| 2018-04-20 Youenn Fablet <youenn@apple.com> |
| |
| Mandate H264 hardware encoder for Mac in libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=184835 |
| |
| Reviewed by Eric Carlson. |
| |
| Tested manually through console traces that hardware VCP encoder code path is actually used instead of software VCP encoder code path. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm: |
| (-[RTCVideoEncoderH264 resetCompressionSessionWithPixelFormat:]): |
| * WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch: Added to cover this change and changes made in bug 184668 and 183961. |
| |
| 2018-04-19 David Kilzer <ddkilzer@apple.com> |
| |
| Enable Objective-C weak references |
| <https://webkit.org/b/184789> |
| <rdar://problem/39571716> |
| |
| Reviewed by Dan Bernstein. |
| |
| * Configurations/Base.xcconfig: |
| (CLANG_ENABLE_OBJC_WEAK): Enable. |
| |
| 2018-04-16 Youenn Fablet <youenn@apple.com> |
| |
| Set H264 VT encoder usage to 1 |
| https://bugs.webkit.org/show_bug.cgi?id=184668 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm: |
| (-[RTCVideoEncoderH264 configureCompressionSession]): |
| |
| 2018-04-10 Youenn Fablet <youenn@apple.com> |
| |
| webrtc/datachannel/basic-tcp.html will crash with an invalid crash |
| https://bugs.webkit.org/show_bug.cgi?id=178285 |
| <rdar://problem/34985374> |
| |
| Reviewed by Eric Carlson. |
| |
| Disable SIGPIPE for WebRTC sockets on Mac as well. |
| |
| * Source/webrtc/rtc_base/physicalsocketserver.cc: |
| * WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch: Added. |
| |
| 2018-04-09 Youenn Fablet <youenn@apple.com> |
| |
| Use special software encoder mode in case there is no VCP not hardware encoder |
| https://bugs.webkit.org/show_bug.cgi?id=183961 |
| |
| Reviewed by Eric Carlson. |
| |
| In case a compression session is not using a hardware encoder and VCP is not active |
| use a specific mode if the resolution is standard. |
| |
| * Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.cpp: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm: |
| |
| 2018-04-05 Alejandro G. Castro <alex@igalia.com> |
| |
| [GTK] Add CMake package search for vpx and libevent libraries |
| https://bugs.webkit.org/show_bug.cgi?id=184257 |
| |
| Reviewed by Michael Catanzaro. |
| |
| Add new cmake search files for libevent, vpx and alsa-lib, this |
| makes a cleaner detection of the libraries. |
| |
| * CMakeLists.txt: Use the new cmake find files to detect the |
| package and add a better error message when the library is not |
| there. |
| * Source/cmake/FindAlsaLib.cmake: Added. |
| * Source/cmake/FindLibEvent.cmake: Added. |
| * Source/cmake/FindVpx.cmake: Added. |
| |
| 2018-04-03 Youenn Fablet <youenn@apple.com> |
| |
| RealtimeOutgoingVideoSourceMac should pass a ObjCFrameBuffer buffer |
| https://bugs.webkit.org/show_bug.cgi?id=184281 |
| rdar://problem/39153262 |
| |
| Reviewed by Jer Noble. |
| |
| Introduce a routine to create the wrapper around native pixel buffers as expected by the new libwebrtc H264 encoder. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Source/webrtc/sdk/WebKit/WebKitUtilities.h: |
| * Source/webrtc/sdk/WebKit/WebKitUtilities.mm: |
| (webrtc::pixelBufferToFrame): |
| |
| 2018-04-02 Alejandro G. Castro <alex@igalia.com> |
| |
| Unreviewed fixing GTK port X86 32bits compilation after r230152. |
| |
| * CMakeLists.txt: |
| |
| 2018-04-02 Alejandro G. Castro <alex@igalia.com> |
| |
| Unreviewed fixing GTK port ARM compilation after r230152. |
| |
| * CMakeLists.txt: Properly avoid SSE implementations for ARM. |
| |
| 2018-04-02 Alejandro G. Castro <alex@igalia.com> |
| |
| [GTK] Make libwebrtc backend buildable for GTK port |
| https://bugs.webkit.org/show_bug.cgi?id=178860 |
| |
| Reviewed by Youenn Fablet. |
| |
| Modified the cmake file and added some assembly code to the |
| boringssl compilation required for the linux compilation generated |
| by libwebrtc. |
| |
| * CMakeLists.txt: This cmake file was unused so we have modified |
| it completely to make it work for our port. It was originally |
| generated from the libwebrtc json file but not anymore. We could |
| change its structure at some point but current one seems a good |
| option for the moment. |
| * Source/webrtc/base/task_queue_libevent.cc: We use system |
| libevent for the moment so we needed to adapt the includes in this file. |
| * Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc: |
| Readded lines removed by mistake in a previous commit. |
| |
| 2018-03-26 Youenn Fablet <youennf@gmail.com> |
| |
| Make VCP encoder usage conditional on using internal SDK |
| https://bugs.webkit.org/show_bug.cgi?id=184009 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h: |
| |
| 2018-03-23 Youenn Fablet <youenn@apple.com> |
| |
| Add support for VCP encoder on MacOS and iOS |
| Build fix. |
| |
| Unreviewed. |
| |
| * Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.cpp: |
| |
| 2018-03-23 Youenn Fablet <youenn@apple.com> |
| |
| Add support for VCP encoder on MacOS and iOS |
| https://bugs.webkit.org/show_bug.cgi?id=183924 |
| |
| Reviewed by Eric Carlson. |
| |
| Soft-Link VideoProcessing functions and use them in H264 encoder. |
| This is conditional on recent MacOS and iOS platforms. |
| |
| * Source/webrtc/sdk/WebKit/EncoderUtilities.h: Added. |
| * Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.cpp: Added. |
| * Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h: Added. |
| * Source/webrtc/sdk/WebKit/WebKitUtilities.mm: |
| (webrtc::createVideoToolboxEncoderFactory): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm: |
| (-[RTCVideoEncoderH264 encode:codecSpecificInfo:frameTypes:]): |
| (-[RTCVideoEncoderH264 resetCompressionSessionWithPixelFormat:]): |
| (-[RTCVideoEncoderH264 destroyCompressionSession]): |
| * WebKit/0001-Using-VCP.patch: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2018-03-23 David Kilzer <ddkilzer@apple.com> |
| |
| Stop using dispatch_set_target_queue() |
| <https://webkit.org/b/183908> |
| <rdar://problem/33553533> |
| |
| Reviewed by Daniel Bates. |
| |
| * Source/webrtc/rtc_base/task_queue_gcd.cc: Remove use of |
| dispatch_set_target_queue() by changing dispatch_queue_create() |
| to dispatch_queue_create_with_target(). |
| * WebKit/0009-Remove-dispatch_set_target_queue.patch: Add patch. |
| Filed this to track upstreaming the change: |
| <https://bugs.chromium.org/p/webrtc/issues/detail?id=9055> |
| * WebKit/patch-libwebrtc: Delete empty patch file. |
| |
| 2018-03-23 Youenn Fablet <youenn@apple.com> |
| |
| Use libwebrtc ObjectiveC H264 encoder and decoder |
| https://bugs.webkit.org/show_bug.cgi?id=183912 |
| |
| Reviewed by Eric Carlson. |
| |
| Add utilities inside libwebrtc to be used by WebKit: |
| - Create ObjectiveC encoder/decoder factories |
| - Notify of application status to invalidate encoders/decoders when in background |
| Implement RTCUIApplicationStatusObserver as a simple boolean that is set by WebCore. |
| This allows limiting the changes made to libwebrtc codec implementations. |
| |
| Minor modifications done to libwebrtc to fix compilation. |
| Add Block_copy/Block_release to codec callbacks. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Source/webrtc/sdk/WebKit/WebKitUtilities.h: Added. |
| * Source/webrtc/sdk/WebKit/WebKitUtilities.mm: Added. |
| (+[RTCUIApplicationStatusObserver sharedInstance]): |
| (+[RTCUIApplicationStatusObserver prepareForUse]): |
| (-[RTCUIApplicationStatusObserver setActive]): |
| (-[RTCUIApplicationStatusObserver setInactive]): |
| (-[RTCUIApplicationStatusObserver isApplicationActive]): |
| (webrtc::setApplicationStatus): |
| (webrtc::createVideoToolboxEncoderFactory): |
| (webrtc::createVideoToolboxDecoderFactory): |
| (webrtc::setH264HardwareEncoderAllowed): |
| (webrtc::isH264HardwareEncoderAllowed): |
| (webrtc::pixelBufferFromFrame): |
| * Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm: |
| (-[RTCCVPixelBuffer dealloc]): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm: |
| (-[RTCVideoDecoderH264 dealloc]): |
| (-[RTCVideoDecoderH264 setCallback:]): |
| (-[RTCVideoDecoderH264 releaseDecoder]): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm: |
| (-[RTCVideoEncoderH264 dealloc]): |
| (-[RTCVideoEncoderH264 setCallback:]): |
| (-[RTCVideoEncoderH264 releaseEncoder]): |
| (-[RTCVideoEncoderH264 resetCompressionSessionWithPixelFormat:]): |
| * WebKit/0001-Adapting-libwebrtc-H264-codec.patch: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2018-03-22 Commit Queue <commit-queue@webkit.org> |
| |
| Unreviewed, rolling out r229876. |
| https://bugs.webkit.org/show_bug.cgi?id=183929 |
| |
| Some webrtc tests are timing out on iOS simulator (Requested |
| by youenn on #webkit). |
| |
| Reverted changeset: |
| |
| "Use libwebrtc ObjectiveC H264 encoder and decoder" |
| https://bugs.webkit.org/show_bug.cgi?id=183912 |
| https://trac.webkit.org/changeset/229876 |
| |
| 2018-03-22 Youenn Fablet <youenn@apple.com> |
| |
| Use libwebrtc ObjectiveC H264 encoder and decoder |
| https://bugs.webkit.org/show_bug.cgi?id=183912 |
| |
| Reviewed by Eric Carlson. |
| |
| Add utilities inside libwebrtc to be used by WebKit: |
| - Create ObjectiveC encoder/decoder factories |
| - Notify of application status to invalidate encoders/decoders when in background |
| Implement RTCUIApplicationStatusObserver as a simple boolean that is set by WebCore. |
| This allows limiting the changes made to libwebrtc codec implementations. |
| |
| Minor modifications done to libwebrtc to fix compilation. |
| Add Block_copy/Block_release to codec callbacks. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Source/webrtc/sdk/WebKit/WebKitUtilities.h: Added. |
| * Source/webrtc/sdk/WebKit/WebKitUtilities.mm: Added. |
| (+[RTCUIApplicationStatusObserver sharedInstance]): |
| (+[RTCUIApplicationStatusObserver prepareForUse]): |
| (-[RTCUIApplicationStatusObserver setActive]): |
| (-[RTCUIApplicationStatusObserver setInactive]): |
| (-[RTCUIApplicationStatusObserver isApplicationActive]): |
| (webrtc::setApplicationStatus): |
| (webrtc::createVideoToolboxEncoderFactory): |
| (webrtc::createVideoToolboxDecoderFactory): |
| (webrtc::setH264HardwareEncoderAllowed): |
| (webrtc::isH264HardwareEncoderAllowed): |
| (webrtc::pixelBufferFromFrame): |
| * Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm: |
| (-[RTCCVPixelBuffer dealloc]): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm: |
| (-[RTCVideoDecoderH264 dealloc]): |
| (-[RTCVideoDecoderH264 setCallback:]): |
| (-[RTCVideoDecoderH264 releaseDecoder]): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm: |
| (-[RTCVideoEncoderH264 dealloc]): |
| (-[RTCVideoEncoderH264 setCallback:]): |
| (-[RTCVideoEncoderH264 releaseEncoder]): |
| (-[RTCVideoEncoderH264 resetCompressionSessionWithPixelFormat:]): |
| * WebKit/0001-Adapting-libwebrtc-H264-codec.patch: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2018-03-14 Youenn Fablet <youenn@apple.com> |
| |
| Update libwebrtc up to 36af4e9614f707f733eb2340fae66d6325aaac5b |
| https://bugs.webkit.org/show_bug.cgi?id=183481 |
| |
| Reviewed by Eric Carlson. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Source/webrtc/: refreshed |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2018-03-12 Tim Horton <timothy_horton@apple.com> |
| |
| Stop using SDK conditionals to control feature definitions |
| https://bugs.webkit.org/show_bug.cgi?id=183430 |
| <rdar://problem/38251619> |
| |
| Reviewed by Dan Bernstein. |
| |
| * Configurations/WebKitTargetConditionals.xcconfig: Renamed. |
| * Configurations/opus.xcconfig: |
| |
| 2018-03-12 Youenn Fablet <youenn@apple.com> |
| |
| Remove empty cpp files in Source/ThirdParty/libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=183529 |
| |
| Unreviewed. |
| Removing further empty files. |
| |
| * Source/webrtc/modules/audio_conference_mixer/BUILD.gn: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/DEPS: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/OWNERS: Removed. |
| * Source/webrtc/modules/video_coding/codecs/OWNERS: Removed. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.mm: Removed. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm: Removed. |
| * Source/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoViewTests.mm: Removed. |
| |
| 2018-03-12 youenn fablet <youenn@apple.com> |
| |
| Remove empty cpp files in Source/ThirdParty/libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=183529 |
| |
| Unreviewed. |
| |
| * libwebrtc.xcodeproj/project.pbxproj: fix the build. |
| |
| 2018-03-09 Youenn Fablet <youenn@apple.com> |
| |
| Remove empty cpp files in Source/ThirdParty/libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=183529 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/third_party/boringssl/boringssl_unittest.cc: Removed. |
| * Source/third_party/boringssl/src/ssl/ssl_privkey_cc.cc: Removed. |
| * Source/webrtc/common_audio/fir_filter.cc: Removed. |
| * Source/webrtc/config.cc: Removed. |
| * Source/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.cc: Removed. |
| * Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor.cc: Removed. |
| * Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc: Removed. |
| * Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.cc: Removed. |
| * Source/webrtc/modules/audio_coding/codecs/ilbc/test/empty.cc: Removed. |
| * Source/webrtc/modules/audio_coding/codecs/isac/empty.cc: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/RTPchange.cc: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/RTPencode.cc: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/RTPjitter.cc: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/RTPtimeshift.cc: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc: Removed. |
| * Source/webrtc/modules/audio_device/test/audio_device_test_api.cc: Removed. |
| * Source/webrtc/modules/audio_processing/aec3/decimator_by_4.cc: Removed. |
| * Source/webrtc/modules/audio_processing/aec3/decimator_by_4_unittest.cc: Removed. |
| * Source/webrtc/modules/audio_processing/agc2/digital_gain_applier.cc: Removed. |
| * Source/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc: Removed. |
| * Source/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc: Removed. |
| * Source/webrtc/modules/congestion_controller/congestion_controller.cc: Removed. |
| * Source/webrtc/modules/congestion_controller/congestion_controller_unittest.cc: Removed. |
| * Source/webrtc/modules/desktop_capture/resolution_change_detector.cc: Removed. |
| * Source/webrtc/modules/video_coding/codecs/test/plot_videoprocessor_integrationtest.cc: Removed. |
| * Source/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.cc: Removed. |
| * Source/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc: Removed. |
| * Source/webrtc/modules/video_coding/sequence_number_util_unittest.cc: Removed. |
| * Source/webrtc/p2p/base/dtlstransportchannel.cc: Removed. |
| * Source/webrtc/p2p/base/dtlstransportchannel_unittest.cc: Removed. |
| * Source/webrtc/p2p/base/transportcontroller.cc: Removed. |
| * Source/webrtc/p2p/base/transportcontroller_unittest.cc: Removed. |
| * Source/webrtc/p2p/quic/quicconnectionhelper.cc: Removed. |
| * Source/webrtc/p2p/quic/quicconnectionhelper_unittest.cc: Removed. |
| * Source/webrtc/p2p/quic/quicsession.cc: Removed. |
| * Source/webrtc/p2p/quic/quicsession_unittest.cc: Removed. |
| * Source/webrtc/p2p/quic/quictransport.cc: Removed. |
| * Source/webrtc/p2p/quic/quictransport_unittest.cc: Removed. |
| * Source/webrtc/p2p/quic/quictransportchannel.cc: Removed. |
| * Source/webrtc/p2p/quic/quictransportchannel_unittest.cc: Removed. |
| * Source/webrtc/p2p/quic/reliablequicstream.cc: Removed. |
| * Source/webrtc/p2p/quic/reliablequicstream_unittest.cc: Removed. |
| * Source/webrtc/pc/quicdatachannel.cc: Removed. |
| * Source/webrtc/pc/quicdatachannel_unittest.cc: Removed. |
| * Source/webrtc/pc/quicdatatransport.cc: Removed. |
| * Source/webrtc/pc/quicdatatransport_unittest.cc: Removed. |
| * Source/webrtc/pc/webrtcsession.cc: Removed. |
