diff --git a/talk/app/webrtc/OWNERS b/talk/app/webrtc/OWNERS deleted file mode 100644 index 20a1fdf80d..0000000000 --- a/talk/app/webrtc/OWNERS +++ /dev/null @@ -1,5 +0,0 @@ -glaznev@webrtc.org -juberti@webrtc.org -perkj@webrtc.org -tkchin@webrtc.org -tommi@webrtc.org diff --git a/talk/app/webrtc/objc/RTCAudioTrack+Internal.h b/talk/app/webrtc/objc/RTCAudioTrack+Internal.h index 3d2a9830b0..1ca2b8ff9d 100644 --- a/talk/app/webrtc/objc/RTCAudioTrack+Internal.h +++ b/talk/app/webrtc/objc/RTCAudioTrack+Internal.h @@ -27,7 +27,7 @@ #import "RTCAudioTrack.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" @interface RTCAudioTrack (Internal) diff --git a/talk/app/webrtc/objc/RTCDataChannel+Internal.h b/talk/app/webrtc/objc/RTCDataChannel+Internal.h index 78063f4cca..9e23b7bd34 100644 --- a/talk/app/webrtc/objc/RTCDataChannel+Internal.h +++ b/talk/app/webrtc/objc/RTCDataChannel+Internal.h @@ -27,7 +27,7 @@ #import "RTCDataChannel.h" -#include "talk/app/webrtc/datachannelinterface.h" +#include "webrtc/api/datachannelinterface.h" #include "webrtc/base/scoped_ref_ptr.h" @interface RTCDataBuffer (Internal) diff --git a/talk/app/webrtc/objc/RTCDataChannel.mm b/talk/app/webrtc/objc/RTCDataChannel.mm index fdb5c99a83..ef45fbe77e 100644 --- a/talk/app/webrtc/objc/RTCDataChannel.mm +++ b/talk/app/webrtc/objc/RTCDataChannel.mm @@ -31,7 +31,7 @@ #import "RTCDataChannel+Internal.h" -#include "talk/app/webrtc/datachannelinterface.h" +#include "webrtc/api/datachannelinterface.h" namespace webrtc { diff --git a/talk/app/webrtc/objc/RTCEnumConverter.mm b/talk/app/webrtc/objc/RTCEnumConverter.mm index fa4608a560..f8ab8383e4 100644 --- a/talk/app/webrtc/objc/RTCEnumConverter.mm +++ b/talk/app/webrtc/objc/RTCEnumConverter.mm @@ -27,7 +27,7 @@ #import "RTCEnumConverter.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" @implementation RTCEnumConverter diff --git a/talk/app/webrtc/objc/RTCICECandidate+Internal.h b/talk/app/webrtc/objc/RTCICECandidate+Internal.h index 7c35ceab22..f5b226f836 100644 --- a/talk/app/webrtc/objc/RTCICECandidate+Internal.h +++ b/talk/app/webrtc/objc/RTCICECandidate+Internal.h @@ -27,7 +27,7 @@ #import "RTCICECandidate.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" @interface RTCICECandidate (Internal) diff --git a/talk/app/webrtc/objc/RTCICEServer+Internal.h b/talk/app/webrtc/objc/RTCICEServer+Internal.h index 1bbe864fe3..92c4816968 100644 --- a/talk/app/webrtc/objc/RTCICEServer+Internal.h +++ b/talk/app/webrtc/objc/RTCICEServer+Internal.h @@ -27,7 +27,7 @@ #import "RTCICEServer.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" @interface RTCICEServer (Internal) diff --git a/talk/app/webrtc/objc/RTCMediaConstraints+Internal.h b/talk/app/webrtc/objc/RTCMediaConstraints+Internal.h index ac52a8fd3d..8b03d8926c 100644 --- a/talk/app/webrtc/objc/RTCMediaConstraints+Internal.h +++ b/talk/app/webrtc/objc/RTCMediaConstraints+Internal.h @@ -29,7 +29,7 @@ #import "RTCMediaConstraintsNative.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" @interface RTCMediaConstraints (Internal) diff --git a/talk/app/webrtc/objc/RTCMediaConstraintsNative.h b/talk/app/webrtc/objc/RTCMediaConstraintsNative.h index 558f2ec5c9..6948465807 100644 --- a/talk/app/webrtc/objc/RTCMediaConstraintsNative.h +++ b/talk/app/webrtc/objc/RTCMediaConstraintsNative.h @@ -28,7 +28,7 @@ #ifndef TALK_APP_WEBRTC_OBJC_RTCMEDIACONSTRAINTSNATIVE_H_ #define TALK_APP_WEBRTC_OBJC_RTCMEDIACONSTRAINTSNATIVE_H_ -#include "talk/app/webrtc/mediaconstraintsinterface.h" +#include "webrtc/api/mediaconstraintsinterface.h" namespace webrtc { class RTCMediaConstraintsNative : public MediaConstraintsInterface { diff --git a/talk/app/webrtc/objc/RTCMediaSource+Internal.h b/talk/app/webrtc/objc/RTCMediaSource+Internal.h index f60dc610c6..2620cfd1b9 100644 --- a/talk/app/webrtc/objc/RTCMediaSource+Internal.h +++ b/talk/app/webrtc/objc/RTCMediaSource+Internal.h @@ -27,7 +27,7 @@ #import "RTCMediaSource.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" @interface RTCMediaSource (Internal) diff --git a/talk/app/webrtc/objc/RTCMediaStream+Internal.h b/talk/app/webrtc/objc/RTCMediaStream+Internal.h index c5e2d7897d..2f17a17dd9 100644 --- a/talk/app/webrtc/objc/RTCMediaStream+Internal.h +++ b/talk/app/webrtc/objc/RTCMediaStream+Internal.h @@ -27,7 +27,7 @@ #import "RTCMediaStream.h" -#include "talk/app/webrtc/mediastreamtrack.h" +#include "webrtc/api/mediastreamtrack.h" @interface RTCMediaStream (Internal) diff --git a/talk/app/webrtc/objc/RTCMediaStream.mm b/talk/app/webrtc/objc/RTCMediaStream.mm index 87f838d7df..543a569930 100644 --- a/talk/app/webrtc/objc/RTCMediaStream.mm +++ b/talk/app/webrtc/objc/RTCMediaStream.mm @@ -35,7 +35,7 @@ #import "RTCMediaStreamTrack+Internal.h" #import "RTCVideoTrack+Internal.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" @implementation RTCMediaStream { NSMutableArray* _audioTracks; diff --git a/talk/app/webrtc/objc/RTCMediaStreamTrack+Internal.h b/talk/app/webrtc/objc/RTCMediaStreamTrack+Internal.h index e5383fe8e0..88c83fff3c 100644 --- a/talk/app/webrtc/objc/RTCMediaStreamTrack+Internal.h +++ b/talk/app/webrtc/objc/RTCMediaStreamTrack+Internal.h @@ -27,7 +27,7 @@ #import "RTCMediaStreamTrack.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" @interface RTCMediaStreamTrack (Internal) diff --git a/talk/app/webrtc/objc/RTCPeerConnection+Internal.h b/talk/app/webrtc/objc/RTCPeerConnection+Internal.h index 96d63ab412..136fdf5840 100644 --- a/talk/app/webrtc/objc/RTCPeerConnection+Internal.h +++ b/talk/app/webrtc/objc/RTCPeerConnection+Internal.h @@ -29,7 +29,7 @@ #import "RTCPeerConnectionDelegate.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" @interface RTCPeerConnection (Internal) diff --git a/talk/app/webrtc/objc/RTCPeerConnection.mm b/talk/app/webrtc/objc/RTCPeerConnection.mm index f814f06ad8..0ec61814f3 100644 --- a/talk/app/webrtc/objc/RTCPeerConnection.mm +++ b/talk/app/webrtc/objc/RTCPeerConnection.mm @@ -45,7 +45,7 @@ #import "RTCStatsDelegate.h" #import "RTCStatsReport+Internal.h" -#include "talk/app/webrtc/jsep.h" +#include "webrtc/api/jsep.h" NSString* const kRTCSessionDescriptionDelegateErrorDomain = @"RTCSDPError"; int const kRTCSessionDescriptionDelegateErrorCode = -1; diff --git a/talk/app/webrtc/objc/RTCPeerConnectionFactory+Internal.h b/talk/app/webrtc/objc/RTCPeerConnectionFactory+Internal.h index 5d6fa12711..d7dd3e551d 100644 --- a/talk/app/webrtc/objc/RTCPeerConnectionFactory+Internal.h +++ b/talk/app/webrtc/objc/RTCPeerConnectionFactory+Internal.h @@ -27,7 +27,7 @@ #import "RTCPeerConnectionFactory.h" -#include "talk/app/webrtc/peerconnectionfactory.h" +#include "webrtc/api/peerconnectionfactory.h" #include "webrtc/base/scoped_ptr.h" @interface RTCPeerConnectionFactory () diff --git a/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm b/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm index b7b8966239..3393eca16c 100644 --- a/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm +++ b/talk/app/webrtc/objc/RTCPeerConnectionFactory.mm @@ -46,11 +46,11 @@ #import "RTCVideoSource+Internal.h" #import "RTCVideoTrack+Internal.h" -#include "talk/app/webrtc/audiotrack.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/videosourceinterface.h" -#include "talk/app/webrtc/videotrack.h" +#include "webrtc/api/audiotrack.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/videosourceinterface.h" +#include "webrtc/api/videotrack.h" #include "webrtc/base/logging.h" #include "webrtc/base/ssladapter.h" diff --git a/talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h b/talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h index 5e8dbbf604..ffa01c6fff 100644 --- a/talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h +++ b/talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h @@ -27,7 +27,7 @@ #import "talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" @interface RTCConfiguration () diff --git a/talk/app/webrtc/objc/RTCPeerConnectionObserver.h b/talk/app/webrtc/objc/RTCPeerConnectionObserver.h index 9b981b9307..0ac37b9ac2 100644 --- a/talk/app/webrtc/objc/RTCPeerConnectionObserver.h +++ b/talk/app/webrtc/objc/RTCPeerConnectionObserver.h @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" #import "RTCPeerConnection.h" #import "RTCPeerConnectionDelegate.h" diff --git a/talk/app/webrtc/objc/RTCSessionDescription+Internal.h b/talk/app/webrtc/objc/RTCSessionDescription+Internal.h index 662c538bfb..d552af1676 100644 --- a/talk/app/webrtc/objc/RTCSessionDescription+Internal.h +++ b/talk/app/webrtc/objc/RTCSessionDescription+Internal.h @@ -27,8 +27,8 @@ #import "RTCSessionDescription.h" -#include "talk/app/webrtc/jsep.h" -#include "talk/app/webrtc/webrtcsession.h" +#include "webrtc/api/jsep.h" +#include "webrtc/api/webrtcsession.h" @interface RTCSessionDescription (Internal) diff --git a/talk/app/webrtc/objc/RTCStatsReport+Internal.h b/talk/app/webrtc/objc/RTCStatsReport+Internal.h index 7a4124642c..a595cfd962 100644 --- a/talk/app/webrtc/objc/RTCStatsReport+Internal.h +++ b/talk/app/webrtc/objc/RTCStatsReport+Internal.h @@ -27,7 +27,7 @@ #import "RTCStatsReport.h" -#include "talk/app/webrtc/statstypes.h" +#include "webrtc/api/statstypes.h" @interface RTCStatsReport (Internal) diff --git a/talk/app/webrtc/objc/RTCVideoCapturer+Internal.h b/talk/app/webrtc/objc/RTCVideoCapturer+Internal.h index 10a72e2572..37b2452428 100644 --- a/talk/app/webrtc/objc/RTCVideoCapturer+Internal.h +++ b/talk/app/webrtc/objc/RTCVideoCapturer+Internal.h @@ -27,7 +27,7 @@ #import "RTCVideoCapturer.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/videosourceinterface.h" @interface RTCVideoCapturer (Internal) diff --git a/talk/app/webrtc/objc/RTCVideoRendererAdapter.h b/talk/app/webrtc/objc/RTCVideoRendererAdapter.h index 20a4cf1458..77198608b6 100644 --- a/talk/app/webrtc/objc/RTCVideoRendererAdapter.h +++ b/talk/app/webrtc/objc/RTCVideoRendererAdapter.h @@ -27,7 +27,7 @@ #import "RTCVideoRenderer.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" @interface RTCVideoRendererAdapter : NSObject diff --git a/talk/app/webrtc/objc/RTCVideoRendererAdapter.mm b/talk/app/webrtc/objc/RTCVideoRendererAdapter.mm index cefd567c94..e601420121 100644 --- a/talk/app/webrtc/objc/RTCVideoRendererAdapter.mm +++ b/talk/app/webrtc/objc/RTCVideoRendererAdapter.mm @@ -29,8 +29,8 @@ #error "This file requires ARC support." #endif -#import "RTCVideoRendererAdapter.h" #import "RTCI420Frame+Internal.h" +#import "RTCVideoRendererAdapter.h" namespace webrtc { diff --git a/talk/app/webrtc/objc/RTCVideoSource+Internal.h b/talk/app/webrtc/objc/RTCVideoSource+Internal.h index c6c4a206b6..a13e71ed83 100644 --- a/talk/app/webrtc/objc/RTCVideoSource+Internal.h +++ b/talk/app/webrtc/objc/RTCVideoSource+Internal.h @@ -27,7 +27,7 @@ #import "RTCVideoSource.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/videosourceinterface.h" @interface RTCVideoSource (Internal) diff --git a/talk/app/webrtc/objc/RTCVideoTrack+Internal.h b/talk/app/webrtc/objc/RTCVideoTrack+Internal.h index c9ec382938..84259ea839 100644 --- a/talk/app/webrtc/objc/RTCVideoTrack+Internal.h +++ b/talk/app/webrtc/objc/RTCVideoTrack+Internal.h @@ -27,8 +27,8 @@ #import "RTCVideoTrack.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnectioninterface.h" @class RTCVideoRenderer; diff --git a/talk/app/webrtc/objc/RTCVideoTrack.mm b/talk/app/webrtc/objc/RTCVideoTrack.mm index 24e5edabc8..e7d9f52ea7 100644 --- a/talk/app/webrtc/objc/RTCVideoTrack.mm +++ b/talk/app/webrtc/objc/RTCVideoTrack.mm @@ -31,10 +31,10 @@ #import "RTCVideoTrack+Internal.h" +#import "RTCMediaSource+Internal.h" #import "RTCMediaStreamTrack+Internal.h" #import "RTCPeerConnectionFactory+Internal.h" #import "RTCVideoRendererAdapter.h" -#import "RTCMediaSource+Internal.h" #import "RTCVideoSource+Internal.h" @implementation RTCVideoTrack { diff --git a/talk/build/common.gypi b/talk/build/common.gypi index 028c0066e9..6da1449da2 100644 --- a/talk/build/common.gypi +++ b/talk/build/common.gypi @@ -33,11 +33,6 @@ # TODO(ronghuawu): For now, disable the Chrome plugins, which causes a # flood of chromium-style warnings. 'clang_use_chrome_plugins%': 0, - 'conditions': [ - ['OS=="android" or OS=="linux"', { - 'java_home%': '= 10.7 above is required for ARC. 'targets': [ @@ -141,7 +34,7 @@ 'target_name': 'libjingle_peerconnection_objc', 'type': 'static_library', 'dependencies': [ - 'libjingle_peerconnection', + '<(webrtc_root)/api/api.gyp:libjingle_peerconnection', ], 'sources': [ 'app/webrtc/objc/RTCAudioTrack+Internal.h', @@ -223,7 +116,7 @@ ], }, 'include_dirs': [ - '<(DEPTH)/talk/app/webrtc', + '<(webrtc_root)/webrtc/api', '<(DEPTH)/talk/app/webrtc/objc', '<(DEPTH)/talk/app/webrtc/objc/public', ], @@ -295,7 +188,6 @@ ], }], ], - 'targets': [ { 'target_name': 'libjingle_p2p', @@ -348,86 +240,5 @@ 'session/media/voicechannel.h', ], }, # target libjingle_p2p - { - 'target_name': 'libjingle_peerconnection', - 'type': 'static_library', - 'dependencies': [ - '<(webrtc_root)/base/base.gyp:rtc_base', - '<(webrtc_root)/media/media.gyp:rtc_media', - 'libjingle_p2p', - ], - 'sources': [ - 'app/webrtc/audiotrack.cc', - 'app/webrtc/audiotrack.h', - 'app/webrtc/datachannel.cc', - 'app/webrtc/datachannel.h', - 'app/webrtc/datachannelinterface.h', - 'app/webrtc/dtlsidentitystore.cc', - 'app/webrtc/dtlsidentitystore.h', - 'app/webrtc/dtmfsender.cc', - 'app/webrtc/dtmfsender.h', - 'app/webrtc/dtmfsenderinterface.h', - 'app/webrtc/jsep.h', - 'app/webrtc/jsepicecandidate.cc', - 'app/webrtc/jsepicecandidate.h', - 'app/webrtc/jsepsessiondescription.cc', - 'app/webrtc/jsepsessiondescription.h', - 'app/webrtc/localaudiosource.cc', - 'app/webrtc/localaudiosource.h', - 'app/webrtc/mediaconstraintsinterface.cc', - 'app/webrtc/mediaconstraintsinterface.h', - 'app/webrtc/mediacontroller.cc', - 'app/webrtc/mediacontroller.h', - 'app/webrtc/mediastream.cc', - 'app/webrtc/mediastream.h', - 'app/webrtc/mediastreaminterface.h', - 'app/webrtc/mediastreamobserver.cc', - 'app/webrtc/mediastreamobserver.h', - 'app/webrtc/mediastreamprovider.h', - 'app/webrtc/mediastreamproxy.h', - 'app/webrtc/mediastreamtrack.h', - 'app/webrtc/mediastreamtrackproxy.h', - 'app/webrtc/notifier.h', - 'app/webrtc/peerconnection.cc', - 'app/webrtc/peerconnection.h', - 'app/webrtc/peerconnectionfactory.cc', - 'app/webrtc/peerconnectionfactory.h', - 'app/webrtc/peerconnectionfactoryproxy.h', - 'app/webrtc/peerconnectioninterface.h', - 'app/webrtc/peerconnectionproxy.h', - 'app/webrtc/proxy.h', - 'app/webrtc/remoteaudiosource.cc', - 'app/webrtc/remoteaudiosource.h', - 'app/webrtc/remotevideocapturer.cc', - 'app/webrtc/remotevideocapturer.h', - 'app/webrtc/rtpreceiver.cc', - 'app/webrtc/rtpreceiver.h', - 'app/webrtc/rtpreceiverinterface.h', - 'app/webrtc/rtpsender.cc', - 'app/webrtc/rtpsender.h', - 'app/webrtc/rtpsenderinterface.h', - 'app/webrtc/sctputils.cc', - 'app/webrtc/sctputils.h', - 'app/webrtc/statscollector.cc', - 'app/webrtc/statscollector.h', - 'app/webrtc/statstypes.cc', - 'app/webrtc/statstypes.h', - 'app/webrtc/streamcollection.h', - 'app/webrtc/videosource.cc', - 'app/webrtc/videosource.h', - 'app/webrtc/videosourceinterface.h', - 'app/webrtc/videosourceproxy.h', - 'app/webrtc/videotrack.cc', - 'app/webrtc/videotrack.h', - 'app/webrtc/videotrackrenderers.cc', - 'app/webrtc/videotrackrenderers.h', - 'app/webrtc/webrtcsdp.cc', - 'app/webrtc/webrtcsdp.h', - 'app/webrtc/webrtcsession.cc', - 'app/webrtc/webrtcsession.h', - 'app/webrtc/webrtcsessiondescriptionfactory.cc', - 'app/webrtc/webrtcsessiondescriptionfactory.h', - ], - }, # target libjingle_peerconnection ], } diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp index 96ca4dc129..f45a99c025 100755 --- a/talk/libjingle_tests.gyp +++ b/talk/libjingle_tests.gyp @@ -31,9 +31,9 @@ 'target_name': 'libjingle_p2p_unittest', 'type': 'executable', 'dependencies': [ + '<(webrtc_root)/api/api.gyp:libjingle_peerconnection', '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils', '<(webrtc_root)/webrtc.gyp:rtc_unittest_main', - 'libjingle.gyp:libjingle_peerconnection', 'libjingle.gyp:libjingle_p2p', ], 'include_dirs': [ @@ -65,101 +65,8 @@ }], ], }, # target libjingle_p2p_unittest - { - 'target_name': 'peerconnection_unittests', - 'type': '<(gtest_target_type)', - 'dependencies': [ - '<(DEPTH)/testing/gmock.gyp:gmock', - '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils', - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/webrtc.gyp:rtc_unittest_main', - 'libjingle.gyp:libjingle_p2p', - 'libjingle.gyp:libjingle_peerconnection', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - '<(DEPTH)/testing/gmock/include', - ], - }, - 'sources': [ - 'app/webrtc/datachannel_unittest.cc', - 'app/webrtc/dtlsidentitystore_unittest.cc', - 'app/webrtc/dtmfsender_unittest.cc', - 'app/webrtc/fakemetricsobserver.cc', - 'app/webrtc/fakemetricsobserver.h', - 'app/webrtc/jsepsessiondescription_unittest.cc', - 'app/webrtc/localaudiosource_unittest.cc', - 'app/webrtc/mediastream_unittest.cc', - 'app/webrtc/peerconnection_unittest.cc', - 'app/webrtc/peerconnectionendtoend_unittest.cc', - 'app/webrtc/peerconnectionfactory_unittest.cc', - 'app/webrtc/peerconnectioninterface_unittest.cc', - # 'app/webrtc/peerconnectionproxy_unittest.cc', - 'app/webrtc/remotevideocapturer_unittest.cc', - 'app/webrtc/rtpsenderreceiver_unittest.cc', - 'app/webrtc/statscollector_unittest.cc', - 'app/webrtc/test/fakeaudiocapturemodule.cc', - 'app/webrtc/test/fakeaudiocapturemodule.h', - 'app/webrtc/test/fakeaudiocapturemodule_unittest.cc', - 'app/webrtc/test/fakeconstraints.h', - 'app/webrtc/test/fakedatachannelprovider.h', - 'app/webrtc/test/fakedtlsidentitystore.h', - 'app/webrtc/test/fakeperiodicvideocapturer.h', - 'app/webrtc/test/fakevideotrackrenderer.h', - 'app/webrtc/test/mockpeerconnectionobservers.h', - 'app/webrtc/test/peerconnectiontestwrapper.h', - 'app/webrtc/test/peerconnectiontestwrapper.cc', - 'app/webrtc/test/testsdpstrings.h', - 'app/webrtc/videosource_unittest.cc', - 'app/webrtc/videotrack_unittest.cc', - 'app/webrtc/webrtcsdp_unittest.cc', - 'app/webrtc/webrtcsession_unittest.cc', - ], - 'conditions': [ - ['OS=="android"', { - 'sources': [ - 'app/webrtc/test/androidtestinitializer.cc', - 'app/webrtc/test/androidtestinitializer.h', - ], - 'dependencies': [ - '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', - 'libjingle.gyp:libjingle_peerconnection_jni', - ], - }], - ['OS=="win" and clang==1', { - 'msvs_settings': { - 'VCCLCompilerTool': { - 'AdditionalOptions': [ - # Disable warnings failing when compiling with Clang on Windows. - # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 - '-Wno-unused-function', - ], - }, - }, - }], - ], - }, # target peerconnection_unittests ], 'conditions': [ - ['OS=="android"', { - 'targets': [ - { - 'target_name': 'libjingle_peerconnection_android_unittest', - 'type': 'none', - 'dependencies': [ - 'libjingle.gyp:libjingle_peerconnection_java', - ], - 'variables': { - 'apk_name': 'libjingle_peerconnection_android_unittest', - 'java_in_dir': 'app/webrtc/androidtests', - 'resource_dir': 'app/webrtc/androidtests/res', - 'native_lib_target': 'libjingle_peerconnection_so', - 'is_test_apk': 1, - }, - 'includes': [ '../build/java_apk.gypi' ], - }, - ], # targets - }], # OS=="android" ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', { # The >=10.7 above is required to make ARC link cleanly (e.g. as # opposed to _compile_ cleanly, which the library under test @@ -204,8 +111,8 @@ 'dependencies': [ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils', '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default', - '<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock', '<(webrtc_root)/webrtc_examples.gyp:apprtc_signaling', + '<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock', ], 'sources': [ 'app/webrtc/objctests/mac/main.mm', @@ -221,17 +128,6 @@ }, # target apprtc_signaling_gunit_test ], }], - ['OS=="android"', { - 'targets': [ - { - 'target_name': 'peerconnection_unittests_apk_target', - 'type': 'none', - 'dependencies': [ - '<(DEPTH)/webrtc/build/apk_tests.gyp:peerconnection_unittests_apk', - ], - }, - ], - }], ['test_isolation_mode != "noop"', { 'targets': [ { @@ -247,19 +143,6 @@ 'libjingle_p2p_unittest.isolate', ], }, - { - 'target_name': 'peerconnection_unittests_run', - 'type': 'none', - 'dependencies': [ - 'peerconnection_unittests', - ], - 'includes': [ - 'build/isolate.gypi', - ], - 'sources': [ - 'peerconnection_unittests.isolate', - ], - }, ], }], ], diff --git a/talk/session/media/channelmanager.cc b/talk/session/media/channelmanager.cc index bfecb5897c..8124f28364 100644 --- a/talk/session/media/channelmanager.cc +++ b/talk/session/media/channelmanager.cc @@ -33,7 +33,15 @@ #include -#include "talk/app/webrtc/mediacontroller.h" +#include "talk/session/media/srtpfilter.h" +#include "webrtc/api/mediacontroller.h" +#include "webrtc/base/bind.h" +#include "webrtc/base/common.h" +#include "webrtc/base/logging.h" +#include "webrtc/base/sigslotrepeater.h" +#include "webrtc/base/stringencode.h" +#include "webrtc/base/stringutils.h" +#include "webrtc/base/trace_event.h" #include "webrtc/media/base/capturemanager.h" #include "webrtc/media/base/device.h" #include "webrtc/media/base/hybriddataengine.h" @@ -42,14 +50,6 @@ #ifdef HAVE_SCTP #include "webrtc/media/sctp/sctpdataengine.h" #endif -#include "talk/session/media/srtpfilter.h" -#include "webrtc/base/bind.h" -#include "webrtc/base/common.h" -#include "webrtc/base/logging.h" -#include "webrtc/base/sigslotrepeater.h" -#include "webrtc/base/stringencode.h" -#include "webrtc/base/stringutils.h" -#include "webrtc/base/trace_event.h" namespace cricket { diff --git a/talk/session/media/channelmanager_unittest.cc b/talk/session/media/channelmanager_unittest.cc index 81faad4eb4..98497a34b4 100644 --- a/talk/session/media/channelmanager_unittest.cc +++ b/talk/session/media/channelmanager_unittest.cc @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/fakemediacontroller.h" #include "talk/session/media/channelmanager.h" +#include "webrtc/api/fakemediacontroller.