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Introduce SimulatedNetworkReceiverInterface and switch DirectTransport on this interface. Also switch part of related users on DefaultNetworkSimulationConfig. This two changes united into single CL to prevent work duplication. Most changes were done because of stop including fake_network_pipe.h into direct_transport.h, so splitting this into 2 CLs will require first fix all imports of fake_network_pipe.h and then replace them on new API imports again. Bug: webrtc:9630 Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6 Reviewed-on: https://webrtc-review.googlesource.com/94762 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24336}
802 lines
30 KiB
C++
802 lines
30 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "test/call_test.h"
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#include <algorithm>
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#include "absl/memory/memory.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "api/video_codecs/video_encoder_config.h"
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#include "call/rtp_transport_controller_send.h"
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#include "modules/audio_mixer/audio_mixer_impl.h"
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#include "modules/congestion_controller/bbr/bbr_factory.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/event.h"
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#include "rtc_base/experiments/congestion_controller_experiment.h"
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#include "test/fake_encoder.h"
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#include "test/testsupport/fileutils.h"
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namespace webrtc {
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namespace test {
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namespace {
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const int kVideoRotationRtpExtensionId = 4;
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}
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CallTest::CallTest()
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: clock_(Clock::GetRealTimeClock()),
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send_event_log_(RtcEventLog::CreateNull()),
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recv_event_log_(RtcEventLog::CreateNull()),
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sender_call_transport_controller_(nullptr),
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audio_send_config_(nullptr),
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audio_send_stream_(nullptr),
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bbr_network_controller_factory_(new BbrNetworkControllerFactory()),
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fake_encoder_factory_([this]() {
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auto encoder = absl::make_unique<test::FakeEncoder>(clock_);
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encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
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return encoder;
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}),
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num_video_streams_(1),
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num_audio_streams_(0),
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num_flexfec_streams_(0),
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audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
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audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
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task_queue_("CallTestTaskQueue") {}
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CallTest::~CallTest() {
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task_queue_.SendTask([this]() {
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fake_send_audio_device_ = nullptr;
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fake_recv_audio_device_ = nullptr;
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video_capturers_.clear();
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});
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}
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void CallTest::RunBaseTest(BaseTest* test) {
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task_queue_.SendTask([this, test]() {
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num_video_streams_ = test->GetNumVideoStreams();
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num_audio_streams_ = test->GetNumAudioStreams();
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num_flexfec_streams_ = test->GetNumFlexfecStreams();
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RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
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Call::Config send_config(send_event_log_.get());
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test->ModifySenderCallConfig(&send_config);
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if (num_audio_streams_ > 0) {
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CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
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test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
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fake_recv_audio_device_.get());
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apm_send_ = AudioProcessingBuilder().Create();
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apm_recv_ = AudioProcessingBuilder().Create();
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EXPECT_EQ(0, fake_send_audio_device_->Init());
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EXPECT_EQ(0, fake_recv_audio_device_->Init());
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AudioState::Config audio_state_config;
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audio_state_config.audio_mixer = AudioMixerImpl::Create();
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audio_state_config.audio_processing = apm_send_;
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audio_state_config.audio_device_module = fake_send_audio_device_;
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send_config.audio_state = AudioState::Create(audio_state_config);
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fake_send_audio_device_->RegisterAudioCallback(
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send_config.audio_state->audio_transport());
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}
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CreateSenderCall(send_config);
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if (sender_call_transport_controller_ != nullptr) {
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test->OnRtpTransportControllerSendCreated(
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sender_call_transport_controller_);
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}
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if (test->ShouldCreateReceivers()) {
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Call::Config recv_config(recv_event_log_.get());
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test->ModifyReceiverCallConfig(&recv_config);
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if (num_audio_streams_ > 0) {
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AudioState::Config audio_state_config;
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audio_state_config.audio_mixer = AudioMixerImpl::Create();
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audio_state_config.audio_processing = apm_recv_;
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audio_state_config.audio_device_module = fake_recv_audio_device_;
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recv_config.audio_state = AudioState::Create(audio_state_config);
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fake_recv_audio_device_->RegisterAudioCallback(
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recv_config.audio_state->audio_transport());
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}
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CreateReceiverCall(recv_config);
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}
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test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
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receive_transport_.reset(test->CreateReceiveTransport(&task_queue_));
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send_transport_.reset(
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test->CreateSendTransport(&task_queue_, sender_call_.get()));
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if (test->ShouldCreateReceivers()) {
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send_transport_->SetReceiver(receiver_call_->Receiver());
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receive_transport_->SetReceiver(sender_call_->Receiver());
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if (num_video_streams_ > 0)
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receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
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if (num_audio_streams_ > 0)
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receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
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} else {
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// Sender-only call delivers to itself.
