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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>
nisse14adba72017-03-20 03:52:39 -070016#include <memory>
danilchapb8b6fbb2015-12-10 05:05:27 -080017#include <set>
Steve Anton296a0ce2018-03-22 15:17:27 -070018#include <string>
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000019#include <vector>
niklase@google.com470e71d2011-07-07 08:21:25 +000020
Danil Chapovalovd264df52018-06-14 12:59:38 +020021#include "absl/types/optional.h"
Yves Gerey988cc082018-10-23 12:03:01 +020022#include "api/rtp_headers.h"
Erik Språngeeaa8f92018-05-17 12:35:56 +020023#include "api/video/video_bitrate_allocation.h"
Yves Gerey988cc082018-10-23 12:03:01 +020024#include "common_types.h" // NOLINT(build/include)
25#include "modules/include/module_common_types.h"
26#include "modules/include/module_fec_types.h"
27#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020028#include "modules/rtp_rtcp/include/rtp_rtcp.h"
Yves Gerey988cc082018-10-23 12:03:01 +020029#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020030#include "modules/rtp_rtcp/source/packet_loss_stats.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"
34#include "modules/rtp_rtcp/source/rtp_sender.h"
35#include "rtc_base/criticalsection.h"
36#include "rtc_base/gtest_prod_util.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000037
niklase@google.com470e71d2011-07-07 08:21:25 +000038namespace webrtc {
39
Yves Gerey988cc082018-10-23 12:03:01 +020040class Clock;
41struct PacedPacketInfo;
42struct RTPVideoHeader;
43
danilchap59cb2bd2016-08-29 11:08:47 -070044class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
pwestin@webrtc.org2853dde2012-05-11 11:08:54 +000045 public:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000046 explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
Danil Chapovalov2a5ce2b2018-02-07 09:38:31 +010047 ~ModuleRtpRtcpImpl() override;
niklase@google.com470e71d2011-07-07 08:21:25 +000048
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000049 // Returns the number of milliseconds until the module want a worker thread to
50 // call Process.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000051 int64_t TimeUntilNextProcess() 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;
wu@webrtc.org822fbd82013-08-15 23:38:54 +000063
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000064 // Sender part.
pwestin@webrtc.org1da1ce02011-10-13 15:19:55 +000065
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000066 int32_t RegisterSendPayload(const CodecInst& voice_codec) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000067
Peter Boström8b79b072016-02-26 16:31:37 +010068 void RegisterVideoSendPayload(int payload_type,
69 const char* payload_name) override;
70
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
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000073 // Register RTP header extension.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000074 int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
75 uint8_t id) override;
Danil Chapovalov585d1aa2018-09-14 18:29:32 +020076 bool RegisterRtpHeaderExtension(const std::string& uri, int id) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000077
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000078 int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000079
stefan53b6cc32017-02-03 08:13:57 -080080 bool HasBweExtensions() const override;
81
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000082 // Get start timestamp.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000083 uint32_t StartTimestamp() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000084
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000085 // Configure start timestamp, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000086 void SetStartTimestamp(uint32_t timestamp) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000087
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000088 uint16_t SequenceNumber() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000089
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +000090 // Set SequenceNumber, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000091 void SetSequenceNumber(uint16_t seq) override;
niklase@google.com470e71d2011-07-07 08:21:25 +000092
Per83d09102016-04-15 14:59:13 +020093 void SetRtpState(const RtpState& rtp_state) override;
94 void SetRtxState(const RtpState& rtp_state) override;
95 RtpState GetRtpState() const override;
96 RtpState GetRtxState() const override;
pbos@webrtc.org2bb1bda2014-07-07 13:06:48 +000097
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000098 uint32_t SSRC() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +000099
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000100 // Configure SSRC, default is a random number.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000101 void SetSSRC(uint32_t ssrc) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000102
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;
mflodman@webrtc.org9f5ebb52013-04-12 14:55:46 +0000111
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000112 void SetRtxSsrc(uint32_t ssrc) 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
Sebastian Jansson1bca65b2018-10-10 09:58:08 +0200129 void SetAsPartOfAllocation(bool part_of_allocation) override;
130
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000131 // Used by the codec module to deliver a video or audio frame for
132 // packetization.
