blob: 5de50fd337018415246f25fb33d1215e2061c124 [file] [log] [blame]
/*
* Copyright (c) 2018 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 "call/degraded_call.h"
#include <utility>
#include "absl/memory/memory.h"
#include "rtc_base/location.h"
namespace webrtc {
DegradedCall::FakeNetworkPipeOnTaskQueue::FakeNetworkPipeOnTaskQueue(
TaskQueueFactory* task_queue_factory,
Clock* clock,
std::unique_ptr<NetworkBehaviorInterface> network_behavior)
: clock_(clock),
task_queue_(task_queue_factory->CreateTaskQueue(
"DegradedSendQueue",
TaskQueueFactory::Priority::NORMAL)),
pipe_(clock, std::move(network_behavior)) {}
void DegradedCall::FakeNetworkPipeOnTaskQueue::SendRtp(
const uint8_t* packet,
size_t length,
const PacketOptions& options,
Transport* transport) {
pipe_.SendRtp(packet, length, options, transport);
Process();
}
void DegradedCall::FakeNetworkPipeOnTaskQueue::SendRtcp(const uint8_t* packet,
size_t length,
Transport* transport) {
pipe_.SendRtcp(packet, length, transport);
Process();
}
void DegradedCall::FakeNetworkPipeOnTaskQueue::AddActiveTransport(
Transport* transport) {
pipe_.AddActiveTransport(transport);
}
void DegradedCall::FakeNetworkPipeOnTaskQueue::RemoveActiveTransport(
Transport* transport) {
pipe_.RemoveActiveTransport(transport);
}
bool DegradedCall::FakeNetworkPipeOnTaskQueue::Process() {
pipe_.Process();
auto time_to_next = pipe_.TimeUntilNextProcess();
if (!time_to_next) {
// Packet was probably sent immediately.
return false;
}
task_queue_.PostTask([this, time_to_next]() {
RTC_DCHECK_RUN_ON(&task_queue_);
int64_t next_process_time = *time_to_next + clock_->TimeInMilliseconds();
if (!next_process_ms_ || next_process_time < *next_process_ms_) {
next_process_ms_ = next_process_time;
task_queue_.PostDelayedTask(
[this]() {
RTC_DCHECK_RUN_ON(&task_queue_);
if (!Process()) {
next_process_ms_.reset();
}
},
*time_to_next);
}
});
return true;
}
DegradedCall::FakeNetworkPipeTransportAdapter::FakeNetworkPipeTransportAdapter(
FakeNetworkPipeOnTaskQueue* fake_network,
Call* call,
Clock* clock,
Transport* real_transport)
: network_pipe_(fake_network),
call_(call),
clock_(clock),
real_transport_(real_transport) {
network_pipe_->AddActiveTransport(real_transport);
}
DegradedCall::FakeNetworkPipeTransportAdapter::
~FakeNetworkPipeTransportAdapter() {
network_pipe_->RemoveActiveTransport(real_transport_);
}
bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtp(
const uint8_t* packet,
size_t length,
const PacketOptions& options) {
// A call here comes from the RTP stack (probably pacer). We intercept it and
// put it in the fake network pipe instead, but report to Call that is has
// been sent, so that the bandwidth estimator sees the delay we add.
network_pipe_->SendRtp(packet, length, options, real_transport_);
if (options.packet_id != -1) {
rtc::SentPacket sent_packet;
sent_packet.packet_id = options.packet_id;
sent_packet.send_time_ms = clock_->TimeInMilliseconds();
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;
call_->OnSentPacket(sent_packet);
}
return true;
}
bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtcp(
const uint8_t* packet,
size_t length) {
network_pipe_->SendRtcp(packet, length, real_transport_);
return true;
}
DegradedCall::DegradedCall(
std::unique_ptr<Call> call,
absl::optional<BuiltInNetworkBehaviorConfig> send_config,
absl::optional<BuiltInNetworkBehaviorConfig> receive_config,
TaskQueueFactory* task_queue_factory)
: clock_(Clock::GetRealTimeClock()),
call_(std::move(call)),
task_queue_factory_(task_queue_factory),
send_config_(send_config),
send_simulated_network_(nullptr),
receive_config_(receive_config) {
if (receive_config_) {
auto network = absl::make_unique<SimulatedNetwork>(*receive_config_);
receive_simulated_network_ = network.get();
receive_pipe_ =
absl::make_unique<webrtc::FakeNetworkPipe>(clock_, std::move(network));
receive_pipe_->SetReceiver(call_->Receiver());
}
if (send_config_) {
auto network = absl::make_unique<SimulatedNetwork>(*send_config_);
send_simulated_network_ = network.get();
send_pipe_ = absl::make_unique<FakeNetworkPipeOnTaskQueue>(
task_queue_factory_, clock_, std::move(network));
}
}
DegradedCall::~DegradedCall() = default;
AudioSendStream* DegradedCall::CreateAudioSendStream(
const AudioSendStream::Config& config) {
if (send_config_) {
auto transport_adapter = absl::make_unique<FakeNetworkPipeTransportAdapter>(
send_pipe_.get(), call_.get(), clock_, config.send_transport);
AudioSendStream::Config degrade_config = config;