| * Source/webrtc/pc/webrtcsession_unittest.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/audio_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/filevideocapturer_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/media_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/native_handle_impl.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/null_audio_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/null_media_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/null_video_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/ownedfactoryandthreads.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/peerconnection_jni.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/rtcstatscollectorcallbackwrapper.cc: Removed. |
| * Source/webrtc/sdk/android/src/jni/video_jni.cc: Removed. |
| * Source/webrtc/system_wrappers/source/atomic32_darwin.cc: Removed. |
| * Source/webrtc/system_wrappers/source/atomic32_non_darwin_unix.cc: Removed. |
| * Source/webrtc/system_wrappers/source/atomic32_win.cc: Removed. |
| * Source/webrtc/system_wrappers/source/logcat_trace_context.cc: Removed. |
| * Source/webrtc/system_wrappers/source/trace_impl.cc: Removed. |
| * Source/webrtc/system_wrappers/source/trace_posix.cc: Removed. |
| * Source/webrtc/system_wrappers/source/trace_win.cc: Removed. |
| * Source/webrtc/test/testsupport/isolated_output.cc: Removed. |
| * Source/webrtc/test/testsupport/isolated_output_unittest.cc: Removed. |
| * Source/webrtc/test/testsupport/trace_to_stderr.cc: Removed. |
| * Source/webrtc/tools/agc/activity_metric.cc: Removed. |
| * Source/webrtc/tools/converter/converter.cc: Removed. |
| * Source/webrtc/tools/converter/rgba_to_i420_converter.cc: Removed. |
| * Source/webrtc/tools/event_log_visualizer/analyzer.cc: Removed. |
| * Source/webrtc/tools/event_log_visualizer/main.cc: Removed. |
| * Source/webrtc/tools/event_log_visualizer/plot_base.cc: Removed. |
| * Source/webrtc/tools/event_log_visualizer/plot_protobuf.cc: Removed. |
| * Source/webrtc/tools/event_log_visualizer/plot_python.cc: Removed. |
| * Source/webrtc/tools/force_mic_volume_max/force_mic_volume_max.cc: Removed. |
| * Source/webrtc/tools/frame_analyzer/frame_analyzer.cc: Removed. |
| * Source/webrtc/tools/frame_analyzer/reference_less_video_analysis.cc: Removed. |
| * Source/webrtc/tools/frame_analyzer/reference_less_video_analysis_lib.cc: Removed. |
| * Source/webrtc/tools/frame_analyzer/reference_less_video_analysis_unittest.cc: Removed. |
| * Source/webrtc/tools/frame_analyzer/video_quality_analysis.cc: Removed. |
| * Source/webrtc/tools/frame_analyzer/video_quality_analysis_unittest.cc: Removed. |
| * Source/webrtc/tools/frame_editing/frame_editing.cc: Removed. |
| * Source/webrtc/tools/frame_editing/frame_editing_lib.cc: Removed. |
| * Source/webrtc/tools/frame_editing/frame_editing_unittest.cc: Removed. |
| * Source/webrtc/tools/network_tester/config_reader.cc: Removed. |
| * Source/webrtc/tools/network_tester/network_tester_unittest.cc: Removed. |
| * Source/webrtc/tools/network_tester/packet_logger.cc: Removed. |
| * Source/webrtc/tools/network_tester/packet_sender.cc: Removed. |
| * Source/webrtc/tools/network_tester/server.cc: Removed. |
| * Source/webrtc/tools/network_tester/test_controller.cc: Removed. |
| * Source/webrtc/tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc: Removed. |
| * Source/webrtc/tools/simple_command_line_parser.cc: Removed. |
| * Source/webrtc/tools/simple_command_line_parser_unittest.cc: Removed. |
| * Source/webrtc/video/vie_encoder.cc: Removed. |
| * Source/webrtc/video/vie_encoder_unittest.cc: Removed. |
| * Source/webrtc/voice_engine/coder.cc: Removed. |
| * Source/webrtc/voice_engine/file_player.cc: Removed. |
| * Source/webrtc/voice_engine/file_player_unittests.cc: Removed. |
| * Source/webrtc/voice_engine/file_recorder.cc: Removed. |
| * Source/webrtc/voice_engine/output_mixer.cc: Removed. |
| * Source/webrtc/voice_engine/statistics.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/automated_mode.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/standard/codec_test.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/voe_conference_test.cc: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/voe_standard_test.cc: Removed. |
| * Source/webrtc/voice_engine/voe_codec_impl.cc: Removed. |
| * Source/webrtc/voice_engine/voe_codec_unittest.cc: Removed. |
| * Source/webrtc/voice_engine/voe_file_impl.cc: Removed. |
| * Source/webrtc/voice_engine/voe_network_impl.cc: Removed. |
| * Source/webrtc/voice_engine/voe_network_unittest.cc: Removed. |
| * Source/webrtc/voice_engine/voe_rtp_rtcp_impl.cc: Removed. |
| * Source/webrtc/voice_engine/voice_engine_fixture.cc: Removed. |
| |
| 2018-03-07 Youenn Fablet <youenn@apple.com> |
| |
| Update to libwebrtc revision 4e70a72571dd26b85c2385e9c618e343428df5d3 |
| https://bugs.webkit.org/show_bug.cgi?id=180843 |
| |
| Unreviewed. |
| Removed empty unused files. |
| |
| * Source/webrtc/audio/test/low_bandwidth_audio_test.h: Removed. |
| * Source/webrtc/config.h: Removed. |
| * Source/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h: Removed. |
| * Source/webrtc/media/engine/webrtccommon.h: Removed. |
| * Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h: Removed. |
| * Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.h: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h: Removed. |
| * Source/webrtc/modules/audio_coding/neteq/test/PayloadTypes.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/memory_pool.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/memory_pool_posix.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/memory_pool_win.h: Removed. |
| * Source/webrtc/modules/audio_conference_mixer/source/time_scheduler.h: Removed. |
| * Source/webrtc/modules/audio_device/test/audio_device_test_defines.h: Removed. |
| * Source/webrtc/modules/audio_processing/aec3/decimator_by_4.h: Removed. |
| * Source/webrtc/modules/audio_processing/agc2/digital_gain_applier.h: Removed. |
| * Source/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h: Removed. |
| * Source/webrtc/modules/congestion_controller/include/congestion_controller.h: Removed. |
| * Source/webrtc/modules/desktop_capture/resolution_change_detector.h: Removed. |
| * Source/webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h: Removed. |
| * Source/webrtc/modules/video_coding/codecs/test/predictive_packet_manipulator.h: Removed. |
| * Source/webrtc/modules/video_coding/sequence_number_util.h: Removed. |
| * Source/webrtc/p2p/base/candidate.h: Removed. |
| * Source/webrtc/p2p/base/dtlstransportchannel.h: Removed. |
| * Source/webrtc/p2p/base/faketransportcontroller.h: Removed. |
| * Source/webrtc/p2p/base/transportcontroller.h: Removed. |
| * Source/webrtc/p2p/quic/quicconnectionhelper.h: Removed. |
| * Source/webrtc/p2p/quic/quicsession.h: Removed. |
| * Source/webrtc/p2p/quic/quictransport.h: Removed. |
| * Source/webrtc/p2p/quic/quictransportchannel.h: Removed. |
| * Source/webrtc/p2p/quic/reliablequicstream.h: Removed. |
| * Source/webrtc/pc/quicdatachannel.h: Removed. |
| * Source/webrtc/pc/quicdatatransport.h: Removed. |
| * Source/webrtc/pc/test/mock_webrtcsession.h: Removed. |
| * Source/webrtc/pc/webrtcsession.h: Removed. |
| * Source/webrtc/sdk/android/src/jni/audio_jni.h: Removed. |
| * Source/webrtc/sdk/android/src/jni/media_jni.h: Removed. |
| * Source/webrtc/sdk/android/src/jni/native_handle_impl.h: Removed. |
| * Source/webrtc/sdk/android/src/jni/ownedfactoryandthreads.h: Removed. |
| * Source/webrtc/sdk/android/src/jni/rtcstatscollectorcallbackwrapper.h: Removed. |
| * Source/webrtc/sdk/android/src/jni/video_jni.h: Removed. |
| * Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCFileVideoCapturer.h: Removed. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.h: Removed. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h: Removed. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.h: Removed. |
| * Source/webrtc/system_wrappers/include/fix_interlocked_exchange_pointer_win.h: Removed. |
| * Source/webrtc/system_wrappers/include/logcat_trace_context.h: Removed. |
| * Source/webrtc/system_wrappers/include/static_instance.h: Removed. |
| * Source/webrtc/system_wrappers/include/trace.h: Removed. |
| * Source/webrtc/system_wrappers/source/trace_impl.h: Removed. |
| * Source/webrtc/system_wrappers/source/trace_posix.h: Removed. |
| * Source/webrtc/system_wrappers/source/trace_win.h: Removed. |
| * Source/webrtc/test/testsupport/isolated_output.h: Removed. |
| * Source/webrtc/test/testsupport/mock/mock_frame_writer.h: Removed. |
| * Source/webrtc/test/testsupport/trace_to_stderr.h: Removed. |
| * Source/webrtc/tools/converter/converter.h: Removed. |
| * Source/webrtc/tools/event_log_visualizer/analyzer.h: Removed. |
| * Source/webrtc/tools/event_log_visualizer/plot_base.h: Removed. |
| * Source/webrtc/tools/event_log_visualizer/plot_protobuf.h: Removed. |
| * Source/webrtc/tools/event_log_visualizer/plot_python.h: Removed. |
| * Source/webrtc/tools/frame_analyzer/reference_less_video_analysis_lib.h: Removed. |
| * Source/webrtc/tools/frame_analyzer/video_quality_analysis.h: Removed. |
| * Source/webrtc/tools/frame_editing/frame_editing_lib.h: Removed. |
| * Source/webrtc/tools/network_tester/config_reader.h: Removed. |
| * Source/webrtc/tools/network_tester/packet_logger.h: Removed. |
| * Source/webrtc/tools/network_tester/packet_sender.h: Removed. |
| * Source/webrtc/tools/network_tester/test_controller.h: Removed. |
| * Source/webrtc/tools/simple_command_line_parser.h: Removed. |
| * Source/webrtc/video/vie_encoder.h: Removed. |
| * Source/webrtc/video_receive_stream.h: Removed. |
| * Source/webrtc/video_send_stream.h: Removed. |
| * Source/webrtc/voice_engine/coder.h: Removed. |
| * Source/webrtc/voice_engine/file_player.h: Removed. |
| * Source/webrtc/voice_engine/file_recorder.h: Removed. |
| * Source/webrtc/voice_engine/include/voe_codec.h: Removed. |
| * Source/webrtc/voice_engine/include/voe_file.h: Removed. |
| * Source/webrtc/voice_engine/include/voe_network.h: Removed. |
| * Source/webrtc/voice_engine/include/voe_rtp_rtcp.h: Removed. |
| * Source/webrtc/voice_engine/mock/mock_voe_observer.h: Removed. |
| * Source/webrtc/voice_engine/monitor_module.h: Removed. |
| * Source/webrtc/voice_engine/output_mixer.h: Removed. |
| * Source/webrtc/voice_engine/statistics.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/automated_mode.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/voe_standard_test.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/voe_test_common.h: Removed. |
| * Source/webrtc/voice_engine/test/auto_test/voe_test_defines.h: Removed. |
| * Source/webrtc/voice_engine/voe_codec_impl.h: Removed. |
| * Source/webrtc/voice_engine/voe_file_impl.h: Removed. |
| * Source/webrtc/voice_engine/voe_network_impl.h: Removed. |
| * Source/webrtc/voice_engine/voe_rtp_rtcp_impl.h: Removed. |
| * Source/webrtc/voice_engine/voice_engine_fixture.h: Removed. |
| |
| 2018-03-07 Youenn Fablet <youenn@apple.com> |
| |
| Update to libwebrtc revision 4e70a72571dd26b85c2385e9c618e343428df5d3 |
| https://bugs.webkit.org/show_bug.cgi?id=180843 |
| |
| Unreviewed. |
| Removed folder as it is now unused. |
| |
| * Source/webrtc/base: Removed. |
| |
| 2017-12-18 Youenn Fablet <youenn@apple.com> |
| |
| Update to libwebrtc revision 4e70a72571dd26b85c2385e9c618e343428df5d3 |
| https://bugs.webkit.org/show_bug.cgi?id=180843 |
| |
| Reviewed by Eric Carlson. |
| |
| Updated libwebrtc as follows: |
| - Boringssl |
| - https://boringssl.googlesource.com/boringssl/ |
| - fc9c67599d9bdeb2e0467085133b81a8e28f77a4 |
| - Libwebrtc |
| - https://webrtc.googlesource.com/src |
| - 4e70a72571dd26b85c2385e9c618e343428df5d3 |
| - Libsrtp |
| - 1d45b8e599dc2db6ea3ae22dbc94a8c504652423 |
| - https://chromium.googlesource.com/chromium/deps/libsrtp.git |
| - Libyuv |
| - 12c904a97c81c3ef4cab0fc8fb1f0485b4ec4e8c |
| - https://chromium.googlesource.com/libyuv/libyuv.git |
| - Usrsctp |
| - f4819e1b177f7bfdd761c147f5a649b9f1a78c06 |
| - https://github.com/sctplab/usrsctp.git |
| |
| Below files have been modified to adapt for WebKit. |
| Patches for various parts are kept in WebKit folder. |
| In addition to these changes, VTB codecs and factories used by WebKit |
| are now added inside libwebrtc in webrtc/sdk/WebKit. |
| Future refactoring should consolidate these files. |
| |
| Not updated the following folders that are not used right now: |
| - Source/third_party/boringssl/linux-x86_64 |
| - Source/third_party/boringssl/mac-x86 |
| - Source/webrtc/data |
| - Source/third_party/boringssl/src/fuzz |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| * Source/third_party/boringssl/src/crypto/fipsmodule/aes/aes.c: |
| * Source/third_party/usrsctp/usrsctplib/netinet/sctp_input.c: |
| (sctp_process_cookie_existing): |
| * Source/third_party/usrsctp/usrsctplib/netinet/sctp_output.c: |
| * Source/third_party/usrsctp/usrsctplib/netinet/sctp_pcb.c: |
| * Source/third_party/usrsctp/usrsctplib/user_atomic.h: |
| * Source/webrtc/api/array_view.h: |
| (rtc::impl::ArrayViewBase::ArrayViewBase): |
| * Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc: |
| * Source/webrtc/api/datachannelinterface.h: |
| (webrtc::DataChannelObserver::OnBufferedAmountChange): |
| * Source/webrtc/api/jsep.h: |
| (webrtc::SessionDescriptionInterface::RemoveCandidates): |
| * Source/webrtc/api/mediastreaminterface.h: |
| (webrtc::VideoTrackInterface::set_content_hint): |
| (webrtc::AudioSourceInterface::SetVolume): |
| (webrtc::AudioSourceInterface::RegisterAudioObserver): |
| (webrtc::AudioSourceInterface::UnregisterAudioObserver): |
| (webrtc::AudioSourceInterface::AddSink): |
| (webrtc::AudioSourceInterface::RemoveSink): |
| (webrtc::AudioTrackInterface::GetSignalLevel): |
| * Source/webrtc/api/mediatypes.cc: |
| * Source/webrtc/api/peerconnectioninterface.h: |
| (webrtc::PeerConnectionInterface::AddTransceiver): |
| (webrtc::PeerConnectionInterface::CreateSender): |
| (webrtc::PeerConnectionInterface::GetStats): |
| (webrtc::PeerConnectionInterface::CreateOffer): |
| (webrtc::PeerConnectionInterface::CreateAnswer): |
| (webrtc::PeerConnectionInterface::SetRemoteDescription): |
| (webrtc::PeerConnectionInterface::UpdateIce): |
| (webrtc::PeerConnectionInterface::SetConfiguration): |
| (webrtc::PeerConnectionInterface::RemoveIceCandidates): |
| (webrtc::PeerConnectionInterface::SetBitrateAllocationStrategy): |
| (webrtc::PeerConnectionInterface::SetAudioPlayout): |
| (webrtc::PeerConnectionInterface::SetAudioRecording): |
| (webrtc::PeerConnectionInterface::StartRtcEventLog): |
| (webrtc::PeerConnectionObserver::OnIceCandidatesRemoved): |
| (webrtc::PeerConnectionObserver::OnIceConnectionReceivingChange): |
| (webrtc::PeerConnectionObserver::OnAddTrack): |
| (webrtc::PeerConnectionObserver::OnRemoveTrack): |
| (webrtc::PeerConnectionFactoryInterface::CreateVideoSource): |
| * Source/webrtc/api/umametrics.h: |
| (webrtc::MetricsObserverInterface::IncrementEnumCounter): |
| * Source/webrtc/api/video_codecs/video_decoder.h: |
| (webrtc::DecodedImageCallback::Decoded): |
| (webrtc::DecodedImageCallback::ReceivedDecodedReferenceFrame): |
| (webrtc::DecodedImageCallback::ReceivedDecodedFrame): |
| * Source/webrtc/api/video_codecs/video_encoder.h: |
| (webrtc::EncodedImageCallback::OnDroppedFrame): |
| * Source/webrtc/common_video/include/frame_callback.h: |
| (webrtc::EncodedFrameObserver::OnEncodeTiming): |
| * Source/webrtc/common_video/video_frame_buffer.cc: |
| * Source/webrtc/logging/rtc_event_log/rtc_event_log.h: |
| (webrtc::RtcEventLog::Create): |
| * Source/webrtc/media/base/mediachannel.h: |
| (cricket::DataMediaChannel::GetStats): |
| (cricket::DataMediaChannel::OnNetworkRouteChanged): |
| * Source/webrtc/media/engine/internaldecoderfactory.cc: |
| * Source/webrtc/media/engine/internalencoderfactory.cc: |
| * Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc: |
| * Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc: |
| * Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc: |
| * Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc: |
| * Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc: |
| * Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc: |
| * Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc: |
| * Source/webrtc/modules/audio_device/android/audio_device_template.h: |
| * Source/webrtc/modules/audio_device/android/audio_record_jni.cc: |
| * Source/webrtc/modules/audio_device/include/audio_device.h: |
| (webrtc::AudioDeviceModule::SetRecordingChannel): |
| (webrtc::AudioDeviceModule::RecordingChannel const): |
| (webrtc::AudioDeviceModule::SetRecordingSampleRate): |
| (webrtc::AudioDeviceModule::RecordingSampleRate const): |
| (webrtc::AudioDeviceModule::SetPlayoutSampleRate): |
| (webrtc::AudioDeviceModule::PlayoutSampleRate const): |
| (webrtc::AudioDeviceModule::SetLoudspeakerStatus): |
| (webrtc::AudioDeviceModule::GetLoudspeakerStatus const): |
| * Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc: |
| * Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h: |
| (webrtc::DefaultTemporalLayersChecker::BufferState::BufferState): |
| * Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h: |
| * Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h: |
| * Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h: |
| * Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h: |
| * Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h: |
| * Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h: |
| (webrtc::TemporalLayers::FrameConfig::operator== const): |
| (webrtc::TemporalLayers::FrameConfig::operator!= const): |
| (webrtc::TemporalLayersChecker::~TemporalLayersChecker): |
| (webrtc::TemporalLayersChecker::BufferState::BufferState): |
| * Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h: |
| * Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc: |
| * Source/webrtc/modules/video_coding/qp_parser.cc: |
| * Source/webrtc/modules/video_coding/video_codec_initializer.cc: |
| * Source/webrtc/ortc/ortcfactory.cc: |
| * Source/webrtc/ortc/rtpparametersconversion.cc: |
| * Source/webrtc/p2p/base/icetransportinternal.h: |
| (cricket::IceTransportInternal::SetIceProtocolType): |
| * Source/webrtc/p2p/base/port.h: |
| (cricket::Port::HandleConnectionDestroyed): |
| * Source/webrtc/p2p/base/stun.h: |
| (cricket::StunAttribute::SetOwner): |
| * Source/webrtc/p2p/base/stunrequest.h: |
| (cricket::StunRequest::Prepare): |
| (cricket::StunRequest::OnResponse): |
| (cricket::StunRequest::OnErrorResponse): |
| * Source/webrtc/rtc_base/checks.h: |
| * Source/webrtc/rtc_base/flags.cc: |
| * Source/webrtc/rtc_base/location.h: |
| * Source/webrtc/rtc_base/messagehandler.h: |
| (rtc::FunctorMessageHandler::OnMessage): |
| * Source/webrtc/rtc_base/network.h: |
| (rtc::NetworkManager::GetAnyAddressNetworks): |
| * Source/webrtc/rtc_base/numerics/safe_conversions.h: |
| (rtc::saturated_cast): |
| * Source/webrtc/rtc_base/numerics/safe_conversions_impl.h: |
| * Source/webrtc/rtc_base/opensslidentity.cc: |
| * Source/webrtc/rtc_base/sanitizer.h: |
| (rtc_AsanPoison): |
| (rtc_AsanUnpoison): |
| (rtc_MsanMarkUninitialized): |
| (rtc_MsanCheckInitialized): |
| * Source/webrtc/rtc_base/socketserver.h: |
| (rtc::SocketServer::SetMessageQueue): |
| * Source/webrtc/rtc_base/stream.h: |
| (rtc::StreamInterface::ConsumeReadData): |
| (rtc::StreamInterface::ConsumeWriteBuffer): |
| * Source/webrtc/rtc_base/stringize_macros.h: |
| * Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp: Added. |
| (webrtc::VideoToolboxVideoDecoderFactory::~VideoToolboxVideoDecoderFactory): |
| (webrtc::VideoToolboxVideoDecoderFactory::Add): |
| (webrtc::VideoToolboxVideoDecoderFactory::Remove): |
| (webrtc::VideoToolboxVideoDecoderFactory::SetActive): |
| (webrtc::VideoToolboxVideoDecoderFactory::CreateVideoDecoder): |
| (webrtc::CreateH264Format): |
| (webrtc::VideoToolboxVideoDecoderFactory::GetSupportedFormats const): |
| * Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.h: Renamed from Source/WebCore/platform/mediastream/libwebrtc/VideoToolBoxDecoderFactory.h. |
| * Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp: Added. |
| (webrtc::VideoToolboxVideoEncoderFactory::~VideoToolboxVideoEncoderFactory): |
| (webrtc::VideoToolboxVideoEncoderFactory::Add): |
| (webrtc::VideoToolboxVideoEncoderFactory::Remove): |
| (webrtc::VideoToolboxVideoEncoderFactory::SetActive): |
| (webrtc::CreateH264Format): |
| (webrtc::VideoToolboxVideoEncoderFactory::GetSupportedFormats const): |
| (webrtc::VideoToolboxVideoEncoderFactory::QueryVideoEncoder const): |
| (webrtc::VideoToolboxVideoEncoderFactory::CreateVideoEncoder): |
| * Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.h: Renamed from Source/WebCore/platform/mediastream/libwebrtc/VideoToolBoxEncoderFactory.h. |
| * Source/webrtc/sdk/WebKit/decoder.h: Added. |
| (webrtc::H264VideoToolboxDecoder::SetActive): |
| * Source/webrtc/sdk/WebKit/decoder.mm: Added. |
| (webrtc::H264VideoToolboxDecoder::H264VideoToolboxDecoder): |
| (webrtc::H264VideoToolboxDecoder::~H264VideoToolboxDecoder): |
| (webrtc::H264VideoToolboxDecoder::ClearFactory): |
| (webrtc::H264VideoToolboxDecoder::InitDecode): |
| (webrtc::H264VideoToolboxDecoder::Decode): |
| * Source/webrtc/sdk/WebKit/encoder.h: Copied from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h. |
| (webrtc::H264VideoToolboxEncoder::ClearFactory): |
| (webrtc::H264VideoToolboxEncoder::SetActive): |
| * Source/webrtc/sdk/WebKit/encoder.mm: Copied from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.mm. |
| (internal::CreateCFDictionary): |
| (internal::CFStringToString): |
| (internal::SetVTSessionProperty): |
| (internal::FrameEncodeParams::FrameEncodeParams): |
| (internal::CopyVideoFrameToPixelBuffer): |
| (internal::CreatePixelBuffer): |
| (internal::VTCompressionOutputCallback): |
| (internal::ExtractProfile): |
| (webrtc::H264VideoToolboxEncoder::H264VideoToolboxEncoder): |
| (webrtc::H264VideoToolboxEncoder::~H264VideoToolboxEncoder): |
| (webrtc::H264VideoToolboxEncoder::InitEncode): |
| (webrtc::H264VideoToolboxEncoder::Encode): |
| * Source/webrtc/sdk/WebKit/encoder_vcp.h: Renamed from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h. |
| (webrtc::H264VideoToolboxEncoderVCP::ClearFactory): |
| * Source/webrtc/sdk/WebKit/encoder_vcp.mm: Renamed from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.mm. |
| (internal::SetVTSessionProperty): |
| (internal::CopyVideoFrameToPixelBuffer): |
| (internal::CreatePixelBuffer): |
| (internal::ExtractProfile): |
| (webrtc::H264VideoToolboxEncoderVCP::H264VideoToolboxEncoderVCP): |
| (webrtc::H264VideoToolboxEncoderVCP::Encode): |
| * Source/webrtc/test/rtp_file_reader.cc: |
| * Source/webrtc/voice_engine/utility.cc: |
| * WebKit/0001-Tweaking-boringssl-include-of-internal.h.patch: Renamed from Source/ThirdParty/libwebrtc/WebKit/patch-boringssl. |
| * WebKit/0002-Fixing-usrctp-library-compilation-errors.patch: Added. |
| * WebKit/0003-Fixing-VP8-files.patch: Added. |
| * WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch: Added. |
| * WebKit/0005-Fix-RTC_FATAL.patch: Added. |
| * WebKit/0006-Disabling-VP8.patch: Added. |
| * WebKit/0007-Fix-RTC_STRINGIZE.patch: Added. |
| * WebKit/0008-Fix-sanitizer.patch: Added. |
| * WebKit/patch-libwebrtc: Removed. |
| * WebKit/patch-usrsctp: Removed. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2018-01-27 Dan Bernstein <mitz@apple.com> |
| |
| HaveInternalSDK includes should be "#include?" |
| https://bugs.webkit.org/show_bug.cgi?id=179670 |
| |
| * Configurations/Base.xcconfig: |
| |
| 2018-01-26 Youenn Fablet <youenn@apple.com> |
| |
| Disable VCP for MacOS |
| https://bugs.webkit.org/show_bug.cgi?id=182183 |
| <rdar://problem/36919791> |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/VideoProcessingSoftLink.h: |
| |
| 2018-01-19 Joseph Pecoraro <pecoraro@apple.com> |
| |
| Follow-up build fix for r227206. |
| |
| Unreviewed. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/VideoProcessingSoftLink.h: |
| Avoid duplicate and different definitions of ALWAYS_INLINE. |
| |
| 2018-01-19 Youenn Fablet <youenn@apple.com> |
| |
| Softlink VideoProcessing in WebKit |
| https://bugs.webkit.org/show_bug.cgi?id=181853 |
| <rdar://problem/36590005> |
| |
| Reviewed by Eric Carlson. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/VideoProcessingSoftLink.cpp: Added. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/VideoProcessingSoftLink.h: Added. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.mm: |
| (internal::SetVTSessionProperty): |
| (webrtc::H264VideoToolboxEncoderVCP::Encode): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.mm: |
| (webrtc::VideoToolboxVideoEncoderFactory::VideoToolboxVideoEncoderFactory): |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2018-01-18 Dan Bernstein <mitz@apple.com> |
| |
| [Xcode] Streamline and future-proof target-macOS-version-dependent build setting definitions |
| https://bugs.webkit.org/show_bug.cgi?id=181803 |
| |
| Reviewed by Tim Horton. |
| |
| * Configurations/Base.xcconfig: Updated. |
| * Configurations/DebugRelease.xcconfig: Ditto. |
| * Configurations/macOSTargetConditionals.xcconfig: Added. Defines helper build settings |
| useful for defining settings that depend on the target macOS version. |
| * Configurations/opus.xcconfig: Adopted macOSTargetConditionals helper. |
| |
| 2018-01-08 David Kilzer <ddkilzer@apple.com> |
| |
| libwebrtc: Fix 'ld: warning: cannot export hidden symbol' messages |
| <https://webkit.org/b/181378> |
| |
| Reviewed by Youenn Fablet. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| - Remove 117 symbols that are not currently exported. These |
| warnings only appear in Release and Production builds. |
| |
| 2018-01-05 Youenn Fablet <youenn@apple.com> |
| |
| Close WebRTC sockets when marked as defunct |
| https://bugs.webkit.org/show_bug.cgi?id=177324 |
| rdar://problem/35244931 |
| |
| Reviewed by Eric Carlson. |
| |
| In case selected sockets return an error when trying to accept an incoming socket, |
| check whether the socket is defunct or not. |
| If so, close it properly. |
| |
| * Source/webrtc/base/asynctcpsocket.cc: |
| * Source/webrtc/base/physicalsocketserver.cc: |
| * Source/webrtc/base/socket.h: |
| |
| 2017-12-15 Dan Bernstein <mitz@apple.com> |
| |
| libwebrtc installs an extra copy of encoder_vcp.h under /usr/local/include |
| https://bugs.webkit.org/show_bug.cgi?id=180858 |
| |
| Reviewed by Anders Carlsson. |
| |
| * libwebrtc.xcodeproj/project.pbxproj: Demoted the header from Private to Project. A script build phase |
| copies it to the correct location under /usr/local/include/webrtc. |
| |
| 2017-12-14 David Kilzer <ddkilzer@apple.com> |
| |
| Enable -Wstrict-prototypes for WebKit |
| <https://webkit.org/b/180757> |
| <rdar://problem/36024132> |
| |
| Rubber-stamped by Joseph Pecoraro. |
| |
| * Configurations/Base.xcconfig: |
| (CLANG_WARN_STRICT_PROTOTYPES): Add. Set to YES. |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/user_socket.c: |
| (wakeup_one): Modernize function argument declarations. |
| (getsockaddr): Ditto. |
| * Source/webrtc/common_audio/signal_processing/include/signal_processing_library.h: |
| (WebRtcSpl_Init): Add 'void' to C function declaration. |
| * Source/webrtc/common_audio/vad/include/webrtc_vad.h: |
| (WebRtcVad_Create): Ditto. |
| * Source/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h: |
| (WebRtcIsacfix_InitTransform): Ditto. |
| * Source/webrtc/modules/audio_processing/agc/legacy/gain_control.h: |
| (WebRtcAgc_Create): Ditto. |
| * Source/webrtc/modules/audio_processing/ns/noise_suppression.h: |
| (WebRtcNs_Create): Ditto. |
| (WebRtcNs_num_freq): Ditto. |
| * Source/webrtc/modules/audio_processing/ns/noise_suppression_x.h: |
| (WebRtcNsx_Create): Ditto. |
| (WebRtcNsx_num_freq): Ditto. |
| |
| 2017-12-11 Youenn Fablet <youenn@apple.com> |
| |
| Use VCP H264 encoder for platforms supporting it |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| rdar://problem/35180773 |
| |
| Reviewed by Eric Carlson. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h: Added. |
| (webrtc::H264VideoToolboxEncoderVCP::SetActive): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.mm: Copied from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm: |
| (internal::CFStringToString): |
| (internal::SetVTSessionProperty): |
| (internal::CopyVideoFrameToPixelBuffer): |
| (internal::CreatePixelBuffer): |
| (internal::VTCompressionOutputCallback): |
| (internal::ExtractProfile): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.mm: |
| (webrtc::VideoToolboxVideoEncoderFactory::VideoToolboxVideoEncoderFactory): |
| (webrtc::VideoToolboxVideoEncoderFactory::CreateSupportedVideoEncoder): |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-12-11 Tim Horton <timothy_horton@apple.com> |
| |
| Stop using deprecated target conditional for simulator builds |
| https://bugs.webkit.org/show_bug.cgi?id=180662 |
| <rdar://problem/35136156> |
| |
| Reviewed by Simon Fraser. |
| |
| * Source/third_party/libyuv/source/mjpeg_decoder.cc: |
| * Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m: |
| (-[ARDAppClient createLocalVideoTrack]): |
| * Source/webrtc/examples/objc/AppRTCMobile/tests/ARDAppClient_xctest.mm: |
| * Source/webrtc/modules/audio_device/ios/audio_device_ios.mm: |
| (webrtc::LogDeviceInfo): |
| |
| 2017-11-06 Commit Queue <commit-queue@webkit.org> |
| |
| Unreviewed, rolling out r224497. |
| https://bugs.webkit.org/show_bug.cgi?id=179335 |
| |
| It is breaking internal builds (Requested by youenn on |
| #webkit). |
| |
| Reverted changeset: |
| |
| "Use VCP H264 encoder for platforms supporting it" |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| https://trac.webkit.org/changeset/224497 |
| |
| 2017-11-06 Youenn Fablet <youenn@apple.com> |
| |
| Use VCP H264 encoder for platforms supporting it |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| rdar://problem/35180773 |
| |
| Reviewed by Eric Carlson. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h: Added. |
| (webrtc::H264VideoToolboxEncoderVCP::SetActive): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.mm: Copied from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm: |
| (internal::CFStringToString): |
| (internal::SetVTSessionProperty): |
| (internal::CopyVideoFrameToPixelBuffer): |
| (internal::CreatePixelBuffer): |
| (internal::VTCompressionOutputCallback): |
| (internal::ExtractProfile): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.mm: |
| (webrtc::VideoToolboxVideoEncoderFactory::VideoToolboxVideoEncoderFactory): |
| (webrtc::VideoToolboxVideoEncoderFactory::CreateSupportedVideoEncoder): |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-11-03 Commit Queue <commit-queue@webkit.org> |
| |
| Unreviewed, rolling out r224428, r224435, and r224440. |
| https://bugs.webkit.org/show_bug.cgi?id=179274 |
| |
| Broke iOS and internal builds (Requested by ryanhaddad on |
| #webkit). |
| |
| Reverted changesets: |
| |
| "Use VCP H264 encoder for platforms supporting it" |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| https://trac.webkit.org/changeset/224428 |
| |
| "Use VCP H264 encoder for platforms supporting it" |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| https://trac.webkit.org/changeset/224435 |
| |
| "Use VCP H264 encoder for platforms supporting it" |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| https://trac.webkit.org/changeset/224440 |
| |
| 2017-11-03 Youenn Fablet <youenn@apple.com> |
| |
| Use VCP H264 encoder for platforms supporting it |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| rdar://problem/35180773 |
| |
| Unreviewed. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h: build fix for iOS. |
| |
| 2017-11-03 Youenn Fablet <youenn@apple.com> |
| |
| Use VCP H264 encoder for platforms supporting it |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| rdar://problem/35180773 |
| |
| Unreviewed. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h: build fix. |
| |
| 2017-11-03 Youenn Fablet <youenn@apple.com> |
| |
| Use VCP H264 encoder for platforms supporting it |
| https://bugs.webkit.org/show_bug.cgi?id=179076 |
| rdar://problem/35180773 |
| |
