Refactored ViESender.
In a later CL: - References -> const or ptr. Review URL: http://webrtc-codereview.appspot.com/291003 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1011 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org
parent
d492f72e43
commit
6d26ef76ea
@ -8,388 +8,186 @@
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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 <cassert>
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#include "vie_sender.h"
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#include "critical_section_wrapper.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 "vie_sender.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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: _engineId(engineId), _channelId(channelId),
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_sendCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
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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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ViESender::ViESender(int engine_id, int channel_id)
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: engine_id_(engine_id),
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channel_id_(channel_id),
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critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
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external_encryption_(NULL),
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encryption_buffer_(NULL),
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transport_(NULL),
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rtp_dump_(NULL) {
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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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ViESender::~ViESender() {
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delete &critsect_;
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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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if (encryption_buffer_) {
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delete[] encryption_buffer_;
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encryption_buffer_ = NULL;
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}
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if (rtp_dump_) {
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rtp_dump_->Stop();
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RtpDump::DestroyRtpDump(rtp_dump_);
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rtp_dump_ = 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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int ViESender::RegisterExternalEncryption(Encryption* encryption) {
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CriticalSectionScoped cs(critsect_);
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if (external_encryption_) {
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return -1;
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}
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encryption_buffer_ = new WebRtc_UWord8[kViEMaxMtu];
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if (encryption_buffer_ == NULL) {
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return -1;
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}
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external_encryption_ = 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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int ViESender::DeregisterExternalEncryption() {
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CriticalSectionScoped cs(critsect_);
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if (external_encryption_ == NULL) {
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return -1;
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}
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if (encryption_buffer_) {
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delete encryption_buffer_;
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encryption_buffer_ = NULL;
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}
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external_encryption_ = 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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int ViESender::RegisterSendTransport(Transport* transport) {
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CriticalSectionScoped cs(critsect_);
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if (transport_) {
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return -1;
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}
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transport_ = 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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int ViESender::DeregisterSendTransport() {
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CriticalSectionScoped cs(critsect_);
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if (transport_ == NULL) {
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return -1;
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}
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transport_ = 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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int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
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CriticalSectionScoped cs(critsect_);
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if (rtp_dump_) {
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// Packet dump is already started, restart it.
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rtp_dump_->Stop();
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} else {
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rtp_dump_ = RtpDump::CreateRtpDump();
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if (rtp_dump_ == NULL) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
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ViEId(engine_id_, channel_id_),
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"StartSRTPDump: Failed to create RTP dump");
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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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if (rtp_dump_->Start(file_nameUTF8) != 0) {
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RtpDump::DestroyRtpDump(rtp_dump_);
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rtp_dump_ = NULL;
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
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ViEId(engine_id_, channel_id_),
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"StartRTPDump: Failed to start RTP dump");
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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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// 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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int ViESender::StopRTPDump() {
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CriticalSectionScoped cs(critsect_);
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if (rtp_dump_) {
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if (rtp_dump_->IsActive()) {
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rtp_dump_->Stop();
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} else {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
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ViEId(engine_id_, channel_id_),
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"StopRTPDump: Dump not active");
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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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RtpDump::DestroyRtpDump(rtp_dump_);
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rtp_dump_ = NULL;
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} else {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
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ViEId(engine_id_, channel_id_),
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"StopRTPDump: RTP dump not started");
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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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// DeregisterSRTCPModule
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// ----------------------------------------------------------------------------
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int ViESender::SendPacket(int vie_id, const void* data, int len) {
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CriticalSectionScoped cs(critsect_);
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if (!transport_) {
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// No transport
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return -1;
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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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assert(ChannelId(vie_id) == channel_id_);
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// ----------------------------------------------------------------------------
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// StartRTPDump
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// ----------------------------------------------------------------------------
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// TODO(mflodman) Change decrypt to get rid of this cast.
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void* tmp_ptr = const_cast<void*>(data);
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unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
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int send_packet_length = len;
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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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if (rtp_dump_) {
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rtp_dump_->DumpPacket(send_packet, send_packet_length);
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}
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if (external_encryption_) {
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external_encryption_->encrypt(channel_id_, send_packet,
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encryption_buffer_, send_packet_length,
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static_cast<int*>(&send_packet_length));
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send_packet = encryption_buffer_;
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}
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return transport_->SendPacket(channel_id_, send_packet, send_packet_length);
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}
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// ----------------------------------------------------------------------------
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// StopRTPDump
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// ----------------------------------------------------------------------------
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int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
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CriticalSectionScoped cs(critsect_);
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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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if (!transport_) {
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return -1;
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}
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assert(ChannelId(vie_id) == channel_id_);
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// Prepare for possible encryption and sending.
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// TODO(mflodman) Change decrypt to get rid of this cast.
