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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 Språng77b75292019-10-28 15:51:36 +010032#include "modules/rtp_rtcp/source/rtp_packet_history.h"
Erik Språng9c771c22019-06-17 16:31:53 +020033#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020034#include "modules/rtp_rtcp/source/rtp_sender.h"
Erik Språng77b75292019-10-28 15:51:36 +010035#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
Steve Anton10542f22019-01-11 09:11:00 -080036#include "rtc_base/critical_section.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020037#include "rtc_base/gtest_prod_util.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:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000048 explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
Danil Chapovalov2a5ce2b2018-02-07 09:38:31 +010049 ~ModuleRtpRtcpImpl() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000050
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000051 // Returns the number of milliseconds until the module want a worker thread to
52 // call Process.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000053 int64_t TimeUntilNextProcess() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000054
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000055 // Process any pending tasks such as timeouts.
pbosa26ac922016-02-25 04:50:01 -080056 void Process() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000057
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000058 // Receiver part.
niklase@google.com470e71d2011-07-07 08:21:25 +000059
stefan@webrtc.orga5cb98c2013-05-29 12:12:51 +000060 // Called when we receive an RTCP packet.
nisse479d3d72017-09-13 07:53:37 -070061 void IncomingRtcpPacket(const uint8_t* incoming_packet,
62 size_t incoming_packet_length) override;
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +000063
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000064 void SetRemoteSSRC(uint32_t ssrc) override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +000065
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000066 // Sender part.
Niels Möller5fe95102019-03-04 16:49:25 +010067 void RegisterSendPayloadFrequency(int payload_type,
68 int payload_frequency) override;
Peter Boström8b79b072016-02-26 16:31:37 +010069
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000070 int32_t DeRegisterSendPayload(int8_t payload_type) override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +000071
Johannes Kron9190b822018-10-29 11:22:05 +010072 void SetExtmapAllowMixed(bool extmap_allow_mixed) override;
73
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000074 // Register RTP header extension.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000075 int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
76 uint8_t id) override;
Sebastian Janssonf39c8152019-10-14 17:32:21 +020077 void RegisterRtpHeaderExtension(absl::string_view uri, int id) override;
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000078 int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
Sebastian Janssonf39c8152019-10-14 17:32:21 +020079 void DeregisterSendRtpHeaderExtension(absl::string_view uri) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000080
Mirko Bonadei999a72a2019-07-12 17:33:46 +000081 bool SupportsPadding() const override;
82 bool SupportsRtxPayloadPadding() const override;
stefan53b6cc32017-02-03 08:13:57 -080083
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000084 // Get start timestamp.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000085 uint32_t StartTimestamp() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000086
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000087 // Configure start timestamp, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000088 void SetStartTimestamp(uint32_t timestamp) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000089
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000090 uint16_t SequenceNumber() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000091
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000092 // Set SequenceNumber, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000093 void SetSequenceNumber(uint16_t seq) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000094
Per83d09102016-04-15 14:59:13 +020095 void SetRtpState(const RtpState& rtp_state) override;
96 void SetRtxState(const RtpState& rtp_state) override;
97 RtpState GetRtpState() const override;
98 RtpState GetRtxState() const override;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000099
Erik Språng6841d252019-10-15 14:29:11 +0200100 uint32_t SSRC() const override { return rtcp_sender_.SSRC(); }
niklase@google.com470e71d2011-07-07 08:21:25 +0000101
Amit Hilbuch77938e62018-12-21 09:23:38 -0800102 void SetRid(const std::string& rid) override;
103
Steve Anton296a0ce2018-03-22 15:17:27 -0700104 void SetMid(const std::string& mid) override;