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send_transport_->SetReceiver(sender_call_->Receiver());
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receive_transport_->SetReceiver(nullptr);
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}
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CreateSendConfig(num_video_streams_, num_audio_streams_,
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num_flexfec_streams_, send_transport_.get());
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if (test->ShouldCreateReceivers()) {
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CreateMatchingReceiveConfigs(receive_transport_.get());
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}
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if (num_video_streams_ > 0) {
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test->ModifyVideoConfigs(GetVideoSendConfig(), &video_receive_configs_,
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GetVideoEncoderConfig());
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}
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if (num_audio_streams_ > 0) {
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test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
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}
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if (num_flexfec_streams_ > 0) {
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test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
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}
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if (num_flexfec_streams_ > 0) {
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CreateFlexfecStreams();
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test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
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}
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if (num_video_streams_ > 0) {
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CreateVideoStreams();
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test->OnVideoStreamsCreated(GetVideoSendStream(), video_receive_streams_);
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}
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if (num_audio_streams_ > 0) {
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CreateAudioStreams();
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test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
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}
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if (num_video_streams_ > 0) {
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int width = kDefaultWidth;
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int height = kDefaultHeight;
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int frame_rate = kDefaultFramerate;
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test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
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CreateFrameGeneratorCapturer(frame_rate, width, height);
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test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_);
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}
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Start();
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});
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test->PerformTest();
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task_queue_.SendTask([this, test]() {
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Stop();
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test->OnStreamsStopped();
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DestroyStreams();
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send_transport_.reset();
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receive_transport_.reset();
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DestroyCalls();
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});
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}
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void CallTest::CreateCalls() {
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CreateCalls(Call::Config(send_event_log_.get()),
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Call::Config(recv_event_log_.get()));
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}
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void CallTest::CreateCalls(const Call::Config& sender_config,
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const Call::Config& receiver_config) {
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CreateSenderCall(sender_config);
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CreateReceiverCall(receiver_config);
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}
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void CallTest::CreateSenderCall() {
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CreateSenderCall(Call::Config(send_event_log_.get()));
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}
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void CallTest::CreateSenderCall(const Call::Config& config) {
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NetworkControllerFactoryInterface* injected_factory =
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config.network_controller_factory;
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if (!injected_factory) {
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if (CongestionControllerExperiment::BbrControllerEnabled()) {
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RTC_LOG(LS_INFO) << "Using BBR network controller factory";
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injected_factory = bbr_network_controller_factory_.get();
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} else {
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RTC_LOG(LS_INFO) << "Using default network controller factory";
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}
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}
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std::unique_ptr<RtpTransportControllerSend> controller_send =
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absl::make_unique<RtpTransportControllerSend>(
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Clock::GetRealTimeClock(), config.event_log, injected_factory,
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config.bitrate_config);
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sender_call_transport_controller_ = controller_send.get();
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sender_call_.reset(Call::Create(config, std::move(controller_send)));
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}
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void CallTest::CreateReceiverCall(const Call::Config& config) {
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receiver_call_.reset(Call::Create(config));
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}
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void CallTest::DestroyCalls() {
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sender_call_.reset();
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receiver_call_.reset();
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}
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void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
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size_t num_video_streams,
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size_t num_used_ssrcs,
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Transport* send_transport) {
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RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, kNumSsrcs);
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*video_config = VideoSendStream::Config(send_transport);
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video_config->encoder_settings.encoder_factory = &fake_encoder_factory_;
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video_config->rtp.payload_name = "FAKE";
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video_config->rtp.payload_type = kFakeVideoSendPayloadType;
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video_config->rtp.extensions.push_back(
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RtpExtension(RtpExtension::kTransportSequenceNumberUri,
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kTransportSequenceNumberExtensionId));