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" #include "webrtc/base/thread.h" diff --git a/webrtc/api/OWNERS b/webrtc/api/OWNERS index cd06158b7f..2a0a6aeae7 100644 --- a/webrtc/api/OWNERS +++ b/webrtc/api/OWNERS @@ -1 +1,6 @@ +pthatcher@webrtc.org +glaznev@webrtc.org +juberti@webrtc.org +perkj@webrtc.org tkchin@webrtc.org +tommi@webrtc.org diff --git a/talk/app/webrtc/androidtests/AndroidManifest.xml b/webrtc/api/androidtests/AndroidManifest.xml similarity index 100% rename from talk/app/webrtc/androidtests/AndroidManifest.xml rename to webrtc/api/androidtests/AndroidManifest.xml diff --git a/talk/app/webrtc/androidtests/OWNERS b/webrtc/api/androidtests/OWNERS similarity index 100% rename from talk/app/webrtc/androidtests/OWNERS rename to webrtc/api/androidtests/OWNERS diff --git a/talk/app/webrtc/androidtests/ant.properties b/webrtc/api/androidtests/ant.properties similarity index 100% rename from talk/app/webrtc/androidtests/ant.properties rename to webrtc/api/androidtests/ant.properties diff --git a/talk/app/webrtc/androidtests/build.xml b/webrtc/api/androidtests/build.xml similarity index 100% rename from talk/app/webrtc/androidtests/build.xml rename to webrtc/api/androidtests/build.xml diff --git a/talk/app/webrtc/androidtests/project.properties b/webrtc/api/androidtests/project.properties similarity index 100% rename from talk/app/webrtc/androidtests/project.properties rename to webrtc/api/androidtests/project.properties diff --git a/talk/app/webrtc/androidtests/res/drawable-hdpi/ic_launcher.png b/webrtc/api/androidtests/res/drawable-hdpi/ic_launcher.png similarity index 100% rename from talk/app/webrtc/androidtests/res/drawable-hdpi/ic_launcher.png rename to webrtc/api/androidtests/res/drawable-hdpi/ic_launcher.png diff --git a/talk/app/webrtc/androidtests/res/drawable-ldpi/ic_launcher.png b/webrtc/api/androidtests/res/drawable-ldpi/ic_launcher.png similarity index 100% rename from talk/app/webrtc/androidtests/res/drawable-ldpi/ic_launcher.png rename to webrtc/api/androidtests/res/drawable-ldpi/ic_launcher.png diff --git a/talk/app/webrtc/androidtests/res/drawable-mdpi/ic_launcher.png b/webrtc/api/androidtests/res/drawable-mdpi/ic_launcher.png similarity index 100% rename from talk/app/webrtc/androidtests/res/drawable-mdpi/ic_launcher.png rename to webrtc/api/androidtests/res/drawable-mdpi/ic_launcher.png diff --git a/talk/app/webrtc/androidtests/res/drawable-xhdpi/ic_launcher.png b/webrtc/api/androidtests/res/drawable-xhdpi/ic_launcher.png similarity index 100% rename from talk/app/webrtc/androidtests/res/drawable-xhdpi/ic_launcher.png rename to webrtc/api/androidtests/res/drawable-xhdpi/ic_launcher.png diff --git a/talk/app/webrtc/androidtests/res/values/strings.xml b/webrtc/api/androidtests/res/values/strings.xml similarity index 100% rename from talk/app/webrtc/androidtests/res/values/strings.xml rename to webrtc/api/androidtests/res/values/strings.xml diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java b/webrtc/api/androidtests/src/org/webrtc/GlRectDrawerTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java rename to webrtc/api/androidtests/src/org/webrtc/GlRectDrawerTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java b/webrtc/api/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java rename to webrtc/api/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/NetworkMonitorTest.java b/webrtc/api/androidtests/src/org/webrtc/NetworkMonitorTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/NetworkMonitorTest.java rename to webrtc/api/androidtests/src/org/webrtc/NetworkMonitorTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/PeerConnectionTest.java b/webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/PeerConnectionTest.java rename to webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/RendererCommonTest.java b/webrtc/api/androidtests/src/org/webrtc/RendererCommonTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/RendererCommonTest.java rename to webrtc/api/androidtests/src/org/webrtc/RendererCommonTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java b/webrtc/api/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java rename to webrtc/api/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java b/webrtc/api/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java rename to webrtc/api/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java b/webrtc/api/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java rename to webrtc/api/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java b/webrtc/api/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java similarity index 100% rename from talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java rename to webrtc/api/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java diff --git a/talk/app/webrtc/androidvideocapturer.cc b/webrtc/api/androidvideocapturer.cc similarity index 98% rename from talk/app/webrtc/androidvideocapturer.cc rename to webrtc/api/androidvideocapturer.cc index 6b0d51c5cc..276067a6c7 100644 --- a/talk/app/webrtc/androidvideocapturer.cc +++ b/webrtc/api/androidvideocapturer.cc @@ -24,9 +24,10 @@ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/androidvideocapturer.h" -#include "talk/app/webrtc/java/jni/native_handle_impl.h" +#include "webrtc/api/androidvideocapturer.h" + +#include "webrtc/api/java/jni/native_handle_impl.h" #include "webrtc/base/common.h" #include "webrtc/base/json.h" #include "webrtc/base/timeutils.h" diff --git a/talk/app/webrtc/androidvideocapturer.h b/webrtc/api/androidvideocapturer.h similarity index 96% rename from talk/app/webrtc/androidvideocapturer.h rename to webrtc/api/androidvideocapturer.h index c6022f7c0c..cba522480c 100644 --- a/talk/app/webrtc/androidvideocapturer.h +++ b/webrtc/api/androidvideocapturer.h @@ -24,8 +24,9 @@ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_ANDROIDVIDEOCAPTURER_H_ -#define TALK_APP_WEBRTC_ANDROIDVIDEOCAPTURER_H_ + +#ifndef WEBRTC_API_ANDROIDVIDEOCAPTURER_H_ +#define WEBRTC_API_ANDROIDVIDEOCAPTURER_H_ #include #include @@ -105,4 +106,4 @@ class AndroidVideoCapturer : public cricket::VideoCapturer { } // namespace webrtc -#endif // TALK_APP_WEBRTC_ANDROIDVIDEOCAPTURER_H_ +#endif // WEBRTC_API_ANDROIDVIDEOCAPTURER_H_ diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp index ac4ea844e6..fe7cd0eacc 100644 --- a/webrtc/api/api.gyp +++ b/webrtc/api/api.gyp @@ -9,6 +9,131 @@ { 'includes': [ '../build/common.gypi', ], 'conditions': [ + ['os_posix == 1 and OS != "mac" and OS != "ios"', { + 'conditions': [ + ['sysroot!=""', { + 'variables': { + 'pkg-config': '../../../build/linux/pkg-config-wrapper "<(sysroot)" "<(target_arch)"', + }, + }, { + 'variables': { + 'pkg-config': 'pkg-config' + }, + }], + ], + }], + ['OS=="android"', { + 'targets': [ + { + 'target_name': 'libjingle_peerconnection_jni', + 'type': 'static_library', + 'dependencies': [ + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default', + 'libjingle_peerconnection', + ], + 'sources': [ + 'androidvideocapturer.cc', + 'androidvideocapturer.h', + 'java/jni/androidmediacodeccommon.h', + 'java/jni/androidmediadecoder_jni.cc', + 'java/jni/androidmediadecoder_jni.h', + 'java/jni/androidmediaencoder_jni.cc', + 'java/jni/androidmediaencoder_jni.h', + 'java/jni/androidnetworkmonitor_jni.cc', + 'java/jni/androidnetworkmonitor_jni.h', + 'java/jni/androidvideocapturer_jni.cc', + 'java/jni/androidvideocapturer_jni.h', + 'java/jni/eglbase_jni.cc', + 'java/jni/eglbase_jni.h', + 'java/jni/surfacetexturehelper_jni.cc', + 'java/jni/surfacetexturehelper_jni.h', + 'java/jni/classreferenceholder.cc', + 'java/jni/classreferenceholder.h', + 'java/jni/jni_helpers.cc', + 'java/jni/jni_helpers.h', + 'java/jni/native_handle_impl.cc', + 'java/jni/native_handle_impl.h', + 'java/jni/peerconnection_jni.cc', + ], + 'include_dirs': [ + '<(libyuv_dir)/include', + ], + # TODO(kjellander): Make the code compile without disabling these flags. + # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307 + 'cflags': [ + '-Wno-sign-compare', + '-Wno-unused-variable', + ], + 'cflags!': [ + '-Wextra', + ], + 'cflags_cc!': [ + '-Wnon-virtual-dtor', + '-Woverloaded-virtual', + ], + 'msvs_disabled_warnings': [ + 4245, # conversion from 'int' to 'size_t', signed/unsigned mismatch. + 4267, # conversion from 'size_t' to 'int', possible loss of data. + 4389, # signed/unsigned mismatch. + ], + 'conditions': [ + ['build_json==1', { + 'dependencies': [ + '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp', + ], + 'export_dependent_settings': [ + '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp', + ], + }], + ], + }, + { + 'target_name': 'libjingle_peerconnection_so', + 'type': 'shared_library', + 'dependencies': [ + 'libjingle_peerconnection', + 'libjingle_peerconnection_jni', + ], + 'sources': [ + 'java/jni/jni_onload.cc', + ], + 'variables': { + # This library uses native JNI exports; tell GYP so that the + # required symbols will be kept. + 'use_native_jni_exports': 1, + }, + }, + { + # |libjingle_peerconnection_java| builds a jar file with name + # libjingle_peerconnection_java.jar using Chrome's build system. + # It includes all Java files needed to setup a PeeerConnection call + # from Android. + 'target_name': 'libjingle_peerconnection_java', + 'type': 'none', + 'dependencies': [ + 'libjingle_peerconnection_so', + ], + 'variables': { + # Designate as Chromium code and point to our lint settings to + # enable linting of the WebRTC code (this is the only way to make + # lint_action invoke the Android linter). + 'android_manifest_path': '<(webrtc_root)/build/android/AndroidManifest.xml', + 'suppressions_file': '<(webrtc_root)/build/android/suppressions.xml', + 'chromium_code': 1, + 'java_in_dir': 'java', + 'webrtc_base_dir': '<(webrtc_root)/base', + 'webrtc_modules_dir': '<(webrtc_root)/modules', + 'additional_src_dirs' : [ + 'java/android', + '<(webrtc_base_dir)/java/src', + '<(webrtc_modules_dir)/audio_device/android/java/src', + '<(webrtc_modules_dir)/video_render/android/java/src', + ], + }, + 'includes': ['../../build/java.gypi'], + }, # libjingle_peerconnection_java + ] + }], ['OS=="ios"', { 'targets': [ { @@ -16,7 +141,7 @@ 'type': 'static_library', 'dependencies': [ '<(webrtc_root)/base/base.gyp:rtc_base_objc', - '../../talk/libjingle.gyp:libjingle_peerconnection', + 'libjingle_peerconnection', ], 'sources': [ 'objc/RTCAVFoundationVideoSource+Private.h', @@ -117,6 +242,128 @@ }, } ], - }], # OS=="ios" - ], + }], # OS=="ios" + ], # conditions + 'targets': [ + { + 'target_name': 'libjingle_peerconnection', + 'type': 'static_library', + 'dependencies': [ + '<(webrtc_root)/media/media.gyp:rtc_media', + '../../talk/libjingle.gyp:libjingle_p2p', + ], + 'sources': [ + 'audiotrack.cc', + 'audiotrack.h', + 'datachannel.cc', + 'datachannel.h', + 'datachannelinterface.h', + 'dtlsidentitystore.cc', + 'dtlsidentitystore.h', + 'dtmfsender.cc', + 'dtmfsender.h', + 'dtmfsenderinterface.h', + 'jsep.h', + 'jsepicecandidate.cc', + 'jsepicecandidate.h', + 'jsepsessiondescription.cc', + 'jsepsessiondescription.h', + 'localaudiosource.cc', + 'localaudiosource.h', + 'mediaconstraintsinterface.cc', + 'mediaconstraintsinterface.h', + 'mediacontroller.cc', + 'mediacontroller.h', + 'mediastream.cc', + 'mediastream.h', + 'mediastreaminterface.h', + 'mediastreamobserver.cc', + 'mediastreamobserver.h', + 'mediastreamprovider.h', + 'mediastreamproxy.h', + 'mediastreamtrack.h', + 'mediastreamtrackproxy.h', + 'notifier.h', + 'peerconnection.cc', + 'peerconnection.h', + 'peerconnectionfactory.cc', + 'peerconnectionfactory.h', + 'peerconnectionfactoryproxy.h', + 'peerconnectioninterface.h', + 'peerconnectionproxy.h', + 'proxy.h', + 'remoteaudiosource.cc', + 'remoteaudiosource.h', + 'remotevideocapturer.cc', + 'remotevideocapturer.h', + 'rtpreceiver.cc', + 'rtpreceiver.h', + 'rtpreceiverinterface.h', + 'rtpsender.cc', + 'rtpsender.h', + 'rtpsenderinterface.h', + 'sctputils.cc', + 'sctputils.h', + 'statscollector.cc', + 'statscollector.h', + 'statstypes.cc', + 'statstypes.h', + 'streamcollection.h', + 'videosource.cc', + 'videosource.h', + 'videosourceinterface.h', + 'videosourceproxy.h', + 'videotrack.cc', + 'videotrack.h', + 'videotrackrenderers.cc', + 'videotrackrenderers.h', + 'webrtcsdp.cc', + 'webrtcsdp.h', + 'webrtcsession.cc', + 'webrtcsession.h', + 'webrtcsessiondescriptionfactory.cc', + 'webrtcsessiondescriptionfactory.h', + ], + # TODO(kjellander): Make the code compile without disabling these flags. + # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307 + 'cflags': [ + '-Wno-sign-compare', + ], + 'cflags_cc!': [ + '-Wnon-virtual-dtor', + '-Woverloaded-virtual', + ], + 'conditions': [ + ['clang==1', { + 'cflags!': [ + '-Wextra', + ], + 'xcode_settings': { + 'WARNING_CFLAGS!': ['-Wextra'], + }, + }, { + 'cflags': [ + '-Wno-maybe-uninitialized', # Only exists for GCC. + ], + }], + ['OS=="win"', { + # Disable warning for signed/unsigned mismatch. + 'msvs_settings': { + 'VCCLCompilerTool': { + 'AdditionalOptions!': ['/we4389'], + }, + }, + }], + ['OS=="win" and clang==1', { + 'msvs_settings': { + 'VCCLCompilerTool': { + 'AdditionalOptions': [ + '-Wno-sign-compare', + ], + }, + }, + }], + ], + }, # target libjingle_peerconnection + ], # targets } diff --git a/webrtc/api/api_tests.gyp b/webrtc/api/api_tests.gyp index cdb23fb290..31bc6999ff 100644 --- a/webrtc/api/api_tests.gyp +++ b/webrtc/api/api_tests.gyp @@ -8,7 +8,135 @@ { 'includes': [ '../build/common.gypi', ], + 'targets': [ + { + 'target_name': 'peerconnection_unittests', + 'type': '<(gtest_target_type)', + 'dependencies': [ + '<(DEPTH)/testing/gmock.gyp:gmock', + '<(webrtc_root)/api/api.gyp:libjingle_peerconnection', + '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils', + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/webrtc.gyp:rtc_unittest_main', + '../../talk/libjingle.gyp:libjingle_p2p', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + '<(DEPTH)/testing/gmock/include', + ], + }, + 'defines': [ + # Feature selection. + 'HAVE_SCTP', + ], + 'sources': [ + 'datachannel_unittest.cc', + 'dtlsidentitystore_unittest.cc', + 'dtmfsender_unittest.cc', + 'fakemetricsobserver.cc', + 'fakemetricsobserver.h', + 'jsepsessiondescription_unittest.cc', + 'localaudiosource_unittest.cc', + 'mediastream_unittest.cc', + 'peerconnection_unittest.cc', + 'peerconnectionendtoend_unittest.cc', + 'peerconnectionfactory_unittest.cc', + 'peerconnectioninterface_unittest.cc', + # 'peerconnectionproxy_unittest.cc', + 'remotevideocapturer_unittest.cc', + 'rtpsenderreceiver_unittest.cc', + 'statscollector_unittest.cc', + 'test/fakeaudiocapturemodule.cc', + 'test/fakeaudiocapturemodule.h', + 'test/fakeaudiocapturemodule_unittest.cc', + 'test/fakeconstraints.h', + 'test/fakedatachannelprovider.h', + 'test/fakedtlsidentitystore.h', + 'test/fakeperiodicvideocapturer.h', + 'test/fakevideotrackrenderer.h', + 'test/mockpeerconnectionobservers.h', + 'test/peerconnectiontestwrapper.h', + 'test/peerconnectiontestwrapper.cc', + 'test/testsdpstrings.h', + 'videosource_unittest.cc', + 'videotrack_unittest.cc', + 'webrtcsdp_unittest.cc', + 'webrtcsession_unittest.cc', + ], + # TODO(kjellander): Make the code compile without disabling these flags. + # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307 + 'cflags': [ + '-Wno-sign-compare', + ], + 'cflags!': [ + '-Wextra', + ], + 'cflags_cc!': [ + '-Wnon-virtual-dtor', + '-Woverloaded-virtual', + ], + 'msvs_disabled_warnings': [ + 4245, # conversion from 'int' to 'size_t', signed/unsigned mismatch. + 4267, # conversion from 'size_t' to 'int', possible loss of data. + 4389, # signed/unsigned mismatch. + ], + 'conditions': [ + ['clang==1', { + # TODO(kjellander): Make the code compile without disabling these flags. + # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307 + 'cflags!': [ + '-Wextra', + ], + 'xcode_settings': { + 'WARNING_CFLAGS!': ['-Wextra'], + }, + }], + ['OS=="android"', { + 'sources': [ + 'test/androidtestinitializer.cc', + 'test/androidtestinitializer.h', + ], + 'dependencies': [ + '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', + '<(webrtc_root)/api/api.gyp:libjingle_peerconnection_jni', + ], + }], + ['OS=="win" and clang==1', { + 'msvs_settings': { + 'VCCLCompilerTool': { + 'AdditionalOptions': [ + # Disable warnings failing when compiling with Clang on Windows. + # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366 + '-Wno-sign-compare', + '-Wno-unused-function', + ], + }, + }, + }], + ], # conditions + }, # target peerconnection_unittests + ], # targets 'conditions': [ + ['OS=="android"', { + 'targets': [ + { + 'target_name': 'libjingle_peerconnection_android_unittest', + 'type': 'none', + 'dependencies': [ + '<(webrtc_root)/api/api.gyp:libjingle_peerconnection_java', + ], + 'variables': { + 'apk_name': 'libjingle_peerconnection_android_unittest', + 'java_in_dir': 'androidtests', + 'resource_dir': 'androidtests/res', + 'native_lib_target': 'libjingle_peerconnection_so', + 'is_test_apk': 1, + 'never_lint': 1, + }, + 'includes': [ '../../build/java_apk.gypi' ], + }, + ], # targets + }], # OS=="android" ['OS=="ios"', { 'targets': [ { @@ -35,8 +163,36 @@ # https://developer.apple.com/library/mac/qa/qa1490/_index.html 'OTHER_LDFLAGS': ['-ObjC'], }, - } + }, ], - }], # OS=="ios" - ], + }], # OS=="ios" + ['OS=="android"', { + 'targets': [ + { + 'target_name': 'peerconnection_unittests_apk_target', + 'type': 'none', + 'dependencies': [ + '<(apk_tests_path):peerconnection_unittests_apk', + ], + }, + ], + }], # OS=="android" + ['test_isolation_mode != "noop"', { + 'targets': [ + { + 'target_name': 'peerconnection_unittests_run', + 'type': 'none', + 'dependencies': [ + 'peerconnection_unittests', + ], + 'includes': [ + '../build/isolate.gypi', + ], + 'sources': [ + 'peerconnection_unittests.isolate', + ], + }, + ], # targets + }], # test_isolation_mode != "noop" + ], # conditions } diff --git a/talk/app/webrtc/audiotrack.cc b/webrtc/api/audiotrack.cc similarity index 97% rename from talk/app/webrtc/audiotrack.cc rename to webrtc/api/audiotrack.cc index b3223cd29f..1ae0535936 100644 --- a/talk/app/webrtc/audiotrack.cc +++ b/webrtc/api/audiotrack.cc @@ -1,6 +1,6 @@ /* * libjingle - * Copyright 2004--2011 Google Inc. + * Copyright 2011 Google Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/audiotrack.h" +#include "webrtc/api/audiotrack.h" #include "webrtc/base/checks.h" diff --git a/talk/app/webrtc/audiotrack.h b/webrtc/api/audiotrack.h similarity index 91% rename from talk/app/webrtc/audiotrack.h rename to webrtc/api/audiotrack.h index 55f4837714..87fc41f43f 100644 --- a/talk/app/webrtc/audiotrack.h +++ b/webrtc/api/audiotrack.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_AUDIOTRACK_H_ -#define TALK_APP_WEBRTC_AUDIOTRACK_H_ +#ifndef WEBRTC_API_AUDIOTRACK_H_ +#define WEBRTC_API_AUDIOTRACK_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/mediastreamtrack.h" -#include "talk/app/webrtc/notifier.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/mediastreamtrack.h" +#include "webrtc/api/notifier.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/base/thread_checker.h" @@ -73,4 +73,4 @@ class AudioTrack : public MediaStreamTrack, } // namespace webrtc -#endif // TALK_APP_WEBRTC_AUDIOTRACK_H_ +#endif // WEBRTC_API_AUDIOTRACK_H_ diff --git a/talk/app/webrtc/datachannel.cc b/webrtc/api/datachannel.cc similarity index 99% rename from talk/app/webrtc/datachannel.cc rename to webrtc/api/datachannel.cc index 05fa5ec71b..855831a0ac 100644 --- a/talk/app/webrtc/datachannel.cc +++ b/webrtc/api/datachannel.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/datachannel.h" +#include "webrtc/api/datachannel.h" #include -#include "talk/app/webrtc/mediastreamprovider.h" -#include "talk/app/webrtc/sctputils.h" +#include "webrtc/api/mediastreamprovider.h" +#include "webrtc/api/sctputils.h" #include "webrtc/base/logging.h" #include "webrtc/base/refcount.h" #include "webrtc/media/sctp/sctpdataengine.h" diff --git a/talk/app/webrtc/datachannel.h b/webrtc/api/datachannel.h similarity index 98% rename from talk/app/webrtc/datachannel.h rename to webrtc/api/datachannel.h index 00e088635d..649cb24088 100644 --- a/talk/app/webrtc/datachannel.h +++ b/webrtc/api/datachannel.h @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_DATACHANNEL_H_ -#define TALK_APP_WEBRTC_DATACHANNEL_H_ +#ifndef WEBRTC_API_DATACHANNEL_H_ +#define WEBRTC_API_DATACHANNEL_H_ #include #include #include -#include "talk/app/webrtc/datachannelinterface.h" -#include "talk/app/webrtc/proxy.h" #include "talk/session/media/channel.h" +#include "webrtc/api/datachannelinterface.h" +#include "webrtc/api/proxy.h" #include "webrtc/base/messagehandler.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/base/sigslot.h" @@ -296,4 +296,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_DATACHANNEL_H_ +#endif // WEBRTC_API_DATACHANNEL_H_ diff --git a/talk/app/webrtc/datachannel_unittest.cc b/webrtc/api/datachannel_unittest.cc similarity index 99% rename from talk/app/webrtc/datachannel_unittest.cc rename to webrtc/api/datachannel_unittest.cc index ff79541478..d5711e8ecd 100644 --- a/talk/app/webrtc/datachannel_unittest.cc +++ b/webrtc/api/datachannel_unittest.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/datachannel.h" -#include "talk/app/webrtc/sctputils.h" -#include "talk/app/webrtc/test/fakedatachannelprovider.h" +#include "webrtc/api/datachannel.h" +#include "webrtc/api/sctputils.h" +#include "webrtc/api/test/fakedatachannelprovider.h" #include "webrtc/base/gunit.h" using webrtc::DataChannel; diff --git a/talk/app/webrtc/datachannelinterface.h b/webrtc/api/datachannelinterface.h similarity index 97% rename from talk/app/webrtc/datachannelinterface.h rename to webrtc/api/datachannelinterface.h index d70972f05a..e291328a6f 100644 --- a/talk/app/webrtc/datachannelinterface.h +++ b/webrtc/api/datachannelinterface.h @@ -28,8 +28,8 @@ // This file contains interfaces for DataChannels // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel -#ifndef TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_ -#define TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_ +#ifndef WEBRTC_API_DATACHANNELINTERFACE_H_ +#define WEBRTC_API_DATACHANNELINTERFACE_H_ #include @@ -156,4 +156,4 @@ class DataChannelInterface : public rtc::RefCountInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_ +#endif // WEBRTC_API_DATACHANNELINTERFACE_H_ diff --git a/talk/app/webrtc/dtlsidentitystore.cc b/webrtc/api/dtlsidentitystore.cc similarity index 98% rename from talk/app/webrtc/dtlsidentitystore.cc rename to webrtc/api/dtlsidentitystore.cc index 390ec0d0b7..79c2075ac2 100644 --- a/talk/app/webrtc/dtlsidentitystore.cc +++ b/webrtc/api/dtlsidentitystore.cc @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/dtlsidentitystore.h" +#include "webrtc/api/dtlsidentitystore.h" #include -#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h" +#include "webrtc/api/webrtcsessiondescriptionfactory.h" #include "webrtc/base/logging.h" using webrtc::DtlsIdentityRequestObserver; diff --git a/talk/app/webrtc/dtlsidentitystore.h b/webrtc/api/dtlsidentitystore.h similarity index 97% rename from talk/app/webrtc/dtlsidentitystore.h rename to webrtc/api/dtlsidentitystore.h index 2a5309d34b..9313b9962c 100644 --- a/talk/app/webrtc/dtlsidentitystore.h +++ b/webrtc/api/dtlsidentitystore.