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/types/optional.h" 19 #include "api/frame_transformer_interface.h" 20 #include "api/scoped_refptr.h" 21 #include "api/transport/webrtc_key_value_config.h" 22 #include "api/video/video_bitrate_allocation.h" 23 #include "modules/rtp_rtcp/include/receive_statistics.h" 24 #include "modules/rtp_rtcp/include/report_block_data.h" 25 #include "modules/rtp_rtcp/include/rtp_packet_sender.h" 26 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" 27 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" 28 #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" 29 #include "modules/rtp_rtcp/source/video_fec_generator.h" 30 #include "rtc_base/constructor_magic.h" 31 32 namespace webrtc { 33 34 // Forward declarations. 35 class FrameEncryptorInterface; 36 class RateLimiter; 37 class RemoteBitrateEstimator; 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 // True for a audio version of the RTP/RTCP module object false will create 50 // a video version. 51 bool audio = false; 52 bool receiver_only = false; 53 54 // The clock to use to read time. If nullptr then system clock will be used. 55 Clock* clock = nullptr; 56 57 ReceiveStatisticsProvider* receive_statistics = nullptr; 58 59 // Transport object that will be called when packets are ready to be sent 60 // out on the network. 61 Transport* outgoing_transport = nullptr; 62 63 // Called when the receiver requests an intra frame. 64 RtcpIntraFrameObserver* intra_frame_callback = nullptr; 65 66 // Called when the receiver sends a loss notification. 67 RtcpLossNotificationObserver* rtcp_loss_notification_observer = nullptr; 68 69 // Called when we receive a changed estimate from the receiver of out 70 // stream. 71 RtcpBandwidthObserver* bandwidth_callback = nullptr; 72 73 NetworkStateEstimateObserver* network_state_estimate_observer = nullptr; 74 TransportFeedbackObserver* transport_feedback_callback = nullptr; 75 VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr; 76 RtcpRttStats* rtt_stats = nullptr; 77 RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr; 78 // Called on receipt of RTCP report block from remote side. 79 // TODO(bugs.webrtc.org/10678): Remove RtcpStatisticsCallback in 80 // favor of ReportBlockDataObserver. 81 // TODO(bugs.webrtc.org/10679): Consider whether we want to use 82 // only getters or only callbacks. If we decide on getters, the 83 // ReportBlockDataObserver should also be removed in favor of 84 // GetLatestReportBlockData(). 85 RtcpStatisticsCallback* rtcp_statistics_callback = nullptr; 86 RtcpCnameCallback* rtcp_cname_callback = nullptr; 87 ReportBlockDataObserver* report_block_data_observer = nullptr; 88 89 // Estimates the bandwidth available for a set of streams from the same 90 // client. 91 RemoteBitrateEstimator* remote_bitrate_estimator = nullptr; 92 93 // Spread any bursts of packets into smaller bursts to minimize packet loss. 94 RtpPacketSender* paced_sender = nullptr; 95 96 // Generates FEC packets. 97 // TODO(sprang): Wire up to RtpSenderEgress. 98 VideoFecGenerator* fec_generator = nullptr; 99 100 BitrateStatisticsObserver* send_bitrate_observer = nullptr; 101 SendSideDelayObserver* send_side_delay_observer = nullptr; 102 RtcEventLog* event_log = nullptr; 103 SendPacketObserver* send_packet_observer = nullptr; 104 RateLimiter* retransmission_rate_limiter = nullptr; 105 StreamDataCountersCallback* rtp_stats_callback = nullptr; 106 107 int rtcp_report_interval_ms = 0; 108 109 // Update network2 instead of pacer_exit field of video timing extension. 110 bool populate_network2_timestamp = false; 111 112 rtc::scoped_refptr<FrameTransformerInterface> frame_transformer; 113 114 // E2EE Custom Video Frame Encryption 115 FrameEncryptorInterface* frame_encryptor = nullptr; 116 // Require all outgoing frames to be encrypted with a FrameEncryptor. 117 bool require_frame_encryption = false; 118 119 // Corresponds to extmap-allow-mixed in SDP negotiation. 120 bool extmap_allow_mixed = false; 121 122 // If true, the RTP sender will always annotate outgoing packets with 123 // MID and RID header extensions, if provided and negotiated. 