/* * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_ #define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_ #include #include #include #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/field_trials_view.h" #include "api/frame_transformer_interface.h" #include "api/scoped_refptr.h" #include "api/video/video_bitrate_allocation.h" #include "modules/rtp_rtcp/include/receive_statistics.h" #include "modules/rtp_rtcp/include/report_block_data.h" #include "modules/rtp_rtcp/include/rtp_packet_sender.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" #include "modules/rtp_rtcp/source/video_fec_generator.h" #include "system_wrappers/include/ntp_time.h" namespace webrtc { // Forward declarations. class FrameEncryptorInterface; class RateLimiter; class RtcEventLog; class RTPSender; class Transport; class VideoBitrateAllocationObserver; class RtpRtcpInterface : public RtcpFeedbackSenderInterface { public: struct Configuration { Configuration() = default; Configuration(Configuration&& rhs) = default; Configuration(const Configuration&) = delete; Configuration& operator=(const Configuration&) = delete; // True for a audio version of the RTP/RTCP module object false will create // a video version. bool audio = false; bool receiver_only = false; // The clock to use to read time. If nullptr then system clock will be used. Clock* clock = nullptr; ReceiveStatisticsProvider* receive_statistics = nullptr; // Transport object that will be called when packets are ready to be sent // out on the network. Transport* outgoing_transport = nullptr; // Called when the receiver requests an intra frame. RtcpIntraFrameObserver* intra_frame_callback = nullptr; // Called when the receiver sends a loss notification. RtcpLossNotificationObserver* rtcp_loss_notification_observer = nullptr; // Called when we receive a changed estimate from the receiver of out // stream. RtcpBandwidthObserver* bandwidth_callback = nullptr; NetworkStateEstimateObserver* network_state_estimate_observer = nullptr; TransportFeedbackObserver* transport_feedback_callback = nullptr; VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr; RtcpRttStats* rtt_stats = nullptr; RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr; // Called on receipt of RTCP report block from remote side. // TODO(bugs.webrtc.org/10679): Consider whether we want to use // only getters or only callbacks. If we decide on getters, the // ReportBlockDataObserver should also be removed in favor of // GetLatestReportBlockData(). RtcpCnameCallback* rtcp_cname_callback = nullptr; ReportBlockDataObserver* report_block_data_observer = nullptr; // Spread any bursts of packets into smaller bursts to minimize packet loss. RtpPacketSender* paced_sender = nullptr; // Generates FEC packets. // TODO(sprang): Wire up to RtpSenderEgress. VideoFecGenerator* fec_generator = nullptr; BitrateStatisticsObserver* send_bitrate_observer = nullptr; SendSideDelayObserver* send_side_delay_observer = nullptr; RtcEventLog* event_log = nullptr; SendPacketObserver* send_packet_observer = nullptr; RateLimiter* retransmission_rate_limiter = nullptr; StreamDataCountersCallback* rtp_stats_callback = nullptr; int rtcp_report_interval_ms = 0; // Update network2 instead of pacer_exit field of video timing extension. bool populate_network2_timestamp = false; rtc::scoped_refptr frame_transformer; // E2EE Custom Video Frame Encryption FrameEncryptorInterface* frame_encryptor = nullptr; // Require all outgoing frames to be encrypted with a FrameEncryptor. bool require_frame_encryption = false; // Corresponds to extmap-allow-mixed in SDP negotiation. bool extmap_allow_mixed = false; // If true, the RTP sender will always annotate outgoing packets with // MID and RID header extensions, if provided and negotiated. // If false, the RTP sender will stop sending MID and RID header extensions, // when it knows that the receiver is ready to demux based on SSRC. This is // done by RTCP RR acking. bool always_send_mid_and_rid = false; // If set, field trials are read from `field_trials`, otherwise // defaults to webrtc::FieldTrialBasedConfig. const FieldTrialsView* field_trials = nullptr; // SSRCs for media and retransmission, respectively. // FlexFec SSRC is fetched from `flexfec_sender`. uint32_t