4c1e7cc19b
This unblocks injecting platform-specific resources, such as power usage signals in Chrome. This CL adds AddAdaptationResource to PeerConnectionInterface and integration tests verifying that if an injected resource is overusing, resolution will soon be reduced. To aid testing, some testing-only classes have been updated. Bug: webrtc:11525 Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31505}
134 lines
5.1 KiB
C++
134 lines
5.1 KiB
C++
/*
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* Copyright 2013 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 PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
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#define PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/audio_codecs/audio_decoder_factory.h"
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#include "api/audio_codecs/audio_encoder_factory.h"
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#include "api/audio_options.h"
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#include "api/data_channel_interface.h"
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#include "api/jsep.h"
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#include "api/media_stream_interface.h"
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#include "api/peer_connection_interface.h"
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#include "api/rtc_error.h"
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#include "api/rtp_receiver_interface.h"
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#include "api/scoped_refptr.h"
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#include "pc/test/fake_audio_capture_module.h"
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#include "pc/test/fake_video_track_renderer.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/thread_checker.h"
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class PeerConnectionTestWrapper
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: public webrtc::PeerConnectionObserver,
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public webrtc::CreateSessionDescriptionObserver,
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public sigslot::has_slots<> {
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public:
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static void Connect(PeerConnectionTestWrapper* caller,
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PeerConnectionTestWrapper* callee);
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PeerConnectionTestWrapper(const std::string& name,
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rtc::Thread* network_thread,
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rtc::Thread* worker_thread);
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virtual ~PeerConnectionTestWrapper();
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bool CreatePc(
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const webrtc::PeerConnectionInterface::RTCConfiguration& config,
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rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory);
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rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory()
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const {
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return peer_connection_factory_;
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}
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webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); }
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rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(
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const std::string& label,
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const webrtc::DataChannelInit& init);
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void WaitForNegotiation();
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// Implements PeerConnectionObserver.
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void OnSignalingChange(
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webrtc::PeerConnectionInterface::SignalingState new_state) override;
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void OnAddTrack(
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rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
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const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
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streams) override;
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void OnDataChannel(
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rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
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void OnRenegotiationNeeded() override {}
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void OnIceConnectionChange(
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webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}
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void OnIceGatheringChange(
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webrtc::PeerConnectionInterface::IceGatheringState new_state) override {}
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void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
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// Implements CreateSessionDescriptionObserver.
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void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
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void OnFailure(webrtc::RTCError) override {}
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void CreateOffer(
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const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
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void CreateAnswer(
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const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options);
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void ReceiveOfferSdp(const std::string& sdp);
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void ReceiveAnswerSdp(const std::string& sdp);
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void AddIceCandidate(const std::string& sdp_mid,
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int sdp_mline_index,
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const std::string& candidate);
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void WaitForCallEstablished();
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void WaitForConnection();
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void WaitForAudio();
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void WaitForVideo();
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void GetAndAddUserMedia(bool audio,
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const cricket::AudioOptions& audio_options,
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bool video);
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// sigslots
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sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
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sigslot::signal3<const std::string&, int, const std::string&>
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SignalOnIceCandidateReady;
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sigslot::signal1<std::string*> SignalOnSdpCreated;
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sigslot::signal1<const std::string&> SignalOnSdpReady;
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sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
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private:
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void SetLocalDescription(webrtc::SdpType type, const std::string& sdp);
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void SetRemoteDescription(webrtc::SdpType type, const std::string& sdp);
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bool CheckForConnection();
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bool CheckForAudio();
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bool CheckForVideo();
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rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
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bool audio,
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const cricket::AudioOptions& audio_options,
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bool video);
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std::string name_;
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rtc::Thread* const network_thread_;
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rtc::Thread* const worker_thread_;
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rtc::ThreadChecker pc_thread_checker_;
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rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
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rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
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peer_connection_factory_;
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rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
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std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
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int num_get_user_media_calls_ = 0;
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bool pending_negotiation_;
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};
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#endif // PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_
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