| /* |
| * 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 "modules/interface/module.h" |
| #include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| |
| namespace webrtc { |
| // forward declaration |
| class RemoteBitrateEstimator; |
| class RemoteBitrateObserver; |
| class Transport; |
| |
| class RtpRtcp : public Module { |
| public: |
| struct Configuration { |
| Configuration() |
| : id(-1), |
| audio(false), |
| clock(NULL), |
| default_module(NULL), |
| incoming_data(NULL), |
| incoming_messages(NULL), |
| outgoing_transport(NULL), |
| rtcp_feedback(NULL), |
| intra_frame_callback(NULL), |
| bandwidth_callback(NULL), |
| audio_messages(NULL), |
| remote_bitrate_estimator(NULL) { |
| } |
| /* 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 |
| * incoming_messages - Callback object that will receive the incoming |
| * RTP messages. |
| * 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 |
| * RTP 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. |
| * remote_bitrate_estimator - Estimates the bandwidth available for a set of |
| * streams from the same client. |
| */ |
| int32_t id; |
| bool audio; |
| RtpRtcpClock* clock; |
| RtpRtcp* default_module; |
| RtpData* incoming_data; |
| RtpFeedback* incoming_messages; |
| Transport* outgoing_transport; |
| RtcpFeedback* rtcp_feedback; |
| RtcpIntraFrameObserver* intra_frame_callback; |
| RtcpBandwidthObserver* bandwidth_callback; |
| RtpAudioFeedback* audio_messages; |
| RemoteBitrateEstimator* remote_bitrate_estimator; |
| }; |
| /* |
| * 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 |
| * |
| ***************************************************************************/ |
| |
| /* |
| * configure a RTP packet timeout value |
| * |
| * RTPtimeoutMS - time in milliseconds after last received RTP packet |
| * RTCPtimeoutMS - time in milliseconds after last received RTCP packet |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetPacketTimeout( |
| const WebRtc_UWord32 RTPtimeoutMS, |
| const WebRtc_UWord32 RTCPtimeoutMS) = 0; |
| |
| /* |
| * Set periodic dead or alive notification |
| * |
| * enable - turn periodic dead or alive notification on/off |
| * sampleTimeSeconds - sample interval in seconds for dead or alive |
| * notifications |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus( |
| const bool enable, |
| const WebRtc_UWord8 sampleTimeSeconds) = 0; |
| |
| /* |
| * Get periodic dead or alive notification status |
| * |
| * enable - periodic dead or alive notification on/off |
| * sampleTimeSeconds - sample interval in seconds for dead or alive |
| * notifications |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 PeriodicDeadOrAliveStatus( |
| bool& enable, |
| WebRtc_UWord8& sampleTimeSeconds) = 0; |
| |
| /* |
| * set voice codec name and payload type |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterReceivePayload( |
| const CodecInst& voiceCodec) = 0; |
| |
| /* |
| * set video codec name and payload type |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterReceivePayload( |
| const VideoCodec& videoCodec) = 0; |
| |
| /* |
| * get payload type for a voice codec |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ReceivePayloadType( |
| const CodecInst& voiceCodec, |
| WebRtc_Word8* plType) = 0; |
| |
| /* |
| * get payload type for a video codec |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ReceivePayloadType( |
| const VideoCodec& videoCodec, |
| WebRtc_Word8* plType) = 0; |
| |
| /* |
| * Remove a registered payload type from list of accepted payloads |
| * |
| * payloadType - payload type of codec |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 DeRegisterReceivePayload( |
| const WebRtc_Word8 payloadType) = 0; |
| |
| /* |
| * (De)register RTP header extension type and id. |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension( |
| const RTPExtensionType type, |
| const WebRtc_UWord8 id) = 0; |
| |
| virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension( |
| const RTPExtensionType type) = 0; |
| |
| /* |
| * Get last received remote timestamp |
| */ |
| virtual WebRtc_UWord32 RemoteTimestamp() const = 0; |
| |
| /* |
| * Get the local time of the last received remote timestamp |
| */ |
| virtual int64_t LocalTimeOfRemoteTimeStamp() const = 0; |
| |
| /* |
| * Get the current estimated remote timestamp |
| * |
| * timestamp - estimated timestamp |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 EstimatedRemoteTimeStamp( |
| WebRtc_UWord32& timestamp) const = 0; |
| |
| /* |
| * Get incoming SSRC |
| */ |
| virtual WebRtc_UWord32 RemoteSSRC() const = 0; |
| |
| /* |
| * Get remote CSRC |
| * |
| * arrOfCSRC - array that will receive the CSRCs |
| * |
