4ef438e2de
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
110 lines
4.0 KiB
C++
110 lines
4.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
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#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
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#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
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#include "webrtc/voice_engine/shared_data.h"
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namespace webrtc {
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class VoERTP_RTCPImpl : public VoERTP_RTCP
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{
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public:
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// RTCP
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virtual int SetRTCPStatus(int channel, bool enable);
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virtual int GetRTCPStatus(int channel, bool& enabled);
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virtual int SetRTCP_CNAME(int channel, const char cName[256]);
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virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]);
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virtual int GetRemoteRTCPData(int channel,
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unsigned int& NTPHigh,
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unsigned int& NTPLow,
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unsigned int& timestamp,
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unsigned int& playoutTimestamp,
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unsigned int* jitter = NULL,
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unsigned short* fractionLost = NULL);
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// SSRC
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virtual int SetLocalSSRC(int channel, unsigned int ssrc);
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virtual int GetLocalSSRC(int channel, unsigned int& ssrc);
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virtual int GetRemoteSSRC(int channel, unsigned int& ssrc);
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// RTP Header Extension for Client-to-Mixer Audio Level Indication
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virtual int SetSendAudioLevelIndicationStatus(int channel,
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bool enable,
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unsigned char id);
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virtual int SetReceiveAudioLevelIndicationStatus(int channel,
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bool enable,
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unsigned char id);
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// RTP Header Extension for Absolute Sender Time
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virtual int SetSendAbsoluteSenderTimeStatus(int channel,
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bool enable,
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unsigned char id);
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virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
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bool enable,
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unsigned char id);
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// Statistics
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virtual int GetRTPStatistics(int channel,
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unsigned int& averageJitterMs,
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unsigned int& maxJitterMs,
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unsigned int& discardedPackets);
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virtual int GetRTCPStatistics(int channel, CallStatistics& stats);
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virtual int GetRemoteRTCPReportBlocks(
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int channel, std::vector<ReportBlock>* report_blocks);
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// RED
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virtual int SetREDStatus(int channel,
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bool enable,
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int redPayloadtype = -1);
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virtual int GetREDStatus(int channel, bool& enabled, int& redPayloadtype);
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//NACK
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virtual int SetNACKStatus(int channel,
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bool enable,
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int maxNoPackets);
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// Store RTP and RTCP packets and dump to file (compatible with rtpplay)
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virtual int StartRTPDump(int channel,
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const char fileNameUTF8[1024],
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RTPDirections direction = kRtpIncoming);
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virtual int StopRTPDump(int channel,
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RTPDirections direction = kRtpIncoming);
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virtual int RTPDumpIsActive(int channel,
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RTPDirections direction = kRtpIncoming);
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virtual int SetVideoEngineBWETarget(int channel, ViENetwork* vie_network,
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int video_channel);
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protected:
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VoERTP_RTCPImpl(voe::SharedData* shared);
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virtual ~VoERTP_RTCPImpl();
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private:
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voe::SharedData* _shared;
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};
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
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