| /* |
| * Copyright 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 TEST_SCENARIO_CALL_CLIENT_H_ |
| #define TEST_SCENARIO_CALL_CLIENT_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/test/time_controller.h" |
| #include "call/call.h" |
| #include "modules/audio_device/include/test_audio_device.h" |
| #include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "test/logging/log_writer.h" |
| #include "test/network/network_emulation.h" |
| #include "test/rtp_header_parser.h" |
| #include "test/scenario/column_printer.h" |
| #include "test/scenario/network_node.h" |
| #include "test/scenario/scenario_config.h" |
| |
| namespace webrtc { |
| |
| namespace test { |
| // Helper class to capture network controller state. |
| class NetworkControleUpdateCache : public NetworkControllerInterface { |
| public: |
| explicit NetworkControleUpdateCache( |
| std::unique_ptr<NetworkControllerInterface> controller); |
| |
| NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override; |
| NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override; |
| NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override; |
| NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override; |
| NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override; |
| NetworkControlUpdate OnSentPacket(SentPacket msg) override; |
| NetworkControlUpdate OnReceivedPacket(ReceivedPacket msg) override; |
| NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override; |
| NetworkControlUpdate OnTargetRateConstraints( |
| TargetRateConstraints msg) override; |
| NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override; |
| NetworkControlUpdate OnTransportPacketsFeedback( |
| TransportPacketsFeedback msg) override; |
| NetworkControlUpdate OnNetworkStateEstimate( |
| NetworkStateEstimate msg) override; |
| |
| NetworkControlUpdate update_state() const; |
| |
| private: |
| NetworkControlUpdate Update(NetworkControlUpdate update); |
| const std::unique_ptr<NetworkControllerInterface> controller_; |
| NetworkControlUpdate update_state_; |
| }; |
| |
| class LoggingNetworkControllerFactory |
| : public NetworkControllerFactoryInterface { |
| public: |
| LoggingNetworkControllerFactory(LogWriterFactoryInterface* log_writer_factory, |
| TransportControllerConfig config); |
| RTC_DISALLOW_COPY_AND_ASSIGN(LoggingNetworkControllerFactory); |
| ~LoggingNetworkControllerFactory(); |
| std::unique_ptr<NetworkControllerInterface> Create( |
| NetworkControllerConfig config) override; |
| TimeDelta GetProcessInterval() const override; |
| // TODO(srte): Consider using the Columnprinter interface for this. |
| void LogCongestionControllerStats(Timestamp at_time); |
| |
| NetworkControlUpdate GetUpdate() const; |
| |
| private: |
| GoogCcDebugFactory goog_cc_factory_; |
| NetworkControllerFactoryInterface* cc_factory_ = nullptr; |
| bool print_cc_state_ = false; |
| NetworkControleUpdateCache* last_controller_ = nullptr; |
| }; |
| |
| struct CallClientFakeAudio { |
| rtc::scoped_refptr<AudioProcessing> apm; |
| rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device; |
| rtc::scoped_refptr<AudioState> audio_state; |
| }; |
| // CallClient represents a participant in a call scenario. It is created by the |
| // Scenario class and is used as sender and receiver when setting up a media |
| // stream session. |
| class CallClient : public EmulatedNetworkReceiverInterface { |
| public: |
| CallClient(TimeController* time_controller, |
| std::unique_ptr<LogWriterFactoryInterface> log_writer_factory, |
| CallClientConfig config); |
| RTC_DISALLOW_COPY_AND_ASSIGN(CallClient); |
| |
| ~CallClient(); |
| ColumnPrinter StatsPrinter(); |
| Call::Stats GetStats(); |
| DataRate send_bandwidth() { |
| return DataRate::BitsPerSec(GetStats().send_bandwidth_bps); |
| } |
| DataRate target_rate() const; |
| DataRate stable_target_rate() const; |
| DataRate padding_rate() const; |
| |
| void OnPacketReceived(EmulatedIpPacket packet) override; |
| std::unique_ptr<RtcEventLogOutput> GetLogWriter(std::string name); |
| |
| // Exposed publicly so that tests can execute tasks such as querying stats |
| // for media streams in the expected runtime environment (essentially what |
| // CallClient does internally for GetStats()). |
| void SendTask(std::function<void()> task); |
| |
| private: |
| friend class Scenario; |
| friend class CallClientPair; |
| friend class SendVideoStream; |
| friend class VideoStreamPair; |
| friend class ReceiveVideoStream; |
| friend class SendAudioStream; |
| friend class ReceiveAudioStream; |
| friend class AudioStreamPair; |
| friend class NetworkNodeTransport; |
| uint32_t GetNextVideoSsrc(); |
| uint32_t GetNextVideoLocalSsrc(); |
| uint32_t GetNextAudioSsrc(); |
| uint32_t GetNextAudioLocalSsrc(); |
| uint32_t GetNextRtxSsrc(); |
| void AddExtensions(std::vector<RtpExtension> extensions); |
| int16_t Bind(EmulatedEndpoint* endpoint); |
| void UnBind(); |
| |
| TimeController* const time_controller_; |
| Clock* clock_; |
| const std::unique_ptr<LogWriterFactoryInterface> log_writer_factory_; |
| std::unique_ptr<RtcEventLog> event_log_; |
| LoggingNetworkControllerFactory network_controller_factory_; |
| CallClientFakeAudio fake_audio_setup_; |
| std::unique_ptr<Call> call_; |
| std::unique_ptr<NetworkNodeTransport> transport_; |
| std::unique_ptr<RtpHeaderParser> const header_parser_; |
| std::vector<std::pair<EmulatedEndpoint*, uint16_t>> endpoints_; |
| |
| int next_video_ssrc_index_ = 0; |
| int next_video_local_ssrc_index_ = 0; |
| int next_rtx_ssrc_index_ = 0; |
| int next_audio_ssrc_index_ = 0; |
| int next_audio_local_ssrc_index_ = 0; |
| std::map<uint32_t, MediaType> ssrc_media_types_; |
| // Defined last so it's destroyed first. |
| TaskQueueForTest task_queue_; |
| |
| const FieldTrialBasedConfig field_trials_; |
| }; |
| |
| class CallClientPair { |
| public: |
| RTC_DISALLOW_COPY_AND_ASSIGN(CallClientPair); |
| ~CallClientPair(); |
| CallClient* first() { return first_; } |
| CallClient* second() { return second_; } |
| std::pair<CallClient*, CallClient*> forward() { return {first(), second()}; } |
| std::pair<CallClient*, CallClient*> reverse() { return {second(), first()}; } |
| |
| private: |
| friend class Scenario; |
| CallClientPair(CallClient* first, CallClient* second) |
| : first_(first), second_(second) {} |
| CallClient* const first_; |
| CallClient* const second_; |
| }; |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // TEST_SCENARIO_CALL_CLIENT_H_ |