blob: 20a7d7dd21e047215b124e93d7bccf5f03d63255 [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 {
namespace {
constexpr int64_t kDoNothingProcessIntervalMs = 5000;
} // namespace
FakeNetworkPipeModule::~FakeNetworkPipeModule() = default;
FakeNetworkPipeModule::FakeNetworkPipeModule(
Clock* clock,
std::unique_ptr<NetworkBehaviorInterface> network_behavior,
Transport* transport)
: pipe_(clock, std::move(network_behavior), transport) {}
void FakeNetworkPipeModule::SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) {
pipe_.SendRtp(packet, length, options);
MaybeResumeProcess();
}
void FakeNetworkPipeModule::SendRtcp(const uint8_t* packet, size_t length) {
pipe_.SendRtcp(packet, length);
MaybeResumeProcess();
}
void FakeNetworkPipeModule::MaybeResumeProcess() {
rtc::CritScope cs(&process_thread_lock_);
if (!pending_process_ && pipe_.TimeUntilNextProcess() && process_thread_) {
process_thread_->WakeUp(nullptr);
}
}
int64_t FakeNetworkPipeModule::TimeUntilNextProcess() {
auto delay = pipe_.TimeUntilNextProcess();
rtc::CritScope cs(&process_thread_lock_);
pending_process_ = delay.has_value();
return delay.value_or(kDoNothingProcessIntervalMs);
}
void FakeNetworkPipeModule::ProcessThreadAttached(
ProcessThread* process_thread) {
rtc::CritScope cs(&process_thread_lock_);
process_thread_ = process_thread;
}
void FakeNetworkPipeModule::Process() {
pipe_.Process();
}
DegradedCall::DegradedCall(
std::unique_ptr<Call> call,
absl::optional<BuiltInNetworkBehaviorConfig> send_config,
absl::optional<BuiltInNetworkBehaviorConfig> receive_config)
: clock_(Clock::GetRealTimeClock()),
call_(std::move(call)),
send_config_(send_config),
send_process_thread_(
send_config_ ? ProcessThread::Create("DegradedSendThread") : nullptr),
num_send_streams_(0),
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_process_thread_) {
send_process_thread_->Start();
}
}
DegradedCall::~DegradedCall() {
if (send_pipe_) {
send_process_thread_->DeRegisterModule(send_pipe_.get());
}
if (send_process_thread_) {
send_process_thread_->Stop();
}
}
AudioSendStream* DegradedCall::CreateAudioSendStream(
const AudioSendStream::Config& config) {
return call_->CreateAudioSendStream(config);
}
void DegradedCall::DestroyAudioSendStream(AudioSendStream* send_stream) {
call_->DestroyAudioSendStream(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) {
if (send_config_ && !send_pipe_) {
auto network = absl::make_unique<SimulatedNetwork>(*send_config_);
send_simulated_network_ = network.get();
send_pipe_ = absl::make_unique<FakeNetworkPipeModule>(
clock_, std::move(network), config.send_transport);
config.send_transport = this;
send_process_thread_->RegisterModule(send_pipe_.get(), RTC_FROM_HERE);
}
++num_send_streams_;
return call_->CreateVideoSendStream(std::move(config),
std::move(encoder_config));
}
VideoSendStream* DegradedCall::CreateVideoSendStream(
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
std::unique_ptr<FecController> fec_controller) {
if (send_config_ && !send_pipe_) {
auto network = absl::make_unique<SimulatedNetwork>(*send_config_);
send_simulated_network_ = network.get();
send_pipe_ = absl::make_unique<FakeNetworkPipeModule>(
clock_, std::move(network), config.send_transport);
config.send_transport = this;
send_process_thread_->RegisterModule(send_pipe_.get(), RTC_FROM_HERE);
}
++num_send_streams_;
return call_->CreateVideoSendStream(
std::move(config), std::move(encoder_config), std::move(fec_controller));
}
void DegradedCall::DestroyVideoSendStream(VideoSendStream* send_stream) {
call_->DestroyVideoSendStream(send_stream);
if (send_pipe_ && num_send_streams_ > 0) {
--num_send_streams_;
if (num_send_streams_ == 0) {
send_process_thread_->DeRegisterModule(send_pipe_.get());
send_pipe_.reset();
}
}
}
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::SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) {
call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
}
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);
}
bool DegradedCall::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.
send_pipe_->SendRtp(packet, length, options);
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::SendRtcp(const uint8_t* packet, size_t length) {
send_pipe_->SendRtcp(packet, length);
return true;
}
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 overikill 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;
}
void DegradedCall::MediaTransportChange(
MediaTransportInterface* media_transport) {
// TODO(bugs.webrtc.org/9719) We should add support for media transport here
// at some point.
}
} // namespace webrtc