Sergey Ulanov525df3f2016-08-02 17:46:41 -0700133 bool SendOutgoingData(FrameType frame_type,
134 int8_t payload_type,
135 uint32_t time_stamp,
136 int64_t capture_time_ms,
137 const uint8_t* payload_data,
138 size_t payload_size,
139 const RTPFragmentationHeader* fragmentation,
140 const RTPVideoHeader* rtp_video_header,
141 uint32_t* transport_frame_id_out) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000142
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000143 bool TimeToSendPacket(uint32_t ssrc,
144 uint16_t sequence_number,
145 int64_t capture_time_ms,
philipela1ed0b32016-06-01 06:31:17 -0700146 bool retransmission,
philipelc7bf32a2017-02-17 03:59:43 -0800147 const PacedPacketInfo& pacing_info) override;
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000148
stefan@webrtc.org508a84b2013-06-17 12:53:37 +0000149 // Returns the number of padding bytes actually sent, which can be more or
150 // less than |bytes|.
philipelc7bf32a2017-02-17 03:59:43 -0800151 size_t TimeToSendPadding(size_t bytes,
152 const PacedPacketInfo& pacing_info) override;
stefan@webrtc.org0a3c1472013-12-05 14:05:07 +0000153
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000154 // RTCP part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000155
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000156 // Get RTCP status.
pbosda903ea2015-10-02 02:36:56 -0700157 RtcpMode RTCP() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000158
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000159 // Configure RTCP status i.e on/off.
pbosda903ea2015-10-02 02:36:56 -0700160 void SetRTCPStatus(RtcpMode method) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000161
162 // Set RTCP CName.
Peter Boström9ba52f82015-06-01 14:12:28 +0200163 int32_t SetCNAME(const char* c_name) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000164
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000165 // Get remote CName.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000166 int32_t RemoteCNAME(uint32_t remote_ssrc,
167 char c_name[RTCP_CNAME_SIZE]) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000168
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000169 // Get remote NTP.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000170 int32_t RemoteNTP(uint32_t* received_ntp_secs,
171 uint32_t* received_ntp_frac,
172 uint32_t* rtcp_arrival_time_secs,
173 uint32_t* rtcp_arrival_time_frac,
174 uint32_t* rtcp_timestamp) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000175
Erik Språng0ea42d32015-06-25 14:46:16 +0200176 int32_t AddMixedCNAME(uint32_t ssrc, const char* c_name) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000177
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000178 int32_t RemoveMixedCNAME(uint32_t ssrc) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000179
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000180 // Get RoundTripTime.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000181 int32_t RTT(uint32_t remote_ssrc,
182 int64_t* rtt,
183 int64_t* avg_rtt,
184 int64_t* min_rtt,
185 int64_t* max_rtt) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000186
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
191 int32_t SendCompoundRTCP(
192 const std::set<RTCPPacketType>& rtcpPacketTypes) override;
mflodman@webrtc.org7c894b72012-11-26 12:40:15 +0000193
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000194 // Statistics of the amount of data sent and received.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000195 int32_t DataCountersRTP(size_t* bytes_sent,
196 uint32_t* packets_sent) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000197
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000198 void GetSendStreamDataCounters(
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000199 StreamDataCounters* rtp_counters,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000200 StreamDataCounters* rtx_counters) const override;
asapersson@webrtc.org97d04892014-12-09 09:47:53 +0000201
bcornell30409b42015-07-10 18:10:05 -0700202 void GetRtpPacketLossStats(
203 bool outgoing,
204 uint32_t ssrc,
205 struct RtpPacketLossStats* loss_stats) const override;
206
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000207 // Get received RTCP report, report block.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000208 int32_t RemoteRTCPStat(
209 std::vector<RTCPReportBlock>* receive_blocks) const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000210
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000211 // (REMB) Receiver Estimated Max Bitrate.
