diff options
50 files changed, 203 insertions, 203 deletions
@@ -191,7 +191,7 @@ deps = { Var('chromium_git') + '/chromium/third_party/ffmpeg.git' + '@' + 'e6e47f514216bbcdbfe796eb1f398c9afece93c8', 'src/third_party/libjingle/source/talk': - Var('chromium_git') + '/external/webrtc/trunk/talk.git' + '@' + '01cbe5bbcb4412882bc787c50c987de64787a37a', # commit position 11522 + Var('chromium_git') + '/external/webrtc/trunk/talk.git' + '@' + 'bb467ca7113e49d3a651f942adb54c7f95425aad', # commit position 11545 'src/third_party/usrsctp/usrsctplib': Var('chromium_git') + '/external/github.com/sctplab/usrsctp' + '@' + 'c60ec8b35c3fe6027d7a3faae89d1c8d7dd3ce98', @@ -215,7 +215,7 @@ deps = { Var('chromium_git') + '/native_client/src/third_party/scons-2.0.1.git' + '@' + '1c1550e17fc26355d08627fbdec13d8291227067', 'src/third_party/webrtc': - Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + '4def4205b4932e1c3d5f004b67a723345d1674ed', # commit position 11523 + Var('chromium_git') + '/external/webrtc/trunk/webrtc.git' + '@' + '608b2be3f94443676004d37fbc28e4e32fe56938', # commit position 11548 'src/third_party/openmax_dl': Var('chromium_git') + '/external/webrtc/deps/third_party/openmax.git' + '@' + Var('openmax_dl_revision'), diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc index 2df3604..7ae75a0 100644 --- a/content/renderer/media/media_stream_audio_processor.cc +++ b/content/renderer/media/media_stream_audio_processor.cc @@ -23,7 +23,7 @@ #include "media/base/audio_fifo.h" #include "media/base/channel_layout.h" #include "third_party/WebKit/public/platform/WebMediaConstraints.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" +#include "third_party/webrtc/api/mediaconstraintsinterface.h" #include "third_party/webrtc/modules/audio_processing/typing_detection.h" namespace content { diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h index dabc42f..478b594 100644 --- a/content/renderer/media/media_stream_audio_processor.h +++ b/content/renderer/media/media_stream_audio_processor.h @@ -18,7 +18,7 @@ #include "content/renderer/media/audio_repetition_detector.h" #include "content/renderer/media/webrtc_audio_device_impl.h" #include "media/base/audio_converter.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h" namespace blink { diff --git a/content/renderer/media/media_stream_audio_processor_options.h b/content/renderer/media/media_stream_audio_processor_options.h index 6a2048a..ea583e1 100644 --- a/content/renderer/media/media_stream_audio_processor_options.h +++ b/content/renderer/media/media_stream_audio_processor_options.h @@ -12,7 +12,7 @@ #include "content/common/content_export.h" #include "content/public/common/media_stream_request.h" #include "third_party/WebKit/public/platform/WebMediaConstraints.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h" namespace webrtc { diff --git a/content/renderer/media/media_stream_audio_processor_unittest.cc b/content/renderer/media/media_stream_audio_processor_unittest.cc index a23d03e..62e4db8 100644 --- a/content/renderer/media/media_stream_audio_processor_unittest.cc +++ b/content/renderer/media/media_stream_audio_processor_unittest.cc @@ -24,7 +24,7 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "third_party/WebKit/public/platform/WebMediaConstraints.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" using ::testing::_; using ::testing::AnyNumber; diff --git a/content/renderer/media/media_stream_audio_source.h b/content/renderer/media/media_stream_audio_source.h index e332b83..b2f44d2 100644 --- a/content/renderer/media/media_stream_audio_source.h +++ b/content/renderer/media/media_stream_audio_source.h @@ -11,7 +11,7 @@ #include "content/renderer/media/media_stream_source.h" #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" #include "content/renderer/media/webrtc_audio_capturer.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/media_stream_audio_track.cc b/content/renderer/media/media_stream_audio_track.cc index 89bc39f..278ab05 100644 --- a/content/renderer/media/media_stream_audio_track.cc +++ b/content/renderer/media/media_stream_audio_track.cc @@ -6,7 +6,7 @@ #include "base/logging.h" #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/media_stream_renderer_factory_impl.cc b/content/renderer/media/media_stream_renderer_factory_impl.cc index 40463f5..3fa439c 100644 --- a/content/renderer/media/media_stream_renderer_factory_impl.cc +++ b/content/renderer/media/media_stream_renderer_factory_impl.cc @@ -17,7 +17,7 @@ #include "third_party/WebKit/public/platform/WebMediaStream.h" #include "third_party/WebKit/public/platform/WebURL.h" #include "third_party/WebKit/public/web/WebMediaStreamRegistry.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/mock_constraint_factory.cc b/content/renderer/media/mock_constraint_factory.cc index 4e2b414..63ae5f0 100644 --- a/content/renderer/media/mock_constraint_factory.cc +++ b/content/renderer/media/mock_constraint_factory.cc @@ -9,7 +9,7 @@ #include "base/strings/utf_string_conversions.h" #include "content/renderer/media/media_stream_audio_processor_options.h" #include "content/renderer/media/mock_constraint_factory.