| // 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_MOCK_PEER_CONNECTION_IMPL_H_ |
| #define CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_ |
| |
| #include <string> |
| |
| #include "base/basictypes.h" |
| #include "base/compiler_specific.h" |
| #include "base/memory/scoped_ptr.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" |
| |
| namespace content { |
| |
| class MockMediaStreamDependencyFactory; |
| class MockStreamCollection; |
| |
| class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface { |
| public: |
| explicit MockPeerConnectionImpl(MockMediaStreamDependencyFactory* factory); |
| |
| // PeerConnectionInterface implementation. |
| virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface> |
| local_streams() OVERRIDE; |
| virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface> |
| remote_streams() OVERRIDE; |
| virtual bool AddStream( |
| webrtc::MediaStreamInterface* local_stream, |
| const webrtc::MediaConstraintsInterface* constraints) OVERRIDE; |
| virtual bool CanSendDtmf(const webrtc::AudioTrackInterface* track) OVERRIDE; |
| virtual bool SendDtmf(const webrtc::AudioTrackInterface* send_track, |
| const std::string& tones, int duration, |
| const webrtc::AudioTrackInterface* play_track) OVERRIDE; |
| virtual void RemoveStream( |
| webrtc::MediaStreamInterface* local_stream) OVERRIDE; |
| virtual ReadyState ready_state() OVERRIDE; |
| virtual bool StartIce(IceOptions options) OVERRIDE; |
| |
| virtual webrtc::SessionDescriptionInterface* CreateOffer( |
| const webrtc::MediaHints& hints) OVERRIDE; |
| virtual webrtc::SessionDescriptionInterface* CreateAnswer( |
| const webrtc::MediaHints& hints, |
| const webrtc::SessionDescriptionInterface* offer) OVERRIDE; |
| virtual bool SetLocalDescription( |
| Action action, |
| webrtc::SessionDescriptionInterface* desc) OVERRIDE; |
| virtual bool SetRemoteDescription( |
| Action action, |
| webrtc::SessionDescriptionInterface* desc) OVERRIDE; |
| virtual bool ProcessIceMessage( |
| const webrtc::IceCandidateInterface* ice_candidate) OVERRIDE; |
| virtual const webrtc::SessionDescriptionInterface* local_description() |
| const OVERRIDE; |
| virtual const webrtc::SessionDescriptionInterface* remote_description() |
| const OVERRIDE; |
| |
| // JSEP01 APIs |
| virtual void CreateOffer( |
| webrtc::CreateSessionDescriptionObserver* observer, |
| const webrtc::MediaConstraintsInterface* constraints) OVERRIDE; |
| virtual void CreateAnswer( |
| webrtc::CreateSessionDescriptionObserver* observer, |
| const webrtc::MediaConstraintsInterface* constraints) OVERRIDE; |
| virtual void SetLocalDescription( |
| webrtc::SetSessionDescriptionObserver* observer, |
| webrtc::SessionDescriptionInterface* desc) OVERRIDE; |
| virtual void SetRemoteDescription( |
| webrtc::SetSessionDescriptionObserver* observer, |
| webrtc::SessionDescriptionInterface* desc) OVERRIDE; |
| virtual bool UpdateIce( |
| const IceServers& configuration, |
| const webrtc::MediaConstraintsInterface* constraints) OVERRIDE; |
| virtual bool AddIceCandidate( |
| const webrtc::IceCandidateInterface* candidate) OVERRIDE; |
| virtual IceState ice_state() OVERRIDE; |
| |
| void AddRemoteStream(webrtc::MediaStreamInterface* stream); |
| void SetReadyState(ReadyState state) { ready_state_ = state; } |
| void SetIceState(IceState state) { ice_state_ = state; } |
| |
| const std::string& stream_label() const { return stream_label_; } |
| bool hint_audio() const { return hint_audio_; } |
| bool hint_video() const { return hint_video_; } |
| Action action() const { return action_; } |
| const std::string& description_sdp() const { return description_sdp_; } |
| IceOptions ice_options() const { return ice_options_; } |
| 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(); |
| } |
| static const char kDummyOffer[]; |
| static const char kDummyAnswer[]; |
| |
| protected: |
| virtual ~MockPeerConnectionImpl(); |
| |
| private: |
| // Used for creating MockSessionDescription. |
| MockMediaStreamDependencyFactory* dependency_factory_; |
| |
| std::string stream_label_; |
| talk_base::scoped_refptr<MockStreamCollection> local_streams_; |
| talk_base::scoped_refptr<MockStreamCollection> remote_streams_; |
| scoped_ptr<webrtc::SessionDescriptionInterface> local_desc_; |
| scoped_ptr<webrtc::SessionDescriptionInterface> remote_desc_; |
| scoped_ptr<webrtc::SessionDescriptionInterface> created_sessiondescription_; |
| bool hint_audio_; |
| bool hint_video_; |
| Action action_; |
| std::string description_sdp_; |
| IceOptions ice_options_; |
| std::string sdp_mid_; |
| int sdp_mline_index_; |
| std::string ice_sdp_; |
| ReadyState ready_state_; |
| IceState ice_state_; |
| |
| DISALLOW_COPY_AND_ASSIGN(MockPeerConnectionImpl); |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_ |