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1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 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/webrtc/mock_peer_connection_dependency_factory.
h" | 5 #include "content/renderer/media/webrtc/mock_peer_connection_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 "content/renderer/media/webrtc_local_audio_track.h" |
15 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 15 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
16 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 16 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
17 #include "third_party/libjingle/source/talk/base/scoped_ref_ptr.h" | |
18 #include "third_party/libjingle/source/talk/media/base/videocapturer.h" | 17 #include "third_party/libjingle/source/talk/media/base/videocapturer.h" |
| 18 #include "third_party/webrtc/base/scoped_ref_ptr.h" |
19 | 19 |
20 using webrtc::AudioSourceInterface; | 20 using webrtc::AudioSourceInterface; |
21 using webrtc::AudioTrackInterface; | 21 using webrtc::AudioTrackInterface; |
22 using webrtc::AudioTrackVector; | 22 using webrtc::AudioTrackVector; |
23 using webrtc::IceCandidateCollection; | 23 using webrtc::IceCandidateCollection; |
24 using webrtc::IceCandidateInterface; | 24 using webrtc::IceCandidateInterface; |
25 using webrtc::MediaStreamInterface; | 25 using webrtc::MediaStreamInterface; |
26 using webrtc::ObserverInterface; | 26 using webrtc::ObserverInterface; |
27 using webrtc::SessionDescriptionInterface; | 27 using webrtc::SessionDescriptionInterface; |
28 using webrtc::VideoRendererInterface; | 28 using webrtc::VideoRendererInterface; |
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83 } | 83 } |
84 | 84 |
85 AudioTrackVector MockMediaStream::GetAudioTracks() { | 85 AudioTrackVector MockMediaStream::GetAudioTracks() { |
86 return audio_track_vector_; | 86 return audio_track_vector_; |
87 } | 87 } |
88 | 88 |
89 VideoTrackVector MockMediaStream::GetVideoTracks() { | 89 VideoTrackVector MockMediaStream::GetVideoTracks() { |
90 return video_track_vector_; | 90 return video_track_vector_; |
91 } | 91 } |
92 | 92 |
93 talk_base::scoped_refptr<AudioTrackInterface> MockMediaStream::FindAudioTrack( | 93 rtc::scoped_refptr<AudioTrackInterface> MockMediaStream::FindAudioTrack( |
94 const std::string& track_id) { | 94 const std::string& track_id) { |
95 AudioTrackVector::iterator it = FindTrack(&audio_track_vector_, track_id); | 95 AudioTrackVector::iterator it = FindTrack(&audio_track_vector_, track_id); |
96 return it == audio_track_vector_.end() ? NULL : *it; | 96 return it == audio_track_vector_.end() ? NULL : *it; |
97 } | 97 } |
98 | 98 |
99 talk_base::scoped_refptr<VideoTrackInterface> MockMediaStream::FindVideoTrack( | 99 rtc::scoped_refptr<VideoTrackInterface> MockMediaStream::FindVideoTrack( |
100 const std::string& track_id) { | 100 const std::string& track_id) { |
101 VideoTrackVector::iterator it = FindTrack(&video_track_vector_, track_id); | 101 VideoTrackVector::iterator it = FindTrack(&video_track_vector_, track_id); |
102 return it == video_track_vector_.end() ? NULL : *it; | 102 return it == video_track_vector_.end() ? NULL : *it; |
103 } | 103 } |
104 | 104 |
105 void MockMediaStream::RegisterObserver(ObserverInterface* observer) { | 105 void MockMediaStream::RegisterObserver(ObserverInterface* observer) { |
106 DCHECK(observers_.find(observer) == observers_.end()); | 106 DCHECK(observers_.find(observer) == observers_.end()); |
107 observers_.insert(observer); | 107 observers_.insert(observer); |
108 } | 108 } |
109 | 109 |
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436 } | 436 } |
437 | 437 |
438 MockPeerConnectionDependencyFactory::~MockPeerConnectionDependencyFactory() {} | 438 MockPeerConnectionDependencyFactory::~MockPeerConnectionDependencyFactory() {} |
439 | 439 |
440 scoped_refptr<webrtc::PeerConnectionInterface> | 440 scoped_refptr<webrtc::PeerConnectionInterface> |
441 MockPeerConnectionDependencyFactory::CreatePeerConnection( | 441 MockPeerConnectionDependencyFactory::CreatePeerConnection( |
442 const webrtc::PeerConnectionInterface::RTCConfiguration& config, | 442 const webrtc::PeerConnectionInterface::RTCConfiguration& config, |
443 const webrtc::MediaConstraintsInterface* constraints, | 443 const webrtc::MediaConstraintsInterface* constraints, |
444 blink::WebFrame* frame, | 444 blink::WebFrame* frame, |
445 webrtc::PeerConnectionObserver* observer) { | 445 webrtc::PeerConnectionObserver* observer) { |
446 return new talk_base::RefCountedObject<MockPeerConnectionImpl>(this); | 446 return new rtc::RefCountedObject<MockPeerConnectionImpl>(this); |
447 } | 447 } |
448 | 448 |
449 scoped_refptr<webrtc::AudioSourceInterface> | 449 scoped_refptr<webrtc::AudioSourceInterface> |
450 MockPeerConnectionDependencyFactory::CreateLocalAudioSource( | 450 MockPeerConnectionDependencyFactory::CreateLocalAudioSource( |
451 const webrtc::MediaConstraintsInterface* constraints) { | 451 const webrtc::MediaConstraintsInterface* constraints) { |
452 last_audio_source_ = | 452 last_audio_source_ = |