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video_config->rtp.extensions.push_back(RtpExtension(
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RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
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if (video_encoder_configs_.empty()) {
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video_encoder_configs_.emplace_back();
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FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
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&video_encoder_configs_.back());
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}
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for (size_t i = 0; i < num_video_streams; ++i)
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video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
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video_config->rtp.extensions.push_back(RtpExtension(
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RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
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}
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void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport) {
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RTC_DCHECK_LE(num_audio_streams, 1);
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RTC_DCHECK_LE(num_flexfec_streams, 1);
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if (num_audio_streams > 0) {
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AudioSendStream::Config audio_send_config(send_transport);
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audio_send_config.rtp.ssrc = kAudioSendSsrc;
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audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
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kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
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audio_send_config.encoder_factory = audio_encoder_factory_;
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SetAudioConfig(audio_send_config);
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}
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// TODO(brandtr): Update this when we support multistream protection.
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if (num_flexfec_streams > 0) {
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SetSendFecConfig({kVideoSendSsrcs[0]});
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}
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}
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void CallTest::SetAudioConfig(const AudioSendStream::Config& config) {
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audio_send_config_ = config;
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}
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void CallTest::SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs) {
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GetVideoSendConfig()->rtp.flexfec.payload_type = kFlexfecPayloadType;
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GetVideoSendConfig()->rtp.flexfec.ssrc = kFlexfecSendSsrc;
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GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs;
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}
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void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) {
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send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
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send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
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send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
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}
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void CallTest::SetReceiveUlpFecConfig(
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VideoReceiveStream::Config* receive_config) {
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receive_config->rtp.red_payload_type = kRedPayloadType;
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receive_config->rtp.ulpfec_payload_type = kUlpfecPayloadType;
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receive_config->rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
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kRedPayloadType;
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}
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void CallTest::CreateSendConfig(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams,
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Transport* send_transport) {
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if (num_video_streams > 0) {
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video_send_configs_.clear();
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video_send_configs_.emplace_back(nullptr);
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CreateVideoSendConfig(&video_send_configs_.back(), num_video_streams, 0,
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send_transport);
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}
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CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
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send_transport);
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}
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void CallTest::CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport) {
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CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport,
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true, absl::nullopt, false, 0);
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}
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void CallTest::CreateMatchingVideoReceiveConfigs(
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport,
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bool send_side_bwe,
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absl::optional<size_t> decode_sub_stream,
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bool receiver_reference_time_report,
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int rtp_history_ms) {
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AddMatchingVideoReceiveConfigs(
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&video_receive_configs_, video_send_config, rtcp_send_transport,
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send_side_bwe, decode_sub_stream, receiver_reference_time_report,
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rtp_history_ms);
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}
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void CallTest::AddMatchingVideoReceiveConfigs(
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std::vector<VideoReceiveStream::Config>* receive_configs,
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const VideoSendStream::Config& video_send_config,
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Transport* rtcp_send_transport,
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bool send_side_bwe,
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absl::optional<size_t> decode_sub_stream,
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bool receiver_reference_time_report,
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int rtp_history_ms) {
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RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
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VideoReceiveStream::Config default_config(rtcp_send_transport);
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default_config.rtp.remb = !send_side_bwe;
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default_config.rtp.transport_cc = send_side_bwe;
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default_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
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for (const RtpExtension& extension : video_send_config.rtp.extensions)
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default_config.rtp.extensions.push_back(extension);
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default_config.rtp.nack.rtp_history_ms = rtp_history_ms;
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// Enable RTT calculation so NTP time estimator will work.
