/* * 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 #include #include #include "absl/strings/string_view.h" #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/task_queue/pending_task_safety_flag.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 "rtc_base/copy_on_write_buffer.h" #include "rtc_base/network/sent_packet.h" #include "rtc_base/task_queue.h" #include "system_wrappers/include/clock.h" namespace webrtc { class DegradedCall : public Call, private PacketReceiver { public: struct TimeScopedNetworkConfig : public BuiltInNetworkBehaviorConfig { TimeDelta duration = TimeDelta::PlusInfinity(); }; explicit DegradedCall( std::unique_ptr call, const std::vector& send_configs, const std::vector& receive_configs); ~DegradedCall() override; // Implements Call. AudioSendStream* CreateAudioSendStream( const AudioSendStream::Config& config) override; void DestroyAudioSendStream(AudioSendStream* send_stream) override; AudioReceiveStreamInterface* CreateAudioReceiveStream( const AudioReceiveStreamInterface::Config& config) override; void DestroyAudioReceiveStream( AudioReceiveStreamInterface* 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; VideoReceiveStreamInterface* CreateVideoReceiveStream( VideoReceiveStreamInterface::Config configuration) override; void DestroyVideoReceiveStream( VideoReceiveStreamInterface* receive_stream) override; FlexfecReceiveStream* CreateFlexfecReceiveStream( const FlexfecReceiveStream::Config config) override; void DestroyFlexfecReceiveStream( FlexfecReceiveStream* receive_stream) override; void AddAdaptationResource(rtc::scoped_refptr resource) override; PacketReceiver* Receiver() override; RtpTransportControllerSendInterface* GetTransportControllerSend() override; Stats GetStats() const override; const FieldTrialsView& trials() const override; TaskQueueBase* network_thread() const override; TaskQueueBase* worker_thread() const override; void SignalChannelNetworkState(MediaType media, NetworkState state) override; void OnAudioTransportOverheadChanged( int transport_overhead_per_packet) override; void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(FlexfecReceiveStream& stream, uint32_t local_ssrc) override; void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream, absl::string_view sync_group) override; void OnSentPacket(const rtc::SentPacket& sent_packet) override; protected: // Implements PacketReceiver. DeliveryStatus DeliverPacket(MediaType media_type, rtc::CopyOnWriteBuffer packet, int64_t packet_time_us) override; private: class FakeNetworkPipeOnTaskQueue { public: FakeNetworkPipeOnTaskQueue( TaskQueueBase* task_queue, const ScopedTaskSafety& task_safety, Clock* clock, std::unique_ptr network_behavior); void SendRtp(const uint8_t* packet, size_t length, const PacketOptions& options, Transport* transport); void SendRtcp(const uint8_t* packet, size_t length, Transport* transport); void AddActiveTransport(Transport* transport); void RemoveActiveTransport(Transport* transport); private: // Try to process packets on the fake network queue. // Returns true if call resulted in a delayed process, false if queue empty. bool Process(); Clock* const clock_; TaskQueueBase* const task_queue_; const ScopedTaskSafety& task_safety_; FakeNetworkPipe pipe_; absl::optional next_process_ms_ RTC_GUARDED_BY(&task_queue_); }; // For audio/video send stream, a TransportAdapter instance is used to // intercept packets to be sent, and put them into a common FakeNetworkPipe // in such as way that they will eventually (unless dropped) be forwarded to // the correct Transport for that stream. class FakeNetworkPipeTransportAdapter : public Transport { public: FakeNetworkPipeTransportAdapter(FakeNetworkPipeOnTaskQueue* fake_network, Call* call, Clock* clock, Transport* real_transport); ~FakeNetworkPipeTransportAdapter(); bool SendRtp(const uint8_t* packet, size_t length, const PacketOptions& options) override; bool SendRtcp(const uint8_t* packet, size_t length) override; private: FakeNetworkPipeOnTaskQueue* const network_pipe_; Call* const call_; Clock* const clock_; Transport* const real_transport_; }; void SetClientBitratePreferences( const webrtc::BitrateSettings& preferences) override {} void UpdateSendNetworkConfig(); void UpdateReceiveNetworkConfig(); Clock* const clock_; const std::unique_ptr call_; ScopedTaskSafety task_safety_; size_t send_config_index_; const std::vector send_configs_; SimulatedNetwork* send_simulated_network_; std::unique_ptr send_pipe_; std::map> audio_send_transport_adapters_; std::map> video_send_transport_adapters_; size_t receive_config_index_; const std::vector receive_configs_; SimulatedNetwork* receive_simulated_network_; std::unique_ptr receive_pipe_; }; } // namespace webrtc #endif // CALL_DEGRADED_CALL_H_