/*
 *  Copyright 2017 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 <stdint.h>

#include <memory>
#include <string>
#include <utility>
#include <vector>

#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/set_remote_description_observer_interface.h"
#include "api/uma_metrics.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "media/base/stream_params.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/port_allocator.h"
#include "pc/media_session.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/sdp_utils.h"
#include "pc/session_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/thread.h"
#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"
#include "test/gtest.h"

// This file contains unit tests that relate to the behavior of the
// SdpOfferAnswer module.
// Tests are writen as integration tests with PeerConnection, since the
// behaviors are still linked so closely that it is hard to test them in
// isolation.

namespace webrtc {

using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;

namespace {

std::unique_ptr<rtc::Thread> CreateAndStartThread() {
  auto thread = rtc::Thread::Create();
  thread->Start();
  return thread;
}

}  // namespace

class SdpOfferAnswerTest : public ::testing::Test {
 public:
  SdpOfferAnswerTest()
      // Note: We use a PeerConnectionFactory with a distinct
      // signaling thread, so that thread handling can be tested.
      : signaling_thread_(CreateAndStartThread()),
        pc_factory_(
            CreatePeerConnectionFactory(nullptr,
                                        nullptr,
                                        signaling_thread_.get(),
                                        FakeAudioCaptureModule::Create(),
                                        CreateBuiltinAudioEncoderFactory(),
                                        CreateBuiltinAudioDecoderFactory(),
                                        CreateBuiltinVideoEncoderFactory(),
                                        CreateBuiltinVideoDecoderFactory(),
                                        nullptr /* audio_mixer */,
                                        nullptr /* audio_processing */)) {
    webrtc::metrics::Reset();
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
    RTCConfiguration config;
    config.sdp_semantics = SdpSemantics::kUnifiedPlan;
    return CreatePeerConnection(config);
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
      const RTCConfiguration& config) {
    auto observer = std::make_unique<MockPeerConnectionObserver>();
    auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
                                                observer.get());
    EXPECT_TRUE(pc.get());
    observer->SetPeerConnectionInterface(pc.get());
    return std::make_unique<PeerConnectionWrapper>(pc_factory_, pc,
                                                   std::move(observer));
  }

 protected:
  std::unique_ptr<rtc::Thread> signaling_thread_;
  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;

 private:
};

TEST_F(SdpOfferAnswerTest, OnTrackReturnsProxiedObject) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  // Verify that caller->observer->OnTrack() has been called with a
  // proxied transceiver object.
  ASSERT_EQ(callee->observer()->on_track_transceivers_.size(), 1u);
  auto transceiver = callee->observer()->on_track_transceivers_[0];
  // Since the signaling thread is not the current thread,
  // this will DCHECK if the transceiver is not proxied.
  transceiver->stopped();
}

}  // namespace webrtc