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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 <cstddef>
#include <cstdint>
#include <map>
#include <memory>
#include <optional>
#include <utility>
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/media_types.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_packet_receiver.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/time_utils.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 (IsRtpPacket(MakeArrayView(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,
ArrayView<const RtpExtension> audio_extensions,
ArrayView<const RtpExtension> video_extensions)
: send_call_(send_call),
task_queue_(task_queue),
demuxer_(payload_type_map),
fake_network_(std::move(pipe)),
audio_extensions_(audio_extensions),
video_extensions_(video_extensions) {
Start();
}
DirectTransport::~DirectTransport() {
next_process_task_.Stop();
}
void DirectTransport::SetReceiver(PacketReceiver* receiver) {
fake_network_->SetReceiver(receiver);
}
bool DirectTransport::SendRtp(ArrayView<const uint8_t> data,
const PacketOptions& options) {
if (send_call_) {
SentPacketInfo sent_packet(options.packet_id, 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 = data.size();
sent_packet.info.packet_type = PacketType::kData;
send_call_->OnSentPacket(sent_packet);
}
const RtpHeaderExtensionMap* extensions = nullptr;
MediaType media_type = demuxer_.GetMediaType(data.data(), data.size());
switch (demuxer_.GetMediaType(data.data(), data.size())) {
case MediaType::AUDIO:
extensions = &audio_extensions_;
break;
case MediaType::VIDEO:
extensions = &video_extensions_;
break;
default:
RTC_CHECK_NOTREACHED();
}
RtpPacketReceived packet(extensions, Timestamp::Micros(TimeMicros()));
if (media_type == MediaType::VIDEO) {
packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
}
RTC_CHECK(packet.Parse(CopyOnWriteBuffer(data)));
fake_network_->DeliverRtpPacket(
media_type, std::move(packet),
[](const RtpPacketReceived& packet) { return false; });
MutexLock lock(&process_lock_);
if (!next_process_task_.Running())
ProcessPackets();
return true;
}
bool DirectTransport::SendRtcp(ArrayView<const uint8_t> data,
const PacketOptions& /* options */) {
fake_network_->DeliverRtcpPacket(CopyOnWriteBuffer(data));
MutexLock lock(&process_lock_);
if (!next_process_task_.Running())
ProcessPackets();
return true;
}
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() {
std::optional<int64_t> initial_delay_ms =
fake_network_->TimeUntilNextProcess();
if (initial_delay_ms == std::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.
MutexLock lock(&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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