local_media_ssrc = 0; absl::optional rtx_send_ssrc; bool need_rtp_packet_infos = false; // If true, the RTP packet history will select RTX packets based on // heuristics such as send time, retransmission count etc, in order to // make padding potentially more useful. // If false, the last packet will always be picked. This may reduce CPU // overhead. bool enable_rtx_padding_prioritization = true; // Estimate RTT as non-sender as described in // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5 bool non_sender_rtt_measurement = false; // If non-empty, sets the value for sending in the RID (and Repaired) RTP // header extension. RIDs are used to identify an RTP stream if SSRCs are // not negotiated. If the RID and Repaired RID extensions are not // registered, the RID will not be sent. std::string rid; }; // Stats for RTCP sender reports (SR) for a specific SSRC. // Refer to https://tools.ietf.org/html/rfc3550#section-6.4.1. struct SenderReportStats { // Arrival NTP timestamp for the last received RTCP SR. NtpTime last_arrival_timestamp; // Received (a.k.a., remote) NTP timestamp for the last received RTCP SR. NtpTime last_remote_timestamp; // Total number of RTP data packets transmitted by the sender since starting // transmission up until the time this SR packet was generated. The count // should be reset if the sender changes its SSRC identifier. uint32_t packets_sent; // Total number of payload octets (i.e., not including header or padding) // transmitted in RTP data packets by the sender since starting transmission // up until the time this SR packet was generated. The count should be reset // if the sender changes its SSRC identifier. uint64_t bytes_sent; // Total number of RTCP SR blocks received. // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent. uint64_t reports_count; }; // Stats about the non-sender SSRC, based on RTCP extended reports (XR). // Refer to https://datatracker.ietf.org/doc/html/rfc3611#section-2. struct NonSenderRttStats { // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime absl::optional round_trip_time; // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime TimeDelta total_round_trip_time = TimeDelta::Zero(); // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements int round_trip_time_measurements = 0; }; // ************************************************************************** // Receiver functions // ************************************************************************** virtual void IncomingRtcpPacket(const uint8_t* incoming_packet, size_t incoming_packet_length) = 0; virtual void SetRemoteSSRC(uint32_t ssrc) = 0; // Called when the local ssrc changes (post initialization) for receive // streams to match with send. Called on the packet receive thread/tq. virtual void SetLocalSsrc(uint32_t ssrc) = 0; // ************************************************************************** // Sender // ************************************************************************** // Sets the maximum size of an RTP packet, including RTP headers. virtual void SetMaxRtpPacketSize(size_t size) = 0; // Returns max RTP packet size. Takes into account RTP headers and // FEC/ULP/RED overhead (when FEC is enabled). virtual size_t MaxRtpPacketSize() const = 0; virtual void RegisterSendPayloadFrequency(int payload_type, int payload_frequency) = 0; // Unregisters a send payload. // `payload_type` - payload type of codec // Returns -1 on failure else 0. virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0; virtual void SetExtmapAllowMixed(bool extmap_allow_mixed) = 0; // Register extension by uri, triggers CHECK on falure. virtual void RegisterRtpHeaderExtension(absl::string_view uri, int id) = 0; virtual void DeregisterSendRtpHeaderExtension(absl::string_view uri) = 0; // Returns true if RTP module is send media, and any of the extensions // required for bandwidth estimation is registered. virtual bool SupportsPadding() const = 0; // Same as SupportsPadding(), but additionally requires that // SetRtxSendStatus() has been called with the kRtxRedundantPayloads option // enabled. virtual bool SupportsRtxPayloadPadding() const = 0; // Returns start timestamp. virtual uint32_t StartTimestamp() const = 0; // Sets start timestamp. Start