453 new talk_base::RefCountedObject<MockAudioSource>(constraints); | 453 new rtc::RefCountedObject<MockAudioSource>(constraints); |
454 return last_audio_source_; | 454 return last_audio_source_; |
455 } | 455 } |
456 | 456 |
457 WebRtcVideoCapturerAdapter* | 457 WebRtcVideoCapturerAdapter* |
458 MockPeerConnectionDependencyFactory::CreateVideoCapturer( | 458 MockPeerConnectionDependencyFactory::CreateVideoCapturer( |
459 bool is_screen_capture) { | 459 bool is_screen_capture) { |
460 return new MockRtcVideoCapturer(is_screen_capture); | 460 return new MockRtcVideoCapturer(is_screen_capture); |
461 } | 461 } |
462 | 462 |
463 scoped_refptr<webrtc::VideoSourceInterface> | 463 scoped_refptr<webrtc::VideoSourceInterface> |
464 MockPeerConnectionDependencyFactory::CreateVideoSource( | 464 MockPeerConnectionDependencyFactory::CreateVideoSource( |
465 cricket::VideoCapturer* capturer, | 465 cricket::VideoCapturer* capturer, |
466 const blink::WebMediaConstraints& constraints) { | 466 const blink::WebMediaConstraints& constraints) { |
467 last_video_source_ = new talk_base::RefCountedObject<MockVideoSource>(); | 467 last_video_source_ = new rtc::RefCountedObject<MockVideoSource>(); |
468 last_video_source_->SetVideoCapturer(capturer); | 468 last_video_source_->SetVideoCapturer(capturer); |
469 return last_video_source_; | 469 return last_video_source_; |
470 } | 470 } |
471 | 471 |
472 scoped_refptr<WebAudioCapturerSource> | 472 scoped_refptr<WebAudioCapturerSource> |
473 MockPeerConnectionDependencyFactory::CreateWebAudioSource( | 473 MockPeerConnectionDependencyFactory::CreateWebAudioSource( |
474 blink::WebMediaStreamSource* source) { | 474 blink::WebMediaStreamSource* source) { |
475 return NULL; | 475 return NULL; |
476 } | 476 } |
477 | 477 |
478 scoped_refptr<webrtc::MediaStreamInterface> | 478 scoped_refptr<webrtc::MediaStreamInterface> |
479 MockPeerConnectionDependencyFactory::CreateLocalMediaStream( | 479 MockPeerConnectionDependencyFactory::CreateLocalMediaStream( |
480 const std::string& label) { | 480 const std::string& label) { |
481 return new talk_base::RefCountedObject<MockMediaStream>(label); | 481 return new rtc::RefCountedObject<MockMediaStream>(label); |
482 } | 482 } |
483 | 483 |
484 scoped_refptr<webrtc::VideoTrackInterface> | 484 scoped_refptr<webrtc::VideoTrackInterface> |
485 MockPeerConnectionDependencyFactory::CreateLocalVideoTrack( | 485 MockPeerConnectionDependencyFactory::CreateLocalVideoTrack( |
486 const std::string& id, | 486 const std::string& id, |
487 webrtc::VideoSourceInterface* source) { | 487 webrtc::VideoSourceInterface* source) { |
488 scoped_refptr<webrtc::VideoTrackInterface> track( | 488 scoped_refptr<webrtc::VideoTrackInterface> track( |
489 new talk_base::RefCountedObject<MockWebRtcVideoTrack>( | 489 new rtc::RefCountedObject<MockWebRtcVideoTrack>( |
490 id, source)); | 490 id, source)); |
491 return track; | 491 return track; |
492 } | 492 } |
493 | 493 |
494 scoped_refptr<webrtc::VideoTrackInterface> | 494 scoped_refptr<webrtc::VideoTrackInterface> |
495 MockPeerConnectionDependencyFactory::CreateLocalVideoTrack( | 495 MockPeerConnectionDependencyFactory::CreateLocalVideoTrack( |
496 const std::string& id, | 496 const std::string& id, |
497 cricket::VideoCapturer* capturer) { | 497 cricket::VideoCapturer* capturer) { |
498 scoped_refptr<MockVideoSource> source = | 498 scoped_refptr<MockVideoSource> source = |
499 new talk_base::RefCountedObject<MockVideoSource>(); | 499 new rtc::RefCountedObject<MockVideoSource>(); |
500 source->SetVideoCapturer(capturer); | 500 source->SetVideoCapturer(capturer); |
501 | 501 |
502 return | 502 return |
503 new talk_base::RefCountedObject<MockWebRtcVideoTrack>(id, source.get()); | 503 new rtc::RefCountedObject<MockWebRtcVideoTrack>(id, source.get()); |
504 } | 504 } |
505 | 505 |
506 SessionDescriptionInterface* | 506 SessionDescriptionInterface* |
507 MockPeerConnectionDependencyFactory::CreateSessionDescription( | 507 MockPeerConnectionDependencyFactory::CreateSessionDescription( |
508 const std::string& type, | 508 const std::string& type, |
509 const std::string& sdp, | 509 const std::string& sdp, |
510 webrtc::SdpParseError* error) { | 510 webrtc::SdpParseError* error) { |
511 return new MockSessionDescription(type, sdp); | 511 return new MockSessionDescription(type, sdp); |
512 } | 512 } |
513 | 513 |
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532 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, | 532 return WebRtcAudioCapturer::CreateCapturer(-1, device_info, |
533 constraints, NULL, audio_source); | 533 constraints, NULL, audio_source); |
534 } | 534 } |
535 | 535 |
536 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack( | 536 void MockPeerConnectionDependencyFactory::StartLocalAudioTrack( |
537 WebRtcLocalAudioTrack* audio_track) { | 537 WebRtcLocalAudioTrack* audio_track) { |
538 audio_track->Start(); | 538 audio_track->Start(); |
539 } | 539 } |
540 | 540 |
541 } // namespace content | 541 } // namespace content |
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