| /* |
| * 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 <vector> |
| |
| #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 |
| * rtcp_feedback - Callback object that will receive the incoming |
| * RTCP messages. |
| * 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 - Telehone 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; |
| Clock* clock; |
| RtpRtcp* default_module; |
| ReceiveStatistics* receive_statistics; |
| Transport* outgoing_transport; |
| RtcpFeedback* rtcp_feedback; |
| RtcpIntraFrameObserver* intra_frame_callback; |
| RtcpBandwidthObserver* bandwidth_callback; |
| RtcpRttStats* rtt_stats; |
| RtpAudioFeedback* audio_messages; |
| RemoteBitrateEstimator* remote_bitrate_estimator; |
| PacedSender* paced_sender; |
| }; |
| |
| /* |
| * 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, |
| uint16_t incoming_packet_length) = 0; |
| |
| virtual void SetRemoteSSRC(const 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(const 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( |
| const bool TCP, |
| const bool IPV6, |
| const 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( |
| const int8_t payloadType) = 0; |
| |
| /* |
| * (De)register RTP header extension type and id. |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t RegisterSendRtpHeaderExtension( |
| const RTPExtensionType type, |
| const uint8_t id) = 0; |
| |
| virtual int32_t DeregisterSendRtpHeaderExtension( |
| const RTPExtensionType type) = 0; |
| |
| /* |
| * get start timestamp |
| */ |
| virtual uint32_t StartTimestamp() const = 0; |
| |
| /* |
| * configure start timestamp, default is a random number |
| * |
| * timestamp - start timestamp |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetStartTimestamp( |
| const uint32_t timestamp) = 0; |
| |
| /* |
| * Get SequenceNumber |
| */ |
| virtual uint16_t SequenceNumber() const = 0; |
| |
| /* |
| * Set SequenceNumber, default is a random number |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetSequenceNumber(const uint16_t seq) = 0; |
| |
| virtual void 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 |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual void SetSSRC(const uint32_t ssrc) = 0; |
| |
| /* |
| * Get CSRC |
| * |
| * arrOfCSRC - array of CSRCs |
| * |
| * return -1 on failure else number of valid entries in the array |
| */ |
| virtual int32_t CSRCs( |
| uint32_t arrOfCSRC[kRtpCsrcSize]) const = 0; |
| |
| /* |
| * Set CSRC |
| * |
| * arrOfCSRC - array of CSRCs |
| * arrLength - number of valid entries in the array |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetCSRCs( |
| const uint32_t arrOfCSRC[kRtpCsrcSize], |
| const uint8_t arrLength) = 0; |
| |
| /* |
| * includes CSRCs in RTP header if enabled |
| * |
| * include CSRC - on/off |
| * |
| * default:on |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetCSRCStatus(const bool include) = 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; |
| |
| // 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) = 0; |
| |
| /* |
| * Get status of sending RTX (RFC 4588) on a specific SSRC. |
| */ |
| virtual void RTXSendStatus(int* modes, uint32_t* ssrc, |
| int* payloadType) const = 0; |
| |
| /* |
| * sends kRtcpByeCode when going from true to false |
| * |
| * sending - on/off |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetSendingStatus(const bool sending) = 0; |
| |
| /* |
| * get send status |
| */ |
| virtual bool Sending() const = 0; |
| |
| /* |
| * Starts/Stops media packets, on by default |
| * |
| * sending - on/off |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetSendingMediaStatus(const 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; |
| |
| /* |
| * Called on any new send bitrate estimate. |
| */ |
| virtual void RegisterVideoBitrateObserver( |
| BitrateStatisticsObserver* observer) = 0; |
| virtual BitrateStatisticsObserver* GetVideoBitrateObserver() 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( |
| const FrameType frameType, |
| const int8_t payloadType, |
| const uint32_t timeStamp, |
| int64_t capture_time_ms, |
| const uint8_t* payloadData, |
| const uint32_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 int TimeToSendPadding(int bytes) = 0; |
| |
| virtual void RegisterSendFrameCountObserver( |
| FrameCountObserver* observer) = 0; |
| virtual FrameCountObserver* GetSendFrameCountObserver() const = 0; |
| |
| virtual bool GetSendSideDelay(int* avg_send_delay_ms, |
| int* max_send_delay_ms) const = 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 |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetRTCPStatus(const RTCPMethod method) = 0; |
| |
| /* |
| * Set RTCP CName (i.e unique identifier) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * Get RTCP CName (i.e unique identifier) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t CNAME(char cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * Get remote CName |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t RemoteCNAME( |