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_DTLSIDENTITYSTORE_H_ -#define TALK_APP_WEBRTC_DTLSIDENTITYSTORE_H_ +#ifndef WEBRTC_API_DTLSIDENTITYSTORE_H_ +#define WEBRTC_API_DTLSIDENTITYSTORE_H_ #include #include @@ -162,4 +162,4 @@ class DtlsIdentityStoreImpl : public DtlsIdentityStoreInterface, } // namespace webrtc -#endif // TALK_APP_WEBRTC_DTLSIDENTITYSTORE_H_ +#endif // WEBRTC_API_DTLSIDENTITYSTORE_H_ diff --git a/talk/app/webrtc/dtlsidentitystore_unittest.cc b/webrtc/api/dtlsidentitystore_unittest.cc similarity index 97% rename from talk/app/webrtc/dtlsidentitystore_unittest.cc rename to webrtc/api/dtlsidentitystore_unittest.cc index e9242216f9..f96cf572db 100644 --- a/talk/app/webrtc/dtlsidentitystore_unittest.cc +++ b/webrtc/api/dtlsidentitystore_unittest.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/dtlsidentitystore.h" +#include "webrtc/api/dtlsidentitystore.h" -#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h" +#include "webrtc/api/webrtcsessiondescriptionfactory.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" #include "webrtc/base/ssladapter.h" diff --git a/talk/app/webrtc/dtmfsender.cc b/webrtc/api/dtmfsender.cc similarity index 99% rename from talk/app/webrtc/dtmfsender.cc rename to webrtc/api/dtmfsender.cc index 30e2ce3873..a10305cc42 100644 --- a/talk/app/webrtc/dtmfsender.cc +++ b/webrtc/api/dtmfsender.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/dtmfsender.h" +#include "webrtc/api/dtmfsender.h" #include diff --git a/talk/app/webrtc/dtmfsender.h b/webrtc/api/dtmfsender.h similarity index 94% rename from talk/app/webrtc/dtmfsender.h rename to webrtc/api/dtmfsender.h index 6d23610c7d..f0f0e6871a 100644 --- a/talk/app/webrtc/dtmfsender.h +++ b/webrtc/api/dtmfsender.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_DTMFSENDER_H_ -#define TALK_APP_WEBRTC_DTMFSENDER_H_ +#ifndef WEBRTC_API_DTMFSENDER_H_ +#define WEBRTC_API_DTMFSENDER_H_ #include -#include "talk/app/webrtc/dtmfsenderinterface.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/proxy.h" +#include "webrtc/api/dtmfsenderinterface.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/proxy.h" #include "webrtc/base/common.h" #include "webrtc/base/messagehandler.h" #include "webrtc/base/refcount.h" @@ -136,4 +136,4 @@ bool GetDtmfCode(char tone, int* code); } // namespace webrtc -#endif // TALK_APP_WEBRTC_DTMFSENDER_H_ +#endif // WEBRTC_API_DTMFSENDER_H_ diff --git a/talk/app/webrtc/dtmfsender_unittest.cc b/webrtc/api/dtmfsender_unittest.cc similarity index 99% rename from talk/app/webrtc/dtmfsender_unittest.cc rename to webrtc/api/dtmfsender_unittest.cc index f686aa2ccc..e754ca229f 100644 --- a/talk/app/webrtc/dtmfsender_unittest.cc +++ b/webrtc/api/dtmfsender_unittest.cc @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/dtmfsender.h" +#include "webrtc/api/dtmfsender.h" #include #include #include -#include "talk/app/webrtc/audiotrack.h" +#include "webrtc/api/audiotrack.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" #include "webrtc/base/timeutils.h" diff --git a/talk/app/webrtc/dtmfsenderinterface.h b/webrtc/api/dtmfsenderinterface.h similarity index 95% rename from talk/app/webrtc/dtmfsenderinterface.h rename to webrtc/api/dtmfsenderinterface.h index 7fbf57af23..327c673ca8 100644 --- a/talk/app/webrtc/dtmfsenderinterface.h +++ b/webrtc/api/dtmfsenderinterface.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ -#define TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ +#ifndef WEBRTC_API_DTMFSENDERINTERFACE_H_ +#define WEBRTC_API_DTMFSENDERINTERFACE_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/base/common.h" #include "webrtc/base/refcount.h" @@ -102,4 +102,4 @@ class DtmfSenderInterface : public rtc::RefCountInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_ +#endif // WEBRTC_API_DTMFSENDERINTERFACE_H_ diff --git a/talk/app/webrtc/fakemediacontroller.h b/webrtc/api/fakemediacontroller.h similarity index 91% rename from talk/app/webrtc/fakemediacontroller.h rename to webrtc/api/fakemediacontroller.h index 5bf3e5fcf8..ec1bd1295f 100644 --- a/talk/app/webrtc/fakemediacontroller.h +++ b/webrtc/api/fakemediacontroller.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_ -#define TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_ +#ifndef WEBRTC_API_FAKEMEDIACONTROLLER_H_ +#define WEBRTC_API_FAKEMEDIACONTROLLER_H_ -#include "talk/app/webrtc/mediacontroller.h" +#include "webrtc/api/mediacontroller.h" #include "webrtc/base/checks.h" namespace cricket { @@ -52,4 +52,4 @@ class FakeMediaController : public webrtc::MediaControllerInterface { webrtc::Call* call_; }; } // namespace cricket -#endif // TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_ +#endif // WEBRTC_API_FAKEMEDIACONTROLLER_H_ diff --git a/talk/app/webrtc/fakemetricsobserver.cc b/webrtc/api/fakemetricsobserver.cc similarity index 98% rename from talk/app/webrtc/fakemetricsobserver.cc rename to webrtc/api/fakemetricsobserver.cc index 4a100a079e..6070ff418c 100644 --- a/talk/app/webrtc/fakemetricsobserver.cc +++ b/webrtc/api/fakemetricsobserver.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/fakemetricsobserver.h" +#include "webrtc/api/fakemetricsobserver.h" #include "webrtc/base/checks.h" namespace webrtc { diff --git a/talk/app/webrtc/fakemetricsobserver.h b/webrtc/api/fakemetricsobserver.h similarity index 92% rename from talk/app/webrtc/fakemetricsobserver.h rename to webrtc/api/fakemetricsobserver.h index e3a22841d8..1f4c2ab838 100644 --- a/talk/app/webrtc/fakemetricsobserver.h +++ b/webrtc/api/fakemetricsobserver.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_ -#define TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_ +#ifndef WEBRTC_API_FAKEMETRICSOBSERVER_H_ +#define WEBRTC_API_FAKEMETRICSOBSERVER_H_ #include #include -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" #include "webrtc/base/thread_checker.h" namespace webrtc { @@ -65,4 +65,4 @@ class FakeMetricsObserver : public MetricsObserverInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_ +#endif // WEBRTC_API_FAKEMETRICSOBSERVER_H_ diff --git a/talk/app/webrtc/java/README b/webrtc/api/java/README similarity index 100% rename from talk/app/webrtc/java/README rename to webrtc/api/java/README diff --git a/talk/app/webrtc/java/android/org/webrtc/Camera2Enumerator.java b/webrtc/api/java/android/org/webrtc/Camera2Enumerator.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/Camera2Enumerator.java rename to webrtc/api/java/android/org/webrtc/Camera2Enumerator.java diff --git a/talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java b/webrtc/api/java/android/org/webrtc/CameraEnumerationAndroid.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java rename to webrtc/api/java/android/org/webrtc/CameraEnumerationAndroid.java diff --git a/talk/app/webrtc/java/android/org/webrtc/CameraEnumerator.java b/webrtc/api/java/android/org/webrtc/CameraEnumerator.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/CameraEnumerator.java rename to webrtc/api/java/android/org/webrtc/CameraEnumerator.java diff --git a/talk/app/webrtc/java/android/org/webrtc/EglBase.java b/webrtc/api/java/android/org/webrtc/EglBase.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/EglBase.java rename to webrtc/api/java/android/org/webrtc/EglBase.java diff --git a/talk/app/webrtc/java/android/org/webrtc/EglBase10.java b/webrtc/api/java/android/org/webrtc/EglBase10.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/EglBase10.java rename to webrtc/api/java/android/org/webrtc/EglBase10.java diff --git a/talk/app/webrtc/java/android/org/webrtc/EglBase14.java b/webrtc/api/java/android/org/webrtc/EglBase14.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/EglBase14.java rename to webrtc/api/java/android/org/webrtc/EglBase14.java diff --git a/talk/app/webrtc/java/android/org/webrtc/GlRectDrawer.java b/webrtc/api/java/android/org/webrtc/GlRectDrawer.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/GlRectDrawer.java rename to webrtc/api/java/android/org/webrtc/GlRectDrawer.java diff --git a/talk/app/webrtc/java/android/org/webrtc/GlShader.java b/webrtc/api/java/android/org/webrtc/GlShader.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/GlShader.java rename to webrtc/api/java/android/org/webrtc/GlShader.java diff --git a/talk/app/webrtc/java/android/org/webrtc/GlTextureFrameBuffer.java b/webrtc/api/java/android/org/webrtc/GlTextureFrameBuffer.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/GlTextureFrameBuffer.java rename to webrtc/api/java/android/org/webrtc/GlTextureFrameBuffer.java diff --git a/talk/app/webrtc/java/android/org/webrtc/GlUtil.java b/webrtc/api/java/android/org/webrtc/GlUtil.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/GlUtil.java rename to webrtc/api/java/android/org/webrtc/GlUtil.java diff --git a/talk/app/webrtc/java/android/org/webrtc/NetworkMonitor.java b/webrtc/api/java/android/org/webrtc/NetworkMonitor.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/NetworkMonitor.java rename to webrtc/api/java/android/org/webrtc/NetworkMonitor.java diff --git a/talk/app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java b/webrtc/api/java/android/org/webrtc/NetworkMonitorAutoDetect.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java rename to webrtc/api/java/android/org/webrtc/NetworkMonitorAutoDetect.java diff --git a/talk/app/webrtc/java/android/org/webrtc/OWNERS b/webrtc/api/java/android/org/webrtc/OWNERS similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/OWNERS rename to webrtc/api/java/android/org/webrtc/OWNERS diff --git a/talk/app/webrtc/java/android/org/webrtc/RendererCommon.java b/webrtc/api/java/android/org/webrtc/RendererCommon.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/RendererCommon.java rename to webrtc/api/java/android/org/webrtc/RendererCommon.java diff --git a/talk/app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java b/webrtc/api/java/android/org/webrtc/SurfaceTextureHelper.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java rename to webrtc/api/java/android/org/webrtc/SurfaceTextureHelper.java diff --git a/talk/app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java b/webrtc/api/java/android/org/webrtc/SurfaceViewRenderer.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java rename to webrtc/api/java/android/org/webrtc/SurfaceViewRenderer.java diff --git a/talk/app/webrtc/java/android/org/webrtc/ThreadUtils.java b/webrtc/api/java/android/org/webrtc/ThreadUtils.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/ThreadUtils.java rename to webrtc/api/java/android/org/webrtc/ThreadUtils.java diff --git a/talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java b/webrtc/api/java/android/org/webrtc/VideoCapturerAndroid.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java rename to webrtc/api/java/android/org/webrtc/VideoCapturerAndroid.java diff --git a/talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java b/webrtc/api/java/android/org/webrtc/VideoRendererGui.java similarity index 100% rename from talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java rename to webrtc/api/java/android/org/webrtc/VideoRendererGui.java diff --git a/talk/app/webrtc/java/jni/OWNERS b/webrtc/api/java/jni/OWNERS similarity index 100% rename from talk/app/webrtc/java/jni/OWNERS rename to webrtc/api/java/jni/OWNERS diff --git a/talk/app/webrtc/java/jni/androidmediacodeccommon.h b/webrtc/api/java/jni/androidmediacodeccommon.h similarity index 93% rename from talk/app/webrtc/java/jni/androidmediacodeccommon.h rename to webrtc/api/java/jni/androidmediacodeccommon.h index 329aa2af90..7044fb4bc0 100644 --- a/talk/app/webrtc/java/jni/androidmediacodeccommon.h +++ b/webrtc/api/java/jni/androidmediacodeccommon.h @@ -26,15 +26,15 @@ * */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_ +#ifndef WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_ +#define WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_ #include #include -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" #include "webrtc/base/thread.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/jni_helpers.h" #include "webrtc/base/logging.h" #include "webrtc/base/thread.h" #include "webrtc/system_wrappers/include/tick_util.h" @@ -110,4 +110,4 @@ static inline bool CheckException(JNIEnv* jni) { } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_ +#endif // WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_ diff --git a/talk/app/webrtc/java/jni/androidmediadecoder_jni.cc b/webrtc/api/java/jni/androidmediadecoder_jni.cc similarity index 99% rename from talk/app/webrtc/java/jni/androidmediadecoder_jni.cc rename to webrtc/api/java/jni/androidmediadecoder_jni.cc index a462b6c70d..b9973be01e 100644 --- a/talk/app/webrtc/java/jni/androidmediadecoder_jni.cc +++ b/webrtc/api/java/jni/androidmediadecoder_jni.cc @@ -29,16 +29,17 @@ #include #include -#include "talk/app/webrtc/java/jni/androidmediadecoder_jni.h" // NOTICE: androidmediadecoder_jni.h must be included before // androidmediacodeccommon.h to avoid build errors. -#include "talk/app/webrtc/java/jni/androidmediacodeccommon.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/native_handle_impl.h" -#include "talk/app/webrtc/java/jni/surfacetexturehelper_jni.h" +#include "webrtc/api/java/jni/androidmediadecoder_jni.h" + #include "third_party/libyuv/include/libyuv/convert.h" #include "third_party/libyuv/include/libyuv/convert_from.h" #include "third_party/libyuv/include/libyuv/video_common.h" +#include "webrtc/api/java/jni/androidmediacodeccommon.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/native_handle_impl.h" +#include "webrtc/api/java/jni/surfacetexturehelper_jni.h" #include "webrtc/base/bind.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" diff --git a/talk/app/webrtc/java/jni/androidmediadecoder_jni.h b/webrtc/api/java/jni/androidmediadecoder_jni.h similarity index 90% rename from talk/app/webrtc/java/jni/androidmediadecoder_jni.h rename to webrtc/api/java/jni/androidmediadecoder_jni.h index b25ec137fc..c79490e9c7 100644 --- a/talk/app/webrtc/java/jni/androidmediadecoder_jni.h +++ b/webrtc/api/java/jni/androidmediadecoder_jni.h @@ -26,10 +26,10 @@ * */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_ +#ifndef WEBRTC_API_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_ +#define WEBRTC_API_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_ -#include "talk/app/webrtc/java/jni/eglbase_jni.h" +#include "webrtc/api/java/jni/eglbase_jni.h" #include "webrtc/media/webrtc/webrtcvideodecoderfactory.h" namespace webrtc_jni { @@ -56,4 +56,4 @@ class MediaCodecVideoDecoderFactory } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_ +#endif // WEBRTC_API_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_ diff --git a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc b/webrtc/api/java/jni/androidmediaencoder_jni.cc similarity index 99% rename from talk/app/webrtc/java/jni/androidmediaencoder_jni.cc rename to webrtc/api/java/jni/androidmediaencoder_jni.cc index aade95ba46..a06b026c84 100644 --- a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc +++ b/webrtc/api/java/jni/androidmediaencoder_jni.cc @@ -26,15 +26,16 @@ * */ -#include "talk/app/webrtc/java/jni/androidmediaencoder_jni.h" // NOTICE: androidmediaencoder_jni.h must be included before // androidmediacodeccommon.h to avoid build errors. -#include "talk/app/webrtc/java/jni/androidmediacodeccommon.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/native_handle_impl.h" +#include "webrtc/api/java/jni/androidmediaencoder_jni.h" + #include "third_party/libyuv/include/libyuv/convert.h" #include "third_party/libyuv/include/libyuv/convert_from.h" #include "third_party/libyuv/include/libyuv/video_common.h" +#include "webrtc/api/java/jni/androidmediacodeccommon.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/native_handle_impl.h" #include "webrtc/base/bind.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" diff --git a/talk/app/webrtc/java/jni/androidmediaencoder_jni.h b/webrtc/api/java/jni/androidmediaencoder_jni.h similarity index 90% rename from talk/app/webrtc/java/jni/androidmediaencoder_jni.h rename to webrtc/api/java/jni/androidmediaencoder_jni.h index 3d361d2a05..e96a489c7f 100644 --- a/talk/app/webrtc/java/jni/androidmediaencoder_jni.h +++ b/webrtc/api/java/jni/androidmediaencoder_jni.h @@ -26,12 +26,12 @@ * */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_ +#ifndef WEBRTC_API_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_ +#define WEBRTC_API_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_ #include -#include "talk/app/webrtc/java/jni/eglbase_jni.h" +#include "webrtc/api/java/jni/eglbase_jni.h" #include "webrtc/media/webrtc/webrtcvideoencoderfactory.h" namespace webrtc_jni { @@ -60,4 +60,4 @@ class MediaCodecVideoEncoderFactory } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_ +#endif // WEBRTC_API_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_ diff --git a/talk/app/webrtc/java/jni/androidnetworkmonitor_jni.cc b/webrtc/api/java/jni/androidnetworkmonitor_jni.cc similarity index 98% rename from talk/app/webrtc/java/jni/androidnetworkmonitor_jni.cc rename to webrtc/api/java/jni/androidnetworkmonitor_jni.cc index 69870dc7b9..a38fa11ca8 100644 --- a/talk/app/webrtc/java/jni/androidnetworkmonitor_jni.cc +++ b/webrtc/api/java/jni/androidnetworkmonitor_jni.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h" +#include "webrtc/api/java/jni/androidnetworkmonitor_jni.h" #include -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/jni_helpers.h" #include "webrtc/base/bind.h" #include "webrtc/base/common.h" #include "webrtc/base/ipaddress.h" diff --git a/talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h b/webrtc/api/java/jni/androidnetworkmonitor_jni.h similarity index 93% rename from talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h rename to webrtc/api/java/jni/androidnetworkmonitor_jni.h index 39898eabab..220a5bcabf 100644 --- a/talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h +++ b/webrtc/api/java/jni/androidnetworkmonitor_jni.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_ +#ifndef WEBRTC_API_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_ +#define WEBRTC_API_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_ #include "webrtc/base/networkmonitor.h" #include -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/jni_helpers.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/thread_checker.h" @@ -105,4 +105,4 @@ class AndroidNetworkMonitorFactory : public rtc::NetworkMonitorFactory { } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_ +#endif // WEBRTC_API_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_ diff --git a/talk/app/webrtc/java/jni/androidvideocapturer_jni.cc b/webrtc/api/java/jni/androidvideocapturer_jni.cc similarity index 97% rename from talk/app/webrtc/java/jni/androidvideocapturer_jni.cc rename to webrtc/api/java/jni/androidvideocapturer_jni.cc index 8813c89de4..a636d6264a 100644 --- a/talk/app/webrtc/java/jni/androidvideocapturer_jni.cc +++ b/webrtc/api/java/jni/androidvideocapturer_jni.cc @@ -26,10 +26,10 @@ * */ -#include "talk/app/webrtc/java/jni/androidvideocapturer_jni.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/native_handle_impl.h" -#include "talk/app/webrtc/java/jni/surfacetexturehelper_jni.h" +#include "webrtc/api/java/jni/androidvideocapturer_jni.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/native_handle_impl.h" +#include "webrtc/api/java/jni/surfacetexturehelper_jni.h" #include "third_party/libyuv/include/libyuv/convert.h" #include "webrtc/base/bind.h" diff --git a/talk/app/webrtc/java/jni/androidvideocapturer_jni.h b/webrtc/api/java/jni/androidvideocapturer_jni.h similarity index 93% rename from talk/app/webrtc/java/jni/androidvideocapturer_jni.h rename to webrtc/api/java/jni/androidvideocapturer_jni.h index 89ecacb3a5..bf611f5448 100644 --- a/talk/app/webrtc/java/jni/androidvideocapturer_jni.h +++ b/webrtc/api/java/jni/androidvideocapturer_jni.h @@ -26,13 +26,13 @@ * */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDVIDEOCAPTURER_JNI_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDVIDEOCAPTURER_JNI_H_ +#ifndef WEBRTC_API_JAVA_JNI_ANDROIDVIDEOCAPTURER_JNI_H_ +#define WEBRTC_API_JAVA_JNI_ANDROIDVIDEOCAPTURER_JNI_H_ #include -#include "talk/app/webrtc/androidvideocapturer.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/androidvideocapturer.h" +#include "webrtc/api/java/jni/jni_helpers.h" #include "webrtc/base/asyncinvoker.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/thread_checker.h" @@ -113,4 +113,4 @@ class AndroidVideoCapturerJni : public webrtc::AndroidVideoCapturerDelegate { } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDVIDEOCAPTURER_JNI_H_ +#endif // WEBRTC_API_JAVA_JNI_ANDROIDVIDEOCAPTURER_JNI_H_ diff --git a/talk/app/webrtc/java/jni/classreferenceholder.cc b/webrtc/api/java/jni/classreferenceholder.cc similarity index 98% rename from talk/app/webrtc/java/jni/classreferenceholder.cc rename to webrtc/api/java/jni/classreferenceholder.cc index 0d52bc5122..0625cc21bb 100644 --- a/talk/app/webrtc/java/jni/classreferenceholder.cc +++ b/webrtc/api/java/jni/classreferenceholder.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. * */ -#include "talk/app/webrtc/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/jni_helpers.h" namespace webrtc_jni { diff --git a/talk/app/webrtc/java/jni/classreferenceholder.h b/webrtc/api/java/jni/classreferenceholder.h similarity index 93% rename from talk/app/webrtc/java/jni/classreferenceholder.h rename to webrtc/api/java/jni/classreferenceholder.h index f345c1dbd3..5edf614401 100644 --- a/talk/app/webrtc/java/jni/classreferenceholder.h +++ b/webrtc/api/java/jni/classreferenceholder.h @@ -31,8 +31,8 @@ // stack. Consequently, we only look up all classes once in app/webrtc. // http://developer.android.com/training/articles/perf-jni.html#faq_FindClass -#ifndef TALK_APP_WEBRTC_JAVA_JNI_CLASSREFERENCEHOLDER_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_CLASSREFERENCEHOLDER_H_ +#ifndef WEBRTC_API_JAVA_JNI_CLASSREFERENCEHOLDER_H_ +#define WEBRTC_API_JAVA_JNI_CLASSREFERENCEHOLDER_H_ #include #include @@ -56,4 +56,4 @@ jclass FindClass(JNIEnv* jni, const char* name); } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_CLASSREFERENCEHOLDER_H_ +#endif // WEBRTC_API_JAVA_JNI_CLASSREFERENCEHOLDER_H_ diff --git a/talk/app/webrtc/java/jni/eglbase_jni.cc b/webrtc/api/java/jni/eglbase_jni.cc similarity index 93% rename from talk/app/webrtc/java/jni/eglbase_jni.cc rename to webrtc/api/java/jni/eglbase_jni.cc index b91aa390f0..26eeeb311a 100644 --- a/talk/app/webrtc/java/jni/eglbase_jni.cc +++ b/webrtc/api/java/jni/eglbase_jni.cc @@ -26,11 +26,11 @@ * */ -#include "talk/app/webrtc/java/jni/eglbase_jni.h" +#include "webrtc/api/java/jni/eglbase_jni.h" -#include "talk/app/webrtc/java/jni/androidmediacodeccommon.