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" +#include "third_party/webrtc/api/mediaconstraintsinterface.h" namespace content { diff --git a/content/renderer/media/mock_data_channel_impl.h b/content/renderer/media/mock_data_channel_impl.h index 9b82669..0b763dc 100644 --- a/content/renderer/media/mock_data_channel_impl.h +++ b/content/renderer/media/mock_data_channel_impl.h @@ -10,7 +10,7 @@ #include <string> #include "base/macros.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" namespace content { diff --git a/content/renderer/media/mock_media_constraint_factory.cc b/content/renderer/media/mock_media_constraint_factory.cc index 66bb1c6..2537d0f 100644 --- a/content/renderer/media/mock_media_constraint_factory.cc +++ b/content/renderer/media/mock_media_constraint_factory.cc @@ -9,7 +9,7 @@ #include "base/strings/utf_string_conversions.h" #include "content/renderer/media/media_stream_audio_processor_options.h" #include "content/renderer/media/mock_media_constraint_factory.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" +#include "third_party/webrtc/api/mediaconstraintsinterface.h" namespace content { diff --git a/content/renderer/media/mock_peer_connection_impl.h b/content/renderer/media/mock_peer_connection_impl.h index dae0b9c..a0b5139 100644 --- a/content/renderer/media/mock_peer_connection_impl.h +++ b/content/renderer/media/mock_peer_connection_impl.h @@ -12,7 +12,7 @@ #include "base/macros.h" #include "base/memory/scoped_ptr.h" #include "testing/gmock/include/gmock/gmock.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" namespace content { diff --git a/content/renderer/media/peer_connection_identity_store.h b/content/renderer/media/peer_connection_identity_store.h index 82afbdc..35d99c5 100644 --- a/content/renderer/media/peer_connection_identity_store.h +++ b/content/renderer/media/peer_connection_identity_store.h @@ -8,7 +8,7 @@ #include "base/macros.h" #include "base/single_thread_task_runner.h" #include "base/threading/thread_checker.h" -#include "third_party/libjingle/source/talk/app/webrtc/dtlsidentitystore.h" +#include "third_party/webrtc/api/dtlsidentitystore.h" #include "url/gurl.h" namespace content { diff --git a/content/renderer/media/peer_connection_tracker.h b/content/renderer/media/peer_connection_tracker.h index 2181ca8..bb01fb9 100644 --- a/content/renderer/media/peer_connection_tracker.h +++ b/content/renderer/media/peer_connection_tracker.h @@ -15,7 +15,7 @@ #include "third_party/WebKit/public/platform/WebMediaStream.h" #include "third_party/WebKit/public/platform/WebRTCPeerConnectionHandlerClient.h" #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" namespace blink { class WebFrame; diff --git a/content/renderer/media/remote_media_stream_impl.h b/content/renderer/media/remote_media_stream_impl.h index 91b9294..75bfbc84 100644 --- a/content/renderer/media/remote_media_stream_impl.h +++ b/content/renderer/media/remote_media_stream_impl.h @@ -14,7 +14,7 @@ #include "base/threading/thread_checker.h" #include "content/common/content_export.h" #include "third_party/WebKit/public/platform/WebMediaStream.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/rtc_data_channel_handler.h b/content/renderer/media/rtc_data_channel_handler.h index 1206d75..ff08ee2 100644 --- a/content/renderer/media/rtc_data_channel_handler.h +++ b/content/renderer/media/rtc_data_channel_handler.h @@ -15,7 +15,7 @@ #include "content/common/content_export.h" #include "third_party/WebKit/public/platform/WebRTCDataChannelHandler.h" #include "third_party/WebKit/public/platform/WebRTCDataChannelHandlerClient.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" namespace content { diff --git a/content/renderer/media/rtc_dtmf_sender_handler.h b/content/renderer/media/rtc_dtmf_sender_handler.h index 6967d48..b18abce 100644 --- a/content/renderer/media/rtc_dtmf_sender_handler.h +++ b/content/renderer/media/rtc_dtmf_sender_handler.h @@ -14,7 +14,7 @@ #include "content/common/content_export.h" #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandler.h" #include "third_party/WebKit/public/platform/WebRTCDTMFSenderHandlerClient.h" -#include "third_party/libjingle/source/talk/app/webrtc/dtmfsenderinterface.h" +#include "third_party/webrtc/api/dtmfsenderinterface.h" namespace content { diff --git a/content/renderer/media/rtc_media_constraints.h b/content/renderer/media/rtc_media_constraints.h index 2c6ab27..8aff145 100644 --- a/content/renderer/media/rtc_media_constraints.h +++ b/content/renderer/media/rtc_media_constraints.h @@ -7,7 +7,7 @@ #include "base/compiler_specific.h" #include "content/common/content_export.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" +#include "third_party/webrtc/api/mediaconstraintsinterface.h" namespace blink { class WebMediaConstraints; diff --git a/content/renderer/media/rtc_peer_connection_handler_unittest.cc b/content/renderer/media/rtc_peer_connection_handler_unittest.cc index 51d195c..697f484 100644 --- a/content/renderer/media/rtc_peer_connection_handler_unittest.cc +++ b/content/renderer/media/rtc_peer_connection_handler_unittest.cc @@ -48,7 +48,7 @@ #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h" #include "third_party/WebKit/public/platform/WebURL.h" #include "third_party/WebKit/public/web/WebHeap.