105
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000106 void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000107
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000108 RTCPSender::FeedbackState GetFeedbackState();
niklase@google.com470e71d2011-07-07 08:21:25 +0000109
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000110 void SetRtxSendStatus(int mode) override;
111 int RtxSendStatus() const override;
Erik Språngc06aef22019-10-17 13:02:27 +0200112 absl::optional<uint32_t> RtxSsrc() const override;
mflodman@webrtc.org9f5ebb52013-04-12 14:55:46 +0000113
Shao Changbine62202f2015-04-21 20:24:50 +0800114 void SetRtxSendPayloadType(int payload_type,
115 int associated_payload_type) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000116
Danil Chapovalovd264df52018-06-14 12:59:38 +0200117 absl::optional<uint32_t> FlexfecSsrc() const override;
brandtr9dfff292016-11-14 05:14:50 -0800118
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000119 // Sends kRtcpByeCode when going from true to false.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000120 int32_t SetSendingStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000121
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000122 bool Sending() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000123
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000124 // Drops or relays media packets.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000125 void SetSendingMediaStatus(bool sending) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000126
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000127 bool SendingMedia() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000128
Erik Språng1e51a382019-12-11 16:47:09 +0100129 bool IsAudioConfigured() const override;
130
Sebastian Jansson1bca65b2018-10-10 09:58:08 +0200131 void SetAsPartOfAllocation(bool part_of_allocation) override;
132
Niels Möller5fe95102019-03-04 16:49:25 +0100133 bool OnSendingRtpFrame(uint32_t timestamp,
134 int64_t capture_time_ms,
135 int payload_type,
136 bool force_sender_report) override;
137
Erik Språng9c771c22019-06-17 16:31:53 +0200138 bool TrySendPacket(RtpPacketToSend* packet,
139 const PacedPacketInfo& pacing_info) override;
140
Erik Språnga9229042019-10-24 12:39:32 +0200141 void OnPacketsAcknowledged(
142 rtc::ArrayView<const uint16_t> sequence_numbers) override;
stefan@webrtc.org0a3c1472013-12-05 14:05:07 +0000143
Erik Språngf6468d22019-07-05 16:53:43 +0200144 std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
145 size_t target_size_bytes) override;
Erik Språng478cb462019-06-26 15:49:27 +0200146
Erik Språng3663f942020-02-07 10:05:15 +0100147 std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
148 rtc::ArrayView<const uint16_t> sequence_numbers) const override;
149
Erik Språng04e1bab2020-05-07 18:18:32 +0200150 size_t ExpectedPerPacketOverhead() const override;
151
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000152 // RTCP part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000153
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000154 // Get RTCP status.
pbosda903ea2015-10-02 02:36:56 -0700155 RtcpMode RTCP() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000156
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000157 // Configure RTCP status i.e on/off.
pbosda903ea2015-10-02 02:36:56 -0700158 void SetRTCPStatus(RtcpMode method) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000159
160 // Set RTCP CName.
Peter Boström9ba52f82015-06-01 14:12:28 +0200161 int32_t SetCNAME(const char* c_name) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000162
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000163 // Get remote CName.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000164 int32_t RemoteCNAME(uint32_t remote_ssrc,
165 char c_name[RTCP_CNAME_SIZE]) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +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
Erik Språng0ea42d32015-06-25 14:46:16 +0200174 int32_t AddMixedCNAME(uint32_t ssrc, const char* c_name) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000175
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000176 int32_t RemoveMixedCNAME(uint32_t ssrc) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000177
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000178 // Get RoundTripTime.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000179 int32_t RTT(uint32_t remote_ssrc,
180 int64_t* rtt,
181 int64_t* avg_rtt,
182 int64_t* min_rtt,
183 int64_t* max_rtt) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000184
Niels Möller5fe95102019-03-04 16:49:25 +0100185 int64_t ExpectedRetransmissionTimeMs() const override;
186
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000187 // Force a send of an RTCP packet.
188 // Normal SR and RR are triggered via the process function.