Danil Chapovalov1de4b622017-12-13 13:35:10 +0100212 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;
Danil Chapovalov51e21aa2017-10-10 17:46:26 +0200213 void UnsetRemb() override;
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000214
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000215 // (TMMBR) Temporary Max Media Bit Rate.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000216 bool TMMBR() const override;
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +0000217
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000218 void SetTMMBRStatus(bool enable) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000219
danilchap59cb2bd2016-08-29 11:08:47 -0700220 void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000221
nisse284542b2017-01-10 08:58:32 -0800222 size_t MaxRtpPacketSize() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000223
nisse284542b2017-01-10 08:58:32 -0800224 void SetMaxRtpPacketSize(size_t max_packet_size) override;
michaelt79e05882016-11-08 02:50:09 -0800225
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000226 // (NACK) Negative acknowledgment part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000227
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000228 int SelectiveRetransmissions() const override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000229
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000230 int SetSelectiveRetransmissions(uint8_t settings) override;
stefan@webrtc.org6a4bef42011-12-22 12:52:41 +0000231
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000232 // Send a Negative acknowledgment packet.
philipel83f831a2016-03-12 03:30:23 -0800233 // TODO(philipel): Deprecate SendNACK and use SendNack instead.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000234 int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override;
stefan@webrtc.org6a4bef42011-12-22 12:52:41 +0000235
philipel83f831a2016-03-12 03:30:23 -0800236 void SendNack(const std::vector<uint16_t>& sequence_numbers) override;
237
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000238 // Store the sent packets, needed to answer to a negative acknowledgment
239 // requests.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000240 void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000241
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000242 bool StorePackets() const override;
wu@webrtc.org822fbd82013-08-15 23:38:54 +0000243
sprang@webrtc.orga6ad6e52013-12-05 09:48:44 +0000244 // Called on receipt of RTCP report block from remote side.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000245 void RegisterRtcpStatisticsCallback(
246 RtcpStatisticsCallback* callback) override;
247 RtcpStatisticsCallback* GetRtcpStatisticsCallback() override;
sprang@webrtc.orga6ad6e52013-12-05 09:48:44 +0000248
sprang233bd872015-09-08 13:25:16 -0700249 bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override;
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000250 // (APP) Application specific data.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000251 int32_t SetRTCPApplicationSpecificData(uint8_t sub_type,
252 uint32_t name,
253 const uint8_t* data,
254 uint16_t length) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000255
asapersson@webrtc.org7d6bd222013-10-31 12:14:34 +0000256 // (XR) Receiver reference time report.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000257 void SetRtcpXrRrtrStatus(bool enable) override;
asapersson@webrtc.org7d6bd222013-10-31 12:14:34 +0000258
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000259 bool RtcpXrRrtrStatus() const override;
asapersson@webrtc.org8d02f5d2013-11-21 08:57:04 +0000260
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000261 // Audio part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000262
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000263 // Send a TelephoneEvent tone using RFC 2833 (4733).
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000264 int32_t SendTelephoneEventOutband(uint8_t key,
265 uint16_t time_ms,
266 uint8_t level) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000267
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000268 // Store the audio level in d_bov for header-extension-for-audio-level-
269 // indication.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000270 int32_t SetAudioLevel(uint8_t level_d_bov) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000271
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000272 // Video part.
