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niklase@google.com470e71d2011-07-07 08:21:25 +00001/*
pwestin@webrtc.orgf6bb77a2012-01-24 17:16:59 +00002 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
niklase@google.com470e71d2011-07-07 08:21:25 +00003 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
12#define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
niklase@google.com470e71d2011-07-07 08:21:25 +000013
Yves Gerey988cc082018-10-23 12:03:01 +020014#include <stddef.h>
15#include <stdint.h>
Jonas Olssona4d87372019-07-05 19:08:33 +020016
nisse14adba72017-03-20 03:52:39 -070017#include <memory>
danilchapb8b6fbb2015-12-10 05:05:27 -080018#include <set>
Steve Anton296a0ce2018-03-22 15:17:27 -070019#include <string>
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000020#include <vector>
niklase@google.com470e71d2011-07-07 08:21:25 +000021
Ali Tofighd14e8892022-05-13 11:42:16 +020022#include "absl/strings/string_view.h"
Danil Chapovalovd264df52018-06-14 12:59:38 +020023#include "absl/types/optional.h"
Yves Gerey988cc082018-10-23 12:03:01 +020024#include "api/rtp_headers.h"
Erik Språngeeaa8f92018-05-17 12:35:56 +020025#include "api/video/video_bitrate_allocation.h"
Yves Gerey988cc082018-10-23 12:03:01 +020026#include "modules/include/module_fec_types.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020027#include "modules/rtp_rtcp/include/rtp_rtcp.h"
Yves Gerey988cc082018-10-23 12:03:01 +020028#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +020029#include "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"
Erik Språngbfcfe032021-08-04 14:45:32 +020030#include "modules/rtp_rtcp/source/packet_sequencer.h"
Yves Gerey988cc082018-10-23 12:03:01 +020031#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020032#include "modules/rtp_rtcp/source/rtcp_receiver.h"
33#include "modules/rtp_rtcp/source/rtcp_sender.h"
Erik Språng77b75292019-10-28 15:51:36 +010034#include "modules/rtp_rtcp/source/rtp_packet_history.h"
Erik Språng9c771c22019-06-17 16:31:53 +020035#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020036#include "modules/rtp_rtcp/source/rtp_sender.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020037#include "rtc_base/gtest_prod_util.h"
Markus Handellf7303e62020-07-09 01:34:42 +020038#include "rtc_base/synchronization/mutex.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000039
niklase@google.com470e71d2011-07-07 08:21:25 +000040namespace webrtc {
41
Yves Gerey988cc082018-10-23 12:03:01 +020042class Clock;
43struct PacedPacketInfo;
44struct RTPVideoHeader;
45
Tommi3a5742c2020-05-20 09:32:51 +020046// DEPRECATED.
danilchap59cb2bd2016-08-29 11:08:47 -070047class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +000048 public:
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020049 explicit ModuleRtpRtcpImpl(
50 const RtpRtcpInterface::Configuration& configuration);
Danil Chapovalov2a5ce2b2018-02-07 09:38:31 +010051 ~ModuleRtpRtcpImpl() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000052
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000053 // Process any pending tasks such as timeouts.
pbosa26ac922016-02-25 04:50:01 -080054 void Process() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000055
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000056 // Receiver part.
niklase@google.com470e71d2011-07-07 08:21:25 +000057
stefan@webrtc.orga5cb98c2013-05-29 12:12:51 +000058 // Called when we receive an RTCP packet.
nisse479d3d72017-09-13 07:53:37 -070059 void IncomingRtcpPacket(const uint8_t* incoming_packet,
60 size_t incoming_packet_length) override;
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +000061
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000062 void SetRemoteSSRC(uint32_t ssrc) override;
Tommi08be9ba2021-06-15 23:01:57 +020063 void SetLocalSsrc(uint32_t ssrc) override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +000064
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000065 // Sender part.
Niels Möller5fe95102019-03-04 16:49:25 +010066 void RegisterSendPayloadFrequency(int payload_type,
67 int payload_frequency) override;
Peter Boström8b79b072016-02-26 16:31:37 +010068
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000069 int32_t DeRegisterSendPayload(int8_t payload_type) override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +000070
Johannes Kron9190b822018-10-29 11:22:05 +010071 void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
72
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000073 // Register RTP header extension.
