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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h')
-rw-r--r-- | third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h | 135 |
1 files changed, 135 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h b/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h new file mode 100644 index 0000000000..cda3ecb73b --- /dev/null +++ b/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h @@ -0,0 +1,135 @@ +/* + * Copyright 2013 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 PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_ +#define PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_ + +#include <memory> +#include <string> +#include <vector> + +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/audio_options.h" +#include "api/data_channel_interface.h" +#include "api/jsep.h" +#include "api/media_stream_interface.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" +#include "api/rtp_receiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "pc/test/fake_audio_capture_module.h" +#include "pc/test/fake_video_track_renderer.h" +#include "rtc_base/third_party/sigslot/sigslot.h" +#include "rtc_base/thread.h" +#include "test/scoped_key_value_config.h" + +class PeerConnectionTestWrapper + : public webrtc::PeerConnectionObserver, + public webrtc::CreateSessionDescriptionObserver, + public sigslot::has_slots<> { + public: + static void Connect(PeerConnectionTestWrapper* caller, + PeerConnectionTestWrapper* callee); + + PeerConnectionTestWrapper(const std::string& name, + rtc::SocketServer* socket_server, + rtc::Thread* network_thread, + rtc::Thread* worker_thread); + virtual ~PeerConnectionTestWrapper(); + + bool CreatePc( + const webrtc::PeerConnectionInterface::RTCConfiguration& config, + rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory, + rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory); + + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory() + const { + return peer_connection_factory_; + } + webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); } + + rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel( + const std::string& label, + const webrtc::DataChannelInit& init); + + void WaitForNegotiation(); + + // Implements PeerConnectionObserver. + void OnSignalingChange( + webrtc::PeerConnectionInterface::SignalingState new_state) override; + void OnAddTrack( + rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, + const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>& + streams) override; + void OnDataChannel( + rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override; + void OnRenegotiationNeeded() override {} + void OnIceConnectionChange( + webrtc::PeerConnectionInterface::IceConnectionState new_state) override {} + void OnIceGatheringChange( + webrtc::PeerConnectionInterface::IceGatheringState new_state) override {} + void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override; + + // Implements CreateSessionDescriptionObserver. + void OnSuccess(webrtc::SessionDescriptionInterface* desc) override; + void OnFailure(webrtc::RTCError) override {} + + void CreateOffer( + const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options); + void CreateAnswer( + const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options); + void ReceiveOfferSdp(const std::string& sdp); + void ReceiveAnswerSdp(const std::string& sdp); + void AddIceCandidate(const std::string& sdp_mid, + int sdp_mline_index, + const std::string& candidate); + void WaitForCallEstablished(); + void WaitForConnection(); + void WaitForAudio(); + void WaitForVideo(); + void GetAndAddUserMedia(bool audio, + const cricket::AudioOptions& audio_options, + bool video); + + // sigslots + sigslot::signal3<const std::string&, int, const std::string&> + SignalOnIceCandidateReady; + sigslot::signal1<const std::string&> SignalOnSdpReady; + sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel; + + private: + void SetLocalDescription(webrtc::SdpType type, const std::string& sdp); + void SetRemoteDescription(webrtc::SdpType type, const std::string& sdp); + bool CheckForConnection(); + bool CheckForAudio(); + bool CheckForVideo(); + rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia( + bool audio, + const cricket::AudioOptions& audio_options, + bool video); + + webrtc::test::ScopedKeyValueConfig field_trials_; + std::string name_; + rtc::SocketServer* const socket_server_; + rtc::Thread* const network_thread_; + rtc::Thread* const worker_thread_; + webrtc::SequenceChecker pc_thread_checker_; + rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_; + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> + peer_connection_factory_; + rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_; + std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_; + int num_get_user_media_calls_ = 0; + bool pending_negotiation_; +}; + +#endif // PC_TEST_PEER_CONNECTION_TEST_WRAPPER_H_ |