mirror of
https://github.com/classilla/tenfourfox.git
synced 2024-06-09 11:29:39 +00:00
118 lines
2.8 KiB
C++
118 lines
2.8 KiB
C++
/*
|
|
* 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
|