/*
 *  Copyright 2004 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 "pc/peer_connection_factory.h"

#include <cstdio>
#include <memory>
#include <utility>
#include <vector>

#include "absl/strings/match.h"
#include "api/fec_controller.h"
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/network_state_predictor.h"
#include "api/peer_connection_factory_proxy.h"
#include "api/peer_connection_proxy.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/field_trial_based_config.h"
#include "api/turn_customizer.h"
#include "api/units/data_rate.h"
#include "api/video_track_source_proxy.h"
#include "media/base/rtp_data_engine.h"
#include "media/sctp/sctp_transport.h"
#include "p2p/base/basic_async_resolver_factory.h"
#include "p2p/base/basic_packet_socket_factory.h"
#include "p2p/base/default_ice_transport_factory.h"
#include "p2p/client/basic_port_allocator.h"
#include "pc/audio_track.h"
#include "pc/local_audio_source.h"
#include "pc/media_stream.h"
#include "pc/peer_connection.h"
#include "pc/rtp_parameters_conversion.h"
#include "pc/video_track.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/field_trial_units.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/system/file_wrapper.h"

namespace webrtc {

rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreateModularPeerConnectionFactory(
    PeerConnectionFactoryDependencies dependencies) {
  rtc::scoped_refptr<PeerConnectionFactory> pc_factory(
      new rtc::RefCountedObject<PeerConnectionFactory>(
          std::move(dependencies)));
  // Call Initialize synchronously but make sure it is executed on
  // |signaling_thread|.
  MethodCall<PeerConnectionFactory, bool> call(
      pc_factory.get(), &PeerConnectionFactory::Initialize);
  bool result = call.Marshal(RTC_FROM_HERE, pc_factory->signaling_thread());

  if (!result) {
    return nullptr;
  }
  return PeerConnectionFactoryProxy::Create(pc_factory->signaling_thread(),
                                            pc_factory);
}

PeerConnectionFactory::PeerConnectionFactory(
    PeerConnectionFactoryDependencies dependencies)
    : context_(new rtc::RefCountedObject<ConnectionContext>(dependencies)),
      task_queue_factory_(std::move(dependencies.task_queue_factory)),
      event_log_factory_(std::move(dependencies.event_log_factory)),
      fec_controller_factory_(std::move(dependencies.fec_controller_factory)),
      network_state_predictor_factory_(
          std::move(dependencies.network_state_predictor_factory)),
      injected_network_controller_factory_(
          std::move(dependencies.network_controller_factory)),
      neteq_factory_(std::move(dependencies.neteq_factory)) {}

PeerConnectionFactory::~PeerConnectionFactory() {
  RTC_DCHECK_RUN_ON(signaling_thread());
}

bool PeerConnectionFactory::Initialize() {
  return context_->Initialize();
}

void PeerConnectionFactory::SetOptions(const Options& options) {
  context_->SetOptions(options);
}

RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
    cricket::MediaType kind) const {
  RTC_DCHECK_RUN_ON(signaling_thread());
  switch (kind) {
    case cricket::MEDIA_TYPE_AUDIO: {
      cricket::AudioCodecs cricket_codecs;
      channel_manager()->GetSupportedAudioSendCodecs(&cricket_codecs);
      return ToRtpCapabilities(
          cricket_codecs,
          channel_manager()->GetDefaultEnabledAudioRtpHeaderExtensions());
    }
    case cricket::MEDIA_TYPE_VIDEO: {
      cricket::VideoCodecs cricket_codecs;
      channel_manager()->GetSupportedVideoSendCodecs(&cricket_codecs);
      return ToRtpCapabilities(
          cricket_codecs,
          channel_manager()->GetDefaultEnabledVideoRtpHeaderExtensions());
    }
    case cricket::MEDIA_TYPE_DATA:
      return RtpCapabilities();
  }
  // Not reached; avoids compile warning.
  FATAL();
}

RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
    cricket::MediaType kind) const {
  RTC_DCHECK_RUN_ON(signaling_thread());
  switch (kind) {
    case cricket::MEDIA_TYPE_AUDIO: {
      cricket::AudioCodecs cricket_codecs;
      channel_manager()->GetSupportedAudioReceiveCodecs(&cricket_codecs);
      return ToRtpCapabilities(
          cricket_codecs,
          channel_manager()->GetDefaultEnabledAudioRtpHeaderExtensions());
    }
    case cricket::MEDIA_TYPE_VIDEO: {
      cricket::VideoCodecs cricket_codecs;
      channel_manager()->GetSupportedVideoReceiveCodecs(&cricket_codecs);
      return ToRtpCapabilities(
          cricket_codecs,
          channel_manager()->GetDefaultEnabledVideoRtpHeaderExtensions());
    }
    case cricket::MEDIA_TYPE_DATA:
      return RtpCapabilities();
  }
  // Not reached; avoids compile warning.
  FATAL();
}

rtc::scoped_refptr<AudioSourceInterface>
PeerConnectionFactory::CreateAudioSource(const cricket::AudioOptions& options) {
  RTC_DCHECK(signaling_thread()->IsCurrent());
  rtc::scoped_refptr<LocalAudioSource> source(
      LocalAudioSource::Create(&options));
  return source;
}

bool PeerConnectionFactory::StartAecDump(FILE* file, int64_t max_size_bytes) {
  RTC_DCHECK(signaling_thread()->IsCurrent());
  return channel_manager()->StartAecDump(FileWrapper(file), max_size_bytes);
}

void PeerConnectionFactory::StopAecDump() {
  RTC_DCHECK(signaling_thread()->IsCurrent());
  channel_manager()->StopAecDump();
}

rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
    const PeerConnectionInterface::RTCConfiguration& configuration,
    std::unique_ptr<cricket::PortAllocator> allocator,
    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
    PeerConnectionObserver* observer) {
  // Convert the legacy API into the new dependency structure.
  PeerConnectionDependencies dependencies(observer);
  dependencies.allocator = std::move(allocator);
  dependencies.cert_generator = std::move(cert_generator);
  // Pass that into the new API.
  return CreatePeerConnection(configuration, std::move(dependencies));
}

rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
    const PeerConnectionInterface::RTCConfiguration& configuration,
    PeerConnectionDependencies dependencies) {
  RTC_DCHECK(signaling_thread()->IsCurrent());
  RTC_DCHECK(!(dependencies.allocator && dependencies.packet_socket_factory))
      << "You can't set both allocator and packet_socket_factory; "
         "the former is going away (see bugs.webrtc.org/7447";

  // Set internal defaults if optional dependencies are not set.
  if (!dependencies.cert_generator) {
    dependencies.cert_generator =
        std::make_unique<rtc::RTCCertificateGenerator>(signaling_thread(),
                                                       network_thread());
  }
  if (!dependencies.allocator) {
    rtc::PacketSocketFactory* packet_socket_factory;
    if (dependencies.packet_socket_factory)
      packet_socket_factory = dependencies.packet_socket_factory.get();
    else
      packet_socket_factory = context_->default_socket_factory();

    dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>(
        context_->default_network_manager(), packet_socket_factory,
        configuration.turn_customizer);
  }

  if (!dependencies.async_resolver_factory) {
    dependencies.async_resolver_factory =
        std::make_unique<webrtc::BasicAsyncResolverFactory>();
  }

  if (!dependencies.ice_transport_factory) {
    dependencies.ice_transport_factory =
        std::make_unique<DefaultIceTransportFactory>();
  }

  dependencies.allocator->SetNetworkIgnoreMask(options().network_ignore_mask);

  std::unique_ptr<RtcEventLog> event_log =
      worker_thread()->Invoke<std::unique_ptr<RtcEventLog>>(
          RTC_FROM_HERE,
          rtc::Bind(&PeerConnectionFactory::CreateRtcEventLog_w, this));

  std::unique_ptr<Call> call = worker_thread()->Invoke<std::unique_ptr<Call>>(
      RTC_FROM_HERE,
      rtc::Bind(&PeerConnectionFactory::CreateCall_w, this, event_log.get()));