124 // If false, the RTP sender will stop sending MID and RID header extensions, 125 // when it knows that the receiver is ready to demux based on SSRC. This is 126 // done by RTCP RR acking. 127 bool always_send_mid_and_rid = false; 128 129 // If set, field trials are read from |field_trials|, otherwise 130 // defaults to webrtc::FieldTrialBasedConfig. 131 const WebRtcKeyValueConfig* field_trials = nullptr; 132 133 // SSRCs for media and retransmission, respectively. 134 // FlexFec SSRC is fetched from |flexfec_sender|. 135 uint32_t local_media_ssrc = 0; 136 absl::optional<uint32_t> rtx_send_ssrc; 137 138 bool need_rtp_packet_infos = false; 139 140 // If true, the RTP packet history will select RTX packets based on 141 // heuristics such as send time, retransmission count etc, in order to 142 // make padding potentially more useful. 143 // If false, the last packet will always be picked. This may reduce CPU 144 // overhead. 145 bool enable_rtx_padding_prioritization = true; 146 147 private: 148 RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); 149 }; 150 151 // ************************************************************************** 152 // Receiver functions 153 // ************************************************************************** 154 155 virtual void IncomingRtcpPacket(const uint8_t* incoming_packet, 156 size_t incoming_packet_length) = 0; 157 158 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; 159 160 // ************************************************************************** 161 // Sender 162 // ************************************************************************** 163 164 // Sets the maximum size of an RTP packet, including RTP headers. 165 virtual void SetMaxRtpPacketSize(size_t size) = 0; 166 167 // Returns max RTP packet size. Takes into account RTP headers and 168 // FEC/ULP/RED overhead (when FEC is enabled). 169 virtual size_t MaxRtpPacketSize() const = 0; 170 171 virtual void RegisterSendPayloadFrequency(int payload_type, 172 int payload_frequency) = 0; 173 174 // Unregisters a send payload. 175 // |payload_type| - payload type of codec 176 // Returns -1 on failure else 0. 177 virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0; 178 179 virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0; 180 181 // Register extension by uri, triggers CHECK on falure. 182 virtual void RegisterRtpHeaderExtension(absl::string_view uri, int id) = 0; 183 184 virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0; 185 virtual void DeregisterSendRtpHeaderExtension(absl::string_view uri) = 0; 186 187 // Returns true if RTP module is send media, and any of the extensions 188 // required for bandwidth estimation is registered. 189 virtual bool SupportsPadding() const = 0; 190 // Same as SupportsPadding(), but additionally requires that 191 // SetRtxSendStatus() has been called with the kRtxRedundantPayloads option 192 // enabled. 193 virtual bool SupportsRtxPayloadPadding() const = 0; 194 195 // Returns start timestamp. 196 virtual uint32_t StartTimestamp() const = 0; 197 198 // Sets start timestamp. Start timestamp is set to a random value if this 199 // function is never called. 200 virtual void SetStartTimestamp(uint32_t timestamp) = 0; 201 202 // Returns SequenceNumber. 203 virtual uint16_t SequenceNumber() const = 0; 204 205 // Sets SequenceNumber, default is a random number. 206 virtual void SetSequenceNumber(uint16_t seq) = 0; 207 208 virtual void SetRtpState(const RtpState& rtp_state) = 0; 209 virtual void SetRtxState(const RtpState& rtp_state) = 0; 210 virtual RtpState GetRtpState() const = 0; 211 virtual RtpState GetRtxState() const = 0; 212 213 // Returns SSRC. 214 virtual uint32_t SSRC() const = 0; 215 216 // Sets the value for sending in the RID (and Repaired) RTP header extension. 217 // RIDs are used to identify an RTP stream if SSRCs are not negotiated. 218 // If the RID and Repaired RID extensions are not registered, the RID will 219 // not be sent. 220 virtual void SetRid(const std::string& rid) = 0; 221 222 // Sets the value for sending in the MID RTP header extension. 223 // The MID RTP header extension should be registered for this to do anything. 224 // Once set, this value can not be changed or removed. 