/* * 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 WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ #define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ #include #include #include "webrtc/modules/interface/module.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" namespace webrtc { // Forward declarations. class PacedSender; class ReceiveStatistics; class RemoteBitrateEstimator; class RtpReceiver; class Transport; class RtpRtcp : public Module { public: struct Configuration { Configuration(); /* id - Unique identifier of this RTP/RTCP module object * audio - True for a audio version of the RTP/RTCP module * object false will create a video version * clock - The clock to use to read time. If NULL object * will be using the system clock. * incoming_data - Callback object that will receive the incoming * data. May not be NULL; default callback will do * nothing. * incoming_messages - Callback object that will receive the incoming * RTP messages. May not be NULL; default callback * will do nothing. * outgoing_transport - Transport object that will be called when packets * are ready to be sent out on the network * intra_frame_callback - Called when the receiver request a intra frame. * bandwidth_callback - Called when we receive a changed estimate from * the receiver of out stream. * audio_messages - Telephone events. May not be NULL; default * callback will do nothing. * remote_bitrate_estimator - Estimates the bandwidth available for a set of * streams from the same client. * paced_sender - Spread any bursts of packets into smaller * bursts to minimize packet loss. */ int32_t id; bool audio; bool receiver_only; Clock* clock; ReceiveStatistics* receive_statistics; Transport* outgoing_transport; RtcpIntraFrameObserver* intra_frame_callback; RtcpBandwidthObserver* bandwidth_callback; RtcpRttStats* rtt_stats; RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; RtpAudioFeedback* audio_messages; RemoteBitrateEstimator* remote_bitrate_estimator; PacedSender* paced_sender; BitrateStatisticsObserver* send_bitrate_observer; FrameCountObserver* send_frame_count_observer; SendSideDelayObserver* send_side_delay_observer; }; /* * Create a RTP/RTCP module object using the system clock. * * configuration - Configuration of the RTP/RTCP module. */ static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration); /************************************************************************** * * Receiver functions * ***************************************************************************/ virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, size_t incoming_packet_length) = 0; virtual void SetRemoteSSRC(uint32_t ssrc) = 0; /************************************************************************** * * Sender * ***************************************************************************/ /* * set MTU * * size - Max transfer unit in bytes, default is 1500 * * return -1 on failure else 0 */ virtual int32_t SetMaxTransferUnit(uint16_t size) = 0; /* * set transtport overhead * default is IPv4 and UDP with no encryption * * TCP - true for TCP false UDP * IPv6 - true for IP version 6 false for version 4 * authenticationOverhead - number of bytes to leave for an * authentication header * * return -1 on failure else 0 */ virtual int32_t SetTransportOverhead( bool TCP, bool IPV6, uint8_t authenticationOverhead = 0) = 0; /* * Get max payload length * * A combination of the configuration MaxTransferUnit and * TransportOverhead. * Does not account FEC/ULP/RED overhead if FEC is enabled. * Does not account for RTP headers */ virtual uint16_t MaxPayloadLength() const = 0; /* * Get max data payload length * * A combination of the configuration MaxTransferUnit, headers and * TransportOverhead. * Takes into account FEC/ULP/RED overhead if FEC is enabled. * Takes into account RTP headers */ virtual uint16_t MaxDataPayloadLength() const = 0; /* * set codec name and payload type * * return -1 on failure else 0 */ virtual int32_t RegisterSendPayload( const CodecInst& voiceCodec) = 0; /* * set codec name and payload type * * return -1 on failure else 0 */ virtual int32_t RegisterSendPayload( const VideoCodec& videoCodec) = 0; /* * Unregister a send payload * * payloadType - payload type of codec * * return -1 on failure else 0 */ virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0; /* * (De)register RTP header extension type and id. * * return -1 on failure else 0 */ virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type, uint8_t id) = 0; virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0; /* * get start