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/androidmediacodeccommon.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/jni_helpers.h" namespace webrtc_jni { diff --git a/talk/app/webrtc/java/jni/eglbase_jni.h b/webrtc/api/java/jni/eglbase_jni.h similarity index 93% rename from talk/app/webrtc/java/jni/eglbase_jni.h rename to webrtc/api/java/jni/eglbase_jni.h index 1015d90b92..de7e39e70a 100644 --- a/talk/app/webrtc/java/jni/eglbase_jni.h +++ b/webrtc/api/java/jni/eglbase_jni.h @@ -26,8 +26,8 @@ * */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_EGLBASE_JNI_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_EGLBASE_JNI_H_ +#ifndef WEBRTC_API_JAVA_JNI_EGLBASE_JNI_H_ +#define WEBRTC_API_JAVA_JNI_EGLBASE_JNI_H_ #include @@ -57,4 +57,4 @@ class EglBase { } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_EGLBASE_JNI_H_ +#endif // WEBRTC_API_JAVA_JNI_EGLBASE_JNI_H_ diff --git a/talk/app/webrtc/java/jni/jni_helpers.cc b/webrtc/api/java/jni/jni_helpers.cc similarity index 99% rename from talk/app/webrtc/java/jni/jni_helpers.cc rename to webrtc/api/java/jni/jni_helpers.cc index 25b340ff79..b07a9c5d14 100644 --- a/talk/app/webrtc/java/jni/jni_helpers.cc +++ b/webrtc/api/java/jni/jni_helpers.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. * */ -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/jni_helpers.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/classreferenceholder.h" #include #include diff --git a/talk/app/webrtc/java/jni/jni_helpers.h b/webrtc/api/java/jni/jni_helpers.h similarity index 97% rename from talk/app/webrtc/java/jni/jni_helpers.h rename to webrtc/api/java/jni/jni_helpers.h index 374962be49..5498158b2f 100644 --- a/talk/app/webrtc/java/jni/jni_helpers.h +++ b/webrtc/api/java/jni/jni_helpers.h @@ -29,8 +29,8 @@ // This file contain convenience functions and classes for JNI. // Before using any of the methods, InitGlobalJniVariables must be called. -#ifndef TALK_APP_WEBRTC_JAVA_JNI_JNI_HELPERS_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_JNI_HELPERS_H_ +#ifndef WEBRTC_API_JAVA_JNI_JNI_HELPERS_H_ +#define WEBRTC_API_JAVA_JNI_JNI_HELPERS_H_ #include #include @@ -143,4 +143,4 @@ class ScopedGlobalRef { } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_JNI_HELPERS_H_ +#endif // WEBRTC_API_JAVA_JNI_JNI_HELPERS_H_ diff --git a/talk/app/webrtc/java/jni/jni_onload.cc b/webrtc/api/java/jni/jni_onload.cc similarity index 95% rename from talk/app/webrtc/java/jni/jni_onload.cc rename to webrtc/api/java/jni/jni_onload.cc index 9664ecdca6..af2804dd0f 100644 --- a/talk/app/webrtc/java/jni/jni_onload.cc +++ b/webrtc/api/java/jni/jni_onload.cc @@ -29,8 +29,8 @@ #undef JNIEXPORT #define JNIEXPORT __attribute__((visibility("default"))) -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/jni_helpers.h" #include "webrtc/base/ssladapter.h" namespace webrtc_jni { diff --git a/talk/app/webrtc/java/jni/native_handle_impl.cc b/webrtc/api/java/jni/native_handle_impl.cc similarity index 98% rename from talk/app/webrtc/java/jni/native_handle_impl.cc rename to webrtc/api/java/jni/native_handle_impl.cc index e2530880e8..8ec549cdb7 100644 --- a/talk/app/webrtc/java/jni/native_handle_impl.cc +++ b/webrtc/api/java/jni/native_handle_impl.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/java/jni/native_handle_impl.h" +#include "webrtc/api/java/jni/native_handle_impl.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/jni_helpers.h" #include "webrtc/base/bind.h" #include "webrtc/base/checks.h" #include "webrtc/base/keep_ref_until_done.h" diff --git a/talk/app/webrtc/java/jni/native_handle_impl.h b/webrtc/api/java/jni/native_handle_impl.h similarity index 94% rename from talk/app/webrtc/java/jni/native_handle_impl.h rename to webrtc/api/java/jni/native_handle_impl.h index 1d0f601d0d..4203bdf402 100644 --- a/talk/app/webrtc/java/jni/native_handle_impl.h +++ b/webrtc/api/java/jni/native_handle_impl.h @@ -26,8 +26,8 @@ * */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_NATIVE_HANDLE_IMPL_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_NATIVE_HANDLE_IMPL_H_ +#ifndef WEBRTC_API_JAVA_JNI_NATIVE_HANDLE_IMPL_H_ +#define WEBRTC_API_JAVA_JNI_NATIVE_HANDLE_IMPL_H_ #include @@ -74,4 +74,4 @@ class AndroidTextureBuffer : public webrtc::NativeHandleBuffer { } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_NATIVE_HANDLE_IMPL_H_ +#endif // WEBRTC_API_JAVA_JNI_NATIVE_HANDLE_IMPL_H_ diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/webrtc/api/java/jni/peerconnection_jni.cc similarity index 99% rename from talk/app/webrtc/java/jni/peerconnection_jni.cc rename to webrtc/api/java/jni/peerconnection_jni.cc index bbbd77f03c..1160b2b134 100644 --- a/talk/app/webrtc/java/jni/peerconnection_jni.cc +++ b/webrtc/api/java/jni/peerconnection_jni.cc @@ -59,20 +59,20 @@ #include #include -#include "talk/app/webrtc/androidvideocapturer.h" -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/java/jni/androidmediadecoder_jni.h" -#include "talk/app/webrtc/java/jni/androidmediaencoder_jni.h" -#include "talk/app/webrtc/java/jni/androidvideocapturer_jni.h" -#include "talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" -#include "talk/app/webrtc/java/jni/native_handle_impl.h" -#include "talk/app/webrtc/mediaconstraintsinterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/rtpreceiverinterface.h" -#include "talk/app/webrtc/rtpsenderinterface.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/androidvideocapturer.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/java/jni/androidmediadecoder_jni.h" +#include "webrtc/api/java/jni/androidmediaencoder_jni.h" +#include "webrtc/api/java/jni/androidnetworkmonitor_jni.h" +#include "webrtc/api/java/jni/androidvideocapturer_jni.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/native_handle_impl.h" +#include "webrtc/api/mediaconstraintsinterface.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/rtpreceiverinterface.h" +#include "webrtc/api/rtpsenderinterface.h" +#include "webrtc/api/videosourceinterface.h" #include "webrtc/base/bind.h" #include "webrtc/base/checks.h" #include "webrtc/base/event_tracer.h" @@ -87,6 +87,7 @@ #include "webrtc/media/devices/videorendererfactory.h" #include "webrtc/media/webrtc/webrtcvideodecoderfactory.h" #include "webrtc/media/webrtc/webrtcvideoencoderfactory.h" +#include "webrtc/modules/video_render/video_render_internal.h" #include "webrtc/system_wrappers/include/field_trial_default.h" #include "webrtc/system_wrappers/include/logcat_trace_context.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.cc b/webrtc/api/java/jni/surfacetexturehelper_jni.cc similarity index 95% rename from talk/app/webrtc/java/jni/surfacetexturehelper_jni.cc rename to webrtc/api/java/jni/surfacetexturehelper_jni.cc index 3e32b9a6fe..335081d5f1 100644 --- a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.cc +++ b/webrtc/api/java/jni/surfacetexturehelper_jni.cc @@ -27,9 +27,9 @@ */ -#include "talk/app/webrtc/java/jni/surfacetexturehelper_jni.h" +#include "webrtc/api/java/jni/surfacetexturehelper_jni.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/classreferenceholder.h" #include "webrtc/base/bind.h" #include "webrtc/base/checks.h" diff --git a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.h b/webrtc/api/java/jni/surfacetexturehelper_jni.h similarity index 91% rename from talk/app/webrtc/java/jni/surfacetexturehelper_jni.h rename to webrtc/api/java/jni/surfacetexturehelper_jni.h index 8dde2b54ed..8953b0249b 100644 --- a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.h +++ b/webrtc/api/java/jni/surfacetexturehelper_jni.h @@ -26,13 +26,13 @@ * */ -#ifndef TALK_APP_WEBRTC_JAVA_JNI_SURFACETEXTUREHELPER_JNI_H_ -#define TALK_APP_WEBRTC_JAVA_JNI_SURFACETEXTUREHELPER_JNI_H_ +#ifndef WEBRTC_API_JAVA_JNI_SURFACETEXTUREHELPER_JNI_H_ +#define WEBRTC_API_JAVA_JNI_SURFACETEXTUREHELPER_JNI_H_ #include -#include "talk/app/webrtc/java/jni/jni_helpers.h" -#include "talk/app/webrtc/java/jni/native_handle_impl.h" +#include "webrtc/api/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/native_handle_impl.h" #include "webrtc/base/refcount.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/common_video/include/video_frame_buffer.h" @@ -76,4 +76,4 @@ class SurfaceTextureHelper : public rtc::RefCountInterface { } // namespace webrtc_jni -#endif // TALK_APP_WEBRTC_JAVA_JNI_SURFACETEXTUREHELPER_JNI_H_ +#endif // WEBRTC_API_JAVA_JNI_SURFACETEXTUREHELPER_JNI_H_ diff --git a/talk/app/webrtc/java/src/org/webrtc/AudioSource.java b/webrtc/api/java/src/org/webrtc/AudioSource.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/AudioSource.java rename to webrtc/api/java/src/org/webrtc/AudioSource.java diff --git a/talk/app/webrtc/java/src/org/webrtc/AudioTrack.java b/webrtc/api/java/src/org/webrtc/AudioTrack.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/AudioTrack.java rename to webrtc/api/java/src/org/webrtc/AudioTrack.java diff --git a/talk/app/webrtc/java/src/org/webrtc/CallSessionFileRotatingLogSink.java b/webrtc/api/java/src/org/webrtc/CallSessionFileRotatingLogSink.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/CallSessionFileRotatingLogSink.java rename to webrtc/api/java/src/org/webrtc/CallSessionFileRotatingLogSink.java diff --git a/talk/app/webrtc/java/src/org/webrtc/DataChannel.java b/webrtc/api/java/src/org/webrtc/DataChannel.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/DataChannel.java rename to webrtc/api/java/src/org/webrtc/DataChannel.java diff --git a/talk/app/webrtc/java/src/org/webrtc/IceCandidate.java b/webrtc/api/java/src/org/webrtc/IceCandidate.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/IceCandidate.java rename to webrtc/api/java/src/org/webrtc/IceCandidate.java diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java b/webrtc/api/java/src/org/webrtc/MediaCodecVideoDecoder.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java rename to webrtc/api/java/src/org/webrtc/MediaCodecVideoDecoder.java diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java b/webrtc/api/java/src/org/webrtc/MediaCodecVideoEncoder.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java rename to webrtc/api/java/src/org/webrtc/MediaCodecVideoEncoder.java diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaConstraints.java b/webrtc/api/java/src/org/webrtc/MediaConstraints.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/MediaConstraints.java rename to webrtc/api/java/src/org/webrtc/MediaConstraints.java diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaSource.java b/webrtc/api/java/src/org/webrtc/MediaSource.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/MediaSource.java rename to webrtc/api/java/src/org/webrtc/MediaSource.java diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaStream.java b/webrtc/api/java/src/org/webrtc/MediaStream.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/MediaStream.java rename to webrtc/api/java/src/org/webrtc/MediaStream.java diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaStreamTrack.java b/webrtc/api/java/src/org/webrtc/MediaStreamTrack.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/MediaStreamTrack.java rename to webrtc/api/java/src/org/webrtc/MediaStreamTrack.java diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java b/webrtc/api/java/src/org/webrtc/PeerConnection.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/PeerConnection.java rename to webrtc/api/java/src/org/webrtc/PeerConnection.java diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java b/webrtc/api/java/src/org/webrtc/PeerConnectionFactory.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java rename to webrtc/api/java/src/org/webrtc/PeerConnectionFactory.java diff --git a/talk/app/webrtc/java/src/org/webrtc/RtpReceiver.java b/webrtc/api/java/src/org/webrtc/RtpReceiver.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/RtpReceiver.java rename to webrtc/api/java/src/org/webrtc/RtpReceiver.java diff --git a/talk/app/webrtc/java/src/org/webrtc/RtpSender.java b/webrtc/api/java/src/org/webrtc/RtpSender.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/RtpSender.java rename to webrtc/api/java/src/org/webrtc/RtpSender.java diff --git a/talk/app/webrtc/java/src/org/webrtc/SdpObserver.java b/webrtc/api/java/src/org/webrtc/SdpObserver.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/SdpObserver.java rename to webrtc/api/java/src/org/webrtc/SdpObserver.java diff --git a/talk/app/webrtc/java/src/org/webrtc/SessionDescription.java b/webrtc/api/java/src/org/webrtc/SessionDescription.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/SessionDescription.java rename to webrtc/api/java/src/org/webrtc/SessionDescription.java diff --git a/talk/app/webrtc/java/src/org/webrtc/StatsObserver.java b/webrtc/api/java/src/org/webrtc/StatsObserver.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/StatsObserver.java rename to webrtc/api/java/src/org/webrtc/StatsObserver.java diff --git a/talk/app/webrtc/java/src/org/webrtc/StatsReport.java b/webrtc/api/java/src/org/webrtc/StatsReport.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/StatsReport.java rename to webrtc/api/java/src/org/webrtc/StatsReport.java diff --git a/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java b/webrtc/api/java/src/org/webrtc/VideoCapturer.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java rename to webrtc/api/java/src/org/webrtc/VideoCapturer.java diff --git a/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java b/webrtc/api/java/src/org/webrtc/VideoRenderer.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java rename to webrtc/api/java/src/org/webrtc/VideoRenderer.java diff --git a/talk/app/webrtc/java/src/org/webrtc/VideoSource.java b/webrtc/api/java/src/org/webrtc/VideoSource.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/VideoSource.java rename to webrtc/api/java/src/org/webrtc/VideoSource.java diff --git a/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java b/webrtc/api/java/src/org/webrtc/VideoTrack.java similarity index 100% rename from talk/app/webrtc/java/src/org/webrtc/VideoTrack.java rename to webrtc/api/java/src/org/webrtc/VideoTrack.java diff --git a/talk/app/webrtc/jsep.h b/webrtc/api/jsep.h similarity index 98% rename from talk/app/webrtc/jsep.h rename to webrtc/api/jsep.h index c12ab85f34..c49a16b451 100644 --- a/talk/app/webrtc/jsep.h +++ b/webrtc/api/jsep.h @@ -27,8 +27,8 @@ // Interfaces matching the draft-ietf-rtcweb-jsep-01. -#ifndef TALK_APP_WEBRTC_JSEP_H_ -#define TALK_APP_WEBRTC_JSEP_H_ +#ifndef WEBRTC_API_JSEP_H_ +#define WEBRTC_API_JSEP_H_ #include #include @@ -152,4 +152,4 @@ class SetSessionDescriptionObserver : public rtc::RefCountInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_JSEP_H_ +#endif // WEBRTC_API_JSEP_H_ diff --git a/talk/app/webrtc/jsepicecandidate.cc b/webrtc/api/jsepicecandidate.cc similarity index 97% rename from talk/app/webrtc/jsepicecandidate.cc rename to webrtc/api/jsepicecandidate.cc index 768bd0a281..172c52e74c 100644 --- a/talk/app/webrtc/jsepicecandidate.cc +++ b/webrtc/api/jsepicecandidate.cc @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/jsepicecandidate.h" +#include "webrtc/api/jsepicecandidate.h" #include -#include "talk/app/webrtc/webrtcsdp.h" +#include "webrtc/api/webrtcsdp.h" #include "webrtc/base/stringencode.h" namespace webrtc { diff --git a/talk/app/webrtc/jsepicecandidate.h b/webrtc/api/jsepicecandidate.h similarity index 94% rename from talk/app/webrtc/jsepicecandidate.h rename to webrtc/api/jsepicecandidate.h index 539376e057..957d7c0bd5 100644 --- a/talk/app/webrtc/jsepicecandidate.h +++ b/webrtc/api/jsepicecandidate.h @@ -27,12 +27,12 @@ // Implements the IceCandidateInterface. -#ifndef TALK_APP_WEBRTC_JSEPICECANDIDATE_H_ -#define TALK_APP_WEBRTC_JSEPICECANDIDATE_H_ +#ifndef WEBRTC_API_JSEPICECANDIDATE_H_ +#define WEBRTC_API_JSEPICECANDIDATE_H_ #include -#include "talk/app/webrtc/jsep.h" +#include "webrtc/api/jsep.h" #include "webrtc/base/constructormagic.h" #include "webrtc/p2p/base/candidate.h" @@ -89,4 +89,4 @@ class JsepCandidateCollection : public IceCandidateCollection { } // namespace webrtc -#endif // TALK_APP_WEBRTC_JSEPICECANDIDATE_H_ +#endif // WEBRTC_API_JSEPICECANDIDATE_H_ diff --git a/talk/app/webrtc/jsepsessiondescription.cc b/webrtc/api/jsepsessiondescription.cc similarity index 98% rename from talk/app/webrtc/jsepsessiondescription.cc rename to webrtc/api/jsepsessiondescription.cc index 226432db69..2ffc2de62b 100644 --- a/talk/app/webrtc/jsepsessiondescription.cc +++ b/webrtc/api/jsepsessiondescription.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/jsepsessiondescription.h" +#include "webrtc/api/jsepsessiondescription.h" -#include "talk/app/webrtc/webrtcsdp.h" +#include "webrtc/api/webrtcsdp.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/arraysize.h" #include "webrtc/base/stringencode.h" diff --git a/talk/app/webrtc/jsepsessiondescription.h b/webrtc/api/jsepsessiondescription.h similarity index 94% rename from talk/app/webrtc/jsepsessiondescription.h rename to webrtc/api/jsepsessiondescription.h index 756352c240..b6e634823f 100644 --- a/talk/app/webrtc/jsepsessiondescription.h +++ b/webrtc/api/jsepsessiondescription.h @@ -27,14 +27,14 @@ // Implements the SessionDescriptionInterface. -#ifndef TALK_APP_WEBRTC_JSEPSESSIONDESCRIPTION_H_ -#define TALK_APP_WEBRTC_JSEPSESSIONDESCRIPTION_H_ +#ifndef WEBRTC_API_JSEPSESSIONDESCRIPTION_H_ +#define WEBRTC_API_JSEPSESSIONDESCRIPTION_H_ #include #include -#include "talk/app/webrtc/jsep.h" -#include "talk/app/webrtc/jsepicecandidate.h" +#include "webrtc/api/jsep.h" +#include "webrtc/api/jsepicecandidate.h" #include "webrtc/base/scoped_ptr.h" namespace cricket { @@ -103,4 +103,4 @@ class JsepSessionDescription : public SessionDescriptionInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_JSEPSESSIONDESCRIPTION_H_ +#endif // WEBRTC_API_JSEPSESSIONDESCRIPTION_H_ diff --git a/talk/app/webrtc/jsepsessiondescription_unittest.cc b/webrtc/api/jsepsessiondescription_unittest.cc similarity index 99% rename from talk/app/webrtc/jsepsessiondescription_unittest.cc rename to webrtc/api/jsepsessiondescription_unittest.cc index dc0773a268..90de058314 100644 --- a/talk/app/webrtc/jsepsessiondescription_unittest.cc +++ b/webrtc/api/jsepsessiondescription_unittest.cc @@ -27,8 +27,8 @@ #include -#include "talk/app/webrtc/jsepicecandidate.h" -#include "talk/app/webrtc/jsepsessiondescription.h" +#include "webrtc/api/jsepicecandidate.h" +#include "webrtc/api/jsepsessiondescription.h" #include "talk/session/media/mediasession.h" #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" diff --git a/talk/app/webrtc/localaudiosource.cc b/webrtc/api/localaudiosource.cc similarity index 97% rename from talk/app/webrtc/localaudiosource.cc rename to webrtc/api/localaudiosource.cc index 9789fe6f54..7aa05bb36f 100644 --- a/talk/app/webrtc/localaudiosource.cc +++ b/webrtc/api/localaudiosource.cc @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/localaudiosource.h" +#include "webrtc/api/localaudiosource.h" #include -#include "talk/app/webrtc/mediaconstraintsinterface.h" +#include "webrtc/api/mediaconstraintsinterface.h" #include "webrtc/media/base/mediaengine.h" using webrtc::MediaConstraintsInterface; diff --git a/talk/app/webrtc/localaudiosource.h b/webrtc/api/localaudiosource.h similarity index 90% rename from talk/app/webrtc/localaudiosource.h rename to webrtc/api/localaudiosource.h index 67750aa26b..5b6133acec 100644 --- a/talk/app/webrtc/localaudiosource.h +++ b/webrtc/api/localaudiosource.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_ -#define TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_ +#ifndef WEBRTC_API_LOCALAUDIOSOURCE_H_ +#define WEBRTC_API_LOCALAUDIOSOURCE_H_ -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/notifier.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/notifier.h" +#include "webrtc/api/peerconnectioninterface.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/media/base/mediachannel.h" @@ -70,4 +70,4 @@ class LocalAudioSource : public Notifier { } // namespace webrtc -#endif // TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_ +#endif // WEBRTC_API_LOCALAUDIOSOURCE_H_ diff --git a/talk/app/webrtc/localaudiosource_unittest.cc b/webrtc/api/localaudiosource_unittest.cc similarity index 98% rename from talk/app/webrtc/localaudiosource_unittest.cc rename to webrtc/api/localaudiosource_unittest.cc index 5ff1a7a664..01469e20a3 100644 --- a/talk/app/webrtc/localaudiosource_unittest.cc +++ b/webrtc/api/localaudiosource_unittest.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/localaudiosource.h" +#include "webrtc/api/localaudiosource.h" #include #include -#include "talk/app/webrtc/test/fakeconstraints.h" +#include "webrtc/api/test/fakeconstraints.h" #include "webrtc/base/gunit.h" #include "webrtc/media/base/fakemediaengine.h" #include "webrtc/media/base/fakevideorenderer.h" diff --git a/talk/app/webrtc/mediaconstraintsinterface.cc b/webrtc/api/mediaconstraintsinterface.cc similarity index 99% rename from talk/app/webrtc/mediaconstraintsinterface.cc rename to webrtc/api/mediaconstraintsinterface.cc index 40db5dc380..b8575ae21d 100644 --- a/talk/app/webrtc/mediaconstraintsinterface.cc +++ b/webrtc/api/mediaconstraintsinterface.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/mediaconstraintsinterface.h" +#include "webrtc/api/mediaconstraintsinterface.h" #include "webrtc/base/stringencode.h" diff --git a/talk/app/webrtc/mediaconstraintsinterface.h b/webrtc/api/mediaconstraintsinterface.h similarity index 97% rename from talk/app/webrtc/mediaconstraintsinterface.h rename to webrtc/api/mediaconstraintsinterface.h index e06c075d4e..d759e005cb 100644 --- a/talk/app/webrtc/mediaconstraintsinterface.h +++ b/webrtc/api/mediaconstraintsinterface.h @@ -30,8 +30,8 @@ // http://www.w3.org/TR/mediacapture-streams/#mediastreamconstraints and also // used in WebRTC: http://dev.w3.org/2011/webrtc/editor/webrtc.html#constraints. -#ifndef TALK_APP_WEBRTC_MEDIACONSTRAINTSINTERFACE_H_ -#define TALK_APP_WEBRTC_MEDIACONSTRAINTSINTERFACE_H_ +#ifndef WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_ +#define WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_ #include #include @@ -138,4 +138,4 @@ bool FindConstraint(const MediaConstraintsInterface* constraints, } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIACONSTRAINTSINTERFACE_H_ +#endif // WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_ diff --git a/talk/app/webrtc/mediacontroller.cc b/webrtc/api/mediacontroller.cc similarity index 98% rename from talk/app/webrtc/mediacontroller.cc rename to webrtc/api/mediacontroller.cc index 24f5877483..7d94a1a569 100644 --- a/talk/app/webrtc/mediacontroller.cc +++ b/webrtc/api/mediacontroller.