| /* |
| * Copyright (c) 2012 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 "webrtc/video_engine/vie_sender.h" |
| |
| #include <assert.h> |
| #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
| |
| #include "webrtc/modules/utility/interface/rtp_dump.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| |
| namespace webrtc { |
| |
| ViESender::ViESender(int channel_id) |
| : channel_id_(channel_id), |
| critsect_(CriticalSectionWrapper::CreateCriticalSection()), |
| transport_(NULL), |
| rtp_dump_(NULL) { |
| } |
| |
| ViESender::~ViESender() { |
| if (rtp_dump_) { |
| rtp_dump_->Stop(); |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } |
| } |
| |
| 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) { |
| return -1; |
| } |
| } |
| if (rtp_dump_->Start(file_nameUTF8) != 0) { |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViESender::StopRTPDump() { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (rtp_dump_) { |
| if (rtp_dump_->IsActive()) { |
| rtp_dump_->Stop(); |
| } |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } else { |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViESender::SendPacket(int vie_id, const void* data, size_t len) { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (!transport_) { |
| // No transport |
| return -1; |
| } |
| assert(ChannelId(vie_id) == channel_id_); |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data), len); |
| } |
| |
| return transport_->SendPacket(channel_id_, data, len); |
| } |
| |
| int ViESender::SendRTCPPacket(int vie_id, const void* data, size_t len) { |
| CriticalSectionScoped cs(critsect_.get()); |
| if (!transport_) { |
| return -1; |
| } |
| assert(ChannelId(vie_id) == channel_id_); |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data), len); |
| } |
| |
| return transport_->SendRTCPPacket(channel_id_, data, len); |
| } |
| |
| } // namespace webrtc |