1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 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 // This sub-API supports the following functionalities: 12 // 13 // - Callbacks for RTP and RTCP events such as modified SSRC or CSRC. 14 // - SSRC handling. 15 // - Transmission of RTCP sender reports. 16 // - Obtaining RTCP data from incoming RTCP sender reports. 17 // - RTP and RTCP statistics (jitter, packet loss, RTT etc.). 18 // - Redundant Coding (RED) 19 // - Writing RTP and RTCP packets to binary files for off-line analysis of 20 // the call quality. 21 // 22 // Usage example, omitting error checking: 23 // 24 // using namespace webrtc; 25 // VoiceEngine* voe = VoiceEngine::Create(); 26 // VoEBase* base = VoEBase::GetInterface(voe); 27 // VoERTP_RTCP* rtp_rtcp = VoERTP_RTCP::GetInterface(voe); 28 // base->Init(); 29 // int ch = base->CreateChannel(); 30 // ... 31 // rtp_rtcp->SetLocalSSRC(ch, 12345); 32 // ... 33 // base->DeleteChannel(ch); 34 // base->Terminate(); 35 // base->Release(); 36 // rtp_rtcp->Release(); 37 // VoiceEngine::Delete(voe); 38 // 39 #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H 40 #define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H 41 42 #include <vector> 43 #include "webrtc/common_types.h" 44 45 namespace webrtc { 46 47 class VoiceEngine; 48 49 // VoERTPObserver 50 class WEBRTC_DLLEXPORT VoERTPObserver { 51 public: 52 virtual void OnIncomingCSRCChanged(int channel, 53 unsigned int CSRC, 54 bool added) = 0; 55 56 virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0; 57 58 protected: ~VoERTPObserver()59 virtual ~VoERTPObserver() {} 60 }; 61 62 // CallStatistics 63 struct CallStatistics { 64 unsigned short fractionLost; 65 unsigned int cumulativeLost; 66 unsigned int extendedMax; 67 unsigned int jitterSamples; 68 int64_t rttMs; 69 size_t bytesSent; 70 int packetsSent; 71 size_t bytesReceived; 72 int packetsReceived; 73 // The capture ntp time (in local timebase) of the first played out audio 74 // frame. 75 int64_t capture_start_ntp_time_ms_; 76 }; 77 78 // See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details. 79 struct SenderInfo { 80 uint32_t NTP_timestamp_high; 81 uint32_t NTP_timestamp_low; 82 uint32_t RTP_timestamp; 83 uint32_t sender_packet_count; 84 uint32_t sender_octet_count; 85 }; 86 87 // See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details. 88 struct ReportBlock { 89 uint32_t sender_SSRC; // SSRC of sender 90 uint32_t source_SSRC; 91 uint8_t fraction_lost; 92 uint32_t cumulative_num_packets_lost; 93 uint32_t extended_highest_sequence_number; 94 uint32_t interarrival_jitter; 95 uint32_t last_SR_timestamp; 96 uint32_t delay_since_last_SR; 97 }; 98 99 // VoERTP_RTCP 100 class WEBRTC_DLLEXPORT VoERTP_RTCP { 101 public: 102 // Factory for the VoERTP_RTCP sub-API. Increases an internal 103 // reference counter if successful. Returns NULL if the API is not 104 // supported or if construction fails. 105 static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine); 106 107 // Releases the VoERTP_RTCP sub-API and decreases an internal 108 // reference counter. Returns the new reference count. This value should 109 // be zero for all sub-API:s before the VoiceEngine object can be safely 110 // deleted. 111 virtual int Release() = 0; 112 113 // Sets the local RTP synchronization source identifier (SSRC) explicitly. 114 virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0; 115 116 // Gets the local RTP SSRC of a specified |channel|. 117 virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0; 118 119 // Gets the SSRC of the incoming RTP packets. 120 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0; 121 122 // Sets the status of rtp-audio-level-indication on a specific |channel|. 