  rtc::scoped_refptr<PeerConnection> pc(
      new rtc::RefCountedObject<PeerConnection>(context_, std::move(event_log),
                                                std::move(call)));
  ActionsBeforeInitializeForTesting(pc);
  if (!pc->Initialize(configuration, std::move(dependencies))) {
    return nullptr;
  }
  return PeerConnectionProxy::Create(signaling_thread(), pc);
}

rtc::scoped_refptr<MediaStreamInterface>
PeerConnectionFactory::CreateLocalMediaStream(const std::string& stream_id) {
  RTC_DCHECK(signaling_thread()->IsCurrent());
  return MediaStreamProxy::Create(signaling_thread(),
                                  MediaStream::Create(stream_id));
}

rtc::scoped_refptr<VideoTrackInterface> PeerConnectionFactory::CreateVideoTrack(
    const std::string& id,
    VideoTrackSourceInterface* source) {
  RTC_DCHECK(signaling_thread()->IsCurrent());
  rtc::scoped_refptr<VideoTrackInterface> track(
      VideoTrack::Create(id, source, worker_thread()));
  return VideoTrackProxy::Create(signaling_thread(), worker_thread(), track);
}

rtc::scoped_refptr<AudioTrackInterface> PeerConnectionFactory::CreateAudioTrack(
    const std::string& id,
    AudioSourceInterface* source) {
  RTC_DCHECK(signaling_thread()->IsCurrent());
  rtc::scoped_refptr<AudioTrackInterface> track(AudioTrack::Create(id, source));
  return AudioTrackProxy::Create(signaling_thread(), track);
}

cricket::ChannelManager* PeerConnectionFactory::channel_manager() {
  return context_->channel_manager();
}

std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() {
  RTC_DCHECK_RUN_ON(worker_thread());

  auto encoding_type = RtcEventLog::EncodingType::Legacy;
  if (IsTrialEnabled("WebRTC-RtcEventLogNewFormat"))
    encoding_type = RtcEventLog::EncodingType::NewFormat;
  return event_log_factory_
             ? event_log_factory_->CreateRtcEventLog(encoding_type)
             : std::make_unique<RtcEventLogNull>();
}

std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
    RtcEventLog* event_log) {
  RTC_DCHECK_RUN_ON(worker_thread());

  webrtc::Call::Config call_config(event_log);
  if (!channel_manager()->media_engine() || !context_->call_factory()) {
    return nullptr;
  }
  call_config.audio_state =
      channel_manager()->media_engine()->voice().GetAudioState();

  FieldTrialParameter<DataRate> min_bandwidth("min",
                                              DataRate::KilobitsPerSec(30));
  FieldTrialParameter<DataRate> start_bandwidth("start",
                                                DataRate::KilobitsPerSec(300));
  FieldTrialParameter<DataRate> max_bandwidth("max",
                                              DataRate::KilobitsPerSec(2000));
  ParseFieldTrial({&min_bandwidth, &start_bandwidth, &max_bandwidth},
                  trials().Lookup("WebRTC-PcFactoryDefaultBitrates"));

  call_config.bitrate_config.min_bitrate_bps =
      rtc::saturated_cast<int>(min_bandwidth->bps());
  call_config.bitrate_config.start_bitrate_bps =
      rtc::saturated_cast<int>(start_bandwidth->bps());
  call_config.bitrate_config.max_bitrate_bps =
      rtc::saturated_cast<int>(max_bandwidth->bps());

  call_config.fec_controller_factory = fec_controller_factory_.get();
  call_config.task_queue_factory = task_queue_factory_.get();
  call_config.network_state_predictor_factory =
      network_state_predictor_factory_.get();
  call_config.neteq_factory = neteq_factory_.get();

  if (IsTrialEnabled("WebRTC-Bwe-InjectedCongestionController")) {
    RTC_LOG(LS_INFO) << "Using injected network controller factory";
    call_config.network_controller_factory =
        injected_network_controller_factory_.get();
  } else {
    RTC_LOG(LS_INFO) << "Using default network controller factory";
  }

  call_config.trials = &trials();

  return std::unique_ptr<Call>(
      context_->call_factory()->CreateCall(call_config));
}

bool PeerConnectionFactory::IsTrialEnabled(absl::string_view key) const {
  return absl::StartsWith(trials().Lookup(key), "Enabled");
}

}  // namespace webrtc