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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
Danil Chapovalovd264df52018-06-14 12:59:38 +020022#include "absl/types/optional.h"
Yves Gerey988cc082018-10-23 12:03:01 +020023#include "api/rtp_headers.h"
Erik Språngeeaa8f92018-05-17 12:35:56 +020024#include "api/video/video_bitrate_allocation.h"
Yves Gerey988cc082018-10-23 12:03:01 +020025#include "modules/include/module_fec_types.h"
26#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.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"
Yves Gerey988cc082018-10-23 12:03:01 +020030#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020031#include "modules/rtp_rtcp/source/rtcp_receiver.h"
32#include "modules/rtp_rtcp/source/rtcp_sender.h"
Erik Språng77b75292019-10-28 15:51:36 +010033#include "modules/rtp_rtcp/source/rtp_packet_history.h"
Erik Språng9c771c22019-06-17 16:31:53 +020034#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020035#include "modules/rtp_rtcp/source/rtp_sender.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020036#include "rtc_base/gtest_prod_util.h"
Markus Handellf7303e62020-07-09 01:34:42 +020037#include "rtc_base/synchronization/mutex.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000038
niklase@google.com470e71d2011-07-07 08:21:25 +000039namespace webrtc {
40
Yves Gerey988cc082018-10-23 12:03:01 +020041class Clock;
42struct PacedPacketInfo;
43struct RTPVideoHeader;
44
Tommi3a5742c2020-05-20 09:32:51 +020045// DEPRECATED.
danilchap59cb2bd2016-08-29 11:08:47 -070046class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +000047 public:
Tomas Gunnarssonf25761d2020-06-03 22:55:33 +020048 explicit ModuleRtpRtcpImpl(
49 const RtpRtcpInterface::Configuration& configuration);
Danil Chapovalov2a5ce2b2018-02-07 09:38:31 +010050 ~ModuleRtpRtcpImpl() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000051
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000052 // Returns the number of milliseconds until the module want a worker thread to
53 // call Process.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000054 int64_t TimeUntilNextProcess() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000055
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000056 // Process any pending tasks such as timeouts.
pbosa26ac922016-02-25 04:50:01 -080057 void Process() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000058
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000059 // Receiver part.
niklase@google.com470e71d2011-07-07 08:21:25 +000060
stefan@webrtc.orga5cb98c2013-05-29 12:12:51 +000061 // Called when we receive an RTCP packet.
nisse479d3d72017-09-13 07:53:37 -070062 void IncomingRtcpPacket(const uint8_t* incoming_packet,
63 size_t incoming_packet_length) override;
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +000064
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000065 void SetRemoteSSRC(uint32_t ssrc) override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +000066
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000067 // Sender part.
Niels Möller5fe95102019-03-04 16:49:25 +010068 void RegisterSendPayloadFrequency(int payload_type,
69 int payload_frequency) override;
Peter Boström8b79b072016-02-26 16:31:37 +010070
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000071 int32_t DeRegisterSendPayload(int8_t payload_type) override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +000072
Johannes Kron9190b822018-10-29 11:22:05 +010073 void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
74
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000075 // Register RTP header extension.
