, NULL);
|
|
return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
|
|
}
|
|
|
|
/* Special handling of large gaps */
|
|
if (!is_rtx && ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout))) {
|
|
gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum,
|
|
gap, max_dropout, max_misorder);
|
|
if (reset) {
|
|
return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Had big gap, waiting for more consecutive packets");
|
|
goto finished;
|
|
}
|
|
}
|
|
|
|
/* We had no huge gap, let's drop all the gap packets */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
|
|
g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
|
|
g_queue_clear (&priv->gap_packets);
|
|
|
|
/* calculate a pts based on rtptime and arrival time (dts) */
|
|
/* If we estimated the DTS, don't consider it in the clock skew calculations */
|
|
pts =
|
|
rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
|
|
rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)),
|
|
gap, is_rtx, &ntp_time);
|
|
|
|
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (pts))) {
|
|
/* A valid timestamp cannot be calculated, discard packet */
|
|
goto discard_invalid;
|
|
}
|
|
|
|
if (G_LIKELY (gap == 0)) {
|
|
/* packet is expected */
|
|
calculate_packet_spacing (jitterbuffer, rtptime, pts);
|
|
do_next_seqnum = TRUE;
|
|
} else {
|
|
|
|
/* we have a gap */
|
|
if (gap > 0) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
|
|
/* fill in the gap with EXPECTED timers */
|
|
gst_rtp_jitter_buffer_handle_missing_packets (jitterbuffer, expected,
|
|
seqnum, pts, gap, now);
|
|
do_next_seqnum = TRUE;
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
|
|
do_next_seqnum = FALSE;
|
|
|
|
/* If an out of order packet arrives before its lost timer has expired
|
|
* remove it to avoid false positive statistics. If this is an RTX
|
|
* packet then the timer will be updated later as part of update_rtx_timers() */
|
|
if (!is_rtx && timer && timer->type == RTP_TIMER_LOST) {
|
|
rtp_timer_queue_unschedule (priv->timers, timer);
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"removing lost timer for late seqnum #%u", seqnum);
|
|
rtp_timer_free (g_steal_pointer (&timer));
|
|
}
|
|
}
|
|
|
|
/* reset spacing estimation when gap */
|
|
priv->ips_rtptime = -1;
|
|
priv->ips_pts = GST_CLOCK_TIME_NONE;
|
|
}
|
|
}
|
|
|
|
if (do_next_seqnum) {
|
|
priv->last_in_pts = pts;
|
|
priv->next_in_seqnum = (seqnum + 1) & 0xffff;
|
|
}
|
|
|
|
if (inband_ntp_time != GST_CLOCK_TIME_NONE) {
|
|
guint64 ext_rtptime;
|
|
|
|
ext_rtptime = priv->jbuf->ext_rtptime;
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
|
|
|
|
priv->last_known_ext_rtptime = ext_rtptime;
|
|
priv->last_known_ntpnstime = inband_ntp_time;
|
|
}
|
|
|
|
if (is_rtx) {
|
|
/* For RTX there must be a corresponding timer or it would be an
|
|
* unsolicited RTX packet that would be dropped */
|
|
g_assert (timer != NULL);
|
|
timer->num_rtx_received++;
|
|
}
|
|
|
|
/* At 2^15, we would detect a seqnum rollover too early, therefore
|
|
* limit the queue size. But let's not limit it to a number that is
|
|
* too small to avoid emptying it needlessly if there is a spurious huge
|
|
* sequence number, let's allow at least 10k packets in any case. */
|
|
while (rtp_jitter_buffer_is_full (priv->jbuf) &&
|
|
priv->srcresult == GST_FLOW_OK) {
|
|
RtpTimer *earliest_timer = rtp_timer_queue_peek_earliest (priv->timers);
|
|
while (earliest_timer) {
|
|
earliest_timer->timeout = -1;
|
|
if (earliest_timer->type == RTP_TIMER_DEADLINE)
|
|
break;
|
|
earliest_timer = rtp_timer_get_next (earliest_timer);
|
|
}
|
|
|
|
update_current_timer (jitterbuffer);
|
|
JBUF_WAIT_QUEUE (priv);
|
|
if (priv->srcresult != GST_FLOW_OK)
|
|
goto out_flushing;
|
|
}
|
|
|
|
/* let's check if this buffer is too late, we can only accept packets with
|
|
* bigger seqnum than the one we last pushed. */
|
|
if (G_LIKELY (priv->last_popped_seqnum != -1)) {
|
|
gint gap;
|
|
|
|
gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
|
|
|
|
/* priv->last_popped_seqnum >= seqnum, we're too late. */
|
|
if (G_UNLIKELY (gap <= 0)) {
|
|
if (priv->do_retransmission) {
|
|
if (is_rtx) {
|
|
/* For RTX there must be a corresponding timer or it would be an
|
|
* unsolicited RTX packet that would be dropped */
|
|
g_assert (timer != NULL);
|
|
|
|
update_rtx_stats (jitterbuffer, timer, dts, FALSE);
|
|
/* Only count the retranmitted packet too late if it has been
|
|
* considered lost. If the original packet arrived before the
|
|
* retransmitted we just count it as a duplicate. */
|
|
if (timer->type != RTP_TIMER_LOST)
|
|
goto rtx_duplicate;
|
|
}
|
|
}
|
|
goto too_late;
|
|
}
|
|
}
|
|
|
|
/* let's drop oldest packet if the queue is already full and drop-on-latency
|
|
* is set. We can only do this when there actually is a latency. When no
|
|
* latency is set, we just pump it in the queue and let the other end push it
|
|
* out as fast as possible. */
|
|
if (priv->latency_ms && priv->drop_on_latency) {
|
|
latency_ts =
|
|
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
|
|
|
|
if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
|
|
RTPJitterBufferItem *old_item;
|
|
|
|
old_item = rtp_jitter_buffer_peek (priv->jbuf);
|
|
|
|
if (IS_DROPABLE (old_item)) {
|
|
old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
|
|
old_item);
|
|
priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff;
|
|
if (priv->post_drop_messages) {
|
|
drop_msg =
|
|
new_drop_message (jitterbuffer, old_item->seqnum, old_item->pts,
|
|
REASON_DROP_ON_LATENCY);
|
|
}
|
|
rtp_jitter_buffer_free_item (old_item);
|
|
}
|
|
/* we might have removed some head buffers, signal the pushing thread to
|
|
* see if it can push now */
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
}
|
|
// If we can calculate a NTP time based solely on the Sender Report, or
|
|
// inband NTP header extension do that so that we can still add a reference
|
|
// timestamp meta to the buffer
|
|
if (!GST_CLOCK_TIME_IS_VALID (ntp_time) &&
|
|
GST_CLOCK_TIME_IS_VALID (priv->last_known_ntpnstime) &&
|
|
priv->last_known_ext_rtptime != -1) {
|
|
guint64 ext_time = priv->last_known_ext_rtptime;
|
|
|
|
ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtptime);
|
|
|
|
if (ext_time >= priv->last_known_ext_rtptime) {
|
|
ntp_time =
|
|
priv->last_known_ntpnstime + gst_util_uint64_scale (ext_time -
|
|
priv->last_known_ext_rtptime, GST_SECOND, priv->clock_rate);
|
|
} else {
|
|
ntp_time =
|
|
priv->last_known_ntpnstime -
|
|
gst_util_uint64_scale (priv->last_known_ext_rtptime - ext_time,