225 virtual void SetMid(const std::string& mid) = 0; 226 227 // Sets CSRC. 228 // |csrcs| - vector of CSRCs 229 virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0; 230 231 // Turns on/off sending RTX (RFC 4588). The modes can be set as a combination 232 // of values of the enumerator RtxMode. 233 virtual void SetRtxSendStatus(int modes) = 0; 234 235 // Returns status of sending RTX (RFC 4588). The returned value can be 236 // a combination of values of the enumerator RtxMode. 237 virtual int RtxSendStatus() const = 0; 238 239 // Returns the SSRC used for RTX if set, otherwise a nullopt. 240 virtual absl::optional<uint32_t> RtxSsrc() const = 0; 241 242 // Sets the payload type to use when sending RTX packets. Note that this 243 // doesn't enable RTX, only the payload type is set. 244 virtual void SetRtxSendPayloadType(int payload_type, 245 int associated_payload_type) = 0; 246 247 // Returns the FlexFEC SSRC, if there is one. 248 virtual absl::optional<uint32_t> FlexfecSsrc() const = 0; 249 250 // Sets sending status. Sends kRtcpByeCode when going from true to false. 251 // Returns -1 on failure else 0. 252 virtual int32_t SetSendingStatus(bool sending) = 0; 253 254 // Returns current sending status. 255 virtual bool Sending() const = 0; 256 257 // Starts/Stops media packets. On by default. 258 virtual void SetSendingMediaStatus(bool sending) = 0; 259 260 // Returns current media sending status. 261 virtual bool SendingMedia() const = 0; 262 263 // Returns whether audio is configured (i.e. Configuration::audio = true). 264 virtual bool IsAudioConfigured() const = 0; 265 266 // Indicate that the packets sent by this module should be counted towards the 267 // bitrate estimate since the stream participates in the bitrate allocation. 268 virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0; 269 270 // Returns bitrate sent (post-pacing) per packet type. 271 virtual RtpSendRates GetSendRates() const = 0; 272 273 virtual RTPSender* RtpSender() = 0; 274 virtual const RTPSender* RtpSender() const = 0; 275 276 // Record that a frame is about to be sent. Returns true on success, and false 277 // if the module isn't ready to send. 278 virtual bool OnSendingRtpFrame(uint32_t timestamp, 279 int64_t capture_time_ms, 280 int payload_type, 281 bool force_sender_report) = 0; 282 283 // Try to send the provided packet. Returns true iff packet matches any of 284 // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the 285 // transport. 286 virtual bool TrySendPacket(RtpPacketToSend* packet, 287 const PacedPacketInfo& pacing_info) = 0; 288 289 // Update the FEC protection parameters to use for delta- and key-frames. 290 // Only used when deferred FEC is active. 291 virtual void SetFecProtectionParams( 292 const FecProtectionParams& delta_params, 293 const FecProtectionParams& key_params) = 0; 294 295 // If deferred FEC generation is enabled, this method should be called after 296 // calling TrySendPacket(). Any generated FEC packets will be removed and 297 // returned from the FEC generator. 298 virtual std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0; 299 300 virtual void OnPacketsAcknowledged( 301 rtc::ArrayView<const uint16_t> sequence_numbers) = 0; 302 303 virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( 304 size_t target_size_bytes) = 0; 305 306 virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos( 307 rtc::ArrayView<const uint16_t> sequence_numbers) const = 0; 308 309 // Returns an expected per packet overhead representing the main RTP header, 310 // any CSRCs, and the registered header extensions that are expected on all 311 // packets (i.e. disregarding things like abs capture time which is only 312 // populated on a subset of packets, but counting MID/RID type extensions 313 // when we expect to send them). 314 virtual size_t ExpectedPerPacketOverhead() const = 0; 315 316 // ************************************************************************** 317 // RTCP 318 // ************************************************************************** 319 320 // Returns RTCP status. 321 virtual RtcpMode RTCP() const = 0; 322 323 // Sets RTCP status i.e on(compound or non-compound)/off. 324 // |method| - RTCP method to use. 325 virtual void SetRTCPStatus(RtcpMode method) = 0; 326 327 // Sets RTCP CName (i.e unique identifier). 