| Reviewed by Eric Carlson. |
| |
| * Configurations/libwebrtc.iOS.exp: |
| * Configurations/libwebrtc.iOSsim.exp: |
| * Configurations/libwebrtc.mac.exp: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.h: Added. |
| (webrtc::H264VideoToolboxEncoderVCP::SetActive): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoProcessing/encoder_vcp.mm: Copied from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm: |
| (internal::CFStringToString): |
| (internal::SetVTSessionProperty): |
| (internal::CopyVideoFrameToPixelBuffer): |
| (internal::CreatePixelBuffer): |
| (internal::VTCompressionOutputCallback): |
| (internal::ExtractProfile): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.mm: |
| (webrtc::VideoToolboxVideoEncoderFactory::VideoToolboxVideoEncoderFactory): |
| (webrtc::VideoToolboxVideoEncoderFactory::CreateSupportedVideoEncoder): |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-10-04 Commit Queue <commit-queue@webkit.org> |
| |
| Unreviewed, rolling out r222775. |
| https://bugs.webkit.org/show_bug.cgi?id=177890 |
| |
| Significantly increased the WebKit build time (Requested by |
| rniwa on #webkit). |
| |
| Reverted changeset: |
| |
| "Build libwebrtc unit tests executables" |
| https://bugs.webkit.org/show_bug.cgi?id=177211 |
| http://trac.webkit.org/changeset/222775 |
| |
| 2017-10-03 Youenn Fablet <youenn@apple.com> |
| |
| Remove no longer needed WebRTC build infrastructure |
| https://bugs.webkit.org/show_bug.cgi?id=177756 |
| |
| Reviewed by Alejandro G. Castro. |
| |
| * WebKit/project.json: Removed. |
| * WebKit/rtc_sdk_framework_objc_info_plist.plist: Removed. |
| |
| 2017-10-03 Youenn Fablet <youenn@apple.com> |
| |
| Build libwebrtc unit tests executables |
| https://bugs.webkit.org/show_bug.cgi?id=177211 |
| |
| Reviewed by Alex Christensen. |
| |
| Adding support for a new target called unittests that will be several executables. |
| Each executable run unit tests dedicated to a part of libwebrtc. |
| |
| Adding one target/executable per unit test suite. |
| Adding one composite target to build all unit test targets. |
| Adding a target to build a static libwebrtctest library. |
| The static libwebrtctest library is then linked to each unit test executable which is also linked to libwebrtc dylib. |
| |
| Some unit tests require a default codec (VP8) that is disabled in libwebrtc. |
| This ends up making some tests crashing. |
| An additional work should follow to execute only the meaningful subset of tests. |
| |
| * Configurations/libwebrtc-base.xcconfig: Added. |
| * Configurations/libwebrtc-test-static.xcconfig: Added. |
| * Configurations/rtc_pc_unittests.xcconfig: Added. |
| * Source/third_party/gflags/gen/posix/include/private/config.h: |
| * Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc: Replacing FATAL by RTC_FATAL. |
| * Source/webrtc/sdk/objc/Framework/Classes/Common/helpers.mm: Removing UIKit dependency. |
| * Source/webrtc/test/gmock.h: Using googletest version instead of checking in testing folder. |
| * Source/webrtc/test/gtest.h: Ditto. |
| * Source/webrtc/test/rtp_file_reader.cc: Replacing FATAL by RTC_FATAL. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-09-29 Matt Lewis <jlewis3@apple.com> |
| |
| Unreviewed, rolling out r222652. |
| |
| This broke an internal build. |
| |
| Reverted changeset: |
| |
| "Build libwebrtc unit tests executables" |
| https://bugs.webkit.org/show_bug.cgi?id=177211 |
| http://trac.webkit.org/changeset/222652 |
| |
| 2017-09-29 Youenn Fablet <youenn@apple.com> |
| |
| Build libwebrtc unit tests executables |
| https://bugs.webkit.org/show_bug.cgi?id=177211 |
| |
| Reviewed by Alex Christensen. |
| |
| Adding support for a new target called unittests that will be several executables. |
| Each executable run unit tests dedicated to a part of libwebrtc. |
| |
| Adding one target/executable per unit test suite. |
| Adding one composite target to build all unit test targets. |
| Adding a target to build a static libwebrtctest library. |
| The static libwebrtctest library is then linked to each unit test executable which is also linked to libwebrtc dylib. |
| |
| Some unit tests require a default codec (VP8) that is disabled in libwebrtc. |
| This ends up making some tests crashing. |
| An additional work should follow to execute only the meaningful subset of tests. |
| |
| * Configurations/libwebrtc-base.xcconfig: Added. |
| * Configurations/libwebrtc-test-static.xcconfig: Added. |
| * Configurations/rtc_pc_unittests.xcconfig: Added. |
| * Source/third_party/gflags/gen/posix/include/private/config.h: |
| * Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc: Replacing FATAL by RTC_FATAL. |
| * Source/webrtc/sdk/objc/Framework/Classes/Common/helpers.mm: Removing UIKit dependency. |
| * Source/webrtc/test/gmock.h: Using googletest version instead of checking in testing folder. |
| * Source/webrtc/test/gtest.h: Ditto. |
| * Source/webrtc/test/rtp_file_reader.cc: Replacing FATAL by RTC_FATAL. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-09-27 Ryan Haddad <ryanhaddad@apple.com> |
| |
| Unreviewed, rolling out r222537. |
| |
| This change broke internal builds. |
| |
| Reverted changeset: |
| |
| "Build libwebrtc unit tests executables" |
| https://bugs.webkit.org/show_bug.cgi?id=177211 |
| http://trac.webkit.org/changeset/222537 |
| |
| 2017-09-26 Youenn Fablet <youenn@apple.com> |
| |
| Build libwebrtc unit tests executables |
| https://bugs.webkit.org/show_bug.cgi?id=177211 |
| |
| Reviewed by Alex Christensen. |
| |
| Adding support for a new target called unittests that will be several executables. |
| Each executable run unit tests dedicated to a part of libwebrtc. |
| |
| Adding one target/executable per unit test suite. |
| Adding one composite target to build all unit test targets. |
| Adding a target to build a static libwebrtctest library. |
| The static libwebrtctest library is then linked to each unit test executable which is also linked to libwebrtc dylib. |
| |
| Some unit tests require a default codec (VP8) that is disabled in libwebrtc. |
| This ends up making some tests crashing. |
| An additional work should follow to execute only the meaningful subset of tests. |
| |
| * Configurations/libwebrtc-base.xcconfig: Added. |
| * Configurations/libwebrtc-test-static.xcconfig: Added. |
| * Configurations/rtc_pc_unittests.xcconfig: Added. |
| * Source/third_party/gflags/gen/posix/include/private/config.h: |
| * Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc: Replacing FATAL by RTC_FATAL. |
| * Source/webrtc/sdk/objc/Framework/Classes/Common/helpers.mm: Removing UIKit dependency. |
| * Source/webrtc/test/gmock.h: Using googletest version instead of checking in testing folder. |
| * Source/webrtc/test/gtest.h: Ditto. |
| * Source/webrtc/test/rtp_file_reader.cc: Replacing FATAL by RTC_FATAL. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-09-26 Youenn Fablet <youenn@apple.com> |
| |
| Remove unnecessary libwebrtc dependencies |
| https://bugs.webkit.org/show_bug.cgi?id=177494 |
| |
| Reviewed by Alex Christensen. |
| |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-09-25 Youenn Fablet <youenn@apple.com> |
| |
| WebRTC video does not resume receiving when switching back to Safari 11 on iOS |
| https://bugs.webkit.org/show_bug.cgi?id=175472 |
| <rdar://problem/33860863> |
| |
| Reviewed by Darin Adler. |
| |
| Adding a method to disable any decoding/encoding task. |
| When reenabling the decoder, the decoder will request an I frame after failing the first initial decoding task. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.h: |
| (webrtc::H264VideoToolboxDecoder::SetActive): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/decoder.mm: |
| (webrtc::H264VideoToolboxDecoder::Decode): |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm: |
| (webrtc::H264VideoToolboxEncoder::Encode): |
| |
| 2017-09-25 Youenn Fablet <youenn@apple.com> |
| |
| Adding per-platform libwebrtc export files |
| https://bugs.webkit.org/show_bug.cgi?id=177465 |
| |
| Reviewed by Alex Christensen. |
| |
| Using per platform export symbol files for libwebrtc.dylib. |
| This allows exporting platform-specific symbols that are used by libwebrtc unit tests. |
| |
| * Configurations/libwebrtc.iOS.exp: Added. |
| * Configurations/libwebrtc.iOSsim.exp: Added. |
| * Configurations/libwebrtc.mac.exp: Added. |
| * Configurations/libwebrtc.exp: Removed. |
| * Configurations/libwebrtc.xcconfig: |
| * libwebrtc.xcodeproj/project.pbxproj: Adding ISAC/fix codec files used for |
| by audio codec unit tests to libwebrtc.dylib. This files will allow us to add support to the ISAC/fix codec. |
| |
| 2017-09-23 Youenn Fablet <youenn@apple.com> |
| |
| Export libwebrtc symbols through an export file |
| https://bugs.webkit.org/show_bug.cgi?id=177344 |
| |
| Reviewed by Darin Adler. |
| |
| Removing export changes made to libwebrtc. |
| Exporting based on libwebrtc.exp file. |
| |
| * Configurations/Base.xcconfig: |
| * Configurations/libwebrtc.exp: Added. |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/api/jsep.h: |
| (): Deleted. |
| * Source/webrtc/api/mediatypes.h: |
| * Source/webrtc/api/peerconnectioninterface.h: |
| * Source/webrtc/api/rtcerror.h: |
| * Source/webrtc/api/stats/rtcstats.h: |
| * Source/webrtc/api/stats/rtcstatsreport.h: |
| (): Deleted. |
| * Source/webrtc/api/video/i420_buffer.h: |
| * Source/webrtc/api/video/video_frame.h: |
| (): Deleted. |
| * Source/webrtc/api/video/video_frame_buffer.h: |
| * Source/webrtc/base/asyncpacketsocket.h: |
| * Source/webrtc/base/asyncresolverinterface.h: |
| (): Deleted. |
| * Source/webrtc/base/checks.h: |
| (): Deleted. |
| * Source/webrtc/base/copyonwritebuffer.h: |
| (): Deleted. |
| * Source/webrtc/base/event.h: |
| (): Deleted. |
| * Source/webrtc/base/export.h: Removed. |
| * Source/webrtc/base/helpers.h: |
| * Source/webrtc/base/ipaddress.h: |
| * Source/webrtc/base/location.h: |
| (): Deleted. |
| * Source/webrtc/base/logging.h: |
| * Source/webrtc/base/messagehandler.h: |
| * Source/webrtc/base/network.h: |
| * Source/webrtc/base/proxyinfo.h: |
| * Source/webrtc/base/socketaddress.h: |
| (): Deleted. |
| * Source/webrtc/base/thread.h: |
| * Source/webrtc/common_video/include/i420_buffer_pool.h: |
| (): Deleted. |
| * Source/webrtc/common_video/include/video_frame_buffer.h: |
| (): Deleted. |
| * Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h: |
| * Source/webrtc/media/engine/webrtcvideoencoderfactory.h: |
| (): Deleted. |
| * Source/webrtc/p2p/base/basicpacketsocketfactory.h: |
| (): Deleted. |
| * Source/webrtc/p2p/client/basicportallocator.h: |
| * Source/webrtc/pc/mediastream.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.h: |
| (): Deleted. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h: |
| (): Deleted. |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/videocodecfactory.h: |
| (): Deleted. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-09-20 Youenn Fablet <youenn@apple.com> |
| |
| Upstream googletest framework |
| https://bugs.webkit.org/show_bug.cgi?id=177252 |
| |
| Reviewed by Alex Christensen. |
| |
| This is used by libwebrtc. |
| |
| * Source/third_party/googletest: Added. |
| |
| 2017-09-15 Alicia Boya García <aboya@igalia.com> |
| |
| Normalize line terminators in jsoncpp Visual Studio files |
| https://bugs.webkit.org/show_bug.cgi?id=176991 |
| |
| Reviewed by Konstantin Tokarev. |
| |
| * Source/third_party/jsoncpp/source/makefiles/vs71/jsoncpp.sln: |
| * Source/third_party/jsoncpp/source/makefiles/vs71/jsontest.vcproj: |
| * Source/third_party/jsoncpp/source/makefiles/vs71/lib_json.vcproj: |
| * Source/third_party/jsoncpp/source/makefiles/vs71/test_lib_json.vcproj: |
| |
| 2017-07-18 Andy Estes <aestes@apple.com> |
| |
| [Xcode] Enable CLANG_WARN_OBJC_LITERAL_CONVERSION |
| https://bugs.webkit.org/show_bug.cgi?id=174631 |
| |
| Reviewed by Sam Weinig. |
| |
| * Configurations/Base.xcconfig: |
| |
| 2017-07-18 Andy Estes <aestes@apple.com> |
| |
| [Xcode] Enable CLANG_WARN_NON_LITERAL_NULL_CONVERSION |
| https://bugs.webkit.org/show_bug.cgi?id=174631 |
| |
| Reviewed by Dan Bernstein. |
| |
| * Configurations/Base.xcconfig: |
| |
| 2017-07-18 Andy Estes <aestes@apple.com> |
| |
| [Xcode] Enable CLANG_WARN_BLOCK_CAPTURE_AUTORELEASING |
| https://bugs.webkit.org/show_bug.cgi?id=174631 |
| |
| Reviewed by Darin Adler. |
| |
| * Configurations/Base.xcconfig: |
| |
| 2017-07-03 Andy Estes <aestes@apple.com> |
| |
| [Xcode] Add an experimental setting to build with ccache |
| https://bugs.webkit.org/show_bug.cgi?id=173875 |
| |
| Reviewed by Tim Horton. |
| |
| * Configurations/DebugRelease.xcconfig: Included ccache.xcconfig. |
| |
| 2017-07-01 Dan Bernstein <mitz@apple.com> |
| |
| [macOS] Remove code only needed when building for OS X Yosemite |
| https://bugs.webkit.org/show_bug.cgi?id=174067 |
| |
| Reviewed by Tim Horton. |
| |
| * Configurations/Base.xcconfig: |
| * Configurations/DebugRelease.xcconfig: |
| |
| 2017-06-27 Youenn Fablet <youenn@apple.com> |
| |
| Update boringssl to c8ff30cbe716c72279a6f6a9d7d7d0d4091220fa |
| https://bugs.webkit.org/show_bug.cgi?id=173676 |
| |
| Reviewed by Alex Christensen. |
| |
| * Configurations/boringssl.xcconfig: Enabling ASM. |
| * Source/third_party/boringssl/BUILD.generated.gni: |
| * Source/third_party/boringssl: Updated folder according new revision. |
| * WebKit/patch-boringssl: Added, needed to fix some files to disable warnings. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-06-27 Youenn Fablet <youenn@apple.com> |
| |
| Refresh usrsctp to Source/ThirdParty/libwebrtc/WebKit/patch-usrsctp and libsrtp to ccf84786f8ef803cb9c75e919e5a3976b9f5a67 |
| https://bugs.webkit.org/show_bug.cgi?id=173673 |
| |
| Reviewed by Sam Weinig. |
| |
| * Source/third_party/libsrtp/README.chromium: |
| * Source/third_party/libsrtp/srtp/srtp.c: |
| (srtp_stream_init_keys): |
| (srtp_calc_aead_iv_srtcp): |
| (srtp_protect_rtcp_aead): |
| (srtp_unprotect_rtcp_aead): |
| * Source/third_party/libsrtp/test/srtp_driver.c: |
| (srtp_validate_encrypted_extensions_headers_gcm): |
| * Source/third_party/usrsctp/usrsctplib/.gitignore: Added. |
| * Source/third_party/usrsctp/usrsctplib/CMakeLists.txt: |
| * Source/third_party/usrsctp/usrsctplib/Makefile.am: |
| * Source/third_party/usrsctp/usrsctplib/README.md: |
| * Source/third_party/usrsctp/usrsctplib/configure.ac: |
| * Source/third_party/usrsctp/usrsctplib/programs/CMakeLists.txt: |
| * Source/third_party/usrsctp/usrsctplib/programs/Makefile.am: |
| * Source/third_party/usrsctp/usrsctplib/programs/client.c: |
| (main): |
| * Source/third_party/usrsctp/usrsctplib/programs/datachan_serv.c: |
| (main): |
| * Source/third_party/usrsctp/usrsctplib/programs/ekr_loop_offload.c: Added. |
| (handle_packets): |
| * Source/third_party/usrsctp/usrsctplib/programs/test_timer.c: Added. |
| (main): |
| * Source/third_party/usrsctp/usrsctplib/usrsctp.pc.in: Added. |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/CMakeLists.txt: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_asconf.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_asconf.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_auth.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_auth.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_bsd_addr.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_bsd_addr.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_cc_functions.c: |
| (sctp_cwnd_update_after_fr): |
| (sctp_hs_cwnd_update_after_fr): |
| (sctp_htcp_cwnd_update_after_fr): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_constants.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_crc32.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_crc32.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_header.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_indata.c: |
| (sctp_build_readq_entry): |
| (sctp_place_control_in_stream): |
| (sctp_abort_in_reasm): |
| (sctp_queue_data_to_stream): |
| (sctp_build_readq_entry_from_ctl): |
| (sctp_handle_old_unordered_data): |
| (sctp_inject_old_unordered_data): |
| (sctp_deliver_reasm_check): |
