blob: 370425c5374930f7562e3f23f71277444a43d2e5 [file] [log] [blame]
/*
* Copyright (c) 2013 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/test/direct_transport.h"
#include "webrtc/call/call.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
namespace test {
DirectTransport::DirectTransport(
Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map)
: DirectTransport(FakeNetworkPipe::Config(), send_call, payload_type_map) {}
DirectTransport::DirectTransport(
const FakeNetworkPipe::Config& config,
Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map)
: DirectTransport(
config,
send_call,
std::unique_ptr<Demuxer>(new DemuxerImpl(payload_type_map))) {}
DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
Call* send_call,
std::unique_ptr<Demuxer> demuxer)
: send_call_(send_call),
packet_event_(false, false),
thread_(NetworkProcess, this, "NetworkProcess"),
clock_(Clock::GetRealTimeClock()),
shutting_down_(false),
fake_network_(clock_, config, std::move(demuxer)) {
thread_.Start();
if (send_call_) {
send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
}
}
DirectTransport::~DirectTransport() { StopSending(); }
void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
fake_network_.SetConfig(config);
}
void DirectTransport::StopSending() {
{
rtc::CritScope crit(&lock_);
shutting_down_ = true;
}
packet_event_.Set();
thread_.Stop();
}
void DirectTransport::SetReceiver(PacketReceiver* receiver) {
fake_network_.SetReceiver(receiver);
}
bool DirectTransport::SendRtp(const uint8_t* data,
size_t length,
const PacketOptions& options) {
if (send_call_) {
rtc::SentPacket sent_packet(options.packet_id,
clock_->TimeInMilliseconds());
send_call_->OnSentPacket(sent_packet);
}
fake_network_.SendPacket(data, length);
packet_event_.Set();
return true;
}
bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
fake_network_.SendPacket(data, length);
packet_event_.Set();
return true;
}
int DirectTransport::GetAverageDelayMs() {
return fake_network_.AverageDelay();
}
DirectTransport::ForceDemuxer::ForceDemuxer(MediaType media_type)
: media_type_(media_type) {}
void DirectTransport::ForceDemuxer::SetReceiver(PacketReceiver* receiver) {
packet_receiver_ = receiver;
}
void DirectTransport::ForceDemuxer::DeliverPacket(
const NetworkPacket* packet,
const PacketTime& packet_time) {
if (!packet_receiver_)
return;
packet_receiver_->DeliverPacket(media_type_, packet->data(),
packet->data_length(), packet_time);
}
bool DirectTransport::NetworkProcess(void* transport) {
return static_cast<DirectTransport*>(transport)->SendPackets();
}
bool DirectTransport::SendPackets() {
fake_network_.Process();
int64_t wait_time_ms = fake_network_.TimeUntilNextProcess();
if (wait_time_ms > 0) {
packet_event_.Wait(static_cast<int>(wait_time_ms));
}
rtc::CritScope crit(&lock_);
return shutting_down_ ? false : true;
}
} // namespace test
} // namespace webrtc