1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H 12 #define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H 13 14 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" 15 16 #include "webrtc/voice_engine/shared_data.h" 17 18 namespace webrtc { 19 20 class VoERTP_RTCPImpl : public VoERTP_RTCP 21 { 22 public: 23 // RTCP 24 virtual int SetRTCPStatus(int channel, bool enable); 25 26 virtual int GetRTCPStatus(int channel, bool& enabled); 27 28 virtual int SetRTCP_CNAME(int channel, const char cName[256]); 29 30 virtual int GetRTCP_CNAME(int channel, char cName[256]); 31 32 virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]); 33 34 virtual int GetRemoteRTCPData(int channel, 35 unsigned int& NTPHigh, 36 unsigned int& NTPLow, 37 unsigned int& timestamp, 38 unsigned int& playoutTimestamp, 39 unsigned int* jitter = NULL, 40 unsigned short* fractionLost = NULL); 41 42 // SSRC 43 virtual int SetLocalSSRC(int channel, unsigned int ssrc); 44 45 virtual int GetLocalSSRC(int channel, unsigned int& ssrc); 46 47 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc); 48 49 // RTP Header Extension for Client-to-Mixer Audio Level Indication 50 virtual int SetSendAudioLevelIndicationStatus(int channel, 51 bool enable, 52 unsigned char id); 53 virtual int SetReceiveAudioLevelIndicationStatus(int channel, 54 bool enable, 55 unsigned char id); 56 57 // RTP Header Extension for Absolute Sender Time 58 virtual int SetSendAbsoluteSenderTimeStatus(int channel, 59 bool enable, 60 unsigned char id); 61 virtual int SetReceiveAbsoluteSenderTimeStatus(int channel, 62 bool enable, 63 unsigned char id); 64 65 // Statistics 66 virtual int GetRTPStatistics(int channel, 67 unsigned int& averageJitterMs, 68 unsigned int& maxJitterMs, 69 unsigned int& discardedPackets); 70 71 virtual int GetRTCPStatistics(int channel, CallStatistics& stats); 72 73 virtual int GetRemoteRTCPReportBlocks( 74 int channel, std::vector<ReportBlock>* report_blocks); 75 76 // RED 77 virtual int SetREDStatus(int channel, 78 bool enable, 79 int redPayloadtype = -1); 80 81 virtual int GetREDStatus(int channel, bool& enabled, int& redPayloadtype); 82 83 //NACK 84 virtual int SetNACKStatus(int channel, 85 bool enable, 86 int maxNoPackets); 87 88 // Store RTP and RTCP packets and dump to file (compatible with rtpplay) 89 virtual int StartRTPDump(int channel, 90 const char fileNameUTF8[1024], 91 RTPDirections direction = kRtpIncoming); 92 93 virtual int StopRTPDump(int channel, 94 RTPDirections direction = kRtpIncoming); 95 96 virtual int RTPDumpIsActive(int channel, 97 RTPDirections direction = kRtpIncoming); 98 99 virtual int SetVideoEngineBWETarget(int channel, ViENetwork* vie_network, 100 int video_channel); 101 protected: 102 VoERTP_RTCPImpl(voe::SharedData* shared); 103 virtual ~VoERTP_RTCPImpl(); 104 105 private: 106 voe::SharedData* _shared; 107 }; 108 109 } // namespace webrtc 110 111 #endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H 112