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174 lines
6.6 KiB
174 lines
6.6 KiB
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef TEST_PEER_SCENARIO_PEER_SCENARIO_CLIENT_H_
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#define TEST_PEER_SCENARIO_PEER_SCENARIO_CLIENT_H_
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#include <functional>
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#include <list>
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "api/peer_connection_interface.h"
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#include "api/test/network_emulation_manager.h"
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#include "api/test/time_controller.h"
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#include "pc/test/frame_generator_capturer_video_track_source.h"
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#include "test/logging/log_writer.h"
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namespace webrtc {
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namespace test {
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// Wrapper for a PeerConnection for use in PeerScenario tests. It's intended to
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// be a minimal wrapper for a peer connection that's simple to use in testing.
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// In particular the constructor hides a lot of the required setup for a peer
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// connection.
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class PeerScenarioClient {
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public:
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struct CallbackHandlers {
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std::vector<std::function<void(PeerConnectionInterface::SignalingState)>>
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on_signaling_change;
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std::vector<std::function<void(rtc::scoped_refptr<DataChannelInterface>)>>
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on_data_channel;
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std::vector<std::function<void()>> on_renegotiation_needed;
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std::vector<
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std::function<void(PeerConnectionInterface::IceConnectionState)>>
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on_standardized_ice_connection_change;
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std::vector<
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std::function<void(PeerConnectionInterface::PeerConnectionState)>>
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on_connection_change;
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std::vector<std::function<void(PeerConnectionInterface::IceGatheringState)>>
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on_ice_gathering_change;
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std::vector<std::function<void(const IceCandidateInterface*)>>
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on_ice_candidate;
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std::vector<std::function<void(const std::string&,
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int,
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const std::string&,
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int,
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const std::string&)>>
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on_ice_candidate_error;
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std::vector<std::function<void(const std::vector<cricket::Candidate>&)>>
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on_ice_candidates_removed;
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std::vector<std::function<void(
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rtc::scoped_refptr<RtpReceiverInterface>,
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&)>>
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on_add_track;
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std::vector<
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std::function<void(rtc::scoped_refptr<RtpTransceiverInterface>)>>
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on_track;
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std::vector<std::function<void(rtc::scoped_refptr<RtpReceiverInterface>)>>
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on_remove_track;
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};
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struct Config {
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// WebRTC only support one audio device that is setup up on construction, so
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// we provide the audio generator configuration here rather than on creation
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// of the tracks. This is unlike video, where multiple capture sources can
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// be used at the same time.
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struct AudioSource {
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int sample_rate = 48000;
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int channels = 1;
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struct PulsedNoise {
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double amplitude = 0.1;
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};
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absl::optional<PulsedNoise> pulsed_noise = PulsedNoise();
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} audio;
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struct Video {
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bool use_fake_codecs = false;
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} video;
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// The created endpoints can be accessed using the map key as |index| in
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// PeerScenarioClient::endpoint(index).
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std::map<int, EmulatedEndpointConfig> endpoints = {
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{0, EmulatedEndpointConfig()}};
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CallbackHandlers handlers;
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PeerConnectionInterface::RTCConfiguration rtc_config;
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Config() { rtc_config.sdp_semantics = SdpSemantics::kUnifiedPlan; }
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};
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struct VideoSendTrackConfig {
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FrameGeneratorCapturerConfig generator;
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bool screencast = false;
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};
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struct AudioSendTrack {
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rtc::scoped_refptr<AudioTrackInterface> track;
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rtc::scoped_refptr<RtpSenderInterface> sender;
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};
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struct VideoSendTrack {
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FrameGeneratorCapturer* capturer;
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FrameGeneratorCapturerVideoTrackSource* source;
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VideoTrackInterface* track;
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RtpSenderInterface* sender;
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};
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PeerScenarioClient(
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NetworkEmulationManager* net,
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rtc::Thread* signaling_thread,
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std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
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Config config);
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PeerConnectionFactoryInterface* factory() { return pc_factory_.get(); }
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PeerConnectionInterface* pc() {
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RTC_DCHECK_RUN_ON(signaling_thread_);
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return peer_connection_.get();
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}
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rtc::Thread* thread() { return signaling_thread_; }
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Clock* clock() { return Clock::GetRealTimeClock(); }
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// Returns the endpoint created from the EmulatedEndpointConfig with the same
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// index in PeerScenarioClient::config.
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EmulatedEndpoint* endpoint(int index = 0);
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AudioSendTrack CreateAudio(std::string track_id,
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cricket::AudioOptions options);
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VideoSendTrack CreateVideo(std::string track_id, VideoSendTrackConfig config);
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void AddVideoReceiveSink(std::string track_id,
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rtc::VideoSinkInterface<VideoFrame>* video_sink);
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CallbackHandlers* handlers() { return &handlers_; }
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// Note that there's no provision for munging SDP as that is deprecated
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// behavior.
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void CreateAndSetSdp(std::function<void(std::string)> offer_handler);
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void SetSdpOfferAndGetAnswer(std::string remote_offer,
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std::function<void(std::string)> answer_handler);
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void SetSdpAnswer(
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std::string remote_answer,
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std::function<void(const SessionDescriptionInterface& answer)>
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done_handler);
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// Adds the given ice candidate when the peer connection is ready.
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void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate);
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private:
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const std::map<int, EmulatedEndpoint*> endpoints_;
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TaskQueueFactory* const task_queue_factory_;
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rtc::Thread* const signaling_thread_;
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const std::unique_ptr<LogWriterFactoryInterface> log_writer_factory_;
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const std::unique_ptr<rtc::Thread> worker_thread_;
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CallbackHandlers handlers_ RTC_GUARDED_BY(signaling_thread_);
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const std::unique_ptr<PeerConnectionObserver> observer_;
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std::map<std::string, std::vector<rtc::VideoSinkInterface<VideoFrame>*>>
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track_id_to_video_sinks_ RTC_GUARDED_BY(signaling_thread_);
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std::list<std::unique_ptr<IceCandidateInterface>> pending_ice_candidates_
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RTC_GUARDED_BY(signaling_thread_);
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rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
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rtc::scoped_refptr<PeerConnectionInterface> peer_connection_
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RTC_GUARDED_BY(signaling_thread_);
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_PEER_SCENARIO_PEER_SCENARIO_CLIENT_H_
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