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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
11#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
12#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
13
pwestin@webrtc.org26f8d9c2012-01-19 15:53:09 +000014#include <map>
Erik Språng242e22b2015-05-11 10:17:43 +020015#include <set>
edjee@google.com79b02892013-04-04 19:43:34 +000016#include <sstream>
17#include <string>
danilchapb8b6fbb2015-12-10 05:05:27 -080018#include <vector>
pwestin@webrtc.org26f8d9c2012-01-19 15:53:09 +000019
danilchap47a740b2015-12-15 00:30:07 -080020#include "webrtc/base/random.h"
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +000021#include "webrtc/base/scoped_ptr.h"
pbos@webrtc.org38344ed2014-09-24 06:05:00 +000022#include "webrtc/base/thread_annotations.h"
pbos@webrtc.orga048d7c2013-05-29 14:27:38 +000023#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
24#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
Henrik Kjellanderff761fb2015-11-04 08:31:52 +010025#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
26#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
Erik Språngbdc0b0d2015-06-22 15:21:24 +020027#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
pbos@webrtc.orga048d7c2013-05-29 14:27:38 +000028#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
29#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
30#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
pbos2d566682015-09-28 09:59:31 -070031#include "webrtc/transport.h"
pbos@webrtc.orga048d7c2013-05-29 14:27:38 +000032#include "webrtc/typedefs.h"
niklase@google.com470e71d2011-07-07 08:21:25 +000033
34namespace webrtc {
pwestin@webrtc.org741da942011-09-20 13:52:04 +000035
wu@webrtc.org822fbd82013-08-15 23:38:54 +000036class ModuleRtpRtcpImpl;
pbos@webrtc.org59f20bb2013-09-09 16:02:19 +000037class RTCPReceiver;
pwestin@webrtc.org741da942011-09-20 13:52:04 +000038
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +000039class NACKStringBuilder {
40 public:
41 NACKStringBuilder();
42 ~NACKStringBuilder();
pbos@webrtc.orgf3e4cee2013-07-31 15:17:19 +000043
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +000044 void PushNACK(uint16_t nack);
45 std::string GetResult();
edjee@google.com79b02892013-04-04 19:43:34 +000046
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +000047 private:
Erik Språng242e22b2015-05-11 10:17:43 +020048 std::ostringstream stream_;
49 int count_;
50 uint16_t prevNack_;
51 bool consecutive_;
edjee@google.com79b02892013-04-04 19:43:34 +000052};
53
asapersson@webrtc.org9ffd8fe2015-01-21 08:22:50 +000054class RTCPSender {
danilchap162abd32015-12-10 02:39:40 -080055 public:
56 struct FeedbackState {
57 FeedbackState();
pbos@webrtc.org59f20bb2013-09-09 16:02:19 +000058
danilchap162abd32015-12-10 02:39:40 -080059 uint8_t send_payload_type;
60 uint32_t frequency_hz;
61 uint32_t packets_sent;
62 size_t media_bytes_sent;
63 uint32_t send_bitrate;
pbos@webrtc.org59f20bb2013-09-09 16:02:19 +000064
danilchap162abd32015-12-10 02:39:40 -080065 uint32_t last_rr_ntp_secs;
66 uint32_t last_rr_ntp_frac;
67 uint32_t remote_sr;
pbos@webrtc.org59f20bb2013-09-09 16:02:19 +000068
danilchap162abd32015-12-10 02:39:40 -080069 bool has_last_xr_rr;
70 RtcpReceiveTimeInfo last_xr_rr;
asapersson@webrtc.org8469f7b2013-10-02 13:15:34 +000071
danilchap162abd32015-12-10 02:39:40 -080072 // Used when generating TMMBR.