| (sctp_add_chk_to_control): |
| (sctp_queue_data_for_reasm): |
| (sctp_find_reasm_entry): |
| (sctp_process_a_data_chunk): |
| (sctp_sack_check): |
| (sctp_process_segment_range): |
| (sctp_check_for_revoked): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_indata.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_input.c: |
| (sctp_process_init): |
| (sctp_process_cookie_existing): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_input.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_output.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_output.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_pcb.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_pcb.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_peeloff.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_ss_functions.c: |
| (sctp_ss_rr_add): |
| (sctp_ss_fcfs_select): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_structs.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_sysctl.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_timer.c: |
| (sctp_recover_sent_list): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_uio.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_usrreq.c: |
| (sctp_init): |
| (sctp_pathmtu_adjustment): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_var.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctputil.c: |
| (sctp_log_strm_del): |
| (sctp_init_asoc): |
| (sctp_notify_send_failed): |
| (sctp_notify_send_failed2): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctputil.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet6/sctp6_usrreq.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet6/sctp6_var.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/user_mbuf.c: |
| (m_get): |
| (mbuf_initialize): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/user_mbuf.h: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/user_socket.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/usrsctp.h: |
| * WebKit/patch-usrsctp: Added. |
| |
| 2017-06-22 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Prevent capturing at unconventional resolutions when using the SW encoder on Mac |
| https://bugs.webkit.org/show_bug.cgi?id=172602 |
| <rdar://problem/32407693> |
| |
| Reviewed by Eric Carlson. |
| |
| Adding a parameter to disable hardware encoder. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/encoder.mm: |
| (webrtc::H264VideoToolboxEncoder::CreateCompressionSession): |
| |
| 2017-06-21 Youenn Fablet <youenn@apple.com> |
| |
| Update libyuv to 8cab2e31d76246263206318f3568d452e7f3ff3e |
| https://bugs.webkit.org/show_bug.cgi?id=173675 |
| |
| Reviewed by Sam Weinig. |
| |
| * Source/third_party/libyuv/.clang-format: Added. |
| * Source/third_party/libyuv/.gitignore: Added. |
| * Source/third_party/libyuv/Android.mk: |
| * Source/third_party/libyuv/BUILD.gn: |
| * Source/third_party/libyuv/CM_linux_packages.cmake: Added. |
| * Source/third_party/libyuv/CMakeLists.txt: |
| * Source/third_party/libyuv/DEPS: |
| * Source/third_party/libyuv/PRESUBMIT.py: |
| (_RunPythonTests): |
| (_RunPythonTests.join): |
| (_CommonChecks): |
| (CheckChangeOnUpload): |
| (CheckChangeOnCommit): |
| * Source/third_party/libyuv/README.chromium: |
| * Source/third_party/libyuv/build_overrides/build.gni: |
| * Source/third_party/libyuv/chromium/.gclient: Removed. |
| * Source/third_party/libyuv/chromium/README: Removed. |
| * Source/third_party/libyuv/cleanup_links.py: Added. |
| (WebRTCLinkSetup): |
| (WebRTCLinkSetup.__init__): |
| (WebRTCLinkSetup.CleanupLinks): |
| (_initialize_database): |
| (main): |
| * Source/third_party/libyuv/codereview.settings: |
| * Source/third_party/libyuv/docs/deprecated_builds.md: |
| * Source/third_party/libyuv/docs/getting_started.md: |
| * Source/third_party/libyuv/gyp_libyuv.py: |
| * Source/third_party/libyuv/include/libyuv/basic_types.h: |
| * Source/third_party/libyuv/include/libyuv/compare.h: |
| * Source/third_party/libyuv/include/libyuv/compare_row.h: |
| * Source/third_party/libyuv/include/libyuv/convert.h: |
| * Source/third_party/libyuv/include/libyuv/convert_argb.h: |
| * Source/third_party/libyuv/include/libyuv/convert_from.h: |
| * Source/third_party/libyuv/include/libyuv/convert_from_argb.h: |
| * Source/third_party/libyuv/include/libyuv/cpu_id.h: |
| * Source/third_party/libyuv/include/libyuv/macros_msa.h: |
| * Source/third_party/libyuv/include/libyuv/mjpeg_decoder.h: |
| * Source/third_party/libyuv/include/libyuv/planar_functions.h: |
| * Source/third_party/libyuv/include/libyuv/rotate.h: |
| * Source/third_party/libyuv/include/libyuv/rotate_argb.h: |
| * Source/third_party/libyuv/include/libyuv/rotate_row.h: |
| * Source/third_party/libyuv/include/libyuv/row.h: |
| * Source/third_party/libyuv/include/libyuv/scale.h: |
| * Source/third_party/libyuv/include/libyuv/scale_argb.h: |
| * Source/third_party/libyuv/include/libyuv/scale_row.h: |
| * Source/third_party/libyuv/include/libyuv/version.h: |
| * Source/third_party/libyuv/include/libyuv/video_common.h: |
| * Source/third_party/libyuv/infra/config/OWNERS: Added. |
| * Source/third_party/libyuv/infra/config/README.md: Added. |
| * Source/third_party/libyuv/infra/config/cq.cfg: Added. |
| * Source/third_party/libyuv/libyuv.gyp: |
| * Source/third_party/libyuv/libyuv.gypi: |
| * Source/third_party/libyuv/libyuv_test.gyp: |
| * Source/third_party/libyuv/linux.mk: |
| * Source/third_party/libyuv/pylintrc: Added. |
| * Source/third_party/libyuv/setup_links.py: Removed. |
| * Source/third_party/libyuv/source/compare.cc: |
| * Source/third_party/libyuv/source/compare_common.cc: |
| * Source/third_party/libyuv/source/compare_gcc.cc: |
| * Source/third_party/libyuv/source/compare_neon.cc: |
| * Source/third_party/libyuv/source/compare_neon64.cc: |
| * Source/third_party/libyuv/source/compare_win.cc: |
| * Source/third_party/libyuv/source/convert.cc: |
| * Source/third_party/libyuv/source/convert_argb.cc: |
| * Source/third_party/libyuv/source/convert_from.cc: |
| * Source/third_party/libyuv/source/convert_from_argb.cc: |
| * Source/third_party/libyuv/source/convert_jpeg.cc: |
| * Source/third_party/libyuv/source/convert_to_argb.cc: |
| * Source/third_party/libyuv/source/convert_to_i420.cc: |
| * Source/third_party/libyuv/source/cpu_id.cc: |
| * Source/third_party/libyuv/source/mjpeg_decoder.cc: |
| * Source/third_party/libyuv/source/mjpeg_validate.cc: |
| * Source/third_party/libyuv/source/planar_functions.cc: |
| * Source/third_party/libyuv/source/rotate.cc: |
| * Source/third_party/libyuv/source/rotate_any.cc: |
| * Source/third_party/libyuv/source/rotate_argb.cc: |
| * Source/third_party/libyuv/source/rotate_common.cc: |
| * Source/third_party/libyuv/source/rotate_dspr2.cc: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_mips.cc. |
| * Source/third_party/libyuv/source/rotate_gcc.cc: |
| * Source/third_party/libyuv/source/rotate_msa.cc: Added. |
| * Source/third_party/libyuv/source/rotate_neon.cc: |
| * Source/third_party/libyuv/source/rotate_neon64.cc: |
| * Source/third_party/libyuv/source/rotate_win.cc: |
| * Source/third_party/libyuv/source/row_any.cc: |
| * Source/third_party/libyuv/source/row_common.cc: |
| * Source/third_party/libyuv/source/row_dspr2.cc: Added. |
| * Source/third_party/libyuv/source/row_gcc.cc: |
| * Source/third_party/libyuv/source/row_mips.cc: Removed. |
| * Source/third_party/libyuv/source/row_msa.cc: |
| * Source/third_party/libyuv/source/row_neon.cc: |
| * Source/third_party/libyuv/source/row_neon64.cc: |
| * Source/third_party/libyuv/source/row_win.cc: |
| * Source/third_party/libyuv/source/scale.cc: |
| * Source/third_party/libyuv/source/scale_any.cc: |
| * Source/third_party/libyuv/source/scale_argb.cc: |
| * Source/third_party/libyuv/source/scale_common.cc: |
| * Source/third_party/libyuv/source/scale_dspr2.cc: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_mips.cc. |
| * Source/third_party/libyuv/source/scale_gcc.cc: |
| * Source/third_party/libyuv/source/scale_msa.cc: Added. |
| * Source/third_party/libyuv/source/scale_neon.cc: |
| * Source/third_party/libyuv/source/scale_neon64.cc: |
| * Source/third_party/libyuv/source/scale_win.cc: |
| * Source/third_party/libyuv/source/video_common.cc: |
| * Source/third_party/libyuv/sync_chromium.py: Removed. |
| * Source/third_party/libyuv/third_party/gflags/BUILD.gn: Removed. |
| * Source/third_party/libyuv/third_party/gflags/LICENSE: Removed. |
| * Source/third_party/libyuv/third_party/gflags/README.libyuv: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/posix/include/gflags/gflags.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/posix/include/gflags/gflags_completions.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/posix/include/gflags/gflags_declare.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/posix/include/gflags/gflags_gflags.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/posix/include/private/config.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/win/include/gflags/gflags.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/win/include/gflags/gflags_completions.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/win/include/gflags/gflags_declare.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/win/include/gflags/gflags_gflags.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gen/win/include/private/config.h: Removed. |
| * Source/third_party/libyuv/third_party/gflags/gflags.gyp: Removed. |
| * Source/third_party/libyuv/tools/gritsettings/README: Removed. |
| * Source/third_party/libyuv/tools/gritsettings/resource_ids: Removed. |
| * Source/third_party/libyuv/tools/valgrind-libyuv/tsan/OWNERS: Removed. |
| * Source/third_party/libyuv/tools/valgrind-libyuv/tsan/PRESUBMIT.py: Removed. |
| * Source/third_party/libyuv/tools/valgrind-libyuv/tsan/suppressions.txt: Removed. |
| * Source/third_party/libyuv/tools/valgrind-libyuv/tsan/suppressions_mac.txt: Removed. |
| * Source/third_party/libyuv/tools/valgrind-libyuv/tsan/suppressions_win32.txt: Removed. |
| * Source/third_party/libyuv/tools_libyuv/OWNERS: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/OWNERS. |
| * Source/third_party/libyuv/tools_libyuv/autoroller/roll_deps.py: Added. |
| (RollError): |
| (ParseDepsDict): |
| (ParseLocalDepsFile): |
| (ParseRemoteCrDepsFile): |
| (ParseCommitPosition): |
| (_RunCommand): |
| (_GetBranches): |
| (_ReadGitilesContent): |
| (ReadRemoteCrFile): |
| (ReadRemoteCrCommit): |
| (ReadUrlContent): |
| (GetMatchingDepsEntries): |
| (BuildDepsentryDict): |
| (BuildDepsentryDict.AddDepsEntries): |
| (CalculateChangedDeps): |
| (CalculateChangedClang): |
| (CalculateChangedClang.GetClangRev): |
| (GenerateCommitMessage): |
| (UpdateDepsFile): |
| (_IsTreeClean): |
| (_EnsureUpdatedMasterBranch): |
| (_CreateRollBranch): |
| (_RemovePreviousRollBranch): |
| (_LocalCommit): |
| (_UploadCL): |
| (_SendToCQ): |
| (main): |
| * Source/third_party/libyuv/tools_libyuv/autoroller/unittests/roll_deps_test.py: Added. |
| (TestError): |
| (FakeCmd): |
| (FakeCmd.__init__): |
| (FakeCmd.add_expectation): |
| (FakeCmd.__call__): |
| (TestRollChromiumRevision): |
| (TestRollChromiumRevision.setUp): |
| (TestRollChromiumRevision.tearDown): |
| (TestRollChromiumRevision.testUpdateDepsFile): |
| (TestRollChromiumRevision.testParseDepsDict): |
| (TestRollChromiumRevision.testParseDepsDict.assertVar): |
| (TestRollChromiumRevision.testGetMatchingDepsEntriesReturnsPathInSimpleCase): |
| (TestRollChromiumRevision.testGetMatchingDepsEntriesHandlesSimilarStartingPaths): |
| (TestRollChromiumRevision.testGetMatchingDepsEntriesHandlesTwoPathsWithIdenticalFirstParts): |
| (TestRollChromiumRevision.testCalculateChangedDeps): |
| (_SetupGitLsRemoteCall): |
| * Source/third_party/libyuv/tools_libyuv/autoroller/unittests/testdata/DEPS: Added. |
| * Source/third_party/libyuv/tools_libyuv/autoroller/unittests/testdata/DEPS.chromium.new: Added. |
| * Source/third_party/libyuv/tools_libyuv/autoroller/unittests/testdata/DEPS.chromium.old: Added. |
| * Source/third_party/libyuv/tools_libyuv/get_landmines.py: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/get_landmines.py. |
| * Source/third_party/libyuv/tools_libyuv/msan/OWNERS: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/msan/OWNERS. |
| * Source/third_party/libyuv/tools_libyuv/msan/blacklist.txt: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/msan/blacklist.txt. |
| * Source/third_party/libyuv/tools_libyuv/ubsan/OWNERS: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/ubsan/OWNERS. |
| * Source/third_party/libyuv/tools_libyuv/ubsan/blacklist.txt: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/ubsan/blacklist.txt. |
| * Source/third_party/libyuv/tools_libyuv/ubsan/vptr_blacklist.txt: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/ubsan/vptr_blacklist.txt. |
| * Source/third_party/libyuv/tools_libyuv/valgrind/libyuv_tests.bat: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/libyuv_tests.bat. |
| * Source/third_party/libyuv/tools_libyuv/valgrind/libyuv_tests.py: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/libyuv_tests.py. |
| (LibyuvTest._DefaultCommand): |
| * Source/third_party/libyuv/tools_libyuv/valgrind/libyuv_tests.sh: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/libyuv_tests.sh. |
| * Source/third_party/libyuv/tools_libyuv/valgrind/memcheck/OWNERS: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/memcheck/OWNERS. |
| * Source/third_party/libyuv/tools_libyuv/valgrind/memcheck/PRESUBMIT.py: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/memcheck/PRESUBMIT.py. |
| (CheckChange): |
| * Source/third_party/libyuv/tools_libyuv/valgrind/memcheck/suppressions.txt: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/memcheck/suppressions.txt. |
| * Source/third_party/libyuv/tools_libyuv/valgrind/memcheck/suppressions_mac.txt: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/memcheck/suppressions_mac.txt. |
| * Source/third_party/libyuv/tools_libyuv/valgrind/memcheck/suppressions_win32.txt: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools/valgrind-libyuv/memcheck/suppressions_win32.txt. |
| * Source/third_party/libyuv/unit_test/color_test.cc: |
| * Source/third_party/libyuv/unit_test/compare_test.cc: |
| * Source/third_party/libyuv/unit_test/convert_test.cc: |
| * Source/third_party/libyuv/unit_test/cpu_test.cc: |
| * Source/third_party/libyuv/unit_test/cpu_thread_test.cc: Added. |
| * Source/third_party/libyuv/unit_test/math_test.cc: |
| * Source/third_party/libyuv/unit_test/planar_test.cc: |
| * Source/third_party/libyuv/unit_test/rotate_argb_test.cc: |
| * Source/third_party/libyuv/unit_test/rotate_test.cc: |
| * Source/third_party/libyuv/unit_test/scale_argb_test.cc: |
| * Source/third_party/libyuv/unit_test/scale_test.cc: |
| * Source/third_party/libyuv/unit_test/unit_test.cc: |
| * Source/third_party/libyuv/unit_test/unit_test.h: |
| (SizeValid): |
| * Source/third_party/libyuv/unit_test/video_common_test.cc: |
| * Source/third_party/libyuv/util/compare.cc: |
| * Source/third_party/libyuv/util/cpuid.c: |
| (main): |
| * Source/third_party/libyuv/util/psnr.cc: |
| * Source/third_party/libyuv/util/psnr_main.cc: |
| * Source/third_party/libyuv/util/ssim.cc: |
| * Source/third_party/libyuv/util/ssim.h: |
| * Source/third_party/libyuv/util/yuvconvert.cc: Renamed from Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/convert.cc. |
| |
| 2017-06-21 Youenn Fablet <youenn@apple.com> |
| |
| Fix build after r218645 |
| https://bugs.webkit.org/show_bug.cgi?id=173668 |
| |
| Unreviewed. |
| |
| * Source/webrtc/base/sigslottester.h: Removing executable right. |
| * Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h: |
| (webrtc::TemporalLayersFactory::Create): Inline a default implementation. |
| * Source/webrtc/modules/video_processing/util/skin_detection.h: Removing executable right. |
| |
| 2017-06-21 Youenn Fablet <youenn@apple.com> |
| |
| Remove expat source code from Source/ThirdParty/libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=173656 |
| |
| Reviewed by Brent Fulgham. |
| |