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" static const char kDummySdp[] = "dummy sdp"; static const char kDummySdpType[] = "dummy type"; diff --git a/content/renderer/media/user_media_client_impl.h b/content/renderer/media/user_media_client_impl.h index 04cc2e0..7beb153 100644 --- a/content/renderer/media/user_media_client_impl.h +++ b/content/renderer/media/user_media_client_impl.h @@ -26,7 +26,7 @@ #include "third_party/WebKit/public/web/WebMediaDevicesRequest.h" #include "third_party/WebKit/public/web/WebUserMediaClient.h" #include "third_party/WebKit/public/web/WebUserMediaRequest.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { class PeerConnectionDependencyFactory; diff --git a/content/renderer/media/webrtc/media_stream_remote_audio_track.cc b/content/renderer/media/webrtc/media_stream_remote_audio_track.cc index 8bb5e0a..17df845 100644 --- a/content/renderer/media/webrtc/media_stream_remote_audio_track.cc +++ b/content/renderer/media/webrtc/media_stream_remote_audio_track.cc @@ -10,7 +10,7 @@ #include "base/logging.h" #include "content/public/renderer/media_stream_audio_sink.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/webrtc/media_stream_remote_video_source.h b/content/renderer/media/webrtc/media_stream_remote_video_source.h index 1f24c75..344748f 100644 --- a/content/renderer/media/webrtc/media_stream_remote_video_source.h +++ b/content/renderer/media/webrtc/media_stream_remote_video_source.h @@ -11,7 +11,7 @@ #include "content/common/content_export.h" #include "content/renderer/media/media_stream_video_source.h" #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/webrtc/media_stream_track_metrics.cc b/content/renderer/media/webrtc/media_stream_track_metrics.cc index 3353692..21f2da6 100644 --- a/content/renderer/media/webrtc/media_stream_track_metrics.cc +++ b/content/renderer/media/webrtc/media_stream_track_metrics.cc @@ -12,7 +12,7 @@ #include "base/thread_task_runner_handle.h" #include "content/common/media/media_stream_track_metrics_host_messages.h" #include "content/renderer/render_thread_impl.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" using webrtc::AudioTrackVector; using webrtc::MediaStreamInterface; diff --git a/content/renderer/media/webrtc/media_stream_track_metrics.h b/content/renderer/media/webrtc/media_stream_track_metrics.h index 4f2cd0e..2f3feca 100644 --- a/content/renderer/media/webrtc/media_stream_track_metrics.h +++ b/content/renderer/media/webrtc/media_stream_track_metrics.h @@ -10,7 +10,7 @@ #include "base/memory/scoped_vector.h" #include "base/threading/non_thread_safe.h" #include "content/common/content_export.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" namespace webrtc { class MediaStreamInterface; diff --git a/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc b/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc index 8f10d55..efb9f73 100644 --- a/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc +++ b/content/renderer/media/webrtc/media_stream_track_metrics_unittest.cc @@ -12,7 +12,7 @@ #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" using webrtc::AudioSourceInterface; using webrtc::AudioTrackInterface; diff --git a/content/renderer/media/webrtc/media_stream_video_webrtc_sink.h b/content/renderer/media/webrtc/media_stream_video_webrtc_sink.h index 57c3702..346d434 100644 --- a/content/renderer/media/webrtc/media_stream_video_webrtc_sink.h +++ b/content/renderer/media/webrtc/media_stream_video_webrtc_sink.h @@ -10,8 +10,8 @@ #include "content/public/renderer/media_stream_video_sink.h" #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" -#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" +#include "third_party/webrtc/api/videosourceinterface.h" namespace content { diff --git a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc index 4d8fe77..c414454 100644 --- a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc +++ b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.cc @@ -15,7 +15,7 @@ #include "content/renderer/media/webrtc_audio_capturer.h" #include "content/renderer/media/webrtc_local_audio_track.h" #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" #include "third_party/webrtc/base/scoped_ref_ptr.h" #include "third_party/webrtc/media/base/videocapturer.h" diff --git a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h index 943b20f..60dec668 100644 --- a/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h +++ b/content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h @@ -12,7 +12,7 @@ #include "base/compiler_specific.h" #include "base/macros.h" #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" +#include "third_party/webrtc/api/mediaconstraintsinterface.h" #include "third_party/webrtc/media/base/videorenderer.h" #include "third_party/webrtc/media/base/videosinkinterface.h" diff --git a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc index a2e3f18..966a295 100644 --- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc +++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc @@ -65,7 +65,7 @@ #include "third_party/WebKit/public/platform/WebURL.h" #include "third_party/WebKit/public/web/WebDocument.h" #include "third_party/WebKit/public/web/WebFrame.