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diff --git a/third_party/libwebrtc/test/call_test.cc b/third_party/libwebrtc/test/call_test.cc
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+++ b/third_party/libwebrtc/test/call_test.cc
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+/*
+ * Copyright (c) 2014 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/call_test.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/create_frame_generator.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "call/fake_network_pipe.h"
+#include "call/packet_receiver.h"
+#include "call/simulated_network.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/fake_encoder.h"
+#include "test/rtp_rtcp_observer.h"
+#include "test/testsupport/file_utils.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+namespace test {
+
+CallTest::CallTest()
+ : clock_(Clock::GetRealTimeClock()),
+ task_queue_factory_(CreateDefaultTaskQueueFactory()),
+ send_event_log_(std::make_unique<RtcEventLogNull>()),
+ recv_event_log_(std::make_unique<RtcEventLogNull>()),
+ audio_send_config_(/*send_transport=*/nullptr),
+ audio_send_stream_(nullptr),
+ frame_generator_capturer_(nullptr),
+ fake_encoder_factory_([this]() {
+ std::unique_ptr<FakeEncoder> fake_encoder;
+ if (video_encoder_configs_[0].codec_type == kVideoCodecVP8) {
+ fake_encoder = std::make_unique<FakeVp8Encoder>(clock_);
+ } else {
+ fake_encoder = std::make_unique<FakeEncoder>(clock_);
+ }
+ fake_encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
+ return fake_encoder;
+ }),
+ fake_decoder_factory_([]() { return std::make_unique<FakeDecoder>(); }),
+ bitrate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()),
+ num_video_streams_(1),
+ num_audio_streams_(0),
+ num_flexfec_streams_(0),
+ audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
+ audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
+ task_queue_(task_queue_factory_->CreateTaskQueue(
+ "CallTestTaskQueue",
+ TaskQueueFactory::Priority::NORMAL)) {}
+
+CallTest::~CallTest() = default;
+
+void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
+ for (const RtpExtension& registered_extension : rtp_extensions_) {
+ if (registered_extension.id == extension.id) {
+ ASSERT_EQ(registered_extension.uri, extension.uri)
+ << "Different URIs associated with ID " << extension.id << ".";
+ ASSERT_EQ(registered_extension.encrypt, extension.encrypt)
+ << "Encryption mismatch associated with ID " << extension.id << ".";
+ return;
+ } else { // Different IDs.
+ // Different IDs referring to the same extension probably indicate
+ // a mistake in the test.
+ ASSERT_FALSE(registered_extension.uri == extension.uri &&
+ registered_extension.encrypt == extension.encrypt)
+ << "URI " << extension.uri
+ << (extension.encrypt ? " with " : " without ")
+ << "encryption already registered with a different "
+ "ID ("
+ << extension.id << " vs. " << registered_extension.id << ").";
+ }
+ }
+ rtp_extensions_.push_back(extension);
+}
+
+void CallTest::RunBaseTest(BaseTest* test) {
+ SendTask(task_queue(), [this, test]() {
+ num_video_streams_ = test->GetNumVideoStreams();
+ num_audio_streams_ = test->GetNumAudioStreams();
+ num_flexfec_streams_ = test->GetNumFlexfecStreams();
+ RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
+ Call::Config send_config(send_event_log_.get());
+ test->ModifySenderBitrateConfig(&send_config.bitrate_config);
+ if (num_audio_streams_ > 0) {
+ CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
+ test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
+ fake_recv_audio_device_.get());
+ apm_send_ = AudioProcessingBuilder().Create();
+ apm_recv_ = AudioProcessingBuilder().Create();
+ EXPECT_EQ(0, fake_send_audio_device_->Init());
+ EXPECT_EQ(0, fake_recv_audio_device_->Init());
+ AudioState::Config audio_state_config;
+ audio_state_config.audio_mixer = AudioMixerImpl::Create();
+ audio_state_config.audio_processing = apm_send_;
+ audio_state_config.audio_device_module = fake_send_audio_device_;
+ send_config.audio_state = AudioState::Create(audio_state_config);
+ fake_send_audio_device_->RegisterAudioCallback(
+ send_config.audio_state->audio_transport());
+ }
+ CreateSenderCall(send_config);
+ if (test->ShouldCreateReceivers()) {
+ Call::Config recv_config(recv_event_log_.get());
+ test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
+ if (num_audio_streams_ > 0) {
+ AudioState::Config audio_state_config;
+ audio_state_config.audio_mixer = AudioMixerImpl::Create();
+ audio_state_config.audio_processing = apm_recv_;
+ audio_state_config.audio_device_module = fake_recv_audio_device_;
+ recv_config.audio_state = AudioState::Create(audio_state_config);
+ fake_recv_audio_device_->RegisterAudioCallback(
+ recv_config.audio_state->audio_transport());
+ }
+ CreateReceiverCall(recv_config);
+ }
+ test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
+ CreateReceiveTransport(test->GetReceiveTransportConfig(), test);
+ CreateSendTransport(test->GetSendTransportConfig(), test);
+ test->OnTransportCreated(send_transport_.get(), send_simulated_network_,
+ receive_transport_.get(),
+ receive_simulated_network_);
+ if (test->ShouldCreateReceivers()) {
+ if (num_video_streams_ > 0)
+ receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
+ if (num_audio_streams_ > 0)
+ receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+ } else {
+ // Sender-only call delivers to itself.
