/* * 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 #include #include #include "absl/types/optional.h" #include "api/call/transport.h" #include "api/fec_controller.h" #include "api/media_types.h" #include "api/rtp_headers.h" #include "api/test/simulated_network.h" #include "api/video_codecs/video_encoder_config.h" #include "call/audio_receive_stream.h" #include "call/audio_send_stream.h" #include "call/call.h" #include "call/fake_network_pipe.h" #include "call/flexfec_receive_stream.h" #include "call/packet_receiver.h" #include "call/rtp_transport_controller_send_interface.h" #include "call/simulated_network.h" #include "call/video_receive_stream.h" #include "call/video_send_stream.h" #include "modules/include/module.h" #include "modules/utility/include/process_thread.h" #include "rtc_base/bitrate_allocation_strategy.h" #include "rtc_base/copy_on_write_buffer.h" #include "rtc_base/network/sent_packet.h" #include "system_wrappers/include/clock.h" namespace webrtc { class FakeNetworkPipeModule : public Module { public: FakeNetworkPipeModule( Clock* clock, std::unique_ptr network_behavior, Transport* transport); ~FakeNetworkPipeModule() override; void SendRtp(const uint8_t* packet, size_t length, const PacketOptions& options); void SendRtcp(const uint8_t* packet, size_t length); // Implements Module interface int64_t TimeUntilNextProcess() override; void ProcessThreadAttached(ProcessThread* process_thread) override; void Process() override; private: void MaybeResumeProcess(); FakeNetworkPipe pipe_; rtc::CriticalSection process_thread_lock_; ProcessThread* process_thread_ RTC_GUARDED_BY(process_thread_lock_) = nullptr; bool pending_process_ RTC_GUARDED_BY(process_thread_lock_) = false; }; class DegradedCall : public Call, private Transport, private PacketReceiver { public: explicit DegradedCall( std::unique_ptr call, absl::optional send_config, absl::optional 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 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 bitrate_allocation_strategy) override; void SignalChannelNetworkState(MediaType media, NetworkState state) override; void OnAudioTransportOverheadChanged( 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_; void MediaTransportChange(MediaTransportInterface* media_transport) override; const absl::optional send_config_; const std::unique_ptr send_process_thread_; SimulatedNetwork* send_simulated_network_; std::unique_ptr send_pipe_; size_t num_send_streams_; const absl::optional receive_config_; SimulatedNetwork* receive_simulated_network_; std::unique_ptr receive_pipe_; }; } // namespace webrtc #endif // CALL_DEGRADED_CALL_H_