|
|
GST_SECOND, priv->clock_rate);
|
|
}
|
|
}
|
|
|
|
if (priv->add_reference_timestamp_meta && GST_CLOCK_TIME_IS_VALID (ntp_time)
|
|
&& priv->reference_timestamp_caps != NULL) {
|
|
buffer = gst_buffer_make_writable (buffer);
|
|
|
|
GST_TRACE_OBJECT (jitterbuffer,
|
|
"adding NTP time reference meta: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (ntp_time));
|
|
|
|
gst_buffer_add_reference_timestamp_meta (buffer,
|
|
priv->reference_timestamp_caps, ntp_time, GST_CLOCK_TIME_NONE);
|
|
}
|
|
|
|
/* If we estimated the DTS, don't consider it in the clock skew calculations
|
|
* later. The code above always sets dts to pts or the other way around if
|
|
* any of those is valid in the buffer, so we know that if we estimated the
|
|
* dts that both are unknown */
|
|
head = rtp_jitter_buffer_append_buffer (priv->jbuf, buffer,
|
|
estimated_dts ? GST_CLOCK_TIME_NONE : dts, pts, seqnum, rtptime,
|
|
&duplicate, &percent);
|
|
|
|
/* now insert the packet into the queue in sorted order. This function returns
|
|
* FALSE if a packet with the same seqnum was already in the queue, meaning we
|
|
* have a duplicate. */
|
|
if (G_UNLIKELY (duplicate)) {
|
|
if (is_rtx) {
|
|
/* For RTX there must be a corresponding timer or it would be an
|
|
* unsolicited RTX packet that would be dropped */
|
|
g_assert (timer != NULL);
|
|
update_rtx_stats (jitterbuffer, timer, dts, FALSE);
|
|
}
|
|
goto duplicate;
|
|
}
|
|
|
|
/* Trigger fast start if needed */
|
|
if (gst_rtp_jitter_buffer_fast_start (jitterbuffer))
|
|
head = TRUE;
|
|
|
|
/* update rtx timers */
|
|
if (priv->do_retransmission)
|
|
update_rtx_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum, is_rtx,
|
|
g_steal_pointer (&timer));
|
|
|
|
/* we had an unhandled SR, handle it now */
|
|
if (priv->last_sr)
|
|
do_handle_sync (jitterbuffer);
|
|
|
|
if (inband_ntp_time != GST_CLOCK_TIME_NONE)
|
|
do_handle_sync_inband (jitterbuffer, inband_ntp_time);
|
|
|
|
if (G_UNLIKELY (head)) {
|
|
/* signal addition of new buffer when the _loop is waiting. */
|
|
if (G_LIKELY (priv->active))
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
|
|
rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
|
|
|
|
msg = check_buffering_percent (jitterbuffer, percent);
|
|
|
|
finished:
|
|
update_current_timer (jitterbuffer);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
if (msg)
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
|
|
if (drop_msg)
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), drop_msg);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
invalid_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received invalid RTP payload, dropping"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_clock_rate:
|
|
{
|
|
GST_WARNING_OBJECT (jitterbuffer,
|
|
"No clock-rate in caps!, dropping buffer");
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
out_flushing:
|
|
{
|
|
ret = priv->srcresult;
|
|
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
have_eos:
|
|
{
|
|
ret = GST_FLOW_EOS;
|
|
GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
too_late:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
|
|
" popped, dropping", seqnum, priv->last_popped_seqnum);
|
|
priv->num_late++;
|
|
if (priv->post_drop_messages) {
|
|
drop_msg = new_drop_message (jitterbuffer, seqnum, pts, REASON_TOO_LATE);
|
|
}
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
duplicate:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
|
|
seqnum);
|
|
priv->num_duplicates++;
|
|
goto finished;
|
|
}
|
|
rtx_duplicate:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Duplicate RTX packet #%d detected, dropping", seqnum);
|
|
priv->num_duplicates++;
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
unsolicited_rtx:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Unsolicited RTX packet #%d detected, dropping", seqnum);
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
discard_invalid:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"cannot calculate a valid pts for #%d (rtx: %d), discard",
|
|
seqnum, is_rtx);
|
|
gst_buffer_unref (buffer);
|
|
goto finished;
|
|
}
|
|
}
|
|
|
|
/* FIXME: hopefully we can do something more efficient here, especially when
|
|
* all packets are in order and/or outside of the currently cached range.
|
|
* Still worthwhile to have it, avoids taking/releasing object lock and pad
|
|
* stream lock for every single buffer in the default chain_list fallback. */
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_chain_list (GstPad * pad, GstObject * parent,
|
|
GstBufferList * buffer_list)
|
|
{
|
|
GstFlowReturn flow_ret = GST_FLOW_OK;
|
|
guint i, n;
|
|
|
|
n = gst_buffer_list_length (buffer_list);
|
|
for (i = 0; i < n; ++i) {
|
|
GstBuffer *buf = gst_buffer_list_get (buffer_list, i);
|
|
|
|
flow_ret = gst_rtp_jitter_buffer_chain (pad, parent, gst_buffer_ref (buf));
|
|
|
|
if (flow_ret != GST_FLOW_OK)
|
|
break;
|
|
}
|
|
gst_buffer_list_unref (buffer_list);
|
|
|
|
return flow_ret;
|
|
}
|
|
|
|
static GstClockTime
|
|
compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
|
|
{
|
|
guint64 ext_time, elapsed;
|
|
guint32 rtp_time;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
priv = jitterbuffer->priv;
|
|
rtp_time = item->rtptime;
|
|
|
|
GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
|
|
G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
|
|
|
|
ext_time = priv->ext_timestamp;
|
|
ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
|
|
if (ext_time < priv->ext_timestamp) {
|
|
ext_time = priv->ext_timestamp;
|
|
} else {
|
|
priv->ext_timestamp = ext_time;
|
|
}
|
|
|
|
if (ext_time > priv->clock_base)
|
|
elapsed = ext_time - priv->clock_base;
|
|
else
|
|
elapsed = 0;
|
|
|
|
elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
|
|
return elapsed;
|
|
}
|
|
|
|
static void
|
|
update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
|
|
RTPJitterBufferItem * item)
|
|
{
|
|
guint64 total, elapsed, left, estimated;
|
|
GstClockTime out_time;
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
if (priv->npt_stop == -1 || priv->ext_timestamp == -1
|
|
|| priv->clock_base == -1 || priv->clock_rate <= 0)
|
|
return;
|
|
|
|
/* compute the elapsed time */
|
|
elapsed = compute_elapsed (jitterbuffer, item);
|
|
|
|
/* do nothing if elapsed time doesn't increment */
|
|
if (priv->last_elapsed && elapsed <= priv->last_elapsed)
|
|
return;
|
|
|
|
priv->last_elapsed = elapsed;
|
|
|
|
/* this is the total time we need to play */
|
|
total = priv->npt_stop - priv->npt_start;