123 virtual int SetSendAudioLevelIndicationStatus(int channel, 124 bool enable, 125 unsigned char id = 1) = 0; 126 127 // Sets the status of receiving rtp-audio-level-indication on a specific 128 // |channel|. 129 virtual int SetReceiveAudioLevelIndicationStatus(int channel, 130 bool enable, 131 unsigned char id = 1) { 132 // TODO(wu): Remove default implementation once talk is updated. 133 return 0; 134 } 135 136 // Sets the status of sending absolute sender time on a specific |channel|. 137 virtual int SetSendAbsoluteSenderTimeStatus(int channel, 138 bool enable, 139 unsigned char id) = 0; 140 141 // Sets status of receiving absolute sender time on a specific |channel|. 142 virtual int SetReceiveAbsoluteSenderTimeStatus(int channel, 143 bool enable, 144 unsigned char id) = 0; 145 146 // Sets the RTCP status on a specific |channel|. 147 virtual int SetRTCPStatus(int channel, bool enable) = 0; 148 149 // Gets the RTCP status on a specific |channel|. 150 virtual int GetRTCPStatus(int channel, bool& enabled) = 0; 151 152 // Sets the canonical name (CNAME) parameter for RTCP reports on a 153 // specific |channel|. 154 virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0; 155 156 // TODO(holmer): Remove this API once it has been removed from 157 // fakewebrtcvoiceengine.h. GetRTCP_CNAME(int channel,char cName[256])158 virtual int GetRTCP_CNAME(int channel, char cName[256]) { return -1; } 159 160 // Gets the canonical name (CNAME) parameter for incoming RTCP reports 161 // on a specific channel. 162 virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0; 163 164 // Gets RTCP data from incoming RTCP Sender Reports. 165 virtual int GetRemoteRTCPData(int channel, 166 unsigned int& NTPHigh, 167 unsigned int& NTPLow, 168 unsigned int& timestamp, 169 unsigned int& playoutTimestamp, 170 unsigned int* jitter = NULL, 171 unsigned short* fractionLost = NULL) = 0; 172 173 // Gets RTP statistics for a specific |channel|. 174 virtual int GetRTPStatistics(int channel, 175 unsigned int& averageJitterMs, 176 unsigned int& maxJitterMs, 177 unsigned int& discardedPackets) = 0; 178 179 // Gets RTCP statistics for a specific |channel|. 180 virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0; 181 182 // Gets the report block parts of the last received RTCP Sender Report (SR), 183 // or RTCP Receiver Report (RR) on a specified |channel|. Each vector 184 // element also contains the SSRC of the sender in addition to a report 185 // block. 186 virtual int GetRemoteRTCPReportBlocks( 187 int channel, 188 std::vector<ReportBlock>* receive_blocks) = 0; 189 190 // Sets the Redundant Coding (RED) status on a specific |channel|. 191 // TODO(minyue): Make SetREDStatus() pure virtual when fakewebrtcvoiceengine 192 // in talk is ready. 193 virtual int SetREDStatus(int channel, bool enable, int redPayloadtype = -1) { 194 return -1; 195 } 196 197 // Gets the RED status on a specific |channel|. 198 // TODO(minyue): Make GetREDStatus() pure virtual when fakewebrtcvoiceengine 199 // in talk is ready. GetREDStatus(int channel,bool & enabled,int & redPayloadtype)200 virtual int GetREDStatus(int channel, bool& enabled, int& redPayloadtype) { 201 return -1; 202 } 203 204 // This function enables Negative Acknowledgment (NACK) using RTCP, 205 // implemented based on RFC 4585. NACK retransmits RTP packets if lost on 206 // the network. This creates a lossless transport at the expense of delay. 207 // If using NACK, NACK should be enabled on both endpoints in a call. 208 virtual int SetNACKStatus(int channel, bool enable, int maxNoPackets) = 0; 209 210 protected: VoERTP_RTCP()211 VoERTP_RTCP() {} ~VoERTP_RTCP()212 virtual ~VoERTP_RTCP() {} 213 }; 214 215 } // namespace webrtc 216 217 #endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H 218