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/mediacontroller.h" +#include "webrtc/api/mediacontroller.h" #include "talk/session/media/channelmanager.h" #include "webrtc/base/bind.h" diff --git a/talk/app/webrtc/mediacontroller.h b/webrtc/api/mediacontroller.h similarity index 94% rename from talk/app/webrtc/mediacontroller.h rename to webrtc/api/mediacontroller.h index 1b51be7ca2..f07ddf5e3d 100644 --- a/talk/app/webrtc/mediacontroller.h +++ b/webrtc/api/mediacontroller.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_MEDIACONTROLLER_H_ -#define TALK_APP_WEBRTC_MEDIACONTROLLER_H_ +#ifndef WEBRTC_API_MEDIACONTROLLER_H_ +#define WEBRTC_API_MEDIACONTROLLER_H_ #include "webrtc/base/thread.h" @@ -52,4 +52,4 @@ class MediaControllerInterface { }; } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIACONTROLLER_H_ +#endif // WEBRTC_API_MEDIACONTROLLER_H_ diff --git a/talk/app/webrtc/mediastream.cc b/webrtc/api/mediastream.cc similarity index 98% rename from talk/app/webrtc/mediastream.cc rename to webrtc/api/mediastream.cc index 0d206300df..fe7db9f928 100644 --- a/talk/app/webrtc/mediastream.cc +++ b/webrtc/api/mediastream.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/mediastream.h" +#include "webrtc/api/mediastream.h" #include "webrtc/base/logging.h" namespace webrtc { diff --git a/talk/app/webrtc/mediastream.h b/webrtc/api/mediastream.h similarity index 93% rename from talk/app/webrtc/mediastream.h rename to webrtc/api/mediastream.h index 240512d9c9..94f21eb7b9 100644 --- a/talk/app/webrtc/mediastream.h +++ b/webrtc/api/mediastream.h @@ -27,14 +27,14 @@ // This file contains the implementation of MediaStreamInterface interface. -#ifndef TALK_APP_WEBRTC_MEDIASTREAM_H_ -#define TALK_APP_WEBRTC_MEDIASTREAM_H_ +#ifndef WEBRTC_API_MEDIASTREAM_H_ +#define WEBRTC_API_MEDIASTREAM_H_ #include #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/notifier.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/notifier.h" namespace webrtc { @@ -72,4 +72,4 @@ class MediaStream : public Notifier { } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIASTREAM_H_ +#endif // WEBRTC_API_MEDIASTREAM_H_ diff --git a/talk/app/webrtc/mediastream_unittest.cc b/webrtc/api/mediastream_unittest.cc similarity index 98% rename from talk/app/webrtc/mediastream_unittest.cc rename to webrtc/api/mediastream_unittest.cc index f19b9456a6..5d6d15d78b 100644 --- a/talk/app/webrtc/mediastream_unittest.cc +++ b/webrtc/api/mediastream_unittest.cc @@ -27,9 +27,9 @@ #include -#include "talk/app/webrtc/audiotrack.h" -#include "talk/app/webrtc/mediastream.h" -#include "talk/app/webrtc/videotrack.h" +#include "webrtc/api/audiotrack.h" +#include "webrtc/api/mediastream.h" +#include "webrtc/api/videotrack.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/gunit.h" diff --git a/talk/app/webrtc/mediastreamhandler.cc b/webrtc/api/mediastreamhandler.cc similarity index 100% rename from talk/app/webrtc/mediastreamhandler.cc rename to webrtc/api/mediastreamhandler.cc diff --git a/talk/app/webrtc/mediastreamhandler.h b/webrtc/api/mediastreamhandler.h similarity index 100% rename from talk/app/webrtc/mediastreamhandler.h rename to webrtc/api/mediastreamhandler.h diff --git a/talk/app/webrtc/mediastreaminterface.h b/webrtc/api/mediastreaminterface.h similarity index 98% rename from talk/app/webrtc/mediastreaminterface.h rename to webrtc/api/mediastreaminterface.h index 2d2dc9ac9c..28f243a99d 100644 --- a/talk/app/webrtc/mediastreaminterface.h +++ b/webrtc/api/mediastreaminterface.h @@ -31,8 +31,8 @@ // interfaces must be used only with PeerConnection. PeerConnectionManager // interface provides the factory methods to create MediaStream and MediaTracks. -#ifndef TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_ +#ifndef WEBRTC_API_MEDIASTREAMINTERFACE_H_ +#define WEBRTC_API_MEDIASTREAMINTERFACE_H_ #include #include @@ -293,4 +293,4 @@ class MediaStreamInterface : public rtc::RefCountInterface, } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIASTREAMINTERFACE_H_ +#endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_ diff --git a/talk/app/webrtc/mediastreamobserver.cc b/webrtc/api/mediastreamobserver.cc similarity index 98% rename from talk/app/webrtc/mediastreamobserver.cc rename to webrtc/api/mediastreamobserver.cc index 2650b9a6f7..3f47f137dd 100644 --- a/talk/app/webrtc/mediastreamobserver.cc +++ b/webrtc/api/mediastreamobserver.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/mediastreamobserver.h" +#include "webrtc/api/mediastreamobserver.h" #include diff --git a/talk/app/webrtc/mediastreamobserver.h b/webrtc/api/mediastreamobserver.h similarity index 92% rename from talk/app/webrtc/mediastreamobserver.h rename to webrtc/api/mediastreamobserver.h index 1dd6c4c118..154694246b 100644 --- a/talk/app/webrtc/mediastreamobserver.h +++ b/webrtc/api/mediastreamobserver.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_ +#ifndef WEBRTC_API_MEDIASTREAMOBSERVER_H_ +#define WEBRTC_API_MEDIASTREAMOBSERVER_H_ -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/base/sigslot.h" @@ -62,4 +62,4 @@ class MediaStreamObserver : public ObserverInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_ +#endif // WEBRTC_API_MEDIASTREAMOBSERVER_H_ diff --git a/talk/app/webrtc/mediastreamprovider.h b/webrtc/api/mediastreamprovider.h similarity index 96% rename from talk/app/webrtc/mediastreamprovider.h rename to webrtc/api/mediastreamprovider.h index 103b3f36d4..4ee27aa2d3 100644 --- a/talk/app/webrtc/mediastreamprovider.h +++ b/webrtc/api/mediastreamprovider.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ +#ifndef WEBRTC_API_MEDIASTREAMPROVIDER_H_ +#define WEBRTC_API_MEDIASTREAMPROVIDER_H_ #include "webrtc/base/basictypes.h" #include "webrtc/base/scoped_ptr.h" @@ -105,4 +105,4 @@ class VideoProviderInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ +#endif // WEBRTC_API_MEDIASTREAMPROVIDER_H_ diff --git a/talk/app/webrtc/mediastreamproxy.h b/webrtc/api/mediastreamproxy.h similarity index 91% rename from talk/app/webrtc/mediastreamproxy.h rename to webrtc/api/mediastreamproxy.h index bde7dcfe2d..635f458e8a 100644 --- a/talk/app/webrtc/mediastreamproxy.h +++ b/webrtc/api/mediastreamproxy.h @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_ +#ifndef WEBRTC_API_MEDIASTREAMPROXY_H_ +#define WEBRTC_API_MEDIASTREAMPROXY_H_ -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/proxy.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/proxy.h" namespace webrtc { @@ -51,4 +51,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIASTREAMPROXY_H_ +#endif // WEBRTC_API_MEDIASTREAMPROXY_H_ diff --git a/talk/app/webrtc/mediastreamtrack.h b/webrtc/api/mediastreamtrack.h similarity index 92% rename from talk/app/webrtc/mediastreamtrack.h rename to webrtc/api/mediastreamtrack.h index 2097d9083d..2e9f774709 100644 --- a/talk/app/webrtc/mediastreamtrack.h +++ b/webrtc/api/mediastreamtrack.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_MEDIASTREAMTRACK_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMTRACK_H_ +#ifndef WEBRTC_API_MEDIASTREAMTRACK_H_ +#define WEBRTC_API_MEDIASTREAMTRACK_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/notifier.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/notifier.h" namespace webrtc { @@ -78,4 +78,4 @@ class MediaStreamTrack : public Notifier { } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIASTREAMTRACK_H_ +#endif // WEBRTC_API_MEDIASTREAMTRACK_H_ diff --git a/talk/app/webrtc/mediastreamtrackproxy.h b/webrtc/api/mediastreamtrackproxy.h similarity index 92% rename from talk/app/webrtc/mediastreamtrackproxy.h rename to webrtc/api/mediastreamtrackproxy.h index e99910e24b..eabb0cf0a5 100644 --- a/talk/app/webrtc/mediastreamtrackproxy.h +++ b/webrtc/api/mediastreamtrackproxy.h @@ -28,11 +28,11 @@ // This file includes proxy classes for tracks. The purpose is // to make sure tracks are only accessed from the signaling thread. -#ifndef TALK_APP_WEBRTC_MEDIASTREAMTRACKPROXY_H_ -#define TALK_APP_WEBRTC_MEDIASTREAMTRACKPROXY_H_ +#ifndef WEBRTC_API_MEDIASTREAMTRACKPROXY_H_ +#define WEBRTC_API_MEDIASTREAMTRACKPROXY_H_ -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/proxy.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/proxy.h" namespace webrtc { @@ -74,4 +74,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_MEDIASTREAMTRACKPROXY_H_ +#endif // WEBRTC_API_MEDIASTREAMTRACKPROXY_H_ diff --git a/talk/app/webrtc/notifier.h b/webrtc/api/notifier.h similarity index 94% rename from talk/app/webrtc/notifier.h rename to webrtc/api/notifier.h index ecc16b9788..a6dbba7c36 100644 --- a/talk/app/webrtc/notifier.h +++ b/webrtc/api/notifier.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_NOTIFIER_H_ -#define TALK_APP_WEBRTC_NOTIFIER_H_ +#ifndef WEBRTC_API_NOTIFIER_H_ +#define WEBRTC_API_NOTIFIER_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/base/common.h" namespace webrtc { @@ -74,4 +74,4 @@ class Notifier : public T { } // namespace webrtc -#endif // TALK_APP_WEBRTC_NOTIFIER_H_ +#endif // WEBRTC_API_NOTIFIER_H_ diff --git a/webrtc/api/objc/RTCAudioTrack+Private.h b/webrtc/api/objc/RTCAudioTrack+Private.h index 36f72c7f25..ce3298ee67 100644 --- a/webrtc/api/objc/RTCAudioTrack+Private.h +++ b/webrtc/api/objc/RTCAudioTrack+Private.h @@ -10,7 +10,7 @@ #import "RTCAudioTrack.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCConfiguration+Private.h b/webrtc/api/objc/RTCConfiguration+Private.h index e14f92bcde..001dac6016 100644 --- a/webrtc/api/objc/RTCConfiguration+Private.h +++ b/webrtc/api/objc/RTCConfiguration+Private.h @@ -10,7 +10,7 @@ #import "RTCConfiguration.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCDataChannel+Private.h b/webrtc/api/objc/RTCDataChannel+Private.h index cc44923d05..179192c83f 100644 --- a/webrtc/api/objc/RTCDataChannel+Private.h +++ b/webrtc/api/objc/RTCDataChannel+Private.h @@ -10,7 +10,7 @@ #import "RTCDataChannel.h" -#include "talk/app/webrtc/datachannelinterface.h" +#include "webrtc/api/datachannelinterface.h" #include "webrtc/base/scoped_ref_ptr.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCDataChannelConfiguration+Private.h b/webrtc/api/objc/RTCDataChannelConfiguration+Private.h index e99ba7c973..13478e78e8 100644 --- a/webrtc/api/objc/RTCDataChannelConfiguration+Private.h +++ b/webrtc/api/objc/RTCDataChannelConfiguration+Private.h @@ -10,7 +10,7 @@ #import "RTCDataChannelConfiguration.h" -#include "talk/app/webrtc/datachannelinterface.h" +#include "webrtc/api/datachannelinterface.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCIceCandidate+Private.h b/webrtc/api/objc/RTCIceCandidate+Private.h index ca95a43e3a..b65f113f36 100644 --- a/webrtc/api/objc/RTCIceCandidate+Private.h +++ b/webrtc/api/objc/RTCIceCandidate+Private.h @@ -10,7 +10,7 @@ #import "RTCIceCandidate.h" -#include "talk/app/webrtc/jsep.h" +#include "webrtc/api/jsep.h" #include "webrtc/base/scoped_ptr.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCIceServer+Private.h b/webrtc/api/objc/RTCIceServer+Private.h index 3890567042..556936de81 100644 --- a/webrtc/api/objc/RTCIceServer+Private.h +++ b/webrtc/api/objc/RTCIceServer+Private.h @@ -10,7 +10,7 @@ #import "RTCIceServer.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCMediaConstraints+Private.h b/webrtc/api/objc/RTCMediaConstraints+Private.h index 2c4b722104..fa582ecae5 100644 --- a/webrtc/api/objc/RTCMediaConstraints+Private.h +++ b/webrtc/api/objc/RTCMediaConstraints+Private.h @@ -10,7 +10,7 @@ #import "RTCMediaConstraints.h" -#include "talk/app/webrtc/mediaconstraintsinterface.h" +#include "webrtc/api/mediaconstraintsinterface.h" #include "webrtc/base/scoped_ptr.h" namespace webrtc { diff --git a/webrtc/api/objc/RTCMediaStream+Private.h b/webrtc/api/objc/RTCMediaStream+Private.h index 2c2662b6fa..4c83288190 100644 --- a/webrtc/api/objc/RTCMediaStream+Private.h +++ b/webrtc/api/objc/RTCMediaStream+Private.h @@ -10,7 +10,7 @@ #import "RTCMediaStream.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCMediaStreamTrack+Private.h b/webrtc/api/objc/RTCMediaStreamTrack+Private.h index fcdcdad2ba..155e31228b 100644 --- a/webrtc/api/objc/RTCMediaStreamTrack+Private.h +++ b/webrtc/api/objc/RTCMediaStreamTrack+Private.h @@ -10,7 +10,7 @@ #import "RTCMediaStreamTrack.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/base/scoped_ptr.h" typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) { diff --git a/webrtc/api/objc/RTCPeerConnectionFactory+Private.h b/webrtc/api/objc/RTCPeerConnectionFactory+Private.h index a5f2350ffa..55a473b4cd 100644 --- a/webrtc/api/objc/RTCPeerConnectionFactory+Private.h +++ b/webrtc/api/objc/RTCPeerConnectionFactory+Private.h @@ -10,7 +10,7 @@ #import "RTCPeerConnectionFactory.h" -#include "talk/app/webrtc/peerconnectionfactory.h" +#include "webrtc/api/peerconnectionfactory.h" #include "webrtc/base/scoped_ref_ptr.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCSessionDescription+Private.h b/webrtc/api/objc/RTCSessionDescription+Private.h index aa0314d3f3..b5c0fff671 100644 --- a/webrtc/api/objc/RTCSessionDescription+Private.h +++ b/webrtc/api/objc/RTCSessionDescription+Private.h @@ -10,7 +10,7 @@ #import "RTCSessionDescription.h" -#include "talk/app/webrtc/jsep.h" +#include "webrtc/api/jsep.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCStatsReport+Private.h b/webrtc/api/objc/RTCStatsReport+Private.h index 5b7dc32a74..5ce5801bee 100644 --- a/webrtc/api/objc/RTCStatsReport+Private.h +++ b/webrtc/api/objc/RTCStatsReport+Private.h @@ -10,7 +10,7 @@ #import "RTCStatsReport.h" -#include "talk/app/webrtc/statstypes.h" +#include "webrtc/api/statstypes.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCVideoRendererAdapter+Private.h b/webrtc/api/objc/RTCVideoRendererAdapter+Private.h index 807eea4052..c181b9b41c 100644 --- a/webrtc/api/objc/RTCVideoRendererAdapter+Private.h +++ b/webrtc/api/objc/RTCVideoRendererAdapter+Private.h @@ -10,7 +10,7 @@ #import "RTCVideoRendererAdapter.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #import "RTCVideoRenderer.h" diff --git a/webrtc/api/objc/RTCVideoSource+Private.h b/webrtc/api/objc/RTCVideoSource+Private.h index 2300848610..c363d3390e 100644 --- a/webrtc/api/objc/RTCVideoSource+Private.h +++ b/webrtc/api/objc/RTCVideoSource+Private.h @@ -10,7 +10,7 @@ #import "RTCVideoSource.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/videosourceinterface.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objc/RTCVideoTrack+Private.h b/webrtc/api/objc/RTCVideoTrack+Private.h index 4f55481b55..cd7de48ebc 100644 --- a/webrtc/api/objc/RTCVideoTrack+Private.h +++ b/webrtc/api/objc/RTCVideoTrack+Private.h @@ -10,7 +10,7 @@ #import "RTCVideoTrack.h" -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" NS_ASSUME_NONNULL_BEGIN diff --git a/webrtc/api/objctests/RTCIceCandidateTest.mm b/webrtc/api/objctests/RTCIceCandidateTest.mm index 391db44ae1..2163ce2f62 100644 --- a/webrtc/api/objctests/RTCIceCandidateTest.mm +++ b/webrtc/api/objctests/RTCIceCandidateTest.mm @@ -1,11 +1,28 @@ /* - * Copyright 2015 The WebRTC project authors. All Rights Reserved. + * libjingle + * Copyright 2015 Google Inc. * - * 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. + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * 1. Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * 3. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED + * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF + * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO + * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; + * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, + * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR + * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF + * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #import diff --git a/webrtc/api/objctests/RTCIceServerTest.mm b/webrtc/api/objctests/RTCIceServerTest.mm index 2e6fb25486..1ddb13c20e 100644 --- a/webrtc/api/objctests/RTCIceServerTest.mm +++ b/webrtc/api/objctests/RTCIceServerTest.mm @@ -1,11 +1,28 @@ /* - * Copyright 2015 The WebRTC project authors. All Rights Reserved. + * libjingle + * Copyright 2015 Google Inc. * - * 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. + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * 1. Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * 3. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED + * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF + * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO + * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; + * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, + * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR + * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF + * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #import diff --git a/webrtc/api/objctests/RTCMediaConstraintsTest.mm b/webrtc/api/objctests/RTCMediaConstraintsTest.mm index 44ffe3d033..c1e1886c37 100644 --- a/webrtc/api/objctests/RTCMediaConstraintsTest.mm +++ b/webrtc/api/objctests/RTCMediaConstraintsTest.mm @@ -1,11 +1,28 @@ /* - * Copyright 2015 The WebRTC project authors. All Rights Reserved. + * libjingle + * Copyright 2015 Google Inc. * - * 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. + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * 1. Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * 3. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED + * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF + * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO + * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; + * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, + * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR + * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF + * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #import diff --git a/webrtc/api/objctests/RTCSessionDescriptionTest.mm b/webrtc/api/objctests/RTCSessionDescriptionTest.mm index 2404dedd3a..6eaa36f99f 100644 --- a/webrtc/api/objctests/RTCSessionDescriptionTest.mm +++ b/webrtc/api/objctests/RTCSessionDescriptionTest.mm @@ -1,11 +1,28 @@ /* - * Copyright 2015 The WebRTC project authors. All Rights Reserved. + * libjingle + * Copyright 2015 Google Inc. * - * 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. + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions are met: + * + * 1. Redistributions of source code must retain the above copyright notice, + * this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright notice, + * this list of conditions and the following disclaimer in the documentation + * and/or other materials provided with the distribution. + * 3. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED + * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF + * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO + * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, + * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; + * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, + * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR + * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF + * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #import diff --git a/talk/app/webrtc/peerconnection.cc b/webrtc/api/peerconnection.cc similarity index 98% rename from talk/app/webrtc/peerconnection.cc rename to webrtc/api/peerconnection.cc index c423b0fade..cdc5861ddd 100644 --- a/talk/app/webrtc/peerconnection.cc +++ b/webrtc/api/peerconnection.cc @@ -25,30 +25,30 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/peerconnection.h" +#include "webrtc/api/peerconnection.h" #include #include // for isdigit #include #include -#include "talk/app/webrtc/audiotrack.h" -#include "talk/app/webrtc/dtmfsender.h" -#include "talk/app/webrtc/jsepicecandidate.h" -#include "talk/app/webrtc/jsepsessiondescription.h" -#include "talk/app/webrtc/mediaconstraintsinterface.h" -#include "talk/app/webrtc/mediastream.h" -#include "talk/app/webrtc/mediastreamobserver.h" -#include "talk/app/webrtc/mediastreamproxy.h" -#include "talk/app/webrtc/mediastreamtrackproxy.h" -#include "talk/app/webrtc/remoteaudiosource.h" -#include "talk/app/webrtc/remotevideocapturer.h" -#include "talk/app/webrtc/rtpreceiver.h" -#include "talk/app/webrtc/rtpsender.h" -#include "talk/app/webrtc/streamcollection.h" -#include "talk/app/webrtc/videosource.h" -#include "talk/app/webrtc/videotrack.h" #include "talk/session/media/channelmanager.h" +#include "webrtc/api/audiotrack.h" +#include "webrtc/api/dtmfsender.h" +#include "webrtc/api/jsepicecandidate.h" +#include "webrtc/api/jsepsessiondescription.h" +#include "webrtc/api/mediaconstraintsinterface.h" +#include "webrtc/api/mediastream.h" +#include "webrtc/api/mediastreamobserver.h" +#include "webrtc/api/mediastreamproxy.h" +#include "webrtc/api/mediastreamtrackproxy.h" +#include "webrtc/api/remoteaudiosource.h" +#include "webrtc/api/remotevideocapturer.h" +#include "webrtc/api/rtpreceiver.h" +#include "webrtc/api/rtpsender.h" +#include "webrtc/api/streamcollection.h" +#include "webrtc/api/videosource.h" +#include "webrtc/api/videotrack.h" #include "webrtc/base/arraysize.h" #include "webrtc/base/logging.h" #include "webrtc/base/stringencode.h" diff --git a/talk/app/webrtc/peerconnection.h b/webrtc/api/peerconnection.h similarity index 97% rename from talk/app/webrtc/peerconnection.h rename to webrtc/api/peerconnection.h index dce24fd293..c7de19ddc2 100644 --- a/talk/app/webrtc/peerconnection.h +++ b/webrtc/api/peerconnection.h @@ -25,19 +25,19 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_PEERCONNECTION_H_ -#define TALK_APP_WEBRTC_PEERCONNECTION_H_ +#ifndef WEBRTC_API_PEERCONNECTION_H_ +#define WEBRTC_API_PEERCONNECTION_H_ #include -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/peerconnectionfactory.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/rtpreceiverinterface.h" -#include "talk/app/webrtc/rtpsenderinterface.h" -#include "talk/app/webrtc/statscollector.h" -#include "talk/app/webrtc/streamcollection.h" -#include "talk/app/webrtc/webrtcsession.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/peerconnectionfactory.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/rtpreceiverinterface.h" +#include "webrtc/api/rtpsenderinterface.h" +#include "webrtc/api/statscollector.h" +#include "webrtc/api/streamcollection.h" +#include "webrtc/api/webrtcsession.h" #include "webrtc/base/scoped_ptr.h" namespace webrtc { @@ -396,4 +396,4 @@ class PeerConnection : public PeerConnectionInterface, } // namespace webrtc -#endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ +#endif // WEBRTC_API_PEERCONNECTION_H_ diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc similarity index 99% rename from talk/app/webrtc/peerconnection_unittest.cc rename to webrtc/api/peerconnection_unittest.cc index cad13e2441..c1e7e3dfd9 100644 --- a/talk/app/webrtc/peerconnection_unittest.cc +++ b/webrtc/api/peerconnection_unittest.cc @@ -33,21 +33,21 @@ #include #include -#include "talk/app/webrtc/dtmfsender.h" -#include "talk/app/webrtc/fakemetricsobserver.h" -#include "talk/app/webrtc/localaudiosource.