niklase@google.com470e71d2011-07-07 08:21:25 +0000273
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000274 // Set method for requesting a new key frame.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000275 int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000276
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000277 // Send a request for a keyframe.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000278 int32_t RequestKeyFrame() override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000279
brandtrf1bb4762016-11-07 03:05:06 -0800280 void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000281
brandtr1743a192016-11-07 03:36:05 -0800282 bool SetFecParameters(const FecProtectionParams& delta_params,
283 const FecProtectionParams& key_params) override;
niklase@google.com470e71d2011-07-07 08:21:25 +0000284
henrik.lundin@webrtc.org1972ff82014-09-11 06:20:28 +0000285 bool LastReceivedNTP(uint32_t* NTPsecs,
286 uint32_t* NTPfrac,
287 uint32_t* remote_sr) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000288
danilchap2b616392016-08-18 06:17:42 -0700289 std::vector<rtcp::TmmbItem> BoundingSet(bool* tmmbr_owner);
niklase@google.com470e71d2011-07-07 08:21:25 +0000290
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000291 void BitrateSent(uint32_t* total_rate,
292 uint32_t* video_rate,
293 uint32_t* fec_rate,
294 uint32_t* nackRate) const override;
stefan@webrtc.org07b45a52012-02-02 08:37:48 +0000295
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000296 void RegisterSendChannelRtpStatisticsCallback(
297 StreamDataCountersCallback* callback) override;
298 StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback()
299 const override;
sprang@webrtc.orgebad7652013-12-05 14:29:02 +0000300
danilchap59cb2bd2016-08-29 11:08:47 -0700301 void OnReceivedNack(
302 const std::vector<uint16_t>& nack_sequence_numbers) override;
303 void OnReceivedRtcpReportBlocks(
304 const ReportBlockList& report_blocks) override;
305 void OnRequestSendReport() override;
henrike@webrtc.orgf5da4da2012-02-15 23:54:59 +0000306
Erik Språng566124a2018-04-23 12:32:22 +0200307 void SetVideoBitrateAllocation(
308 const VideoBitrateAllocation& bitrate) override;
sprang5e38c962016-12-01 05:18:09 -0800309
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000310 protected:
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000311 bool UpdateRTCPReceiveInformationTimers();
niklase@google.com470e71d2011-07-07 08:21:25 +0000312
nisse14adba72017-03-20 03:52:39 -0700313 RTPSender* rtp_sender() { return rtp_sender_.get(); }
314 const RTPSender* rtp_sender() const { return rtp_sender_.get(); }
nissea33c62e2017-03-14 00:49:45 -0700315
316 RTCPSender* rtcp_sender() { return &rtcp_sender_; }
317 const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
318
319 RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
320 const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
321
322 const Clock* clock() const { return clock_; }
323
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000324 private:
asapersson@webrtc.orge7b1e112013-12-16 14:40:36 +0000325 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000326 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000327 int64_t RtcpReportInterval();
stefan@webrtc.org28a331e2013-09-17 07:49:56 +0000328 void SetRtcpReceiverSsrcs(uint32_t main_ssrc);
niklase@google.com470e71d2011-07-07 08:21:25 +0000329
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000330 void set_rtt_ms(int64_t rtt_ms);
331 int64_t rtt_ms() const;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000332
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000333 bool TimeToSendFullNackList(int64_t now) const;
334
nisse14adba72017-03-20 03:52:39 -0700335 std::unique_ptr<RTPSender> rtp_sender_;
nisse150708e2017-03-16 05:02:53 -0700336 RTCPSender rtcp_sender_;
337 RTCPReceiver rtcp_receiver_;
338
339 const Clock* const clock_;
340
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000341 const bool audio_;
sprang168794c2017-07-06 04:38:06 -0700342
343 const RtpKeepAliveConfig keepalive_config_;
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000344 int64_t last_bitrate_process_time_;
345 int64_t last_rtt_process_time_;
sprang168794c2017-07-06 04:38:06 -0700346 int64_t next_process_time_;
347 int64_t next_keepalive_time_;
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +0000348 uint16_t packet_overhead_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000349
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000350 // Send side
Danil Chapovalov9eb6ce12017-12-15 12:25:01 +0100351 int64_t nack_last_time_sent_full_ms_;
asapersson@webrtc.orgba8138b2014-12-08 13:29:02 +0000352 uint16_t nack_last_seq_number_sent_;
stefan@webrtc.org9354cc92012-06-07 08:10:14 +0000353
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000354 KeyFrameRequestMethod key_frame_req_method_;
355
356 RemoteBitrateEstimator* remote_bitrate_;
357
Tommi5f223652018-03-26 13:28:26 +0200358 RtcpRttStats* const rtt_stats_;
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000359
bcornell30409b42015-07-10 18:10:05 -0700360 PacketLossStats send_loss_stats_;
361 PacketLossStats receive_loss_stats_;
362
asapersson@webrtc.org1ae1d0c2013-11-20 12:46:11 +0000363 // The processed RTT from RtcpRttStats.
danilchap7c9426c2016-04-14 03:05:31 -0700364 rtc::CriticalSection critical_section_rtt_;
pkasting@chromium.org16825b12015-01-12 21:51:21 +0000365 int64_t rtt_ms_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000366};
phoglund@webrtc.orgacfdd962013-01-16 10:27:33 +0000367
368} // namespace webrtc
369
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200370#endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_