/*
 *  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 <cstddef>
#include <cstdint>
#include <list>
#include <map>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include <vector>

#include "api/audio/audio_device.h"
#include "api/audio/builtin_audio_processing_builder.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/call/transport.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/field_trials.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/create_frame_generator.h"
#include "api/test/simulated_network.h"
#include "api/transport/bitrate_settings.h"
#include "api/units/time_delta.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_rotation.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/call.h"
#include "call/call_config.h"
#include "call/fake_network_pipe.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread.h"
#include "test/create_test_environment.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/fake_encoder.h"
#include "test/fake_vp8_encoder.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.h"
#include "test/network/simulated_network.h"
#include "test/rtp_rtcp_observer.h"
#include "test/video_test_constants.h"
#include "video/config/video_encoder_config.h"

namespace webrtc {
namespace test {

CallTest::CallTest(FieldTrials field_trials)
    : field_trials_(std::move(field_trials)),
      env_(CreateTestEnvironment({.field_trials = &field_trials_})),
      send_env_(env_),
      recv_env_(env_),
      audio_send_config_(/*send_transport=*/nullptr),
      audio_send_stream_(nullptr),
      frame_generator_capturer_(nullptr),
      fake_encoder_factory_(
          [this](const Environment& env, const SdpVideoFormat& format) {
            std::unique_ptr<FakeEncoder> fake_encoder;
            if (video_encoder_configs_[0].codec_type == kVideoCodecVP8) {
              fake_encoder = std::make_unique<FakeVp8Encoder>(env);
            } else {
              fake_encoder = std::make_unique<FakeEncoder>(env);
            }
            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()),
      network_thread_(Thread::CreateWithSocketServer()),
      task_queue_(env_.task_queue_factory().CreateTaskQueue(
          "CallTestTaskQueue",
          TaskQueueFactory::Priority::kNormal)) {
  network_thread_->Start();
}

CallTest::~CallTest() = default;

void CallTest::SetSendEventLog(std::unique_ptr<RtcEventLog> event_log) {
  EnvironmentFactory f(env_);
  f.Set(std::move(event_log));
  send_env_ = f.Create();
}

void CallTest::SetRecvEventLog(std::unique_ptr<RtcEventLog> event_log) {
  EnvironmentFactory f(env_);
  f.Set(std::move(event_log));
  recv_env_ = f.Create();
}

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);
    CallConfig send_config = SendCallConfig();
    CallConfig recv_config = RecvCallConfig();
    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_ = BuiltinAudioProcessingBuilder().Build(send_config.env);
      apm_recv_ = BuiltinAudioProcessingBuilder().Build(recv_config.env);
      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(std::move(send_config));
    if (test->ShouldCreateReceivers()) {
      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(std::move(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()) {
      network_thread()->BlockingCall([this]() {
        receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
        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 = VideoTestConstants::kDefaultWidth;
      int height = VideoTestConstants::kDefaultHeight;
      int frame_rate = VideoTestConstants::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;
  });
}

CallConfig CallTest::SendCallConfig() const {
  CallConfig sender_config(send_env_, network_thread_.get());
  sender_config.network_state_predictor_factory =
      network_state_predictor_factory_.get();
  sender_config.network_controller_factory = network_controller_factory_.get();
  return sender_config;
}

CallConfig CallTest::RecvCallConfig() const {
  return CallConfig(recv_env_, network_thread_.get());
}

void CallTest::CreateCalls() {
  CreateCalls(SendCallConfig(), RecvCallConfig());
}

void CallTest::CreateCalls(CallConfig sender_config,
                           CallConfig receiver_config) {
  CreateSenderCall(std::move(sender_config));
  CreateReceiverCall(std::move(receiver_config));
}

void CallTest::CreateSenderCall() {
  CreateSenderCall(SendCallConfig());
}

void CallTest::CreateSenderCall(CallConfig config) {
  sender_call_ = Call::Create(std::move(config));
}

void CallTest::CreateReceiverCall(CallConfig config) {
  receiver_call_ = Call::Create(std::move(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,
                VideoTestConstants::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 =
      VideoTestConstants::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(
        VideoTestConstants::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 = VideoTestConstants::kAudioSendSsrc;
    AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
                         &audio_send_config.rtp.extensions);
    audio_send_config.include_in_congestion_control_allocation = true;

    audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
        VideoTestConstants::kAudioSendPayloadType,
        {"opus", 48000, 2, {{"stereo", "1"}}});
    audio_send_config.min_bitrate_bps = 6000;
    audio_send_config.max_bitrate_bps = 60000;
    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({VideoTestConstants::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 =
      VideoTestConstants::kFlexfecPayloadType;
  GetVideoSendConfig()->rtp.flexfec.ssrc = VideoTestConstants::kFlexfecSendSsrc;
  GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs;
}