Sebastian Janssonf39c8152019-10-14 17:32:21 +020076 void RegisterRtpHeaderExtension(absl::string_view uri, int id) override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000077 int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
Sebastian Janssonf39c8152019-10-14 17:32:21 +020078 void DeregisterSendRtpHeaderExtension(absl::string_view uri) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000079
Mirko Bonadei999a72a2019-07-12 17:33:46 +000080 bool SupportsPadding() const override;
81 bool SupportsRtxPayloadPadding() const override;
stefan53b6cc32017-02-03 08:13:57 -080082
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000083 // Get start timestamp.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000084 uint32_t StartTimestamp() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000085
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000086 // Configure start timestamp, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000087 void SetStartTimestamp(uint32_t timestamp) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000088
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000089 uint16_t SequenceNumber() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000090
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000091 // Set SequenceNumber, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000092 void SetSequenceNumber(uint16_t seq) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000093
Per83d09102016-04-15 14:59:13 +020094 void SetRtpState(const RtpState& rtp_state) override;
95 void SetRtxState(const RtpState& rtp_state) override;
96 RtpState GetRtpState() const override;
97 RtpState GetRtxState() const override;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000098
Erik Språng6841d252019-10-15 14:29:11 +020099 uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
niklase@google.com470e71d2011-07-07 08:21:25 +0000100
Amit Hilbuch77938e62018-12-21 09:23:38 -0800101 void SetRid(const std::string& rid) override;
102
Steve Anton296a0ce2018-03-22 15:17:27 -0700103 void SetMid(const std::string& mid) override;
104
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000105 void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000106
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000107 RTCPSender::FeedbackState GetFeedbackState();
niklase@google.com470e71d2011-07-07 08:21:25 +0000108
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000109 void SetRtxSendStatus(int mode) override;
110 int RtxSendStatus() const override;
Erik Språngc06aef22019-10-17 13:02:27 +0200111 absl::optional<uint32_t> RtxSsrc() const override;
mflodman@webrtc.org9f5ebb52013-04-12 14:55:46 +0000112
Shao Changbine62202f2015-04-21 20:24:50 +0800113 void SetRtxSendPayloadType(int payload_type,
114 int associated_payload_type) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000115
Danil Chapovalovd264df52018-06-14 12:59:38 +0200116 absl::optional<uint32_t> FlexfecSsrc() const override;
brandtr9dfff292016-11-14 05:14:50 -0800117
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000118 // Sends kRtcpByeCode when going from true to false.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000119 int32_t SetSendingStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000120
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000121 bool Sending() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000122
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000123 // Drops or relays media packets.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000124 void SetSendingMediaStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000125
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000126 bool SendingMedia() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000127
Erik Språng1e51a382019-12-11 16:47:09 +0100128 bool IsAudioConfigured() const override;
129
Sebastian Jansson1bca65b2018-10-10 09:58:08 +0200130 void SetAsPartOfAllocation(bool part_of_allocation) override;
131
Niels Möller5fe95102019-03-04 16:49:25 +0100132 bool OnSendingRtpFrame(uint32_t timestamp,
133 int64_t capture_time_ms,
134 int payload_type,
135 bool force_sender_report) override;
136
Erik Språng9c771c22019-06-17 16:31:53 +0200137 bool TrySendPacket(RtpPacketToSend* packet,
138 const PacedPacketInfo& pacing_info) override;
139
Erik Språng1d50cb62020-07-02 17:41:32 +0200140 void SetFecProtectionParams(const FecProtectionParams& delta_params,
141 const FecProtectionParams& key_params) override;
142
143 std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override;
144
Erik Språnga9229042019-10-24 12:39:32 +0200145 void OnPacketsAcknowledged(
146 rtc::ArrayView<const uint16_t> sequence_numbers) override;
stefan@webrtc.org0a3c1472013-12-05 14:05:07 +0000147
Erik Språngf6468d22019-07-05 16:53:43 +0200148 std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
149 size_t target_size_bytes) override;
Erik Språng478cb462019-06-26 15:49:27 +0200150
Erik Språng3663f942020-02-07 10:05:15 +0100151 std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
152 rtc::ArrayView<const uint16_t> sequence_numbers) const override;
153
Erik Språng04e1bab2020-05-07 18:18:32 +0200154 size_t ExpectedPerPacketOverhead() const override;
155
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000156 // RTCP part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000157
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000158 // Get RTCP status.
pbosda903ea2015-10-02 02:36:56 -0700159 RtcpMode RTCP() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000160
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000161 // Configure RTCP status i.e on/off.
pbosda903ea2015-10-02 02:36:56 -0700162 void SetRTCPStatus(RtcpMode method) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000163
164 // Set RTCP CName.
