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/*
* 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 "test/direct_transport.h"
#include "absl/memory/memory.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/time_utils.h"
#include "test/rtp_header_parser.h"
namespace webrtc {
namespace test {
Demuxer::Demuxer(const std::map<uint8_t, MediaType>& payload_type_map)
: payload_type_map_(payload_type_map) {}
MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
const size_t packet_length) const {
if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) {
RTC_CHECK_GE(packet_length, 2);
const uint8_t payload_type = packet_data[1] & 0x7f;
std::map<uint8_t, MediaType>::const_iterator it =
payload_type_map_.find(payload_type);
RTC_CHECK(it != payload_type_map_.end())
<< "payload type " << static_cast<int>(payload_type) << " unknown.";
return it->second;
}
return MediaType::ANY;
}
DirectTransport::DirectTransport(
TaskQueueBase* task_queue,
std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map)
: send_call_(send_call),
task_queue_(task_queue),
demuxer_(payload_type_map),
fake_network_(std::move(pipe)) {
Start();
}
DirectTransport::~DirectTransport() {
next_process_task_.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, rtc::TimeMillis());
sent_packet.info.included_in_feedback = options.included_in_feedback;
sent_packet.info.included_in_allocation = options.included_in_allocation;
sent_packet.info.packet_size_bytes = length;
sent_packet.info.packet_type = rtc::PacketType::kData;
send_call_->OnSentPacket(sent_packet);
}
SendPacket(data, length);
return true;
}
bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
SendPacket(data, length);
return true;
}
void DirectTransport::SendPacket(const uint8_t* data, size_t length) {
MediaType media_type = demuxer_.GetMediaType(data, length);
int64_t send_time_us = rtc::TimeMicros();
fake_network_->DeliverPacket(media_type, rtc::CopyOnWriteBuffer(data, length),
send_time_us);
rtc::CritScope cs(&process_lock_);
if (!next_process_task_.Running())
ProcessPackets();
}
int DirectTransport::GetAverageDelayMs() {
return fake_network_->AverageDelay();
}
void DirectTransport::Start() {
RTC_DCHECK(task_queue_);
if (send_call_) {
send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
}
}
void DirectTransport::ProcessPackets() {
absl::optional<int64_t> initial_delay_ms =
fake_network_->TimeUntilNextProcess();
if (initial_delay_ms == absl::nullopt)
return;
next_process_task_ = RepeatingTaskHandle::DelayedStart(
task_queue_, TimeDelta::Millis(*initial_delay_ms), [this] {
fake_network_->Process();
if (auto delay_ms = fake_network_->TimeUntilNextProcess())
return TimeDelta::Millis(*delay_ms);
// Otherwise stop the task.
rtc::CritScope cs(&process_lock_);
next_process_task_.Stop();
// Since this task is stopped, return value doesn't matter.
return TimeDelta::Zero();
});
}
} // namespace test
} // namespace webrtc
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