|
|
GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (total));
|
|
|
|
/* this is how much time there is left */
|
|
if (total > elapsed)
|
|
left = total - elapsed;
|
|
else
|
|
left = 0;
|
|
|
|
/* if we have less time left that the size of the buffer, we will not
|
|
* be able to keep it filled, disabled buffering then */
|
|
if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
|
|
", disable buffering close to EOS", GST_TIME_ARGS (left));
|
|
rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
|
|
}
|
|
|
|
/* this is the current time as running-time */
|
|
out_time = item->pts;
|
|
|
|
if (elapsed > 0)
|
|
estimated = gst_util_uint64_scale (out_time, total, elapsed);
|
|
else {
|
|
/* if there is almost nothing left,
|
|
* we may never advance enough to end up in the above case */
|
|
if (total < GST_SECOND)
|
|
estimated = GST_SECOND;
|
|
else
|
|
estimated = -1;
|
|
}
|
|
GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
|
|
|
|
if (estimated != -1 && priv->estimated_eos != estimated) {
|
|
rtp_timer_queue_set_eos (priv->timers, estimated,
|
|
timeout_offset (jitterbuffer));
|
|
priv->estimated_eos = estimated;
|
|
}
|
|
}
|
|
|
|
/* take a buffer from the queue and push it */
|
|
static GstFlowReturn
|
|
pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
RTPJitterBufferItem *item;
|
|
GstBuffer *outbuf = NULL;
|
|
GstEvent *outevent = NULL;
|
|
GstQuery *outquery = NULL;
|
|
GstClockTime dts, pts;
|
|
gint percent = -1;
|
|
gboolean do_push = TRUE;
|
|
guint type;
|
|
GstMessage *msg;
|
|
|
|
/* when we get here we are ready to pop and push the buffer */
|
|
item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
|
|
type = item->type;
|
|
|
|
switch (type) {
|
|
case ITEM_TYPE_BUFFER:
|
|
|
|
/* we need to make writable to change the flags and timestamps */
|
|
outbuf = gst_buffer_make_writable (item->data);
|
|
|
|
if (G_UNLIKELY (priv->discont)) {
|
|
/* set DISCONT flag when we missed a packet. We pushed the buffer writable
|
|
* into the jitterbuffer so we can modify now. */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
|
priv->discont = FALSE;
|
|
}
|
|
if (G_UNLIKELY (priv->ts_discont)) {
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
|
|
priv->ts_discont = FALSE;
|
|
}
|
|
|
|
dts =
|
|
gst_segment_position_from_running_time (&priv->segment,
|
|
GST_FORMAT_TIME, item->dts);
|
|
pts =
|
|
gst_segment_position_from_running_time (&priv->segment,
|
|
GST_FORMAT_TIME, item->pts);
|
|
|
|
/* if this is a new frame, check if ts_offset needs to be updated */
|
|
if (pts != priv->last_pts) {
|
|
update_offset (jitterbuffer);
|
|
}
|
|
|
|
/* apply timestamp with offset to buffer now */
|
|
GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
|
|
GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
|
|
|
|
/* update the elapsed time when we need to check against the npt stop time. */
|
|
update_estimated_eos (jitterbuffer, item);
|
|
|
|
priv->last_pts = pts;
|
|
priv->last_out_time = GST_BUFFER_PTS (outbuf);
|
|
break;
|
|
case ITEM_TYPE_LOST:
|
|
priv->discont = TRUE;
|
|
if (!priv->do_lost)
|
|
do_push = FALSE;
|
|
/* FALLTHROUGH */
|
|
case ITEM_TYPE_EVENT:
|
|
outevent = item->data;
|
|
break;
|
|
case ITEM_TYPE_QUERY:
|
|
outquery = item->data;
|
|
break;
|
|
}
|
|
|
|
/* now we are ready to push the buffer. Save the seqnum and release the lock
|
|
* so the other end can push stuff in the queue again. */
|
|
if (seqnum != -1) {
|
|
priv->last_popped_seqnum = seqnum;
|
|
priv->next_seqnum = (seqnum + item->count) & 0xffff;
|
|
}
|
|
msg = check_buffering_percent (jitterbuffer, percent);
|
|
|
|
if (type == ITEM_TYPE_EVENT && outevent &&
|
|
GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
|
|
g_assert (priv->eos);
|
|
while (rtp_timer_queue_length (priv->timers) > 0) {
|
|
/* Stopping timers */
|
|
unschedule_current_timer (jitterbuffer);
|
|
JBUF_WAIT_TIMER_CHECK (priv, out_flushing_wait);
|
|
}
|
|
}
|
|
|
|
JBUF_UNLOCK (priv);
|
|
|
|
item->data = NULL;
|
|
rtp_jitter_buffer_free_item (item);
|
|
|
|
if (msg)
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
|
|
|
|
switch (type) {
|
|
case ITEM_TYPE_BUFFER:
|
|
/* push buffer */
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
|
|
seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
|
|
GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
|
|
priv->num_pushed++;
|
|
GST_BUFFER_DTS (outbuf) = GST_CLOCK_TIME_NONE;
|
|
result = gst_pad_push (priv->srcpad, outbuf);
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
break;
|
|
case ITEM_TYPE_LOST:
|
|
case ITEM_TYPE_EVENT:
|
|
/* We got not enough consecutive packets with a huge gap, we can
|
|
* as well just drop them here now on EOS */
|
|
if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
|
|
g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
|
|
g_queue_clear (&priv->gap_packets);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
|
|
", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
|
|
|
|
if (do_push)
|
|
gst_pad_push_event (priv->srcpad, outevent);
|
|
else if (outevent)
|
|
gst_event_unref (outevent);
|
|
|
|
result = GST_FLOW_OK;
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
break;
|
|
case ITEM_TYPE_QUERY:
|
|
{
|
|
gboolean res;
|
|
|
|
res = gst_pad_peer_query (priv->srcpad, outquery);
|
|
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
result = GST_FLOW_OK;
|
|
GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
|
|
JBUF_SIGNAL_QUERY (priv, res);
|
|
break;
|
|
}
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
out_flushing:
|
|
{
|
|
return priv->srcresult;
|
|
}
|
|
|
|
out_flushing_wait:
|
|
{
|
|
rtp_jitter_buffer_free_item (item);
|
|
return priv->srcresult;
|
|
}
|
|
}
|
|
|
|
#define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
|
|
|
|
/* Peek a buffer and compare the seqnum to the expected seqnum.
|
|
* If all is fine, the buffer is pushed.
|
|
* If something is wrong, we wait for some event
|
|
*/
|
|
static GstFlowReturn
|
|
handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstFlowReturn result;
|
|
RTPJitterBufferItem *item;
|
|
guint seqnum;
|
|
guint32 next_seqnum;
|
|
|
|
/* only push buffers when PLAYING and active and not buffering */
|
|
if (priv->blocked || !priv->active ||
|
|
rtp_jitter_buffer_is_buffering (priv->jbuf)) {
|
|
return GST_FLOW_WAIT;
|
|
}
|
|
|
|
/* peek a buffer, we're just looking at the sequence number.
|
|
* If all is fine, we'll pop and push it. If the sequence number is wrong we
|
|
* wait for a timeout or something to change.