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" +#include "third_party/webrtc/api/mediaconstraintsinterface.h" #include "third_party/webrtc/base/ssladapter.h" #include "third_party/webrtc/modules/video_coding/codecs/h264/include/h264.h" diff --git a/content/renderer/media/webrtc/peer_connection_dependency_factory.h b/content/renderer/media/webrtc/peer_connection_dependency_factory.h index a5bdb34..212eefb 100644 --- a/content/renderer/media/webrtc/peer_connection_dependency_factory.h +++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.h @@ -16,8 +16,8 @@ #include "content/renderer/media/webrtc/stun_field_trial.h" #include "content/renderer/p2p/socket_dispatcher.h" #include "ipc/ipc_platform_file.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" -#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" +#include "third_party/webrtc/api/videosourceinterface.h" #include "third_party/webrtc/p2p/stunprober/stunprober.h" namespace base { diff --git a/content/renderer/media/webrtc/track_observer.h b/content/renderer/media/webrtc/track_observer.h index c0a6ad4..2a542b8 100644 --- a/content/renderer/media/webrtc/track_observer.h +++ b/content/renderer/media/webrtc/track_observer.h @@ -10,7 +10,7 @@ #include "base/memory/ref_counted.h" #include "base/single_thread_task_runner.h" #include "content/common/content_export.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc index 715309e..2679aff 100644 --- a/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc +++ b/content/renderer/media/webrtc/webrtc_audio_sink_adapter.cc @@ -5,7 +5,7 @@ #include "base/logging.h" #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h" #include "media/base/audio_bus.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc index 3c86a1f..20a3969 100644 --- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc +++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.cc @@ -11,7 +11,7 @@ #include "content/renderer/media/webrtc/webrtc_audio_sink_adapter.h" #include "content/renderer/media/webrtc_local_audio_track.h" #include "content/renderer/render_thread_impl.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h index c52066e..cfe4a98 100644 --- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h +++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h @@ -13,7 +13,7 @@ #include "base/synchronization/lock.h" #include "base/threading/thread_checker.h" #include "content/common/content_export.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h" +#include "third_party/webrtc/api/mediastreamtrack.h" #include "third_party/webrtc/media/base/audiorenderer.h" namespace cricket { diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc index e34ddb7..0230e06 100644 --- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc +++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc @@ -10,7 +10,7 @@ #include "content/renderer/media/webrtc_local_audio_track.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" using ::testing::_; using ::testing::AnyNumber; diff --git a/content/renderer/media/webrtc/webrtc_media_stream_adapter.h b/content/renderer/media/webrtc/webrtc_media_stream_adapter.h index 3b8a3e7..eba23db 100644 --- a/content/renderer/media/webrtc/webrtc_media_stream_adapter.h +++ b/content/renderer/media/webrtc/webrtc_media_stream_adapter.h @@ -11,7 +11,7 @@ #include "content/common/content_export.h" #include "content/renderer/media/media_stream.h" #include "third_party/WebKit/public/platform/WebMediaStream.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace content { diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc index 5b17248..49fc185 100644 --- a/content/renderer/media/webrtc_audio_renderer.cc +++ b/content/renderer/media/webrtc_audio_renderer.cc @@ -22,7 +22,7 @@ #include "media/audio/audio_parameters.h" #include "media/audio/sample_rates.h" #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" #include "third_party/webrtc/media/base/audiorenderer.h" #if defined(OS_WIN) diff --git a/content/renderer/media/webrtc_audio_renderer_unittest.cc b/content/renderer/media/webrtc_audio_renderer_unittest.cc index 86710d5..7797ffa 100644 --- a/content/renderer/media/webrtc_audio_renderer_unittest.cc +++ b/content/renderer/media/webrtc_audio_renderer_unittest.cc @@ -25,7 +25,7 @@ #include "third_party/WebKit/public/platform/WebMediaStream.h" #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" #include "third_party/WebKit/public/web/WebHeap.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" using testing::Return; using testing::_; diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc index 60123b0..986a536 100644 --- a/content/renderer/media/webrtc_local_audio_track_unittest.cc +++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc @@ -19,7 +19,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "third_party/WebKit/public/platform/WebMediaConstraints.h" #include "third_party/WebKit/public/web/WebHeap.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" using ::testing::_; using ::testing::AnyNumber; diff --git a/remoting/host/cast_extension_session.cc b/remoting/host/cast_extension_session.cc index f9c6bb6..b602178 100644 --- a/remoting/host/cast_extension_session.cc +++ b/remoting/host/cast_extension_session.cc @@ -19,9 +19,9 @@ #include "remoting/protocol/port_allocator_factory.h" #include "remoting/protocol/transport_context.h" #include "remoting/protocol/webrtc_video_capturer_adapter.