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 ViENetwork; 48 class VoiceEngine; 49 50 // VoERTPObserver 51 class WEBRTC_DLLEXPORT VoERTPObserver 52 { 53 public: 54 virtual void OnIncomingCSRCChanged( 55 int channel, unsigned int CSRC, bool added) = 0; 56 57 virtual void OnIncomingSSRCChanged( 58 int channel, unsigned int SSRC) = 0; 59 60 protected: 61 virtual ~VoERTPObserver() {} 62 }; 63 64 // VoERTCPObserver 65 class WEBRTC_DLLEXPORT VoERTCPObserver 66 { 67 public: 68 virtual void OnApplicationDataReceived( 69 int channel, unsigned char subType, 70 unsigned int name, const unsigned char* data, 71 unsigned short dataLengthInBytes) = 0; 72 73 protected: 74 virtual ~VoERTCPObserver() {} 75 }; 76 77 // CallStatistics 78 struct CallStatistics 79 { 80 unsigned short fractionLost; 81 unsigned int cumulativeLost; 82 unsigned int extendedMax; 83 unsigned int jitterSamples; 84 int rttMs; 85 int bytesSent; 86 int packetsSent; 87 int bytesReceived; 88 int packetsReceived; 89 // The capture ntp time (in local timebase) of the first played out audio 90 // frame. 91 int64_t capture_start_ntp_time_ms_; 92 }; 93 94 // See section 6.4.1 in http://www.ietf.org/rfc/rfc3550.txt for details. 95 struct SenderInfo { 96 uint32_t NTP_timestamp_high; 97 uint32_t NTP_timestamp_low; 98 uint32_t RTP_timestamp; 99 uint32_t sender_packet_count; 100 uint32_t sender_octet_count; 101 }; 102 103 // See section 6.4.2 in http://www.ietf.org/rfc/rfc3550.txt for details. 104 struct ReportBlock { 105 uint32_t sender_SSRC; // SSRC of sender 106 uint32_t source_SSRC; 107 uint8_t fraction_lost; 108 uint32_t cumulative_num_packets_lost; 109 uint32_t extended_highest_sequence_number; 110 uint32_t interarrival_jitter; 111 uint32_t last_SR_timestamp; 112 uint32_t delay_since_last_SR; 113 }; 114 115 // VoERTP_RTCP 116 class WEBRTC_DLLEXPORT VoERTP_RTCP 117 { 118 public: 119 120 // Factory for the VoERTP_RTCP sub-API. Increases an internal 121 // reference counter if successful. Returns NULL if the API is not 122 // supported or if construction fails. 123 static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine); 124 125 // Releases the VoERTP_RTCP sub-API and decreases an internal 126 // reference counter. Returns the new reference count. This value should 127 // be zero for all sub-API:s before the VoiceEngine object can be safely 128 // deleted. 129 virtual int Release() = 0; 130 131 // Sets the local RTP synchronization source identifier (SSRC) explicitly. 132 virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0; 133 134 // Gets the local RTP SSRC of a specified |channel|. 135 virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0; 136 137 // Gets the SSRC of the incoming RTP packets. 138 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0; 139 140 // Sets the status of rtp-audio-level-indication on a specific |channel|. 141 virtual int SetSendAudioLevelIndicationStatus(int channel, 142 bool enable, 143 unsigned char id = 1) = 0; 144 145 // Sets the status of receiving rtp-audio-level-indication on a specific 146 // |channel|. 147 virtual int SetReceiveAudioLevelIndicationStatus(int channel, 148 bool enable, 149 unsigned char id = 1) { 150 // TODO(wu): Remove default implementation once talk is updated. 151 return 0; 152 } 153 154 // Sets the status of sending absolute sender time on a specific |channel|. 155 virtual int SetSendAbsoluteSenderTimeStatus(int channel, 156 bool enable, 157 unsigned char id) = 0; 158 159 // Sets status of receiving absolute sender time on a specific |channel|. 160 virtual int SetReceiveAbsoluteSenderTimeStatus(int channel, 161 bool enable, 162 unsigned char id) = 0; 163 164 // Sets the RTCP status on a specific |channel|. 165 virtual int SetRTCPStatus(int channel, bool enable) = 0; 166 167 // Gets the RTCP status on a specific |channel|. 168 virtual int GetRTCPStatus(int channel, bool& enabled) = 0; 169 170 // Sets the canonical name (CNAME) parameter for RTCP reports on a 171 // specific |channel|. 172 virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0; 173 174 // TODO(holmer): Remove this API once it has been removed from 175 // fakewebrtcvoiceengine.h. 176 virtual int GetRTCP_CNAME(int channel, char cName[256]) { 177 return -1; 178 } 179 180 // Gets the canonical name (CNAME) parameter for incoming RTCP reports 181 // on a specific channel. 182 virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0; 183 184 // Gets RTCP data from incoming RTCP Sender Reports. 185 virtual int GetRemoteRTCPData( 186 int channel, unsigned int& NTPHigh, unsigned int& NTPLow, 187 unsigned int& timestamp, unsigned int& playoutTimestamp, 188 unsigned int* jitter = NULL, unsigned short* fractionLost = NULL) = 0; 189 190 // Gets RTP statistics for a specific |channel|. 