void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) {
  send_config->rtp.ulpfec.red_payload_type =
      VideoTestConstants::kRedPayloadType;
  send_config->rtp.ulpfec.ulpfec_payload_type =
      VideoTestConstants::kUlpfecPayloadType;
  send_config->rtp.ulpfec.red_rtx_payload_type =
      VideoTestConstants::kRtxRedPayloadType;
}

void CallTest::SetReceiveUlpFecConfig(
    VideoReceiveStreamInterface::Config* receive_config) {
  receive_config->rtp.red_payload_type = VideoTestConstants::kRedPayloadType;
  receive_config->rtp.ulpfec_payload_type =
      VideoTestConstants::kUlpfecPayloadType;
  receive_config->rtp
      .rtx_associated_payload_types[VideoTestConstants::kRtxRedPayloadType] =
      VideoTestConstants::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_, std::nullopt, false,
                                    0);
}

void CallTest::CreateMatchingVideoReceiveConfigs(
    const VideoSendStream::Config& video_send_config,
    Transport* rtcp_send_transport,
    VideoDecoderFactory* decoder_factory,
    std::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,
    std::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.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
          [VideoTestConstants::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());
  }
}

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,
    scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
    Transport* transport,
    std::string sync_group) {
  AudioReceiveStreamInterface::Config audio_config;
  audio_config.rtcp_send_transport = transport;
  audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
  audio_config.decoder_factory = audio_decoder_factory;
  audio_config.decoder_map = {
      {VideoTestConstants::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.remote_ssrc = send_config.rtp.flexfec.ssrc;
  config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
  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, std::nullopt,
                                           std::nullopt),
          framerate * speed, env_.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>(
          &env_.clock(),
          test::CreateSquareFrameGenerator(width, height, std::nullopt,
                                           std::nullopt),
          framerate, env_.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(env_, std::move(capturer), nullptr, 1.f);
  fake_recv_audio_device_ =
      TestAudioDeviceModule::Create(env_, 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_) {
    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_) {
      video_send_streams_[i] = sender_call_->CreateVideoSendStream(
          video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
          nullptr, fec_controller_factory_->CreateFecController(send_env_));
    } 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>(
      env_, network_thread_.get(), sender_call_.get(), observer,
      test::PacketTransport::kSender, payload_type_map_,
      std::make_unique<FakeNetworkPipe>(&env_.clock(), 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>(
      env_, network_thread_.get(), nullptr, observer,
      test::PacketTransport::kReceiver, payload_type_map_,
      std::make_unique<FakeNetworkPipe>(&env_.clock(), 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::StartVideoSources() {
  for (size_t i = 0; i < video_sources_.size(); ++i) {
    video_sources_[i]->Start();
  }
}

void CallTest::StartVideoStreams() {
  StartVideoSources();
  for (size_t i = 0; i < video_send_streams_.size(); ++i) {
    video_send_streams_[i]->Start();
  }
  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);
}

std::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
    const std::string& uri) const {
  for (const auto& extension : rtp_extensions_) {
    if (extension.uri == uri) {
      return extension;
    }
  }
  return std::nullopt;
}

void CallTest::AddRtpExtensionByUri(
    const std::string& uri,
    std::vector<RtpExtension>* extensions) const {
  const std::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
  if (extension) {
    extensions->push_back(*extension);
  }
}

const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
    {VideoTestConstants::kVideoSendPayloadType, MediaType::VIDEO},
    {VideoTestConstants::kFakeVideoSendPayloadType, MediaType::VIDEO},
    {VideoTestConstants::kSendRtxPayloadType, MediaType::VIDEO},
    {VideoTestConstants::kPayloadTypeVP8, MediaType::VIDEO},
    {VideoTestConstants::kPayloadTypeVP9, MediaType::VIDEO},
    {VideoTestConstants::kPayloadTypeH264, MediaType::VIDEO},
    {VideoTestConstants::kPayloadTypeGeneric, MediaType::VIDEO},
    {VideoTestConstants::kRedPayloadType, MediaType::VIDEO},
    {VideoTestConstants::kRtxRedPayloadType, MediaType::VIDEO},
    {VideoTestConstants::kUlpfecPayloadType, MediaType::VIDEO},
    {VideoTestConstants::kFlexfecPayloadType, MediaType::VIDEO},
    {VideoTestConstants::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(AudioDeviceModule* send_audio_device,
                                         AudioDeviceModule* 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