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void* tmp_ptr = const_cast<void*>(data);
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unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
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int send_packet_length = len;
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if (rtp_dump_) {
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rtp_dump_->DumpPacket(send_packet, send_packet_length);
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}
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if (external_encryption_) {
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external_encryption_->encrypt_rtcp(
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channel_id_, send_packet, encryption_buffer_, send_packet_length,
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static_cast<int*>(&send_packet_length));
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send_packet = encryption_buffer_;
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}
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return transport_->SendRTCPPacket(channel_id_, send_packet,
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send_packet_length);
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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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assert(ChannelId(vieId) == _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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assert(ChannelId(vieId) == _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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{
|
||||
WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo, ViEId(_engineId, _channelId), " %d bytes is allocated as RTCP output => memory is now corrupted", KMaxPacketSize);
|
||||
return -1;
|
||||
}
|
||||
sendPacket = _ptrSrtcpBuffer;
|
||||
}
|
||||
#endif
|
||||
if (_ptrExternalEncryption)
|
||||
{
|
||||
_ptrExternalEncryption->encrypt_rtcp(_channelId, sendPacket,
|
||||
_ptrEncryptionBuffer,
|
||||
sendPacketLength,
|
||||
(int*) &sendPacketLength);
|
||||
sendPacket = _ptrEncryptionBuffer;
|
||||
}
|
||||
|
||||
return _ptrTransport->SendRTCPPacket(_channelId, sendPacket,
|
||||
sendPacketLength);
|
||||
}
|
||||
} // namespace webrtc
|
||||
} // namespace webrtc
|
||||
|
||||
@ -8,75 +8,57 @@
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
/*
|
||||
* vie_sender.h
|
||||
*/
|
||||
// ViESender is responsible for encrypting, if enabled, packets and send to
|
||||
// network.
|
||||
|
||||
#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_SENDER_H_
|
||||
#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_SENDER_H_
|
||||
#ifndef WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
|
||||
#define WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
|
||||
|
||||
// Defines
|
||||
#include "engine_configurations.h"
|
||||
#include "vie_defines.h"
|
||||
#include "typedefs.h"
|
||||
#include "common_types.h"
|
||||
|
||||
// Forward declarations
|
||||
|
||||
#ifdef WEBRTC_SRTP
|
||||
class SrtpModule;
|
||||
#endif
|
||||
#include "engine_configurations.h"
|
||||
#include "typedefs.h"
|
||||
#include "vie_defines.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
class CriticalSectionWrapper;
|
||||
class RtpDump;
|
||||
class Transport;
|
||||
class VideoCodingModule;
|
||||
|
||||
class ViESender: public Transport
|
||||
{
|
||||
public:
|
||||
ViESender(int engineId, int channelId);
|
||||
~ViESender();
|
||||
class ViESender: public Transport {
|
||||
public:
|
||||
ViESender(int engine_id, int channel_id);
|
||||
~ViESender();
|
||||
|
||||
int RegisterExternalEncryption(Encryption* encryption);
|
||||
int DeregisterExternalEncryption();
|
||||
// Registers an encryption class to use before sending packets.
|
||||
int RegisterExternalEncryption(Encryption* encryption);
|
||||
int DeregisterExternalEncryption();
|
||||
|
||||
int RegisterSendTransport(Transport* transport);
|
||||
int DeregisterSendTransport();
|
||||
// Registers transport to use for sending RTP and RTCP.
|
||||
int RegisterSendTransport(Transport* transport);
|
||||
int DeregisterSendTransport();
|
||||
|
||||
#ifdef WEBRTC_SRTP
|
||||
int RegisterSRTPModule(SrtpModule* srtpModule);
|
||||
int DeregisterSRTPModule();
|
||||
// Stores all incoming packets to file.
|
||||
int StartRTPDump(const char file_nameUTF8[1024]);
|
||||
int StopRTPDump();
|
||||
|
||||
int RegisterSRTCPModule(SrtpModule* srtpModule);
|
||||
int DeregisterSRTCPModule();
|
||||
#endif
|
||||
// Implements Transport.
|
||||
virtual int SendPacket(int vie_id, const void* data, int len);
|
||||
virtual int SendRTCPPacket(int vie_id, const void* data, int len);
|
||||
|
||||
int StartRTPDump(const char fileNameUTF8[1024]);
|
||||
int StopRTPDump();
|
||||
private:
|
||||
int engine_id_;
|
||||
int channel_id_;
|
||||
|
||||
// Implements Transport
|
||||
virtual int SendPacket(int vieId, const void *data, int len);
|
||||
virtual int SendRTCPPacket(int vieId, const void *data, int len);
|
||||
CriticalSectionWrapper& critsect_;
|
||||
|
||||
private:
|
||||
int _engineId;
|
||||
int _channelId;
|
||||
CriticalSectionWrapper& _sendCritsect;
|
||||
|
||||
#ifdef WEBRTC_SRTP
|
||||
SrtpModule* _ptrSrtp;
|
||||
SrtpModule* _ptrSrtcp;
|
||||
#endif
|
||||
|
||||
Encryption* _ptrExternalEncryption;
|
||||
WebRtc_UWord8* _ptrSrtpBuffer;
|
||||
WebRtc_UWord8* _ptrSrtcpBuffer;
|
||||
WebRtc_UWord8* _ptrEncryptionBuffer;
|
||||
Transport* _ptrTransport;
|
||||
RtpDump* _rtpDump;
|
||||
Encryption* external_encryption_;
|
||||
WebRtc_UWord8* encryption_buffer_;
|
||||
Transport* transport_;
|
||||
RtpDump* rtp_dump_;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
#endif // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_SENDER_H_
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user