| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <cassert> |
| |
| #include "critical_section_wrapper.h" |
| #include "rtp_dump.h" |
| #include "vie_sender.h" |
| #include "trace.h" |
| |
| namespace webrtc { |
| |
| ViESender::ViESender(int engine_id, int channel_id) |
| : engine_id_(engine_id), |
| channel_id_(channel_id), |
| critsect_(CriticalSectionWrapper::CreateCriticalSection()), |
| external_encryption_(NULL), |
| encryption_buffer_(NULL), |
| transport_(NULL), |
| rtp_dump_(NULL) { |
| } |
| |
| ViESender::~ViESender() { |
| if (encryption_buffer_) { |
| delete[] encryption_buffer_; |
| encryption_buffer_ = NULL; |
| } |
| |
| if (rtp_dump_) { |
| rtp_dump_->Stop(); |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } |
| } |
| |
| int ViESender::RegisterExternalEncryption(Encryption* encryption) { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (external_encryption_) { |
| return -1; |
| } |
| encryption_buffer_ = new WebRtc_UWord8[kViEMaxMtu]; |
| if (encryption_buffer_ == NULL) { |
| return -1; |
| } |
| external_encryption_ = encryption; |
| return 0; |
| } |
| |
| int ViESender::DeregisterExternalEncryption() { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (external_encryption_ == NULL) { |
| return -1; |
| } |
| if (encryption_buffer_) { |
| delete encryption_buffer_; |
| encryption_buffer_ = NULL; |
| } |
| external_encryption_ = NULL; |
| return 0; |
| } |
| |
| int ViESender::RegisterSendTransport(Transport* transport) { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (transport_) { |
| return -1; |
| } |
| transport_ = transport; |
| return 0; |
| } |
| |
| int ViESender::DeregisterSendTransport() { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (transport_ == NULL) { |
| return -1; |
| } |
| transport_ = NULL; |
| return 0; |
| } |
| |
| int ViESender::StartRTPDump(const char file_nameUTF8[1024]) { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (rtp_dump_) { |
| // Packet dump is already started, restart it. |
| rtp_dump_->Stop(); |
| } else { |
| rtp_dump_ = RtpDump::CreateRtpDump(); |
| if (rtp_dump_ == NULL) { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StartSRTPDump: Failed to create RTP dump"); |
| return -1; |
| } |
| } |
| if (rtp_dump_->Start(file_nameUTF8) != 0) { |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StartRTPDump: Failed to start RTP dump"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViESender::StopRTPDump() { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (rtp_dump_) { |
| if (rtp_dump_->IsActive()) { |
| rtp_dump_->Stop(); |
| } else { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StopRTPDump: Dump not active"); |
| } |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } else { |
| WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "StopRTPDump: RTP dump not started"); |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViESender::SendPacket(int vie_id, const void* data, int len) { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (!transport_) { |
| // No transport |
| return -1; |
| } |
| |
| assert(ChannelId(vie_id) == channel_id_); |
| |
| // TODO(mflodman) Change decrypt to get rid of this cast. |
| void* tmp_ptr = const_cast<void*>(data); |
| unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr); |
| int send_packet_length = len; |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket(send_packet, send_packet_length); |
| } |
| |
| if (external_encryption_) { |
| external_encryption_->encrypt(channel_id_, send_packet, |
| encryption_buffer_, send_packet_length, |
| static_cast<int*>(&send_packet_length)); |
| send_packet = encryption_buffer_; |
| } |
| |
| const int bytes_sent = transport_->SendPacket(channel_id_, send_packet, |
| send_packet_length); |
| if (bytes_sent != send_packet_length) { |
| WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "ViESender::SendPacket - Transport failed to send RTP packet"); |
| } |
| return bytes_sent; |
| } |
| |
| int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) { |
| CriticalSectionScoped cs(critsect_.get()); |
| |
| if (!transport_) { |
| return -1; |
| } |
| |
| assert(ChannelId(vie_id) == channel_id_); |
| |
| // Prepare for possible encryption and sending. |
| // TODO(mflodman) Change decrypt to get rid of this cast. |
| void* tmp_ptr = const_cast<void*>(data); |
| unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr); |
| int send_packet_length = len; |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket(send_packet, send_packet_length); |
| } |
| |
| if (external_encryption_) { |
| external_encryption_->encrypt_rtcp( |
| channel_id_, send_packet, encryption_buffer_, send_packet_length, |
| static_cast<int*>(&send_packet_length)); |
| send_packet = encryption_buffer_; |
| } |
| |
| const int bytes_sent = transport_->SendRTCPPacket(channel_id_, send_packet, |
| send_packet_length); |
| if (bytes_sent != send_packet_length) { |
| WEBRTC_TRACE( |
| webrtc::kTraceWarning, webrtc::kTraceVideo, |
| ViEId(engine_id_, channel_id_), |
| "ViESender::SendRTCPPacket - Transport failed to send RTCP packet"); |
| } |
| return bytes_sent; |
| } |
| |
| } // namespace webrtc |