+ send_transport_->SetReceiver(sender_call_->Receiver());
+ receive_transport_->SetReceiver(nullptr);
+ }
+
+ CreateSendConfig(num_video_streams_, num_audio_streams_,
+ num_flexfec_streams_, send_transport_.get());
+ if (test->ShouldCreateReceivers()) {
+ CreateMatchingReceiveConfigs();
+ }
+ if (num_video_streams_ > 0) {
+ test->ModifyVideoConfigs(GetVideoSendConfig(), &video_receive_configs_,
+ GetVideoEncoderConfig());
+ }
+ if (num_audio_streams_ > 0) {
+ test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
+ }
+ if (num_flexfec_streams_ > 0) {
+ test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
+ }
+
+ if (num_flexfec_streams_ > 0) {
+ CreateFlexfecStreams();
+ test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
+ }
+ if (num_video_streams_ > 0) {
+ CreateVideoStreams();
+ test->OnVideoStreamsCreated(GetVideoSendStream(), video_receive_streams_);
+ }
+ if (num_audio_streams_ > 0) {
+ CreateAudioStreams();
+ test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
+ }
+
+ if (num_video_streams_ > 0) {
+ int width = kDefaultWidth;
+ int height = kDefaultHeight;
+ int frame_rate = kDefaultFramerate;
+ test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
+ test->ModifyVideoDegradationPreference(&degradation_preference_);
+ CreateFrameGeneratorCapturer(frame_rate, width, height);
+ test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_);
+ }
+
+ Start();
+ });
+
+ test->PerformTest();
+
+ SendTask(task_queue(), [this, test]() {
+ Stop();
+ test->OnStreamsStopped();
+ DestroyStreams();
+ send_transport_.reset();
+ receive_transport_.reset();
+
+ frame_generator_capturer_ = nullptr;
+ DestroyCalls();
+
+ fake_send_audio_device_ = nullptr;
+ fake_recv_audio_device_ = nullptr;
+ });
+}
+
+void CallTest::CreateCalls() {
+ CreateCalls(Call::Config(send_event_log_.get()),
+ Call::Config(recv_event_log_.get()));
+}
+
+void CallTest::CreateCalls(const Call::Config& sender_config,
+ const Call::Config& receiver_config) {
+ CreateSenderCall(sender_config);
+ CreateReceiverCall(receiver_config);
+}
+
+void CallTest::CreateSenderCall() {
+ CreateSenderCall(Call::Config(send_event_log_.get()));
+}
+
+void CallTest::CreateSenderCall(const Call::Config& config) {
+ auto sender_config = config;
+ sender_config.task_queue_factory = task_queue_factory_.get();
+ sender_config.network_state_predictor_factory =
+ network_state_predictor_factory_.get();
+ sender_config.network_controller_factory = network_controller_factory_.get();
+ sender_config.trials = &field_trials_;
+ sender_call_.reset(Call::Create(sender_config));
+}
+
+void CallTest::CreateReceiverCall(const Call::Config& config) {
+ auto receiver_config = config;
+ receiver_config.task_queue_factory = task_queue_factory_.get();
+ receiver_config.trials = &field_trials_;
+ receiver_call_.reset(Call::Create(receiver_config));
+}
+
+void CallTest::DestroyCalls() {
+ send_transport_.reset();
+ receive_transport_.reset();
+ sender_call_.reset();
+ receiver_call_.reset();
+}
+
+void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
+ size_t num_video_streams,
+ size_t num_used_ssrcs,
+ Transport* send_transport) {
+ RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, kNumSsrcs);
+ *video_config = VideoSendStream::Config(send_transport);
+ video_config->encoder_settings.encoder_factory = &fake_encoder_factory_;
+ video_config->encoder_settings.bitrate_allocator_factory =
+ bitrate_allocator_factory_.get();
+ video_config->rtp.payload_name = "FAKE";
+ video_config->rtp.payload_type = kFakeVideoSendPayloadType;
+ video_config->rtp.extmap_allow_mixed = true;
+ AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kAbsSendTimeUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kTimestampOffsetUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri00,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kDependencyDescriptorUri,
+ &video_config->rtp.extensions);
+ if (video_encoder_configs_.empty()) {
+ video_encoder_configs_.emplace_back();
+ FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
+ &video_encoder_configs_.back());
+ }
+ for (size_t i = 0; i < num_video_streams; ++i)
+ video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
+ AddRtpExtensionByUri(RtpExtension::kVideoRotationUri,
+ &video_config->rtp.extensions);
+ AddRtpExtensionByUri(RtpExtension::kColorSpaceUri,
+ &video_config->rtp.extensions);
+}
+
+void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
+ size_t num_flexfec_streams,
+ Transport* send_transport) {
+ RTC_DCHECK_LE(num_audio_streams, 1);
+ RTC_DCHECK_LE(num_flexfec_streams, 1);
+ if (num_audio_streams > 0) {
+ AudioSendStream::Config audio_send_config(send_transport);
+ audio_send_config.rtp.ssrc = kAudioSendSsrc;
+ AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
+ &audio_send_config.rtp.extensions);
+
+ audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+ kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
+ audio_send_config.encoder_factory = audio_encoder_factory_;
+ SetAudioConfig(audio_send_config);
+ }
+
+ // TODO(brandtr): Update this when we support multistream protection.
+ if (num_flexfec_streams > 0) {
+ SetSendFecConfig({kVideoSendSsrcs[0]});
+ }
+}
+
+void CallTest::SetAudioConfig(const AudioSendStream::Config& config) {
+ audio_send_config_ = config;
+}
+
+void CallTest::SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs) {
+ GetVideoSendConfig()->rtp.flexfec.payload_type = kFlexfecPayloadType;
+ GetVideoSendConfig()->rtp.flexfec.ssrc = kFlexfecSendSsrc;
+ GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs;
+}
+
+void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) {
+ send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
+ send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
+ send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
+}
+
+void CallTest::SetReceiveUlpFecConfig(
+ VideoReceiveStreamInterface::Config* receive_config) {
+ receive_config->rtp.red_payload_type = kRedPayloadType;
+ receive_config->rtp.ulpfec_payload_type = kUlpfecPayloadType;
+ receive_config->rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
+ kRedPayloadType;
+}
+
+void CallTest::CreateSendConfig(size_t num_video_streams,
+ size_t num_audio_streams,
+ size_t num_flexfec_streams,
+ Transport* send_transport) {
+ if (num_video_streams > 0) {
+ video_send_configs_.clear();
+ video_send_configs_.emplace_back(nullptr);
+ CreateVideoSendConfig(&video_send_configs_.back(), num_video_streams, 0,
+ send_transport);
+ }
+ CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
+ send_transport);
+}
+
+void CallTest::CreateMatchingVideoReceiveConfigs(
+ const VideoSendStream::Config& video_send_config,
+ Transport* rtcp_send_transport) {
+ CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport,
+ &fake_decoder_factory_, absl::nullopt,
+ false, 0);
+}
+
+void CallTest::CreateMatchingVideoReceiveConfigs(
+ const VideoSendStream::Config& video_send_config,
+ Transport* rtcp_send_transport,
+ VideoDecoderFactory* decoder_factory,
+ absl::optional<size_t> decode_sub_stream,
+ bool receiver_reference_time_report,
+ int rtp_history_ms) {
+ AddMatchingVideoReceiveConfigs(
+ &video_receive_configs_, video_send_config, rtcp_send_transport,
+ decoder_factory, decode_sub_stream, receiver_reference_time_report,
+ rtp_history_ms);
+}
+
+void CallTest::AddMatchingVideoReceiveConfigs(
+ std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
+ const VideoSendStream::Config& video_send_config,
+ Transport* rtcp_send_transport,
+ VideoDecoderFactory* decoder_factory,
+ absl::optional<size_t> decode_sub_stream,
+ bool receiver_reference_time_report,
+ int rtp_history_ms) {
+ RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
+ VideoReceiveStreamInterface::Config default_config(rtcp_send_transport);
+ default_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+ for (const RtpExtension& extension : video_send_config.rtp.extensions)
+ default_config.rtp.extensions.push_back(extension);
+ default_config.rtp.nack.rtp_history_ms = rtp_history_ms;
+ // Enable RTT calculation so NTP time estimator will work.
+ default_config.rtp.rtcp_xr.receiver_reference_time_report =
+ receiver_reference_time_report;
+ default_config.renderer = &fake_renderer_;
+
+ for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
+ VideoReceiveStreamInterface::Config video_recv_config(
+ default_config.Copy());
+ video_recv_config.decoders.clear();
+ if (!video_send_config.rtp.rtx.ssrcs.empty()) {
+ video_recv_config.rtp.rtx_ssrc = video_send_config.rtp.rtx.ssrcs[i];
+ video_recv_config.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
+ video_send_config.rtp.payload_type;
+ }
+ video_recv_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
+ VideoReceiveStreamInterface::Decoder decoder;
+
+ decoder.payload_type = video_send_config.rtp.payload_type;
+ decoder.video_format = SdpVideoFormat(video_send_config.rtp.payload_name);
+ // Force fake decoders on non-selected simulcast streams.
+ if (!decode_sub_stream || i == *decode_sub_stream) {
+ video_recv_config.decoder_factory = decoder_factory;
+ } else {
+ video_recv_config.decoder_factory = &fake_decoder_factory_;
+ }
+ video_recv_config.decoders.push_back(decoder);
+ receive_configs->emplace_back(std::move(video_recv_config));
+ }
+}
+
+void CallTest::CreateMatchingAudioAndFecConfigs(
+ Transport* rtcp_send_transport) {
+ RTC_DCHECK_GE(1, num_audio_streams_);
+ if (num_audio_streams_ == 1) {
+ CreateMatchingAudioConfigs(rtcp_send_transport, "");
+ }
+
+ // TODO(brandtr): Update this when we support multistream protection.
+ RTC_DCHECK(num_flexfec_streams_ <= 1);
+ if (num_flexfec_streams_ == 1) {
+ CreateMatchingFecConfig(rtcp_send_transport, *GetVideoSendConfig());
+ for (const RtpExtension& extension : GetVideoSendConfig()->rtp.extensions)
+ GetFlexFecConfig()->rtp.extensions.push_back(extension);
+ }
+}
+
+void CallTest::CreateMatchingAudioConfigs(Transport* transport,
+ std::string sync_group) {
+ audio_receive_configs_.push_back(CreateMatchingAudioConfig(
+ audio_send_config_, audio_decoder_factory_, transport, sync_group));
+}
+
+AudioReceiveStreamInterface::Config CallTest::CreateMatchingAudioConfig(
+ const AudioSendStream::Config& send_config,
+ rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
+ Transport* transport,
+ std::string sync_group) {
+ AudioReceiveStreamInterface::Config audio_config;
+ audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
+ audio_config.rtcp_send_transport = transport;
+ audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
+ audio_config.rtp.extensions = send_config.rtp.extensions;
+ audio_config.decoder_factory = audio_decoder_factory;
+ audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
+ audio_config.sync_group = sync_group;
+ return audio_config;
+}
+
+void CallTest::CreateMatchingFecConfig(
+ Transport* transport,
+ const VideoSendStream::Config& send_config) {
+ FlexfecReceiveStream::Config config(transport);
+ config.payload_type = send_config.rtp.flexfec.payload_type;
+ config.rtp.remote_ssrc = send_config.rtp.flexfec.ssrc;
+ config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
+ config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+ if (!video_receive_configs_.empty()) {
+ video_receive_configs_[0].rtp.protected_by_flexfec = true;
+ video_receive_configs_[0].rtp.packet_sink_ = this;
+ }
+ flexfec_receive_configs_.push_back(config);
+}
+
+void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
+ video_receive_configs_.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();
+ auto frame_generator_capturer =
+ std::make_unique<test::FrameGeneratorCapturer>(
+ clock,
+ test::CreateSquareFrameGenerator(width, height, absl::nullopt,
+ absl::nullopt),
+ framerate * speed, *task_queue_factory_);
+ frame_generator_capturer_ = frame_generator_capturer.get();
+ frame_generator_capturer->Init();
+ video_sources_.push_back(std::move(frame_generator_capturer));
+ ConnectVideoSourcesToStreams();
+}
+
+void CallTest::CreateFrameGeneratorCapturer(int framerate,
+ int width,
+ int height) {
+ video_sources_.clear();
+ auto frame_generator_capturer =
+ std::make_unique<test::FrameGeneratorCapturer>(
+ clock_,
+ test::CreateSquareFrameGenerator(width, height, absl::nullopt,
+ absl::nullopt),
+ framerate, *task_queue_factory_);
+ frame_generator_capturer_ = frame_generator_capturer.get();
+ frame_generator_capturer->Init();
+ video_sources_.push_back(std::move(frame_generator_capturer));
+ ConnectVideoSourcesToStreams();
+}
+
+void CallTest::CreateFakeAudioDevices(
+ std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+ std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
+ fake_send_audio_device_ = TestAudioDeviceModule::Create(
+ task_queue_factory_.get(), std::move(capturer), nullptr, 1.f);
+ fake_recv_audio_device_ = TestAudioDeviceModule::Create(
+ task_queue_factory_.get(), 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()));
+ }
+}
+
+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]));
+ }
+}
+
+void CallTest::CreateSendTransport(const BuiltInNetworkBehaviorConfig& config,
+ RtpRtcpObserver* observer) {
+ PacketReceiver* receiver =
+ receiver_call_ ? receiver_call_->Receiver() : nullptr;
+
+ auto network = std::make_unique<SimulatedNetwork>(config);
+ send_simulated_network_ = network.get();
+ send_transport_ = std::make_unique<PacketTransport>(
+ task_queue(), sender_call_.get(), observer,
+ test::PacketTransport::kSender, payload_type_map_,
+ std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
+ std::move(network), receiver),
+ rtp_extensions_, rtp_extensions_);
+}
+
+void CallTest::CreateReceiveTransport(
+ const BuiltInNetworkBehaviorConfig& config,
+ RtpRtcpObserver* observer) {
+ auto network = std::make_unique<SimulatedNetwork>(config);
+ receive_simulated_network_ = network.get();
+ receive_transport_ = std::make_unique<PacketTransport>(
+ task_queue(), nullptr, observer, test::PacketTransport::kReceiver,
+ payload_type_map_,
+ std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
+ std::move(network),
+ sender_call_->Receiver()),
+ rtp_extensions_, rtp_extensions_);
+}
+
+void CallTest::ConnectVideoSourcesToStreams() {
+ for (size_t i = 0; i < video_sources_.size(); ++i)
+ video_send_streams_[i]->SetSource(video_sources_[i].get(),
+ degradation_preference_);
+}
+
+void CallTest::Start() {
+ StartVideoStreams();
+ if (audio_send_stream_) {
+ audio_send_stream_->Start();
+ }
+ for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
+ audio_recv_stream->Start();
+}
+
+void CallTest::StartVideoStreams() {
+ for (size_t i = 0; i < video_send_streams_.size(); ++i) {
+ std::vector<bool> active_rtp_streams(
+ video_send_configs_[i].rtp.ssrcs.size(), true);
+ video_send_streams_[i]->StartPerRtpStream(active_rtp_streams);
+ }
+ for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
+ video_recv_stream->Start();
+}
+
+void CallTest::Stop() {
+ for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
+ audio_recv_stream->Stop();
+ if (audio_send_stream_) {
+ audio_send_stream_->Stop();
+ }
+ StopVideoStreams();
+}
+
+void CallTest::StopVideoStreams() {
+ for (VideoSendStream* video_send_stream : video_send_streams_)
+ video_send_stream->Stop();
+ for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
+ video_recv_stream->Stop();
+}
+
+void CallTest::DestroyStreams() {
+ if (audio_send_stream_)
+ sender_call_->DestroyAudioSendStream(audio_send_stream_);
+ audio_send_stream_ = nullptr;
+ for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
+ receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
+
+ DestroyVideoSendStreams();
+
+ for (VideoReceiveStreamInterface* 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();
+ video_sources_.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];
+}
+
+void CallTest::OnRtpPacket(const RtpPacketReceived& packet) {
+ // All FlexFEC streams protect all of the video streams.
+ for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
+ flexfec_recv_stream->OnRtpPacket(packet);
+}
+
+absl::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
+ const std::string& uri) const {
+ for (const auto& extension : rtp_extensions_) {
+ if (extension.uri == uri) {
+ return extension;
+ }
+ }
+ return absl::nullopt;
+}
+
+void CallTest::AddRtpExtensionByUri(
+ const std::string& uri,
+ std::vector<RtpExtension>* extensions) const {
+ const absl::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
+ if (extension) {
+ extensions->push_back(*extension);
+ }
+}
+
+constexpr size_t CallTest::kNumSsrcs;
+const int CallTest::kDefaultWidth;
+const int CallTest::kDefaultHeight;
+const int CallTest::kDefaultFramerate;
+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 std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
+ {CallTest::kVideoSendPayloadType, MediaType::VIDEO},
+ {CallTest::kFakeVideoSendPayloadType, MediaType::VIDEO},
+ {CallTest::kSendRtxPayloadType, MediaType::VIDEO},
+ {CallTest::kPayloadTypeVP8, MediaType::VIDEO},
+ {CallTest::kPayloadTypeVP9, MediaType::VIDEO},
+ {CallTest::kPayloadTypeH264, MediaType::VIDEO},
+ {CallTest::kPayloadTypeGeneric, MediaType::VIDEO},
+ {CallTest::kRedPayloadType, MediaType::VIDEO},
+ {CallTest::kRtxRedPayloadType, MediaType::VIDEO},
+ {CallTest::kUlpfecPayloadType, MediaType::VIDEO},
+ {CallTest::kFlexfecPayloadType, MediaType::VIDEO},
+ {CallTest::kAudioSendPayloadType, MediaType::AUDIO}};
+
+BaseTest::BaseTest() {}
+
+BaseTest::BaseTest(TimeDelta timeout) : RtpRtcpObserver(timeout) {}
+
+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::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {}
+
+void BaseTest::ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) {
+}
+
+void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
+
+void BaseTest::OnTransportCreated(PacketTransport* to_receiver,
+ SimulatedNetworkInterface* sender_network,
+ PacketTransport* to_sender,
+ SimulatedNetworkInterface* receiver_network) {
+}
+
+BuiltInNetworkBehaviorConfig BaseTest::GetSendTransportConfig() const {
+ return BuiltInNetworkBehaviorConfig();
+}
+BuiltInNetworkBehaviorConfig BaseTest::GetReceiveTransportConfig() const {
+ return BuiltInNetworkBehaviorConfig();
+}
+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<VideoReceiveStreamInterface::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) {}
+
+void BaseTest::ModifyVideoCaptureStartResolution(int* width,
+ int* heigt,
+ int* frame_rate) {}
+
+void BaseTest::ModifyVideoDegradationPreference(
+ DegradationPreference* degradation_preference) {}
+
+void BaseTest::OnVideoStreamsCreated(
+ VideoSendStream* send_stream,
+ const std::vector<VideoReceiveStreamInterface*>& receive_streams) {}
+
+void BaseTest::ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {}
+
+void BaseTest::OnAudioStreamsCreated(
+ AudioSendStream* send_stream,
+ const std::vector<AudioReceiveStreamInterface*>& 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(TimeDelta timeout) : BaseTest(timeout) {}
+
+bool SendTest::ShouldCreateReceivers() const {
+ return false;
+}
+
+EndToEndTest::EndToEndTest() {}
+
+EndToEndTest::EndToEndTest(TimeDelta timeout) : BaseTest(timeout) {}
+
+bool EndToEndTest::ShouldCreateReceivers() const {
+ return true;
+}
+
+} // namespace test
+} // namespace webrtc