Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(159)

Side by Side Diff: content/renderer/media/media_stream_dependency_factory.cc

Issue 14247018: Implement WebRTC in Chrome for TV (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: rebase Created 7 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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/media_stream_dependency_factory.h" 5 #include "content/renderer/media/media_stream_dependency_factory.h"
6 6
7 #include <vector> 7 #include <vector>
8 8
9 #include "base/synchronization/waitable_event.h" 9 #include "base/synchronization/waitable_event.h"
10 #include "base/utf_string_conversions.h" 10 #include "base/utf_string_conversions.h"
(...skipping 15 matching lines...) Expand all
26 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSourc e.h" 26 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSourc e.h"
27 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamTrack .h" 27 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamTrack .h"
28 #include "third_party/WebKit/Source/WebKit/chromium/public/WebFrame.h" 28 #include "third_party/WebKit/Source/WebKit/chromium/public/WebFrame.h"
29 29
30 #if defined(USE_OPENSSL) 30 #if defined(USE_OPENSSL)
31 #include "third_party/libjingle/source/talk/base/ssladapter.h" 31 #include "third_party/libjingle/source/talk/base/ssladapter.h"
32 #else 32 #else
33 #include "net/socket/nss_ssl_util.h" 33 #include "net/socket/nss_ssl_util.h"
34 #endif 34 #endif
35 35
36 #if defined(GOOGLE_TV)
37 #include "content/renderer/media/rtc_video_decoder_factory_tv.h"
38 #endif
39
36 namespace content { 40 namespace content {
37 41
38 // Constant constraint keys which disables all audio constraints. 42 // Constant constraint keys which disables all audio constraints.
39 // Only used in combination with WebAudio sources. 43 // Only used in combination with WebAudio sources.
40 struct { 44 struct {
41 const char* key; 45 const char* key;
42 const char* value; 46 const char* value;
43 } const kWebAudioConstraints[] = { 47 } const kWebAudioConstraints[] = {
44 {webrtc::MediaConstraintsInterface::kEchoCancellation, 48 {webrtc::MediaConstraintsInterface::kEchoCancellation,
45 webrtc::MediaConstraintsInterface::kValueFalse}, 49 webrtc::MediaConstraintsInterface::kValueFalse},
(...skipping 406 matching lines...) Expand 10 before | Expand all | Expand 10 after
452 return type == WebKit::WebMediaStreamSource::TypeAudio ? 456 return type == WebKit::WebMediaStreamSource::TypeAudio ?
453 native_stream->RemoveTrack(native_stream->FindAudioTrack(track_id)) : 457 native_stream->RemoveTrack(native_stream->FindAudioTrack(track_id)) :
454 native_stream->RemoveTrack(native_stream->FindVideoTrack(track_id)); 458 native_stream->RemoveTrack(native_stream->FindVideoTrack(track_id));
455 } 459 }
456 460
457 bool MediaStreamDependencyFactory::CreatePeerConnectionFactory() { 461 bool MediaStreamDependencyFactory::CreatePeerConnectionFactory() {
458 DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()"; 462 DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()";
459 if (!pc_factory_) { 463 if (!pc_factory_) {
460 DCHECK(!audio_device_); 464 DCHECK(!audio_device_);
461 audio_device_ = new WebRtcAudioDeviceImpl(); 465 audio_device_ = new WebRtcAudioDeviceImpl();
466
467 cricket::WebRtcVideoDecoderFactory* decoder_factory = NULL;
468 #if defined(GOOGLE_TV)
469 // PeerConnectionFactory will hold the ownership of this
470 // VideoDecoderFactory.
471 decoder_factory = new RTCVideoDecoderFactoryTv(this);
472 #endif
473
462 scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory( 474 scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
463 webrtc::CreatePeerConnectionFactory(worker_thread_, 475 webrtc::CreatePeerConnectionFactory(worker_thread_,
464 signaling_thread_, 476 signaling_thread_,
465 audio_device_)); 477 audio_device_,
478 decoder_factory));
466 if (factory) 479 if (factory)
467 pc_factory_ = factory; 480 pc_factory_ = factory;
468 else 481 else
469 audio_device_ = NULL; 482 audio_device_ = NULL;
470 } 483 }
471 return pc_factory_.get() != NULL; 484 return pc_factory_.get() != NULL;
472 } 485 }
473 486
474 bool MediaStreamDependencyFactory::PeerConnectionFactoryCreated() { 487 bool MediaStreamDependencyFactory::PeerConnectionFactoryCreated() {
475 return pc_factory_.get() != NULL; 488 return pc_factory_.get() != NULL;
(...skipping 271 matching lines...) Expand 10 before | Expand all | Expand 10 after
747 // processed before returning. We wait for the above task to finish before 760 // processed before returning. We wait for the above task to finish before
748 // letting the the function continue to avoid any potential race issues. 761 // letting the the function continue to avoid any potential race issues.
749 chrome_worker_thread_.Stop(); 762 chrome_worker_thread_.Stop();
750 } else { 763 } else {
751 NOTREACHED() << "Worker thread not running."; 764 NOTREACHED() << "Worker thread not running.";
752 } 765 }
753 } 766 }
754 } 767 }
755 768
756 } // namespace content 769 } // namespace content
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698