webrtc_m130/webrtc/api/videosourceinterface.h

/*
 *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef WEBRTC_API_VIDEOSOURCEINTERFACE_H_
#define WEBRTC_API_VIDEOSOURCEINTERFACE_H_

#include "webrtc/api/mediastreaminterface.h"

// TODO(perkj): Remove this file once Chrome build files doesn't depend on it.

#endif  // WEBRTC_API_VIDEOSOURCEINTERFACE_H_
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d 2013-07-10 00:45:36 +00:00			`/*`
Fix license headers in webrtc/api. In addition to the code moved from talk/app/webrtc there were some files in webrtc/api/objctests that still had the libjingle license header. BUG=webrtc:5418 TBR=tkchin@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1680293005 Cr-Commit-Position: refs/heads/master@{#11552} 2016-02-10 07:54:43 -08:00			`* Copyright 2012 The WebRTC project authors. All Rights Reserved.`
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d 2013-07-10 00:45:36 +00:00			`*`
Fix license headers in webrtc/api. In addition to the code moved from talk/app/webrtc there were some files in webrtc/api/objctests that still had the libjingle license header. BUG=webrtc:5418 TBR=tkchin@webrtc.org NOTRY=True Review URL: https://codereview.webrtc.org/1680293005 Cr-Commit-Position: refs/heads/master@{#11552} 2016-02-10 07:54:43 -08:00			`* Use of this source code is governed by a BSD-style license`
			`* that can be found in the LICENSE file in the root of the source`
			`* tree. An additional intellectual property rights grant can be found`
			`* in the file PATENTS. All contributing project authors may`
			`* be found in the AUTHORS file in the root of the source tree.`
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d 2013-07-10 00:45:36 +00:00			`*/`

Move talk/app/webrtc to webrtc/api The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545} 2016-02-10 10:53:12 +01:00			`#ifndef WEBRTC_API_VIDEOSOURCEINTERFACE_H_`
			`#define WEBRTC_API_VIDEOSOURCEINTERFACE_H_`
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d 2013-07-10 00:45:36 +00:00
Move talk/app/webrtc to webrtc/api The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545} 2016-02-10 10:53:12 +01:00			`#include "webrtc/api/mediastreaminterface.h"`
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d 2013-07-10 00:45:36 +00:00
Renamed VideoSourceInterface to VideoTrackSourceInterface. Moved VideoSourceInterface to MediaStreamInterface.h Renamed VideoSourceTest to VideoCapturerTrackSourceTest Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format. BUG=webrtc:5426 TBR=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1770003002 . Cr-Commit-Position: refs/heads/master@{#11893} 2016-03-08 01:27:48 +01:00			`// TODO(perkj): Remove this file once Chrome build files doesn't depend on it.`
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d 2013-07-10 00:45:36 +00:00
Move talk/app/webrtc to webrtc/api The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545} 2016-02-10 10:53:12 +01:00			`#endif // WEBRTC_API_VIDEOSOURCEINTERFACE_H_`