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnection.h" -#include "talk/app/webrtc/peerconnectionfactory.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/test/fakeaudiocapturemodule.h" -#include "talk/app/webrtc/test/fakeconstraints.h" -#include "talk/app/webrtc/test/fakedtlsidentitystore.h" -#include "talk/app/webrtc/test/fakeperiodicvideocapturer.h" -#include "talk/app/webrtc/test/fakevideotrackrenderer.h" -#include "talk/app/webrtc/test/mockpeerconnectionobservers.h" -#include "talk/app/webrtc/videosourceinterface.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/dtmfsender.h" +#include "webrtc/api/fakemetricsobserver.h" +#include "webrtc/api/localaudiosource.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnection.h" +#include "webrtc/api/peerconnectionfactory.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/test/fakeaudiocapturemodule.h" +#include "webrtc/api/test/fakeconstraints.h" +#include "webrtc/api/test/fakedtlsidentitystore.h" +#include "webrtc/api/test/fakeperiodicvideocapturer.h" +#include "webrtc/api/test/fakevideotrackrenderer.h" +#include "webrtc/api/test/mockpeerconnectionobservers.h" +#include "webrtc/api/videosourceinterface.h" #include "webrtc/base/gunit.h" #include "webrtc/base/physicalsocketserver.h" #include "webrtc/base/scoped_ptr.h" diff --git a/talk/peerconnection_unittests.isolate b/webrtc/api/peerconnection_unittests.isolate similarity index 100% rename from talk/peerconnection_unittests.isolate rename to webrtc/api/peerconnection_unittests.isolate diff --git a/talk/app/webrtc/peerconnectionendtoend_unittest.cc b/webrtc/api/peerconnectionendtoend_unittest.cc similarity index 98% rename from talk/app/webrtc/peerconnectionendtoend_unittest.cc rename to webrtc/api/peerconnectionendtoend_unittest.cc index f71ce61eb2..adcfe57102 100644 --- a/talk/app/webrtc/peerconnectionendtoend_unittest.cc +++ b/webrtc/api/peerconnectionendtoend_unittest.cc @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/test/peerconnectiontestwrapper.h" +#include "webrtc/api/test/peerconnectiontestwrapper.h" // Notice that mockpeerconnectionobservers.h must be included after the above! -#include "talk/app/webrtc/test/mockpeerconnectionobservers.h" +#include "webrtc/api/test/mockpeerconnectionobservers.h" #ifdef WEBRTC_ANDROID -#include "talk/app/webrtc/test/androidtestinitializer.h" +#include "webrtc/api/test/androidtestinitializer.h" #endif #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/webrtc/api/peerconnectionfactory.cc similarity index 95% rename from talk/app/webrtc/peerconnectionfactory.cc rename to webrtc/api/peerconnectionfactory.cc index a781eb3e25..66545930e5 100644 --- a/talk/app/webrtc/peerconnectionfactory.cc +++ b/webrtc/api/peerconnectionfactory.cc @@ -1,6 +1,6 @@ /* * libjingle - * Copyright 2004--2011 Google Inc. + * Copyright 2004 Google Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: @@ -25,21 +25,21 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/peerconnectionfactory.h" +#include "webrtc/api/peerconnectionfactory.h" #include -#include "talk/app/webrtc/audiotrack.h" -#include "talk/app/webrtc/localaudiosource.h" -#include "talk/app/webrtc/mediastream.h" -#include "talk/app/webrtc/mediastreamproxy.h" -#include "talk/app/webrtc/mediastreamtrackproxy.h" -#include "talk/app/webrtc/peerconnection.h" -#include "talk/app/webrtc/peerconnectionfactoryproxy.h" -#include "talk/app/webrtc/peerconnectionproxy.h" -#include "talk/app/webrtc/videosource.h" -#include "talk/app/webrtc/videosourceproxy.h" -#include "talk/app/webrtc/videotrack.h" +#include "webrtc/api/audiotrack.h" +#include "webrtc/api/localaudiosource.h" +#include "webrtc/api/mediastream.h" +#include "webrtc/api/mediastreamproxy.h" +#include "webrtc/api/mediastreamtrackproxy.h" +#include "webrtc/api/peerconnection.h" +#include "webrtc/api/peerconnectionfactoryproxy.h" +#include "webrtc/api/peerconnectionproxy.h" +#include "webrtc/api/videosource.h" +#include "webrtc/api/videosourceproxy.h" +#include "webrtc/api/videotrack.h" #include "webrtc/base/bind.h" #include "webrtc/media/webrtc/webrtcmediaengine.h" #include "webrtc/media/webrtc/webrtcvideodecoderfactory.h" diff --git a/talk/app/webrtc/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h similarity index 93% rename from talk/app/webrtc/peerconnectionfactory.h rename to webrtc/api/peerconnectionfactory.h index a38218aebe..701173668c 100644 --- a/talk/app/webrtc/peerconnectionfactory.h +++ b/webrtc/api/peerconnectionfactory.h @@ -25,16 +25,16 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_ -#define TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_ +#ifndef WEBRTC_API_PEERCONNECTIONFACTORY_H_ +#define WEBRTC_API_PEERCONNECTIONFACTORY_H_ #include -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/mediacontroller.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" #include "talk/session/media/channelmanager.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/mediacontroller.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnectioninterface.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/base/thread.h" @@ -129,4 +129,4 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_ +#endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_ diff --git a/talk/app/webrtc/peerconnectionfactory_unittest.cc b/webrtc/api/peerconnectionfactory_unittest.cc similarity index 98% rename from talk/app/webrtc/peerconnectionfactory_unittest.cc rename to webrtc/api/peerconnectionfactory_unittest.cc index d0d2f00101..a526ea5e63 100644 --- a/talk/app/webrtc/peerconnectionfactory_unittest.cc +++ b/webrtc/api/peerconnectionfactory_unittest.cc @@ -28,14 +28,14 @@ #include #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnectionfactory.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnectionfactory.h" #ifdef WEBRTC_ANDROID -#include "talk/app/webrtc/test/androidtestinitializer.h" +#include "webrtc/api/test/androidtestinitializer.h" #endif -#include "talk/app/webrtc/test/fakedtlsidentitystore.h" -#include "talk/app/webrtc/test/fakevideotrackrenderer.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/test/fakedtlsidentitystore.h" +#include "webrtc/api/test/fakevideotrackrenderer.h" +#include "webrtc/api/videosourceinterface.h" #include "webrtc/base/gunit.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread.h" diff --git a/talk/app/webrtc/peerconnectionfactoryproxy.h b/webrtc/api/peerconnectionfactoryproxy.h similarity index 93% rename from talk/app/webrtc/peerconnectionfactoryproxy.h rename to webrtc/api/peerconnectionfactoryproxy.h index 1d0b6aab5a..65f09693c6 100644 --- a/talk/app/webrtc/peerconnectionfactoryproxy.h +++ b/webrtc/api/peerconnectionfactoryproxy.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_PEERCONNECTIONFACTORYPROXY_H_ -#define TALK_APP_WEBRTC_PEERCONNECTIONFACTORYPROXY_H_ +#ifndef WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_ +#define WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_ #include #include -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/proxy.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/proxy.h" #include "webrtc/base/bind.h" namespace webrtc { @@ -83,4 +83,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORYPROXY_H_ +#endif // WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_ diff --git a/talk/app/webrtc/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h similarity index 97% rename from talk/app/webrtc/peerconnectioninterface.h rename to webrtc/api/peerconnectioninterface.h index 940f0fb9e6..5cdb097507 100644 --- a/talk/app/webrtc/peerconnectioninterface.h +++ b/webrtc/api/peerconnectioninterface.h @@ -65,23 +65,23 @@ // 7. Once a candidate have been found PeerConnection will call the observer // function OnIceCandidate. Send these candidates to the remote peer. -#ifndef TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ -#define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ +#ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_ +#define WEBRTC_API_PEERCONNECTIONINTERFACE_H_ #include #include #include -#include "talk/app/webrtc/datachannelinterface.h" -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/dtmfsenderinterface.h" -#include "talk/app/webrtc/jsep.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/rtpreceiverinterface.h" -#include "talk/app/webrtc/rtpsenderinterface.h" -#include "talk/app/webrtc/statstypes.h" -#include "talk/app/webrtc/umametrics.h" +#include "webrtc/api/datachannelinterface.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/dtmfsenderinterface.h" +#include "webrtc/api/jsep.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/rtpreceiverinterface.h" +#include "webrtc/api/rtpsenderinterface.h" +#include "webrtc/api/statstypes.h" +#include "webrtc/api/umametrics.h" #include "webrtc/base/fileutils.h" #include "webrtc/base/network.h" #include "webrtc/base/rtccertificate.h" @@ -619,4 +619,4 @@ CreatePeerConnectionFactory( } // namespace webrtc -#endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ +#endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc similarity index 99% rename from talk/app/webrtc/peerconnectioninterface_unittest.cc rename to webrtc/api/peerconnectioninterface_unittest.cc index c29718fe34..b93cd7787f 100644 --- a/talk/app/webrtc/peerconnectioninterface_unittest.cc +++ b/webrtc/api/peerconnectioninterface_unittest.cc @@ -28,25 +28,25 @@ #include #include -#include "talk/app/webrtc/audiotrack.h" -#include "talk/app/webrtc/jsepsessiondescription.h" -#include "talk/app/webrtc/mediastream.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnection.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/rtpreceiverinterface.h" -#include "talk/app/webrtc/rtpsenderinterface.h" -#include "talk/app/webrtc/streamcollection.h" -#ifdef WEBRTC_ANDROID -#include "talk/app/webrtc/test/androidtestinitializer.h" -#endif -#include "talk/app/webrtc/test/fakeconstraints.h" -#include "talk/app/webrtc/test/fakedtlsidentitystore.h" -#include "talk/app/webrtc/test/mockpeerconnectionobservers.h" -#include "talk/app/webrtc/test/testsdpstrings.h" -#include "talk/app/webrtc/videosource.h" -#include "talk/app/webrtc/videotrack.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/audiotrack.h" +#include "webrtc/api/jsepsessiondescription.h" +#include "webrtc/api/mediastream.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnection.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/rtpreceiverinterface.h" +#include "webrtc/api/rtpsenderinterface.h" +#include "webrtc/api/streamcollection.h" +#ifdef WEBRTC_ANDROID +#include "webrtc/api/test/androidtestinitializer.h" +#endif +#include "webrtc/api/test/fakeconstraints.h" +#include "webrtc/api/test/fakedtlsidentitystore.h" +#include "webrtc/api/test/mockpeerconnectionobservers.h" +#include "webrtc/api/test/testsdpstrings.h" +#include "webrtc/api/videosource.h" +#include "webrtc/api/videotrack.h" #include "webrtc/base/gunit.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/ssladapter.h" diff --git a/talk/app/webrtc/peerconnectionproxy.h b/webrtc/api/peerconnectionproxy.h similarity index 94% rename from talk/app/webrtc/peerconnectionproxy.h rename to webrtc/api/peerconnectionproxy.h index d76a6b87f3..9faf01484a 100644 --- a/talk/app/webrtc/peerconnectionproxy.h +++ b/webrtc/api/peerconnectionproxy.h @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_ -#define TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_ +#ifndef WEBRTC_API_PEERCONNECTIONPROXY_H_ +#define WEBRTC_API_PEERCONNECTIONPROXY_H_ -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/proxy.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/proxy.h" namespace webrtc { @@ -85,4 +85,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_PEERCONNECTIONPROXY_H_ +#endif // WEBRTC_API_PEERCONNECTIONPROXY_H_ diff --git a/talk/app/webrtc/portallocatorfactory.cc b/webrtc/api/portallocatorfactory.cc similarity index 96% rename from talk/app/webrtc/portallocatorfactory.cc rename to webrtc/api/portallocatorfactory.cc index 64d714cd50..a5a98b0e97 100644 --- a/talk/app/webrtc/portallocatorfactory.cc +++ b/webrtc/api/portallocatorfactory.cc @@ -27,4 +27,4 @@ // TODO(deadbeef): Remove this file once chromium build files no longer // reference it. -#include "talk/app/webrtc/portallocatorfactory.h" +#include "webrtc/api/portallocatorfactory.h" diff --git a/talk/app/webrtc/portallocatorfactory.h b/webrtc/api/portallocatorfactory.h similarity index 91% rename from talk/app/webrtc/portallocatorfactory.h rename to webrtc/api/portallocatorfactory.h index bb6cf4741f..bce71311f5 100644 --- a/talk/app/webrtc/portallocatorfactory.h +++ b/webrtc/api/portallocatorfactory.h @@ -27,7 +27,7 @@ // TODO(deadbeef): Remove this file once chromium build files no longer // reference it. -#ifndef TALK_APP_WEBRTC_PORTALLOCATORFACTORY_H_ -#define TALK_APP_WEBRTC_PORTALLOCATORFACTORY_H_ +#ifndef WEBRTC_API_PORTALLOCATORFACTORY_H_ +#define WEBRTC_API_PORTALLOCATORFACTORY_H_ -#endif // TALK_APP_WEBRTC_PORTALLOCATORFACTORY_H_ +#endif // WEBRTC_API_PORTALLOCATORFACTORY_H_ diff --git a/talk/app/webrtc/proxy.h b/webrtc/api/proxy.h similarity index 99% rename from talk/app/webrtc/proxy.h rename to webrtc/api/proxy.h index 76a5c1eff2..384e1897ef 100644 --- a/talk/app/webrtc/proxy.h +++ b/webrtc/api/proxy.h @@ -52,8 +52,8 @@ // // The proxy can be created using TestProxy::Create(Thread*, TestInterface*). -#ifndef TALK_APP_WEBRTC_PROXY_H_ -#define TALK_APP_WEBRTC_PROXY_H_ +#ifndef WEBRTC_API_PROXY_H_ +#define WEBRTC_API_PROXY_H_ #include "webrtc/base/event.h" #include "webrtc/base/thread.h" @@ -388,4 +388,4 @@ class MethodCall5 : public rtc::Message, } // namespace webrtc -#endif // TALK_APP_WEBRTC_PROXY_H_ +#endif // WEBRTC_API_PROXY_H_ diff --git a/talk/app/webrtc/proxy_unittest.cc b/webrtc/api/proxy_unittest.cc similarity index 99% rename from talk/app/webrtc/proxy_unittest.cc rename to webrtc/api/proxy_unittest.cc index 6fc89a5d10..8fa7363869 100644 --- a/talk/app/webrtc/proxy_unittest.cc +++ b/webrtc/api/proxy_unittest.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/proxy.h" +#include "webrtc/api/proxy.h" #include diff --git a/talk/app/webrtc/remoteaudiosource.cc b/webrtc/api/remoteaudiosource.cc similarity index 98% rename from talk/app/webrtc/remoteaudiosource.cc rename to webrtc/api/remoteaudiosource.cc index e904dd9192..9a0900d920 100644 --- a/talk/app/webrtc/remoteaudiosource.cc +++ b/webrtc/api/remoteaudiosource.cc @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/remoteaudiosource.h" +#include "webrtc/api/remoteaudiosource.h" #include #include #include -#include "talk/app/webrtc/mediastreamprovider.h" +#include "webrtc/api/mediastreamprovider.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/thread.h" diff --git a/talk/app/webrtc/remoteaudiosource.h b/webrtc/api/remoteaudiosource.h similarity index 93% rename from talk/app/webrtc/remoteaudiosource.h rename to webrtc/api/remoteaudiosource.h index 0e28157459..a46b13044c 100644 --- a/talk/app/webrtc/remoteaudiosource.h +++ b/webrtc/api/remoteaudiosource.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_ -#define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_ +#ifndef WEBRTC_API_REMOTEAUDIOSOURCE_H_ +#define WEBRTC_API_REMOTEAUDIOSOURCE_H_ #include #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/notifier.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/notifier.h" #include "webrtc/audio/audio_sink.h" #include "webrtc/base/criticalsection.h" #include "webrtc/media/base/audiorenderer.h" @@ -93,4 +93,4 @@ class RemoteAudioSource : public Notifier { } // namespace webrtc -#endif // TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_ +#endif // WEBRTC_API_REMOTEAUDIOSOURCE_H_ diff --git a/talk/app/webrtc/remoteaudiotrack.cc b/webrtc/api/remoteaudiotrack.cc similarity index 100% rename from talk/app/webrtc/remoteaudiotrack.cc rename to webrtc/api/remoteaudiotrack.cc diff --git a/talk/app/webrtc/remoteaudiotrack.h b/webrtc/api/remoteaudiotrack.h similarity index 100% rename from talk/app/webrtc/remoteaudiotrack.h rename to webrtc/api/remoteaudiotrack.h diff --git a/talk/app/webrtc/remotevideocapturer.cc b/webrtc/api/remotevideocapturer.cc similarity index 98% rename from talk/app/webrtc/remotevideocapturer.cc rename to webrtc/api/remotevideocapturer.cc index 7e69eedf8e..b7be8f88dd 100644 --- a/talk/app/webrtc/remotevideocapturer.cc +++ b/webrtc/api/remotevideocapturer.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/remotevideocapturer.h" +#include "webrtc/api/remotevideocapturer.h" #include "webrtc/base/logging.h" #include "webrtc/media/base/videoframe.h" diff --git a/talk/app/webrtc/remotevideocapturer.h b/webrtc/api/remotevideocapturer.h similarity index 93% rename from talk/app/webrtc/remotevideocapturer.h rename to webrtc/api/remotevideocapturer.h index 02f48b7c01..15c113425b 100644 --- a/talk/app/webrtc/remotevideocapturer.h +++ b/webrtc/api/remotevideocapturer.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_REMOTEVIDEOCAPTURER_H_ -#define TALK_APP_WEBRTC_REMOTEVIDEOCAPTURER_H_ +#ifndef WEBRTC_API_REMOTEVIDEOCAPTURER_H_ +#define WEBRTC_API_REMOTEVIDEOCAPTURER_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/media/base/videocapturer.h" #include "webrtc/media/base/videorenderer.h" @@ -62,4 +62,4 @@ class RemoteVideoCapturer : public cricket::VideoCapturer { } // namespace webrtc -#endif // TALK_APP_WEBRTC_REMOTEVIDEOCAPTURER_H_ +#endif // WEBRTC_API_REMOTEVIDEOCAPTURER_H_ diff --git a/talk/app/webrtc/remotevideocapturer_unittest.cc b/webrtc/api/remotevideocapturer_unittest.cc similarity index 98% rename from talk/app/webrtc/remotevideocapturer_unittest.cc rename to webrtc/api/remotevideocapturer_unittest.cc index 8e79325bff..f8906e30e2 100644 --- a/talk/app/webrtc/remotevideocapturer_unittest.cc +++ b/webrtc/api/remotevideocapturer_unittest.cc @@ -27,7 +27,7 @@ #include -#include "talk/app/webrtc/remotevideocapturer.h" +#include "webrtc/api/remotevideocapturer.h" #include "webrtc/base/gunit.h" #include "webrtc/media/webrtc/webrtcvideoframe.h" diff --git a/talk/app/webrtc/rtpreceiver.cc b/webrtc/api/rtpreceiver.cc similarity index 97% rename from talk/app/webrtc/rtpreceiver.cc rename to webrtc/api/rtpreceiver.cc index 34efe5c415..11d074a7b4 100644 --- a/talk/app/webrtc/rtpreceiver.cc +++ b/webrtc/api/rtpreceiver.cc @@ -25,9 +25,9 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/rtpreceiver.h" +#include "webrtc/api/rtpreceiver.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/videosourceinterface.h" namespace webrtc { diff --git a/talk/app/webrtc/rtpreceiver.h b/webrtc/api/rtpreceiver.h similarity index 93% rename from talk/app/webrtc/rtpreceiver.h rename to webrtc/api/rtpreceiver.h index db021baf68..016ec6a986 100644 --- a/talk/app/webrtc/rtpreceiver.h +++ b/webrtc/api/rtpreceiver.h @@ -29,13 +29,13 @@ // An RtpReceiver associates a MediaStreamTrackInterface with an underlying // transport (provided by AudioProviderInterface/VideoProviderInterface) -#ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_ -#define TALK_APP_WEBRTC_RTPRECEIVER_H_ +#ifndef WEBRTC_API_RTPRECEIVER_H_ +#define WEBRTC_API_RTPRECEIVER_H_ #include -#include "talk/app/webrtc/mediastreamprovider.h" -#include "talk/app/webrtc/rtpreceiverinterface.h" +#include "webrtc/api/mediastreamprovider.h" +#include "webrtc/api/rtpreceiverinterface.h" #include "webrtc/base/basictypes.h" namespace webrtc { @@ -101,4 +101,4 @@ class VideoRtpReceiver : public rtc::RefCountedObject { } // namespace webrtc -#endif // TALK_APP_WEBRTC_RTPRECEIVER_H_ +#endif // WEBRTC_API_RTPRECEIVER_H_ diff --git a/talk/app/webrtc/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h similarity index 90% rename from talk/app/webrtc/rtpreceiverinterface.h rename to webrtc/api/rtpreceiverinterface.h index 120a50f18c..961d8697e1 100644 --- a/talk/app/webrtc/rtpreceiverinterface.h +++ b/webrtc/api/rtpreceiverinterface.h @@ -28,13 +28,13 @@ // This file contains interfaces for RtpReceivers // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface -#ifndef TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_ -#define TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_ +#ifndef WEBRTC_API_RTPRECEIVERINTERFACE_H_ +#define WEBRTC_API_RTPRECEIVERINTERFACE_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/proxy.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/proxy.h" #include "webrtc/base/refcount.h" #include "webrtc/base/scoped_ref_ptr.h" @@ -63,4 +63,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_RTPRECEIVERINTERFACE_H_ +#endif // WEBRTC_API_RTPRECEIVERINTERFACE_H_ diff --git a/talk/app/webrtc/rtpsender.cc b/webrtc/api/rtpsender.cc similarity index 98% rename from talk/app/webrtc/rtpsender.cc rename to webrtc/api/rtpsender.cc index a30bf0b163..f20f464305 100644 --- a/talk/app/webrtc/rtpsender.cc +++ b/webrtc/api/rtpsender.cc @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/rtpsender.h" +#include "webrtc/api/rtpsender.h" -#include "talk/app/webrtc/localaudiosource.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/localaudiosource.h" +#include "webrtc/api/videosourceinterface.h" #include "webrtc/base/helpers.h" namespace webrtc { diff --git a/talk/app/webrtc/rtpsender.h b/webrtc/api/rtpsender.h similarity index 96% rename from talk/app/webrtc/rtpsender.h rename to webrtc/api/rtpsender.h index c68f64be40..45b765de9f 100644 --- a/talk/app/webrtc/rtpsender.h +++ b/webrtc/api/rtpsender.h @@ -29,14 +29,14 @@ // An RtpSender associates a MediaStreamTrackInterface with an underlying // transport (provided by AudioProviderInterface/VideoProviderInterface) -#ifndef TALK_APP_WEBRTC_RTPSENDER_H_ -#define TALK_APP_WEBRTC_RTPSENDER_H_ +#ifndef WEBRTC_API_RTPSENDER_H_ +#define WEBRTC_API_RTPSENDER_H_ #include -#include "talk/app/webrtc/mediastreamprovider.h" -#include "talk/app/webrtc/rtpsenderinterface.h" -#include "talk/app/webrtc/statscollector.h" +#include "webrtc/api/mediastreamprovider.h" +#include "webrtc/api/rtpsenderinterface.h" +#include "webrtc/api/statscollector.