degrade_config.send_transport = transport_adapter.get();
AudioSendStream* send_stream = call_->CreateAudioSendStream(degrade_config);
if (send_stream) {
audio_send_transport_adapters_[send_stream] =
std::move(transport_adapter);
}
return send_stream;
}
return call_->CreateAudioSendStream(config);
}
void DegradedCall::DestroyAudioSendStream(AudioSendStream* send_stream) {
call_->DestroyAudioSendStream(send_stream);
audio_send_transport_adapters_.erase(send_stream);
}
AudioReceiveStream* DegradedCall::CreateAudioReceiveStream(
const AudioReceiveStream::Config& config) {
return call_->CreateAudioReceiveStream(config);
}
void DegradedCall::DestroyAudioReceiveStream(
AudioReceiveStream* receive_stream) {
call_->DestroyAudioReceiveStream(receive_stream);
}
VideoSendStream* DegradedCall::CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config) {
std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
if (send_config_) {
transport_adapter = absl::make_unique<FakeNetworkPipeTransportAdapter>(
send_pipe_.get(), call_.get(), clock_, config.send_transport);
config.send_transport = transport_adapter.get();
}
VideoSendStream* send_stream = call_->CreateVideoSendStream(
std::move(config), std::move(encoder_config));
if (send_stream && transport_adapter) {
video_send_transport_adapters_[send_stream] = std::move(transport_adapter);
}
return send_stream;
}
VideoSendStream* DegradedCall::CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller) {
std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
if (send_config_) {
transport_adapter = absl::make_unique<FakeNetworkPipeTransportAdapter>(
send_pipe_.get(), call_.get(), clock_, config.send_transport);
config.send_transport = transport_adapter.get();
}
VideoSendStream* send_stream = call_->CreateVideoSendStream(
std::move(config), std::move(encoder_config), std::move(fec_controller));
if (send_stream && transport_adapter) {
video_send_transport_adapters_[send_stream] = std::move(transport_adapter);
}
return send_stream;
}
void DegradedCall::DestroyVideoSendStream(VideoSendStream* send_stream) {
call_->DestroyVideoSendStream(send_stream);
video_send_transport_adapters_.erase(send_stream);
}
VideoReceiveStream* DegradedCall::CreateVideoReceiveStream(
VideoReceiveStream::Config configuration) {
return call_->CreateVideoReceiveStream(std::move(configuration));
}
void DegradedCall::DestroyVideoReceiveStream(
VideoReceiveStream* receive_stream) {
call_->DestroyVideoReceiveStream(receive_stream);
}
FlexfecReceiveStream* DegradedCall::CreateFlexfecReceiveStream(
const FlexfecReceiveStream::Config& config) {
return call_->CreateFlexfecReceiveStream(config);
}
void DegradedCall::DestroyFlexfecReceiveStream(
FlexfecReceiveStream* receive_stream) {
call_->DestroyFlexfecReceiveStream(receive_stream);
}
PacketReceiver* DegradedCall::Receiver() {
if (receive_config_) {
return this;
}
return call_->Receiver();
}
RtpTransportControllerSendInterface*
DegradedCall::GetTransportControllerSend() {
return call_->GetTransportControllerSend();
}
Call::Stats DegradedCall::GetStats() const {
return call_->GetStats();
}
void DegradedCall::SignalChannelNetworkState(MediaType media,
NetworkState state) {
call_->SignalChannelNetworkState(media, state);
}
void DegradedCall::OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) {
call_->OnAudioTransportOverheadChanged(transport_overhead_per_packet);
}
void DegradedCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
if (send_config_) {
// If we have a degraded send-transport, we have already notified call
// about the supposed network send time. Discard the actual network send
// time in order to properly fool the BWE.
return;
}
call_->OnSentPacket(sent_packet);
}
PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
PacketReceiver::DeliveryStatus status = receive_pipe_->DeliverPacket(
media_type, std::move(packet), packet_time_us);
// This is not optimal, but there are many places where there are thread
// checks that fail if we're not using the worker thread call into this
// method. If we want to fix this we probably need a task queue to do handover
// of all overriden methods, which feels like overkill for the current use
// case.
// By just having this thread call out via the Process() method we work around
// that, with the tradeoff that a non-zero delay may become a little larger
// than anticipated at very low packet rates.
receive_pipe_->Process();
return status;
}
} // namespace webrtc