73 ModuleRtpRtcpImpl* module;
74 };
Erik Språng61be2a42015-04-27 13:32:52 +020075
danilchap162abd32015-12-10 02:39:40 -080076 RTCPSender(bool audio,
77 Clock* clock,
78 ReceiveStatistics* receive_statistics,
79 RtcpPacketTypeCounterObserver* packet_type_counter_observer,
80 Transport* outgoing_transport);
81 virtual ~RTCPSender();
niklase@google.com470e71d2011-07-07 08:21:25 +000082
danilchap162abd32015-12-10 02:39:40 -080083 RtcpMode Status() const;
84 void SetRTCPStatus(RtcpMode method);
niklase@google.com470e71d2011-07-07 08:21:25 +000085
danilchap162abd32015-12-10 02:39:40 -080086 bool Sending() const;
87 int32_t SetSendingStatus(const FeedbackState& feedback_state,
88 bool enabled); // combine the functions
niklase@google.com470e71d2011-07-07 08:21:25 +000089
danilchap162abd32015-12-10 02:39:40 -080090 int32_t SetNackStatus(bool enable);
niklase@google.com470e71d2011-07-07 08:21:25 +000091
danilchap162abd32015-12-10 02:39:40 -080092 void SetStartTimestamp(uint32_t start_timestamp);
stefan@webrtc.org7c3523c2012-09-11 07:00:42 +000093
danilchap162abd32015-12-10 02:39:40 -080094 void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms);
stefan@webrtc.org7c3523c2012-09-11 07:00:42 +000095
danilchap162abd32015-12-10 02:39:40 -080096 void SetSSRC(uint32_t ssrc);
niklase@google.com470e71d2011-07-07 08:21:25 +000097
danilchap162abd32015-12-10 02:39:40 -080098 void SetRemoteSSRC(uint32_t ssrc);
niklase@google.com470e71d2011-07-07 08:21:25 +000099
danilchap162abd32015-12-10 02:39:40 -0800100 int32_t SetCNAME(const char* cName);
niklase@google.com470e71d2011-07-07 08:21:25 +0000101
danilchap162abd32015-12-10 02:39:40 -0800102 int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name);
niklase@google.com470e71d2011-07-07 08:21:25 +0000103
danilchap162abd32015-12-10 02:39:40 -0800104 int32_t RemoveMixedCNAME(uint32_t SSRC);
niklase@google.com470e71d2011-07-07 08:21:25 +0000105
danilchap162abd32015-12-10 02:39:40 -0800106 int64_t SendTimeOfSendReport(uint32_t sendReport);
niklase@google.com470e71d2011-07-07 08:21:25 +0000107
danilchap162abd32015-12-10 02:39:40 -0800108 bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const;
asapersson@webrtc.org8469f7b2013-10-02 13:15:34 +0000109
danilchap162abd32015-12-10 02:39:40 -0800110 bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000111
danilchap162abd32015-12-10 02:39:40 -0800112 int32_t SendRTCP(const FeedbackState& feedback_state,
113 RTCPPacketType packetType,
114 int32_t nackSize = 0,
115 const uint16_t* nackList = 0,
116 bool repeat = false,
117 uint64_t pictureID = 0);
niklase@google.com470e71d2011-07-07 08:21:25 +0000118
danilchap162abd32015-12-10 02:39:40 -0800119 int32_t SendCompoundRTCP(const FeedbackState& feedback_state,
120 const std::set<RTCPPacketType>& packetTypes,
121 int32_t nackSize = 0,
122 const uint16_t* nackList = 0,
123 bool repeat = false,
124 uint64_t pictureID = 0);
Erik Språng242e22b2015-05-11 10:17:43 +0200125
danilchap162abd32015-12-10 02:39:40 -0800126 bool REMB() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000127
danilchap162abd32015-12-10 02:39:40 -0800128 void SetREMBStatus(bool enable);
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000129
danilchap162abd32015-12-10 02:39:40 -0800130 void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs);
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000131
danilchap162abd32015-12-10 02:39:40 -0800132 bool TMMBR() const;
mflodman@webrtc.org84dc3d12011-12-22 10:26:13 +0000133
danilchap162abd32015-12-10 02:39:40 -0800134 void SetTMMBRStatus(bool enable);
niklase@google.com470e71d2011-07-07 08:21:25 +0000135
danilchap162abd32015-12-10 02:39:40 -0800136 int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit);
niklase@google.com470e71d2011-07-07 08:21:25 +0000137
danilchap162abd32015-12-10 02:39:40 -0800138 int32_t SetApplicationSpecificData(uint8_t subType,
139 uint32_t name,
140 const uint8_t* data,
141 uint16_t length);
142 int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
asapersson@webrtc.org5249cc82011-12-16 14:31:37 +0000143
danilchap162abd32015-12-10 02:39:40 -0800144 void SendRtcpXrReceiverReferenceTime(bool enable);
niklase@google.com470e71d2011-07-07 08:21:25 +0000145
danilchap162abd32015-12-10 02:39:40 -0800146 bool RtcpXrReceiverReferenceTime() const;
niklase@google.com470e71d2011-07-07 08:21:25 +0000147
danilchap162abd32015-12-10 02:39:40 -0800148 void SetCsrcs(const std::vector<uint32_t>& csrcs);
niklase@google.com470e71d2011-07-07 08:21:25 +0000149
danilchap162abd32015-12-10 02:39:40 -0800150 void SetTargetBitrate(unsigned int target_bitrate);
151 bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
mflodman@webrtc.org117c1192012-01-13 08:52:58 +0000152
danilchap162abd32015-12-10 02:39:40 -0800153 private:
154 class RtcpContext;
Erik Språng242e22b2015-05-11 10:17:43 +0200155
danilchap162abd32015-12-10 02:39:40 -0800156 // Determine which RTCP messages should be sent and setup flags.