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/scoped_ptr.h" @@ -192,4 +192,4 @@ class VideoRtpSender : public ObserverInterface, } // namespace webrtc -#endif // TALK_APP_WEBRTC_RTPSENDER_H_ +#endif // WEBRTC_API_RTPSENDER_H_ diff --git a/talk/app/webrtc/rtpsenderinterface.h b/webrtc/api/rtpsenderinterface.h similarity index 93% rename from talk/app/webrtc/rtpsenderinterface.h rename to webrtc/api/rtpsenderinterface.h index f96ff1ef6b..740e9857d7 100644 --- a/talk/app/webrtc/rtpsenderinterface.h +++ b/webrtc/api/rtpsenderinterface.h @@ -28,14 +28,14 @@ // This file contains interfaces for RtpSenders // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface -#ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ -#define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ +#ifndef WEBRTC_API_RTPSENDERINTERFACE_H_ +#define WEBRTC_API_RTPSENDERINTERFACE_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/proxy.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/proxy.h" #include "webrtc/base/refcount.h" #include "webrtc/base/scoped_ref_ptr.h" @@ -87,4 +87,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ +#endif // WEBRTC_API_RTPSENDERINTERFACE_H_ diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/webrtc/api/rtpsenderreceiver_unittest.cc similarity index 98% rename from talk/app/webrtc/rtpsenderreceiver_unittest.cc rename to webrtc/api/rtpsenderreceiver_unittest.cc index bcd9ea0350..faca6579f8 100644 --- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc +++ b/webrtc/api/rtpsenderreceiver_unittest.cc @@ -28,16 +28,16 @@ #include #include -#include "talk/app/webrtc/audiotrack.h" -#include "talk/app/webrtc/mediastream.h" -#include "talk/app/webrtc/remoteaudiosource.h" -#include "talk/app/webrtc/rtpreceiver.h" -#include "talk/app/webrtc/rtpsender.h" -#include "talk/app/webrtc/streamcollection.h" -#include "talk/app/webrtc/videosource.h" -#include "talk/app/webrtc/videotrack.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/api/audiotrack.h" +#include "webrtc/api/mediastream.h" +#include "webrtc/api/remoteaudiosource.h" +#include "webrtc/api/rtpreceiver.h" +#include "webrtc/api/rtpsender.h" +#include "webrtc/api/streamcollection.h" +#include "webrtc/api/videosource.h" +#include "webrtc/api/videotrack.h" #include "webrtc/base/gunit.h" #include "webrtc/media/base/fakevideocapturer.h" #include "webrtc/media/base/mediachannel.h" diff --git a/talk/app/webrtc/sctputils.cc b/webrtc/api/sctputils.cc similarity index 99% rename from talk/app/webrtc/sctputils.cc rename to webrtc/api/sctputils.cc index 2239599511..84cb2931a6 100644 --- a/talk/app/webrtc/sctputils.cc +++ b/webrtc/api/sctputils.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/sctputils.h" +#include "webrtc/api/sctputils.h" #include "webrtc/base/buffer.h" #include "webrtc/base/bytebuffer.h" diff --git a/talk/app/webrtc/sctputils.h b/webrtc/api/sctputils.h similarity index 93% rename from talk/app/webrtc/sctputils.h rename to webrtc/api/sctputils.h index f16873c4c3..a3bdb5c16a 100644 --- a/talk/app/webrtc/sctputils.h +++ b/webrtc/api/sctputils.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_SCTPUTILS_H_ -#define TALK_APP_WEBRTC_SCTPUTILS_H_ +#ifndef WEBRTC_API_SCTPUTILS_H_ +#define WEBRTC_API_SCTPUTILS_H_ #include -#include "talk/app/webrtc/datachannelinterface.h" +#include "webrtc/api/datachannelinterface.h" namespace rtc { class Buffer; @@ -55,4 +55,4 @@ bool WriteDataChannelOpenMessage(const std::string& label, void WriteDataChannelOpenAckMessage(rtc::Buffer* payload); } // namespace webrtc -#endif // TALK_APP_WEBRTC_SCTPUTILS_H_ +#endif // WEBRTC_API_SCTPUTILS_H_ diff --git a/talk/app/webrtc/sctputils_unittest.cc b/webrtc/api/sctputils_unittest.cc similarity index 99% rename from talk/app/webrtc/sctputils_unittest.cc rename to webrtc/api/sctputils_unittest.cc index e0e203f5cd..8e29d4c8f6 100644 --- a/talk/app/webrtc/sctputils_unittest.cc +++ b/webrtc/api/sctputils_unittest.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/sctputils.h" +#include "webrtc/api/sctputils.h" #include "webrtc/base/bytebuffer.h" #include "webrtc/base/gunit.h" diff --git a/talk/app/webrtc/statscollector.cc b/webrtc/api/statscollector.cc similarity index 99% rename from talk/app/webrtc/statscollector.cc rename to webrtc/api/statscollector.cc index 883766a36c..c326ea14f9 100644 --- a/talk/app/webrtc/statscollector.cc +++ b/webrtc/api/statscollector.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/statscollector.h" +#include "webrtc/api/statscollector.h" #include #include -#include "talk/app/webrtc/peerconnection.h" +#include "webrtc/api/peerconnection.h" #include "talk/session/media/channel.h" #include "webrtc/base/base64.h" #include "webrtc/base/checks.h" diff --git a/talk/app/webrtc/statscollector.h b/webrtc/api/statscollector.h similarity index 95% rename from talk/app/webrtc/statscollector.h rename to webrtc/api/statscollector.h index 56db79de20..caeac82b03 100644 --- a/talk/app/webrtc/statscollector.h +++ b/webrtc/api/statscollector.h @@ -28,17 +28,17 @@ // This file contains a class used for gathering statistics from an ongoing // libjingle PeerConnection. -#ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_ -#define TALK_APP_WEBRTC_STATSCOLLECTOR_H_ +#ifndef WEBRTC_API_STATSCOLLECTOR_H_ +#define WEBRTC_API_STATSCOLLECTOR_H_ #include #include #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/statstypes.h" -#include "talk/app/webrtc/webrtcsession.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/statstypes.h" +#include "webrtc/api/webrtcsession.h" namespace webrtc { @@ -166,4 +166,4 @@ class StatsCollector { } // namespace webrtc -#endif // TALK_APP_WEBRTC_STATSCOLLECTOR_H_ +#endif // WEBRTC_API_STATSCOLLECTOR_H_ diff --git a/talk/app/webrtc/statscollector_unittest.cc b/webrtc/api/statscollector_unittest.cc similarity index 99% rename from talk/app/webrtc/statscollector_unittest.cc rename to webrtc/api/statscollector_unittest.cc index 1b16b0c4fa..b99aa12430 100644 --- a/talk/app/webrtc/statscollector_unittest.cc +++ b/webrtc/api/statscollector_unittest.cc @@ -29,18 +29,18 @@ #include -#include "talk/app/webrtc/statscollector.h" +#include "webrtc/api/statscollector.h" -#include "talk/app/webrtc/mediastream.h" -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/mediastreamtrack.h" -#include "talk/app/webrtc/peerconnection.h" -#include "talk/app/webrtc/peerconnectionfactory.h" -#include "talk/app/webrtc/test/fakedatachannelprovider.h" -#include "talk/app/webrtc/videotrack.h" #include "talk/session/media/channelmanager.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" +#include "webrtc/api/mediastream.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/mediastreamtrack.h" +#include "webrtc/api/peerconnection.h" +#include "webrtc/api/peerconnectionfactory.h" +#include "webrtc/api/test/fakedatachannelprovider.h" +#include "webrtc/api/videotrack.h" #include "webrtc/base/base64.h" #include "webrtc/base/fakesslidentity.h" #include "webrtc/base/gunit.h" diff --git a/talk/app/webrtc/statstypes.cc b/webrtc/api/statstypes.cc similarity index 99% rename from talk/app/webrtc/statstypes.cc rename to webrtc/api/statstypes.cc index 954f90f496..ab58cb11d4 100644 --- a/talk/app/webrtc/statstypes.cc +++ b/webrtc/api/statstypes.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/statstypes.h" +#include "webrtc/api/statstypes.h" #include diff --git a/talk/app/webrtc/statstypes.h b/webrtc/api/statstypes.h similarity index 99% rename from talk/app/webrtc/statstypes.h rename to webrtc/api/statstypes.h index 6f547e1a52..753cba61da 100644 --- a/talk/app/webrtc/statstypes.h +++ b/webrtc/api/statstypes.h @@ -28,8 +28,8 @@ // This file contains structures used for retrieving statistics from an ongoing // libjingle session. -#ifndef TALK_APP_WEBRTC_STATSTYPES_H_ -#define TALK_APP_WEBRTC_STATSTYPES_H_ +#ifndef WEBRTC_API_STATSTYPES_H_ +#define WEBRTC_API_STATSTYPES_H_ #include #include @@ -416,4 +416,4 @@ class StatsCollection { } // namespace webrtc -#endif // TALK_APP_WEBRTC_STATSTYPES_H_ +#endif // WEBRTC_API_STATSTYPES_H_ diff --git a/talk/app/webrtc/streamcollection.h b/webrtc/api/streamcollection.h similarity index 95% rename from talk/app/webrtc/streamcollection.h rename to webrtc/api/streamcollection.h index 07a30a68c8..fc9a891066 100644 --- a/talk/app/webrtc/streamcollection.h +++ b/webrtc/api/streamcollection.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_STREAMCOLLECTION_H_ -#define TALK_APP_WEBRTC_STREAMCOLLECTION_H_ +#ifndef WEBRTC_API_STREAMCOLLECTION_H_ +#define WEBRTC_API_STREAMCOLLECTION_H_ #include #include -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/peerconnectioninterface.h" namespace webrtc { @@ -122,4 +122,4 @@ class StreamCollection : public StreamCollectionInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_STREAMCOLLECTION_H_ +#endif // WEBRTC_API_STREAMCOLLECTION_H_ diff --git a/talk/app/webrtc/test/DEPS b/webrtc/api/test/DEPS similarity index 100% rename from talk/app/webrtc/test/DEPS rename to webrtc/api/test/DEPS diff --git a/talk/app/webrtc/test/androidtestinitializer.cc b/webrtc/api/test/androidtestinitializer.cc similarity index 94% rename from talk/app/webrtc/test/androidtestinitializer.cc rename to webrtc/api/test/androidtestinitializer.cc index 883c2d8178..17118c05c6 100644 --- a/talk/app/webrtc/test/androidtestinitializer.cc +++ b/webrtc/api/test/androidtestinitializer.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/test/androidtestinitializer.h" +#include "webrtc/api/test/androidtestinitializer.h" #include @@ -36,8 +36,8 @@ #include "base/android/context_utils.h" #include "base/android/jni_android.h" -#include "talk/app/webrtc/java/jni/classreferenceholder.h" -#include "talk/app/webrtc/java/jni/jni_helpers.h" +#include "webrtc/api/java/jni/classreferenceholder.h" +#include "webrtc/api/java/jni/jni_helpers.h" #include "webrtc/base/checks.h" #include "webrtc/base/ssladapter.h" #include "webrtc/voice_engine/include/voe_base.h" diff --git a/talk/app/webrtc/test/androidtestinitializer.h b/webrtc/api/test/androidtestinitializer.h similarity index 90% rename from talk/app/webrtc/test/androidtestinitializer.h rename to webrtc/api/test/androidtestinitializer.h index e6992825dd..2d178ac351 100644 --- a/talk/app/webrtc/test/androidtestinitializer.h +++ b/webrtc/api/test/androidtestinitializer.h @@ -25,8 +25,8 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_TEST_ANDROIDTESTINITIALIZER_H_ -#define TALK_APP_WEBRTC_TEST_ANDROIDTESTINITIALIZER_H_ +#ifndef WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_ +#define WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_ namespace webrtc { @@ -34,4 +34,4 @@ void InitializeAndroidObjects(); } // namespace webrtc -#endif // TALK_APP_WEBRTC_TEST_ANDROIDTESTINITIALIZER_H_ +#endif // WEBRTC_API_TEST_ANDROIDTESTINITIALIZER_H_ diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.cc b/webrtc/api/test/fakeaudiocapturemodule.cc similarity index 99% rename from talk/app/webrtc/test/fakeaudiocapturemodule.cc rename to webrtc/api/test/fakeaudiocapturemodule.cc index 3564d28d25..2dfa267105 100644 --- a/talk/app/webrtc/test/fakeaudiocapturemodule.cc +++ b/webrtc/api/test/fakeaudiocapturemodule.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/test/fakeaudiocapturemodule.h" +#include "webrtc/api/test/fakeaudiocapturemodule.h" #include "webrtc/base/common.h" #include "webrtc/base/refcount.h" diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.h b/webrtc/api/test/fakeaudiocapturemodule.h similarity index 98% rename from talk/app/webrtc/test/fakeaudiocapturemodule.h rename to webrtc/api/test/fakeaudiocapturemodule.h index fdac0b9ed2..315c251ab8 100644 --- a/talk/app/webrtc/test/fakeaudiocapturemodule.h +++ b/webrtc/api/test/fakeaudiocapturemodule.h @@ -34,8 +34,8 @@ // Note P postfix of a function indicates that it should only be called by the // processing thread. -#ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ -#define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ +#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_ +#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_ #include "webrtc/base/basictypes.h" #include "webrtc/base/criticalsection.h" @@ -284,4 +284,4 @@ class FakeAudioCaptureModule rtc::CriticalSection crit_callback_; }; -#endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ +#endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_ diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/webrtc/api/test/fakeaudiocapturemodule_unittest.cc similarity index 99% rename from talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc rename to webrtc/api/test/fakeaudiocapturemodule_unittest.cc index 4e3bafef72..b95d2d7669 100644 --- a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc +++ b/webrtc/api/test/fakeaudiocapturemodule_unittest.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/test/fakeaudiocapturemodule.h" +#include "webrtc/api/test/fakeaudiocapturemodule.h" #include diff --git a/talk/app/webrtc/test/fakeconstraints.h b/webrtc/api/test/fakeconstraints.h similarity index 95% rename from talk/app/webrtc/test/fakeconstraints.h rename to webrtc/api/test/fakeconstraints.h index 8673d85097..155e5ea3fd 100644 --- a/talk/app/webrtc/test/fakeconstraints.h +++ b/webrtc/api/test/fakeconstraints.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_TEST_FAKECONSTRAINTS_H_ -#define TALK_APP_WEBRTC_TEST_FAKECONSTRAINTS_H_ +#ifndef WEBRTC_API_TEST_FAKECONSTRAINTS_H_ +#define WEBRTC_API_TEST_FAKECONSTRAINTS_H_ #include #include -#include "talk/app/webrtc/mediaconstraintsinterface.h" +#include "webrtc/api/mediaconstraintsinterface.h" #include "webrtc/base/stringencode.h" namespace webrtc { @@ -130,4 +130,4 @@ class FakeConstraints : public webrtc::MediaConstraintsInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_TEST_FAKECONSTRAINTS_H_ +#endif // WEBRTC_API_TEST_FAKECONSTRAINTS_H_ diff --git a/talk/app/webrtc/test/fakedatachannelprovider.h b/webrtc/api/test/fakedatachannelprovider.h similarity index 96% rename from talk/app/webrtc/test/fakedatachannelprovider.h rename to webrtc/api/test/fakedatachannelprovider.h index ff44e585fe..32c2b52d50 100644 --- a/talk/app/webrtc/test/fakedatachannelprovider.h +++ b/webrtc/api/test/fakedatachannelprovider.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_TEST_FAKEDATACHANNELPROVIDER_H_ -#define TALK_APP_WEBRTC_TEST_FAKEDATACHANNELPROVIDER_H_ +#ifndef WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_ +#define WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_ -#include "talk/app/webrtc/datachannel.h" +#include "webrtc/api/datachannel.h" class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { public: @@ -158,4 +158,4 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface { std::set send_ssrcs_; std::set recv_ssrcs_; }; -#endif // TALK_APP_WEBRTC_TEST_FAKEDATACHANNELPROVIDER_H_ +#endif // WEBRTC_API_TEST_FAKEDATACHANNELPROVIDER_H_ diff --git a/talk/app/webrtc/test/fakedtlsidentitystore.h b/webrtc/api/test/fakedtlsidentitystore.h similarity index 96% rename from talk/app/webrtc/test/fakedtlsidentitystore.h rename to webrtc/api/test/fakedtlsidentitystore.h index 98074c742a..404e2ae0d1 100644 --- a/talk/app/webrtc/test/fakedtlsidentitystore.h +++ b/webrtc/api/test/fakedtlsidentitystore.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_TEST_FAKEDTLSIDENTITYSERVICE_H_ -#define TALK_APP_WEBRTC_TEST_FAKEDTLSIDENTITYSERVICE_H_ +#ifndef WEBRTC_API_TEST_FAKEDTLSIDENTITYSERVICE_H_ +#define WEBRTC_API_TEST_FAKEDTLSIDENTITYSERVICE_H_ #include #include -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/peerconnectioninterface.h" #include "webrtc/base/rtccertificate.h" static const struct { @@ -181,4 +181,4 @@ class FakeDtlsIdentityStore : public webrtc::DtlsIdentityStoreInterface, int key_index_ = 0; }; -#endif // TALK_APP_WEBRTC_TEST_FAKEDTLSIDENTITYSERVICE_H_ +#endif // WEBRTC_API_TEST_FAKEDTLSIDENTITYSERVICE_H_ diff --git a/talk/app/webrtc/test/fakeperiodicvideocapturer.h b/webrtc/api/test/fakeperiodicvideocapturer.h similarity index 95% rename from talk/app/webrtc/test/fakeperiodicvideocapturer.h rename to webrtc/api/test/fakeperiodicvideocapturer.h index 7e6863c9da..2ce648e7a0 100644 --- a/talk/app/webrtc/test/fakeperiodicvideocapturer.h +++ b/webrtc/api/test/fakeperiodicvideocapturer.h @@ -28,8 +28,8 @@ // FakePeriodicVideoCapturer implements a fake cricket::VideoCapturer that // creates video frames periodically after it has been started. -#ifndef TALK_APP_WEBRTC_TEST_FAKEPERIODICVIDEOCAPTURER_H_ -#define TALK_APP_WEBRTC_TEST_FAKEPERIODICVIDEOCAPTURER_H_ +#ifndef WEBRTC_API_TEST_FAKEPERIODICVIDEOCAPTURER_H_ +#define WEBRTC_API_TEST_FAKEPERIODICVIDEOCAPTURER_H_ #include "webrtc/base/thread.h" #include "webrtc/media/base/fakevideocapturer.h" @@ -86,4 +86,4 @@ class FakePeriodicVideoCapturer : public cricket::FakeVideoCapturer { } // namespace webrtc -#endif // TALK_APP_WEBRTC_TEST_FAKEPERIODICVIDEOCAPTURER_H_ +#endif // WEBRTC_API_TEST_FAKEPERIODICVIDEOCAPTURER_H_ diff --git a/talk/app/webrtc/test/fakevideotrackrenderer.h b/webrtc/api/test/fakevideotrackrenderer.h similarity index 92% rename from talk/app/webrtc/test/fakevideotrackrenderer.h rename to webrtc/api/test/fakevideotrackrenderer.h index 97e7eea5c3..4a7477b754 100644 --- a/talk/app/webrtc/test/fakevideotrackrenderer.h +++ b/webrtc/api/test/fakevideotrackrenderer.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_TEST_FAKEVIDEOTRACKRENDERER_H_ -#define TALK_APP_WEBRTC_TEST_FAKEVIDEOTRACKRENDERER_H_ +#ifndef WEBRTC_API_TEST_FAKEVIDEOTRACKRENDERER_H_ +#define WEBRTC_API_TEST_FAKEVIDEOTRACKRENDERER_H_ -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/media/base/fakevideorenderer.h" namespace webrtc { @@ -68,4 +68,4 @@ class FakeVideoTrackRenderer : public VideoRendererInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_TEST_FAKEVIDEOTRACKRENDERER_H_ +#endif // WEBRTC_API_TEST_FAKEVIDEOTRACKRENDERER_H_ diff --git a/talk/app/webrtc/test/mockpeerconnectionobservers.h b/webrtc/api/test/mockpeerconnectionobservers.h similarity index 96% rename from talk/app/webrtc/test/mockpeerconnectionobservers.h rename to webrtc/api/test/mockpeerconnectionobservers.h index f1bdbee9f5..bae85383eb 100644 --- a/talk/app/webrtc/test/mockpeerconnectionobservers.h +++ b/webrtc/api/test/mockpeerconnectionobservers.h @@ -27,12 +27,12 @@ // This file contains mock implementations of observers used in PeerConnection. -#ifndef TALK_APP_WEBRTC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ -#define TALK_APP_WEBRTC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ +#ifndef WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ +#define WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ #include -#include "talk/app/webrtc/datachannelinterface.h" +#include "webrtc/api/datachannelinterface.h" namespace webrtc { @@ -240,4 +240,4 @@ class MockStatsObserver : public webrtc::StatsObserver { } // namespace webrtc -#endif // TALK_APP_WEBRTC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ +#endif // WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.cc b/webrtc/api/test/peerconnectiontestwrapper.cc similarity index 97% rename from talk/app/webrtc/test/peerconnectiontestwrapper.cc rename to webrtc/api/test/peerconnectiontestwrapper.cc index 86b7842517..7f9ab59805 100644 --- a/talk/app/webrtc/test/peerconnectiontestwrapper.cc +++ b/webrtc/api/test/peerconnectiontestwrapper.cc @@ -27,11 +27,11 @@ #include -#include "talk/app/webrtc/test/fakedtlsidentitystore.h" -#include "talk/app/webrtc/test/fakeperiodicvideocapturer.h" -#include "talk/app/webrtc/test/mockpeerconnectionobservers.h" -#include "talk/app/webrtc/test/peerconnectiontestwrapper.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/test/fakedtlsidentitystore.h" +#include "webrtc/api/test/fakeperiodicvideocapturer.h" +#include "webrtc/api/test/mockpeerconnectionobservers.h" +#include "webrtc/api/test/peerconnectiontestwrapper.h" +#include "webrtc/api/videosourceinterface.h" #include "webrtc/base/gunit.h" #include "webrtc/p2p/client/fakeportallocator.h" diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.h b/webrtc/api/test/peerconnectiontestwrapper.h similarity index 92% rename from talk/app/webrtc/test/peerconnectiontestwrapper.h rename to webrtc/api/test/peerconnectiontestwrapper.h index 883f2f2454..f4600ea88c 100644 --- a/talk/app/webrtc/test/peerconnectiontestwrapper.h +++ b/webrtc/api/test/peerconnectiontestwrapper.h @@ -25,13 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ -#define TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ +#ifndef WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ +#define WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/test/fakeaudiocapturemodule.h" -#include "talk/app/webrtc/test/fakeconstraints.h" -#include "talk/app/webrtc/test/fakevideotrackrenderer.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/test/fakeaudiocapturemodule.h" +#include "webrtc/api/test/fakeconstraints.h" +#include "webrtc/api/test/fakevideotrackrenderer.h" #include "webrtc/base/sigslot.h" class PeerConnectionTestWrapper @@ -112,4 +112,4 @@ class PeerConnectionTestWrapper rtc::scoped_ptr renderer_; }; -#endif // TALK_APP_WEBRTC_TEST_PEERCONNECTIONTESTWRAPPER_H_ +#endif // WEBRTC_API_TEST_PEERCONNECTIONTESTWRAPPER_H_ diff --git a/talk/app/webrtc/test/testsdpstrings.h b/webrtc/api/test/testsdpstrings.h similarity index 97% rename from talk/app/webrtc/test/testsdpstrings.h rename to webrtc/api/test/testsdpstrings.h index e27c9a2f88..d806e71414 100644 --- a/talk/app/webrtc/test/testsdpstrings.h +++ b/webrtc/api/test/testsdpstrings.h @@ -27,8 +27,8 @@ // This file contain SDP strings used for testing. -#ifndef TALK_APP_WEBRTC_TEST_TESTSDPSTRINGS_H_ -#define TALK_APP_WEBRTC_TEST_TESTSDPSTRINGS_H_ +#ifndef WEBRTC_API_TEST_TESTSDPSTRINGS_H_ +#define WEBRTC_API_TEST_TESTSDPSTRINGS_H_ namespace webrtc { @@ -144,4 +144,4 @@ static const char kAudioSdpWithUnsupportedCodecs[] = } // namespace webrtc -#endif // TALK_APP_WEBRTC_TEST_TESTSDPSTRINGS_H_ +#endif // WEBRTC_API_TEST_TESTSDPSTRINGS_H_ diff --git a/talk/app/webrtc/umametrics.h b/webrtc/api/umametrics.h similarity index 97% rename from talk/app/webrtc/umametrics.h rename to webrtc/api/umametrics.h index 14fac962f4..f72ad3be23 100644 --- a/talk/app/webrtc/umametrics.h +++ b/webrtc/api/umametrics.h @@ -27,8 +27,8 @@ // This file contains enums related to IPv4/IPv6 metrics. -#ifndef TALK_APP_WEBRTC_UMAMETRICS_H_ -#define TALK_APP_WEBRTC_UMAMETRICS_H_ +#ifndef WEBRTC_API_UMAMETRICS_H_ +#define WEBRTC_API_UMAMETRICS_H_ namespace webrtc { @@ -125,4 +125,4 @@ enum IceCandidatePairType { } // namespace webrtc -#endif // TALK_APP_WEBRTC_UMAMETRICS_H_ +#endif // WEBRTC_API_UMAMETRICS_H_ diff --git a/talk/app/webrtc/videosource.cc b/webrtc/api/videosource.cc similarity index 99% rename from talk/app/webrtc/videosource.cc rename to webrtc/api/videosource.cc index 08c971738b..a94c9377e2 100644 --- a/talk/app/webrtc/videosource.cc +++ b/webrtc/api/videosource.cc @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/videosource.h" +#include "webrtc/api/videosource.h" #include #include -#include "talk/app/webrtc/mediaconstraintsinterface.h" +#include "webrtc/api/mediaconstraintsinterface.h" #include "talk/session/media/channelmanager.h" #include "webrtc/base/arraysize.h" diff --git a/talk/app/webrtc/videosource.h b/webrtc/api/videosource.h similarity index 93% rename from talk/app/webrtc/videosource.h rename to webrtc/api/videosource.h index bae5d30c05..262bc441f7 100644 --- a/talk/app/webrtc/videosource.h +++ b/webrtc/api/videosource.h @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_VIDEOSOURCE_H_ -#define TALK_APP_WEBRTC_VIDEOSOURCE_H_ +#ifndef WEBRTC_API_VIDEOSOURCE_H_ +#define WEBRTC_API_VIDEOSOURCE_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/notifier.h" -#include "talk/app/webrtc/videosourceinterface.h" -#include "talk/app/webrtc/videotrackrenderers.