timestamp */ virtual uint32_t StartTimestamp() const = 0; /* * configure start timestamp, default is a random number * * timestamp - start timestamp */ virtual void SetStartTimestamp(uint32_t timestamp) = 0; /* * Get SequenceNumber */ virtual uint16_t SequenceNumber() const = 0; /* * Set SequenceNumber, default is a random number */ virtual void SetSequenceNumber(uint16_t seq) = 0; // Returns true if the ssrc matched this module, false otherwise. virtual bool SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) = 0; virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0; /* * Get SSRC */ virtual uint32_t SSRC() const = 0; /* * configure SSRC, default is a random number */ virtual void SetSSRC(uint32_t ssrc) = 0; /* * Set CSRC * * csrcs - vector of CSRCs */ virtual void SetCsrcs(const std::vector& csrcs) = 0; /* * Turn 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; /* * Get 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; // Gets the payload type pair of (RTX, associated) to use when sending RTX // packets. virtual std::pair RtxSendPayloadType() const = 0; /* * sends kRtcpByeCode when going from true to false * * sending - on/off * * return -1 on failure else 0 */ virtual int32_t SetSendingStatus(bool sending) = 0; /* * get send status */ virtual bool Sending() const = 0; /* * Starts/Stops media packets, on by default * * sending - on/off */ virtual void SetSendingMediaStatus(bool sending) = 0; /* * get send status */ virtual bool SendingMedia() const = 0; /* * get sent bitrate in Kbit/s */ virtual void BitrateSent(uint32_t* totalRate, uint32_t* videoRate, uint32_t* fecRate, uint32_t* nackRate) const = 0; /* * Used by the codec module to deliver a video or audio frame for * packetization. * * frameType - type of frame to send * payloadType - payload type of frame to send * timestamp - timestamp of frame to send * payloadData - payload buffer of frame to send * payloadSize - size of payload buffer to send * fragmentation - fragmentation offset data for fragmented frames such * as layers or RED * * return -1 on failure else 0 */ virtual int32_t SendOutgoingData( FrameType frameType, int8_t payloadType, uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* fragmentation = NULL, const RTPVideoHeader* rtpVideoHdr = NULL) = 0; virtual bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms, bool retransmission) = 0; virtual size_t TimeToSendPadding(size_t bytes) = 0; // Called on generation of new statistics after an RTP send. virtual void RegisterSendChannelRtpStatisticsCallback( StreamDataCountersCallback* callback) = 0; virtual StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback() const = 0; /************************************************************************** * * RTCP * ***************************************************************************/ /* * Get RTCP status */ virtual RTCPMethod RTCP() const = 0; /* * configure RTCP status i.e on(compound or non- compound)/off * * method - RTCP method to use */ virtual void SetRTCPStatus(RTCPMethod method) = 0; /* * Set RTCP CName (i.e unique identifier) * * return -1 on failure else 0 */ virtual int32_t SetCNAME(const char* c_name) = 0; /* * Get remote CName * * return -1 on failure else 0 */ virtual int32_t RemoteCNAME(uint32_t remoteSSRC, char cName[RTCP_CNAME_SIZE]) const = 0; /* * Get remote NTP * * return -1 on failure else 0 */ virtual int32_t RemoteNTP( uint32_t *ReceivedNTPsecs, uint32_t *ReceivedNTPfrac, uint32_t *RTCPArrivalTimeSecs, uint32_t *RTCPArrivalTimeFrac, uint32_t *rtcp_timestamp) const = 0; /* * AddMixedCNAME * * return -1 on failure else 0 */ virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0; /* * RemoveMixedCNAME * * return -1 on failure else 0 */ virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0; /* * Get RoundTripTime * * return -1 on failure else 0 */ virtual int32_t RTT(uint32_t remoteSSRC, int64_t* RTT, int64_t* avgRTT, int64_t* minRTT, int64_t* maxRTT) const = 0; /* * Force a send of a RTCP packet * periodic SR and RR are triggered via the process function * * return -1 on failure else 0 */ virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0; /* * Force a send of a RTCP packet with more than one packet type. * periodic SR and RR are triggered via the process function * * return -1 on failure else 0 */ virtual int32_t SendCompoundRTCP( const std::set& rtcpPacketTypes) = 0; /* * Good state of RTP receiver inform sender */ virtual int32_t SendRTCPReferencePictureSelection( const uint64_t pictureID) = 0; /* * Send a RTCP Slice Loss Indication (SLI) * 6 least significant bits of pictureID */ virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0; /* * Statistics of the amount of data sent * * return -1 on failure else 0 */ virtual int32_t DataCountersRTP( size_t* bytesSent, uint32_t* packetsSent) const = 0; /* * Get send statistics for the RTP and RTX stream. */ virtual void GetSendStreamDataCounters( StreamDataCounters* rtp_counters, StreamDataCounters* rtx_counters) const = 0; /* * Get packet loss statistics for the RTP stream. */ virtual void GetRtpPacketLossStats( bool outgoing, uint32_t ssrc, struct RtpPacketLossStats* loss_stats) const = 0; /* * Get received RTCP sender info * * return -1 on failure else 0 */ virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0; /* * Get received RTCP report block * * return -1 on failure else 0 */ virtual int32_t RemoteRTCPStat( std::vector* receiveBlocks) const = 0; /* * (APP) Application specific data * * return -1 on failure else 0 */ virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType, uint32_t name, const uint8_t* data, uint16_t length) = 0; /* * (XR) VOIP metric * * return -1 on failure else 0 */ virtual int32_t SetRTCPVoIPMetrics( const RTCPVoIPMetric* VoIPMetric) = 0; /* * (XR) Receiver Reference Time Report */ virtual void SetRtcpXrRrtrStatus(bool enable) = 0; virtual bool RtcpXrRrtrStatus() const = 0; /* * (REMB) Receiver Estimated Max Bitrate */ virtual bool REMB() const = 0; virtual void SetREMBStatus(bool enable) = 0; virtual void SetREMBData(uint32_t bitrate, const std::vector& ssrcs) = 0; /* * (IJ) Extended jitter report. */ virtual bool IJ() const = 0; virtual void SetIJStatus(bool enable) = 0; /* * (TMMBR) Temporary Max Media Bit Rate */ virtual bool TMMBR() const = 0; virtual void SetTMMBRStatus(bool enable) = 0; /* * (NACK) */ /* * TODO(holmer): Propagate this API to VideoEngine. * Returns the currently configured selective retransmission settings. */ virtual int SelectiveRetransmissions() const = 0; /* * TODO(holmer): Propagate this API to VideoEngine. * Sets the selective retransmission settings, which will decide which * packets will be retransmitted if NACKed. Settings are constructed by * combining the constants in enum RetransmissionMode with bitwise OR. * All packets are retransmitted if kRetransmitAllPackets is set, while no * packets are retransmitted if kRetransmitOff is set. * By default all packets except FEC packets are retransmitted. For VP8 * with temporal scalability only base layer packets are retransmitted. * * Returns -1 on failure, otherwise 0. */ virtual int SetSelectiveRetransmissions(uint8_t settings) = 0; /* * Send a Negative acknowledgement packet * * return -1 on failure else 0 */ virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0; /* * Store the sent packets, needed to answer to a Negative acknowledgement * 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; /************************************************************************** * * Audio * ***************************************************************************/ /* * set audio packet size, used to determine when it's time to send a DTMF * packet in silence (CNG) * * return -1 on failure else 0 */ virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0; /* * Send a TelephoneEvent tone using RFC 2833 (4733) * * return -1 on failure else 0 */ virtual int32_t SendTelephoneEventOutband(uint8_t key, uint16_t time_ms, uint8_t level) = 0; /* * Set payload type for Redundant Audio Data RFC 2198 * * return -1 on failure else 0 */ virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0; /* * Get payload type for Redundant Audio Data RFC 2198 * * return -1 on failure else 0 */ virtual int32_t SendREDPayloadType( int8_t& payloadType) const = 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 the target send bitrate */ virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0; /* * Turn on/off generic FEC * * return -1 on failure else 0 */ virtual int32_t SetGenericFECStatus(bool enable, uint8_t payloadTypeRED, uint8_t payloadTypeFEC) = 0; /* * Get generic FEC setting * * return -1 on failure else 0 */ virtual int32_t GenericFECStatus(bool& enable, uint8_t& payloadTypeRED, uint8_t& payloadTypeFEC) = 0; virtual int32_t SetFecParameters( const FecProtectionParams* delta_params, const FecProtectionParams* key_params) = 0; /* * Set method for requestion a new key frame * * return -1 on failure else 0 */ virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; /* * send a request for a keyframe * * return -1 on failure else 0 */ virtual int32_t RequestKeyFrame() = 0; }; } // namespace webrtc #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_