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default_config.rtp.rtcp_xr.receiver_reference_time_report =
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receiver_reference_time_report;
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default_config.renderer = &fake_renderer_;
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for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
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VideoReceiveStream::Config video_recv_config(default_config.Copy());
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video_recv_config.decoders.clear();
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if (!video_send_config.rtp.rtx.ssrcs.empty()) {
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video_recv_config.rtp.rtx_ssrc = video_send_config.rtp.rtx.ssrcs[i];
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video_recv_config.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
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video_send_config.rtp.payload_type;
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}
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video_recv_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
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VideoReceiveStream::Decoder decoder;
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// Force fake decoders on non-selected simulcast streams.
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if (!decode_sub_stream || i == *decode_sub_stream) {
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decoder = test::CreateMatchingDecoder(video_send_config);
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} else {
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decoder.decoder = new test::FakeDecoder();
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decoder.payload_type = video_send_config.rtp.payload_type;
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decoder.payload_name = video_send_config.rtp.payload_name;
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}
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allocated_decoders_.emplace_back(decoder.decoder);
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video_recv_config.decoders.push_back(decoder);
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receive_configs->emplace_back(std::move(video_recv_config));
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}
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}
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void CallTest::CreateMatchingAudioAndFecConfigs(
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Transport* rtcp_send_transport) {
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RTC_DCHECK_GE(1, num_audio_streams_);
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if (num_audio_streams_ == 1) {
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CreateMatchingAudioConfigs(rtcp_send_transport, "");
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}
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// TODO(brandtr): Update this when we support multistream protection.
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RTC_DCHECK(num_flexfec_streams_ <= 1);
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if (num_flexfec_streams_ == 1) {
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CreateMatchingFecConfig(rtcp_send_transport, *GetVideoSendConfig());
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for (const RtpExtension& extension : GetVideoSendConfig()->rtp.extensions)
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GetFlexFecConfig()->rtp_header_extensions.push_back(extension);
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}
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}
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void CallTest::CreateMatchingAudioConfigs(Transport* transport,
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std::string sync_group) {
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audio_receive_configs_.push_back(CreateMatchingAudioConfig(
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audio_send_config_, audio_decoder_factory_, transport, sync_group));
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}
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AudioReceiveStream::Config CallTest::CreateMatchingAudioConfig(
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const AudioSendStream::Config& send_config,
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rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
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Transport* transport,
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std::string sync_group) {
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AudioReceiveStream::Config audio_config;
