- put references to RtpDump to #ifdef USE_RTPDUMP where possible
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@ -7,8 +7,11 @@
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#include "HL_Exception.h"
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#include "HL_String.h"
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#if defined(USE_RTP_DUMP)
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# include "jrtplib/src/rtprawpacket.h"
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# include "jrtplib/src/rtpipv4address.h"
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#endif
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#if !defined(TARGET_WIN)
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# include <alloca.h>
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#endif
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@ -97,6 +100,7 @@ int RtpHelper::findPayloadLength(const void* buffer, size_t length)
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return -1;
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}
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#if defined(USE_RTPDUMP)
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RtpDump::RtpDump(const char *filename)
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:mFilename(filename)
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{}
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@ -172,6 +176,8 @@ void RtpDump::flush()
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}
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fclose(f);
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}
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#endif
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// -------------- MediaStreamId --------------------
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bool MediaStreamId::operator < (const MediaStreamId& right) const
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@ -206,7 +212,10 @@ void writeToJson(const MediaStreamId& id, std::ostringstream& oss)
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oss << " \"src\": \"" << id.mSource.toStdString() << "\"," << std::endl
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<< " \"dst\": \"" << id.mDestination.toStdString() << "\"," << std::endl
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<< " \"ssrc\": \"" << StringHelper::toHex(id.mSSRC) << "\"," << std::endl
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<< " \"link_id\": \"" << id.mLinkId.toString() << "\"" << std::endl;
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#if !defined(USE_NULL_UUID)
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<< " \"link_id\": \"" << id.mLinkId.toString() << "\"" << std::endl
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#endif
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;
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}
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std::string MediaStreamId::getDetectDescription() const
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@ -6,8 +6,15 @@
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#ifndef __HL_RTP_H
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#define __HL_RTP_H
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#if defined(USE_RTPDUMP)
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# include "jrtplib/src/rtppacket.h"
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#endif
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#if !defined(USE_NULL_UUID)
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# include "HL_Uuid.h"
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#endif
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#include "HL_InternetAddress.h"
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#include <vector>
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@ -42,6 +49,7 @@ public:
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static int findPayloadLength(const void* buffer, size_t length);
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};
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#if defined(USE_RTPDUMP)
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class RtpDump
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{
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protected:
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@ -66,6 +74,7 @@ public:
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void add(const void* data, size_t len);
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void flush();
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};
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#endif
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struct MediaStreamId
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{
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@ -73,7 +82,9 @@ struct MediaStreamId
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InternetAddress mDestination;
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uint32_t mSSRC = 0;
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bool mSsrcIsId = true;
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#if !defined(USE_NULL_UUID)
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Uuid mLinkId;
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#endif
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bool operator < (const MediaStreamId& s2) const;
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bool operator == (const MediaStreamId& right) const;
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@ -24,7 +24,7 @@ using namespace MT;
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AudioStream::AudioStream(const CodecList::Settings& settings)
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:mPacketTime(0), mEncodedTime(0), mCodecSettings(settings),
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mRemoteTelephoneCodec(0), mRtpSession(), mTransmittingPayloadType(-1),
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mRtpSender(mStat), mRtpDump(NULL)
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mRtpSender(mStat)
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{
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mOutputBuffer.setCapacity(16384);
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mCapturedAudio.setCapacity(16384);
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@ -75,11 +75,13 @@ AudioStream::~AudioStream()
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if (mRtpSession.IsActive())
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mRtpSession.Destroy();
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#if defined(USE_RTPDUMP)
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if (mRtpDump)
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{
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mRtpDump->flush();
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delete mRtpDump;
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}
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#endif
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mCaptureResampler8.stop();
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mCaptureResampler16.stop();
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@ -79,7 +79,9 @@ namespace MT
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NativeRtpSender mRtpSender;
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AudioStreamMap mStreamMap; // Map of media streams. Key is RTP's SSRC value.
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Audio::DataWindow mOutputBuffer;
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RtpDump* mRtpDump;
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#if defined(USE_RTPDUMP)
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RtpDump* mRtpDump = nullptr;
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#endif
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Audio::Resampler mCaptureResampler8,
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mCaptureResampler16,
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mCaptureResampler32,
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@ -1,4 +1,4 @@
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/* Copyright(C) 2007-2014 VoIP objects (voipobjects.com)
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/* Copyright(C) 2007-2019 VoIP objects (voipobjects.com)
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* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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@ -9,7 +9,7 @@
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using namespace MT;
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NativeRtpSender::NativeRtpSender(Statistics& stat)
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:mDumpWriter(NULL), mStat(stat), mSrtpSession(NULL)
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:mStat(stat), mSrtpSession(nullptr)
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{
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}
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@ -22,8 +22,10 @@ bool NativeRtpSender::SendRTP(const void *data, size_t len)
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if (mTarget.mRtp.isEmpty() || !mSocket.mRtp)
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return false;
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#if defined(USE_RTPDUMP)
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if (mDumpWriter)
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mDumpWriter->add(data, len);
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#endif
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// Copy data to intermediary buffer bigger that original
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int sendLength = len;
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@ -99,6 +101,7 @@ RtpPair<PDatagramSocket>& NativeRtpSender::socket()
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return mSocket;
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}
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#if defined(USE_RTPDUMP)
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void NativeRtpSender::setDumpWriter(RtpDump *dump)
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{
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mDumpWriter = dump;
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@ -108,6 +111,7 @@ RtpDump* NativeRtpSender::dumpWriter()
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{
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return mDumpWriter;
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}
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#endif
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void NativeRtpSender::setSrtpSession(SrtpSession* srtp)
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{
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@ -39,9 +39,10 @@ namespace MT
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void setSocket(const RtpPair<PDatagramSocket>& socket);
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RtpPair<PDatagramSocket>& socket();
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#if defined(USE_RTPDUMP)
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void setDumpWriter(RtpDump* dump);
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RtpDump* dumpWriter();
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#endif
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void setSrtpSession(SrtpSession* srtp);
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SrtpSession* srtpSession();
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@ -49,7 +50,9 @@ namespace MT
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RtpPair<PDatagramSocket> mSocket;
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RtpPair<InternetAddress> mTarget;
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Statistics& mStat;
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RtpDump* mDumpWriter;
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#if defined(USE_RTPDUMP)
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RtpDump* mDumpWriter = nullptr;
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#endif
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SrtpSession* mSrtpSession;
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char mSendBuffer[MAX_VALID_UDPPACKET_SIZE];
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};
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