157 void PrepareReport(const std::set<RTCPPacketType>& packetTypes,
158 const FeedbackState& feedback_state)
159 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000160
danilchap162abd32015-12-10 02:39:40 -0800161 int32_t AddReportBlock(const RTCPReportBlock& report_block)
162 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000163
danilchap162abd32015-12-10 02:39:40 -0800164 bool PrepareReportBlock(const FeedbackState& feedback_state,
165 uint32_t ssrc,
166 StreamStatistician* statistician,
167 RTCPReportBlock* report_block);
stefan@webrtc.org286fe0b2013-08-21 20:58:21 +0000168
danilchap162abd32015-12-10 02:39:40 -0800169 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context)
170 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
171 rtc::scoped_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context)
172 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
173 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context)
174 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
175 rtc::scoped_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context)
176 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
177 rtc::scoped_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context)
178 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
179 rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBR(const RtcpContext& context)
180 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
181 rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBN(const RtcpContext& context)
182 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
183 rtc::scoped_ptr<rtcp::RtcpPacket> BuildAPP(const RtcpContext& context)
184 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
185 rtc::scoped_ptr<rtcp::RtcpPacket> BuildVoIPMetric(const RtcpContext& context)
186 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
187 rtc::scoped_ptr<rtcp::RtcpPacket> BuildBYE(const RtcpContext& context)
188 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
189 rtc::scoped_ptr<rtcp::RtcpPacket> BuildFIR(const RtcpContext& context)
190 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
191 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSLI(const RtcpContext& context)
192 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
193 rtc::scoped_ptr<rtcp::RtcpPacket> BuildRPSI(const RtcpContext& context)
194 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
195 rtc::scoped_ptr<rtcp::RtcpPacket> BuildNACK(const RtcpContext& context)
196 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
197 rtc::scoped_ptr<rtcp::RtcpPacket> BuildReceiverReferenceTime(
198 const RtcpContext& context)
199 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
200 rtc::scoped_ptr<rtcp::RtcpPacket> BuildDlrr(const RtcpContext& context)
201 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
asapersson@webrtc.org8469f7b2013-10-02 13:15:34 +0000202
danilchap162abd32015-12-10 02:39:40 -0800203 private:
204 const bool audio_;
205 Clock* const clock_;
danilchap47a740b2015-12-15 00:30:07 -0800206 Random random_ GUARDED_BY(critical_section_rtcp_sender_);
danilchap162abd32015-12-10 02:39:40 -0800207 RtcpMode method_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000208
danilchap162abd32015-12-10 02:39:40 -0800209 Transport* const transport_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000210
danilchap162abd32015-12-10 02:39:40 -0800211 rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_;
212 bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_);
213 bool sending_ GUARDED_BY(critical_section_rtcp_sender_);
214 bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000215
danilchap162abd32015-12-10 02:39:40 -0800216 int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000217
danilchap162abd32015-12-10 02:39:40 -0800218 uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
219 uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_);
220 int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_);
221 uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
222 // SSRC that we receive on our RTP channel
223 uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_);
224 std::string cname_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000225
danilchap162abd32015-12-10 02:39:40 -0800226 ReceiveStatistics* receive_statistics_
227 GUARDED_BY(critical_section_rtcp_sender_);
228 std::map<uint32_t, rtcp::ReportBlock> report_blocks_
229 GUARDED_BY(critical_section_rtcp_sender_);
230 std::map<uint32_t, std::string> csrc_cnames_
231 GUARDED_BY(critical_section_rtcp_sender_);
stefan@webrtc.org286fe0b2013-08-21 20:58:21 +0000232
danilchap162abd32015-12-10 02:39:40 -0800233 // Sent
234 uint32_t last_send_report_[RTCP_NUMBER_OF_SR] GUARDED_BY(
235 critical_section_rtcp_sender_); // allow packet loss and RTT above 1 sec
236 int64_t last_rtcp_time_[RTCP_NUMBER_OF_SR] GUARDED_BY(
237 critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000238
danilchap162abd32015-12-10 02:39:40 -0800239 // Sent XR receiver reference time report.