Peter Boström9ba52f82015-06-01 14:12:28 +0200165 int32_t SetCNAME(const char* c_name) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000166
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000167 // Get remote NTP.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000168 int32_t RemoteNTP(uint32_t* received_ntp_secs,
169 uint32_t* received_ntp_frac,
170 uint32_t* rtcp_arrival_time_secs,
171 uint32_t* rtcp_arrival_time_frac,
172 uint32_t* rtcp_timestamp) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000173
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000174 // Get RoundTripTime.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000175 int32_t RTT(uint32_t remote_ssrc,
176 int64_t* rtt,
177 int64_t* avg_rtt,
178 int64_t* min_rtt,
179 int64_t* max_rtt) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000180
Niels Möller5fe95102019-03-04 16:49:25 +0100181 int64_t ExpectedRetransmissionTimeMs() const override;
182
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000183 // Force a send of an RTCP packet.
184 // Normal SR and RR are triggered via the process function.
Erik Språng242e22b2015-05-11 10:17:43 +0200185 int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
186
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000187 void GetSendStreamDataCounters(
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000188 StreamDataCounters* rtp_counters,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000189 StreamDataCounters* rtx_counters) const override;
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000190
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000191 // Get received RTCP report, report block.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000192 int32_t RemoteRTCPStat(
193 std::vector<RTCPReportBlock>* receive_blocks) const override;
Henrik Boström6e436d12019-05-27 12:19:33 +0200194 // A snapshot of the most recent Report Block with additional data of
195 // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
196 // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
197 // which is the SSRC of the corresponding outbound RTP stream, is unique.
198 std::vector<ReportBlockData> GetLatestReportBlockData() 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 Språng77b75292019-10-28 15:51:36 +0100275 // Handles final time timestamping/stats/etc and handover to Transport.
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +0200276 DEPRECATED_RtpSenderEgress packet_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100277 // If no paced sender configured, this class will be used to pass packets
278 // from |packet_generator_| to |packet_sender_|.
Tomas Gunnarsson593e6a42020-06-07 22:32:31 +0200279 DEPRECATED_RtpSenderEgress::NonPacedPacketSender non_paced_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100280 // Handles creation of RTP packets to be sent.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100281 RTPSender packet_generator;
Erik Språng77b75292019-10-28 15:51:36 +0100282 };
283
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000284 void set_rtt_ms(int64_t rtt_ms);
285 int64_t rtt_ms() const;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000286
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000287 bool TimeToSendFullNackList(int64_t now) const;
288
Niels Mölleraf6ea0c2020-11-20 12:21:21 +0100289 // Returns true if the module is configured to store packets.
290 bool StorePackets() const;
291
292 // Returns current Receiver Reference Time Report (RTTR) status.
293 bool RtcpXrRrtrStatus() const;
294
Erik Språng77b75292019-10-28 15:51:36 +0100295 std::unique_ptr<RtpSenderContext> rtp_sender_;
296
nisse150708e2017-03-16 05:02:53 -0700297 RTCPSender rtcp_sender_;
298 RTCPReceiver rtcp_receiver_;
299
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100300 Clock* const clock_;
nisse150708e2017-03-16 05:02:53 -0700301
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000302 int64_t last_bitrate_process_time_;
303 int64_t last_rtt_process_time_;
sprang168794c2017-07-06 04:38:06 -0700304 int64_t next_process_time_;
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000305 uint16_t packet_overhead_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000306
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000307 // Send side
Danil Chapovalov9eb6ce12017-12-15 12:25:01 +0100308 int64_t nack_last_time_sent_full_ms_;
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000309 uint16_t nack_last_seq_number_sent_;
stefan@webrtc.org9354cc92012-06-07 08:10:14 +0000310
Niels Möller5fe95102019-03-04 16:49:25 +0100311 RemoteBitrateEstimator* const remote_bitrate_;
312
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_