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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/pc/peer_connection_wrapper.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_wrapper.h')
-rw-r--r-- | third_party/libwebrtc/pc/peer_connection_wrapper.h | 201 |
1 files changed, 201 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/peer_connection_wrapper.h b/third_party/libwebrtc/pc/peer_connection_wrapper.h new file mode 100644 index 0000000000..c503a48099 --- /dev/null +++ b/third_party/libwebrtc/pc/peer_connection_wrapper.h @@ -0,0 +1,201 @@ +/* + * Copyright 2017 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_PEER_CONNECTION_WRAPPER_H_ +#define PC_PEER_CONNECTION_WRAPPER_H_ + +#include <memory> +#include <string> +#include <vector> + +#include "api/data_channel_interface.h" +#include "api/function_view.h" +#include "api/jsep.h" +#include "api/media_stream_interface.h" +#include "api/media_types.h" +#include "api/peer_connection_interface.h" +#include "api/rtc_error.h" +#include "api/rtp_sender_interface.h" +#include "api/rtp_transceiver_interface.h" +#include "api/scoped_refptr.h" +#include "api/stats/rtc_stats_report.h" +#include "pc/test/mock_peer_connection_observers.h" + +namespace webrtc { + +// Class that wraps a PeerConnection so that it is easier to use in unit tests. +// Namely, gives a synchronous API for the event-callback-based API of +// PeerConnection and provides an observer object that stores information from +// PeerConnectionObserver callbacks. +// +// This is intended to be subclassed if additional information needs to be +// stored with the PeerConnection (e.g., fake PeerConnection parameters so that +// tests can be written against those interactions). The base +// PeerConnectionWrapper should only have helper methods that are broadly +// useful. More specific helper methods should be created in the test-specific +// subclass. +// +// The wrapper is intended to be constructed by specialized factory methods on +// a test fixture class then used as a local variable in each test case. +class PeerConnectionWrapper { + public: + // Constructs a PeerConnectionWrapper from the given PeerConnection. + // The given PeerConnectionFactory should be the factory that created the + // PeerConnection and the MockPeerConnectionObserver should be the observer + // that is watching the PeerConnection. + PeerConnectionWrapper( + rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory, + rtc::scoped_refptr<PeerConnectionInterface> pc, + std::unique_ptr<MockPeerConnectionObserver> observer); + virtual ~PeerConnectionWrapper(); + + PeerConnectionFactoryInterface* pc_factory(); + PeerConnectionInterface* pc(); + MockPeerConnectionObserver* observer(); + + // Calls the underlying PeerConnection's CreateOffer method and returns the + // resulting SessionDescription once it is available. If the method call + // failed, null is returned. + std::unique_ptr<SessionDescriptionInterface> CreateOffer( + const PeerConnectionInterface::RTCOfferAnswerOptions& options, + std::string* error_out = nullptr); + // Calls CreateOffer with default options. + std::unique_ptr<SessionDescriptionInterface> CreateOffer(); + // Calls CreateOffer and sets a copy of the offer as the local description. + std::unique_ptr<SessionDescriptionInterface> CreateOfferAndSetAsLocal( + const PeerConnectionInterface::RTCOfferAnswerOptions& options); + // Calls CreateOfferAndSetAsLocal with default options. + std::unique_ptr<SessionDescriptionInterface> CreateOfferAndSetAsLocal(); + + // Calls the underlying PeerConnection's CreateAnswer method and returns the + // resulting SessionDescription once it is available. If the method call + // failed, null is returned. + std::unique_ptr<SessionDescriptionInterface> CreateAnswer( + const PeerConnectionInterface::RTCOfferAnswerOptions& options, + std::string* error_out = nullptr); + // Calls CreateAnswer with the default options. + std::unique_ptr<SessionDescriptionInterface> CreateAnswer(); + // Calls CreateAnswer and sets a copy of the offer as the local description. + std::unique_ptr<SessionDescriptionInterface> CreateAnswerAndSetAsLocal( + const PeerConnectionInterface::RTCOfferAnswerOptions& options); + // Calls CreateAnswerAndSetAsLocal with default options. + std::unique_ptr<SessionDescriptionInterface> CreateAnswerAndSetAsLocal(); + std::unique_ptr<SessionDescriptionInterface> CreateRollback(); + + // Calls the underlying PeerConnection's SetLocalDescription method with the + // given session description and waits for the success/failure response. + // Returns true if the description was successfully set. + bool SetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc, + std::string* error_out = nullptr); + // Calls the underlying PeerConnection's SetRemoteDescription method with the + // given session description and waits for the success/failure response. + // Returns true if the description was successfully set. + bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc, + std::string* error_out = nullptr); + bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc, + RTCError* error_out); + + // Does a round of offer/answer with the local PeerConnectionWrapper + // generating the offer and the given PeerConnectionWrapper generating the + // answer. + // Equivalent to: + // 1. this->CreateOffer(offer_options) + // 2. this->SetLocalDescription(offer) + // 3. answerer->SetRemoteDescription(offer) + // 4. answerer->CreateAnswer(answer_options) + // 5. answerer->SetLocalDescription(answer) + // 6. this->SetRemoteDescription(answer) + // Returns true if all steps succeed, false otherwise. + // Suggested usage: + // ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get())); + bool ExchangeOfferAnswerWith(PeerConnectionWrapper* answerer); + bool ExchangeOfferAnswerWith( + PeerConnectionWrapper* answerer, + const PeerConnectionInterface::RTCOfferAnswerOptions& offer_options, + const PeerConnectionInterface::RTCOfferAnswerOptions& answer_options); + + // The following are wrappers for the underlying PeerConnection's + // AddTransceiver method. They return the result of calling AddTransceiver + // with the given arguments, DCHECKing if there is an error. + rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver( + cricket::MediaType media_type); + rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver( + cricket::MediaType media_type, + const RtpTransceiverInit& init); + rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver( + rtc::scoped_refptr<MediaStreamTrackInterface> track); + rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver( + rtc::scoped_refptr<MediaStreamTrackInterface> track, + const RtpTransceiverInit& init); + + // Returns a new dummy audio track with the given label. + rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack( + const std::string& label); + + // Returns a new dummy video track with the given label. + rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack( + const std::string& label); + + // Wrapper for the underlying PeerConnection's AddTrack method. DCHECKs if + // AddTrack fails. + rtc::scoped_refptr<RtpSenderInterface> AddTrack( + rtc::scoped_refptr<MediaStreamTrackInterface> track, + const std::vector<std::string>& stream_ids = {}); + + rtc::scoped_refptr<RtpSenderInterface> AddTrack( + rtc::scoped_refptr<MediaStreamTrackInterface> track, + const std::vector<std::string>& stream_ids, + const std::vector<RtpEncodingParameters>& init_send_encodings); + + // Calls the underlying PeerConnection's AddTrack method with an audio media + // stream track not bound to any source. + rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack( + const std::string& track_label, + const std::vector<std::string>& stream_ids = {}); + + // Calls the underlying PeerConnection's AddTrack method with a video media + // stream track fed by a FakeVideoTrackSource. + rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack( + const std::string& track_label, + const std::vector<std::string>& stream_ids = {}); + + // Calls the underlying PeerConnection's CreateDataChannel method with default + // initialization parameters. + rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( + const std::string& label); + + // Returns the signaling state of the underlying PeerConnection. + PeerConnectionInterface::SignalingState signaling_state(); + + // Returns true if ICE has finished gathering candidates. + bool IsIceGatheringDone(); + + // Returns true if ICE has established a connection. + bool IsIceConnected(); + + // Calls GetStats() on the underlying PeerConnection and returns the resulting + // report. If GetStats() fails, this method returns null and fails the test. + rtc::scoped_refptr<const RTCStatsReport> GetStats(); + + private: + std::unique_ptr<SessionDescriptionInterface> CreateSdp( + rtc::FunctionView<void(CreateSessionDescriptionObserver*)> fn, + std::string* error_out); + bool SetSdp(rtc::FunctionView<void(SetSessionDescriptionObserver*)> fn, + std::string* error_out); + + rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_; + std::unique_ptr<MockPeerConnectionObserver> observer_; + rtc::scoped_refptr<PeerConnectionInterface> pc_; +}; + +} // namespace webrtc + +#endif // PC_PEER_CONNECTION_WRAPPER_H_ |