1 /* 2 * Copyright (c) 2020 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 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_ 12 #define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_ 13 14 #include <memory> 15 #include <string> 16 #include <vector> 17 18 #include "absl/strings/string_view.h" 19 #include "absl/types/optional.h" 20 #include "api/field_trials_view.h" 21 #include "api/frame_transformer_interface.h" 22 #include "api/scoped_refptr.h" 23 #include "api/video/video_bitrate_allocation.h" 24 #include "modules/rtp_rtcp/include/receive_statistics.h" 25 #include "modules/rtp_rtcp/include/report_block_data.h" 26 #include "modules/rtp_rtcp/include/rtp_packet_sender.h" 27 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 28 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" 29 #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" 30 #include "modules/rtp_rtcp/source/video_fec_generator.h" 31 #include "system_wrappers/include/ntp_time.h" 32 33 namespace webrtc { 34 35 // Forward declarations. 36 class FrameEncryptorInterface; 37 class RateLimiter; 38 class RtcEventLog; 39 class RTPSender; 40 class Transport; 41 class VideoBitrateAllocationObserver; 42 43 class RtpRtcpInterface : public RtcpFeedbackSenderInterface { 44 public: 45 struct Configuration { 46 Configuration() = default; 47 Configuration(Configuration&& rhs) = default; 48 49 Configuration(const Configuration&) = delete; 50 Configuration& operator=(const Configuration&) = delete; 51 52 // True for a audio version of the RTP/RTCP module object false will create 53 // a video version. 54 bool audio = false; 55 bool receiver_only = false; 56 57 // The clock to use to read time. If nullptr then system clock will be used. 58 Clock* clock = nullptr; 59 60 ReceiveStatisticsProvider* receive_statistics = nullptr; 61 62 // Transport object that will be called when packets are ready to be sent 63 // out on the network. 64 Transport* outgoing_transport = nullptr; 65 66 // Called when the receiver requests an intra frame. 67 RtcpIntraFrameObserver* intra_frame_callback = nullptr; 68 69 // Called when the receiver sends a loss notification. 70 RtcpLossNotificationObserver* rtcp_loss_notification_observer = nullptr; 71 72 // Called when we receive a changed estimate from the receiver of out 73 // stream. 74 RtcpBandwidthObserver* bandwidth_callback = nullptr; 75 76 NetworkStateEstimateObserver* network_state_estimate_observer = nullptr; 77 TransportFeedbackObserver* transport_feedback_callback = nullptr; 78 VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr; 79 RtcpRttStats* rtt_stats = nullptr; 80 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr; 81 // Called on receipt of RTCP report block from remote side. 82 // TODO(bugs.webrtc.org/10679): Consider whether we want to use 83 // only getters or only callbacks. If we decide on getters, the 84 // ReportBlockDataObserver should also be removed in favor of 85 // GetLatestReportBlockData(). 86 RtcpCnameCallback* rtcp_cname_callback = nullptr; 87 ReportBlockDataObserver* report_block_data_observer = nullptr; 88 89 // Spread any bursts of packets into smaller bursts to minimize packet loss. 90 RtpPacketSender* paced_sender = nullptr; 91 92 // Generates FEC packets. 93 // TODO(sprang): Wire up to RtpSenderEgress. 94 VideoFecGenerator* fec_generator = nullptr; 95 96 BitrateStatisticsObserver* send_bitrate_observer = nullptr; 97 SendSideDelayObserver* send_side_delay_observer = nullptr; 98 RtcEventLog* event_log = nullptr; 99 SendPacketObserver* send_packet_observer = nullptr; 100 RateLimiter* retransmission_rate_limiter = nullptr; 101 StreamDataCountersCallback* rtp_stats_callback = nullptr; 102 103 int rtcp_report_interval_ms = 0; 104 105 // Update network2 instead of pacer_exit field of video timing extension. 106 bool populate_network2_timestamp = false; 107 108 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer; 109 110 // E2EE Custom Video Frame Encryption 111 FrameEncryptorInterface* frame_encryptor = nullptr; 112 // Require all outgoing frames to be encrypted with a FrameEncryptor. 113 bool require_frame_encryption = false; 114 115 // Corresponds to extmap-allow-mixed in SDP negotiation. 116 bool extmap_allow_mixed = false; 117 118 // If true, the RTP sender will always annotate outgoing packets with 119 // MID and RID header extensions, if provided and negotiated. 120 // If false, the RTP sender will stop sending MID and RID header extensions, 121 // when it knows that the receiver is ready to demux based on SSRC. This is 122 // done by RTCP RR acking. 123 bool always_send_mid_and_rid = false; 124 125 // If set, field trials are read from `field_trials`, otherwise 126 // defaults to webrtc::FieldTrialBasedConfig. 127 const FieldTrialsView* field_trials = nullptr; 128 129 // SSRCs for media and retransmission, respectively. 