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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
Yves Gerey988cc082018-10-23 12:03:01 +020029#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020030#include "modules/rtp_rtcp/source/rtcp_receiver.h"
31#include "modules/rtp_rtcp/source/rtcp_sender.h"
Erik Språng9c771c22019-06-17 16:31:53 +020032#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020033#include "modules/rtp_rtcp/source/rtp_sender.h"
Steve Anton10542f22019-01-11 09:11:00 -080034#include "rtc_base/critical_section.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020035#include "rtc_base/gtest_prod_util.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000036
niklase@google.com470e71d2011-07-07 08:21:25 +000037namespace webrtc {
38
Yves Gerey988cc082018-10-23 12:03:01 +020039class Clock;
40struct PacedPacketInfo;
41struct RTPVideoHeader;
42
danilchap59cb2bd2016-08-29 11:08:47 -070043class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +000044 public:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000045 explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
Danil Chapovalov2a5ce2b2018-02-07 09:38:31 +010046 ~ModuleRtpRtcpImpl() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000047
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000048 // Returns the number of milliseconds until the module want a worker thread to
49 // call Process.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000050 int64_t TimeUntilNextProcess() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000051
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000052 // Process any pending tasks such as timeouts.
pbosa26ac922016-02-25 04:50:01 -080053 void Process() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000054
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000055 // Receiver part.
niklase@google.com470e71d2011-07-07 08:21:25 +000056
stefan@webrtc.orga5cb98c2013-05-29 12:12:51 +000057 // Called when we receive an RTCP packet.
nisse479d3d72017-09-13 07:53:37 -070058 void IncomingRtcpPacket(const uint8_t* incoming_packet,
59 size_t incoming_packet_length) override;
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +000060
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000061 void SetRemoteSSRC(uint32_t ssrc) override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +000062
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000063 // Sender part.
Niels Möller5fe95102019-03-04 16:49:25 +010064 void RegisterSendPayloadFrequency(int payload_type,
65 int payload_frequency) override;
Peter Boström8b79b072016-02-26 16:31:37 +010066
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000067 int32_t DeRegisterSendPayload(int8_t payload_type) override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +000068
Johannes Kron9190b822018-10-29 11:22:05 +010069 void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
70
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000071 // Register RTP header extension.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000072 int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
73 uint8_t id) override;
Sebastian Janssonf39c8152019-10-14 17:32:21 +020074 void RegisterRtpHeaderExtension(absl::string_view uri, int id) override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000075 int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
Sebastian Janssonf39c8152019-10-14 17:32:21 +020076 void DeregisterSendRtpHeaderExtension(absl::string_view uri) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000077
Mirko Bonadei999a72a2019-07-12 17:33:46 +000078 bool SupportsPadding() const override;
79 bool SupportsRtxPayloadPadding() const override;
stefan53b6cc32017-02-03 08:13:57 -080080
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000081 // Get start timestamp.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000082 uint32_t StartTimestamp() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000083
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000084 // Configure start timestamp, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000085 void SetStartTimestamp(uint32_t timestamp) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000086
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000087 uint16_t SequenceNumber() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000088
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000089 // Set SequenceNumber, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000090 void SetSequenceNumber(uint16_t seq) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000091
Per83d09102016-04-15 14:59:13 +020092 void SetRtpState(const RtpState& rtp_state) override;
93 void SetRtxState(const RtpState& rtp_state) override;
94 RtpState GetRtpState() const override;
95 RtpState GetRtxState() const override;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000096
Erik Språng6841d252019-10-15 14:29:11 +020097 uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
niklase@google.com470e71d2011-07-07 08:21:25 +000098
Amit Hilbuch77938e62018-12-21 09:23:38 -080099 void SetRid(const std::string& rid) override;
100
Steve Anton296a0ce2018-03-22 15:17:27 -0700101 void SetMid(const std::string& mid) override;
102
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000103 void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000104
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000105 RTCPSender::FeedbackState GetFeedbackState();
niklase@google.com470e71d2011-07-07 08:21:25 +0000106
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000107 void SetRtxSendStatus(int mode) override;
108 int RtxSendStatus() 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
Sebastian Jansson1bca65b2018-10-10 09:58:08 +0200125 void SetAsPartOfAllocation(bool part_of_allocation) override;
126
Niels Möller5fe95102019-03-04 16:49:25 +0100127 bool OnSendingRtpFrame(uint32_t timestamp,
128 int64_t capture_time_ms,
129 int payload_type,
130 bool force_sender_report) override;
131
Erik Språng9c771c22019-06-17 16:31:53 +0200132 bool TrySendPacket(RtpPacketToSend* packet,
133 const PacedPacketInfo& pacing_info) override;
134
stefan@webrtc.org0a3c1472013-12-05 14:05:07 +0000135
Erik Språngf6468d22019-07-05 16:53:43 +0200136 std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
137 size_t target_size_bytes) override;
Erik Språng478cb462019-06-26 15:49:27 +0200138
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000139 // RTCP part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000140
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000141 // Get RTCP status.
pbosda903ea2015-10-02 02:36:56 -0700142 RtcpMode RTCP() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000143
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000144 // Configure RTCP status i.e on/off.
pbosda903ea2015-10-02 02:36:56 -0700145 void SetRTCPStatus(RtcpMode method) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000146
147 // Set RTCP CName.