Sebastian Janssonf39c8152019-10-14 17:32:21 +020074 void RegisterRtpHeaderExtension(absl::string_view uri, int id) override;
Sebastian Janssonf39c8152019-10-14 17:32:21 +020075 void DeregisterSendRtpHeaderExtension(absl::string_view uri) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000076
Mirko Bonadei999a72a2019-07-12 17:33:46 +000077 bool SupportsPadding() const override;
78 bool SupportsRtxPayloadPadding() const override;
stefan53b6cc32017-02-03 08:13:57 -080079
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000080 // Get start timestamp.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000081 uint32_t StartTimestamp() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000082
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000083 // Configure start timestamp, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000084 void SetStartTimestamp(uint32_t timestamp) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000085
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000086 uint16_t SequenceNumber() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000087
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000088 // Set SequenceNumber, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000089 void SetSequenceNumber(uint16_t seq) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000090
Per83d09102016-04-15 14:59:13 +020091 void SetRtpState(const RtpState& rtp_state) override;
92 void SetRtxState(const RtpState& rtp_state) override;
93 RtpState GetRtpState() const override;
94 RtpState GetRtxState() const override;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000095
Ivo Creusen8c40d512021-07-13 12:53:22 +000096 void SetNonSenderRttMeasurement(bool enabled) override {}
97
Erik Språng6841d252019-10-15 14:29:11 +020098 uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
niklase@google.com470e71d2011-07-07 08:21:25 +000099
Ali Tofighd14e8892022-05-13 11:42:16 +0200100 void SetMid(absl::string_view mid) override;
Steve Anton296a0ce2018-03-22 15:17:27 -0700101
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000102 void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000103
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000104 RTCPSender::FeedbackState GetFeedbackState();
niklase@google.com470e71d2011-07-07 08:21:25 +0000105
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000106 void SetRtxSendStatus(int mode) override;
107 int RtxSendStatus() const override;
Erik Språngc06aef22019-10-17 13:02:27 +0200108 absl::optional<uint32_t> RtxSsrc() const override;
mflodman@webrtc.org9f5ebb52013-04-12 14:55:46 +0000109
Shao Changbine62202f2015-04-21 20:24:50 +0800110 void SetRtxSendPayloadType(int payload_type,
111 int associated_payload_type) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000112
Danil Chapovalovd264df52018-06-14 12:59:38 +0200113 absl::optional<uint32_t> FlexfecSsrc() const override;
brandtr9dfff292016-11-14 05:14:50 -0800114
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000115 // Sends kRtcpByeCode when going from true to false.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000116 int32_t SetSendingStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000117
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000118 bool Sending() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000119
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000120 // Drops or relays media packets.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000121 void SetSendingMediaStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000122
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000123 bool SendingMedia() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000124
Erik Språng1e51a382019-12-11 16:47:09 +0100125 bool IsAudioConfigured() const override;
126
Sebastian Jansson1bca65b2018-10-10 09:58:08 +0200127 void SetAsPartOfAllocation(bool part_of_allocation) override;
128
Niels Möller5fe95102019-03-04 16:49:25 +0100129 bool OnSendingRtpFrame(uint32_t timestamp,
130 int64_t capture_time_ms,
131 int payload_type,
132 bool force_sender_report) override;
133
Erik Språng9c771c22019-06-17 16:31:53 +0200134 bool TrySendPacket(RtpPacketToSend* packet,
135 const PacedPacketInfo& pacing_info) override;
136
Erik Språng1d50cb62020-07-02 17:41:32 +0200137 void SetFecProtectionParams(const FecProtectionParams& delta_params,
138 const FecProtectionParams& key_params) override;
139
140 std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
141
Erik Språnga9229042019-10-24 12:39:32 +0200142 void OnPacketsAcknowledged(
143 rtc::ArrayView<const uint16_t> sequence_numbers) override;
stefan@webrtc.org0a3c1472013-12-05 14:05:07 +0000144
Erik Språngf6468d22019-07-05 16:53:43 +0200145 std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
146 size_t target_size_bytes) override;
Erik Språng478cb462019-06-26 15:49:27 +0200147
Erik Språng3663f942020-02-07 10:05:15 +0100148 std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
149 rtc::ArrayView<const uint16_t> sequence_numbers) const override;
150
Erik Språng04e1bab2020-05-07 18:18:32 +0200151 size_t ExpectedPerPacketOverhead() const override;
152
Erik Språngb6bbdeb2021-08-13 16:12:41 +0200153 void OnPacketSendingThreadSwitched() override;
154
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000155 // RTCP part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000156
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000157 // Get RTCP status.
pbosda903ea2015-10-02 02:36:56 -0700158 RtcpMode RTCP() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000159
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000160 // Configure RTCP status i.e on/off.
pbosda903ea2015-10-02 02:36:56 -0700161 void SetRTCPStatus(RtcpMode method) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000162
163 // Set RTCP CName.
