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      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