|
|
* The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
|
|
item = rtp_jitter_buffer_peek (priv->jbuf);
|
|
if (item == NULL) {
|
|
goto wait;
|
|
}
|
|
|
|
/* get the seqnum and the next expected seqnum */
|
|
seqnum = item->seqnum;
|
|
if (seqnum == -1) {
|
|
return pop_and_push_next (jitterbuffer, seqnum);
|
|
}
|
|
|
|
next_seqnum = priv->next_seqnum;
|
|
|
|
/* get the gap between this and the previous packet. If we don't know the
|
|
* previous packet seqnum assume no gap. */
|
|
if (G_UNLIKELY (next_seqnum == -1)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
|
|
/* we don't know what the next_seqnum should be, the chain function should
|
|
* have scheduled a DEADLINE timer that will increment next_seqnum when it
|
|
* fires, so wait for that */
|
|
result = GST_FLOW_WAIT;
|
|
} else {
|
|
gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
|
|
|
|
if (G_LIKELY (gap == 0)) {
|
|
/* no missing packet, pop and push */
|
|
result = pop_and_push_next (jitterbuffer, seqnum);
|
|
} else if (G_UNLIKELY (gap < 0)) {
|
|
/* if we have a packet that we already pushed or considered dropped, pop it
|
|
* off and get the next packet */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
|
|
seqnum, next_seqnum);
|
|
item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
|
|
rtp_jitter_buffer_free_item (item);
|
|
result = GST_FLOW_OK;
|
|
} else {
|
|
/* the chain function has scheduled timers to request retransmission or
|
|
* when to consider the packet lost, wait for that */
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"Sequence number GAP detected: expected %d instead of %d (%d missing)",
|
|
next_seqnum, seqnum, gap);
|
|
/* if we have reached EOS, just keep processing */
|
|
/* Also do the same if we block input because the JB is full */
|
|
if (priv->eos || rtp_jitter_buffer_is_full (priv->jbuf)) {
|
|
result = pop_and_push_next (jitterbuffer, seqnum);
|
|
result = GST_FLOW_OK;
|
|
} else {
|
|
result = GST_FLOW_WAIT;
|
|
}
|
|
}
|
|
}
|
|
|
|
return result;
|
|
|
|
wait:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
|
|
if (priv->eos) {
|
|
return GST_FLOW_EOS;
|
|
} else {
|
|
return GST_FLOW_WAIT;
|
|
}
|
|
}
|
|
}
|
|
|
|
static GstClockTime
|
|
get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
|
|
{
|
|
GstClockTime rtx_retry_timeout;
|
|
GstClockTime rtx_min_retry_timeout;
|
|
|
|
if (priv->rtx_retry_timeout == -1) {
|
|
if (priv->avg_rtx_rtt == 0)
|
|
rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
|
|
else
|
|
/* we want to ask for a retransmission after we waited for a
|
|
* complete RTT and the additional jitter */
|
|
rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
|
|
} else {
|
|
rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
|
|
}
|
|
/* make sure we don't retry too often. On very low latency networks,
|
|
* the RTT and jitter can be very low. */
|
|
if (priv->rtx_min_retry_timeout == -1) {
|
|
rtx_min_retry_timeout = priv->packet_spacing;
|
|
} else {
|
|
rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
|
|
}
|
|
rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
|
|
|
|
return rtx_retry_timeout;
|
|
}
|
|
|
|
static GstClockTime
|
|
get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
|
|
GstClockTime rtx_retry_timeout)
|
|
{
|
|
GstClockTime rtx_retry_period;
|
|
|
|
if (priv->rtx_retry_period == -1) {
|
|
/* we retry up to the configured jitterbuffer size but leaving some
|
|
* room for the retransmission to arrive in time */
|
|
if (rtx_retry_timeout > priv->latency_ns) {
|
|
rtx_retry_period = 0;
|
|
} else {
|
|
rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
|
|
}
|
|
} else {
|
|
rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
|
|
}
|
|
return rtx_retry_period;
|
|
}
|
|
|
|
/*
|
|
1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th)
|
|
2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th)
|
|
3. For very large measurements (> avg * 2), consider them "outliers"
|
|
and count them a lot less (1/48th)
|
|
*/
|
|
static void
|
|
update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt)
|
|
{
|
|
gint weight;
|
|
|
|
if (priv->avg_rtx_rtt == 0) {
|
|
priv->avg_rtx_rtt = rtt;
|
|
return;
|
|
}
|
|
|
|
if (rtt > 2 * priv->avg_rtx_rtt)
|
|
weight = 48;
|
|
else if (rtt > priv->avg_rtx_rtt)
|
|
weight = 8;
|
|
else
|
|
weight = 16;
|
|
|
|
priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight;
|
|
}
|
|
|
|
static void
|
|
update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, const RtpTimer * timer,
|
|
GstClockTime dts, gboolean success)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstClockTime delay;
|
|
|
|
if (success) {
|
|
/* we scheduled a retry for this packet and now we have it */
|
|
priv->num_rtx_success++;
|
|
/* all the previous retry attempts failed */
|
|
priv->num_rtx_failed += timer->num_rtx_retry - 1;
|
|
} else {
|
|
/* All retries failed or was too late */
|
|
priv->num_rtx_failed += timer->num_rtx_retry;
|
|
}
|
|
|
|
/* number of retries before (hopefully) receiving the packet */
|
|
if (priv->avg_rtx_num == 0.0)
|
|
priv->avg_rtx_num = timer->num_rtx_retry;
|
|
else
|
|
priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
|
|
|
|
/* Calculate the delay between retransmission request and receiving this
|
|
* packet. We have a valid delay if and only if this packet is a response to
|
|
* our last request. If not we don't know if this is a response to an
|
|
* earlier request and delay could be way off. For RTT is more important
|
|
* with correct values than to update for every packet. */
|
|
if (timer->num_rtx_retry == timer->num_rtx_received &&
|
|
dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
|
|
delay = dts - timer->rtx_last;
|
|
update_avg_rtx_rtt (priv, delay);
|
|
} else {
|
|
delay = 0;
|
|
}
|
|
|
|
GST_LOG_OBJECT (jitterbuffer,
|
|
"RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
|
|
G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
|
|
G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
|
|
GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
|
|
priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
|
|
priv->avg_rtx_num, GST_TIME_ARGS (delay),
|
|
GST_TIME_ARGS (priv->avg_rtx_rtt));
|
|
}
|
|
|
|
/* the timeout for when we expected a packet expired */
|
|
static gboolean
|
|
do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer,
|
|
GstClockTime now, GQueue * events)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstEvent *event;
|
|
guint delay, delay_ms, avg_rtx_rtt_ms;
|
|
guint rtx_retry_timeout_ms, rtx_retry_period_ms;
|
|
guint rtx_deadline_ms;
|
|
GstClockTime rtx_retry_period;
|
|
GstClockTime rtx_retry_timeout;
|
|
GstClock *clock;
|
|
GstClockTimeDiff offset = 0;
|
|
GstClockTime timeout;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "expected #%d didn't arrive, now %"
|
|
GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
|
|
|
|
rtx_retry_timeout = get_rtx_retry_timeout (priv);
|
|
rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
|
|
|
|
/* delay expresses how late this packet is currently */
|
|
delay = now - timer->rtx_base;
|
|
|
|
delay_ms = GST_TIME_AS_MSECONDS (delay);
|
|
rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
|
|
rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
|
|
avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
|
|
rtx_deadline_ms =
|
|
priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
|
|
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
|
|
gst_structure_new ("GstRTPRetransmissionRequest",
|
|
"seqnum", G_TYPE_UINT, (guint) timer->seqnum,
|
|
"running-time", G_TYPE_UINT64, timer->rtx_base,
|
|
"delay", G_TYPE_UINT, delay_ms,
|
|
"retry", G_TYPE_UINT, timer->num_rtx_retry,
|
|
"frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
|
|
"period", G_TYPE_UINT, rtx_retry_period_ms,
|
|
"deadline", G_TYPE_UINT, rtx_deadline_ms,
|
|
"packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
|
|
"avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
|
|
g_queue_push_tail (events, event);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
|
|
|
|
priv->num_rtx_requests++;
|
|
timer->num_rtx_retry++;
|
|
|
|
GST_OBJECT_LOCK (jitterbuffer);
|
|
if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
|
|
timer->rtx_last = gst_clock_get_time (clock);
|
|
timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
|
|
} else {
|
|
timer->rtx_last = now;
|
|
}
|
|
GST_OBJECT_UNLOCK (jitterbuffer);
|
|
|
|
/*
|
|
Calculate the timeout for the next retransmission attempt:
|
|
We have just successfully sent one RTX request, and we need to
|
|
find out when to schedule the next one.
|
|
|
|
The rtx_retry_timeout tells us the logical timeout between RTX
|
|
requests based on things like round-trip time, jitter and packet spacing,
|
|
and is how long we are going to wait before attempting another RTX packet
|
|
*/
|
|
timeout = timer->rtx_last + rtx_retry_timeout;
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"timer #%i new timeout %" GST_TIME_FORMAT ", rtx retry timeout %"
|
|
GST_TIME_FORMAT ", num_retry %u", timer->seqnum, GST_TIME_ARGS (timeout),
|
|
GST_TIME_ARGS (rtx_retry_timeout), timer->num_rtx_retry);
|
|
if ((priv->rtx_max_retries != -1
|
|
&& timer->num_rtx_retry >= priv->rtx_max_retries)
|
|
|| (timeout > timer->rtx_base + rtx_retry_period)) {
|
|
/* too many retransmission request, we now convert the timer
|
|
* to a lost timer, leave the num_rtx_retry as it is for stats */
|
|
timer->type = RTP_TIMER_LOST;
|
|
timeout = timer->rtx_base;
|
|
offset = timeout_offset (jitterbuffer);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "reschedule #%i as LOST timer for %"
|
|
GST_TIME_FORMAT, timer->seqnum,
|
|
GST_TIME_ARGS (timer->rtx_base + offset));
|
|
}
|
|
rtp_timer_queue_update_timer (priv->timers, timer, timer->seqnum,
|
|
timeout, 0, offset, FALSE);
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
/* a packet is lost */
|
|
static gboolean
|
|
do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer,
|
|
GstClockTime now)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstClockTime timestamp;
|
|
|
|
timestamp = apply_offset (jitterbuffer, get_pts_timeout (timer));
|
|
insert_lost_event (jitterbuffer, timer->seqnum, 1, timestamp,
|
|
timer->duration, timer->num_rtx_retry);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
|
|
/* Store info to update stats if the packet arrives too late */
|
|
timer->timeout = now + priv->rtx_stats_timeout * GST_MSECOND;
|
|
timer->type = RTP_TIMER_LOST;
|
|
rtp_timer_queue_insert (priv->rtx_stats_timers, timer);
|
|
} else {
|
|
rtp_timer_free (timer);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer,
|
|
GstClockTime now)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
|
|
rtp_timer_free (timer);
|
|
if (!priv->eos) {
|
|
GstEvent *event;
|
|
|
|
/* there was no EOS in the buffer, put one in there now */
|
|
event = gst_event_new_eos ();
|
|
if (priv->segment_seqnum != GST_SEQNUM_INVALID)
|
|
gst_event_set_seqnum (event, priv->segment_seqnum);
|
|
queue_event (jitterbuffer, event);
|
|
}
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer,
|
|
GstClockTime now)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
|
|
|
|
/* timer seqnum might have been obsoleted by caps seqnum-base,
|
|
* only mess with current ongoing seqnum if still unknown */
|
|
if (priv->next_seqnum == -1)
|
|
priv->next_seqnum = timer->seqnum;
|
|
rtp_timer_free (timer);
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
do_timeout (GstRtpJitterBuffer * jitterbuffer, RtpTimer * timer,
|
|
GstClockTime now, GQueue * events)
|
|
{
|
|
gboolean removed = FALSE;
|
|
|
|
switch (timer->type) {
|
|
case RTP_TIMER_EXPECTED:
|
|
removed = do_expected_timeout (jitterbuffer, timer, now, events);
|
|
break;
|
|
case RTP_TIMER_LOST:
|
|
removed = do_lost_timeout (jitterbuffer, timer, now);
|
|
break;
|
|
case RTP_TIMER_DEADLINE:
|
|
removed = do_deadline_timeout (jitterbuffer, timer, now);
|
|
break;
|
|
case RTP_TIMER_EOS:
|
|
removed = do_eos_timeout (jitterbuffer, timer, now);
|
|
break;
|
|
}
|
|
return removed;
|
|
}
|
|
|
|
static void
|
|
push_rtx_events_unlocked (GstRtpJitterBuffer * jitterbuffer, GQueue * events)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstEvent *event;
|
|
|
|
while ((event = (GstEvent *) g_queue_pop_head (events)))
|
|
gst_pad_push_event (priv->sinkpad, event);
|
|
}
|
|
|
|
/* called with JBUF lock
|
|
*
|
|
* Pushes all events in @events queue.
|
|
*
|
|
* Returns: %TRUE if the timer thread is not longer running
|
|
*/
|
|
static void
|
|
push_rtx_events (GstRtpJitterBuffer * jitterbuffer, GQueue * events)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
|
|
if (events->length == 0)
|
|
return;
|
|
|
|
JBUF_UNLOCK (priv);
|
|
push_rtx_events_unlocked (jitterbuffer, events);
|
|
JBUF_LOCK (priv);
|
|
}
|
|
|
|
/* called when we need to wait for the next timeout.
|
|
*
|
|
* We loop over the array of recorded timeouts and wait for the earliest one.
|
|
* When it timed out, do the logic associated with the timer.
|
|
*
|
|
* If there are no timers, we wait on a gcond until something new happens.
|
|
*/
|
|
static void
|
|
wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
|
|
GstClockTime now = 0;
|
|
|
|
JBUF_LOCK (priv);
|
|
while (priv->timer_running) {
|
|
RtpTimer *timer = NULL;
|
|
GQueue events = G_QUEUE_INIT;
|
|
|
|
/* don't produce data in paused */
|
|
while (priv->blocked) {
|
|
JBUF_WAIT_TIMER (priv);
|
|
if (!priv->timer_running)
|
|
goto stopping;
|
|
}
|
|
|
|
/* If we have a clock, update "now" now with the very
|
|
* latest running time we have. If timers are unscheduled below we
|
|
* otherwise wouldn't update now (it's only updated when timers
|
|
* expire), and also for the very first loop iteration now would
|
|
* otherwise always be 0
|
|
*/
|
|
GST_OBJECT_LOCK (jitterbuffer);
|
|
if (priv->eos) {
|
|
now = GST_CLOCK_TIME_NONE;
|
|
} else if (GST_ELEMENT_CLOCK (jitterbuffer)) {
|
|
now =
|
|
gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
|
|
GST_ELEMENT_CAST (jitterbuffer)->base_time;
|
|
}
|
|
GST_OBJECT_UNLOCK (jitterbuffer);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (now));
|
|
|
|
/* Clear expired rtx-stats timers */
|
|
if (priv->do_retransmission)
|
|
rtp_timer_queue_remove_until (priv->rtx_stats_timers, now);
|
|
|
|
/* Iterate expired "normal" timers */
|
|
while ((timer = rtp_timer_queue_pop_until (priv->timers, now)))
|
|
do_timeout (jitterbuffer, timer, now, &events);
|
|
|
|
timer = rtp_timer_queue_peek_earliest (priv->timers);
|
|
if (timer) {
|
|
GstClock *clock;
|
|
GstClockTime sync_time;
|
|
GstClockID id;
|
|
GstClockReturn ret;
|
|
GstClockTimeDiff clock_jitter;
|
|
|
|
/* we poped all immediate and due timer, so this should just never
|
|
* happens */
|
|
g_assert (GST_CLOCK_TIME_IS_VALID (timer->timeout));
|
|
|
|
GST_OBJECT_LOCK (jitterbuffer);
|
|
clock = GST_ELEMENT_CLOCK (jitterbuffer);
|
|
if (!clock) {
|
|
GST_OBJECT_UNLOCK (jitterbuffer);
|
|
/* let's just push if there is no clock */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
|
|
now = timer->timeout;
|
|
push_rtx_events (jitterbuffer, &events);
|
|
continue;
|
|
}
|
|
|
|
/* prepare for sync against clock */
|
|
sync_time = timer->timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
|
|
/* add latency of peer to get input time */
|
|
sync_time += priv->peer_latency;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "timer #%i sync to timestamp %"
|
|
GST_TIME_FORMAT " with sync time %" GST_TIME_FORMAT, timer->seqnum,
|
|
GST_TIME_ARGS (get_pts_timeout (timer)), GST_TIME_ARGS (sync_time));
|
|
|
|
/* create an entry for the clock */
|
|
id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
|
|
priv->timer_timeout = timer->timeout;
|
|
priv->timer_seqnum = timer->seqnum;
|
|
GST_OBJECT_UNLOCK (jitterbuffer);
|
|
|
|
/* release the lock so that the other end can push stuff or unlock */
|
|
JBUF_UNLOCK (priv);
|
|
|
|
push_rtx_events_unlocked (jitterbuffer, &events);
|
|
|
|
ret = gst_clock_id_wait (id, &clock_jitter);
|
|
|
|
JBUF_LOCK (priv);
|
|
|
|
if (!priv->timer_running) {
|
|
g_queue_clear_full (&events, (GDestroyNotify) gst_event_unref);
|
|
gst_clock_id_unref (id);
|
|
priv->clock_id = NULL;
|
|
break;
|
|
}
|
|
|
|
if (ret != GST_CLOCK_UNSCHEDULED) {
|
|
now = priv->timer_timeout + MAX (clock_jitter, 0);
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
|
|
GST_STIME_ARGS (clock_jitter));
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
|
|
}
|
|
|
|
/* and free the entry */
|
|
gst_clock_id_unref (id);
|
|
priv->clock_id = NULL;
|
|
} else {
|
|
push_rtx_events_unlocked (jitterbuffer, &events);
|
|
|
|
/* when draining the timers, the pusher thread will reuse our
|
|
* condition to wait for completion. Signal that thread before
|
|
* sleeping again here */
|
|
if (priv->eos)
|
|
JBUF_SIGNAL_TIMER (priv);
|
|
|
|
/* no timers, wait for activity */
|
|
JBUF_WAIT_TIMER (priv);
|
|
}
|
|
}
|
|
stopping:
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
|
|
return;
|
|
}
|
|
|
|
/*
|
|
* This function implements the main pushing loop on the source pad.
|
|
*
|
|
* It first tries to push as many buffers as possible. If there is a seqnum
|
|
* mismatch, we wait for the next timeouts.
|
|
*/
|
|
static void
|
|
gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
JBUF_LOCK_CHECK (priv, flushing);
|
|
do {
|
|
result = handle_next_buffer (jitterbuffer);
|
|
if (G_LIKELY (result == GST_FLOW_WAIT)) {
|
|
/* now wait for the next event */
|
|
JBUF_SIGNAL_QUEUE (priv);
|
|
JBUF_WAIT_EVENT (priv, flushing);
|
|
result = GST_FLOW_OK;
|
|
}
|
|
} while (result == GST_FLOW_OK);
|
|
/* store result for upstream */
|
|
priv->srcresult = result;
|
|
/* if we get here we need to pause */
|
|
goto pause;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
result = priv->srcresult;
|
|
goto pause;
|
|
}
|
|
pause:
|
|
{
|
|
GstEvent *event;
|
|
|
|
JBUF_SIGNAL_QUERY (priv, FALSE);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
|
|
gst_flow_get_name (result));
|
|
gst_pad_pause_task (priv->srcpad);
|
|
if (result == GST_FLOW_EOS) {
|
|
event = gst_event_new_eos ();
|
|
if (priv->segment_seqnum != GST_SEQNUM_INVALID)
|
|
gst_event_set_seqnum (event, priv->segment_seqnum);
|
|
gst_pad_push_event (priv->srcpad, event);
|
|
}
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
do_handle_sync_inband (GstRtpJitterBuffer * jitterbuffer, guint64 ntpnstime)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstStructure *s;
|
|
guint64 base_rtptime, base_time;
|
|
guint32 clock_rate;
|
|
guint64 last_rtptime;
|
|
const gchar *cname = NULL;
|
|
GList *l;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
/* get the last values from the jitterbuffer */
|
|
rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
|
|
&clock_rate, &last_rtptime);
|
|
|
|
for (l = priv->cname_ssrc_mappings; l; l = l->next) {
|
|
const CNameSSRCMapping *map = l->data;
|
|
|
|
if (map->ssrc == priv->last_ssrc) {
|
|
cname = map->cname;
|
|
break;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"inband NTP-64 %" GST_TIME_FORMAT " rtptime %" G_GUINT64_FORMAT ", base %"
|
|
G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT ", clock-base %"
|
|
G_GUINT64_FORMAT ", CNAME %s", GST_TIME_ARGS (ntpnstime), last_rtptime,
|
|
base_rtptime, clock_rate, priv->clock_base, GST_STR_NULL (cname));
|
|
|
|
/* no CNAME known yet for this ssrc */
|
|
if (cname == NULL) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "no CNAME for this packet known yet");
|
|
return;
|
|
}
|
|
|
|
if (priv->last_ntpnstime != GST_CLOCK_TIME_NONE
|
|
&& ntpnstime - priv->last_ntpnstime < priv->sync_interval * GST_MSECOND) {
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"discarding RTCP sender packet for sync; "
|
|
"previous sender info too recent " "(previous NTP %" G_GUINT64_FORMAT
|
|
")", priv->last_ntpnstime);
|
|
return;
|
|
}
|
|
priv->last_ntpnstime = ntpnstime;
|
|
|
|
s = gst_structure_new ("application/x-rtp-sync",
|
|
"base-rtptime", G_TYPE_UINT64, base_rtptime,
|
|
"base-time", G_TYPE_UINT64, base_time,
|
|
"clock-rate", G_TYPE_UINT, clock_rate,
|
|
"clock-base", G_TYPE_UINT64, priv->clock_base & G_MAXUINT32,
|
|
"cname", G_TYPE_STRING, cname,
|
|
"ssrc", G_TYPE_UINT, priv->last_ssrc,
|
|
"inband-ext-rtptime", G_TYPE_UINT64, last_rtptime,
|
|
"inband-ntpnstime", G_TYPE_UINT64, ntpnstime, NULL);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
|
|
JBUF_UNLOCK (priv);
|
|
g_signal_emit (jitterbuffer,
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
|
|
JBUF_LOCK (priv);
|
|
gst_structure_free (s);
|
|
}
|
|
|
|
/* collect the info from the latest RTCP packet and the jitterbuffer sync, do
|
|
* some sanity checks and then emit the handle-sync signal with the parameters.
|
|
* This function must be called with the LOCK */
|
|
static void
|
|
do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv;
|
|
guint64 base_rtptime, base_time;
|
|
guint32 clock_rate;
|
|
guint64 last_rtptime;
|
|
guint64 clock_base;
|
|
guint64 ext_rtptime, diff;
|
|
gboolean valid = TRUE, keep = FALSE;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
/* get the last values from the jitterbuffer */
|
|
rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
|
|
&clock_rate, &last_rtptime);
|
|
|
|
clock_base = priv->clock_base;
|
|
ext_rtptime = priv->last_sr_ext_rtptime;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer,
|
|
"ext SR %" G_GUINT64_FORMAT ", NTP %" G_GUINT64_FORMAT ", base %"
|
|
G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT ", clock-base %"
|
|
G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT, ext_rtptime,
|
|
priv->last_sr_ntpnstime, base_rtptime, clock_rate, clock_base,
|
|
last_rtptime);
|
|
|
|
if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
|
|
/* we keep this SR packet for later. When we get a valid RTP packet the
|
|
* above values will be set and we can try to use the SR packet */
|
|
GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
|
|
keep = TRUE;
|
|
} else {
|
|
/* we can't accept anything that happened before we did the last resync */
|
|
if (base_rtptime > ext_rtptime) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
|
|
valid = FALSE;
|
|
} else {
|
|
/* the SR RTP timestamp must be something close to what we last observed
|
|
* in the jitterbuffer */
|
|
if (ext_rtptime > last_rtptime) {
|
|
/* check how far ahead it is to our RTP timestamps */
|
|
diff = ext_rtptime - last_rtptime;
|
|
/* if bigger than 1 second, we drop it */
|
|
if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
|
|
diff >
|
|
gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
|
|
clock_rate, 1000)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
|
|
/* should drop this, but some RTSP servers end up with bogus
|
|
* way too ahead RTCP packet when repeated PAUSE/PLAY,
|
|
* so still trigger rptbin sync but invalidate RTCP data
|
|
* (sync might use other methods) */
|
|
ext_rtptime = -1;
|
|
}
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
|
|
G_GUINT64_FORMAT, last_rtptime, diff);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (keep) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
|
|
} else if (valid) {
|
|
GstStructure *s;
|
|
GList *l;
|
|
|
|
s = gst_structure_new ("application/x-rtp-sync",
|
|
"base-rtptime", G_TYPE_UINT64, base_rtptime,
|
|
"base-time", G_TYPE_UINT64, base_time,
|
|
"clock-rate", G_TYPE_UINT, clock_rate,
|
|
"clock-base", G_TYPE_UINT64, priv->clock_base & G_MAXUINT32,
|
|
"ssrc", G_TYPE_UINT, priv->last_sr_ssrc,
|
|
"sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
|
|
"sr-ntpnstime", G_TYPE_UINT64, priv->last_sr_ntpnstime,
|
|
"sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
|
|
|
|
for (l = priv->cname_ssrc_mappings; l; l = l->next) {
|
|
const CNameSSRCMapping *map = l->data;
|
|
|
|
if (map->ssrc == priv->last_ssrc) {
|
|
gst_structure_set (s, "cname", G_TYPE_STRING, map->cname, NULL);
|
|
break;
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
|
|
gst_buffer_replace (&priv->last_sr, NULL);
|
|
JBUF_UNLOCK (priv);
|
|
g_signal_emit (jitterbuffer,
|
|
gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
|
|
JBUF_LOCK (priv);
|
|
gst_structure_free (s);
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
|
|
gst_buffer_replace (&priv->last_sr, NULL);
|
|
}
|
|
}
|
|
|
|
#define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
|
|
for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
|
|
(b) = gst_rtcp_packet_move_to_next ((packet)))
|
|
|
|
#define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
|
|
for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
|
|
(b) = gst_rtcp_packet_sdes_next_item ((packet)))
|
|
|
|
#define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
|
|
for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
|
|
(b) = gst_rtcp_packet_sdes_next_entry ((packet)))
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
guint32 ssrc;
|
|
GstRTCPPacket packet;
|
|
guint64 ext_rtptime, ntptime;
|
|
GstClockTime ntpnstime = GST_CLOCK_TIME_NONE;
|
|
guint32 rtptime;
|
|
GstRTCPBuffer rtcp = { NULL, };
|
|
gchar *cname = NULL;
|
|
gboolean have_sr = FALSE;
|
|
gboolean more;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
|
|
if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
|
|
goto invalid_buffer;
|
|
|
|
priv = jitterbuffer->priv;
|
|
|
|
gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
|
|
|
|
GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) {
|
|
/* first packet must be SR or RR or else the validate would have failed */
|
|
switch (gst_rtcp_packet_get_type (&packet)) {
|
|
case GST_RTCP_TYPE_SR:
|
|
/* only parse first. There is only supposed to be one SR in the packet
|
|
* but we will deal with malformed packets gracefully by trying the
|
|
* next RTCP packet */
|
|
if (have_sr)
|
|
continue;
|
|
|
|
/* get NTP and RTP times */
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
|
|
NULL, NULL);
|
|
|
|
/* convert ntptime to nanoseconds */
|
|
ntpnstime =
|
|
gst_util_uint64_scale (ntptime, GST_SECOND,
|
|
G_GUINT64_CONSTANT (1) << 32);
|
|
|
|
have_sr = TRUE;
|
|
|
|
break;
|
|
case GST_RTCP_TYPE_SDES:
|
|
{
|
|
gboolean more_items;
|
|
|
|
/* Bail out if we have not seen an SR item yet. */
|
|
if (!have_sr)
|
|
goto ignore_buffer;
|
|
|
|
GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
|
|
gboolean more_entries;
|
|
|
|
/* skip items that are not about the SSRC of the sender */
|
|
if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
|
|
continue;
|
|
|
|
/* find the CNAME entry */
|
|
GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
|
|
GstRTCPSDESType type;
|
|
guint8 len;
|
|
const guint8 *data;
|
|
|
|
gst_rtcp_packet_sdes_get_entry (&packet, &type, &len,
|
|
(guint8 **) & data);
|
|
|
|
if (type == GST_RTCP_SDES_CNAME) {
|
|
cname = g_strndup ((const gchar *) data, len);
|
|
goto out;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* only deal with first SDES, there is only supposed to be one SDES in
|
|
* the RTCP packet but we deal with bad packets gracefully. */
|
|
goto out;
|
|
}
|
|
default:
|
|
/* we can ignore these packets */
|
|
break;
|
|
}
|
|
}
|
|
out:
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x from CNAME %s",
|
|
ssrc, GST_STR_NULL (cname));
|
|
|
|
if (!have_sr)
|
|
goto empty_buffer;
|
|
|
|
JBUF_LOCK (priv);
|
|
if (cname)
|
|
insert_cname_ssrc_mapping (jitterbuffer, cname, ssrc);
|
|
|
|
/* convert the RTP timestamp to our extended timestamp, using the same offset
|
|
* we used in the jitterbuffer */
|
|
ext_rtptime = priv->jbuf->ext_rtptime;
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
|
|
|
|
priv->last_sr_ext_rtptime = ext_rtptime;
|
|
priv->last_sr_ssrc = ssrc;
|
|
priv->last_sr_ntpnstime = ntpnstime;
|
|
|
|
priv->last_known_ext_rtptime = ext_rtptime;
|
|
priv->last_known_ntpnstime = ntpnstime;
|
|
|
|
if (G_UNLIKELY (priv->last_ssrc == -1)) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "SSRC changed from %u to %u",
|
|
priv->last_ssrc, ssrc);
|
|
priv->last_ssrc = ssrc;
|
|
}
|
|
|
|
if (priv->last_ntpnstime != GST_CLOCK_TIME_NONE
|
|
&& ntpnstime - priv->last_ntpnstime < priv->sync_interval * GST_MSECOND) {
|
|
gst_buffer_replace (&priv->last_sr, NULL);
|
|
GST_DEBUG_OBJECT (jitterbuffer, "discarding RTCP sender packet for sync; "
|
|
"previous sender info too recent "
|
|
"(previous NTP %" G_GUINT64_FORMAT ")", priv->last_ntpnstime);
|
|
} else {
|
|
gst_buffer_replace (&priv->last_sr, buffer);
|
|
do_handle_sync (jitterbuffer);
|
|
priv->last_ntpnstime = ntpnstime;
|
|
}
|
|
|
|
JBUF_UNLOCK (priv);
|
|
|
|
done:
|
|
g_free (cname);
|
|
gst_buffer_unref (buffer);
|
|
|
|
return ret;
|
|
|
|
invalid_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received invalid RTCP payload, dropping"));
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
empty_buffer:
|
|
{
|
|
/* this is not fatal but should be filtered earlier */
|
|
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
|
|
("Received empty RTCP payload, dropping"));
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
ignore_buffer:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
ret = GST_FLOW_OK;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
if (GST_QUERY_IS_SERIALIZED (query)) {
|
|
JBUF_LOCK_CHECK (priv, out_flushing);
|
|
if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
|
|
RTP_JITTER_BUFFER_MODE_BUFFER) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
|
|
if (rtp_jitter_buffer_append_query (priv->jbuf, query))
|
|
JBUF_SIGNAL_EVENT (priv);
|
|
JBUF_WAIT_QUERY (priv, out_flushing);
|
|
res = priv->last_query;
|
|
} else {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
|
|
res = FALSE;
|
|
}
|
|
JBUF_UNLOCK (priv);
|
|
} else {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
}
|
|
break;
|
|
}
|
|
return res;
|
|
/* ERRORS */
|
|
out_flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
|
|
JBUF_UNLOCK (priv);
|
|
return FALSE;
|
|
}
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
gboolean res = FALSE;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
/* We need to send the query upstream and add the returned latency to our
|
|
* own */
|
|
GstClockTime min_latency, max_latency;
|
|
gboolean us_live;
|
|
GstClockTime our_latency;
|
|
|
|
if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
|
|
gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
/* store this so that we can safely sync on the peer buffers. */
|
|
JBUF_LOCK (priv);
|
|
priv->peer_latency = min_latency;
|
|
our_latency = priv->latency_ns;
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (our_latency));
|
|
|
|
/* we add some latency but can buffer an infinite amount of time */
|
|
min_latency += our_latency;
|
|
max_latency = -1;
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
|
|
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
gst_query_set_latency (query, TRUE, min_latency, max_latency);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstClockTime start, last_out;
|
|
GstFormat fmt;
|
|
|
|
gst_query_parse_position (query, &fmt, NULL);
|
|
if (fmt != GST_FORMAT_TIME) {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
JBUF_LOCK (priv);
|
|
start = priv->npt_start;
|
|
last_out = priv->last_out_time;
|
|
JBUF_UNLOCK (priv);
|
|
|
|
GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
|
|
", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
|
|
GST_TIME_ARGS (last_out));
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
|
|
/* bring 0-based outgoing time to stream time */
|
|
gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
|
|
res = TRUE;
|
|
} else {
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
{
|
|
guint new_latency, old_latency;
|
|
|
|
new_latency = g_value_get_uint (value);
|
|
|
|
JBUF_LOCK (priv);
|
|
old_latency = priv->latency_ms;
|
|
priv->latency_ms = new_latency;
|
|
priv->latency_ns = priv->latency_ms * GST_MSECOND;
|
|
rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
/* post message if latency changed, this will inform the parent pipeline
|
|
* that a latency reconfiguration is possible/needed. */
|
|
if (new_latency != old_latency) {
|
|
GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (new_latency * GST_MSECOND));
|
|
|
|
gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
|
|
gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
|
|
}
|
|
break;
|
|
}
|
|
case PROP_DROP_ON_LATENCY:
|
|
JBUF_LOCK (priv);
|
|
priv->drop_on_latency = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
JBUF_LOCK (priv);
|
|
if (priv->max_ts_offset_adjustment != 0) {
|
|
gint64 new_offset = g_value_get_int64 (value);
|
|
|
|
if (new_offset > priv->ts_offset) {
|
|
priv->ts_offset_remainder = new_offset - priv->ts_offset;
|
|
} else {
|
|
priv->ts_offset_remainder = -(priv->ts_offset - new_offset);
|
|
}
|
|
} else {
|
|
priv->ts_offset = g_value_get_int64 (value);
|
|
priv->ts_offset_remainder = 0;
|
|
update_timer_offsets (jitterbuffer);
|
|
}
|
|
priv->ts_discont = TRUE;
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
|
|
JBUF_LOCK (priv);
|
|
priv->max_ts_offset_adjustment = g_value_get_uint64 (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_DO_LOST:
|
|
JBUF_LOCK (priv);
|
|
priv->do_lost = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_POST_DROP_MESSAGES:
|
|
JBUF_LOCK (priv);
|
|