| * Source/third_party/expat/BUILD.gn: Removed. |
| * Source/third_party/expat/OWNERS: Removed. |
| * Source/third_party/expat/README.chromium: Removed. |
| * Source/third_party/expat/files/COPYING: Removed. |
| * Source/third_party/expat/files/Changes: Removed. |
| * Source/third_party/expat/files/MANIFEST: Removed. |
| * Source/third_party/expat/files/README: Removed. |
| * Source/third_party/expat/files/lib/amigaconfig.h: Removed. |
| * Source/third_party/expat/files/lib/ascii.h: Removed. |
| * Source/third_party/expat/files/lib/asciitab.h: Removed. |
| * Source/third_party/expat/files/lib/expat.h: Removed. |
| * Source/third_party/expat/files/lib/expat_config.h: Removed. |
| * Source/third_party/expat/files/lib/expat_external.h: Removed. |
| * Source/third_party/expat/files/lib/iasciitab.h: Removed. |
| * Source/third_party/expat/files/lib/internal.h: Removed. |
| * Source/third_party/expat/files/lib/latin1tab.h: Removed. |
| * Source/third_party/expat/files/lib/libexpat.def: Removed. |
| * Source/third_party/expat/files/lib/libexpatw.def: Removed. |
| * Source/third_party/expat/files/lib/macconfig.h: Removed. |
| * Source/third_party/expat/files/lib/nametab.h: Removed. |
| * Source/third_party/expat/files/lib/utf8tab.h: Removed. |
| * Source/third_party/expat/files/lib/winconfig.h: Removed. |
| * Source/third_party/expat/files/lib/winconfig.h.original: Removed. |
| * Source/third_party/expat/files/lib/xmlparse.c: Removed. |
| * Source/third_party/expat/files/lib/xmlparse.c.original: Removed. |
| * Source/third_party/expat/files/lib/xmlrole.c: Removed. |
| * Source/third_party/expat/files/lib/xmlrole.h: Removed. |
| * Source/third_party/expat/files/lib/xmltok.c: Removed. |
| * Source/third_party/expat/files/lib/xmltok.h: Removed. |
| * Source/third_party/expat/files/lib/xmltok_impl.c: Removed. |
| * Source/third_party/expat/files/lib/xmltok_impl.c.original: Removed. |
| * Source/third_party/expat/files/lib/xmltok_impl.h: Removed. |
| * Source/third_party/expat/files/lib/xmltok_ns.c: Removed. |
| * Source/third_party/expat/fuzz/OWNERS: Removed. |
| * Source/third_party/expat/fuzz/expat_xml_parse_fuzzer.cc: Removed. |
| |
| 2017-06-21 Youenn Fablet <youenn@apple.com> |
| |
| Refresh libwebrtc code up to a87675d4a160e2c49c3e754cd9ca291d6c8f36ae |
| https://bugs.webkit.org/show_bug.cgi?id=173602 |
| |
| Reviewed by Eric Carlson. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source: Updated to a87675d4a160e2c49c3e754cd9ca291d6c8f36ae and reapplied WebKit specific changes. |
| * WebKit/patch-libwebrtc: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-06-19 Commit Queue <commit-queue@webkit.org> |
| |
| Unreviewed, rolling out r218505. |
| https://bugs.webkit.org/show_bug.cgi?id=173563 |
| |
| "It would break internal builds" (Requested by youenn on |
| #webkit). |
| |
| Reverted changeset: |
| |
| "[WebRTC] Prevent capturing at unconventional resolutions when |
| using the SW encoder on Mac" |
| https://bugs.webkit.org/show_bug.cgi?id=172602 |
| http://trac.webkit.org/changeset/218505 |
| |
| 2017-06-19 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Prevent capturing at unconventional resolutions when using the SW encoder on Mac |
| https://bugs.webkit.org/show_bug.cgi?id=172602 |
| <rdar://problem/32407693> |
| |
| Reviewed by Eric Carlson. |
| |
| Adding a parameter to disable hardware encoder. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_encoder.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_encoder.mm: |
| (webrtc::H264VideoToolboxEncoder::CreateCompressionSession): |
| |
| 2017-06-10 Dan Bernstein <mitz@apple.com> |
| |
| Reverted r218056 because it made the IDE reindex constantly. |
| |
| * Configurations/DebugRelease.xcconfig: |
| |
| 2017-06-10 Dan Bernstein <mitz@apple.com> |
| |
| [Xcode] With Xcode 9 developer beta, everything rebuilds when switching between command-line and IDE |
| https://bugs.webkit.org/show_bug.cgi?id=173223 |
| |
| Reviewed by Sam Weinig. |
| |
| The rebuilds were happening due to a difference in the compiler options that the IDE and |
| xcodebuild were specifying. Only the IDE was passing the -index-store-path option. To make |
| xcodebuild pass that option, too, set CLANG_INDEX_STORE_ENABLE to YES if it is unset, and |
| specify an appropriate path in CLANG_INDEX_STORE_PATH. |
| |
| * Configurations/DebugRelease.xcconfig: |
| |
| 2017-06-07 Youenn Fablet <youenn@apple.com> |
| |
| Add WebRTC stats logging |
| https://bugs.webkit.org/show_bug.cgi?id=173045 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/api/stats/rtcstats.h: Exporting RTCStats ToString. |
| |
| 2017-05-28 Dan Bernstein <mitz@apple.com> |
| |
| [Xcode] ALWAYS_SEARCH_USER_PATHS is set to YES |
| https://bugs.webkit.org/show_bug.cgi?id=172691 |
| |
| Reviewed by Tim Horton. |
| |
| * Configurations/Base.xcconfig: Set ALWAYS_SEARCH_USER_PATHS to NO. |
| |
| 2017-05-16 Youenn Fablet <youenn@apple.com> |
| |
| RealtimeOutgoingVideoSource should support sinkWants for rotation |
| https://bugs.webkit.org/show_bug.cgi?id=172123 |
| <rdar://problem/32200017> |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/api/video/i420_buffer.h: Exporting rotate routine. |
| |
| 2017-05-08 Youenn Fablet <youenn@apple.com> |
| |
| TURNS gathering is not working properly |
| https://bugs.webkit.org/show_bug.cgi?id=171747 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/base/openssladapter.cc: Adding support for SNI in case of TLS ice candidate gathering. |
| |
| 2017-04-29 Dan Bernstein <mitz@apple.com> |
| |
| [Xcode] libwebrtc SRCROOT includes examples |
| https://bugs.webkit.org/show_bug.cgi?id=171478 |
| |
| Reviewed by Tim Horton. |
| |
| * Configurations/Base.xcconfig: Exclude the Source/webrtc/examples subdirectory from |
| installsrc. Its contents are not used for building any of the targets in the project. |
| |
| 2017-04-19 Youenn Fablet <youenn@apple.com> |
| |
| [Mac] Allow customizing H264 encoder |
| https://bugs.webkit.org/show_bug.cgi?id=170829 |
| |
| Reviewed by Alex Christensen. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_encoder.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_encoder.mm: |
| (webrtc::H264VideoToolboxEncoder::ResetCompressionSession): |
| (webrtc::H264VideoToolboxEncoder::CreateCompressionSession): Default implementation, fixing memory leak for dictionary. |
| * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc: |
| |
| 2017-04-18 Youenn Fablet <youenn@apple.com> |
| |
| Add NDEBUG and CodeStripping to libwebrtc build system |
| https://bugs.webkit.org/show_bug.cgi?id=170954 |
| |
| Reviewed by Alex Christensen. |
| |
| This optimizes libwebrtc library size and efficiency. |
| This allows allocating libwebrtc objects in WebCore without issues. |
| |
| * Configurations/Base.xcconfig: |
| * Configurations/boringssl.xcconfig: |
| * Configurations/libsrtp.xcconfig: |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| * Configurations/opus.xcconfig: |
| * Configurations/usrsctp.xcconfig: |
| |
| 2017-04-17 Youenn Fablet <youenn@apple.com> |
| |
| Add an external libwebrtc encoder factory in WebCore |
| https://bugs.webkit.org/show_bug.cgi?id=170883 |
| |
| Reviewed by Alex Christensen. |
| |
| Exporting some symbols. |
| Allowing to customize the creation of the H264 encoder. |
| |
| * Source/webrtc/media/base/codec.h: |
| * Source/webrtc/media/engine/webrtcvideoencoderfactory.h |
| * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc: |
| * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.h: |
| * Source/webrtc/video_decoder.h |
| * Source/webrtc/video_encoder.h |
| |
| 2017-04-14 Mark Lam <mark.lam@apple.com> |
| |
| Update architectures in xcconfig files. |
| https://bugs.webkit.org/show_bug.cgi?id=170867 |
| <rdar://problem/31628104> |
| |
| Reviewed by Joseph Pecoraro. |
| |
| * Configurations/opus.xcconfig: |
| |
| 2017-04-12 Dan Bernstein <mitz@apple.com> |
| |
| [Mac] Future-proof .xcconfig files |
| https://bugs.webkit.org/show_bug.cgi?id=170802 |
| |
| Reviewed by Tim Horton. |
| |
| * Configurations/Base.xcconfig: |
| * Configurations/DebugRelease.xcconfig: |
| * Configurations/opus.xcconfig: |
| |
| 2017-04-07 Alex Christensen <achristensen@webkit.org> |
| |
| Enable SSE4 and NEON optimizations of libopus where available |
| https://bugs.webkit.org/show_bug.cgi?id=170592 |
| |
| Reviewed by Youenn Fablet. |
| |
| * Configurations/opus.xcconfig: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-04-06 Youenn Fablet <youenn@apple.com> |
| |
| WebRTC aborts when trying to sleep on a wrong thread |
| https://bugs.webkit.org/show_bug.cgi?id=170492 |
| <rdar://problem/31446377> |
| |
| Reviewed by Eric Carlson. |
| |
| Libwebrtc network thread is set up so that it does not accept blocking calls to other threads. |
| as per ChannelManager::Init() in channelmanager.cc. |
| But rtc::Thread::SleepMs expects to block it. |
| Marking thread as blockable before calling SleepMs and resetting the value if needed afterwards. |
| * Source/webrtc/media/sctp/sctptransport.cc: |
| |
| 2017-03-27 Alejandro G. Castro <alex@igalia.com> |
| |
| Fixes for libwebrtc logging after r214288 |
| https://bugs.webkit.org/show_bug.cgi?id=170116 |
| |
| Reviewed by Youenn Fablet. |
| |
| * Source/webrtc/base/logging.cc: Added the critical section |
| requirement and the call to the new getter for g_log_crit. |
| |
| 2017-03-27 Alex Christensen <achristensen@webkit.org> |
| |
| Build libwebrtc with even more warnings |
| https://bugs.webkit.org/show_bug.cgi?id=169997 |
| |
| Reviewed by Tim Horton. |
| |
| There are still OSAtomic* functions I don't want to worry about right now, |
| so I'm keeping a few -Wno-deprecated-declarations, but everything else can go. |
| |
| * Configurations/libsrtp.xcconfig: |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| |
| 2017-03-27 Youenn Fablet <youenn@apple.com> |
| |
| Add support for RTCRtpReceiver/RTCRtpSender getParameters |
| https://bugs.webkit.org/show_bug.cgi?id=170057 |
| |
| Reviewed by Alex Christensen. |
| |
| * Source/webrtc/api/mediatypes.h: |
| |
| 2017-03-22 Alex Christensen <achristensen@webkit.org> |
| |
| Fix warnings in libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=169973 |
| |
| Reviewed by Geoffrey Garen. |
| |
| * Configurations/boringssl.xcconfig: |
| * Configurations/libsrtp.xcconfig: |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| * Configurations/libyuv.xcconfig: |
| * Configurations/opus.xcconfig: |
| * Configurations/usrsctp.xcconfig: |
| Build with more warnings. |
| opus still needs some incompatible pointer warnings disabled because it converts |
| const opus_int16 * to const opus_val16 * and opus_int32 * to opus_val32 * |
| and that's ok because its a codec and that's what codecs do. |
| * Source/webrtc/base/logging.cc: |
| * Source/webrtc/base/logging.h: |
| * Source/webrtc/base/neverdestroyed.h: Added. |
| (webrtc::NeverDestroyed::NeverDestroyed): |
| (webrtc::NeverDestroyed::operator T&): |
| (webrtc::NeverDestroyed::get): |
| (webrtc::NeverDestroyed::operator&): |
| (webrtc::NeverDestroyed::asPtr): |
| Added webrtc::NeverDestroyed which may or may not be based on WTF::NeverDestroyed. |
| This allows us to avoid exit time destructors, which would slow down program termination for no reason. |
| * Source/webrtc/base/virtualsocketserver.cc: |
| * Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc: |
| Adopt NeverDestroyed in function scope so we don't have global constructors or destructors. |
| * Source/webrtc/modules/audio_processing/beamformer/array_util.h: |
| (webrtc::DegreesToRadians): |
| (webrtc::RadiansToDegrees): |
| Add constexpr so we can calculate values at compile time instead of launch time. |
| * Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc: |
| * Source/webrtc/system_wrappers/source/clock.cc: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Don't compile ssl_test.cc. We don't need it. |
| |
| 2017-03-10 Youenn Fablet <youenn@apple.com> |
| |
| Move libwebrtc backend to using tracks |
| https://bugs.webkit.org/show_bug.cgi?id=169472 |
| |
| Reviewed by Alex Christensen. |
| |
| * Source/webrtc/pc/rtcstatscollector.cc: Moving from using media stream to tracks. |
| |
| 2017-03-08 Youenn Fablet <youenn@apple.com> |
| |
| Use H264 hardware encoder for Mac libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=169383 |
| |
| Reviewed by Alex Christensen. |
| |
| Switching to H264 hardware encoder if available for Mac. |
| Adding logs in case hardware encoder cannot be used. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_encoder.mm: |
| (webrtc::H264VideoToolboxEncoder::ResetCompressionSession): |
| (webrtc::H264VideoToolboxEncoder::ConfigureCompressionSession): |
| |
| 2017-03-07 Youenn Fablet <youenn@apple.com> |
| |
| TurnPort::OnSocketConnect is crashing |
| https://bugs.webkit.org/show_bug.cgi?id=169284 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/p2p/base/turnport.cc: Fixing the assertion. |
| |
| 2017-03-06 Youenn Fablet <youenn@apple.com> |
| |
| Bring back WebKit specific changes to disable temporarily libwebrtc video adaptation |
| https://bugs.webkit.org/show_bug.cgi?id=169229 |
| |
| Reviewed by Alex Christensen. |
| |
| * Source/webrtc/modules/video_coding/video_sender.cc: disabling frame dropping. |
| * Source/webrtc/video/vie_encoder.cc: disabling resolution decrease based on CPU overuse. |
| |
| 2017-03-06 Alex Christensen <achristensen@webkit.org> |
| |
| Fix Production libwebrtc build after r213418 |
| https://bugs.webkit.org/show_bug.cgi?id=169217 |
| <rdar://problem/30876775> |
| |
| Reviewed by Tim Horton. |
| |
| * Source/webrtc/base/checks.h: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| MakeCheckOpString was a weak export, and it wasn't needed. |
| There is an internal build that checks for weak exports and fails if there is one. |
| Run the check-for-weak-vtables-and-externals script for libwebrtc.dylib like we do for the other frameworks. |
| |
| 2017-03-04 Dan Bernstein <mitz@apple.com> |
| |
| [Cocoa] libwebrtc.dylib’s current version is fixed at 1.0.0 |
| https://bugs.webkit.org/show_bug.cgi?id=169170 |
| |
| Reviewed by Alex Christensen. |
| |
| * Configurations/Version.xcconfig: Copied from Source/JavaScriptCore/Configurations/Version.xcconfig. |
| This defines DYLIB_CURRENT_VERSION. |
| * Configurations/libwebrtc.xcconfig: Include Version.xcconfig. |
| |
| 2017-03-04 Alex Christensen <achristensen@webkit.org> |
| |
| Cleanup after r213418 |
| https://bugs.webkit.org/show_bug.cgi?id=169165 |
| |
| Reviewed by Youenn Fablet. |
| |
| * WebKit/patch-libwebrtc: |
| I made another change after the last patch I uploaded to stop crashing. |
| This should be reflected in our patch. |
| |
| 2017-03-03 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Update libwebrtc source code |
| https://bugs.webkit.org/show_bug.cgi?id=168599 |
| |
| Reviewed by Alex Christensen. |
| |
| Very long list of file changes omitted. |
| |
| We updated to git commit 716e726ef0b322e8317b749613691da043bfc61c |
| of https://chromium.googlesource.com/external/webrtc and applied |
| the changes that are now in WebKit/patch-libwebrtc |
| |
| 2017-03-03 Alex Christensen <achristensen@webkit.org> |
| |
| Remove empty build directories. |
| |
| * build: Removed. |
| * build/Debug: Removed. |
| |
| 2017-03-01 Joseph Pecoraro <pecoraro@apple.com> |
| |
| [WebRTC] Install libwebrtc.dylib inside of WebCore.framework |
| https://bugs.webkit.org/show_bug.cgi?id=168859 |
| |
| Reviewed by Dan Bernstein. |
| |
| * Configurations/Base.xcconfig: |
| Define some general configuration variables. |
| |
| * Configurations/DebugRelease.xcconfig: |
| Define WK_RELOCATABLE_FRAMEWORKS for Debug/Release builds. |
| |
| * Configurations/libwebrtc.xcconfig: |
| Set INSTALL_PATH to be inside WebCore.framework's sub-Frameworks directory |
| unless WK_USE_OVERRIDE_FRAMEWORKS_DIR. Set the install name of the dylib to |
| be relative to WebCore / WebKit when frameworks are relocatable, such as |
| WK_USE_OVERRIDE_FRAMEWORKS_DIR or WK_RELOCATABLE_FRAMEWORKS. |
| |