130 // FlexFec SSRC is fetched from `flexfec_sender`. 131 uint32_t local_media_ssrc = 0; 132 absl::optional<uint32_t> rtx_send_ssrc; 133 134 bool need_rtp_packet_infos = false; 135 136 // If true, the RTP packet history will select RTX packets based on 137 // heuristics such as send time, retransmission count etc, in order to 138 // make padding potentially more useful. 139 // If false, the last packet will always be picked. This may reduce CPU 140 // overhead. 141 bool enable_rtx_padding_prioritization = true; 142 143 // Estimate RTT as non-sender as described in 144 // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5 145 bool non_sender_rtt_measurement = false; 146 147 // If non-empty, sets the value for sending in the RID (and Repaired) RTP 148 // header extension. RIDs are used to identify an RTP stream if SSRCs are 149 // not negotiated. If the RID and Repaired RID extensions are not 150 // registered, the RID will not be sent. 151 std::string rid; 152 }; 153 154 // Stats for RTCP sender reports (SR) for a specific SSRC. 155 // Refer to https://tools.ietf.org/html/rfc3550#section-6.4.1. 156 struct SenderReportStats { 157 // Arrival NTP timestamp for the last received RTCP SR. 158 NtpTime last_arrival_timestamp; 159 // Received (a.k.a., remote) NTP timestamp for the last received RTCP SR. 160 NtpTime last_remote_timestamp; 161 // Total number of RTP data packets transmitted by the sender since starting 162 // transmission up until the time this SR packet was generated. The count 163 // should be reset if the sender changes its SSRC identifier. 164 uint32_t packets_sent; 165 // Total number of payload octets (i.e., not including header or padding) 166 // transmitted in RTP data packets by the sender since starting transmission 167 // up until the time this SR packet was generated. The count should be reset 168 // if the sender changes its SSRC identifier. 169 uint64_t bytes_sent; 170 // Total number of RTCP SR blocks received. 171 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent. 172 uint64_t reports_count; 173 }; 174 // Stats about the non-sender SSRC, based on RTCP extended reports (XR). 175 // Refer to https://datatracker.ietf.org/doc/html/rfc3611#section-2. 176 struct NonSenderRttStats { 177 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime 178 absl::optional<TimeDelta> round_trip_time; 179 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime 180 TimeDelta total_round_trip_time = TimeDelta::Zero(); 181 // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements 182 int round_trip_time_measurements = 0; 183 }; 184 185 // ************************************************************************** 186 // Receiver functions 187 // ************************************************************************** 188 189 virtual void IncomingRtcpPacket(const uint8_t* incoming_packet, 190 size_t incoming_packet_length) = 0; 191 192 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; 193 194 // Called when the local ssrc changes (post initialization) for receive 195 // streams to match with send. Called on the packet receive thread/tq. 196 virtual void SetLocalSsrc(uint32_t ssrc) = 0; 197 198 // ************************************************************************** 199 // Sender 200 // ************************************************************************** 201 202 // Sets the maximum size of an RTP packet, including RTP headers. 203 virtual void SetMaxRtpPacketSize(size_t size) = 0; 204 205 // Returns max RTP packet size. Takes into account RTP headers and 206 // FEC/ULP/RED overhead (when FEC is enabled). 207 virtual size_t MaxRtpPacketSize() const = 0; 208 209 virtual void RegisterSendPayloadFrequency(int payload_type, 210 int payload_frequency) = 0; 211 212 // Unregisters a send payload. 213 // `payload_type` - payload type of codec 214 // Returns -1 on failure else 0. 215 virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0; 216 217 virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0; 218 219 // Register extension by uri, triggers CHECK on falure. 220 virtual void RegisterRtpHeaderExtension(absl::string_view uri, int id) = 0; 221 222 virtual void DeregisterSendRtpHeaderExtension(absl::string_view uri) = 0; 223 224 // Returns true if RTP module is send media, and any of the extensions 225 // required for bandwidth estimation is registered. 226 virtual bool SupportsPadding() const = 0; 227 // Same as SupportsPadding(), but additionally requires that 228 // SetRtxSendStatus() has been called with the kRtxRedundantPayloads option 229 // enabled. 230 virtual bool SupportsRtxPayloadPadding() const = 0; 231 232 // Returns start timestamp. 233 virtual uint32_t StartTimestamp() const = 0; 234 235 // Sets start timestamp. Start timestamp is set to a random value if this 236 // function is never called. 