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" -#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" -#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" +#include "third_party/webrtc/api/test/fakeconstraints.h" +#include "third_party/webrtc/api/videosourceinterface.h" namespace remoting { diff --git a/remoting/host/cast_extension_session.h b/remoting/host/cast_extension_session.h index e584abe..18e4b27 100644 --- a/remoting/host/cast_extension_session.h +++ b/remoting/host/cast_extension_session.h @@ -15,7 +15,7 @@ #include "base/values.h" #include "jingle/glue/thread_wrapper.h" #include "remoting/host/host_extension_session.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" #include "third_party/webrtc/base/scoped_ref_ptr.h" #include "third_party/webrtc/modules/desktop_capture/desktop_capturer.h" diff --git a/remoting/protocol/webrtc_connection_to_client.cc b/remoting/protocol/webrtc_connection_to_client.cc index 3173dcf..1febc42 100644 --- a/remoting/protocol/webrtc_connection_to_client.cc +++ b/remoting/protocol/webrtc_connection_to_client.cc @@ -23,10 +23,10 @@ #include "remoting/protocol/webrtc_transport.h" #include "remoting/protocol/webrtc_video_capturer_adapter.h" #include "remoting/protocol/webrtc_video_stream.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" -#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" -#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" +#include "third_party/webrtc/api/test/fakeconstraints.h" +#include "third_party/webrtc/api/videosourceinterface.h" namespace remoting { namespace protocol { diff --git a/remoting/protocol/webrtc_data_stream_adapter.h b/remoting/protocol/webrtc_data_stream_adapter.h index a05a149..9d6262d 100644 --- a/remoting/protocol/webrtc_data_stream_adapter.h +++ b/remoting/protocol/webrtc_data_stream_adapter.h @@ -12,7 +12,7 @@ #include "base/memory/weak_ptr.h" #include "remoting/protocol/errors.h" #include "remoting/protocol/message_channel_factory.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" #include "third_party/webrtc/base/refcount.h" namespace rtc { diff --git a/remoting/protocol/webrtc_transport.cc b/remoting/protocol/webrtc_transport.cc index eb13567..105330c 100644 --- a/remoting/protocol/webrtc_transport.cc +++ b/remoting/protocol/webrtc_transport.cc @@ -15,7 +15,7 @@ #include "jingle/glue/thread_wrapper.h" #include "remoting/protocol/stream_message_pipe_adapter.h" #include "remoting/protocol/transport_context.h" -#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" +#include "third_party/webrtc/api/test/fakeconstraints.h" #include "third_party/webrtc/libjingle/xmllite/xmlelement.h" #include "third_party/webrtc/modules/audio_device/include/fake_audio_device.h" diff --git a/remoting/protocol/webrtc_transport.h b/remoting/protocol/webrtc_transport.h index 7762691..bf42467 100644 --- a/remoting/protocol/webrtc_transport.h +++ b/remoting/protocol/webrtc_transport.h @@ -16,7 +16,7 @@ #include "remoting/protocol/transport.h" #include "remoting/protocol/webrtc_data_stream_adapter.h" #include "remoting/signaling/signal_strategy.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" namespace webrtc { class FakeAudioDeviceModule; diff --git a/remoting/protocol/webrtc_video_renderer_adapter.h b/remoting/protocol/webrtc_video_renderer_adapter.h index 87efcc9..e9cbb72 100644 --- a/remoting/protocol/webrtc_video_renderer_adapter.h +++ b/remoting/protocol/webrtc_video_renderer_adapter.h @@ -7,7 +7,7 @@ #include "base/memory/ref_counted.h" #include "base/memory/weak_ptr.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" namespace base { class SingleThreadTaskRunner; diff --git a/remoting/protocol/webrtc_video_stream.cc b/remoting/protocol/webrtc_video_stream.cc index 07e215e..f377613 100644 --- a/remoting/protocol/webrtc_video_stream.cc +++ b/remoting/protocol/webrtc_video_stream.cc @@ -6,10 +6,10 @@ #include "base/logging.h" #include "remoting/protocol/webrtc_video_capturer_adapter.h" -#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" -#include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" -#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" -#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" +#include "third_party/webrtc/api/mediastreaminterface.h" +#include "third_party/webrtc/api/peerconnectioninterface.h" +#include "third_party/webrtc/api/test/fakeconstraints.h" +#include "third_party/webrtc/api/videosourceinterface.h" namespace remoting { namespace protocol { diff --git a/third_party/libjingle/BUILD.gn b/third_party/libjingle/BUILD.gn index bc2c3e7..be97212 100644 --- a/third_party/libjingle/BUILD.gn +++ b/third_party/libjingle/BUILD.gn @@ -296,6 +296,78 @@ if (enable_webrtc) { # as is supported in the GYP build. It's not clear what this is used for. source_set("libjingle_webrtc_common") { sources = [ + "../webrtc/api/audiotrack.cc", + "../webrtc/api/audiotrack.h", + "../webrtc/api/datachannel.cc", + "../webrtc/api/datachannel.h", + "../webrtc/api/dtlsidentitystore.cc", + "../webrtc/api/dtlsidentitystore.h", + "../webrtc/api/dtmfsender.cc", + "../webrtc/api/dtmfsender.h", + "../webrtc/api/jsep.h", + "../webrtc/api/jsepicecandidate.cc", + "../webrtc/api/jsepicecandidate.h", + "../webrtc/api/jsepsessiondescription.cc", + "../webrtc/api/jsepsessiondescription.h", + "../webrtc/api/localaudiosource.cc", + "../webrtc/api/localaudiosource.h", + "../webrtc/api/mediaconstraintsinterface.cc", + "../webrtc/api/mediaconstraintsinterface.h", + "../webrtc/api/mediacontroller.cc", + "../webrtc/api/mediacontroller.h", + "../webrtc/api/mediastream.cc", + "../webrtc/api/mediastream.h", + "../webrtc/api/mediastreamhandler.cc", + "../webrtc/api/mediastreamhandler.h", + "../webrtc/api/mediastreaminterface.h", + "../webrtc/api/mediastreamobserver.cc", + "../webrtc/api/mediastreamobserver.h", + "../webrtc/api/mediastreamprovider.h", + "../webrtc/api/mediastreamproxy.h", + "../webrtc/api/mediastreamtrack.h", + "../webrtc/api/mediastreamtrackproxy.h", + "../webrtc/api/notifier.h", + "../webrtc/api/peerconnection.cc", + "../webrtc/api/peerconnection.h", + "../webrtc/api/peerconnectionfactory.cc", + "../webrtc/api/peerconnectionfactory.h", + "../webrtc/api/peerconnectioninterface.h", + "../webrtc/api/portallocatorfactory.cc", + "../webrtc/api/portallocatorfactory.h", + "../webrtc/api/remoteaudiosource.cc", + "../webrtc/api/remoteaudiosource.h", + "../webrtc/api/remoteaudiotrack.cc", + "../webrtc/api/remoteaudiotrack.h", + "../webrtc/api/remotevideocapturer.cc", + "../webrtc/api/remotevideocapturer.h", + "../webrtc/api/rtpreceiver.cc", + "../webrtc/api/rtpreceiver.h", + "../webrtc/api/rtpreceiverinterface.h", + "../webrtc/api/rtpsender.cc", + "../webrtc/api/rtpsender.h", + "../webrtc/api/rtpsenderinterface.h", + "../webrtc/api/sctputils.cc", + "../webrtc/api/sctputils.h", + "../webrtc/api/statscollector.cc", + "../webrtc/api/statscollector.h", + "../webrtc/api/statstypes.cc", + "../webrtc/api/statstypes.h", + "../webrtc/api/streamcollection.h", + "../webrtc/api/umametrics.h", + "../webrtc/api/videosource.cc", + "../webrtc/api/videosource.h", + "../webrtc/api/videosourceinterface.h", + "../webrtc/api/videosourceproxy.h", + "../webrtc/api/videotrack.cc", + "../webrtc/api/videotrack.h", + "../webrtc/api/videotrackrenderers.cc", + "../webrtc/api/videotrackrenderers.h", + "../webrtc/api/webrtcsdp.cc", + "../webrtc/api/webrtcsdp.h", + "../webrtc/api/webrtcsession.cc", + "../webrtc/api/webrtcsession.h", + "../webrtc/api/webrtcsessiondescriptionfactory.cc", + "../webrtc/api/webrtcsessiondescriptionfactory.h", "../webrtc/media/base/audiorenderer.h", "../webrtc/media/base/capturemanager.cc", "../webrtc/media/base/capturemanager.h", @@ -338,78 +410,6 @@ if (enable_webrtc) { "../webrtc/media/webrtc/webrtcvideoframefactory.cc", "../webrtc/media/webrtc/webrtcvideoframefactory.h", "../webrtc/media/webrtc/webrtcvoe.h", - "source/talk/app/webrtc/audiotrack.cc", - "source/talk/app/webrtc/audiotrack.h", - "source/talk/app/webrtc/datachannel.cc", - "source/talk/app/webrtc/datachannel.h", - "source/talk/app/webrtc/dtlsidentitystore.cc", - "source/talk/app/webrtc/dtlsidentitystore.h", - "source/talk/app/webrtc/dtmfsender.cc", - "source/talk/app/webrtc/dtmfsender.h", - "source/talk/app/webrtc/jsep.h", - "source/talk/app/webrtc/jsepicecandidate.cc", - "source/talk/app/webrtc/jsepicecandidate.h", - "source/talk/app/webrtc/jsepsessiondescription.cc", - "source/talk/app/webrtc/jsepsessiondescription.h", - "source/talk/app/webrtc/localaudiosource.cc", - "source/talk/app/webrtc/localaudiosource.h", - "source/talk/app/webrtc/mediaconstraintsinterface.cc", - "source/talk/app/webrtc/mediaconstraintsinterface.h", - "source/talk/app/webrtc/mediacontroller.cc", - "source/talk/app/webrtc/mediacontroller.h", - "source/talk/app/webrtc/mediastream.cc", - "source/talk/app/webrtc/mediastream.h", - "source/talk/app/webrtc/mediastreamhandler.cc", - "source/talk/app/webrtc/mediastreamhandler.h", - "source/talk/app/webrtc/mediastreaminterface.h", - "source/talk/app/webrtc/mediastreamobserver.cc", - "source/talk/app/webrtc/mediastreamobserver.h", - "source/talk/app/webrtc/mediastreamprovider.h", - "source/talk/app/webrtc/mediastreamproxy.h", - "source/talk/app/webrtc/mediastreamtrack.h", - "source/talk/app/webrtc/mediastreamtrackproxy.h", - "source/talk/app/webrtc/notifier.h", - "source/talk/app/webrtc/peerconnection.cc", - "source/talk/app/webrtc/peerconnection.h", - "source/talk/app/webrtc/peerconnectionfactory.cc", - "source/talk/app/webrtc/peerconnectionfactory.h", - "source/talk/app/webrtc/peerconnectioninterface.h", - "source/talk/app/webrtc/portallocatorfactory.cc", - "source/talk/app/webrtc/portallocatorfactory.h", - "source/talk/app/webrtc/remoteaudiosource.cc", - "source/talk/app/webrtc/remoteaudiosource.h", - "source/talk/app/webrtc/remoteaudiotrack.cc", - "source/talk/app/webrtc/remoteaudiotrack.h", - "source/talk/app/webrtc/remotevideocapturer.cc", - "source/talk/app/webrtc/remotevideocapturer.h", - "source/talk/app/webrtc/rtpreceiver.cc", - "source/talk/app/webrtc/rtpreceiver.h", - "source/talk/app/webrtc/rtpreceiverinterface.h", - "source/talk/app/webrtc/rtpsender.cc", - "source/talk/app/webrtc/rtpsender.h", - "source/talk/app/webrtc/rtpsenderinterface.h", - "source/talk/app/webrtc/sctputils.cc", - "source/talk/app/webrtc/sctputils.h", - "source/talk/app/webrtc/statscollector.cc", - "source/talk/app/webrtc/statscollector.h", - "source/talk/app/webrtc/statstypes.cc", - "source/talk/app/webrtc/statstypes.h", - "source/talk/app/webrtc/streamcollection.h", - "source/talk/app/webrtc/umametrics.h", - "source/talk/app/webrtc/videosource.cc", - "source/talk/app/webrtc/videosource.h", - "source/talk/app/webrtc/videosourceinterface.h", - "source/talk/app/webrtc/videosourceproxy.h", - "source/talk/app/webrtc/videotrack.cc", - "source/talk/app/webrtc/videotrack.h", - "source/talk/app/webrtc/videotrackrenderers.cc", - "source/talk/app/webrtc/videotrackrenderers.h", - "source/talk/app/webrtc/webrtcsdp.cc", - "source/talk/app/webrtc/webrtcsdp.h", - "source/talk/app/webrtc/webrtcsession.cc", - "source/talk/app/webrtc/webrtcsession.h", - "source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc", - "source/talk/app/webrtc/webrtcsessiondescriptionfactory.h", "source/talk/session/media/audiomonitor.cc", "source/talk/session/media/audiomonitor.h", "source/talk/session/media/bundlefilter.cc", diff --git a/third_party/libjingle/README.chromium b/third_party/libjingle/README.chromium index af65653..63591ad 100644 --- a/third_party/libjingle/README.chromium +++ b/third_party/libjingle/README.chromium @@ -1,7 +1,7 @@ Name: libjingle URL: http://www.webrtc.org Version: unknown -Revision: 11522 +Revision: 11545 License: BSD License File: source/talk/COPYING Security Critical: yes diff --git a/third_party/libjingle/libjingle.gyp b/third_party/libjingle/libjingle.gyp index a079f01..c2d7352 100644 --- a/third_party/libjingle/libjingle.gyp +++ b/third_party/libjingle/libjingle.gyp @@ -256,6 +256,78 @@ 'target_name': 'libjingle_webrtc_common', 'type': 'static_library', 'sources': [ + '<(DEPTH)/third_party/webrtc/api/audiotrack.cc', + '<(DEPTH)/third_party/webrtc/api/audiotrack.h', + '<(DEPTH)/third_party/webrtc/api/datachannel.cc', + '<(DEPTH)/third_party/webrtc/api/datachannel.h', + '<(DEPTH)/third_party/webrtc/api/dtlsidentitystore.cc', + '<(DEPTH)/third_party/webrtc/api/dtlsidentitystore.h', + '<(DEPTH)/third_party/webrtc/api/dtmfsender.cc', + '<(DEPTH)/third_party/webrtc/api/dtmfsender.h', + '<(DEPTH)/third_party/webrtc/api/jsep.h', + '<(DEPTH)/third_party/webrtc/api/jsepicecandidate.cc', + '<(DEPTH)/third_party/webrtc/api/jsepicecandidate.h', + '<(DEPTH)/third_party/webrtc/api/jsepsessiondescription.cc', + '<(DEPTH)/third_party/webrtc/api/jsepsessiondescription.h', + '<(DEPTH)/third_party/webrtc/api/localaudiosource.cc', + '<(DEPTH)/third_party/webrtc/api/localaudiosource.h', + '<(DEPTH)/third_party/webrtc/api/mediaconstraintsinterface.cc', + '<(DEPTH)/third_party/webrtc/api/mediaconstraintsinterface.h', + '<(DEPTH)/third_party/webrtc/api/mediacontroller.cc', + '<(DEPTH)/third_party/webrtc/api/mediacontroller.h', + '<(DEPTH)/third_party/webrtc/api/mediastream.cc', + '<(DEPTH)/third_party/webrtc/api/mediastream.h', + '<(DEPTH)/third_party/webrtc/api/mediastreamhandler.cc', + '<(DEPTH)/third_party/webrtc/api/mediastreamhandler.h', + '<(DEPTH)/third_party/webrtc/api/mediastreaminterface.h', + '<(DEPTH)/third_party/webrtc/api/mediastreamobserver.cc', + '<(DEPTH)/third_party/webrtc/api/mediastreamobserver.h', + '<(DEPTH)/third_party/webrtc/api/mediastreamprovider.h', + '<(DEPTH)/third_party/webrtc/api/mediastreamproxy.h', + '<(DEPTH)/third_party/webrtc/api/mediastreamtrack.h', + '<(DEPTH)/third_party/webrtc/api/mediastreamtrackproxy.h', + '<(DEPTH)/third_party/webrtc/api/notifier.h', + '<(DEPTH)/third_party/webrtc/api/peerconnection.cc', + '<(DEPTH)/third_party/webrtc/api/peerconnection.h', + '<(DEPTH)/third_party/webrtc/api/peerconnectionfactory.cc', + '<(DEPTH)/third_party/webrtc/api/peerconnectionfactory.h', + '<(DEPTH)/third_party/webrtc/api/peerconnectioninterface.h', + '<(DEPTH)/third_party/webrtc/api/portallocatorfactory.cc', + '<(DEPTH)/third_party/webrtc/api/portallocatorfactory.h', + '<(DEPTH)/third_party/webrtc/api/remoteaudiosource.cc', + '<(DEPTH)/third_party/webrtc/api/remoteaudiosource.h', + '<(DEPTH)/third_party/webrtc/api/remoteaudiotrack.cc', + '<(DEPTH)/third_party/webrtc/api/remoteaudiotrack.h', + '<(DEPTH)/third_party/webrtc/api/remotevideocapturer.cc', + '<(DEPTH)/third_party/webrtc/api/remotevideocapturer.h', + '<(DEPTH)/third_party/webrtc/api/rtpreceiver.cc', + '<(DEPTH)/third_party/webrtc/api/rtpreceiver.h', + '<(DEPTH)/third_party/webrtc/api/rtpreceiverinterface.h', + '<(DEPTH)/third_party/webrtc/api/rtpsender.cc', + '<(DEPTH)/third_party/webrtc/api/rtpsender.h', + '<(DEPTH)/third_party/webrtc/api/rtpsenderinterface.h', + '<(DEPTH)/third_party/webrtc/api/sctputils.cc', + '<(DEPTH)/third_party/webrtc/api/sctputils.h', + '<(DEPTH)/third_party/webrtc/api/statscollector.cc', + '<(DEPTH)/third_party/webrtc/api/statscollector.h', + '<(DEPTH)/third_party/webrtc/api/statstypes.cc', + '<(DEPTH)/third_party/webrtc/api/statstypes.h', + '<(DEPTH)/third_party/webrtc/api/streamcollection.h', + '<(DEPTH)/third_party/webrtc/api/umametrics.h', + '<(DEPTH)/third_party/webrtc/api/videosource.cc', + '<(DEPTH)/third_party/webrtc/api/videosource.h', + '<(DEPTH)/third_party/webrtc/api/videosourceinterface.h', + '<(DEPTH)/third_party/webrtc/api/videosourceproxy.h', + '<(DEPTH)/third_party/webrtc/api/videotrack.cc', + '<(DEPTH)/third_party/webrtc/api/videotrack.h', + '<(DEPTH)/third_party/webrtc/api/videotrackrenderers.cc', + '<(DEPTH)/third_party/webrtc/api/videotrackrenderers.h', + '<(DEPTH)/third_party/webrtc/api/webrtcsdp.cc', + '<(DEPTH)/third_party/webrtc/api/webrtcsdp.h', + '<(DEPTH)/third_party/webrtc/api/webrtcsession.cc', + '<(DEPTH)/third_party/webrtc/api/webrtcsession.h', + '<(DEPTH)/third_party/webrtc/api/webrtcsessiondescriptionfactory.cc', + '<(DEPTH)/third_party/webrtc/api/webrtcsessiondescriptionfactory.h', '<(DEPTH)/third_party/webrtc/media/base/audiorenderer.h', '<(DEPTH)/third_party/webrtc/media/base/capturemanager.cc', '<(DEPTH)/third_party/webrtc/media/base/capturemanager.h', @@ -298,78 +370,6 @@ '<(DEPTH)/third_party/webrtc/media/webrtc/webrtcvideoframefactory.cc', '<(DEPTH)/third_party/webrtc/media/webrtc/webrtcvideoframefactory.h', '<(DEPTH)/third_party/webrtc/media/webrtc/webrtcvoe.h', - '<(libjingle_source)/talk/app/webrtc/audiotrack.cc', - '<(libjingle_source)/talk/app/webrtc/audiotrack.h', - '<(libjingle_source)/talk/app/webrtc/datachannel.cc', - '<(libjingle_source)/talk/app/webrtc/datachannel.h', - '<(libjingle_source)/talk/app/webrtc/dtlsidentitystore.cc', - '<(libjingle_source)/talk/app/webrtc/dtlsidentitystore.h', - '<(libjingle_source)/talk/app/webrtc/dtmfsender.cc', - '<(libjingle_source)/talk/app/webrtc/dtmfsender.h', - '<(libjingle_source)/talk/app/webrtc/jsep.h', - '<(libjingle_source)/talk/app/webrtc/jsepicecandidate.cc', - '<(libjingle_source)/talk/app/webrtc/jsepicecandidate.h', - '<(libjingle_source)/talk/app/webrtc/jsepsessiondescription.cc', - '<(libjingle_source)/talk/app/webrtc/jsepsessiondescription.h', - '<(libjingle_source)/talk/app/webrtc/localaudiosource.cc', - '<(libjingle_source)/talk/app/webrtc/localaudiosource.h', - '<(libjingle_source)/talk/app/webrtc/mediaconstraintsinterface.cc', - '<(libjingle_source)/talk/app/webrtc/mediaconstraintsinterface.h', - '<(libjingle_source)/talk/app/webrtc/mediacontroller.cc', - '<(libjingle_source)/talk/app/webrtc/mediacontroller.h', - '<(libjingle_source)/talk/app/webrtc/mediastream.cc', - '<(libjingle_source)/talk/app/webrtc/mediastream.h', - '<(libjingle_source)/talk/app/webrtc/mediastreamhandler.cc', - '<(libjingle_source)/talk/app/webrtc/mediastreamhandler.h', - '<(libjingle_source)/talk/app/webrtc/mediastreaminterface.h', - '<(libjingle_source)/talk/app/webrtc/mediastreamobserver.cc', - '<(libjingle_source)/talk/app/webrtc/mediastreamobserver.h', - '<(libjingle_source)/talk/app/webrtc/mediastreamprovider.h', - '<(libjingle_source)/talk/app/webrtc/mediastreamproxy.h', - '<(libjingle_source)/talk/app/webrtc/mediastreamtrack.h', - '<(libjingle_source)/talk/app/webrtc/mediastreamtrackproxy.h', - '<(libjingle_source)/talk/app/webrtc/notifier.h', - '<(libjingle_source)/talk/app/webrtc/peerconnection.cc', - '<(libjingle_source)/talk/app/webrtc/peerconnection.h', - '<(libjingle_source)/talk/app/webrtc/peerconnectionfactory.cc', - '<(libjingle_source)/talk/app/webrtc/peerconnectionfactory.h', - '<(libjingle_source)/talk/app/webrtc/peerconnectioninterface.h', - '<(libjingle_source)/talk/app/webrtc/portallocatorfactory.cc', - '<(libjingle_source)/talk/app/webrtc/portallocatorfactory.h', - '<(libjingle_source)/talk/app/webrtc/remoteaudiosource.cc', - '<(libjingle_source)/talk/app/webrtc/remoteaudiosource.h', - '<(libjingle_source)/talk/app/webrtc/remoteaudiotrack.cc', - '<(libjingle_source)/talk/app/webrtc/remoteaudiotrack.h', - '<(libjingle_source)/talk/app/webrtc/remotevideocapturer.cc', - '<(libjingle_source)/talk/app/webrtc/remotevideocapturer.h', - '<(libjingle_source)/talk/app/webrtc/rtpreceiver.cc', - '<(libjingle_source)/talk/app/webrtc/rtpreceiver.h', - '<(libjingle_source)/talk/app/webrtc/rtpreceiverinterface.h', - '<(libjingle_source)/talk/app/webrtc/rtpsender.cc', - '<(libjingle_source)/talk/app/webrtc/rtpsender.h', - '<(libjingle_source)/talk/app/webrtc/rtpsenderinterface.h', - '<(libjingle_source)/talk/app/webrtc/sctputils.cc', - '<(libjingle_source)/talk/app/webrtc/sctputils.h', - '<(libjingle_source)/talk/app/webrtc/statscollector.cc', - '<(libjingle_source)/talk/app/webrtc/statscollector.h', - '<(libjingle_source)/talk/app/webrtc/statstypes.cc', - '<(libjingle_source)/talk/app/webrtc/statstypes.h', - '<(libjingle_source)/talk/app/webrtc/streamcollection.h', - '<(libjingle_source)/talk/app/webrtc/umametrics.h', - '<(libjingle_source)/talk/app/webrtc/videosource.cc', - '<(libjingle_source)/talk/app/webrtc/videosource.h', - '<(libjingle_source)/talk/app/webrtc/videosourceinterface.h', - '<(libjingle_source)/talk/app/webrtc/videosourceproxy.h', - '<(libjingle_source)/talk/app/webrtc/videotrack.cc', - '<(libjingle_source)/talk/app/webrtc/videotrack.h', - '<(libjingle_source)/talk/app/webrtc/videotrackrenderers.cc', - '<(libjingle_source)/talk/app/webrtc/videotrackrenderers.h', - '<(libjingle_source)/talk/app/webrtc/webrtcsdp.cc', - '<(libjingle_source)/talk/app/webrtc/webrtcsdp.h', - '<(libjingle_source)/talk/app/webrtc/webrtcsession.cc', - '<(libjingle_source)/talk/app/webrtc/webrtcsession.h', - '<(libjingle_source)/talk/app/webrtc/webrtcsessiondescriptionfactory.cc', - '<(libjingle_source)/talk/app/webrtc/webrtcsessiondescriptionfactory.h', '<(libjingle_source)/talk/session/media/audiomonitor.cc', '<(libjingle_source)/talk/session/media/audiomonitor.h', '<(libjingle_source)/talk/session/media/bundlefilter.cc', |