Peter Boström9ba52f82015-06-01 14:12:28 +0200148 int32_t SetCNAME(const char* c_name) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000149
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000150 // Get remote CName.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000151 int32_t RemoteCNAME(uint32_t remote_ssrc,
152 char c_name[RTCP_CNAME_SIZE]) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000153
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000154 // Get remote NTP.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000155 int32_t RemoteNTP(uint32_t* received_ntp_secs,
156 uint32_t* received_ntp_frac,
157 uint32_t* rtcp_arrival_time_secs,
158 uint32_t* rtcp_arrival_time_frac,
159 uint32_t* rtcp_timestamp) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000160
Erik Språng0ea42d32015-06-25 14:46:16 +0200161 int32_t AddMixedCNAME(uint32_t ssrc, const char* c_name) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000162
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000163 int32_t RemoveMixedCNAME(uint32_t ssrc) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000164
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000165 // Get RoundTripTime.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000166 int32_t RTT(uint32_t remote_ssrc,
167 int64_t* rtt,
168 int64_t* avg_rtt,
169 int64_t* min_rtt,
170 int64_t* max_rtt) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000171
Niels Möller5fe95102019-03-04 16:49:25 +0100172 int64_t ExpectedRetransmissionTimeMs() const override;
173
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000174 // Force a send of an RTCP packet.
175 // Normal SR and RR are triggered via the process function.
Erik Språng242e22b2015-05-11 10:17:43 +0200176 int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
177
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000178 // Statistics of the amount of data sent and received.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000179 int32_t DataCountersRTP(size_t* bytes_sent,
180 uint32_t* packets_sent) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000181
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000182 void GetSendStreamDataCounters(
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000183 StreamDataCounters* rtp_counters,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000184 StreamDataCounters* rtx_counters) const override;
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000185
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000186 // Get received RTCP report, report block.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000187 int32_t RemoteRTCPStat(
188 std::vector<RTCPReportBlock>* receive_blocks) const override;
Henrik Boström6e436d12019-05-27 12:19:33 +0200189 // A snapshot of the most recent Report Block with additional data of
190 // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
191 // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
192 // which is the SSRC of the corresponding outbound RTP stream, is unique.
193 std::vector<ReportBlockData> GetLatestReportBlockData() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000194
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000195 // (REMB) Receiver Estimated Max Bitrate.
Danil Chapovalov1de4b622017-12-13 13:35:10 +0100196 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
Danil Chapovalov51e21aa2017-10-10 17:46:26 +0200197 void UnsetRemb() override;
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000198
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000199 // (TMMBR) Temporary Max Media Bit Rate.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000200 bool TMMBR() const override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +0000201
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000202 void SetTMMBRStatus(bool enable) override;
niklase@google.com470e71d2011-07-07 08:21:25 +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
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000222 bool StorePackets() const override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000223
sprang@webrtc.orga6ad6e52013-12-05 09:48:44 +0000224 // Called on receipt of RTCP report block from remote side.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000225 void RegisterRtcpStatisticsCallback(
226 RtcpStatisticsCallback* callback) override;
227 RtcpStatisticsCallback* GetRtcpStatisticsCallback() override;
Niels Möller4d7c4052019-08-05 12:45:19 +0200228 void RegisterRtcpCnameCallback(RtcpCnameCallback* callback) override;
229
Henrik Boström87e3f9d2019-05-27 10:44:24 +0200230 void SetReportBlockDataObserver(ReportBlockDataObserver* observer) override;
sprang@webrtc.orga6ad6e52013-12-05 09:48:44 +0000231
sprang233bd872015-09-08 13:25:16 -0700232 bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override;
Sebastian Janssone1795f42019-07-24 11:38:03 +0200233 bool SendNetworkStateEstimatePacket(
234 const rtcp::RemoteEstimate& packet) override;
Per Kjellander16999812019-10-10 12:57:28 +0200235 void SendCombinedRtcpPacket(
236 std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override;
237
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000238 // (APP) Application specific data.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000239 int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
240 uint32_t name,
241 const uint8_t* data,
242 uint16_t length) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000243
asapersson@webrtc.org7d6bd222013-10-31 12:14:34 +0000244 // (XR) Receiver reference time report.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000245 void SetRtcpXrRrtrStatus(bool enable) override;
asapersson@webrtc.org7d6bd222013-10-31 12:14:34 +0000246
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000247 bool RtcpXrRrtrStatus() const override;
asapersson@webrtc.org8d02f5d2013-11-21 08:57:04 +0000248
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000249 // Video part.
