f06f923ef0
Bug: webrtc:9239 Change-Id: I04f4370f624346bf72c7e4e090b57987b558213b Reviewed-on: https://webrtc-review.googlesource.com/74420 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24396}
118 lines
4.6 KiB
C++
118 lines
4.6 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_PEERCONNECTIONTESTWRAPPER_H_
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#define PC_TEST_PEERCONNECTIONTESTWRAPPER_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/peerconnectioninterface.h"
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#include "api/test/fakeconstraints.h"
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#include "pc/test/fakeaudiocapturemodule.h"
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#include "pc/test/fakevideotrackrenderer.h"
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#include "rtc_base/third_party/sigslot/sigslot.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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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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// 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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const webrtc::FakeConstraints& video_constraints);
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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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const webrtc::FakeConstraints& video_constraints);
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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::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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};
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#endif // PC_TEST_PEERCONNECTIONTESTWRAPPER_H_
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