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/notifier.h" +#include "webrtc/api/videosourceinterface.h" +#include "webrtc/api/videotrackrenderers.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/sigslot.h" #include "webrtc/media/base/videosinkinterface.h" @@ -113,4 +113,4 @@ class VideoSource : public Notifier, } // namespace webrtc -#endif // TALK_APP_WEBRTC_VIDEOSOURCE_H_ +#endif // WEBRTC_API_VIDEOSOURCE_H_ diff --git a/talk/app/webrtc/videosource_unittest.cc b/webrtc/api/videosource_unittest.cc similarity index 99% rename from talk/app/webrtc/videosource_unittest.cc rename to webrtc/api/videosource_unittest.cc index bb91127000..26543ad336 100644 --- a/talk/app/webrtc/videosource_unittest.cc +++ b/webrtc/api/videosource_unittest.cc @@ -28,10 +28,10 @@ #include #include -#include "talk/app/webrtc/remotevideocapturer.h" -#include "talk/app/webrtc/test/fakeconstraints.h" -#include "talk/app/webrtc/videosource.h" #include "talk/session/media/channelmanager.h" +#include "webrtc/api/remotevideocapturer.h" +#include "webrtc/api/test/fakeconstraints.h" +#include "webrtc/api/videosource.h" #include "webrtc/base/gunit.h" #include "webrtc/media/base/fakemediaengine.h" #include "webrtc/media/base/fakevideocapturer.h" diff --git a/talk/app/webrtc/videosourceinterface.h b/webrtc/api/videosourceinterface.h similarity index 93% rename from talk/app/webrtc/videosourceinterface.h rename to webrtc/api/videosourceinterface.h index d74bf3bff9..5491576684 100644 --- a/talk/app/webrtc/videosourceinterface.h +++ b/webrtc/api/videosourceinterface.h @@ -25,10 +25,10 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_VIDEOSOURCEINTERFACE_H_ -#define TALK_APP_WEBRTC_VIDEOSOURCEINTERFACE_H_ +#ifndef WEBRTC_API_VIDEOSOURCEINTERFACE_H_ +#define WEBRTC_API_VIDEOSOURCEINTERFACE_H_ -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/media/base/mediachannel.h" #include "webrtc/media/base/videorenderer.h" @@ -65,4 +65,4 @@ class VideoSourceInterface : public MediaSourceInterface { } // namespace webrtc -#endif // TALK_APP_WEBRTC_VIDEOSOURCEINTERFACE_H_ +#endif // WEBRTC_API_VIDEOSOURCEINTERFACE_H_ diff --git a/talk/app/webrtc/videosourceproxy.h b/webrtc/api/videosourceproxy.h similarity index 91% rename from talk/app/webrtc/videosourceproxy.h rename to webrtc/api/videosourceproxy.h index 01abaf657d..99a3b1e02a 100644 --- a/talk/app/webrtc/videosourceproxy.h +++ b/webrtc/api/videosourceproxy.h @@ -25,11 +25,11 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_VIDEOSOURCEPROXY_H_ -#define TALK_APP_WEBRTC_VIDEOSOURCEPROXY_H_ +#ifndef WEBRTC_API_VIDEOSOURCEPROXY_H_ +#define WEBRTC_API_VIDEOSOURCEPROXY_H_ -#include "talk/app/webrtc/proxy.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/proxy.h" +#include "webrtc/api/videosourceinterface.h" namespace webrtc { @@ -51,4 +51,4 @@ END_PROXY() } // namespace webrtc -#endif // TALK_APP_WEBRTC_VIDEOSOURCEPROXY_H_ +#endif // WEBRTC_API_VIDEOSOURCEPROXY_H_ diff --git a/talk/app/webrtc/videotrack.cc b/webrtc/api/videotrack.cc similarity index 98% rename from talk/app/webrtc/videotrack.cc rename to webrtc/api/videotrack.cc index c649275fae..4c87c3940f 100644 --- a/talk/app/webrtc/videotrack.cc +++ b/webrtc/api/videotrack.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/videotrack.h" +#include "webrtc/api/videotrack.h" #include diff --git a/talk/app/webrtc/videotrack.h b/webrtc/api/videotrack.h similarity index 89% rename from talk/app/webrtc/videotrack.h rename to webrtc/api/videotrack.h index b321c42646..399e513189 100644 --- a/talk/app/webrtc/videotrack.h +++ b/webrtc/api/videotrack.h @@ -25,14 +25,14 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_VIDEOTRACK_H_ -#define TALK_APP_WEBRTC_VIDEOTRACK_H_ +#ifndef WEBRTC_API_VIDEOTRACK_H_ +#define WEBRTC_API_VIDEOTRACK_H_ #include -#include "talk/app/webrtc/mediastreamtrack.h" -#include "talk/app/webrtc/videosourceinterface.h" -#include "talk/app/webrtc/videotrackrenderers.h" +#include "webrtc/api/mediastreamtrack.h" +#include "webrtc/api/videosourceinterface.h" +#include "webrtc/api/videotrackrenderers.h" #include "webrtc/base/scoped_ref_ptr.h" namespace webrtc { @@ -62,4 +62,4 @@ class VideoTrack : public MediaStreamTrack { } // namespace webrtc -#endif // TALK_APP_WEBRTC_VIDEOTRACK_H_ +#endif // WEBRTC_API_VIDEOTRACK_H_ diff --git a/talk/app/webrtc/videotrack_unittest.cc b/webrtc/api/videotrack_unittest.cc similarity index 96% rename from talk/app/webrtc/videotrack_unittest.cc rename to webrtc/api/videotrack_unittest.cc index 88552a55a0..717cba697f 100644 --- a/talk/app/webrtc/videotrack_unittest.cc +++ b/webrtc/api/videotrack_unittest.cc @@ -27,11 +27,11 @@ #include -#include "talk/app/webrtc/remotevideocapturer.h" -#include "talk/app/webrtc/test/fakevideotrackrenderer.h" -#include "talk/app/webrtc/videosource.h" -#include "talk/app/webrtc/videotrack.h" #include "talk/session/media/channelmanager.h" +#include "webrtc/api/remotevideocapturer.h" +#include "webrtc/api/test/fakevideotrackrenderer.h" +#include "webrtc/api/videosource.h" +#include "webrtc/api/videotrack.h" #include "webrtc/base/gunit.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/media/base/fakemediaengine.h" diff --git a/talk/app/webrtc/videotrackrenderers.cc b/webrtc/api/videotrackrenderers.cc similarity index 98% rename from talk/app/webrtc/videotrackrenderers.cc rename to webrtc/api/videotrackrenderers.cc index 81f6530c1f..83615d44d2 100644 --- a/talk/app/webrtc/videotrackrenderers.cc +++ b/webrtc/api/videotrackrenderers.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/videotrackrenderers.h" +#include "webrtc/api/videotrackrenderers.h" #include "webrtc/media/webrtc/webrtcvideoframe.h" namespace webrtc { diff --git a/talk/app/webrtc/videotrackrenderers.h b/webrtc/api/videotrackrenderers.h similarity index 93% rename from talk/app/webrtc/videotrackrenderers.h rename to webrtc/api/videotrackrenderers.h index f66f8db8fd..1ce5afad47 100644 --- a/talk/app/webrtc/videotrackrenderers.h +++ b/webrtc/api/videotrackrenderers.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_VIDEOTRACKRENDERERS_H_ -#define TALK_APP_WEBRTC_VIDEOTRACKRENDERERS_H_ +#ifndef WEBRTC_API_VIDEOTRACKRENDERERS_H_ +#define WEBRTC_API_VIDEOTRACKRENDERERS_H_ #include -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/media/base/videorenderer.h" @@ -68,4 +68,4 @@ class VideoTrackRenderers : public cricket::VideoRenderer { } // namespace webrtc -#endif // TALK_APP_WEBRTC_VIDEOTRACKRENDERERS_H_ +#endif // WEBRTC_API_VIDEOTRACKRENDERERS_H_ diff --git a/talk/app/webrtc/webrtcsdp.cc b/webrtc/api/webrtcsdp.cc similarity index 99% rename from talk/app/webrtc/webrtcsdp.cc rename to webrtc/api/webrtcsdp.cc index 16c0e6939e..1f06b695b1 100644 --- a/talk/app/webrtc/webrtcsdp.cc +++ b/webrtc/api/webrtcsdp.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/webrtcsdp.h" +#include "webrtc/api/webrtcsdp.h" #include #include @@ -34,9 +34,9 @@ #include #include -#include "talk/app/webrtc/jsepicecandidate.h" -#include "talk/app/webrtc/jsepsessiondescription.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/jsepicecandidate.h" +#include "webrtc/api/jsepsessiondescription.h" #include "webrtc/base/arraysize.h" #include "webrtc/base/common.h" #include "webrtc/base/logging.h" diff --git a/talk/app/webrtc/webrtcsdp.h b/webrtc/api/webrtcsdp.h similarity index 96% rename from talk/app/webrtc/webrtcsdp.h rename to webrtc/api/webrtcsdp.h index fcbbdad3d3..a75f735417 100644 --- a/talk/app/webrtc/webrtcsdp.h +++ b/webrtc/api/webrtcsdp.h @@ -34,8 +34,8 @@ // * draft-lennox-mmusic-sdp-source-selection-02 - // Mechanisms for Media Source Selection in SDP -#ifndef TALK_APP_WEBRTC_WEBRTCSDP_H_ -#define TALK_APP_WEBRTC_WEBRTCSDP_H_ +#ifndef WEBRTC_API_WEBRTCSDP_H_ +#define WEBRTC_API_WEBRTCSDP_H_ #include @@ -78,4 +78,4 @@ bool SdpDeserializeCandidate(const std::string& message, SdpParseError* error); } // namespace webrtc -#endif // TALK_APP_WEBRTC_WEBRTCSDP_H_ +#endif // WEBRTC_API_WEBRTCSDP_H_ diff --git a/talk/app/webrtc/webrtcsdp_unittest.cc b/webrtc/api/webrtcsdp_unittest.cc similarity index 99% rename from talk/app/webrtc/webrtcsdp_unittest.cc rename to webrtc/api/webrtcsdp_unittest.cc index 3e438ffb9f..24dbd584b3 100644 --- a/talk/app/webrtc/webrtcsdp_unittest.cc +++ b/webrtc/api/webrtcsdp_unittest.cc @@ -29,12 +29,12 @@ #include #include -#include "talk/app/webrtc/jsepsessiondescription.h" -#ifdef WEBRTC_ANDROID -#include "talk/app/webrtc/test/androidtestinitializer.h" -#endif -#include "talk/app/webrtc/webrtcsdp.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/jsepsessiondescription.h" +#ifdef WEBRTC_ANDROID +#include "webrtc/api/test/androidtestinitializer.h" +#endif +#include "webrtc/api/webrtcsdp.h" #include "webrtc/base/gunit.h" #include "webrtc/base/logging.h" #include "webrtc/base/messagedigest.h" diff --git a/talk/app/webrtc/webrtcsession.cc b/webrtc/api/webrtcsession.cc similarity index 99% rename from talk/app/webrtc/webrtcsession.cc rename to webrtc/api/webrtcsession.cc index a848408ec5..15feb539ce 100644 --- a/talk/app/webrtc/webrtcsession.cc +++ b/webrtc/api/webrtcsession.cc @@ -25,7 +25,7 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/webrtcsession.h" +#include "webrtc/api/webrtcsession.h" #include @@ -34,15 +34,15 @@ #include #include -#include "talk/app/webrtc/jsepicecandidate.h" -#include "talk/app/webrtc/jsepsessiondescription.h" -#include "talk/app/webrtc/mediaconstraintsinterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/sctputils.h" -#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h" #include "talk/session/media/channel.h" #include "talk/session/media/channelmanager.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/jsepicecandidate.h" +#include "webrtc/api/jsepsessiondescription.h" +#include "webrtc/api/mediaconstraintsinterface.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/sctputils.h" +#include "webrtc/api/webrtcsessiondescriptionfactory.h" #include "webrtc/audio/audio_sink.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/checks.h" diff --git a/talk/app/webrtc/webrtcsession.h b/webrtc/api/webrtcsession.h similarity index 98% rename from talk/app/webrtc/webrtcsession.h rename to webrtc/api/webrtcsession.h index 7378736b15..0632fe24ca 100644 --- a/talk/app/webrtc/webrtcsession.h +++ b/webrtc/api/webrtcsession.h @@ -25,19 +25,19 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_ -#define TALK_APP_WEBRTC_WEBRTCSESSION_H_ +#ifndef WEBRTC_API_WEBRTCSESSION_H_ +#define WEBRTC_API_WEBRTCSESSION_H_ #include #include -#include "talk/app/webrtc/datachannel.h" -#include "talk/app/webrtc/dtmfsender.h" -#include "talk/app/webrtc/mediacontroller.h" -#include "talk/app/webrtc/mediastreamprovider.h" -#include "talk/app/webrtc/peerconnectioninterface.h" -#include "talk/app/webrtc/statstypes.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/datachannel.h" +#include "webrtc/api/dtmfsender.h" +#include "webrtc/api/mediacontroller.h" +#include "webrtc/api/mediastreamprovider.h" +#include "webrtc/api/peerconnectioninterface.h" +#include "webrtc/api/statstypes.h" #include "webrtc/base/sigslot.h" #include "webrtc/base/sslidentity.h" #include "webrtc/base/thread.h" @@ -519,4 +519,4 @@ class WebRtcSession : public AudioProviderInterface, }; } // namespace webrtc -#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ +#endif // WEBRTC_API_WEBRTCSESSION_H_ diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc similarity index 99% rename from talk/app/webrtc/webrtcsession_unittest.cc rename to webrtc/api/webrtcsession_unittest.cc index 8d8a782538..250e4153d6 100644 --- a/talk/app/webrtc/webrtcsession_unittest.cc +++ b/webrtc/api/webrtcsession_unittest.cc @@ -28,22 +28,22 @@ #include #include -#include "talk/app/webrtc/audiotrack.h" -#include "talk/app/webrtc/fakemediacontroller.h" -#include "talk/app/webrtc/fakemetricsobserver.h" -#include "talk/app/webrtc/jsepicecandidate.h" -#include "talk/app/webrtc/jsepsessiondescription.h" -#include "talk/app/webrtc/peerconnection.h" -#include "talk/app/webrtc/sctputils.h" -#include "talk/app/webrtc/streamcollection.h" -#include "talk/app/webrtc/streamcollection.h" -#include "talk/app/webrtc/test/fakeconstraints.h" -#include "talk/app/webrtc/test/fakedtlsidentitystore.h" -#include "talk/app/webrtc/videotrack.h" -#include "talk/app/webrtc/webrtcsession.h" -#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h" #include "talk/session/media/channelmanager.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/audiotrack.h" +#include "webrtc/api/fakemediacontroller.h" +#include "webrtc/api/fakemetricsobserver.h" +#include "webrtc/api/jsepicecandidate.h" +#include "webrtc/api/jsepsessiondescription.h" +#include "webrtc/api/peerconnection.h" +#include "webrtc/api/sctputils.h" +#include "webrtc/api/streamcollection.h" +#include "webrtc/api/streamcollection.h" +#include "webrtc/api/test/fakeconstraints.h" +#include "webrtc/api/test/fakedtlsidentitystore.h" +#include "webrtc/api/videotrack.h" +#include "webrtc/api/webrtcsession.h" +#include "webrtc/api/webrtcsessiondescriptionfactory.h" #include "webrtc/base/fakenetwork.h" #include "webrtc/base/firewallsocketserver.h" #include "webrtc/base/gunit.h" diff --git a/talk/app/webrtc/webrtcsessiondescriptionfactory.cc b/webrtc/api/webrtcsessiondescriptionfactory.cc similarity index 98% rename from talk/app/webrtc/webrtcsessiondescriptionfactory.cc rename to webrtc/api/webrtcsessiondescriptionfactory.cc index f08b77eb40..4421465923 100644 --- a/talk/app/webrtc/webrtcsessiondescriptionfactory.cc +++ b/webrtc/api/webrtcsessiondescriptionfactory.cc @@ -25,15 +25,15 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h" +#include "webrtc/api/webrtcsessiondescriptionfactory.h" #include -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/jsep.h" -#include "talk/app/webrtc/jsepsessiondescription.h" -#include "talk/app/webrtc/mediaconstraintsinterface.h" -#include "talk/app/webrtc/webrtcsession.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/jsep.h" +#include "webrtc/api/jsepsessiondescription.h" +#include "webrtc/api/mediaconstraintsinterface.h" +#include "webrtc/api/webrtcsession.h" #include "webrtc/base/sslidentity.h" using cricket::MediaSessionOptions; diff --git a/talk/app/webrtc/webrtcsessiondescriptionfactory.h b/webrtc/api/webrtcsessiondescriptionfactory.h similarity index 96% rename from talk/app/webrtc/webrtcsessiondescriptionfactory.h rename to webrtc/api/webrtcsessiondescriptionfactory.h index 3281bb5c88..7d2cdee42d 100644 --- a/talk/app/webrtc/webrtcsessiondescriptionfactory.h +++ b/webrtc/api/webrtcsessiondescriptionfactory.h @@ -25,12 +25,12 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#ifndef TALK_APP_WEBRTC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_ -#define TALK_APP_WEBRTC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_ +#ifndef WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_ +#define WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_ -#include "talk/app/webrtc/dtlsidentitystore.h" -#include "talk/app/webrtc/peerconnectioninterface.h" #include "talk/session/media/mediasession.h" +#include "webrtc/api/dtlsidentitystore.h" +#include "webrtc/api/peerconnectioninterface.h" #include "webrtc/base/messagehandler.h" #include "webrtc/base/rtccertificate.h" #include "webrtc/p2p/base/transportdescriptionfactory.h" @@ -190,4 +190,4 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler, }; } // namespace webrtc -#endif // TALK_APP_WEBRTC_WEBRTCSESSIONDESCRIPTIONFACTORY_H_ +#endif // WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_ diff --git a/webrtc/build/android/test_runner.py b/webrtc/build/android/test_runner.py index 3772005ba3..fd8ca2fc07 100755 --- a/webrtc/build/android/test_runner.py +++ b/webrtc/build/android/test_runner.py @@ -39,7 +39,7 @@ def main(): 'common_video_unittests': 'webrtc/common_video/common_video_unittests.isolate', 'peerconnection_unittests': - 'talk/peerconnection_unittests.isolate', + 'webrtc/api/peerconnection_unittests.isolate', 'modules_tests': 'webrtc/modules/modules_tests.isolate', 'modules_unittests': 'webrtc/modules/modules_unittests.isolate', 'rtc_unittests': 'webrtc/rtc_unittests.isolate', diff --git a/webrtc/build/apk_tests.gyp b/webrtc/build/apk_tests.gyp index 45cb7b6eee..f7e9a90220 100644 --- a/webrtc/build/apk_tests.gyp +++ b/webrtc/build/apk_tests.gyp @@ -68,8 +68,8 @@ 'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)peerconnection_unittests<(SHARED_LIB_SUFFIX)', }, 'dependencies': [ - '<(DEPTH)/talk/libjingle_tests.gyp:peerconnection_unittests', - '<(DEPTH)/talk/libjingle.gyp:libjingle_peerconnection_java', + '<(webrtc_root)/api/api_tests.gyp:peerconnection_unittests', + '<(webrtc_root)/api/api.gyp:libjingle_peerconnection_java', ], 'includes': [ '../../build/apk_test.gypi', diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi index 2d8127128f..6d79aa3db9 100644 --- a/webrtc/build/common.gypi +++ b/webrtc/build/common.gypi @@ -134,6 +134,9 @@ # Determines whether NEON code will be built. 'build_with_neon%': 0, + # Disable this to skip building source requiring GTK. + 'use_gtk%': 1, + # Enable this to use HW H.264 encoder/decoder on iOS/Mac PeerConnections. # Enabling this may break interop with Android clients that support H264. 'use_objc_h264%': 0, @@ -193,6 +196,9 @@ 'include_tests%': 1, 'restrict_webrtc_logging%': 0, }], + ['OS=="android" or OS=="linux"', { + 'java_home%': ' #include -#include "talk/app/webrtc/test/fakeconstraints.h" -#include "talk/app/webrtc/videosourceinterface.h" +#include "webrtc/api/videosourceinterface.h" +#include "webrtc/api/test/fakeconstraints.h" #include "webrtc/base/common.h" #include "webrtc/base/json.h" #include "webrtc/base/logging.h" diff --git a/webrtc/examples/peerconnection/client/conductor.h b/webrtc/examples/peerconnection/client/conductor.h index 21d838aae4..db2f77b646 100644 --- a/webrtc/examples/peerconnection/client/conductor.h +++ b/webrtc/examples/peerconnection/client/conductor.h @@ -17,8 +17,8 @@ #include #include -#include "talk/app/webrtc/mediastreaminterface.h" -#include "talk/app/webrtc/peerconnectioninterface.h" +#include "webrtc/api/mediastreaminterface.h" +#include "webrtc/api/peerconnectioninterface.h" #include "webrtc/examples/peerconnection/client/main_wnd.h" #include "webrtc/examples/peerconnection/client/peer_connection_client.h" #include "webrtc/base/scoped_ptr.h" diff --git a/webrtc/examples/peerconnection/client/main_wnd.h b/webrtc/examples/peerconnection/client/main_wnd.h index 6d39b38aa9..5cf38df8d7 100644 --- a/webrtc/examples/peerconnection/client/main_wnd.h +++ b/webrtc/examples/peerconnection/client/main_wnd.h @@ -15,7 +15,7 @@ #include #include -#include "talk/app/webrtc/mediastreaminterface.h" +#include "webrtc/api/mediastreaminterface.h" #include "webrtc/base/win32.h" #include "webrtc/examples/peerconnection/client/peer_connection_client.h" #include "webrtc/media/base/mediachannel.h" diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index 974ad9231b..be911088d4 100644 --- a/webrtc/webrtc.gyp +++ b/webrtc/webrtc.gyp @@ -6,7 +6,35 @@ # in the file PATENTS. All contributing project authors may # be found in the AUTHORS file in the root of the source tree. { + 'variables': { + 'webrtc_all_dependencies': [ + 'base/base.gyp:*', + 'sound/sound.gyp:*', + 'common.gyp:*', + 'common_audio/common_audio.gyp:*', + 'common_video/common_video.gyp:*', + 'media/media.gyp:*', + 'modules/modules.gyp:*', + 'p2p/p2p.gyp:*', + 'system_wrappers/system_wrappers.gyp:*', + 'tools/tools.gyp:*', + 'voice_engine/voice_engine.gyp:*', + '<(webrtc_vp8_dir)/vp8.gyp:*', + '<(webrtc_vp9_dir)/vp9.gyp:*', + ], + }, 'conditions': [ + ['build_with_chromium==0', { + # TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/ + # has been moved to webrtc/. It can't be processed by Chromium since the + # reference to buid/java.gypi is using an absolute path (and includes + # entries cannot contain variables). + 'variables': { + 'webrtc_all_dependencies': [ + 'api/api.gyp:*', + ], + }, + }], ['include_tests==1', { 'includes': [ 'libjingle/xmllite/xmllite_tests.gypi', @@ -54,23 +82,6 @@ 'call/webrtc_call.gypi', 'video/webrtc_video.gypi', ], - 'variables': { - 'webrtc_all_dependencies': [ - 'base/base.gyp:*', - 'sound/sound.gyp:*', - 'common.gyp:*', - 'common_audio/common_audio.gyp:*', - 'common_video/common_video.gyp:*', - 'media/media.gyp:*', - 'modules/modules.gyp:*', - 'p2p/p2p.gyp:*', - 'system_wrappers/system_wrappers.gyp:*', - 'tools/tools.gyp:*', - 'voice_engine/voice_engine.gyp:*', - '<(webrtc_vp8_dir)/vp8.gyp:*', - '<(webrtc_vp9_dir)/vp9.gyp:*', - ], - }, 'targets': [ { 'target_name': 'webrtc_all', @@ -82,6 +93,7 @@ 'conditions': [ ['include_tests==1', { 'dependencies': [ + 'api/api_tests.gyp:*', 'common_video/common_video_unittests.gyp:*', 'rtc_unittests', 'system_wrappers/system_wrappers_tests.gyp:*', @@ -91,14 +103,6 @@ 'webrtc_tests', ], }], - ['OS=="ios"', { - 'dependencies': [ - # TODO(tkchin): Move this target to webrtc_all_dependencies once it - # has more than iOS specific targets. - # TODO(tkchin): Figure out where to add this in BUILD.gn. - 'api/api.gyp:*', - ], - }], ], }, { diff --git a/webrtc/webrtc_examples.gyp b/webrtc/webrtc_examples.gyp index fd39b0304c..51cd7929d5 100755 --- a/webrtc/webrtc_examples.gyp +++ b/webrtc/webrtc_examples.gyp @@ -78,7 +78,7 @@ 'examples/peerconnection/client/peer_connection_client.h', ], 'dependencies': [ - '../talk/libjingle.gyp:libjingle_peerconnection', + 'api/api.gyp:libjingle_peerconnection', '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default', ], 'conditions': [ @@ -364,7 +364,7 @@ 'target_name': 'AppRTCDemo', 'type': 'none', 'dependencies': [ - '../talk/libjingle.gyp:libjingle_peerconnection_java', + 'api/api.gyp:libjingle_peerconnection_java', ], 'variables': { 'apk_name': 'AppRTCDemo',