| 2017-02-28 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] CPU Overuse libwebrtc detector is decreasing the quality of the video |
| https://bugs.webkit.org/show_bug.cgi?id=168990 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/video/vie_encoder.cc: Disabling temporarily overuse detector. |
| |
| 2017-02-28 Alex Christensen <achristensen@webkit.org> |
| |
| [WebRTC] Fix an internal production build |
| https://bugs.webkit.org/show_bug.cgi?id=168992 |
| |
| Reviewed by Youenn Fablet. |
| |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Link with Foundation and CoreFoundation frameworks. |
| |
| 2017-02-28 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] LibWebRTC frame dropper is not working consistently |
| https://bugs.webkit.org/show_bug.cgi?id=168973 |
| |
| Reviewed by Eric Carlson. |
| |
| * Source/webrtc/modules/video_coding/video_sender.cc: Disable temporarily the frame dropper as it is sometimes |
| dropping too many frames. |
| |
| 2017-02-27 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] RealtimOutgoingVideoSource should not need to do image conversion |
| https://bugs.webkit.org/show_bug.cgi?id=168802 |
| |
| Reviewed by Jon Lee. |
| |
| Exporting new symbols. |
| Including headers in the project file. |
| |
| * Source/webrtc/common_video/include/corevideo_frame_buffer.h: |
| * Source/webrtc/common_video/include/i420_buffer_pool.h: |
| * Source/webrtc/common_video/include/video_frame_buffer.h: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-24 Alex Christensen <achristensen@webkit.org> |
| |
| Remove unneeded protobuf tests directory. |
| |
| Rubber-stamped by Joe Pecoraro. |
| |
| This directory contained a swift file that was causing problems in an internal verification step. |
| |
| * Source/third_party/protobuf/objectivec/Tests: Removed. |
| (And everything in this subdirectory) |
| |
| 2017-02-22 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Disable libwebrtc stderr logging in release mode |
| https://bugs.webkit.org/show_bug.cgi?id=168734 |
| |
| Reviewed by Tim Horton. |
| |
| * Source/webrtc/base/logging.h: |
| |
| 2017-02-21 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC][Mac] Activate libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=167293 |
| <rdar://problem/30401864> |
| |
| Reviewed by Alex Christensen. |
| |
| Doing some clean-up. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/base/checks.h: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-21 Alex Christensen <achristensen@webkit.org> |
| |
| Don't build libwebrtc on 32-bit architectures |
| https://bugs.webkit.org/show_bug.cgi?id=168692 |
| |
| Reviewed by Dan Bernstein. |
| |
| * Configurations/Base.xcconfig: |
| |
| 2017-02-21 Youenn Fablet <youenn@apple.com> |
| |
| [Xcode] libwebrtc installhdrs doesn’t install any of the headers |
| https://bugs.webkit.org/show_bug.cgi?id=168634 |
| |
| Reviewed by Alex Christensen. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-21 Alex Christensen <achristensen@webkit.org> |
| |
| Unreviewed, rolling out r212699. |
| |
| Internal build not ready |
| |
| Reverted changeset: |
| |
| "[WebRTC][Mac] Activate libwebrtc" |
| https://bugs.webkit.org/show_bug.cgi?id=167293 |
| http://trac.webkit.org/changeset/212699 |
| |
| 2017-02-20 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC][Mac] Activate libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=167293 |
| <rdar://problem/30401864> |
| |
| Reviewed by Alex Christensen. |
| |
| Doing some clean-up. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/base/checks.h: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-17 Alex Christensen <achristensen@webkit.org> |
| |
| Fix iOS ASAN build after r212401 |
| https://bugs.webkit.org/show_bug.cgi?id=168398 |
| |
| * libwebrtc.xcodeproj/project.pbxproj: |
| libwebrtc.dylib needs some symbols from CFNetwork, |
| like CFNetworkCopySystemProxySettings |
| |
| 2017-02-16 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Fix some missing exports after r212401 |
| https://bugs.webkit.org/show_bug.cgi?id=168449 |
| |
| Reviewed by Alex Christensen. |
| |
| * Source/webrtc/api/jsep.h: |
| * Source/webrtc/base/checks.h: |
| |
| 2017-02-15 Alex Christensen <achristensen@webkit.org> |
| |
| Fix ASAN build after r212401 |
| https://bugs.webkit.org/show_bug.cgi?id=168398 |
| |
| * Source/webrtc/media/engine/webrtcvideocapturer.cc: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-15 Alex Christensen <achristensen@webkit.org> |
| |
| Make libwebrtc.dylib |
| https://bugs.webkit.org/show_bug.cgi?id=168335 |
| |
| Reviewed by Dan Bernstein. |
| |
| We were building libwebrtc as a static library, which would prevent us from weak linking with it. |
| We need to explicitly export what we use from WebCore or WebKit2, and RTCLogging.mm now needs to |
| be built on Mac, so we make it not automatically reference counted to make it work on 32-bit El Capitan. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/api/jsep.h: |
| * Source/webrtc/api/mediastream.h: |
| * Source/webrtc/api/notifier.h: |
| (webrtc::Notifier::Notifier): Deleted. |
| (webrtc::Notifier::RegisterObserver): Deleted. |
| (webrtc::Notifier::UnregisterObserver): Deleted. |
| (webrtc::Notifier::FireOnChanged): Deleted. |
| * Source/webrtc/api/peerconnectioninterface.h: |
| * Source/webrtc/base/asyncpacketsocket.h: |
| * Source/webrtc/base/asyncresolverinterface.h: |
| (rtc::AsyncResolverInterface::address): Deleted. |
| * Source/webrtc/base/copyonwritebuffer.h: |
| (rtc::CopyOnWriteBuffer::CopyOnWriteBuffer): Deleted. |
| (rtc::CopyOnWriteBuffer::data): Deleted. |
| (rtc::CopyOnWriteBuffer::cdata): Deleted. |
| (rtc::CopyOnWriteBuffer::size): Deleted. |
| (rtc::CopyOnWriteBuffer::capacity): Deleted. |
| (rtc::CopyOnWriteBuffer::operator=): Deleted. |
| (rtc::CopyOnWriteBuffer::operator!=): Deleted. |
| (rtc::CopyOnWriteBuffer::operator[]): Deleted. |
| (rtc::CopyOnWriteBuffer::SetData): Deleted. |
| (rtc::CopyOnWriteBuffer::AppendData): Deleted. |
| (rtc::CopyOnWriteBuffer::swap): Deleted. |
| (rtc::CopyOnWriteBuffer::IsConsistent): Deleted. |
| * Source/webrtc/base/event.h: |
| * Source/webrtc/base/export.h: Added. |
| * Source/webrtc/base/helpers.h: |
| * Source/webrtc/base/ipaddress.h: |
| (rtc::IPAddress::IPAddress): Deleted. |
| (rtc::IPAddress::~IPAddress): Deleted. |
| (rtc::IPAddress::operator=): Deleted. |
| (rtc::IPAddress::family): Deleted. |
| * Source/webrtc/base/location.h: |
| (rtc::Location::function_name): Deleted. |
| (rtc::Location::file_and_line): Deleted. |
| * Source/webrtc/base/messagehandler.h: |
| (rtc::MessageHandler::MessageHandler): Deleted. |
| * Source/webrtc/base/network.h: |
| (rtc::NetworkManagerBase::ipv6_enabled): Deleted. |
| (rtc::NetworkManagerBase::set_ipv6_enabled): Deleted. |
| (rtc::NetworkManagerBase::set_max_ipv6_networks): Deleted. |
| (rtc::NetworkManagerBase::max_ipv6_networks): Deleted. |
| (rtc::NetworkManagerBase::set_enumeration_permission): Deleted. |
| (rtc::BasicNetworkManager::started): Deleted. |
| (rtc::BasicNetworkManager::set_network_ignore_list): Deleted. |
| (rtc::BasicNetworkManager::set_ignore_non_default_routes): Deleted. |
| (rtc::Network::default_local_address_provider): Deleted. |
| (rtc::Network::set_default_local_address_provider): Deleted. |
| (rtc::Network::name): Deleted. |
| (rtc::Network::description): Deleted. |
| (rtc::Network::prefix): Deleted. |
| (rtc::Network::prefix_length): Deleted. |
| (rtc::Network::key): Deleted. |
| (rtc::Network::ip): Deleted. |
| (rtc::Network::AddIP): Deleted. |
| (rtc::Network::GetIPs): Deleted. |
| (rtc::Network::ClearIPs): Deleted. |
| (rtc::Network::scope_id): Deleted. |
| (rtc::Network::set_scope_id): Deleted. |
| (rtc::Network::ignored): Deleted. |
| (rtc::Network::set_ignored): Deleted. |
| (rtc::Network::type): Deleted. |
| (rtc::Network::set_type): Deleted. |
| (rtc::Network::GetCost): Deleted. |
| (rtc::Network::id): Deleted. |
| (rtc::Network::set_id): Deleted. |
| (rtc::Network::preference): Deleted. |
| (rtc::Network::set_preference): Deleted. |
| (rtc::Network::active): Deleted. |
| (rtc::Network::set_active): Deleted. |
| * Source/webrtc/base/proxyinfo.h: |
| * Source/webrtc/base/refcountedobject.h: |
| (rtc::RefCountedObject::RefCountedObject): Deleted. |
| (rtc::RefCountedObject::AddRef): Deleted. |
| (rtc::RefCountedObject::Release): Deleted. |
| (rtc::RefCountedObject::HasOneRef): Deleted. |
| (rtc::RefCountedObject::~RefCountedObject): Deleted. |
| * Source/webrtc/base/socketaddress.h: |
| (rtc::SocketAddress::hostname): Deleted. |
| (rtc::SocketAddress::family): Deleted. |
| (rtc::SocketAddress::scope_id): Deleted. |
| (rtc::SocketAddress::SetScopeID): Deleted. |
| (rtc::SocketAddress::operator !=): Deleted. |
| * Source/webrtc/base/thread.h: |
| * Source/webrtc/common_types.h: |
| * Source/webrtc/common_video/include/video_frame_buffer.h: |
| (webrtc::I420Buffer::Copy): Deleted. |
| (webrtc::I420Buffer::CropAndScaleFrom): Deleted. |
| (webrtc::I420Buffer::ScaleFrom): Deleted. |
| * Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h: |
| * Source/webrtc/p2p/base/basicpacketsocketfactory.h: |
| * Source/webrtc/p2p/client/basicportallocator.h: |
| (cricket::BasicPortAllocator::network_ignore_mask): Deleted. |
| (cricket::BasicPortAllocator::network_manager): Deleted. |
| (cricket::BasicPortAllocator::socket_factory): Deleted. |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm: |
| (RTCFileName): |
| * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.h: |
| * Source/webrtc/video_frame.h: |
| (webrtc::VideoFrame::timestamp_us): Deleted. |
| (webrtc::VideoFrame::set_timestamp_us): Deleted. |
| (webrtc::VideoFrame::set_timestamp): Deleted. |
| (webrtc::VideoFrame::timestamp): Deleted. |
| (webrtc::VideoFrame::transport_frame_id): Deleted. |
| (webrtc::VideoFrame::set_ntp_time_ms): Deleted. |
| (webrtc::VideoFrame::ntp_time_ms): Deleted. |
| (webrtc::VideoFrame::rotation): Deleted. |
| (webrtc::VideoFrame::set_rotation): Deleted. |
| (webrtc::VideoFrame::set_render_time_ms): Deleted. |
| (webrtc::VideoFrame::render_time_ms): Deleted. |
| (webrtc::VideoFrame::is_texture): Deleted. |
| * build: Added. |
| * build/Debug: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-15 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Remove libwebrtc ObjectiveC files that use UIKit |
| https://bugs.webkit.org/show_bug.cgi?id=168392 |
| |
| Reviewed by Alex Christensen. |
| |
| Removing default AudioDeviceModule as WebKit is providing its own. |
| Removing checks for active application in H264 codec as WebKit should be made responsible for that. |
| Removing no longer needed ObjectiveC files. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| * Source/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_decoder.cc: |
| * Source/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_encoder.mm: |
| (webrtc::H264VideoToolboxEncoder::Encode): |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-14 Ryan Haddad <ryanhaddad@apple.com> |
| |
| Unreviewed, rolling out r212326. |
| |
| This change broke certain build configurations. |
| |
| Reverted changeset: |
| |
| "Make libwebrtc.dylib" |
| https://bugs.webkit.org/show_bug.cgi?id=168335 |
| http://trac.webkit.org/changeset/212326 |
| |
| 2017-02-14 Alex Christensen <achristensen@webkit.org> |
| |
| Make libwebrtc.dylib |
| https://bugs.webkit.org/show_bug.cgi?id=168335 |
| |
| Reviewed by Dan Bernstein. |
| |
| We were building libwebrtc as a static library, which would prevent us from weak linking with it. |
| We need to explicitly export what we use from WebCore or WebKit2, and RTCLogging.mm now needs to |
| be built on Mac, so we make it not automatically reference counted to make it work on 32-bit El Capitan. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/api/jsep.h: |
| * Source/webrtc/api/mediastream.h: |
| * Source/webrtc/api/notifier.h: |
| (webrtc::Notifier::Notifier): Deleted. |
| (webrtc::Notifier::RegisterObserver): Deleted. |
| (webrtc::Notifier::UnregisterObserver): Deleted. |
| (webrtc::Notifier::FireOnChanged): Deleted. |
| * Source/webrtc/api/peerconnectioninterface.h: |
| * Source/webrtc/base/asyncpacketsocket.h: |
| * Source/webrtc/base/asyncresolverinterface.h: |
| (rtc::AsyncResolverInterface::address): Deleted. |
| * Source/webrtc/base/copyonwritebuffer.h: |
| (rtc::CopyOnWriteBuffer::CopyOnWriteBuffer): Deleted. |
| (rtc::CopyOnWriteBuffer::data): Deleted. |
| (rtc::CopyOnWriteBuffer::cdata): Deleted. |
| (rtc::CopyOnWriteBuffer::size): Deleted. |
| (rtc::CopyOnWriteBuffer::capacity): Deleted. |
| (rtc::CopyOnWriteBuffer::operator=): Deleted. |
| (rtc::CopyOnWriteBuffer::operator!=): Deleted. |
| (rtc::CopyOnWriteBuffer::operator[]): Deleted. |
| (rtc::CopyOnWriteBuffer::SetData): Deleted. |
| (rtc::CopyOnWriteBuffer::AppendData): Deleted. |
| (rtc::CopyOnWriteBuffer::swap): Deleted. |
| (rtc::CopyOnWriteBuffer::IsConsistent): Deleted. |
| * Source/webrtc/base/event.h: |
| * Source/webrtc/base/export.h: Added. |
| * Source/webrtc/base/helpers.h: |
| * Source/webrtc/base/ipaddress.h: |
| (rtc::IPAddress::IPAddress): Deleted. |
| (rtc::IPAddress::~IPAddress): Deleted. |
| (rtc::IPAddress::operator=): Deleted. |
| (rtc::IPAddress::family): Deleted. |
| * Source/webrtc/base/location.h: |
| (rtc::Location::function_name): Deleted. |
| (rtc::Location::file_and_line): Deleted. |
| * Source/webrtc/base/messagehandler.h: |
| (rtc::MessageHandler::MessageHandler): Deleted. |
| * Source/webrtc/base/network.h: |
| (rtc::NetworkManagerBase::ipv6_enabled): Deleted. |
| (rtc::NetworkManagerBase::set_ipv6_enabled): Deleted. |
| (rtc::NetworkManagerBase::set_max_ipv6_networks): Deleted. |
| (rtc::NetworkManagerBase::max_ipv6_networks): Deleted. |
| (rtc::NetworkManagerBase::set_enumeration_permission): Deleted. |
| (rtc::BasicNetworkManager::started): Deleted. |
| (rtc::BasicNetworkManager::set_network_ignore_list): Deleted. |
| (rtc::BasicNetworkManager::set_ignore_non_default_routes): Deleted. |
| (rtc::Network::default_local_address_provider): Deleted. |
| (rtc::Network::set_default_local_address_provider): Deleted. |
| (rtc::Network::name): Deleted. |
| (rtc::Network::description): Deleted. |
| (rtc::Network::prefix): Deleted. |
| (rtc::Network::prefix_length): Deleted. |
| (rtc::Network::key): Deleted. |
| (rtc::Network::ip): Deleted. |
| (rtc::Network::AddIP): Deleted. |
| (rtc::Network::GetIPs): Deleted. |
| (rtc::Network::ClearIPs): Deleted. |
| (rtc::Network::scope_id): Deleted. |
| (rtc::Network::set_scope_id): Deleted. |
| (rtc::Network::ignored): Deleted. |
| (rtc::Network::set_ignored): Deleted. |
| (rtc::Network::type): Deleted. |
| (rtc::Network::set_type): Deleted. |
| (rtc::Network::GetCost): Deleted. |
| (rtc::Network::id): Deleted. |
| (rtc::Network::set_id): Deleted. |
| (rtc::Network::preference): Deleted. |
| (rtc::Network::set_preference): Deleted. |
| (rtc::Network::active): Deleted. |
| (rtc::Network::set_active): Deleted. |
| * Source/webrtc/base/proxyinfo.h: |
| * Source/webrtc/base/refcountedobject.h: |
| (rtc::RefCountedObject::RefCountedObject): Deleted. |
| (rtc::RefCountedObject::AddRef): Deleted. |
| (rtc::RefCountedObject::Release): Deleted. |
| (rtc::RefCountedObject::HasOneRef): Deleted. |
| (rtc::RefCountedObject::~RefCountedObject): Deleted. |
| * Source/webrtc/base/socketaddress.h: |
| (rtc::SocketAddress::hostname): Deleted. |
| (rtc::SocketAddress::family): Deleted. |
| (rtc::SocketAddress::scope_id): Deleted. |
| (rtc::SocketAddress::SetScopeID): Deleted. |
| (rtc::SocketAddress::operator !=): Deleted. |
| * Source/webrtc/base/thread.h: |
| * Source/webrtc/common_types.h: |
| * Source/webrtc/common_video/include/video_frame_buffer.h: |
| (webrtc::I420Buffer::Copy): Deleted. |
| (webrtc::I420Buffer::CropAndScaleFrom): Deleted. |
| (webrtc::I420Buffer::ScaleFrom): Deleted. |
| * Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h: |
| * Source/webrtc/p2p/base/basicpacketsocketfactory.h: |
| * Source/webrtc/p2p/client/basicportallocator.h: |
| (cricket::BasicPortAllocator::network_ignore_mask): Deleted. |
| (cricket::BasicPortAllocator::network_manager): Deleted. |
