| /* |
| * 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 <memory> |
| #include <string> |
| #include <vector> |
| |
| #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<FrameTransformerInterface> 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<uint32_t> 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; |
| // Received (a.k.a., remote) RTP timestamp from the last received RTCP SR. |
| uint32_t last_remote_rtp_timestamp = 0; |
| // 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 = 0; |
| // 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 = 0; |
| // Total number of RTCP SR blocks received. |
| // https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-reportssent. |
| uint64_t reports_count = 0; |
| }; |
| // 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<TimeDelta> 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( |
| rtc::ArrayView<const uint8_t> incoming_packet) = 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; |
| |
| // 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<uint32_t> 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<uint32_t> 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<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() = 0; |
| |
| virtual void OnAbortedRetransmissions( |
| rtc::ArrayView<const uint16_t> sequence_numbers) = 0; |
| |
| virtual void OnPacketsAcknowledged( |
| rtc::ArrayView<const uint16_t> sequence_numbers) = 0; |
| |
| virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( |
| size_t target_size_bytes) = 0; |
| |
| virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos( |
| rtc::ArrayView<const uint16_t> 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 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<ReportBlockData> GetLatestReportBlockData() const = 0; |
| // Returns stats based on the received RTCP SRs. |
| virtual absl::optional<SenderReportStats> GetSenderReportStats() const = 0; |
| // Returns non-sender RTT stats, based on DLRR. |
| virtual absl::optional<NonSenderRttStats> 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<uint32_t> 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<uint16_t>& 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_ |