191 virtual int GetRTPStatistics( 192 int channel, unsigned int& averageJitterMs, unsigned int& maxJitterMs, 193 unsigned int& discardedPackets) = 0; 194 195 // Gets RTCP statistics for a specific |channel|. 196 virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0; 197 198 // Gets the report block parts of the last received RTCP Sender Report (SR), 199 // or RTCP Receiver Report (RR) on a specified |channel|. Each vector 200 // element also contains the SSRC of the sender in addition to a report 201 // block. 202 virtual int GetRemoteRTCPReportBlocks( 203 int channel, std::vector<ReportBlock>* receive_blocks) = 0; 204 205 // Sets the Redundant Coding (RED) status on a specific |channel|. 206 // TODO(minyue): Make SetREDStatus() pure virtual when fakewebrtcvoiceengine 207 // in talk is ready. 208 virtual int SetREDStatus( 209 int channel, bool enable, int redPayloadtype = -1) { return -1; } 210 211 // Gets the RED status on a specific |channel|. 212 // TODO(minyue): Make GetREDStatus() pure virtual when fakewebrtcvoiceengine 213 // in talk is ready. 214 virtual int GetREDStatus( 215 int channel, bool& enabled, int& redPayloadtype) { return -1; } 216 217 // Sets the Forward Error Correction (FEC) status on a specific |channel|. 218 // TODO(minyue): Remove SetFECStatus() when SetFECStatus() is replaced by 219 // SetREDStatus() in fakewebrtcvoiceengine. 220 virtual int SetFECStatus( 221 int channel, bool enable, int redPayloadtype = -1) { 222 return SetREDStatus(channel, enable, redPayloadtype); 223 }; 224 225 // Gets the FEC status on a specific |channel|. 226 // TODO(minyue): Remove GetFECStatus() when GetFECStatus() is replaced by 227 // GetREDStatus() in fakewebrtcvoiceengine. 228 virtual int GetFECStatus( 229 int channel, bool& enabled, int& redPayloadtype) { 230 return SetREDStatus(channel, enabled, redPayloadtype); 231 } 232 233 // This function enables Negative Acknowledgment (NACK) using RTCP, 234 // implemented based on RFC 4585. NACK retransmits RTP packets if lost on 235 // the network. This creates a lossless transport at the expense of delay. 236 // If using NACK, NACK should be enabled on both endpoints in a call. 237 virtual int SetNACKStatus(int channel, 238 bool enable, 239 int maxNoPackets) = 0; 240 241 // Enables capturing of RTP packets to a binary file on a specific 242 // |channel| and for a given |direction|. The file can later be replayed 243 // using e.g. RTP Tools rtpplay since the binary file format is 244 // compatible with the rtpdump format. 245 virtual int StartRTPDump( 246 int channel, const char fileNameUTF8[1024], 247 RTPDirections direction = kRtpIncoming) = 0; 248 249 // Disables capturing of RTP packets to a binary file on a specific 250 // |channel| and for a given |direction|. 251 virtual int StopRTPDump( 252 int channel, RTPDirections direction = kRtpIncoming) = 0; 253 254 // Gets the the current RTP capturing state for the specified 255 // |channel| and |direction|. 256 virtual int RTPDumpIsActive( 257 int channel, RTPDirections direction = kRtpIncoming) = 0; 258 259 // Sets video engine channel to receive incoming audio packets for 260 // aggregated bandwidth estimation. Takes ownership of the ViENetwork 261 // interface. 262 virtual int SetVideoEngineBWETarget(int channel, ViENetwork* vie_network, 263 int video_channel) { 264 return 0; 265 } 266 267 // Will be removed. Don't use. 268 virtual int RegisterRTPObserver(int channel, 269 VoERTPObserver& observer) { return -1; }; 270 virtual int DeRegisterRTPObserver(int channel) { return -1; }; 271 virtual int RegisterRTCPObserver( 272 int channel, VoERTCPObserver& observer) { return -1; }; 273 virtual int DeRegisterRTCPObserver(int channel) { return -1; }; 274 virtual int GetRemoteCSRCs(int channel, 275 unsigned int arrCSRC[15]) { return -1; }; 276 virtual int InsertExtraRTPPacket( 277 int channel, unsigned char payloadType, bool markerBit, 278 const char* payloadData, unsigned short payloadSize) { return -1; }; 279 virtual int GetRemoteRTCPSenderInfo( 280 int channel, SenderInfo* sender_info) { return -1; }; 281 virtual int SendApplicationDefinedRTCPPacket( 282 int channel, unsigned char subType, unsigned int name, 283 const char* data, unsigned short dataLengthInBytes) { return -1; }; 284 virtual int GetLastRemoteTimeStamp(int channel, 285 uint32_t* lastRemoteTimeStamp) { return -1; }; 286 287 protected: 288 VoERTP_RTCP() {} 289 virtual ~VoERTP_RTCP() {} 290 }; 291 292 } // namespace webrtc 293 294 #endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H 295
