| /* |
| * Copyright (c) 2011 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 "module.h" |
| #include "rtp_rtcp_defines.h" |
| |
| namespace webrtc { |
| // forward declaration |
| class Transport; |
| |
| class RtpRtcp : public Module |
| { |
| public: |
| /* |
| * create a RTP/RTCP module object |
| * |
| * 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 |
| */ |
| static RtpRtcp* CreateRtpRtcp(const WebRtc_Word32 id, |
| const bool audio); |
| |
| /* |
| * destroy a RTP/RTCP module object |
| * |
| * module - object to destroy |
| */ |
| static void DestroyRtpRtcp(RtpRtcp* module); |
| |
| /* |
| * Returns version of the module and its components |
| * |
| * version - buffer to which the version will be written |
| * remainingBufferInBytes - remaining number of WebRtc_Word8 in the version buffer |
| * position - position of the next empty WebRtc_Word8 in the version buffer |
| */ |
| static WebRtc_Word32 GetVersion(WebRtc_Word8* version, |
| WebRtc_UWord32& remainingBufferInBytes, |
| WebRtc_UWord32& position); |
| |
| /* |
| * Change the unique identifier of this object |
| * |
| * id - new unique identifier of this RTP/RTCP module object |
| */ |
| virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0; |
| |
| /* |
| * De-muxing functionality for conferencing |
| * |
| * register a module that will act as a default module for this module |
| * used for feedback messages back to the encoder when one encoded stream |
| * is sent to multiple destinations |
| * |
| * module - default module |
| */ |
| virtual WebRtc_Word32 RegisterDefaultModule(RtpRtcp* module) = 0; |
| |
| /* |
| * unregister the default module |
| * will stop the demuxing feedback |
| */ |
| virtual WebRtc_Word32 DeRegisterDefaultModule() = 0; |
| |
| /* |
| * returns true if a default module is registered, false otherwise |
| */ |
| virtual bool DefaultModuleRegistered() = 0; |
| |
| /* |
| * returns number of registered child modules |
| */ |
| virtual WebRtc_UWord32 NumberChildModules() = 0; |
| |
| /* |
| * Lip-sync between voice-video |
| * |
| * module - audio module |
| * |
| * Note: only allowed on a video module |
| */ |
| virtual WebRtc_Word32 RegisterSyncModule(RtpRtcp* module) = 0; |
| |
| /* |
| * Turn off lip-sync between voice-video |
| */ |
| virtual WebRtc_Word32 DeRegisterSyncModule() = 0; |
| |
| /************************************************************************** |
| * |
| * Receiver functions |
| * |
| ***************************************************************************/ |
| |
| /* |
| * Initialize receive side |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 InitReceiver() = 0; |
| |
| /* |
| * Used by the module to deliver the incoming data to the codec module |
| * |
| * incomingDataCallback - callback object that will receive the incoming data |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback) = 0; |
| |
| /* |
| * Used by the module to deliver messages to the codec module/appliation |
| * |
| * incomingMessagesCallback - callback object that will receive the incoming messages |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback) = 0; |
| |
| /* |
| * 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 codec name and payload type |
| * |
| * payloadName - payload name of codec |
| * payloadType - payload type of codec |
| * frequency - (audio specific) frequency of codec |
| * channels - (audio specific) number of channels in codec (1 = mono, 2 = stereo) |
| * rate - (audio) rate of codec |
| * (video) maxBitrate of codec, bits/sec |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterReceivePayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE], |
| const WebRtc_Word8 payloadType, |
| const WebRtc_UWord32 frequency = 0, |
| const WebRtc_UWord8 channels = 1, |
| const WebRtc_UWord32 rate = 0) = 0; |
| |
| /* |
| * Remove a registerd 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; |
| |
| /* |
| * get configured payload type |
| * |
| * payloadName - payload name of codec |
| * frequency - frequency of codec, ignored for video |
| * payloadType - payload type of codec, ignored for video |
| * channels - number of channels in codec (1 = mono, 2 = stereo) |
| * rate - (audio) rate of codec (ignored if set to 0) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ReceivePayloadType(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE], |
| const WebRtc_UWord32 frequency, |
| const WebRtc_UWord8 channels, |
| WebRtc_Word8* payloadType, |
| const WebRtc_UWord32 rate = 0) const = 0; |
| |
| /* |
| * get configured payload |
| * |
| * payloadType - payload type of codec |
| * payloadName - payload name of codec |
| * frequency - frequency of codec |
| * channels - number of channels in codec (1 = mono, 2 = stereo) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 ReceivePayload(const WebRtc_Word8 payloadType, |
| WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE], |
| WebRtc_UWord32* frequency, |
| WebRtc_UWord8* channels, |
| WebRtc_UWord32* rate = NULL) const = 0; |
| |
| /* |
| * Get last received remote timestamp |
| */ |
| virtual WebRtc_UWord32 RemoteTimestamp() 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 Current incoming payload |
| * |
| * payloadName - payload name of codec |
| * payloadType - payload type of codec |
| * frequency - frequency of codec |
| * channels - number of channels in codec (2 = stereo) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemotePayload(WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE], |
| WebRtc_Word8* payloadType, |
| WebRtc_UWord32* frequency, |
| WebRtc_UWord8* channels) 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; |
| |
| /* |
| * 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; |
| |
| |
| /* |
| * Option when not using the RegisterSyncModule function |
| * |
| * Inform the module about the received audion NTP |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 IncomingAudioNTP(const WebRtc_UWord32 audioReceivedNTPsecs, |
| const WebRtc_UWord32 audioReceivedNTPfrac, |
| const WebRtc_UWord32 audioRTCPArrivalTimeSecs, |
