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/*
* 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, int 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),
static_cast<uint16_t>(len));
}
return transport_->SendPacket(channel_id_, data, len);
}
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_);
if (rtp_dump_) {
rtp_dump_->DumpPacket(static_cast<const uint8_t*>(data),
static_cast<uint16_t>(len));
}
return transport_->SendRTCPPacket(channel_id_, data, len);
}
} // namespace webrtc
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