| /* |
| * 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 <stddef.h> |
| #include <stdint.h> |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #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 "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" |
| #include "video/config/video_encoder_config.h" |
| |
| namespace webrtc { |
| class DegradedCall : public Call, private PacketReceiver { |
| public: |
| struct TimeScopedNetworkConfig : public BuiltInNetworkBehaviorConfig { |
| TimeDelta duration = TimeDelta::PlusInfinity(); |
| }; |
| |
| explicit DegradedCall( |
| std::unique_ptr<Call> call, |
| const std::vector<TimeScopedNetworkConfig>& send_configs, |
| const std::vector<TimeScopedNetworkConfig>& 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<FecController> 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> 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, |
| rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive, |
| Clock* clock, |
| std::unique_ptr<NetworkBehaviorInterface> 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_; |
| rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive_; |
| FakeNetworkPipe pipe_; |
| absl::optional<int64_t> next_process_ms_ RTC_GUARDED_BY(&task_queue_); |
| }; |
| |
| class ThreadedPacketReceiver : public PacketReceiver { |
| public: |
| ThreadedPacketReceiver(webrtc::TaskQueueBase* worker_thread, |
| webrtc::TaskQueueBase* network_thread, |
| rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive, |
| PacketReceiver* receiver); |
| ~ThreadedPacketReceiver() override; |
| |
| DeliveryStatus DeliverPacket(MediaType media_type, |
| rtc::CopyOnWriteBuffer packet, |
| int64_t packet_time_us) override; |
| |
| private: |
| webrtc::TaskQueueBase* const worker_thread_; |
| webrtc::TaskQueueBase* const network_thread_; |
| rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive_; |
| webrtc::PacketReceiver* const receiver_; |
| }; |
| |
| // 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> call_; |
| // For cancelling tasks on the network thread when DegradedCall is destroyed |
| rtc::scoped_refptr<PendingTaskSafetyFlag> call_alive_; |
| size_t send_config_index_; |
| const std::vector<TimeScopedNetworkConfig> send_configs_; |
| SimulatedNetwork* send_simulated_network_; |
| std::unique_ptr<FakeNetworkPipeOnTaskQueue> send_pipe_; |
| std::map<AudioSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>> |
| audio_send_transport_adapters_; |
| std::map<VideoSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>> |
| video_send_transport_adapters_; |
| |
| size_t receive_config_index_; |
| const std::vector<TimeScopedNetworkConfig> receive_configs_; |
| SimulatedNetwork* receive_simulated_network_; |
| std::unique_ptr<FakeNetworkPipe> receive_pipe_; |
| std::unique_ptr<ThreadedPacketReceiver> packet_receiver_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_DEGRADED_CALL_H_ |