237 virtual void SetStartTimestamp(uint32_t timestamp) = 0; 238 239 // Returns SequenceNumber. 240 virtual uint16_t SequenceNumber() const = 0; 241 242 // Sets SequenceNumber, default is a random number. 243 virtual void SetSequenceNumber(uint16_t seq) = 0; 244 245 virtual void SetRtpState(const RtpState& rtp_state) = 0; 246 virtual void SetRtxState(const RtpState& rtp_state) = 0; 247 virtual RtpState GetRtpState() const = 0; 248 virtual RtpState GetRtxState() const = 0; 249 250 // This can be used to enable/disable receive-side RTT. 251 virtual void SetNonSenderRttMeasurement(bool enabled) = 0; 252 253 // Returns SSRC. 254 virtual uint32_t SSRC() const = 0; 255 256 // Sets the value for sending in the MID RTP header extension. 257 // The MID RTP header extension should be registered for this to do anything. 258 // Once set, this value can not be changed or removed. 259 virtual void SetMid(absl::string_view mid) = 0; 260 261 // Sets CSRC. 262 // `csrcs` - vector of CSRCs 263 virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0; 264 265 // Turns on/off sending RTX (RFC 4588). The modes can be set as a combination 266 // of values of the enumerator RtxMode. 267 virtual void SetRtxSendStatus(int modes) = 0; 268 269 // Returns status of sending RTX (RFC 4588). The returned value can be 270 // a combination of values of the enumerator RtxMode. 271 virtual int RtxSendStatus() const = 0; 272 273 // Returns the SSRC used for RTX if set, otherwise a nullopt. 274 virtual absl::optional<uint32_t> RtxSsrc() const = 0; 275 276 // Sets the payload type to use when sending RTX packets. Note that this 277 // doesn't enable RTX, only the payload type is set. 278 virtual void SetRtxSendPayloadType(int payload_type, 279 int associated_payload_type) = 0; 280 281 // Returns the FlexFEC SSRC, if there is one. 282 virtual absl::optional<uint32_t> FlexfecSsrc() const = 0; 283 284 // Sets sending status. Sends kRtcpByeCode when going from true to false. 285 // Returns -1 on failure else 0. 286 virtual int32_t SetSendingStatus(bool sending) = 0; 287 288 // Returns current sending status. 289 virtual bool Sending() const = 0; 290 291 // Starts/Stops media packets. On by default. 292 virtual void SetSendingMediaStatus(bool sending) = 0; 293 294 // Returns current media sending status. 295 virtual bool SendingMedia() const = 0; 296 297 // Returns whether audio is configured (i.e. Configuration::audio = true). 298 virtual bool IsAudioConfigured() const = 0; 299 300 // Indicate that the packets sent by this module should be counted towards the 301 // bitrate estimate since the stream participates in the bitrate allocation. 302 virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0; 303 304 // Returns bitrate sent (post-pacing) per packet type. 305 virtual RtpSendRates GetSendRates() const = 0; 306 307 virtual RTPSender* RtpSender() = 0; 308 virtual const RTPSender* RtpSender() const = 0; 309 310 // Record that a frame is about to be sent. Returns true on success, and false 311 // if the module isn't ready to send. 312 virtual bool OnSendingRtpFrame(uint32_t timestamp, 313 int64_t capture_time_ms, 314 int payload_type, 315 bool force_sender_report) = 0; 316 317 // Try to send the provided packet. Returns true iff packet matches any of 318 // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the 319 // transport. 320 virtual bool TrySendPacket(RtpPacketToSend* packet, 321 const PacedPacketInfo& pacing_info) = 0; 322 323 // Update the FEC protection parameters to use for delta- and key-frames. 324 // Only used when deferred FEC is active. 325 virtual void SetFecProtectionParams( 326 const FecProtectionParams& delta_params, 327 const FecProtectionParams& key_params) = 0; 328 329 // If deferred FEC generation is enabled, this method should be called after 330 // calling TrySendPacket(). Any generated FEC packets will be removed and 331 // returned from the FEC generator. 332 virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0; 333 334 virtual void OnAbortedRetransmissions( 335 rtc::ArrayView<const uint16_t> sequence_numbers) = 0; 336 337 virtual void OnPacketsAcknowledged( 338 rtc::ArrayView<const uint16_t> sequence_numbers) = 0; 339 340 virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( 341 size_t target_size_bytes) = 0; 342 343 virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos( 344 rtc::ArrayView<const uint16_t> sequence_numbers) const = 0; 345 346 // Returns an expected per packet overhead representing the main RTP header, 347 // any CSRCs, and the registered header extensions that are expected on all 348 // packets (i.e. disregarding things like abs capture time which is only 349 // populated on a subset of packets, but counting MID/RID type extensions 350 // when we expect to send them). 