240 // <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>.
241 std::map<uint32_t, int64_t> last_xr_rr_
242 GUARDED_BY(critical_section_rtcp_sender_);
asapersson@webrtc.org8469f7b2013-10-02 13:15:34 +0000243
danilchap162abd32015-12-10 02:39:40 -0800244 // send CSRCs
245 std::vector<uint32_t> csrcs_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000246
danilchap162abd32015-12-10 02:39:40 -0800247 // Full intra request
248 uint8_t sequence_number_fir_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000249
danilchap162abd32015-12-10 02:39:40 -0800250 // REMB
251 uint32_t remb_bitrate_ GUARDED_BY(critical_section_rtcp_sender_);
252 std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(critical_section_rtcp_sender_);
pwestin@webrtc.org741da942011-09-20 13:52:04 +0000253
danilchap162abd32015-12-10 02:39:40 -0800254 TMMBRHelp tmmbr_help_ GUARDED_BY(critical_section_rtcp_sender_);
255 uint32_t tmmbr_send_ GUARDED_BY(critical_section_rtcp_sender_);
256 uint32_t packet_oh_send_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000257
danilchap162abd32015-12-10 02:39:40 -0800258 // APP
259 uint8_t app_sub_type_ GUARDED_BY(critical_section_rtcp_sender_);
260 uint32_t app_name_ GUARDED_BY(critical_section_rtcp_sender_);
261 rtc::scoped_ptr<uint8_t[]> app_data_
262 GUARDED_BY(critical_section_rtcp_sender_);
263 uint16_t app_length_ GUARDED_BY(critical_section_rtcp_sender_);
niklase@google.com470e71d2011-07-07 08:21:25 +0000264
danilchap162abd32015-12-10 02:39:40 -0800265 // True if sending of XR Receiver reference time report is enabled.
266 bool xr_send_receiver_reference_time_enabled_
267 GUARDED_BY(critical_section_rtcp_sender_);
asapersson@webrtc.org8469f7b2013-10-02 13:15:34 +0000268
danilchap162abd32015-12-10 02:39:40 -0800269 // XR VoIP metric
270 RTCPVoIPMetric xr_voip_metric_ GUARDED_BY(critical_section_rtcp_sender_);
edjee@google.com79b02892013-04-04 19:43:34 +0000271
danilchap162abd32015-12-10 02:39:40 -0800272 RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
273 RtcpPacketTypeCounter packet_type_counter_
274 GUARDED_BY(critical_section_rtcp_sender_);
asapersson@webrtc.org2dd31342014-10-29 12:42:30 +0000275
danilchap162abd32015-12-10 02:39:40 -0800276 RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_);
Erik Språng242e22b2015-05-11 10:17:43 +0200277
danilchap162abd32015-12-10 02:39:40 -0800278 void SetFlag(RTCPPacketType type, bool is_volatile)
279 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
280 void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile)
281 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
282 bool IsFlagPresent(RTCPPacketType type) const
283 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
284 bool ConsumeFlag(RTCPPacketType type, bool forced = false)
285 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
286 bool AllVolatileFlagsConsumed() const
287 EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_);
288 struct ReportFlag {
289 ReportFlag(RTCPPacketType type, bool is_volatile)
290 : type(type), is_volatile(is_volatile) {}
291 bool operator<(const ReportFlag& flag) const { return type < flag.type; }
292 bool operator==(const ReportFlag& flag) const { return type == flag.type; }
293 const RTCPPacketType type;
294 const bool is_volatile;
295 };
Erik Språng242e22b2015-05-11 10:17:43 +0200296
danilchap162abd32015-12-10 02:39:40 -0800297 std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_);
Erik Språng242e22b2015-05-11 10:17:43 +0200298
danilchap162abd32015-12-10 02:39:40 -0800299 typedef rtc::scoped_ptr<rtcp::RtcpPacket> (RTCPSender::*BuilderFunc)(
300 const RtcpContext&);
301 std::map<RTCPPacketType, BuilderFunc> builders_;
niklase@google.com470e71d2011-07-07 08:21:25 +0000302};
pbos@webrtc.orgd900e8b2013-07-03 15:12:26 +0000303} // namespace webrtc
niklase@google.com470e71d2011-07-07 08:21:25 +0000304
danilchap162abd32015-12-10 02:39:40 -0800305#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_