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audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
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audio_config.rtcp_send_transport = transport;
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audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
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audio_config.rtp.transport_cc =
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send_config.send_codec_spec
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? send_config.send_codec_spec->transport_cc_enabled
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: false;
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audio_config.rtp.extensions = send_config.rtp.extensions;
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audio_config.decoder_factory = audio_decoder_factory;
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audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
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audio_config.sync_group = sync_group;
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return audio_config;
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}
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void CallTest::CreateMatchingFecConfig(
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Transport* transport,
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const VideoSendStream::Config& send_config) {
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FlexfecReceiveStream::Config config(transport);
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config.payload_type = send_config.rtp.flexfec.payload_type;
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config.remote_ssrc = send_config.rtp.flexfec.ssrc;
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config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
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config.local_ssrc = kReceiverLocalVideoSsrc;
|
|
if (!video_receive_configs_.empty())
|
|
video_receive_configs_[0].rtp.protected_by_flexfec = true;
|
|
flexfec_receive_configs_.push_back(config);
|
|
}
|
|
|
|
void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
|
|
video_receive_configs_.clear();
|
|
allocated_decoders_.clear();
|
|
for (VideoSendStream::Config& video_send_config : video_send_configs_) {
|
|
CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport);
|
|
}
|
|
CreateMatchingAudioAndFecConfigs(rtcp_send_transport);
|
|
}
|
|
|
|
void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
|
|
float speed,
|
|
int framerate,
|
|
int width,
|
|
int height) {
|
|
video_sources_.clear();
|
|
video_capturers_.clear();
|
|
frame_generator_capturer_ = test::FrameGeneratorCapturer::Create(
|
|
width, height, absl::nullopt, absl::nullopt, framerate * speed, clock);
|
|
video_capturers_.emplace_back(
|
|
std::unique_ptr<FrameGeneratorCapturer>(frame_generator_capturer_));
|
|
video_sources_.push_back(video_capturers_.back().get());
|
|
ConnectVideoSourcesToStreams();
|
|
}
|
|
|
|
void CallTest::CreateFrameGeneratorCapturer(int framerate,
|
|
int width,
|
|
int height) {
|
|
video_sources_.clear();
|
|
video_capturers_.clear();
|
|
frame_generator_capturer_ = test::FrameGeneratorCapturer::Create(
|
|
width, height, absl::nullopt, absl::nullopt, framerate, clock_);
|
|
video_capturers_.emplace_back(
|
|
std::unique_ptr<FrameGeneratorCapturer>(frame_generator_capturer_));
|
|
video_sources_.push_back(video_capturers_.back().get());
|
|
ConnectVideoSourcesToStreams();
|
|
}
|
|
|
|
void CallTest::CreateFakeAudioDevices(
|
|
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
|
|
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
|
|
fake_send_audio_device_ = TestAudioDeviceModule::CreateTestAudioDeviceModule(
|
|
std::move(capturer), nullptr, 1.f);
|
|
fake_recv_audio_device_ = TestAudioDeviceModule::CreateTestAudioDeviceModule(
|
|
nullptr, std::move(renderer), 1.f);
|
|
}
|
|
|
|
void CallTest::CreateVideoStreams() {
|
|
RTC_DCHECK(video_receive_streams_.empty());
|
|
CreateVideoSendStreams();
|
|
for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
|
|
video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
|
|
video_receive_configs_[i].Copy()));
|
|
}
|
|
|
|
AssociateFlexfecStreamsWithVideoStreams();
|
|
}
|
|
|
|
void CallTest::CreateVideoSendStreams() {
|
|
RTC_DCHECK(video_send_streams_.empty());
|
|
|
|
// We currently only support testing external fec controllers with a single
|
|
// VideoSendStream.
|
|
if (fec_controller_factory_.get()) {
|
|
RTC_DCHECK_LE(video_send_configs_.size(), 1);
|
|
}
|
|
|
|
// TODO(http://crbug/818127):
|
|
// Remove this workaround when ALR is not screenshare-specific.
|
|
std::list<size_t> streams_creation_order;
|
|
for (size_t i = 0; i < video_send_configs_.size(); ++i) {
|
|
// If dual streams are created, add the screenshare stream last.
