400 lines
11 KiB
C++
400 lines
11 KiB
C++
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/*
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* Copyright (c) 2011 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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/*
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* vie_sender.cc
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*/
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#include "vie_sender.h"
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#include "critical_section_wrapper.h"
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#include "rtp_rtcp.h"
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#ifdef WEBRTC_SRTP
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#include "SrtpModule.h"
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#endif
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#include "rtp_dump.h"
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#include "trace.h"
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namespace webrtc {
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// ----------------------------------------------------------------------------
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// Constructor
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// ----------------------------------------------------------------------------
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ViESender::ViESender(int engineId, int channelId,
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RtpRtcp& rtpRtcpModule)
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: _engineId(engineId), _channelId(channelId),
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_sendCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
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_rtpRtcp(rtpRtcpModule),
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#ifdef WEBRTC_SRTP
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_ptrSrtp(NULL),
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_ptrSrtcp(NULL),
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#endif
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_ptrExternalEncryption(NULL), _ptrSrtpBuffer(NULL),
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_ptrSrtcpBuffer(NULL), _ptrEncryptionBuffer(NULL), _ptrTransport(NULL),
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_rtpDump(NULL)
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{
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}
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// ----------------------------------------------------------------------------
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// Destructor
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// ----------------------------------------------------------------------------
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ViESender::~ViESender()
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{
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delete &_sendCritsect;
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if (_ptrSrtpBuffer)
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{
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delete[] _ptrSrtpBuffer;
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_ptrSrtpBuffer = NULL;
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}
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if (_ptrSrtcpBuffer)
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{
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delete[] _ptrSrtcpBuffer;
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_ptrSrtcpBuffer = NULL;
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}
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if (_ptrEncryptionBuffer)
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{
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delete[] _ptrEncryptionBuffer;
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_ptrEncryptionBuffer = NULL;
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}
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if (_rtpDump)
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{
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_rtpDump->Stop();
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RtpDump::DestroyRtpDump(_rtpDump);
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_rtpDump = NULL;
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}
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}
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// ----------------------------------------------------------------------------
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// RegisterExternalEncryption
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// ----------------------------------------------------------------------------
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int ViESender::RegisterExternalEncryption(Encryption* encryption)
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrExternalEncryption)
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{
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return -1;
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}
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_ptrEncryptionBuffer = new WebRtc_UWord8[kViEMaxMtu];
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if (_ptrEncryptionBuffer == NULL)
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{
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return -1;
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}
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_ptrExternalEncryption = encryption;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// DeregisterExternalEncryption
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// ----------------------------------------------------------------------------
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int ViESender::DeregisterExternalEncryption()
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrExternalEncryption == NULL)
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{
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return -1;
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}
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if (_ptrEncryptionBuffer)
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{
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delete _ptrEncryptionBuffer;
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_ptrEncryptionBuffer = NULL;
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}
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_ptrExternalEncryption = NULL;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// RegisterSendTransport
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// ----------------------------------------------------------------------------
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int ViESender::RegisterSendTransport(Transport* transport)
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrTransport)
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{
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return -1;
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}
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_ptrTransport = transport;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// DeregisterSendTransport
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// ----------------------------------------------------------------------------
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int ViESender::DeregisterSendTransport()
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrTransport == NULL)
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{
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return -1;
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}
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_ptrTransport = NULL;
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return 0;
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}
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#ifdef WEBRTC_SRTP
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// ----------------------------------------------------------------------------
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// RegisterSRTPModule
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// ----------------------------------------------------------------------------
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int ViESender::RegisterSRTPModule(SrtpModule* srtpModule)
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrSrtp ||
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srtpModule == NULL)
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{
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return -1;
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}
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_ptrSrtpBuffer = new WebRtc_UWord8[KMaxPacketSize];
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if (_ptrSrtpBuffer == NULL)
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{
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return -1;
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}
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_ptrSrtp = srtpModule;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// DeregisterSRTPModule
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// ----------------------------------------------------------------------------
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int ViESender::DeregisterSRTPModule()
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrSrtp == NULL)
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{
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return -1;
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}
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if (_ptrSrtpBuffer)
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{
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delete [] _ptrSrtpBuffer;
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_ptrSrtpBuffer = NULL;
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}
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_ptrSrtp = NULL;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// RegisterSRTCPModule
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// ----------------------------------------------------------------------------
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int ViESender::RegisterSRTCPModule(SrtpModule* srtcpModule)
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrSrtcp ||
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srtcpModule == NULL)
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{
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return -1;
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}
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_ptrSrtcpBuffer = new WebRtc_UWord8[KMaxPacketSize];
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if (_ptrSrtcpBuffer == NULL)
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{
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return -1;
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}
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_ptrSrtcp = srtcpModule;
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return 0;
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}
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// ----------------------------------------------------------------------------
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// DeregisterSRTCPModule
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// ----------------------------------------------------------------------------
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int ViESender::DeregisterSRTCPModule()
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_ptrSrtcp == NULL)
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{
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return -1;
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}