Erik Språng242e22b2015-05-11 10:17:43 +0200189 int32_t SendRTCP(RTCPPacketType rtcpPacketType) override;
190
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000191 // Statistics of the amount of data sent and received.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000192 int32_t DataCountersRTP(size_t* bytes_sent,
193 uint32_t* packets_sent) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000194
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000195 void GetSendStreamDataCounters(
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000196 StreamDataCounters* rtp_counters,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000197 StreamDataCounters* rtx_counters) const override;
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000198
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000199 // Get received RTCP report, report block.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000200 int32_t RemoteRTCPStat(
201 std::vector<RTCPReportBlock>* receive_blocks) const override;
Henrik Boström6e436d12019-05-27 12:19:33 +0200202 // A snapshot of the most recent Report Block with additional data of
203 // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
204 // Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
205 // which is the SSRC of the corresponding outbound RTP stream, is unique.
206 std::vector<ReportBlockData> GetLatestReportBlockData() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000207
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000208 // (REMB) Receiver Estimated Max Bitrate.
Danil Chapovalov1de4b622017-12-13 13:35:10 +0100209 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
Danil Chapovalov51e21aa2017-10-10 17:46:26 +0200210 void UnsetRemb() override;
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000211
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000212 // (TMMBR) Temporary Max Media Bit Rate.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000213 bool TMMBR() const override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +0000214
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000215 void SetTMMBRStatus(bool enable) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000216
danilchap59cb2bd2016-08-29 11:08:47 -0700217 void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000218
nisse284542b2017-01-10 08:58:32 -0800219 size_t MaxRtpPacketSize() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000220
nisse284542b2017-01-10 08:58:32 -0800221 void SetMaxRtpPacketSize(size_t max_packet_size) override;
michaelt79e05882016-11-08 02:50:09 -0800222
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000223 // (NACK) Negative acknowledgment part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000224
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000225 // Send a Negative acknowledgment packet.
philipel83f831a2016-03-12 03:30:23 -0800226 // TODO(philipel): Deprecate SendNACK and use SendNack instead.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000227 int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
stefan@webrtc.org6a4bef42011-12-22 12:52:41 +0000228
philipel83f831a2016-03-12 03:30:23 -0800229 void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
230
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000231 // Store the sent packets, needed to answer to a negative acknowledgment
232 // requests.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000233 void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000234
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000235 bool StorePackets() const override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000236
Per Kjellander16999812019-10-10 12:57:28 +0200237 void SendCombinedRtcpPacket(
238 std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override;
239
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000240 // (APP) Application specific data.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000241 int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
242 uint32_t name,
243 const uint8_t* data,
244 uint16_t length) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000245
asapersson@webrtc.org7d6bd222013-10-31 12:14:34 +0000246 // (XR) Receiver reference time report.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000247 void SetRtcpXrRrtrStatus(bool enable) override;
asapersson@webrtc.org7d6bd222013-10-31 12:14:34 +0000248
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000249 bool RtcpXrRrtrStatus() const override;
asapersson@webrtc.org8d02f5d2013-11-21 08:57:04 +0000250
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000251 // Video part.
Elad Alon7d6a4c02019-02-25 13:00:51 +0100252 int32_t SendLossNotification(uint16_t last_decoded_seq_num,
253 uint16_t last_received_seq_num,
Elad Alone86af2c2019-06-03 14:37:50 +0200254 bool decodability_flag,
255 bool buffering_allowed) override;
Elad Alon7d6a4c02019-02-25 13:00:51 +0100256
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000257 bool LastReceivedNTP(uint32_t* NTPsecs,
258 uint32_t* NTPfrac,
259 uint32_t* remote_sr) const;
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
Erik Språngbf46cfe2020-05-11 18:22:02 +0200266 RtpSendRates GetSendRates() const override;
267
danilchap59cb2bd2016-08-29 11:08:47 -0700268 void OnReceivedNack(
269 const std::vector<uint16_t>& nack_sequence_numbers) override;
270 void OnReceivedRtcpReportBlocks(
271 const ReportBlockList& report_blocks) override;
272 void OnRequestSendReport() override;
henrike@webrtc.orgf5da4da2012-02-15 23:54:59 +0000273
Erik Språng566124a2018-04-23 12:32:22 +0200274 void SetVideoBitrateAllocation(
275 const VideoBitrateAllocation& bitrate) override;
sprang5e38c962016-12-01 05:18:09 -0800276
Niels Möller5fe95102019-03-04 16:49:25 +0100277 RTPSender* RtpSender() override;
278 const RTPSender* RtpSender() const override;
279
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000280 protected:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000281 bool UpdateRTCPReceiveInformationTimers();
niklase@google.com470e71d2011-07-07 08:21:25 +0000282
Erik Språng77b75292019-10-28 15:51:36 +0100283 RTPSender* rtp_sender() {
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100284 return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
Erik Språng77b75292019-10-28 15:51:36 +0100285 }
286 const RTPSender* rtp_sender() const {
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100287 return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
Erik Språng77b75292019-10-28 15:51:36 +0100288 }
nissea33c62e2017-03-14 00:49:45 -0700289
290 RTCPSender* rtcp_sender() { return &rtcp_sender_; }
291 const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
292
293 RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
294 const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
295
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100296 Clock* clock() const { return clock_; }
nissea33c62e2017-03-14 00:49:45 -0700297
Erik Språngbf46cfe2020-05-11 18:22:02 +0200298 // TODO(sprang): Remove when usage is gone.