351 virtual size_t ExpectedPerPacketOverhead() const = 0; 352 353 // Access to packet state (e.g. sequence numbering) must only be access by 354 // one thread at a time. It may be only one thread, or a construction thread 355 // that calls SetRtpState() - handing over to a pacer thread that calls 356 // TrySendPacket() - and at teardown ownership is handed to a destruciton 357 // thread that calls GetRtpState(). 358 // This method is used to signal that "ownership" of the rtp state is being 359 // transferred to another thread. 360 virtual void OnPacketSendingThreadSwitched() = 0; 361 362 // ************************************************************************** 363 // RTCP 364 // ************************************************************************** 365 366 // Returns RTCP status. 367 virtual RtcpMode RTCP() const = 0; 368 369 // Sets RTCP status i.e on(compound or non-compound)/off. 370 // `method` - RTCP method to use. 371 virtual void SetRTCPStatus(RtcpMode method) = 0; 372 373 // Sets RTCP CName (i.e unique identifier). 374 // Returns -1 on failure else 0. 375 virtual int32_t SetCNAME(absl::string_view cname) = 0; 376 377 // Returns remote NTP. 378 // Returns -1 on failure else 0. 379 virtual int32_t RemoteNTP(uint32_t* received_ntp_secs, 380 uint32_t* received_ntp_frac, 381 uint32_t* rtcp_arrival_time_secs, 382 uint32_t* rtcp_arrival_time_frac, 383 uint32_t* rtcp_timestamp) const = 0; 384 385 // Returns current RTT (round-trip time) estimate. 386 // Returns -1 on failure else 0. 387 virtual int32_t RTT(uint32_t remote_ssrc, 388 int64_t* rtt, 389 int64_t* avg_rtt, 390 int64_t* min_rtt, 391 int64_t* max_rtt) const = 0; 392 393 // Returns the estimated RTT, with fallback to a default value. 394 virtual int64_t ExpectedRetransmissionTimeMs() const = 0; 395 396 // Forces a send of a RTCP packet. Periodic SR and RR are triggered via the 397 // process function. 398 // Returns -1 on failure else 0. 399 virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0; 400 401 // Returns send statistics for the RTP and RTX stream. 402 virtual void GetSendStreamDataCounters( 403 StreamDataCounters* rtp_counters, 404 StreamDataCounters* rtx_counters) const = 0; 405 406 // A snapshot of Report Blocks with additional data of interest to statistics. 407 // Within this list, the sender-source SSRC pair is unique and per-pair the 408 // ReportBlockData represents the latest Report Block that was received for 409 // that pair. 410 virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0; 411 // Returns stats based on the received RTCP SRs. 412 virtual absl::optional<SenderReportStats> GetSenderReportStats() const = 0; 413 // Returns non-sender RTT stats, based on DLRR. 414 virtual absl::optional<NonSenderRttStats> GetNonSenderRttStats() const = 0; 415 416 // (REMB) Receiver Estimated Max Bitrate. 417 // Schedules sending REMB on next and following sender/receiver reports. 418 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override = 0; 419 // Stops sending REMB on next and following sender/receiver reports. 420 void UnsetRemb() override = 0; 421 422 // (NACK) 423 424 // Sends a Negative acknowledgement packet. 425 // Returns -1 on failure else 0. 426 // TODO(philipel): Deprecate this and start using SendNack instead, mostly 427 // because we want a function that actually send NACK for the specified 428 // packets. 429 virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0; 430 431 // Sends NACK for the packets specified. 432 // Note: This assumes the caller keeps track of timing and doesn't rely on 433 // the RTP module to do this. 434 virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0; 435 436 // Store the sent packets, needed to answer to a Negative acknowledgment 437 // requests. 438 virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0; 439 440 virtual void SetVideoBitrateAllocation( 441 const VideoBitrateAllocation& bitrate) = 0; 442 443 // ************************************************************************** 444 // Video 445 // ************************************************************************** 446 447 // Requests new key frame. 448 // using PLI, https://tools.ietf.org/html/rfc4585#section-6.3.1.1 SendPictureLossIndication()449 void SendPictureLossIndication() { SendRTCP(kRtcpPli); } 450 // using FIR, https://tools.ietf.org/html/rfc5104#section-4.3.1.2 SendFullIntraRequest()451 void SendFullIntraRequest() { SendRTCP(kRtcpFir); } 452 453 // Sends a LossNotification RTCP message. 454 // Returns -1 on failure else 0. 455 virtual int32_t SendLossNotification(uint16_t last_decoded_seq_num, 456 uint16_t last_received_seq_num, 457 bool decodability_flag, 458 bool buffering_allowed) = 0; 459 }; 460 461 } // namespace webrtc 462 463 #endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_ 464