|
|
if (video_encoder_configs_[i].content_type ==
|
|
VideoEncoderConfig::ContentType::kScreen) {
|
|
streams_creation_order.push_back(i);
|
|
} else {
|
|
streams_creation_order.push_front(i);
|
|
}
|
|
}
|
|
|
|
video_send_streams_.resize(video_send_configs_.size(), nullptr);
|
|
|
|
for (size_t i : streams_creation_order) {
|
|
if (fec_controller_factory_.get()) {
|
|
video_send_streams_[i] = sender_call_->CreateVideoSendStream(
|
|
video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
|
|
fec_controller_factory_->CreateFecController());
|
|
} else {
|
|
video_send_streams_[i] = sender_call_->CreateVideoSendStream(
|
|
video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy());
|
|
}
|
|
}
|
|
}
|
|
|
|
void CallTest::CreateVideoSendStream(const VideoEncoderConfig& encoder_config) {
|
|
RTC_DCHECK(video_send_streams_.empty());
|
|
video_send_streams_.push_back(sender_call_->CreateVideoSendStream(
|
|
GetVideoSendConfig()->Copy(), encoder_config.Copy()));
|
|
}
|
|
|
|
void CallTest::CreateAudioStreams() {
|
|
RTC_DCHECK(audio_send_stream_ == nullptr);
|
|
RTC_DCHECK(audio_receive_streams_.empty());
|
|
audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
|
|
for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
|
|
audio_receive_streams_.push_back(
|
|
receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
|
|
}
|
|
}
|
|
|
|
void CallTest::CreateFlexfecStreams() {
|
|
for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) {
|
|
flexfec_receive_streams_.push_back(
|
|
receiver_call_->CreateFlexfecReceiveStream(
|
|
flexfec_receive_configs_[i]));
|
|
}
|
|
|
|
AssociateFlexfecStreamsWithVideoStreams();
|
|
}
|
|
|
|
void CallTest::ConnectVideoSourcesToStreams() {
|
|
for (size_t i = 0; i < video_sources_.size(); ++i)
|
|
video_send_streams_[i]->SetSource(video_sources_[i],
|
|
degradation_preference_);
|
|
}
|
|
|
|
void CallTest::AssociateFlexfecStreamsWithVideoStreams() {
|
|
// All FlexFEC streams protect all of the video streams.
|
|
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) {
|
|
for (VideoReceiveStream* video_recv_stream : video_receive_streams_) {
|
|
video_recv_stream->AddSecondarySink(flexfec_recv_stream);
|
|
}
|
|
}
|
|
}
|
|
|
|
void CallTest::DissociateFlexfecStreamsFromVideoStreams() {
|
|
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) {
|
|
for (VideoReceiveStream* video_recv_stream : video_receive_streams_) {
|
|
video_recv_stream->RemoveSecondarySink(flexfec_recv_stream);
|
|
}
|
|
}
|
|
}
|
|
|
|
void CallTest::Start() {
|
|
StartVideoStreams();
|
|
if (audio_send_stream_) {
|
|
audio_send_stream_->Start();
|
|
}
|
|
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
|
|
audio_recv_stream->Start();
|
|
StartVideoCapture();
|
|
}
|
|
|
|
void CallTest::StartVideoStreams() {
|
|
for (VideoSendStream* video_send_stream : video_send_streams_)
|
|
video_send_stream->Start();
|
|
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
|
|
video_recv_stream->Start();
|
|
}
|
|
|
|
void CallTest::StartVideoCapture() {
|
|
for (auto& capturer : video_capturers_)
|
|
capturer->Start();
|
|
}
|
|
|
|
void CallTest::Stop() {
|
|
StopVideoCapture();
|
|
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
|
|
audio_recv_stream->Stop();
|
|
if (audio_send_stream_) {
|
|
audio_send_stream_->Stop();
|
|
}
|
|
StopVideoStreams();
|
|
}
|
|
|
|
void CallTest::StopVideoCapture() {
|
|
for (auto& capturer : video_capturers_)
|
|
capturer->Stop();
|
|
}
|
|
|
|
void CallTest::StopVideoStreams() {