timestamp is set to a random value if this // function is never called. virtual void SetStartTimestamp(uint32_t timestamp) = 0; // Returns SequenceNumber. virtual uint16_t SequenceNumber() const = 0; // Sets SequenceNumber, default is a random number. virtual void SetSequenceNumber(uint16_t seq) = 0; virtual void SetRtpState(const RtpState& rtp_state) = 0; virtual void SetRtxState(const RtpState& rtp_state) = 0; virtual RtpState GetRtpState() const = 0; virtual RtpState GetRtxState() const = 0; // This can be used to enable/disable receive-side RTT. virtual void SetNonSenderRttMeasurement(bool enabled) = 0; // Returns SSRC. virtual uint32_t SSRC() const = 0; // Sets the value for sending in the MID RTP header extension. // The MID RTP header extension should be registered for this to do anything. // Once set, this value can not be changed or removed. virtual void SetMid(absl::string_view mid) = 0; // Sets CSRC. // `csrcs` - vector of CSRCs virtual void SetCsrcs(const std::vector& csrcs) = 0; // Turns on/off sending RTX (RFC 4588). The modes can be set as a combination // of values of the enumerator RtxMode. virtual void SetRtxSendStatus(int modes) = 0; // Returns status of sending RTX (RFC 4588). The returned value can be // a combination of values of the enumerator RtxMode. virtual int RtxSendStatus() const = 0; // Returns the SSRC used for RTX if set, otherwise a nullopt. virtual absl::optional RtxSsrc() const = 0; // Sets the payload type to use when sending RTX packets. Note that this // doesn't enable RTX, only the payload type is set. virtual void SetRtxSendPayloadType(int payload_type, int associated_payload_type) = 0; // Returns the FlexFEC SSRC, if there is one. virtual absl::optional FlexfecSsrc() const = 0; // Sets sending status. Sends kRtcpByeCode when going from true to false. // Returns -1 on failure else 0. virtual int32_t SetSendingStatus(bool sending) = 0; // Returns current sending status. virtual bool Sending() const = 0; // Starts/Stops media packets. On by default. virtual void SetSendingMediaStatus(bool sending) = 0; // Returns current media sending status. virtual bool SendingMedia() const = 0; // Returns whether audio is configured (i.e. Configuration::audio = true). virtual bool IsAudioConfigured() const = 0; // Indicate that the packets sent by this module should be counted towards the // bitrate estimate since the stream participates in the bitrate allocation. virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0; // Returns bitrate sent (post-pacing) per packet type. virtual RtpSendRates GetSendRates() const = 0; virtual RTPSender* RtpSender() = 0; virtual const RTPSender* RtpSender() const = 0; // Record that a frame is about to be sent. Returns true on success, and false // if the module isn't ready to send. virtual bool OnSendingRtpFrame(uint32_t timestamp, int64_t capture_time_ms, int payload_type, bool force_sender_report) = 0; // Try to send the provided packet. Returns true iff packet matches any of // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the // transport. virtual bool TrySendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info) = 0; // Update the FEC protection parameters to use for delta- and key-frames. // Only used when deferred FEC is active. virtual void SetFecProtectionParams( const FecProtectionParams& delta_params, const FecProtectionParams& key_params) = 0; // If deferred FEC generation is enabled, this method should be called after // calling TrySendPacket(). Any generated FEC packets will be removed and // returned from the FEC generator. virtual std::vector> FetchFecPackets() = 0; virtual void OnAbortedRetransmissions( rtc::ArrayView sequence_numbers) = 0; virtual void OnPacketsAcknowledged( rtc::ArrayView sequence_numbers) = 0; virtual std::vector> GeneratePadding( size_t target_size_bytes) = 0; virtual std::vector GetSentRtpPacketInfos( rtc::ArrayView sequence_numbers) const = 0; // Returns an expected per packet overhead representing the main RTP header, // any CSRCs, and the registered header extensions that are expected on all // packets (i.e. disregarding things like abs capture time which is only // populated on a subset of packets, but counting MID/RID type extensions // when we expect to send