| /* | 
 |  *  Copyright (c) 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. | 
 |  */ | 
 |  | 
 | #ifndef CALL_DEGRADED_CALL_H_ | 
 | #define CALL_DEGRADED_CALL_H_ | 
 |  | 
 | #include <memory> | 
 |  | 
 | #include "absl/types/optional.h" | 
 | #include "api/call/transport.h" | 
 | #include "api/test/simulated_network.h" | 
 | #include "call/call.h" | 
 | #include "call/fake_network_pipe.h" | 
 | #include "call/simulated_network.h" | 
 | #include "modules/utility/include/process_thread.h" | 
 | #include "system_wrappers/include/clock.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | class DegradedCall : public Call, private Transport, private PacketReceiver { | 
 |  public: | 
 |   explicit DegradedCall( | 
 |       std::unique_ptr<Call> call, | 
 |       absl::optional<DefaultNetworkSimulationConfig> send_config, | 
 |       absl::optional<DefaultNetworkSimulationConfig> receive_config); | 
 |   ~DegradedCall() override; | 
 |  | 
 |   // Implements Call. | 
 |   AudioSendStream* CreateAudioSendStream( | 
 |       const AudioSendStream::Config& config) override; | 
 |   void DestroyAudioSendStream(AudioSendStream* send_stream) override; | 
 |  | 
 |   AudioReceiveStream* CreateAudioReceiveStream( | 
 |       const AudioReceiveStream::Config& config) override; | 
 |   void DestroyAudioReceiveStream(AudioReceiveStream* receive_stream) override; | 
 |  | 
 |   VideoSendStream* CreateVideoSendStream( | 
 |       VideoSendStream::Config config, | 
 |       VideoEncoderConfig encoder_config) override; | 
 |   VideoSendStream* CreateVideoSendStream( | 
 |       VideoSendStream::Config config, | 
 |       VideoEncoderConfig encoder_config, | 
 |       std::unique_ptr<FecController> fec_controller) override; | 
 |   void DestroyVideoSendStream(VideoSendStream* send_stream) override; | 
 |  | 
 |   VideoReceiveStream* CreateVideoReceiveStream( | 
 |       VideoReceiveStream::Config configuration) override; | 
 |   void DestroyVideoReceiveStream(VideoReceiveStream* receive_stream) override; | 
 |  | 
 |   FlexfecReceiveStream* CreateFlexfecReceiveStream( | 
 |       const FlexfecReceiveStream::Config& config) override; | 
 |   void DestroyFlexfecReceiveStream( | 
 |       FlexfecReceiveStream* receive_stream) override; | 
 |  | 
 |   PacketReceiver* Receiver() override; | 
 |  | 
 |   RtpTransportControllerSendInterface* GetTransportControllerSend() override; | 
 |  | 
 |   Stats GetStats() const override; | 
 |  | 
 |   void SetBitrateAllocationStrategy( | 
 |       std::unique_ptr<rtc::BitrateAllocationStrategy> | 
 |           bitrate_allocation_strategy) override; | 
 |  | 
 |   void SignalChannelNetworkState(MediaType media, NetworkState state) override; | 
 |  | 
 |   void OnTransportOverheadChanged(MediaType media, | 
 |                                   int transport_overhead_per_packet) override; | 
 |  | 
 |   void OnSentPacket(const rtc::SentPacket& sent_packet) override; | 
 |  | 
 |  protected: | 
 |   // Implements Transport. | 
 |   bool SendRtp(const uint8_t* packet, | 
 |                size_t length, | 
 |                const PacketOptions& options) override; | 
 |  | 
 |   bool SendRtcp(const uint8_t* packet, size_t length) override; | 
 |  | 
 |   // Implements PacketReceiver. | 
 |   DeliveryStatus DeliverPacket(MediaType media_type, | 
 |                                rtc::CopyOnWriteBuffer packet, | 
 |                                int64_t packet_time_us) override; | 
 |  | 
 |  private: | 
 |   Clock* const clock_; | 
 |   const std::unique_ptr<Call> call_; | 
 |  | 
 |   const absl::optional<DefaultNetworkSimulationConfig> send_config_; | 
 |   const std::unique_ptr<ProcessThread> send_process_thread_; | 
 |   SimulatedNetwork* send_simulated_network_; | 
 |   std::unique_ptr<FakeNetworkPipe> send_pipe_; | 
 |   size_t num_send_streams_; | 
 |  | 
 |   const absl::optional<DefaultNetworkSimulationConfig> receive_config_; | 
 |   SimulatedNetwork* receive_simulated_network_; | 
 |   std::unique_ptr<FakeNetworkPipe> receive_pipe_; | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // CALL_DEGRADED_CALL_H_ |