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| 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #include "content/renderer/media/mock_peer_connection_impl.h" | 5 #include "content/renderer/media/mock_peer_connection_impl.h" |
| 6 | 6 |
| 7 #include <stddef.h> | 7 #include <stddef.h> |
| 8 | 8 |
| 9 #include <vector> | 9 #include <vector> |
| 10 | 10 |
| 11 #include "base/logging.h" | 11 #include "base/logging.h" |
| 12 #include "content/renderer/media/mock_data_channel_impl.h" | 12 #include "content/renderer/media/mock_data_channel_impl.h" |
| 13 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.
h" | 13 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.
h" |
| 14 #include "third_party/webrtc/api/rtpreceiverinterface.h" |
| 15 #include "third_party/webrtc/base/refcountedobject.h" |
| 14 | 16 |
| 15 using testing::_; | 17 using testing::_; |
| 16 using webrtc::AudioTrackInterface; | 18 using webrtc::AudioTrackInterface; |
| 17 using webrtc::CreateSessionDescriptionObserver; | 19 using webrtc::CreateSessionDescriptionObserver; |
| 18 using webrtc::DtmfSenderInterface; | 20 using webrtc::DtmfSenderInterface; |
| 19 using webrtc::DtmfSenderObserverInterface; | 21 using webrtc::DtmfSenderObserverInterface; |
| 20 using webrtc::IceCandidateInterface; | 22 using webrtc::IceCandidateInterface; |
| 21 using webrtc::MediaStreamInterface; | 23 using webrtc::MediaStreamInterface; |
| 22 using webrtc::PeerConnectionInterface; | 24 using webrtc::PeerConnectionInterface; |
| 23 using webrtc::SessionDescriptionInterface; | 25 using webrtc::SessionDescriptionInterface; |
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| 107 ~MockDtmfSender() override {} | 109 ~MockDtmfSender() override {} |
| 108 | 110 |
| 109 private: | 111 private: |
| 110 rtc::scoped_refptr<AudioTrackInterface> track_; | 112 rtc::scoped_refptr<AudioTrackInterface> track_; |
| 111 DtmfSenderObserverInterface* observer_; | 113 DtmfSenderObserverInterface* observer_; |
| 112 std::string tones_; | 114 std::string tones_; |
| 113 int duration_; | 115 int duration_; |
| 114 int inter_tone_gap_; | 116 int inter_tone_gap_; |
| 115 }; | 117 }; |
| 116 | 118 |
| 119 class FakeRtpReceiver |
| 120 : public rtc::RefCountedObject<webrtc::RtpReceiverInterface> { |
| 121 public: |
| 122 FakeRtpReceiver(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track) |
| 123 : track_(track) {} |
| 124 |
| 125 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track() const override { |
| 126 return track_; |
| 127 } |
| 128 |
| 129 cricket::MediaType media_type() const override { |
| 130 NOTIMPLEMENTED(); |
| 131 return cricket::MEDIA_TYPE_AUDIO; |
| 132 } |
| 133 |
| 134 std::string id() const override { |
| 135 NOTIMPLEMENTED(); |
| 136 return ""; |
| 137 } |
| 138 |
| 139 webrtc::RtpParameters GetParameters() const override { |
| 140 NOTIMPLEMENTED(); |
| 141 return webrtc::RtpParameters(); |
| 142 } |
| 143 |
| 144 bool SetParameters(const webrtc::RtpParameters& parameters) override { |
| 145 NOTIMPLEMENTED(); |
| 146 return false; |
| 147 } |
| 148 |
| 149 void SetObserver(webrtc::RtpReceiverObserverInterface* observer) override { |
| 150 NOTIMPLEMENTED(); |
| 151 } |
| 152 |
| 153 private: |
| 154 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track_; |
| 155 }; |
| 156 |
| 117 const char MockPeerConnectionImpl::kDummyOffer[] = "dummy offer"; | 157 const char MockPeerConnectionImpl::kDummyOffer[] = "dummy offer"; |
| 118 const char MockPeerConnectionImpl::kDummyAnswer[] = "dummy answer"; | 158 const char MockPeerConnectionImpl::kDummyAnswer[] = "dummy answer"; |