| const 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( |
| const uint32_t SSRC, |
| const char cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * RemoveMixedCNAME |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t RemoveMixedCNAME(const uint32_t SSRC) = 0; |
| |
| /* |
| * Get RoundTripTime |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t RTT(const uint32_t remoteSSRC, |
| uint16_t* RTT, |
| uint16_t* avgRTT, |
| uint16_t* minRTT, |
| uint16_t* maxRTT) const = 0 ; |
| |
| /* |
| * Reset RoundTripTime statistics |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t ResetRTT(const uint32_t remoteSSRC)= 0 ; |
| |
| /* |
| * Force a send of a RTCP packet |
| * normal SR and RR are triggered via the process function |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SendRTCP( |
| uint32_t rtcpPacketType = kRtcpReport) = 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( |
| const uint8_t pictureID) = 0; |
| |
| /* |
| * Reset RTP data counters for the sending side |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t ResetSendDataCountersRTP() = 0; |
| |
| /* |
| * statistics of the amount of data sent and received |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t DataCountersRTP( |
| uint32_t* bytesSent, |
| uint32_t* packetsSent) 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<RTCPReportBlock>* receiveBlocks) const = 0; |
| /* |
| * Set received RTCP report block |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t AddRTCPReportBlock( |
| const uint32_t SSRC, |
| const RTCPReportBlock* receiveBlock) = 0; |
| |
| /* |
| * RemoveRTCPReportBlock |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t RemoveRTCPReportBlock(const uint32_t SSRC) = 0; |
| |
| /* |
| * Get number of sent and received RTCP packet types. |
| */ |
| virtual void GetRtcpPacketTypeCounters( |
| RtcpPacketTypeCounter* packets_sent, |
| RtcpPacketTypeCounter* packets_received) const = 0; |
| |
| /* |
| * (APP) Application specific data |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetRTCPApplicationSpecificData( |
| const uint8_t subType, |
| const uint32_t name, |
| const uint8_t* data, |
| const 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 int32_t SetREMBStatus(const bool enable) = 0; |
| |
| virtual int32_t SetREMBData(const uint32_t bitrate, |
| const uint8_t numberOfSSRC, |
| const uint32_t* SSRC) = 0; |
| |
| /* |
| * (IJ) Extended jitter report. |
| */ |
| virtual bool IJ() const = 0; |
| |
| virtual int32_t SetIJStatus(const bool enable) = 0; |
| |
| /* |
| * (TMMBR) Temporary Max Media Bit Rate |
| */ |
| virtual bool TMMBR() const = 0; |
| |
| /* |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetTMMBRStatus(const 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, |
| const uint16_t size) = 0; |
| |
| /* |
| * Store the sent packets, needed to answer to a Negative acknowledgement |
| * requests |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetStorePacketsStatus( |
| const bool enable, |
| const 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 RegisterSendChannelRtcpStatisticsCallback( |
| RtcpStatisticsCallback* callback) = 0; |
| virtual RtcpStatisticsCallback* |
| GetSendChannelRtcpStatisticsCallback() = 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( |
| const uint16_t packetSizeSamples) = 0; |
| |
| /* |
| * SendTelephoneEventActive |
| * |
| * return true if we currently send a telephone event and 100 ms after an |
| * event is sent used to prevent the telephone event tone to be recorded |
| * by the microphone and send inband just after the tone has ended. |
| */ |
| virtual bool SendTelephoneEventActive( |
| int8_t& telephoneEvent) const = 0; |
| |
| /* |
| * Send a TelephoneEvent tone using RFC 2833 (4733) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SendTelephoneEventOutband( |
| const uint8_t key, |
| const uint16_t time_ms, |
| const uint8_t level) = 0; |
| |
| /* |
| * Set payload type for Redundant Audio Data RFC 2198 |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetSendREDPayloadType( |
| const 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(const uint8_t level_dBov) = 0; |
| |
| /************************************************************************** |
| * |
| * Video |
| * |
| ***************************************************************************/ |
| |
| /* |
| * Set the estimated camera delay in MS |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetCameraDelay(const int32_t delayMS) = 0; |
| |
| /* |
| * Set the target send bitrate |
| */ |
| virtual void SetTargetSendBitrate( |
| const std::vector<uint32_t>& stream_bitrates) = 0; |
| |
| /* |
| * Turn on/off generic FEC |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SetGenericFECStatus( |
| const bool enable, |
| const uint8_t payloadTypeRED, |
| const 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( |
| const 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_ |