Elad Alon7d6a4c02019-02-25 13:00:51 +0100250 int32_t SendLossNotification(uint16_t last_decoded_seq_num,
251 uint16_t last_received_seq_num,
Elad Alone86af2c2019-06-03 14:37:50 +0200252 bool decodability_flag,
253 bool buffering_allowed) override;
Elad Alon7d6a4c02019-02-25 13:00:51 +0100254
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000255 bool LastReceivedNTP(uint32_t* NTPsecs,
256 uint32_t* NTPfrac,
257 uint32_t* remote_sr) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000258
danilchap2b616392016-08-18 06:17:42 -0700259 std::vector<rtcp::TmmbItem> BoundingSet(bool* tmmbr_owner);
niklase@google.com470e71d2011-07-07 08:21:25 +0000260
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000261 void BitrateSent(uint32_t* total_rate,
262 uint32_t* video_rate,
263 uint32_t* fec_rate,
264 uint32_t* nackRate) const override;
stefan@webrtc.org07b45a52012-02-02 08:37:48 +0000265
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000266 void RegisterSendChannelRtpStatisticsCallback(
267 StreamDataCountersCallback* callback) override;
268 StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
269 const override;
sprang@webrtc.orgebad7652013-12-05 14:29:02 +0000270
danilchap59cb2bd2016-08-29 11:08:47 -0700271 void OnReceivedNack(
272 const std::vector<uint16_t>& nack_sequence_numbers) override;
273 void OnReceivedRtcpReportBlocks(
274 const ReportBlockList& report_blocks) override;
275 void OnRequestSendReport() override;
henrike@webrtc.orgf5da4da2012-02-15 23:54:59 +0000276
Erik Språng566124a2018-04-23 12:32:22 +0200277 void SetVideoBitrateAllocation(
278 const VideoBitrateAllocation& bitrate) override;
sprang5e38c962016-12-01 05:18:09 -0800279
Niels Möller5fe95102019-03-04 16:49:25 +0100280 RTPSender* RtpSender() override;
281 const RTPSender* RtpSender() const override;
282
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000283 protected:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000284 bool UpdateRTCPReceiveInformationTimers();
niklase@google.com470e71d2011-07-07 08:21:25 +0000285
nisse14adba72017-03-20 03:52:39 -0700286 RTPSender* rtp_sender() { return rtp_sender_.get(); }
287 const RTPSender* rtp_sender() const { return rtp_sender_.get(); }
nissea33c62e2017-03-14 00:49:45 -0700288
289 RTCPSender* rtcp_sender() { return &rtcp_sender_; }
290 const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
291
292 RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
293 const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
294
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100295 Clock* clock() const { return clock_; }
nissea33c62e2017-03-14 00:49:45 -0700296
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000297 private:
asapersson@webrtc.orge7b1e112013-12-16 14:40:36 +0000298 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000299 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
niklase@google.com470e71d2011-07-07 08:21:25 +0000300
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000301 void set_rtt_ms(int64_t rtt_ms);
302 int64_t rtt_ms() const;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000303
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000304 bool TimeToSendFullNackList(int64_t now) const;
305
nisse14adba72017-03-20 03:52:39 -0700306 std::unique_ptr<RTPSender> rtp_sender_;
nisse150708e2017-03-16 05:02:53 -0700307 RTCPSender rtcp_sender_;
308 RTCPReceiver rtcp_receiver_;
309
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100310 Clock* const clock_;
nisse150708e2017-03-16 05:02:53 -0700311
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000312 int64_t last_bitrate_process_time_;
313 int64_t last_rtt_process_time_;
sprang168794c2017-07-06 04:38:06 -0700314 int64_t next_process_time_;
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000315 uint16_t packet_overhead_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000316
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000317 // Send side
Danil Chapovalov9eb6ce12017-12-15 12:25:01 +0100318 int64_t nack_last_time_sent_full_ms_;
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000319 uint16_t nack_last_seq_number_sent_;
stefan@webrtc.org9354cc92012-06-07 08:10:14 +0000320
Niels Möller5fe95102019-03-04 16:49:25 +0100321 RemoteBitrateEstimator* const remote_bitrate_;
322
323 RtcpAckObserver* const ack_observer_;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000324
Tommi5f223652018-03-26 13:28:26 +0200325 RtcpRttStats* const rtt_stats_;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000326
327 // The processed RTT from RtcpRttStats.
danilchap7c9426c2016-04-14 03:05:31 -0700328 rtc::CriticalSection critical_section_rtt_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000329 int64_t rtt_ms_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000330};
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000331
332} // namespace webrtc
333
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200334#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_