Ali Tofighd14e8892022-05-13 11:42:16 +0200164 int32_t SetCNAME(absl::string_view c_name) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000165
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000166 // Get remote NTP.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000167 int32_t RemoteNTP(uint32_t* received_ntp_secs,
168 uint32_t* received_ntp_frac,
169 uint32_t* rtcp_arrival_time_secs,
170 uint32_t* rtcp_arrival_time_frac,
171 uint32_t* rtcp_timestamp) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000172
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000173 // Get RoundTripTime.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000174 int32_t RTT(uint32_t remote_ssrc,
175 int64_t* rtt,
176 int64_t* avg_rtt,
177 int64_t* min_rtt,
178 int64_t* max_rtt) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000179
Niels Möller5fe95102019-03-04 16:49:25 +0100180 int64_t ExpectedRetransmissionTimeMs() const override;
181
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000182 // Force a send of an RTCP packet.
183 // Normal SR and RR are triggered via the process function.
Erik Språng242e22b2015-05-11 10:17:43 +0200184 int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
185
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000186 void GetSendStreamDataCounters(
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000187 StreamDataCounters* rtp_counters,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000188 StreamDataCounters* rtx_counters) const override;
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000189
Henrik Boström6e436d12019-05-27 12:19:33 +0200190 // A snapshot of the most recent Report Block with additional data of
191 // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
192 // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
193 // which is the SSRC of the corresponding outbound RTP stream, is unique.
194 std::vector<ReportBlockData> GetLatestReportBlockData() const override;
Alessio Bazzicabc1c93d2021-03-12 17:45:26 +0100195 absl::optional<SenderReportStats> GetSenderReportStats() const override;
Ivo Creusen2562cf02021-09-03 14:51:22 +0000196 // Round trip time statistics computed from the XR block contained in the last
197 // report.
198 absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000199
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000200 // (REMB) Receiver Estimated Max Bitrate.
Danil Chapovalov1de4b622017-12-13 13:35:10 +0100201 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
Danil Chapovalov51e21aa2017-10-10 17:46:26 +0200202 void UnsetRemb() override;
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000203
danilchap59cb2bd2016-08-29 11:08:47 -0700204 void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000205
nisse284542b2017-01-10 08:58:32 -0800206 size_t MaxRtpPacketSize() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000207
nisse284542b2017-01-10 08:58:32 -0800208 void SetMaxRtpPacketSize(size_t max_packet_size) override;
michaelt79e05882016-11-08 02:50:09 -0800209
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000210 // (NACK) Negative acknowledgment part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000211
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000212 // Send a Negative acknowledgment packet.
philipel83f831a2016-03-12 03:30:23 -0800213 // TODO(philipel): Deprecate SendNACK and use SendNack instead.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000214 int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
stefan@webrtc.org6a4bef42011-12-22 12:52:41 +0000215
philipel83f831a2016-03-12 03:30:23 -0800216 void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
217
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000218 // Store the sent packets, needed to answer to a negative acknowledgment
219 // requests.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000220 void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000221
Per Kjellander16999812019-10-10 12:57:28 +0200222 void SendCombinedRtcpPacket(
223 std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override;
224
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000225 // Video part.