priv->post_drop_messages = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_DROP_MESSAGES_INTERVAL:
|
|
JBUF_LOCK (priv);
|
|
priv->drop_messages_interval_ms = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MODE:
|
|
JBUF_LOCK (priv);
|
|
rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_DO_RETRANSMISSION:
|
|
JBUF_LOCK (priv);
|
|
priv->do_retransmission = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_NEXT_SEQNUM:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_next_seqnum = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DELAY:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_delay = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MIN_DELAY:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_min_delay = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DELAY_REORDER:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_delay_reorder = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_retry_timeout = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MIN_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_min_retry_timeout = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_PERIOD:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_retry_period = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MAX_RETRIES:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_max_retries = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DEADLINE:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_deadline_ms = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_STATS_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
priv->rtx_stats_timeout = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MAX_RTCP_RTP_TIME_DIFF:
|
|
JBUF_LOCK (priv);
|
|
priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MAX_DROPOUT_TIME:
|
|
JBUF_LOCK (priv);
|
|
priv->max_dropout_time = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MAX_MISORDER_TIME:
|
|
JBUF_LOCK (priv);
|
|
priv->max_misorder_time = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RFC7273_SYNC:
|
|
JBUF_LOCK (priv);
|
|
rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
|
|
g_value_get_boolean (value));
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_FASTSTART_MIN_PACKETS:
|
|
JBUF_LOCK (priv);
|
|
priv->faststart_min_packets = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_ADD_REFERENCE_TIMESTAMP_META:
|
|
JBUF_LOCK (priv);
|
|
priv->add_reference_timestamp_meta = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_SYNC_INTERVAL:
|
|
JBUF_LOCK (priv);
|
|
priv->sync_interval = g_value_get_uint (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RFC7273_USE_SYSTEM_CLOCK:
|
|
JBUF_LOCK (priv);
|
|
priv->rfc7273_use_system_clock = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RFC7273_REFERENCE_TIMESTAMP_META_ONLY:
|
|
JBUF_LOCK (priv);
|
|
priv->rfc7273_reference_timestamp_meta_only = g_value_get_boolean (value);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_jitter_buffer_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpJitterBuffer *jitterbuffer;
|
|
GstRtpJitterBufferPrivate *priv;
|
|
|
|
jitterbuffer = GST_RTP_JITTER_BUFFER (object);
|
|
priv = jitterbuffer->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->latency_ms);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_DROP_ON_LATENCY:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->drop_on_latency);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_TS_OFFSET:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int64 (value, priv->ts_offset);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint64 (value, priv->max_ts_offset_adjustment);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_DO_LOST:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->do_lost);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_POST_DROP_MESSAGES:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->post_drop_messages);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_DROP_MESSAGES_INTERVAL:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->drop_messages_interval_ms);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MODE:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_PERCENT:
|
|
{
|
|
gint percent;
|
|
|
|
JBUF_LOCK (priv);
|
|
if (priv->srcresult != GST_FLOW_OK)
|
|
percent = 100;
|
|
else
|
|
percent = rtp_jitter_buffer_get_percent (priv->jbuf);
|
|
|
|
g_value_set_int (value, percent);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
}
|
|
case PROP_DO_RETRANSMISSION:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->do_retransmission);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_NEXT_SEQNUM:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->rtx_next_seqnum);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DELAY:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_delay);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MIN_DELAY:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->rtx_min_delay);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DELAY_REORDER:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_delay_reorder);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_retry_timeout);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MIN_RETRY_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_min_retry_timeout);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_RETRY_PERIOD:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_retry_period);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_MAX_RETRIES:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_max_retries);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_DEADLINE:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->rtx_deadline_ms);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RTX_STATS_TIMEOUT:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->rtx_stats_timeout);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_STATS:
|
|
g_value_take_boxed (value,
|
|
gst_rtp_jitter_buffer_create_stats (jitterbuffer));
|
|
break;
|
|
case PROP_MAX_RTCP_RTP_TIME_DIFF:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MAX_DROPOUT_TIME:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->max_dropout_time);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_MAX_MISORDER_TIME:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->max_misorder_time);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RFC7273_SYNC:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value,
|
|
rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_FASTSTART_MIN_PACKETS:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->faststart_min_packets);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_ADD_REFERENCE_TIMESTAMP_META:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->add_reference_timestamp_meta);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_SYNC_INTERVAL:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_uint (value, priv->sync_interval);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RFC7273_USE_SYSTEM_CLOCK:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->rfc7273_use_system_clock);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
case PROP_RFC7273_REFERENCE_TIMESTAMP_META_ONLY:
|
|
JBUF_LOCK (priv);
|
|
g_value_set_boolean (value, priv->rfc7273_reference_timestamp_meta_only);
|
|
JBUF_UNLOCK (priv);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstStructure *
|
|
gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
|
|
{
|
|
GstRtpJitterBufferPrivate *priv = jbuf->priv;
|
|
GstStructure *s;
|
|
|
|
JBUF_LOCK (priv);
|
|
s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
|
|
"num-pushed", G_TYPE_UINT64, priv->num_pushed,
|
|
"num-lost", G_TYPE_UINT64, priv->num_lost,
|
|
"num-late", G_TYPE_UINT64, priv->num_late,
|
|
"num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
|
|
"avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
|
|
"rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
|
|
"rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
|
|
"rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
|
|
"rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);
|
|
JBUF_UNLOCK (priv);
|
|
|
|
return s;
|
|
}
|