chrome.bluetoothSocket: clean-up Listen functions
[chromium-blink-merge.git] / content / renderer / media / mock_peer_connection_impl.h
blobd563746aeed5e7f236787f031fa0fff89ce682a0
1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
5 #ifndef CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_
6 #define CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_
8 #include <string>
10 #include "base/basictypes.h"
11 #include "base/compiler_specific.h"
12 #include "base/logging.h"
13 #include "base/memory/scoped_ptr.h"
14 #include "testing/gmock/include/gmock/gmock.h"
15 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h"
17 namespace content {
19 class MockPeerConnectionDependencyFactory;
20 class MockStreamCollection;
22 class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface {
23 public:
24 explicit MockPeerConnectionImpl(MockPeerConnectionDependencyFactory* factory);
26 // PeerConnectionInterface implementation.
27 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface>
28 local_streams() OVERRIDE;
29 virtual talk_base::scoped_refptr<webrtc::StreamCollectionInterface>
30 remote_streams() OVERRIDE;
31 virtual bool AddStream(
32 webrtc::MediaStreamInterface* local_stream,
33 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE;
34 virtual void RemoveStream(
35 webrtc::MediaStreamInterface* local_stream) OVERRIDE;
36 virtual talk_base::scoped_refptr<webrtc::DtmfSenderInterface>
37 CreateDtmfSender(webrtc::AudioTrackInterface* track) OVERRIDE;
38 virtual talk_base::scoped_refptr<webrtc::DataChannelInterface>
39 CreateDataChannel(const std::string& label,
40 const webrtc::DataChannelInit* config) OVERRIDE;
42 virtual bool GetStats(webrtc::StatsObserver* observer,
43 webrtc::MediaStreamTrackInterface* track) {
44 return false;
46 virtual bool GetStats(webrtc::StatsObserver* observer,
47 webrtc::MediaStreamTrackInterface* track,
48 StatsOutputLevel level) OVERRIDE;
50 // Set Call this function to make sure next call to GetStats fail.
51 void SetGetStatsResult(bool result) { getstats_result_ = result; }
53 virtual SignalingState signaling_state() OVERRIDE {
54 NOTIMPLEMENTED();
55 return PeerConnectionInterface::kStable;
57 virtual IceState ice_state() OVERRIDE {
58 NOTIMPLEMENTED();
59 return PeerConnectionInterface::kIceNew;
61 virtual IceConnectionState ice_connection_state() OVERRIDE {
62 NOTIMPLEMENTED();
63 return PeerConnectionInterface::kIceConnectionNew;
65 virtual IceGatheringState ice_gathering_state() OVERRIDE {
66 NOTIMPLEMENTED();
67 return PeerConnectionInterface::kIceGatheringNew;
69 virtual void Close() OVERRIDE {
70 NOTIMPLEMENTED();
73 virtual const webrtc::SessionDescriptionInterface* local_description()
74 const OVERRIDE;
75 virtual const webrtc::SessionDescriptionInterface* remote_description()
76 const OVERRIDE;
78 // JSEP01 APIs
79 virtual void CreateOffer(
80 webrtc::CreateSessionDescriptionObserver* observer,
81 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE;
82 virtual void CreateAnswer(
83 webrtc::CreateSessionDescriptionObserver* observer,
84 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE;
85 MOCK_METHOD2(SetLocalDescription,
86 void(webrtc::SetSessionDescriptionObserver* observer,
87 webrtc::SessionDescriptionInterface* desc));
88 void SetLocalDescriptionWorker(
89 webrtc::SetSessionDescriptionObserver* observer,
90 webrtc::SessionDescriptionInterface* desc) ;
91 MOCK_METHOD2(SetRemoteDescription,
92 void(webrtc::SetSessionDescriptionObserver* observer,
93 webrtc::SessionDescriptionInterface* desc));
94 void SetRemoteDescriptionWorker(
95 webrtc::SetSessionDescriptionObserver* observer,
96 webrtc::SessionDescriptionInterface* desc);
97 virtual bool UpdateIce(
98 const IceServers& configuration,
99 const webrtc::MediaConstraintsInterface* constraints) OVERRIDE;
100 virtual bool AddIceCandidate(
101 const webrtc::IceCandidateInterface* candidate) OVERRIDE;
102 virtual void RegisterUMAObserver(webrtc::UMAObserver* observer) OVERRIDE;
104 void AddRemoteStream(webrtc::MediaStreamInterface* stream);
106 const std::string& stream_label() const { return stream_label_; }
107 bool hint_audio() const { return hint_audio_; }
108 bool hint_video() const { return hint_video_; }
109 const std::string& description_sdp() const { return description_sdp_; }
110 const std::string& sdp_mid() const { return sdp_mid_; }
111 int sdp_mline_index() const { return sdp_mline_index_; }
112 const std::string& ice_sdp() const { return ice_sdp_; }
113 webrtc::SessionDescriptionInterface* created_session_description() const {
114 return created_sessiondescription_.get();
116 static const char kDummyOffer[];
117 static const char kDummyAnswer[];
119 protected:
120 virtual ~MockPeerConnectionImpl();
122 private:
123 // Used for creating MockSessionDescription.
124 MockPeerConnectionDependencyFactory* dependency_factory_;
126 std::string stream_label_;
127 talk_base::scoped_refptr<MockStreamCollection> local_streams_;
128 talk_base::scoped_refptr<MockStreamCollection> remote_streams_;
129 scoped_ptr<webrtc::SessionDescriptionInterface> local_desc_;
130 scoped_ptr<webrtc::SessionDescriptionInterface> remote_desc_;
131 scoped_ptr<webrtc::SessionDescriptionInterface> created_sessiondescription_;
132 bool hint_audio_;
133 bool hint_video_;
134 bool getstats_result_;
135 std::string description_sdp_;
136 std::string sdp_mid_;
137 int sdp_mline_index_;
138 std::string ice_sdp_;
140 DISALLOW_COPY_AND_ASSIGN(MockPeerConnectionImpl);
143 } // namespace content
145 #endif // CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_