them). virtual size_t ExpectedPerPacketOverhead() const = 0; // Access to packet state (e.g. sequence numbering) must only be access by // one thread at a time. It may be only one thread, or a construction thread // that calls SetRtpState() - handing over to a pacer thread that calls // TrySendPacket() - and at teardown ownership is handed to a destruciton // thread that calls GetRtpState(). // This method is used to signal that "ownership" of the rtp state is being // transferred to another thread. virtual void OnPacketSendingThreadSwitched() = 0; // ************************************************************************** // RTCP // ************************************************************************** // Returns RTCP status. virtual RtcpMode RTCP() const = 0; // Sets RTCP status i.e on(compound or non-compound)/off. // `method` - RTCP method to use. virtual void SetRTCPStatus(RtcpMode method) = 0; // Sets RTCP CName (i.e unique identifier). // Returns -1 on failure else 0. virtual int32_t SetCNAME(absl::string_view cname) = 0; // Returns remote NTP. // Returns -1 on failure else 0. virtual int32_t RemoteNTP(uint32_t* received_ntp_secs, uint32_t* received_ntp_frac, uint32_t* rtcp_arrival_time_secs, uint32_t* rtcp_arrival_time_frac, uint32_t* rtcp_timestamp) const = 0; // Returns current RTT (round-trip time) estimate. // Returns -1 on failure else 0. virtual int32_t RTT(uint32_t remote_ssrc, int64_t* rtt, int64_t* avg_rtt, int64_t* min_rtt, int64_t* max_rtt) const = 0; // Returns the estimated RTT, with fallback to a default value. virtual int64_t ExpectedRetransmissionTimeMs() const = 0; // Forces a send of a RTCP packet. Periodic SR and RR are triggered via the // process function. // Returns -1 on failure else 0. virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0; // Returns send statistics for the RTP and RTX stream. virtual void GetSendStreamDataCounters( StreamDataCounters* rtp_counters, StreamDataCounters* rtx_counters) const = 0; // A snapshot of Report Blocks with additional data of interest to statistics. // Within this list, the sender-source SSRC pair is unique and per-pair the // ReportBlockData represents the latest Report Block that was received for // that pair. virtual std::vector GetLatestReportBlockData() const = 0; // Returns stats based on the received RTCP SRs. virtual absl::optional GetSenderReportStats() const = 0; // Returns non-sender RTT stats, based on DLRR. virtual absl::optional GetNonSenderRttStats() const = 0; // (REMB) Receiver Estimated Max Bitrate. // Schedules sending REMB on next and following sender/receiver reports. void SetRemb(int64_t bitrate_bps, std::vector ssrcs) override = 0; // Stops sending REMB on next and following sender/receiver reports. void UnsetRemb() override = 0; // (NACK) // Sends a Negative acknowledgement packet. // Returns -1 on failure else 0. // TODO(philipel): Deprecate this and start using SendNack instead, mostly // because we want a function that actually send NACK for the specified // packets. virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0; // Sends NACK for the packets specified. // Note: This assumes the caller keeps track of timing and doesn't rely on // the RTP module to do this. virtual void SendNack(const std::vector& sequence_numbers) = 0; // Store the sent packets, needed to answer to a Negative acknowledgment // requests. virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0; virtual void SetVideoBitrateAllocation( const VideoBitrateAllocation& bitrate) = 0; // ************************************************************************** // Video // ************************************************************************** // Requests new key frame. // using PLI, https://tools.ietf.org/html/rfc4585#section-6.3.1.1 void SendPictureLossIndication() { SendRTCP(kRtcpPli); } // using FIR, https://tools.ietf.org/html/rfc5104#section-4.3.1.2 void SendFullIntraRequest() { SendRTCP(kRtcpFir); } // Sends a LossNotification RTCP message. // Returns -1 on failure else 0. virtual int32_t SendLossNotification(uint16_t last_decoded_seq_num, uint16_t last_received_seq_num, bool decodability_flag, bool buffering_allowed) = 0; }; } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_INTERFACE_H_