|
|
for (VideoSendStream* video_send_stream : video_send_streams_)
|
|
video_send_stream->Stop();
|
|
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
|
|
video_recv_stream->Stop();
|
|
}
|
|
|
|
void CallTest::DestroyStreams() {
|
|
DissociateFlexfecStreamsFromVideoStreams();
|
|
|
|
if (audio_send_stream_)
|
|
sender_call_->DestroyAudioSendStream(audio_send_stream_);
|
|
audio_send_stream_ = nullptr;
|
|
for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
|
|
receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
|
|
|
|
DestroyVideoSendStreams();
|
|
|
|
for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
|
|
receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
|
|
|
|
for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
|
|
receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
|
|
|
|
video_receive_streams_.clear();
|
|
allocated_decoders_.clear();
|
|
}
|
|
|
|
void CallTest::DestroyVideoSendStreams() {
|
|
for (VideoSendStream* video_send_stream : video_send_streams_)
|
|
sender_call_->DestroyVideoSendStream(video_send_stream);
|
|
video_send_streams_.clear();
|
|
}
|
|
|
|
void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
|
|
frame_generator_capturer_->SetFakeRotation(rotation);
|
|
}
|
|
|
|
void CallTest::SetVideoDegradation(DegradationPreference preference) {
|
|
GetVideoSendStream()->SetSource(frame_generator_capturer_, preference);
|
|
}
|
|
|
|
VideoSendStream::Config* CallTest::GetVideoSendConfig() {
|
|
return &video_send_configs_[0];
|
|
}
|
|
|
|
void CallTest::SetVideoSendConfig(const VideoSendStream::Config& config) {
|
|
video_send_configs_.clear();
|
|
video_send_configs_.push_back(config.Copy());
|
|
}
|
|
|
|
VideoEncoderConfig* CallTest::GetVideoEncoderConfig() {
|
|
return &video_encoder_configs_[0];
|
|
}
|
|
|
|
void CallTest::SetVideoEncoderConfig(const VideoEncoderConfig& config) {
|
|
video_encoder_configs_.clear();
|
|
video_encoder_configs_.push_back(config.Copy());
|
|
}
|
|
|
|
VideoSendStream* CallTest::GetVideoSendStream() {
|
|
return video_send_streams_[0];
|
|
}
|
|
FlexfecReceiveStream::Config* CallTest::GetFlexFecConfig() {
|
|
return &flexfec_receive_configs_[0];
|
|
}
|
|
|
|
constexpr size_t CallTest::kNumSsrcs;
|
|
const int CallTest::kDefaultWidth;
|
|
const int CallTest::kDefaultHeight;
|
|
const int CallTest::kDefaultFramerate;
|
|
const int CallTest::kDefaultTimeoutMs = 30 * 1000;
|
|
const int CallTest::kLongTimeoutMs = 120 * 1000;
|
|
const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {
|
|
0xBADCAFD, 0xBADCAFE, 0xBADCAFF, 0xBADCB00, 0xBADCB01, 0xBADCB02};
|
|
const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {
|
|
0xC0FFED, 0xC0FFEE, 0xC0FFEF, 0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
|
|
const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
|
|
const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF;
|
|
const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
|
|
const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
|
|
const int CallTest::kNackRtpHistoryMs = 1000;
|
|
|
|
const uint8_t CallTest::kDefaultKeepalivePayloadType =
|
|
RtpKeepAliveConfig().payload_type;
|
|
|
|
const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
|
|
{CallTest::kVideoSendPayloadType, MediaType::VIDEO},
|
|
{CallTest::kFakeVideoSendPayloadType, MediaType::VIDEO},
|
|
{CallTest::kSendRtxPayloadType, MediaType::VIDEO},
|
|
{CallTest::kRedPayloadType, MediaType::VIDEO},
|
|
{CallTest::kRtxRedPayloadType, MediaType::VIDEO},