| 119 | 159 |
| 120 MockPeerConnectionImpl::MockPeerConnectionImpl( | 160 MockPeerConnectionImpl::MockPeerConnectionImpl( |
| 121 MockPeerConnectionDependencyFactory* factory, | 161 MockPeerConnectionDependencyFactory* factory, |
| 122 webrtc::PeerConnectionObserver* observer) | 162 webrtc::PeerConnectionObserver* observer) |
| 123 : dependency_factory_(factory), | 163 : dependency_factory_(factory), |
| 124 local_streams_(new rtc::RefCountedObject<MockStreamCollection>), | 164 local_streams_(new rtc::RefCountedObject<MockStreamCollection>), |
| 125 remote_streams_(new rtc::RefCountedObject<MockStreamCollection>), | 165 remote_streams_(new rtc::RefCountedObject<MockStreamCollection>), |
| 126 hint_audio_(false), | 166 hint_audio_(false), |
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| 161 } | 201 } |
| 162 | 202 |
| 163 rtc::scoped_refptr<DtmfSenderInterface> | 203 rtc::scoped_refptr<DtmfSenderInterface> |
| 164 MockPeerConnectionImpl::CreateDtmfSender(AudioTrackInterface* track) { | 204 MockPeerConnectionImpl::CreateDtmfSender(AudioTrackInterface* track) { |
| 165 if (!track) { | 205 if (!track) { |
| 166 return NULL; | 206 return NULL; |
| 167 } | 207 } |
| 168 return new rtc::RefCountedObject<MockDtmfSender>(track); | 208 return new rtc::RefCountedObject<MockDtmfSender>(track); |
| 169 } | 209 } |
| 170 | 210 |
| 211 std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> |
| 212 MockPeerConnectionImpl::GetReceivers() const { |
| 213 std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> receivers; |
| 214 for (size_t i = 0; i < remote_streams_->count(); ++i) { |
| 215 for (const auto& audio_track : remote_streams_->at(i)->GetAudioTracks()) { |
| 216 receivers.push_back(new FakeRtpReceiver(audio_track)); |
| 217 } |
| 218 for (const auto& video_track : remote_streams_->at(i)->GetVideoTracks()) { |
| 219 receivers.push_back(new FakeRtpReceiver(video_track)); |
| 220 } |
| 221 } |
| 222 return receivers; |
| 223 } |
| 224 |
| 171 rtc::scoped_refptr<webrtc::DataChannelInterface> | 225 rtc::scoped_refptr<webrtc::DataChannelInterface> |
| 172 MockPeerConnectionImpl::CreateDataChannel(const std::string& label, | 226 MockPeerConnectionImpl::CreateDataChannel(const std::string& label, |
| 173 const webrtc::DataChannelInit* config) { | 227 const webrtc::DataChannelInit* config) { |
| 174 return new rtc::RefCountedObject<MockDataChannel>(label, config); | 228 return new rtc::RefCountedObject<MockDataChannel>(label, config); |
| 175 } | 229 } |
| 176 | 230 |
| 177 bool MockPeerConnectionImpl::GetStats( | 231 bool MockPeerConnectionImpl::GetStats( |
| 178 webrtc::StatsObserver* observer, | 232 webrtc::StatsObserver* observer, |
| 179 webrtc::MediaStreamTrackInterface* track, | 233 webrtc::MediaStreamTrackInterface* track, |
| 180 StatsOutputLevel level) { | 234 StatsOutputLevel level) { |
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| 282 sdp_mline_index_ = candidate->sdp_mline_index(); | 336 sdp_mline_index_ = candidate->sdp_mline_index(); |
| 283 return candidate->ToString(&ice_sdp_); | 337 return candidate->ToString(&ice_sdp_); |
| 284 } | 338 } |
| 285 | 339 |
| 286 void MockPeerConnectionImpl::RegisterUMAObserver( | 340 void MockPeerConnectionImpl::RegisterUMAObserver( |
| 287 webrtc::UMAObserver* observer) { | 341 webrtc::UMAObserver* observer) { |
| 288 NOTIMPLEMENTED(); | 342 NOTIMPLEMENTED(); |
| 289 } | 343 } |
| 290 | 344 |
| 291 } // namespace content | 345 } // namespace content |
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