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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_media_stream_dependency_factory.h" | 5 #include "content/renderer/media/mock_media_stream_dependency_factory.h" |
| 6 | 6 |
| 7 #include "base/logging.h" | 7 #include "base/logging.h" |
| 8 #include "base/strings/utf_string_conversions.h" | 8 #include "base/strings/utf_string_conversions.h" |
| 9 #include "content/renderer/media/mock_peer_connection_impl.h" | 9 #include "content/renderer/media/mock_peer_connection_impl.h" |
| 10 #include "content/renderer/media/webaudio_capturer_source.h" | 10 #include "content/renderer/media/webaudio_capturer_source.h" |
| 11 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 11 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| 12 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" | 12 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" |
| 13 #include "content/renderer/media/webrtc_audio_capturer.h" | 13 #include "content/renderer/media/webrtc_audio_capturer.h" |
| 14 #include "content/renderer/media/webrtc_local_audio_track.h" |
| 14 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 15 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
| 15 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 16 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
| 16 #include "third_party/libjingle/source/talk/base/scoped_ref_ptr.h" | 17 #include "third_party/libjingle/source/talk/base/scoped_ref_ptr.h" |
| 17 #include "third_party/libjingle/source/talk/media/base/videocapturer.h" | 18 #include "third_party/libjingle/source/talk/media/base/videocapturer.h" |
| 18 | 19 |
| 19 using webrtc::AudioSourceInterface; | 20 using webrtc::AudioSourceInterface; |
| 20 using webrtc::AudioTrackInterface; | 21 using webrtc::AudioTrackInterface; |
| 21 using webrtc::AudioTrackVector; | 22 using webrtc::AudioTrackVector; |
| 22 using webrtc::IceCandidateCollection; | 23 using webrtc::IceCandidateCollection; |
| 23 using webrtc::IceCandidateInterface; | 24 using webrtc::IceCandidateInterface; |
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| 277 return | 278 return |
| 278 static_cast<MockRtcVideoCapturer*>(capturer_.get())->GetLastFrameHeight(); | 279 static_cast<MockRtcVideoCapturer*>(capturer_.get())->GetLastFrameHeight(); |
| 279 } | 280 } |
| 280 | 281 |
| 281 int MockVideoSource::GetFrameNum() const { | 282 int MockVideoSource::GetFrameNum() const { |
| 282 DCHECK(capturer_); | 283 DCHECK(capturer_); |
| 283 return static_cast<MockRtcVideoCapturer*>(capturer_.get())->GetFrameNum(); | 284 return static_cast<MockRtcVideoCapturer*>(capturer_.get())->GetFrameNum(); |
| 284 } | 285 } |
| 285 | 286 |
| 286 MockWebRtcVideoTrack::MockWebRtcVideoTrack( | 287 MockWebRtcVideoTrack::MockWebRtcVideoTrack( |
| 287 std::string id, | 288 const std::string& id, |
| 288 webrtc::VideoSourceInterface* source) | 289 webrtc::VideoSourceInterface* source) |
| 289 : enabled_(false), | 290 : enabled_(false), |
| 290 id_(id), | 291 id_(id), |
| 291 state_(MediaStreamTrackInterface::kLive), | 292 state_(MediaStreamTrackInterface::kLive), |
| 292 source_(source), | 293 source_(source), |
| 293 observer_(NULL), | 294 observer_(NULL), |
| 294 renderer_(NULL) { | 295 renderer_(NULL) { |
| 295 } | 296 } |
| 296 | 297 |
| 297 MockWebRtcVideoTrack::~MockWebRtcVideoTrack() {} | 298 MockWebRtcVideoTrack::~MockWebRtcVideoTrack() {} |
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| 423 return true; | 424 return true; |
| 424 } | 425 } |
| 425 | 426 |
| 426 private: | 427 private: |
| 427 std::string sdp_mid_; | 428 std::string sdp_mid_; |
| 428 int sdp_mline_index_; | 429 int sdp_mline_index_; |
| 429 std::string sdp_; | 430 std::string sdp_; |
| 430 }; | 431 }; |
| 431 | 432 |
| 432 MockMediaStreamDependencyFactory::MockMediaStreamDependencyFactory() | 433 MockMediaStreamDependencyFactory::MockMediaStreamDependencyFactory() |
| 433 : MediaStreamDependencyFactory(NULL) { | 434 : MediaStreamDependencyFactory(NULL), |
| 435 fail_to_create_next_audio_capturer_(false) { |
| 434 } | 436 } |
| 435 | 437 |
| 436 MockMediaStreamDependencyFactory::~MockMediaStreamDependencyFactory() {} | 438 MockMediaStreamDependencyFactory::~MockMediaStreamDependencyFactory() {} |
| 437 | 439 |
| 438 scoped_refptr<webrtc::PeerConnectionInterface> | 440 scoped_refptr<webrtc::PeerConnectionInterface> |
| 439 MockMediaStreamDependencyFactory::CreatePeerConnection( | 441 MockMediaStreamDependencyFactory::CreatePeerConnection( |
| 440 const webrtc::PeerConnectionInterface::IceServers& ice_servers, | 442 const webrtc::PeerConnectionInterface::IceServers& ice_servers, |
| 441 const webrtc::MediaConstraintsInterface* constraints, | 443 const webrtc::MediaConstraintsInterface* constraints, |
| 442 blink::WebFrame* frame, | 444 blink::WebFrame* frame, |
| 443 webrtc::PeerConnectionObserver* observer) { | 445 webrtc::PeerConnectionObserver* observer) { |
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| 513 MockMediaStreamDependencyFactory::CreateIceCandidate( | 515 MockMediaStreamDependencyFactory::CreateIceCandidate( |
| 514 const std::string& sdp_mid, | 516 const std::string& sdp_mid, |
| 515 int sdp_mline_index, | 517 int sdp_mline_index, |
| 516 const std::string& sdp) { | 518 const std::string& sdp) { |
| 517 return new MockIceCandidate(sdp_mid, sdp_mline_index, sdp); | 519 return new MockIceCandidate(sdp_mid, sdp_mline_index, sdp); |
| 518 } | 520 } |
| 519 | 521 |
| 520 scoped_refptr<WebRtcAudioCapturer> | 522 scoped_refptr<WebRtcAudioCapturer> |
| 521 MockMediaStreamDependencyFactory::CreateAudioCapturer( | 523 MockMediaStreamDependencyFactory::CreateAudioCapturer( |
| 522 int render_view_id, const StreamDeviceInfo& device_info, | 524 int render_view_id, const StreamDeviceInfo& device_info, |
| 523 const blink::WebMediaConstraints& constraints) { | 525 const blink::WebMediaConstraints& constraints, |
| 526 MediaStreamAudioSource* audio_source) { |
| 527 if (fail_to_create_next_audio_capturer_) { |
| 528 fail_to_create_next_audio_capturer_ = false; |
| 529 return NULL; |
| 530 } |
| 531 DCHECK(audio_source); |
| 524 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, | 532 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, |
| 525 constraints, NULL); | 533 constraints, NULL, audio_source); |
| 526 } | 534 } |
| 527 | 535 |
| 528 void MockMediaStreamDependencyFactory::StartLocalAudioTrack( | 536 void MockMediaStreamDependencyFactory::StartLocalAudioTrack( |
| 529 WebRtcLocalAudioTrack* audio_track) { | 537 WebRtcLocalAudioTrack* audio_track) { |
| 530 return; | 538 audio_track->Start(); |
| 531 } | 539 } |
| 532 | 540 |
| 533 } // namespace content | 541 } // namespace content |
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