|
|
{CallTest::kUlpfecPayloadType, MediaType::VIDEO},
|
|
{CallTest::kFlexfecPayloadType, MediaType::VIDEO},
|
|
{CallTest::kAudioSendPayloadType, MediaType::AUDIO},
|
|
{CallTest::kDefaultKeepalivePayloadType, MediaType::ANY}};
|
|
|
|
BaseTest::BaseTest() {}
|
|
|
|
BaseTest::BaseTest(int timeout_ms) : RtpRtcpObserver(timeout_ms) {}
|
|
|
|
BaseTest::~BaseTest() {}
|
|
|
|
std::unique_ptr<TestAudioDeviceModule::Capturer> BaseTest::CreateCapturer() {
|
|
return TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
|
|
}
|
|
|
|
std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() {
|
|
return TestAudioDeviceModule::CreateDiscardRenderer(48000);
|
|
}
|
|
|
|
void BaseTest::OnFakeAudioDevicesCreated(
|
|
TestAudioDeviceModule* send_audio_device,
|
|
TestAudioDeviceModule* recv_audio_device) {}
|
|
|
|
void BaseTest::ModifySenderCallConfig(Call::Config* config) {}
|
|
|
|
void BaseTest::ModifyReceiverCallConfig(Call::Config* config) {}
|
|
|
|
void BaseTest::OnRtpTransportControllerSendCreated(
|
|
RtpTransportControllerSend* controller) {}
|
|
|
|
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
|
|
|
|
test::PacketTransport* BaseTest::CreateSendTransport(
|
|
SingleThreadedTaskQueueForTesting* task_queue,
|
|
Call* sender_call) {
|
|
return new PacketTransport(
|
|
task_queue, sender_call, this, test::PacketTransport::kSender,
|
|
CallTest::payload_type_map_, DefaultNetworkSimulationConfig());
|
|
}
|
|
|
|
test::PacketTransport* BaseTest::CreateReceiveTransport(
|
|
SingleThreadedTaskQueueForTesting* task_queue) {
|
|
return new PacketTransport(
|
|
task_queue, nullptr, this, test::PacketTransport::kReceiver,
|
|
CallTest::payload_type_map_, DefaultNetworkSimulationConfig());
|
|
}
|
|
|
|
size_t BaseTest::GetNumVideoStreams() const {
|
|
return 1;
|
|
}
|
|
|
|
size_t BaseTest::GetNumAudioStreams() const {
|
|
return 0;
|
|
}
|
|
|
|
size_t BaseTest::GetNumFlexfecStreams() const {
|
|
return 0;
|
|
}
|
|
|
|
void BaseTest::ModifyVideoConfigs(
|
|
VideoSendStream::Config* send_config,
|
|
std::vector<VideoReceiveStream::Config>* receive_configs,
|
|
VideoEncoderConfig* encoder_config) {}
|
|
|
|
void BaseTest::ModifyVideoCaptureStartResolution(int* width,
|
|
int* heigt,
|
|
int* frame_rate) {}
|
|
|
|
void BaseTest::OnVideoStreamsCreated(
|
|
VideoSendStream* send_stream,
|
|
const std::vector<VideoReceiveStream*>& receive_streams) {}
|
|
|
|
void BaseTest::ModifyAudioConfigs(
|
|
AudioSendStream::Config* send_config,
|
|
std::vector<AudioReceiveStream::Config>* receive_configs) {}
|
|
|
|
void BaseTest::OnAudioStreamsCreated(
|
|
AudioSendStream* send_stream,
|
|
const std::vector<AudioReceiveStream*>& receive_streams) {}
|
|
|
|
void BaseTest::ModifyFlexfecConfigs(
|
|
std::vector<FlexfecReceiveStream::Config>* receive_configs) {}
|
|
|
|
void BaseTest::OnFlexfecStreamsCreated(
|
|
const std::vector<FlexfecReceiveStream*>& receive_streams) {}
|
|
|
|
void BaseTest::OnFrameGeneratorCapturerCreated(
|
|
FrameGeneratorCapturer* frame_generator_capturer) {}
|
|
|
|
void BaseTest::OnStreamsStopped() {}
|
|
|
|
SendTest::SendTest(int timeout_ms) : BaseTest(timeout_ms) {}
|
|
|
|
bool SendTest::ShouldCreateReceivers() const {
|
|
return false;
|
|
}
|
|
|
|
EndToEndTest::EndToEndTest() {}
|
|
|
|
EndToEndTest::EndToEndTest(int timeout_ms) : BaseTest(timeout_ms) {}
|
|
|
|
bool EndToEndTest::ShouldCreateReceivers() const {
|
|
return true;
|
|
}
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|