| * return -1 on failure else the number of valid entries in the list |
| */ |
| virtual WebRtc_Word32 RemoteCSRCs( |
| WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0; |
| |
| /* |
| * get the currently configured SSRC filter |
| * |
| * allowedSSRC - SSRC that will be allowed through |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const = 0; |
| |
| /* |
| * set a SSRC to be used as a filter for incoming RTP streams |
| * |
| * allowedSSRC - SSRC that will be allowed through |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetSSRCFilter(const bool enable, |
| const WebRtc_UWord32 allowedSSRC) = 0; |
| |
| /* |
| * Turn on/off receiving RTX (RFC 4588) on a specific SSRC. |
| */ |
| virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable, |
| const WebRtc_UWord32 SSRC) = 0; |
| |
| /* |
| * Get status of receiving RTX (RFC 4588) on a specific SSRC. |
| */ |
| virtual WebRtc_Word32 RTXReceiveStatus(bool* enable, |
| WebRtc_UWord32* SSRC) const = 0; |
| |
| /* |
| * called by the network module when we receive a packet |
| * |
| * incomingPacket - incoming packet buffer |
| * packetLength - length of incoming buffer |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPacket, |
| const WebRtc_UWord16 packetLength) = 0; |
| |
| /************************************************************************** |
| * |
| * Sender |
| * |
| ***************************************************************************/ |
| |
| /* |
| * set MTU |
| * |
| * size - Max transfer unit in bytes, default is 1500 |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 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 WebRtc_Word32 SetTransportOverhead( |
| const bool TCP, |
| const bool IPV6, |
| const WebRtc_UWord8 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 WebRtc_UWord16 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 WebRtc_UWord16 MaxDataPayloadLength() const = 0; |
| |
| /* |
| * set codec name and payload type |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterSendPayload( |
| const CodecInst& voiceCodec) = 0; |
| |
| /* |
| * set codec name and payload type |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterSendPayload( |
| const VideoCodec& videoCodec) = 0; |
| |
| /* |
| * Unregister a send payload |
| * |
| * payloadType - payload type of codec |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 DeRegisterSendPayload( |
| const WebRtc_Word8 payloadType) = 0; |
| |
| /* |
| * (De)register RTP header extension type and id. |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterSendRtpHeaderExtension( |
| const RTPExtensionType type, |
| const WebRtc_UWord8 id) = 0; |
| |
| virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension( |
| const RTPExtensionType type) = 0; |
| |
| /* |
| * Enable/disable traffic smoothing of sending stream. |
| */ |
| virtual void SetTransmissionSmoothingStatus(const bool enable) = 0; |
| |
| virtual bool TransmissionSmoothingStatus() const = 0; |
| |
| /* |
| * get start timestamp |
| */ |
| virtual WebRtc_UWord32 StartTimestamp() const = 0; |
| |
| /* |
| * configure start timestamp, default is a random number |
| * |
| * timestamp - start timestamp |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetStartTimestamp( |
| const WebRtc_UWord32 timestamp) = 0; |
| |
| /* |
| * Get SequenceNumber |
| */ |
| virtual WebRtc_UWord16 SequenceNumber() const = 0; |
| |
| /* |
| * Set SequenceNumber, default is a random number |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq) = 0; |
| |
| /* |
| * Get SSRC |
| */ |
| virtual WebRtc_UWord32 SSRC() const = 0; |
| |
| /* |
| * configure SSRC, default is a random number |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc) = 0; |
| |
| /* |
| * Get CSRC |
| * |
| * arrOfCSRC - array of CSRCs |
| * |
| * return -1 on failure else number of valid entries in the array |
| */ |
| virtual WebRtc_Word32 CSRCs( |
| WebRtc_UWord32 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 WebRtc_Word32 SetCSRCs( |
| const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], |
| const WebRtc_UWord8 arrLength) = 0; |
| |
| /* |
| * includes CSRCs in RTP header if enabled |
| * |
| * include CSRC - on/off |
| * |
| * default:on |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetCSRCStatus(const bool include) = 0; |
| |
| /* |
| * Turn on/off sending RTX (RFC 4588) on a specific SSRC. |
| */ |
| virtual WebRtc_Word32 SetRTXSendStatus(const bool enable, |
| const bool setSSRC, |
| const WebRtc_UWord32 SSRC) = 0; |
| |
| /* |
| * Get status of sending RTX (RFC 4588) on a specific SSRC. |
| */ |
| virtual WebRtc_Word32 RTXSendStatus(bool* enable, |
| WebRtc_UWord32* SSRC) const = 0; |
| |
| /* |
| * sends kRtcpByeCode when going from true to false |
| * |
| * sending - on/off |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 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 WebRtc_Word32 SetSendingMediaStatus(const bool sending) = 0; |
| |
| /* |
| * get send status |
| */ |
| virtual bool SendingMedia() const = 0; |
| |
| /* |
| * get sent bitrate in Kbit/s |
| */ |
| virtual void BitrateSent(WebRtc_UWord32* totalRate, |
| WebRtc_UWord32* videoRate, |
| WebRtc_UWord32* fecRate, |
| WebRtc_UWord32* nackRate) const = 0; |
| |
| /* |
| * Get the receive-side estimate of the available bandwidth. |
| */ |
| virtual int EstimatedReceiveBandwidth( |
| WebRtc_UWord32* available_bandwidth) 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 WebRtc_Word32 SendOutgoingData( |
| const FrameType frameType, |
| const WebRtc_Word8 payloadType, |
| const WebRtc_UWord32 timeStamp, |
| int64_t capture_time_ms, |
| const WebRtc_UWord8* payloadData, |
| const WebRtc_UWord32 payloadSize, |
| const RTPFragmentationHeader* fragmentation = NULL, |
| const RTPVideoHeader* rtpVideoHdr = NULL) = 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 WebRtc_Word32 SetRTCPStatus(const RTCPMethod method) = 0; |
| |
| /* |
| * Set RTCP CName (i.e unique identifier) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * Get RTCP CName (i.e unique identifier) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * Get remote CName |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemoteCNAME( |
| const WebRtc_UWord32 remoteSSRC, |
| char cName[RTCP_CNAME_SIZE]) const = 0; |
| |
| /* |
| * Get remote NTP |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemoteNTP( |
| WebRtc_UWord32 *ReceivedNTPsecs, |
| WebRtc_UWord32 *ReceivedNTPfrac, |
| WebRtc_UWord32 *RTCPArrivalTimeSecs, |
| WebRtc_UWord32 *RTCPArrivalTimeFrac, |
| WebRtc_UWord32 *rtcp_timestamp) const = 0; |
| |
| /* |
| * AddMixedCNAME |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 AddMixedCNAME( |
| const WebRtc_UWord32 SSRC, |
| const char cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * RemoveMixedCNAME |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC) = 0; |
| |
| /* |
| * Get RoundTripTime |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC, |
| WebRtc_UWord16* RTT, |
| WebRtc_UWord16* avgRTT, |
| WebRtc_UWord16* minRTT, |
| WebRtc_UWord16* maxRTT) const = 0 ; |
| |
| /* |
| * Reset RoundTripTime statistics |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 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 WebRtc_Word32 SendRTCP( |
| WebRtc_UWord32 rtcpPacketType = kRtcpReport) = 0; |
| |
| /* |
| * Good state of RTP receiver inform sender |
| */ |
| virtual WebRtc_Word32 SendRTCPReferencePictureSelection( |
| const WebRtc_UWord64 pictureID) = 0; |
| |
| /* |
| * Send a RTCP Slice Loss Indication (SLI) |
| * 6 least significant bits of pictureID |
| */ |
| virtual WebRtc_Word32 SendRTCPSliceLossIndication( |
| const WebRtc_UWord8 pictureID) = 0; |
| |
| /* |
| * Reset RTP statistics |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ResetStatisticsRTP() = 0; |
| |
| /* |
| * statistics of our localy created statistics of the received RTP stream |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 StatisticsRTP( |
| WebRtc_UWord8* fraction_lost, // scale 0 to 255 |
| WebRtc_UWord32* cum_lost, // number of lost packets |
| WebRtc_UWord32* ext_max, // highest sequence number received |
| WebRtc_UWord32* jitter, |
| WebRtc_UWord32* max_jitter = NULL) const = 0; |
| |
| /* |
| * Reset RTP data counters for the receiving side |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ResetReceiveDataCountersRTP() = 0; |
| |
| /* |
| * Reset RTP data counters for the sending side |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ResetSendDataCountersRTP() = 0; |
| |
| /* |
| * statistics of the amount of data sent and received |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 DataCountersRTP( |
| WebRtc_UWord32* bytesSent, |
| WebRtc_UWord32* packetsSent, |
| WebRtc_UWord32* bytesReceived, |
| WebRtc_UWord32* packetsReceived) const = 0; |
| /* |
| * Get received RTCP sender info |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0; |
| |
| /* |
| * Get received RTCP report block |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemoteRTCPStat( |
| std::vector<RTCPReportBlock>* receiveBlocks) const = 0; |
| /* |
| * Set received RTCP report block |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 AddRTCPReportBlock( |
| const WebRtc_UWord32 SSRC, |
| const RTCPReportBlock* receiveBlock) = 0; |
| |
| /* |
| * RemoveRTCPReportBlock |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC) = 0; |
| |
| /* |
| * (APP) Application specific data |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetRTCPApplicationSpecificData( |
| const WebRtc_UWord8 subType, |
| const WebRtc_UWord32 name, |
| const WebRtc_UWord8* data, |
| const WebRtc_UWord16 length) = 0; |
| /* |
| * (XR) VOIP metric |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetRTCPVoIPMetrics( |
| const RTCPVoIPMetric* VoIPMetric) = 0; |
| |
| /* |
| * (REMB) Receiver Estimated Max Bitrate |
| */ |
| virtual bool REMB() const = 0; |
| |
| virtual WebRtc_Word32 SetREMBStatus(const bool enable) = 0; |
| |
| virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate, |
| const WebRtc_UWord8 numberOfSSRC, |
| const WebRtc_UWord32* SSRC) = 0; |
| |
| /* |
| * (IJ) Extended jitter report. |
| */ |
| virtual bool IJ() const = 0; |
| |
| virtual WebRtc_Word32 SetIJStatus(const bool enable) = 0; |
| |
| /* |
| * (TMMBR) Temporary Max Media Bit Rate |
| */ |
| virtual bool TMMBR() const = 0; |
| |
| /* |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetTMMBRStatus(const bool enable) = 0; |
| |
| /* |
| * (NACK) |
| */ |
| virtual NACKMethod NACK() const = 0; |
| |
| /* |
| * Turn negative acknowledgement requests on/off |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method) = 0; |
| |
| /* |
| * 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 WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList, |
| const WebRtc_UWord16 size) = 0; |
| |
| /* |
| * Store the sent packets, needed to answer to a Negative acknowledgement |
| * requests |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetStorePacketsStatus( |
| const bool enable, |
| const WebRtc_UWord16 numberToStore = 200) = 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 WebRtc_Word32 SetAudioPacketSize( |
| const WebRtc_UWord16 packetSizeSamples) = 0; |
| |
| /* |
| * Outband TelephoneEvent(DTMF) detection |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetTelephoneEventStatus( |
| const bool enable, |
| const bool forwardToDecoder, |
| const bool detectEndOfTone = false) = 0; |
| |
| /* |
| * Is outband TelephoneEvent(DTMF) turned on/off? |
| */ |
| virtual bool TelephoneEvent() const = 0; |
| |
| /* |
| * Returns true if received DTMF events are forwarded to the decoder using |
| * the OnPlayTelephoneEvent callback. |
| */ |
| virtual bool TelephoneEventForwardToDecoder() const = 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( |
| WebRtc_Word8& telephoneEvent) const = 0; |
| |
| /* |
| * Send a TelephoneEvent tone using RFC 2833 (4733) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SendTelephoneEventOutband( |
| const WebRtc_UWord8 key, |
| const WebRtc_UWord16 time_ms, |
| const WebRtc_UWord8 level) = 0; |
| |
| /* |
| * Set payload type for Redundant Audio Data RFC 2198 |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetSendREDPayloadType( |
| const WebRtc_Word8 payloadType) = 0; |
| |
| /* |
| * Get payload type for Redundant Audio Data RFC 2198 |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SendREDPayloadType( |
| WebRtc_Word8& payloadType) const = 0; |
| |
| /* |
| * Set status and ID for header-extension-for-audio-level-indication. |
| * See http://tools.ietf.org/html/rfc6464 for more details. |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus( |
| const bool enable, |
| const WebRtc_UWord8 ID) = 0; |
| |
| /* |
| * Get status and ID for header-extension-for-audio-level-indication. |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus( |
| bool& enable, |
| WebRtc_UWord8& ID) 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 WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov) = 0; |
| |
| /************************************************************************** |
| * |
| * Video |
| * |
| ***************************************************************************/ |
| |
| /* |
| * Set the estimated camera delay in MS |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS) = 0; |
| |
| /* |
| * Set the target send bitrate |
| */ |
| virtual void SetTargetSendBitrate(const WebRtc_UWord32 bitrate) = 0; |
| |
| /* |
| * Turn on/off generic FEC |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetGenericFECStatus( |
| const bool enable, |
| const WebRtc_UWord8 payloadTypeRED, |
| const WebRtc_UWord8 payloadTypeFEC) = 0; |
| |
| /* |
| * Get generic FEC setting |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 GenericFECStatus(bool& enable, |
| WebRtc_UWord8& payloadTypeRED, |
| WebRtc_UWord8& payloadTypeFEC) = 0; |
| |
| |
| virtual WebRtc_Word32 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 WebRtc_Word32 SetKeyFrameRequestMethod( |
| const KeyFrameRequestMethod method) = 0; |
| |
| /* |
| * send a request for a keyframe |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RequestKeyFrame() = 0; |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ |
