/*
 *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include <memory>

#include "api/test/create_frame_generator.h"
#include "api/test/frame_generator_interface.h"
#include "api/test/simulated_network.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "rtc_base/task_queue_for_test.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/frame_forwarder.h"
#include "test/gtest.h"
#include "test/null_transport.h"

namespace webrtc {

class CallOperationEndToEndTest : public test::CallTest {};

TEST_F(CallOperationEndToEndTest, ReceiverCanBeStartedTwice) {
  CreateCalls();

  test::NullTransport transport;
  CreateSendConfig(1, 0, 0, &transport);
  CreateMatchingReceiveConfigs(&transport);

  CreateVideoStreams();

  video_receive_streams_[0]->Start();
  video_receive_streams_[0]->Start();

  DestroyStreams();
}

TEST_F(CallOperationEndToEndTest, ReceiverCanBeStoppedTwice) {
  CreateCalls();

  test::NullTransport transport;
  CreateSendConfig(1, 0, 0, &transport);
  CreateMatchingReceiveConfigs(&transport);

  CreateVideoStreams();

  video_receive_streams_[0]->Stop();
  video_receive_streams_[0]->Stop();

  DestroyStreams();
}

TEST_F(CallOperationEndToEndTest, ReceiverCanBeStoppedAndRestarted) {
  CreateCalls();

  test::NullTransport transport;
  CreateSendConfig(1, 0, 0, &transport);
  CreateMatchingReceiveConfigs(&transport);

  CreateVideoStreams();

  video_receive_streams_[0]->Stop();
  video_receive_streams_[0]->Start();
  video_receive_streams_[0]->Stop();

  DestroyStreams();
}

TEST_F(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
  static const int kWidth = 320;
  static const int kHeight = 240;
  // This constant is chosen to be higher than the timeout in the video_render
  // module. This makes sure that frames aren't dropped if there are no other
  // frames in the queue.
  static const int kRenderDelayMs = 1000;

  class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
   public:
    void OnFrame(const VideoFrame& video_frame) override {
      SleepMs(kRenderDelayMs);
      event_.Set();
    }

    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }

    rtc::Event event_;
  } renderer;

  test::FrameForwarder frame_forwarder;
  std::unique_ptr<test::DirectTransport> sender_transport;
  std::unique_ptr<test::DirectTransport> receiver_transport;

  SendTask(
      RTC_FROM_HERE, task_queue(),
      [this, &renderer, &frame_forwarder, &sender_transport,
       &receiver_transport]() {
        CreateCalls();

        sender_transport = std::make_unique<test::DirectTransport>(
            task_queue(),
            std::make_unique<FakeNetworkPipe>(
                Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
                                               BuiltInNetworkBehaviorConfig())),
            sender_call_.get(), payload_type_map_);
        receiver_transport = std::make_unique<test::DirectTransport>(
            task_queue(),
            std::make_unique<FakeNetworkPipe>(
                Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
                                               BuiltInNetworkBehaviorConfig())),
            receiver_call_.get(), payload_type_map_);
        sender_transport->SetReceiver(receiver_call_->Receiver());
        receiver_transport->SetReceiver(sender_call_->Receiver());

        CreateSendConfig(1, 0, 0, sender_transport.get());
        CreateMatchingReceiveConfigs(receiver_transport.get());

        video_receive_configs_[0].renderer = &renderer;

        CreateVideoStreams();
        Start();

        // Create frames that are smaller than the send width/height, this is
        // done to check that the callbacks are done after processing video.
        std::unique_ptr<test::FrameGeneratorInterface> frame_generator(
            test::CreateSquareFrameGenerator(kWidth, kHeight, absl::nullopt,
                                             absl::nullopt));
        GetVideoSendStream()->SetSource(
            &frame_forwarder, DegradationPreference::MAINTAIN_FRAMERATE);

        test::FrameGeneratorInterface::VideoFrameData frame_data =
            frame_generator->NextFrame();
        VideoFrame frame = VideoFrame::Builder()
                               .set_video_frame_buffer(frame_data.buffer)
                               .set_update_rect(frame_data.update_rect)
                               .build();
        frame_forwarder.IncomingCapturedFrame(frame);
      });

  EXPECT_TRUE(renderer.Wait())
      << "Timed out while waiting for the frame to render.";

  SendTask(RTC_FROM_HERE, task_queue(),
           [this, &sender_transport, &receiver_transport]() {
             Stop();
             DestroyStreams();
             sender_transport.reset();
             receiver_transport.reset();
             DestroyCalls();
           });
}

TEST_F(CallOperationEndToEndTest, TransmitsFirstFrame) {
  class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
   public:
    void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }

    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }

    rtc::Event event_;
  } renderer;

  std::unique_ptr<test::FrameGeneratorInterface> frame_generator;
  test::FrameForwarder frame_forwarder;

  std::unique_ptr<test::DirectTransport> sender_transport;
  std::unique_ptr<test::DirectTransport> receiver_transport;

  SendTask(
      RTC_FROM_HERE, task_queue(),
      [this, &renderer, &frame_generator, &frame_forwarder, &sender_transport,
       &receiver_transport]() {
        CreateCalls();

        sender_transport = std::make_unique<test::DirectTransport>(
            task_queue(),
            std::make_unique<FakeNetworkPipe>(
                Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
                                               BuiltInNetworkBehaviorConfig())),
            sender_call_.get(), payload_type_map_);
        receiver_transport = std::make_unique<test::DirectTransport>(
            task_queue(),
            std::make_unique<FakeNetworkPipe>(
                Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
                                               BuiltInNetworkBehaviorConfig())),
            receiver_call_.get(), payload_type_map_);
        sender_transport->SetReceiver(receiver_call_->Receiver());
        receiver_transport->SetReceiver(sender_call_->Receiver());

        CreateSendConfig(1, 0, 0, sender_transport.get());
        CreateMatchingReceiveConfigs(receiver_transport.get());
        video_receive_configs_[0].renderer = &renderer;

        CreateVideoStreams();
        Start();

        frame_generator = test::CreateSquareFrameGenerator(
            kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt);
        GetVideoSendStream()->SetSource(
            &frame_forwarder, DegradationPreference::MAINTAIN_FRAMERATE);
        test::FrameGeneratorInterface::VideoFrameData frame_data =
            frame_generator->NextFrame();
        VideoFrame frame = VideoFrame::Builder()
                               .set_video_frame_buffer(frame_data.buffer)
                               .set_update_rect(frame_data.update_rect)
                               .build();
        frame_forwarder.IncomingCapturedFrame(frame);
      });

  EXPECT_TRUE(renderer.Wait())
      << "Timed out while waiting for the frame to render.";

  SendTask(RTC_FROM_HERE, task_queue(),
           [this, &sender_transport, &receiver_transport]() {
             Stop();
             DestroyStreams();
             sender_transport.reset();
             receiver_transport.reset();
             DestroyCalls();
           });
}

}  // namespace webrtc