/* * 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_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H #define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H #include "voe_rtp_rtcp.h" #include "ref_count.h" #include "shared_data.h" namespace webrtc { class VoERTP_RTCPImpl : public virtual voe::SharedData, public VoERTP_RTCP, public voe::RefCount { public: virtual int Release(); // Registration of observers for RTP and RTCP callbacks virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer); virtual int DeRegisterRTPObserver(int channel); virtual int RegisterRTCPObserver(int channel, VoERTCPObserver& observer); virtual int DeRegisterRTCPObserver(int channel); // RTCP virtual int SetRTCPStatus(int channel, bool enable); virtual int GetRTCPStatus(int channel, bool& enabled); virtual int SetRTCP_CNAME(int channel, const char cName[256]); virtual int GetRTCP_CNAME(int channel, char cName[256]); virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]); virtual int GetRemoteRTCPData(int channel, unsigned int& NTPHigh, unsigned int& NTPLow, unsigned int& timestamp, unsigned int& playoutTimestamp, unsigned int* jitter = NULL, unsigned short* fractionLost = NULL); virtual int SendApplicationDefinedRTCPPacket( int channel, const unsigned char subType, unsigned int name, const char* data, unsigned short dataLengthInBytes); // SSRC virtual int SetLocalSSRC(int channel, unsigned int ssrc); virtual int GetLocalSSRC(int channel, unsigned int& ssrc); virtual int GetRemoteSSRC(int channel, unsigned int& ssrc); // RTP Header Extension for Client-to-Mixer Audio Level Indication virtual int SetRTPAudioLevelIndicationStatus(int channel, bool enable, unsigned char ID); virtual int GetRTPAudioLevelIndicationStatus(int channel, bool& enabled, unsigned char& ID); // CSRC virtual int GetRemoteCSRCs(int channel, unsigned int arrCSRC[15]); // Statistics virtual int GetRTPStatistics(int channel, unsigned int& averageJitterMs, unsigned int& maxJitterMs, unsigned int& discardedPackets); virtual int GetRTCPStatistics(int channel, CallStatistics& stats); // RTP keepalive mechanism (maintains NAT mappings associated to RTP flows) virtual int SetRTPKeepaliveStatus(int channel, bool enable, unsigned char unknownPayloadType, int deltaTransmitTimeSeconds = 15); virtual int GetRTPKeepaliveStatus(int channel, bool& enabled, unsigned char& unknownPayloadType, int& deltaTransmitTimeSeconds); // FEC virtual int SetFECStatus(int channel, bool enable, int redPayloadtype = -1); virtual int GetFECStatus(int channel, bool& enabled, int& redPayloadtype); // Store RTP and RTCP packets and dump to file (compatible with rtpplay) virtual int StartRTPDump(int channel, const char fileNameUTF8[1024], RTPDirections direction = kRtpIncoming); virtual int StopRTPDump(int channel, RTPDirections direction = kRtpIncoming); virtual int RTPDumpIsActive(int channel, RTPDirections direction = kRtpIncoming); // Insert (and transmits) extra RTP packet into active RTP audio stream virtual int InsertExtraRTPPacket(int channel, unsigned char payloadType, bool markerBit, const char* payloadData, unsigned short payloadSize); protected: VoERTP_RTCPImpl(); virtual ~VoERTP_RTCPImpl(); }; } // namespace webrtc #endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H