| const WebRtc_UWord32 audioRTCPArrivalTimeFrac) = 0; |
| |
| |
| /************************************************************************** |
| * |
| * Sender |
| * |
| ***************************************************************************/ |
| |
| /* |
| * Initialize send side |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 InitSender() = 0; |
| |
| /* |
| * Used by the module to send RTP and RTCP packet to the network module |
| * |
| * outgoingTransport - transport object that will be called when packets are ready to be sent out on the network |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport) = 0; |
| |
| /* |
| * 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 RTPKeepaliveStatus |
| * |
| * enable - on/off |
| * unknownPayloadType - payload type to use for RTP keepalive |
| * deltaTransmitTimeMS - delta time between RTP keepalive packets |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetRTPKeepaliveStatus(const bool enable, |
| const WebRtc_Word8 unknownPayloadType, |
| const WebRtc_UWord16 deltaTransmitTimeMS) = 0; |
| |
| /* |
| * Get RTPKeepaliveStatus |
| * |
| * enable - on/off |
| * unknownPayloadType - payload type in use for RTP keepalive |
| * deltaTransmitTimeMS - delta time between RTP keepalive packets |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RTPKeepaliveStatus(bool* enable, |
| WebRtc_Word8* unknownPayloadType, |
| WebRtc_UWord16* deltaTransmitTimeMS) const = 0; |
| |
| /* |
| * check if RTPKeepaliveStatus is enabled |
| */ |
| virtual bool RTPKeepalive() const = 0; |
| |
| /* |
| * set codec name and payload type |
| * |
| * payloadName - payload name of codec |
| * payloadType - payload type of codec |
| * frequency - frequency of codec |
| * channels - number of channels in codec (1 = mono, 2 = stereo) |
| * rate - (audio) rate of codec |
| * (video) maxBitrate of codec, bits/sec |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterSendPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE], |
| const WebRtc_Word8 payloadType, |
| const WebRtc_UWord32 frequency = 0, |
| const WebRtc_UWord8 channels = 1, |
| const WebRtc_UWord32 rate = 0) = 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; |
| |
| /* |
| * 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; |
| |
| /* |
| * 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 WebRtc_UWord32 BitrateSent() 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, |
| const WebRtc_UWord8* payloadData, |
| const WebRtc_UWord32 payloadSize, |
| const RTPFragmentationHeader* fragmentation = NULL, |
| const RTPVideoTypeHeader* rtpTypeHdr = NULL) = 0; |
| |
| /************************************************************************** |
| * |
| * RTCP |
| * |
| ***************************************************************************/ |
| |
| /* |
| * RegisterIncomingRTCPCallback |
| * |
| * incomingMessagesCallback - callback object that will receive messages from RTCP |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback) = 0; |
| |
| /* |
| * 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 WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * Get RTCP CName (i.e unique identifier) |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0; |
| |
| /* |
| * Get remote CName |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC, |
| WebRtc_Word8 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) const = 0; |
| |
| /* |
| * AddMixedCNAME |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC, |
| const WebRtc_Word8 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( const WebRtc_UWord32 remoteSSRC, |
| RTCPReportBlock* receiveBlock) = 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; |
| |
| /* |
| * (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; |
| |
| /* |
| * local bw estimation changed |
| * |
| * for video called by internal estimator |
| * for audio (iSAC) called by engine, geting the data from the decoder |
| */ |
| virtual void OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit) = 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; |
| |
| /* |
| * 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 |
| * |
| ***************************************************************************/ |
| |
| /* |
| * RegisterAudioCallback |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback) = 0; |
| |
| /* |
| * 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 teh 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 https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/ |
| * 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 |
| * |
| ***************************************************************************/ |
| |
| /* |
| * Register a callback object that will receive callbacks for video related events |
| * such as an incoming key frame request. |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback) = 0; |
| |
| /* |
| * Set the estimated camera delay in MS |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS) = 0; |
| |
| /* |
| * Set the start and max send bitrate |
| * used by the bandwidth management |
| * |
| * Not calling this or setting startBitrateKbit to 0 disables the bandwidth management |
| * |
| * minBitrateKbit = 0 equals no min bitrate |
| * maxBitrateKbit = 0 equals no max bitrate |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetSendBitrate(const WebRtc_UWord32 startBitrate, |
| const WebRtc_UWord16 minBitrateKbit, |
| const WebRtc_UWord16 maxBitrateKbit) = 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; |
| |
| /* |
| * Set FEC code rate of key and delta frames |
| * codeRate on a scale of 0 to 255 where 255 is 100% added packets, hence protect up to 50% packet loss |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate, |
| const WebRtc_UWord8 deltaFrameCodeRate) = 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(const FrameType frameType = kVideoFrameKey) = 0; |
| |
| /* |
| * Only for H.263 to interop with bad endpoints |
| */ |
| virtual WebRtc_Word32 SetH263InverseLogic(const bool enable) = 0; |
| }; |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ |
