| // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_MOCK_PEER_CONNECTION_IMPL_H_ |
| #define CONTENT_RENDERER_MEDIA_WEBRTC_MOCK_PEER_CONNECTION_IMPL_H_ |
| |
| #include <memory> |
| #include <string> |
| |
| #include "base/compiler_specific.h" |
| #include "base/logging.h" |
| #include "base/macros.h" |
| #include "base/optional.h" |
| #include "testing/gmock/include/gmock/gmock.h" |
| #include "third_party/webrtc/api/dtls_transport_interface.h" |
| #include "third_party/webrtc/api/peer_connection_interface.h" |
| #include "third_party/webrtc/api/stats/rtc_stats_report.h" |
| |
| namespace content { |
| |
| class MockPeerConnectionDependencyFactory; |
| class MockStreamCollection; |
| |
| class FakeRtpSender : public webrtc::RtpSenderInterface { |
| public: |
| FakeRtpSender(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track, |
| std::vector<std::string> stream_ids); |
| ~FakeRtpSender() override; |
| |
| bool SetTrack(webrtc::MediaStreamTrackInterface* track) override; |
| rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track() const override; |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> dtls_transport() |
| const override; |
| uint32_t ssrc() const override; |
| cricket::MediaType media_type() const override; |
| std::string id() const override; |
| std::vector<std::string> stream_ids() const override; |
| std::vector<webrtc::RtpEncodingParameters> init_send_encodings() |
| const override; |
| webrtc::RtpParameters GetParameters() const override; |
| webrtc::RTCError SetParameters( |
| const webrtc::RtpParameters& parameters) override; |
| rtc::scoped_refptr<webrtc::DtmfSenderInterface> GetDtmfSender() |
| const override; |
| void SetTransport( |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> transport) { |
| transport_ = transport; |
| } |
| |
| private: |
| rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track_; |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> transport_; |
| std::vector<std::string> stream_ids_; |
| }; |
| |
| class FakeRtpReceiver : public webrtc::RtpReceiverInterface { |
| public: |
| FakeRtpReceiver(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track, |
| std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> |
| streams = {}); |
| ~FakeRtpReceiver() override; |
| |
| rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track() const override; |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> dtls_transport() |
| const override; |
| std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> streams() |
| const override; |
| std::vector<std::string> stream_ids() const override; |
| cricket::MediaType media_type() const override; |
| std::string id() const override; |
| webrtc::RtpParameters GetParameters() const override; |
| bool SetParameters(const webrtc::RtpParameters& parameters) override; |
| void SetObserver(webrtc::RtpReceiverObserverInterface* observer) override; |
| std::vector<webrtc::RtpSource> GetSources() const override; |
| void SetTransport( |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> transport) { |
| transport_ = transport; |
| } |
| |
| private: |
| rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track_; |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> transport_; |
| std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> streams_; |
| }; |
| |
| class FakeRtpTransceiver : public webrtc::RtpTransceiverInterface { |
| public: |
| FakeRtpTransceiver( |
| cricket::MediaType media_type, |
| rtc::scoped_refptr<FakeRtpSender> sender, |
| rtc::scoped_refptr<FakeRtpReceiver> receiver, |
| base::Optional<std::string> mid, |
| bool stopped, |
| webrtc::RtpTransceiverDirection direction, |
| base::Optional<webrtc::RtpTransceiverDirection> current_direction); |
| ~FakeRtpTransceiver() override; |
| |
| FakeRtpTransceiver& operator=(const FakeRtpTransceiver& other) = default; |
| |
| cricket::MediaType media_type() const override; |
| absl::optional<std::string> mid() const override; |
| rtc::scoped_refptr<webrtc::RtpSenderInterface> sender() const override; |
| rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver() const override; |
| bool stopped() const override; |
| webrtc::RtpTransceiverDirection direction() const override; |
| void SetDirection(webrtc::RtpTransceiverDirection new_direction) override; |
| absl::optional<webrtc::RtpTransceiverDirection> current_direction() |
| const override; |
| void Stop() override; |
| void SetTransport( |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> transport); |
| |
| private: |
| cricket::MediaType media_type_; |
| rtc::scoped_refptr<FakeRtpSender> sender_; |
| rtc::scoped_refptr<FakeRtpReceiver> receiver_; |
| absl::optional<std::string> mid_; |
| bool stopped_; |
| webrtc::RtpTransceiverDirection direction_; |
| absl::optional<webrtc::RtpTransceiverDirection> current_direction_; |
| }; |
| |
| class FakeDtlsTransport : public webrtc::DtlsTransportInterface { |
| public: |
| FakeDtlsTransport(); |
| rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport() override; |
| webrtc::DtlsTransportInformation Information() override; |
| void RegisterObserver( |
| webrtc::DtlsTransportObserverInterface* observer) override {} |
| void UnregisterObserver() override {} |
| }; |
| |
| // TODO(hbos): The use of fakes and mocks is the wrong approach for testing of |
| // this. It introduces complexity, is error prone (not testing the right thing |
| // and bugs in the mocks). This class is a maintenance burden and should be |
| // removed. https://crbug.com/788659 |
| class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface { |
| public: |
| explicit MockPeerConnectionImpl(MockPeerConnectionDependencyFactory* factory, |
| webrtc::PeerConnectionObserver* observer); |
| |
| // PeerConnectionInterface implementation. |
| rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() |
| override { |
| NOTIMPLEMENTED(); |
| return nullptr; |
| } |
| rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() |
| override { |
| NOTIMPLEMENTED(); |
| return nullptr; |
| } |
| bool AddStream(webrtc::MediaStreamInterface* local_stream) override { |
| NOTIMPLEMENTED(); |
| return false; |
| } |
| void RemoveStream(webrtc::MediaStreamInterface* local_stream) override { |
| NOTIMPLEMENTED(); |
| } |
| webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> AddTrack( |
| rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track, |
| const std::vector<std::string>& stream_ids) override; |
| bool RemoveTrack(webrtc::RtpSenderInterface* sender) override; |
| std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> GetSenders() |
| const override; |
| std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> GetReceivers() |
| const override; |
| rtc::scoped_refptr<webrtc::DataChannelInterface> |
| CreateDataChannel(const std::string& label, |
| const webrtc::DataChannelInit* config) override; |
| bool GetStats(webrtc::StatsObserver* observer, |
| webrtc::MediaStreamTrackInterface* track, |
| StatsOutputLevel level) override; |
| void GetStats(webrtc::RTCStatsCollectorCallback* callback) override; |
| void GetStats( |
| rtc::scoped_refptr<webrtc::RtpSenderInterface> selector, |
| rtc::scoped_refptr<webrtc::RTCStatsCollectorCallback> callback) override; |
| void GetStats( |
| rtc::scoped_refptr<webrtc::RtpReceiverInterface> selector, |
| rtc::scoped_refptr<webrtc::RTCStatsCollectorCallback> callback) override; |
| |
| // Call this function to make sure next call to legacy GetStats fail. |
| void SetGetStatsResult(bool result) { getstats_result_ = result; } |
| // Set the report that |GetStats(RTCStatsCollectorCallback*)| returns. |
| void SetGetStatsReport(webrtc::RTCStatsReport* report); |
| rtc::scoped_refptr<webrtc::DtlsTransportInterface> LookupDtlsTransportByMid( |
| const std::string& mid) override { |
| return nullptr; |
| } |
| |
| SignalingState signaling_state() override { |
| NOTIMPLEMENTED(); |
| return PeerConnectionInterface::kStable; |
| } |
| IceConnectionState ice_connection_state() override { |
| NOTIMPLEMENTED(); |
| return PeerConnectionInterface::kIceConnectionNew; |
| } |
| IceGatheringState ice_gathering_state() override { |
| NOTIMPLEMENTED(); |
| return PeerConnectionInterface::kIceGatheringNew; |
| } |
| |
| bool StartRtcEventLog(rtc::PlatformFile file, |
| int64_t max_size_bytes) override { |
| NOTIMPLEMENTED(); |
| return false; |
| } |
| void StopRtcEventLog() override { NOTIMPLEMENTED(); } |
| |
| MOCK_METHOD0(Close, void()); |
| |
| const webrtc::SessionDescriptionInterface* local_description() const override; |
| const webrtc::SessionDescriptionInterface* remote_description() |
| const override; |
| |
| // JSEP01 APIs |
| void CreateOffer(webrtc::CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| void CreateAnswer(webrtc::CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| MOCK_METHOD2(SetLocalDescription, |
| void(webrtc::SetSessionDescriptionObserver* observer, |
| webrtc::SessionDescriptionInterface* desc)); |
| void SetLocalDescriptionWorker( |
| webrtc::SetSessionDescriptionObserver* observer, |
| webrtc::SessionDescriptionInterface* desc); |
| // TODO(hbos): Remove once no longer mandatory to implement. |
| MOCK_METHOD2(SetRemoteDescription, |
| void(webrtc::SetSessionDescriptionObserver* observer, |
| webrtc::SessionDescriptionInterface* desc)); |
| void SetRemoteDescription( |
| std::unique_ptr<webrtc::SessionDescriptionInterface> desc, |
| rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface> |
| observer) override { |
| SetRemoteDescriptionForMock(&desc, &observer); |
| } |
| // Work-around due to MOCK_METHOD being unable to handle move-only arguments. |
| MOCK_METHOD2( |
| SetRemoteDescriptionForMock, |
| void(std::unique_ptr<webrtc::SessionDescriptionInterface>* desc, |
| rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>* |
| observer)); |
| void SetRemoteDescriptionWorker( |
| webrtc::SetSessionDescriptionObserver* observer, |
| webrtc::SessionDescriptionInterface* desc); |
| bool SetConfiguration(const RTCConfiguration& configuration, |
| webrtc::RTCError* error) override; |
| bool AddIceCandidate(const webrtc::IceCandidateInterface* candidate) override; |
| |
| webrtc::RTCError SetBitrate(const webrtc::BitrateSettings& bitrate) override; |
| |
| void AddRemoteStream(webrtc::MediaStreamInterface* stream); |
| |
| const std::string& stream_label() const { return stream_label_; } |
| bool hint_audio() const { return hint_audio_; } |
| bool hint_video() const { return hint_video_; } |
| const std::string& description_sdp() const { return description_sdp_; } |
| const std::string& sdp_mid() const { return sdp_mid_; } |
| int sdp_mline_index() const { return sdp_mline_index_; } |
| const std::string& ice_sdp() const { return ice_sdp_; } |
| webrtc::SessionDescriptionInterface* created_session_description() const { |
| return created_sessiondescription_.get(); |
| } |
| webrtc::PeerConnectionObserver* observer() { |
| return observer_; |
| } |
| void set_setconfiguration_error_type(webrtc::RTCErrorType error_type) { |
| setconfiguration_error_type_ = error_type; |
| } |
| static const char kDummyOffer[]; |
| static const char kDummyAnswer[]; |
| |
| protected: |
| ~MockPeerConnectionImpl() override; |
| |
| private: |
| // Used for creating MockSessionDescription. |
| MockPeerConnectionDependencyFactory* dependency_factory_; |
| |
| std::string stream_label_; |
| std::vector<std::string> local_stream_ids_; |
| rtc::scoped_refptr<MockStreamCollection> remote_streams_; |
| std::vector<rtc::scoped_refptr<FakeRtpSender>> senders_; |
| std::unique_ptr<webrtc::SessionDescriptionInterface> local_desc_; |
| std::unique_ptr<webrtc::SessionDescriptionInterface> remote_desc_; |
| std::unique_ptr<webrtc::SessionDescriptionInterface> |
| created_sessiondescription_; |
| bool hint_audio_; |
| bool hint_video_; |
| bool getstats_result_; |
| std::string description_sdp_; |
| std::string sdp_mid_; |
| int sdp_mline_index_; |
| std::string ice_sdp_; |
| webrtc::PeerConnectionObserver* observer_; |
| webrtc::RTCErrorType setconfiguration_error_type_ = |
| webrtc::RTCErrorType::NONE; |
| rtc::scoped_refptr<webrtc::RTCStatsReport> stats_report_; |
| |
| DISALLOW_COPY_AND_ASSIGN(MockPeerConnectionImpl); |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_WEBRTC_MOCK_PEER_CONNECTION_IMPL_H_ |