Erik Språngcff20c22019-10-28 12:28:16 +0100299 DataRate SendRate() const;
300 DataRate NackOverheadRate() const;
301
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000302 private:
asapersson@webrtc.orge7b1e112013-12-16 14:40:36 +0000303 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000304 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
niklase@google.com470e71d2011-07-07 08:21:25 +0000305
Erik Språng77b75292019-10-28 15:51:36 +0100306 struct RtpSenderContext {
307 explicit RtpSenderContext(const RtpRtcp::Configuration& config);
308 // Storage of packets, for retransmissions and padding, if applicable.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100309 RtpPacketHistory packet_history;
Erik Språng77b75292019-10-28 15:51:36 +0100310 // Handles final time timestamping/stats/etc and handover to Transport.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100311 RtpSenderEgress packet_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100312 // If no paced sender configured, this class will be used to pass packets
313 // from |packet_generator_| to |packet_sender_|.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100314 RtpSenderEgress::NonPacedPacketSender non_paced_sender;
Erik Språng77b75292019-10-28 15:51:36 +0100315 // Handles creation of RTP packets to be sent.
Erik Språng9cdc9cc2019-10-28 18:24:32 +0100316 RTPSender packet_generator;
Erik Språng77b75292019-10-28 15:51:36 +0100317 };
318
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000319 void set_rtt_ms(int64_t rtt_ms);
320 int64_t rtt_ms() const;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000321
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000322 bool TimeToSendFullNackList(int64_t now) const;
323
Erik Språng77b75292019-10-28 15:51:36 +0100324 std::unique_ptr<RtpSenderContext> rtp_sender_;
325
nisse150708e2017-03-16 05:02:53 -0700326 RTCPSender rtcp_sender_;
327 RTCPReceiver rtcp_receiver_;
328
Sebastian Janssonaa01f272019-01-30 11:28:59 +0100329 Clock* const clock_;
nisse150708e2017-03-16 05:02:53 -0700330
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000331 int64_t last_bitrate_process_time_;
332 int64_t last_rtt_process_time_;
sprang168794c2017-07-06 04:38:06 -0700333 int64_t next_process_time_;
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000334 uint16_t packet_overhead_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000335
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000336 // Send side
Danil Chapovalov9eb6ce12017-12-15 12:25:01 +0100337 int64_t nack_last_time_sent_full_ms_;
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000338 uint16_t nack_last_seq_number_sent_;
stefan@webrtc.org9354cc92012-06-07 08:10:14 +0000339
Niels Möller5fe95102019-03-04 16:49:25 +0100340 RemoteBitrateEstimator* const remote_bitrate_;
341
Tommi5f223652018-03-26 13:28:26 +0200342 RtcpRttStats* const rtt_stats_;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000343
344 // The processed RTT from RtcpRttStats.
danilchap7c9426c2016-04-14 03:05:31 -0700345 rtc::CriticalSection critical_section_rtt_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000346 int64_t rtt_ms_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000347};
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000348
349} // namespace webrtc
350
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200351#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_