328 // Returns -1 on failure else 0. 329 virtual int32_t SetCNAME(const char* cname) = 0; 330 331 // Returns remote NTP. 332 // Returns -1 on failure else 0. 333 virtual int32_t RemoteNTP(uint32_t* received_ntp_secs, 334 uint32_t* received_ntp_frac, 335 uint32_t* rtcp_arrival_time_secs, 336 uint32_t* rtcp_arrival_time_frac, 337 uint32_t* rtcp_timestamp) const = 0; 338 339 // Returns current RTT (round-trip time) estimate. 340 // Returns -1 on failure else 0. 341 virtual int32_t RTT(uint32_t remote_ssrc, 342 int64_t* rtt, 343 int64_t* avg_rtt, 344 int64_t* min_rtt, 345 int64_t* max_rtt) const = 0; 346 347 // Returns the estimated RTT, with fallback to a default value. 348 virtual int64_t ExpectedRetransmissionTimeMs() const = 0; 349 350 // Forces a send of a RTCP packet. Periodic SR and RR are triggered via the 351 // process function. 352 // Returns -1 on failure else 0. 353 virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0; 354 355 // Returns send statistics for the RTP and RTX stream. 356 virtual void GetSendStreamDataCounters( 357 StreamDataCounters* rtp_counters, 358 StreamDataCounters* rtx_counters) const = 0; 359 360 // Returns received RTCP report block. 361 // Returns -1 on failure else 0. 362 // TODO(https://crbug.com/webrtc/10678): Remove this in favor of 363 // GetLatestReportBlockData(). 364 virtual int32_t RemoteRTCPStat( 365 std::vector<RTCPReportBlock>* receive_blocks) const = 0; 366 // A snapshot of Report Blocks with additional data of interest to statistics. 367 // Within this list, the sender-source SSRC pair is unique and per-pair the 368 // ReportBlockData represents the latest Report Block that was received for 369 // that pair. 370 virtual std::vector<ReportBlockData> GetLatestReportBlockData() const = 0; 371 372 // (XR) Sets Receiver Reference Time Report (RTTR) status. 373 virtual void SetRtcpXrRrtrStatus(bool enable) = 0; 374 375 // Returns current Receiver Reference Time Report (RTTR) status. 376 virtual bool RtcpXrRrtrStatus() const = 0; 377 378 // (REMB) Receiver Estimated Max Bitrate. 379 // Schedules sending REMB on next and following sender/receiver reports. 380 void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override = 0; 381 // Stops sending REMB on next and following sender/receiver reports. 382 void UnsetRemb() override = 0; 383 384 // (NACK) 385 386 // Sends a Negative acknowledgement packet. 387 // Returns -1 on failure else 0. 388 // TODO(philipel): Deprecate this and start using SendNack instead, mostly 389 // because we want a function that actually send NACK for the specified 390 // packets. 391 virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0; 392 393 // Sends NACK for the packets specified. 394 // Note: This assumes the caller keeps track of timing and doesn't rely on 395 // the RTP module to do this. 396 virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0; 397 398 // Store the sent packets, needed to answer to a Negative acknowledgment 399 // requests. 400 virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0; 401 402 // Returns true if the module is configured to store packets. 403 virtual bool StorePackets() const = 0; 404 405 virtual void SetVideoBitrateAllocation( 406 const VideoBitrateAllocation& bitrate) = 0; 407 408 // ************************************************************************** 409 // Video 410 // ************************************************************************** 411 412 // Requests new key frame. 413 // using PLI, https://tools.ietf.org/html/rfc4585#section-6.3.1.1 SendPictureLossIndication()414 void SendPictureLossIndication() { SendRTCP(kRtcpPli); } 415 // using FIR, https://tools.ietf.org/html/rfc5104#section-4.3.1.2 SendFullIntraRequest()416 void SendFullIntraRequest() { SendRTCP(kRtcpFir); } 417 418 // Sends a LossNotification RTCP message. 419 // Returns -1 on failure else 0. 420 virtual int32_t SendLossNotification(uint16_t last_decoded_seq_num, 421 uint16_t last_received_seq_num, 422 bool decodability_flag, 423 bool buffering_allowed) = 0; 424 }; 425 426 } // namespace webrtc 427 428 #endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_ 429