Elad Alon7d6a4c02019-02-25 13:00:51 +0100226 int32_t SendLossNotification(uint16_t last_decoded_seq_num,
227 uint16_t last_received_seq_num,
Elad Alone86af2c2019-06-03 14:37:50 +0200228 bool decodability_flag,
229 bool buffering_allowed) override;
Elad Alon7d6a4c02019-02-25 13:00:51 +0100230
Erik Språngbf46cfe2020-05-11 18:22:02 +0200231 RtpSendRates GetSendRates() const override;
232
danilchap59cb2bd2016-08-29 11:08:47 -0700233 void OnReceivedNack(
234 const std::vector<uint16_t>& nack_sequence_numbers) override;
235 void OnReceivedRtcpReportBlocks(
236 const ReportBlockList& report_blocks) override;
237 void OnRequestSendReport() override;
henrike@webrtc.orgf5da4da2012-02-15 23:54:59 +0000238
Erik Språng566124a2018-04-23 12:32:22 +0200239 void SetVideoBitrateAllocation(
240 const VideoBitrateAllocation& bitrate) override;
sprang5e38c962016-12-01 05:18:09 -0800241
Niels Möller5fe95102019-03-04 16:49:25 +0100242 RTPSender* RtpSender() override;
243 const RTPSender* RtpSender() const override;
244
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000245 protected:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000246 bool UpdateRTCPReceiveInformationTimers();
niklase@google.com470e71d2011-07-07 08:21:25 +0000247
Erik Språng77b75292019-10-28 15:51:36 +0100248 RTPSender* rtp_sender() {
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100249 return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
Erik Språng77b75292019-10-28 15:51:36 +0100250 }
251 const RTPSender* rtp_sender() const {
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100252 return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
Erik Språng77b75292019-10-28 15:51:36 +0100253 }
nissea33c62e2017-03-14 00:49:45 -0700254
255 RTCPSender* rtcp_sender() { return &rtcp_sender_; }
256 const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
257
258 RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
259 const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
260
Tomas Gunnarsson79ca92d2020-06-18 17:30:15 +0200261 void SetMediaHasBeenSent(bool media_has_been_sent) {
262 rtp_sender_->packet_sender.SetMediaHasBeenSent(media_has_been_sent);
263 }
264
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100265 Clock* clock() const { return clock_; }
nissea33c62e2017-03-14 00:49:45 -0700266
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000267 private:
asapersson@webrtc.orge7b1e112013-12-16 14:40:36 +0000268 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000269 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
niklase@google.com470e71d2011-07-07 08:21:25 +0000270
Erik Språng77b75292019-10-28 15:51:36 +0100271 struct RtpSenderContext {
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +0200272 explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config);
Erik Språng77b75292019-10-28 15:51:36 +0100273 // Storage of packets, for retransmissions and padding, if applicable.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100274 RtpPacketHistory packet_history;
Erik Språngbfcfe032021-08-04 14:45:32 +0200275 // Handles sequence number assignment and padding timestamp generation.
Erik Språng5f1d4062021-08-12 11:34:03 +0200276 mutable Mutex sequencer_mutex;
277 PacketSequencer sequencer_ RTC_GUARDED_BY(sequencer_mutex);
Erik Språng77b75292019-10-28 15:51:36 +0100278 // Handles final time timestamping/stats/etc and handover to Transport.
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +0200279 DEPRECATED_RtpSenderEgress packet_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100280 // If no paced sender configured, this class will be used to pass packets
Artem Titov913cfa72021-07-28 23:57:33 +0200281 // from `packet_generator_` to `packet_sender_`.
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +0200282 DEPRECATED_RtpSenderEgress::NonPacedPacketSender non_paced_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100283 // Handles creation of RTP packets to be sent.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100284 RTPSender packet_generator;
Erik Språng77b75292019-10-28 15:51:36 +0100285 };
286
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000287 void set_rtt_ms(int64_t rtt_ms);
288 int64_t rtt_ms() const;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000289
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000290 bool TimeToSendFullNackList(int64_t now) const;
291
Niels Mölleraf6ea0c2020-11-20 12:21:21 +0100292 // Returns true if the module is configured to store packets.
293 bool StorePackets() const;
294
295 // Returns current Receiver Reference Time Report (RTTR) status.
296 bool RtcpXrRrtrStatus() const;
297
Erik Språng77b75292019-10-28 15:51:36 +0100298 std::unique_ptr<RtpSenderContext> rtp_sender_;
299
nisse150708e2017-03-16 05:02:53 -0700300 RTCPSender rtcp_sender_;
301 RTCPReceiver rtcp_receiver_;
302
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100303 Clock* const clock_;
nisse150708e2017-03-16 05:02:53 -0700304
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000305 int64_t last_bitrate_process_time_;
306 int64_t last_rtt_process_time_;
307 uint16_t packet_overhead_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000308
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000309 // Send side
Danil Chapovalov9eb6ce12017-12-15 12:25:01 +0100310 int64_t nack_last_time_sent_full_ms_;
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000311 uint16_t nack_last_seq_number_sent_;
stefan@webrtc.org9354cc92012-06-07 08:10:14 +0000312
Tommi5f223652018-03-26 13:28:26 +0200313 RtcpRttStats* const rtt_stats_;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000314
315 // The processed RTT from RtcpRttStats.
Markus Handellf7303e62020-07-09 01:34:42 +0200316 mutable Mutex mutex_rtt_;
Niels Möllercd982132020-11-26 16:19:56 +0100317 int64_t rtt_ms_ RTC_GUARDED_BY(mutex_rtt_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000318};
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000319
320} // namespace webrtc
321
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200322#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_