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if (_ptrSrtcpBuffer)
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{
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delete [] _ptrSrtcpBuffer;
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_ptrSrtcpBuffer = NULL;
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}
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_ptrSrtcp = NULL;
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return 0;
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}
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#endif
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// ----------------------------------------------------------------------------
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// StartRTPDump
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// ----------------------------------------------------------------------------
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int ViESender::StartRTPDump(const char fileNameUTF8[1024])
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_rtpDump)
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{
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// Restart it if it already exists and is started
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_rtpDump->Stop();
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} else
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{
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_rtpDump = RtpDump::CreateRtpDump();
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if (_rtpDump == NULL)
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(_engineId,
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_channelId),
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"%s: Failed to create RTP dump", __FUNCTION__);
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return -1;
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}
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}
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if (_rtpDump->Start(fileNameUTF8) != 0)
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{
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RtpDump::DestroyRtpDump(_rtpDump);
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_rtpDump = NULL;
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
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ViEId(_engineId, _channelId),
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"%s: Failed to start RTP dump", __FUNCTION__);
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return -1;
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}
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return 0;
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}
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// ----------------------------------------------------------------------------
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// StopRTPDump
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// ----------------------------------------------------------------------------
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int ViESender::StopRTPDump()
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (_rtpDump)
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{
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if (_rtpDump->IsActive())
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{
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_rtpDump->Stop();
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} else
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(_engineId,
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_channelId),
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"%s: Dump not active", __FUNCTION__);
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}
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RtpDump::DestroyRtpDump(_rtpDump);
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_rtpDump = NULL;
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} else
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
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ViEId(_engineId, _channelId), "%s: RTP dump not started",
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__FUNCTION__);
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return -1;
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}
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return 0;
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}
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// ----------------------------------------------------------------------------
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// SendPacket
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// ----------------------------------------------------------------------------
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int ViESender::SendPacket(int vieId, const void *data, int len)
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (!_ptrTransport)
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{
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// No transport
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return -1;
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}
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int channelId = ChannelId(vieId);
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assert(channelId == _channelId);
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// Prepare for possible encryption and sending
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WebRtc_UWord8* sendPacket = (WebRtc_UWord8*) data;
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int sendPacketLength = len;
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if (_rtpDump)
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{
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_rtpDump->DumpPacket(sendPacket, sendPacketLength);
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}
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#ifdef WEBRTC_SRTP
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if (_ptrSrtp)
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{
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_ptrSrtp->encrypt(_channelId, sendPacket, _ptrSrtpBuffer, sendPacketLength, (int*) &sendPacketLength);
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if (sendPacketLength <= 0)
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(_engineId, _channelId), "RTP encryption failed for channel");
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return -1;
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}
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else if (sendPacketLength > KMaxPacketSize)
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{
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WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo, ViEId(_engineId, _channelId),
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" %d bytes is allocated as RTP output => memory is now corrupted", KMaxPacketSize);
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return -1;
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}
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sendPacket = _ptrSrtpBuffer;
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}
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#endif
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if (_ptrExternalEncryption)
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{
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_ptrExternalEncryption->encrypt(_channelId, sendPacket,
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_ptrEncryptionBuffer, sendPacketLength,
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(int*) &sendPacketLength);
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sendPacket = _ptrEncryptionBuffer;
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}
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return _ptrTransport->SendPacket(_channelId, sendPacket, sendPacketLength);
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}
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// ----------------------------------------------------------------------------
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// SendRTCPPacket
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// ----------------------------------------------------------------------------
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int ViESender::SendRTCPPacket(int vieId, const void *data, int len)
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{
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CriticalSectionScoped cs(_sendCritsect);
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if (!_ptrTransport)
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{
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// No transport
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return -1;
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}
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int channelId = ChannelId(vieId);
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assert(channelId == _channelId);
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// Prepare for possible encryption and sending
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WebRtc_UWord8* sendPacket = (WebRtc_UWord8*) data;
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int sendPacketLength = len;
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if (_rtpDump)
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{
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_rtpDump->DumpPacket(sendPacket, sendPacketLength);
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}
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#ifdef WEBRTC_SRTP
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if (_ptrSrtcp)
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{
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_ptrSrtcp->encrypt_rtcp(_channelId, sendPacket, _ptrSrtcpBuffer, sendPacketLength, (int*) &sendPacketLength);
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if (sendPacketLength <= 0)
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(_engineId, _channelId), "RTCP encryption failed for channel");
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return -1;
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}
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else if (sendPacketLength > KMaxPacketSize)
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{
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WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo, ViEId(_engineId, _channelId), " %d bytes is allocated as RTCP output => memory is now corrupted", KMaxPacketSize);
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return -1;
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}
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sendPacket = _ptrSrtcpBuffer;
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}
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#endif
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if (_ptrExternalEncryption)
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{
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_ptrExternalEncryption->encrypt_rtcp(_channelId, sendPacket,
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_ptrEncryptionBuffer,
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sendPacketLength,
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(int*) &sendPacketLength);
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sendPacket = _ptrEncryptionBuffer;
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}
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return _ptrTransport->SendRTCPPacket(_channelId, sendPacket,
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sendPacketLength);
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}
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} // namespace webrtc
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