| /* |
| * 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 <memory> |
| |
| #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/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 DegradedCall : public Call, private Transport, private PacketReceiver { |
| public: |
| explicit DegradedCall( |
| std::unique_ptr<Call> call, |
| absl::optional<BuiltInNetworkBehaviorConfig> send_config, |
| absl::optional<BuiltInNetworkBehaviorConfig> 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 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> call_; |
| |
| void MediaTransportChange(MediaTransportInterface* media_transport) override; |
| const absl::optional<BuiltInNetworkBehaviorConfig> 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<BuiltInNetworkBehaviorConfig> receive_config_; |
| SimulatedNetwork* receive_simulated_network_; |
| std::unique_ptr<FakeNetworkPipe> receive_pipe_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // CALL_DEGRADED_CALL_H_ |