/* * Copyright (c) 2012 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_INCLUDE_RTP_RTCP_H_ #define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ #include #include #include #include #include #include "absl/strings/string_view.h" #include "absl/types/optional.h" #include "api/transport/webrtc_key_value_config.h" #include "api/video/video_bitrate_allocation.h" #include "modules/include/module.h" #include "modules/rtp_rtcp/include/flexfec_sender.h" #include "modules/rtp_rtcp/include/receive_statistics.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtp_sender.h" #include "rtc_base/constructor_magic.h" #include "rtc_base/deprecation.h" namespace webrtc { // Forward declarations. class FrameEncryptorInterface; class OverheadObserver; class RateLimiter; class ReceiveStatisticsProvider; class RemoteBitrateEstimator; class RtcEventLog; class Transport; class VideoBitrateAllocationObserver; namespace rtcp { class TransportFeedback; } class RtpRtcp : public Module, public RtcpFeedbackSenderInterface { public: struct Configuration { Configuration(); // 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 request a intra frame. RtcpIntraFrameObserver* intra_frame_callback = nullptr; // Called when we receive a changed estimate from the receiver of out // stream. RtcpBandwidthObserver* bandwidth_callback = nullptr; TransportFeedbackObserver* transport_feedback_callback = nullptr; VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr; RtcpRttStats* rtt_stats = nullptr; RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr; // Estimates the bandwidth available for a set of streams from the same // client. RemoteBitrateEstimator* remote_bitrate_estimator = nullptr; // Spread any bursts of packets into smaller bursts to minimize packet loss. RtpPacketSender* paced_sender = nullptr; // Generate FlexFEC packets. // TODO(brandtr): Remove when FlexfecSender is wired up to PacedSender. FlexfecSender* flexfec_sender = nullptr; TransportSequenceNumberAllocator* transport_sequence_number_allocator = 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; OverheadObserver* overhead_observer = nullptr; RtcpAckObserver* ack_observer = nullptr; RtpKeepAliveConfig keepalive_config; int rtcp_report_interval_ms = 0; // Update network2 instead of pacer_exit field of video timing extension. bool populate_network2_timestamp = false; // 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 set, field trials are read from |field_trials|, otherwise // defaults to webrtc::FieldTrialBasedConfig. const WebRtcKeyValueConfig* field_trials = nullptr; private: RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); }; // Creates an RTP/RTCP module object using provided |configuration|. static std::unique_ptr Create(const Configuration& configuration); // Prefer factory function just above. RTC_DEPRECATED static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration); // ************************************************************************** // Receiver functions // ************************************************************************** virtual void IncomingRtcpPacket(const uint8_t* incoming_packet, size_t incoming_packet_length) = 0; virtual void SetRemoteSSRC(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 RegisterAudioSendPayload(int payload_type, absl::string_view payload_name, int frequency, int channels, int rate) = 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; // (De)registers RTP header extension type and id. // Returns -1 on failure else 0. virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type, uint8_t id) = 0; // Register extension by uri, returns false on failure. virtual bool RegisterRtpHeaderExtension(const std::string& uri, int id) = 0; virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0; virtual bool HasBweExtensions() 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; // Returns SSRC. uint32_t SSRC() const override = 0; // Sets SSRC, default is a random number. virtual void SetSSRC(uint32_t ssrc) = 0; // 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. virtual void SetRid(const std::string& rid) = 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(const std::string& 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; // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX, // only the SSRC is set. virtual void SetRtxSsrc(uint32_t ssrc) = 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; // 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; // Fetches the current send bitrates in bits/s. virtual void BitrateSent(uint32_t* total_rate, uint32_t* video_rate, uint32_t* fec_rate, uint32_t* nack_rate) const = 0; virtual RTPSender* RtpSender() = 0; virtual const RTPSender* RtpSender() const = 0; // Used by the codec module to deliver a video or audio frame for // packetization. // |frame_type| - type of frame to send // |payload_type| - payload type of frame to send // |timestamp| - timestamp of frame to send // |payload_data| - payload buffer of frame to send // |payload_size| - size of payload buffer to send // |fragmentation| - fragmentation offset data for fragmented frames such // as layers or RED // |transport_frame_id_out| - set to RTP timestamp. // Returns true on success. virtual bool SendOutgoingData(FrameType frame_type, int8_t payload_type, uint32_t timestamp, int64_t capture_time_ms, const uint8_t* payload_data, size_t payload_size, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtp_video_header, uint32_t* transport_frame_id_out) = 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; virtual bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission, const PacedPacketInfo& pacing_info) = 0; virtual size_t TimeToSendPadding(size_t bytes, const PacedPacketInfo& pacing_info) = 0; // Called on generation of new statistics after an RTP send. virtual void RegisterSendChannelRtpStatisticsCallback( StreamDataCountersCallback* callback) = 0; virtual StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback() const = 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(const char* cname) = 0; // Returns remote CName. // Returns -1 on failure else 0. virtual int32_t RemoteCNAME(uint32_t remote_ssrc, char cname[RTCP_CNAME_SIZE]) const = 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 -1 on failure else 0. virtual int32_t AddMixedCNAME(uint32_t ssrc, const char* cname) = 0; // Returns -1 on failure else 0. virtual int32_t RemoveMixedCNAME(uint32_t ssrc) = 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; // Forces a send of a RTCP packet with more than one packet type. // periodic SR and RR are triggered via the process function // Returns -1 on failure else 0. virtual int32_t SendCompoundRTCP( const std::set& rtcp_packet_types) = 0; // Returns statistics of the amount of data sent. // Returns -1 on failure else 0. virtual int32_t DataCountersRTP(size_t* bytes_sent, uint32_t* packets_sent) const = 0; // Returns send statistics for the RTP and RTX stream. virtual void GetSendStreamDataCounters( StreamDataCounters* rtp_counters, StreamDataCounters* rtx_counters) const = 0; // Returns packet loss statistics for the RTP stream. virtual void GetRtpPacketLossStats( bool outgoing, uint32_t ssrc, struct RtpPacketLossStats* loss_stats) const = 0; // Returns received RTCP report block. // Returns -1 on failure else 0. virtual int32_t RemoteRTCPStat( std::vector* receive_blocks) const = 0; // (APP) Sets application specific data. // Returns -1 on failure else 0. virtual int32_t SetRTCPApplicationSpecificData(uint8_t sub_type, uint32_t name, const uint8_t* data, uint16_t length) = 0; // (XR) Sets Receiver Reference Time Report (RTTR) status. virtual void SetRtcpXrRrtrStatus(bool enable) = 0; // Returns current Receiver Reference Time Report (RTTR) status. virtual bool RtcpXrRrtrStatus() 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; // (TMMBR) Temporary Max Media Bit Rate virtual bool TMMBR() const = 0; virtual void SetTMMBRStatus(bool enable) = 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; // Returns true if the module is configured to store packets. virtual bool StorePackets() const = 0; // Called on receipt of RTCP report block from remote side. virtual void RegisterRtcpStatisticsCallback( RtcpStatisticsCallback* callback) = 0; virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; // BWE feedback packets. bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override = 0; virtual void SetVideoBitrateAllocation( const VideoBitrateAllocation& bitrate) = 0; // ************************************************************************** // Audio // ************************************************************************** // Sends a TelephoneEvent tone using RFC 2833 (4733). // Returns -1 on failure else 0. virtual int32_t SendTelephoneEventOutband(uint8_t key, uint16_t time_ms, uint8_t level) = 0; // Store the audio level in dBov for header-extension-for-audio-level- // indication. // This API shall be called before transmision of an RTP packet to ensure // that the |level| part of the extended RTP header is updated. // return -1 on failure else 0. virtual int32_t SetAudioLevel(uint8_t level_dbov) = 0; // ************************************************************************** // Video // ************************************************************************** // Set method for requestion a new key frame. // Returns -1 on failure else 0. virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; // Sends a request for a keyframe. // Returns -1 on failure else 0. virtual int32_t RequestKeyFrame() = 0; // 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) = 0; }; } // namespace webrtc #endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_