| (cricket::BasicPortAllocator::socket_factory): Deleted. |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm: |
| (RTCFileName): |
| * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.h: |
| * Source/webrtc/video_frame.h: |
| (webrtc::VideoFrame::timestamp_us): Deleted. |
| (webrtc::VideoFrame::set_timestamp_us): Deleted. |
| (webrtc::VideoFrame::set_timestamp): Deleted. |
| (webrtc::VideoFrame::timestamp): Deleted. |
| (webrtc::VideoFrame::transport_frame_id): Deleted. |
| (webrtc::VideoFrame::set_ntp_time_ms): Deleted. |
| (webrtc::VideoFrame::ntp_time_ms): Deleted. |
| (webrtc::VideoFrame::rotation): Deleted. |
| (webrtc::VideoFrame::set_rotation): Deleted. |
| (webrtc::VideoFrame::set_render_time_ms): Deleted. |
| (webrtc::VideoFrame::render_time_ms): Deleted. |
| (webrtc::VideoFrame::is_texture): Deleted. |
| * build: Added. |
| * build/Debug: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-14 Alex Christensen <achristensen@webkit.org> |
| |
| Remove android-specific files from ThirdParty/libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=168272 |
| |
| Reviewed by Brady Eidson. |
| |
| * Source/third_party/boringssl/src/third_party/android-cmake: Removed. |
| * Source/third_party/boringssl/src/third_party/android-cmake/AndroidNdkGdb.cmake: Removed. |
| * Source/third_party/boringssl/src/third_party/android-cmake/AndroidNdkModules.cmake: Removed. |
| * Source/third_party/boringssl/src/third_party/android-cmake/LICENSE: Removed. |
| * Source/third_party/boringssl/src/third_party/android-cmake/METADATA: Removed. |
| * Source/third_party/boringssl/src/third_party/android-cmake/README.md: Removed. |
| * Source/third_party/boringssl/src/third_party/android-cmake/android.toolchain.cmake: Removed. |
| * Source/third_party/boringssl/src/third_party/android-cmake/ndk_links.md: Removed. |
| * Source/third_party/boringssl/src/util/run_android_tests.go: Removed. |
| * Source/third_party/libyuv/util/android: Removed. |
| * Source/third_party/libyuv/util/android/test_runner.py: Removed. |
| * Source/webrtc/androidjunit: Removed. |
| * Source/webrtc/androidjunit/OWNERS: Removed. |
| * Source/webrtc/androidjunit/src: Removed. |
| * Source/webrtc/androidjunit/src/org: Removed. |
| * Source/webrtc/androidjunit/src/org/webrtc: Removed. |
| * Source/webrtc/androidjunit/src/org/webrtc/CameraEnumerationTest.java: Removed. |
| * Source/webrtc/api/android: Removed. |
| * Source/webrtc/api/android/PRESUBMIT.py: Removed. |
| * Source/webrtc/api/android/README: Removed. |
| * Source/webrtc/api/android/java: Removed. |
| * Source/webrtc/api/android/java/src: Removed. |
| * Source/webrtc/api/android/java/src/org: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/AudioSource.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/AudioTrack.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/CallSessionFileRotatingLogSink.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/Camera1Capturer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/Camera1Enumerator.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/Camera1Session.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/Camera2Capturer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/Camera2Enumerator.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/Camera2Session.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/CameraCapturer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/CameraEnumerationAndroid.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/CameraEnumerator.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/CameraSession.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/CameraVideoCapturer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/DataChannel.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/EglBase.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/EglBase10.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/EglBase14.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/EglRenderer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/FileVideoCapturer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/GlRectDrawer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/GlShader.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/GlTextureFrameBuffer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/GlUtil.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/IceCandidate.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/MediaCodecVideoDecoder.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/MediaCodecVideoEncoder.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/MediaConstraints.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/MediaSource.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/MediaStream.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/MediaStreamTrack.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/Metrics.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/NetworkMonitor.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/NetworkMonitorAutoDetect.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/OWNERS: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/PeerConnection.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/PeerConnectionFactory.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/RendererCommon.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/RtpParameters.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/RtpReceiver.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/RtpSender.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/ScreenCapturerAndroid.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/SdpObserver.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/SessionDescription.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/StatsObserver.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/StatsReport.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/SurfaceTextureHelper.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/SurfaceViewRenderer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/VideoCapturer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/VideoCapturerAndroid.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/VideoFileRenderer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/VideoRenderer.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/VideoRendererGui.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/VideoSource.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/VideoTrack.java: Removed. |
| * Source/webrtc/api/android/java/src/org/webrtc/YuvConverter.java: Removed. |
| * Source/webrtc/api/android/jni: Removed. |
| * Source/webrtc/api/android/jni/OWNERS: Removed. |
| * Source/webrtc/api/android/jni/androidmediacodeccommon.h: Removed. |
| * Source/webrtc/api/android/jni/androidmediadecoder_jni.cc: Removed. |
| * Source/webrtc/api/android/jni/androidmediadecoder_jni.h: Removed. |
| * Source/webrtc/api/android/jni/androidmediaencoder_jni.cc: Removed. |
| * Source/webrtc/api/android/jni/androidmediaencoder_jni.h: Removed. |
| * Source/webrtc/api/android/jni/androidmetrics_jni.cc: Removed. |
| * Source/webrtc/api/android/jni/androidnetworkmonitor_jni.cc: Removed. |
| * Source/webrtc/api/android/jni/androidnetworkmonitor_jni.h: Removed. |
| * Source/webrtc/api/android/jni/androidvideotracksource_jni.cc: Removed. |
| * Source/webrtc/api/android/jni/classreferenceholder.cc: Removed. |
| * Source/webrtc/api/android/jni/classreferenceholder.h: Removed. |
| * Source/webrtc/api/android/jni/jni_helpers.cc: Removed. |
| * Source/webrtc/api/android/jni/jni_helpers.h: Removed. |
| * Source/webrtc/api/android/jni/jni_onload.cc: Removed. |
| * Source/webrtc/api/android/jni/native_handle_impl.cc: Removed. |
| * Source/webrtc/api/android/jni/native_handle_impl.h: Removed. |
| * Source/webrtc/api/android/jni/peerconnection_jni.cc: Removed. |
| * Source/webrtc/api/android/jni/surfacetexturehelper_jni.cc: Removed. |
| * Source/webrtc/api/android/jni/surfacetexturehelper_jni.h: Removed. |
| * Source/webrtc/api/androidtests: Removed. |
| * Source/webrtc/api/androidtests/AndroidManifest.xml: Removed. |
| * Source/webrtc/api/androidtests/OWNERS: Removed. |
| * Source/webrtc/api/androidtests/ant.properties: Removed. |
| * Source/webrtc/api/androidtests/build.xml: Removed. |
| * Source/webrtc/api/androidtests/project.properties: Removed. |
| * Source/webrtc/api/androidtests/res: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-hdpi: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-hdpi/ic_launcher.png: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-ldpi: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-ldpi/ic_launcher.png: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-mdpi: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-mdpi/ic_launcher.png: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-xhdpi: Removed. |
| * Source/webrtc/api/androidtests/res/drawable-xhdpi/ic_launcher.png: Removed. |
| * Source/webrtc/api/androidtests/res/values: Removed. |
| * Source/webrtc/api/androidtests/res/values/strings.xml: Removed. |
| * Source/webrtc/api/androidtests/src: Removed. |
| * Source/webrtc/api/androidtests/src/org: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/Camera2CapturerTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/EglRendererTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/GlRectDrawerTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/NetworkMonitorTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/RendererCommonTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java: Removed. |
| * Source/webrtc/api/androidtests/src/org/webrtc/WebRtcJniBootTest.java: Removed. |
| * Source/webrtc/api/androidvideotracksource.cc: Removed. |
| * Source/webrtc/api/androidvideotracksource.h: Removed. |
| * Source/webrtc/api/test/androidtestinitializer.cc: Removed. |
| * Source/webrtc/api/test/androidtestinitializer.h: Removed. |
| * Source/webrtc/base/ifaddrs-android.cc: Removed. |
| * Source/webrtc/base/ifaddrs-android.h: Removed. |
| * Source/webrtc/build/android: Removed. |
| * Source/webrtc/build/android/AndroidManifest.xml: Removed. |
| * Source/webrtc/build/android/suppressions.xml: Removed. |
| * Source/webrtc/build/android/test_runner.py: Removed. |
| * Source/webrtc/examples/androidapp: Removed. |
| * Source/webrtc/examples/androidapp/AndroidManifest.xml: Removed. |
| * Source/webrtc/examples/androidapp/OWNERS: Removed. |
| * Source/webrtc/examples/androidapp/README: Removed. |
| * Source/webrtc/examples/androidapp/ant.properties: Removed. |
| * Source/webrtc/examples/androidapp/build.xml: Removed. |
| * Source/webrtc/examples/androidapp/project.properties: Removed. |
| * Source/webrtc/examples/androidapp/res: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-hdpi: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-hdpi/disconnect.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-hdpi/ic_action_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-hdpi/ic_action_return_from_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-hdpi/ic_launcher.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-hdpi/ic_loopback_call.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-ldpi: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-ldpi/disconnect.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-ldpi/ic_action_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-ldpi/ic_action_return_from_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-ldpi/ic_launcher.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-ldpi/ic_loopback_call.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-mdpi: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-mdpi/disconnect.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-mdpi/ic_action_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-mdpi/ic_action_return_from_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-mdpi/ic_launcher.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-mdpi/ic_loopback_call.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-xhdpi: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-xhdpi/disconnect.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-xhdpi/ic_action_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-xhdpi/ic_action_return_from_full_screen.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-xhdpi/ic_launcher.png: Removed. |
| * Source/webrtc/examples/androidapp/res/drawable-xhdpi/ic_loopback_call.png: Removed. |
| * Source/webrtc/examples/androidapp/res/layout: Removed. |
| * Source/webrtc/examples/androidapp/res/layout/activity_call.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/layout/activity_connect.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/layout/fragment_call.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/layout/fragment_hud.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/menu: Removed. |
| * Source/webrtc/examples/androidapp/res/menu/connect_menu.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/values: Removed. |
| * Source/webrtc/examples/androidapp/res/values-v17: Removed. |
| * Source/webrtc/examples/androidapp/res/values-v17/styles.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/values-v21: Removed. |
| * Source/webrtc/examples/androidapp/res/values-v21/styles.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/values/arrays.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/values/strings.xml: Removed. |
| * Source/webrtc/examples/androidapp/res/xml: Removed. |
| * Source/webrtc/examples/androidapp/res/xml/preferences.xml: Removed. |
| * Source/webrtc/examples/androidapp/src: Removed. |
| * Source/webrtc/examples/androidapp/src/org: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCClient.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallFragment.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CaptureQualityController.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/HudFragment.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PercentFrameLayout.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/SettingsFragment.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/UnhandledExceptionHandler.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/util: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/util/AppRTCUtils.java: Removed. |
| * Source/webrtc/examples/androidapp/src/org/appspot/apprtc/util/AsyncHttpURLConnection.java: Removed. |
| * Source/webrtc/examples/androidapp/start_loopback_stubbed_camera_saved_video_out.py: Removed. |
| * Source/webrtc/examples/androidapp/third_party: Removed. |
| * Source/webrtc/examples/androidapp/third_party/autobanh: Removed. |
| * Source/webrtc/examples/androidapp/third_party/autobanh/BUILD.gn: Removed. |
| * Source/webrtc/examples/androidapp/third_party/autobanh/LICENSE: Removed. |
| * Source/webrtc/examples/androidapp/third_party/autobanh/LICENSE.md: Removed. |
| * Source/webrtc/examples/androidapp/third_party/autobanh/NOTICE: Removed. |
| * Source/webrtc/examples/androidapp/third_party/autobanh/lib: Removed. |
| * Source/webrtc/examples/androidapp/third_party/autobanh/lib/autobanh.jar: Removed. |
| * Source/webrtc/examples/androidjunit: Removed. |
| * Source/webrtc/examples/androidjunit/README: Removed. |
| * Source/webrtc/examples/androidjunit/src: Removed. |
| * Source/webrtc/examples/androidjunit/src/org: Removed. |
| * Source/webrtc/examples/androidjunit/src/org/appspot: Removed. |
| * Source/webrtc/examples/androidjunit/src/org/appspot/apprtc: Removed. |
| * Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java: Removed. |
| * Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java: Removed. |
| * Source/webrtc/examples/androidtests: Removed. |
| * Source/webrtc/examples/androidtests/AndroidManifest.xml: Removed. |
| * Source/webrtc/examples/androidtests/README: Removed. |
| * Source/webrtc/examples/androidtests/ant.properties: Removed. |
| * Source/webrtc/examples/androidtests/build.xml: Removed. |
| * Source/webrtc/examples/androidtests/project.properties: Removed. |
| * Source/webrtc/examples/androidtests/src: Removed. |
| * Source/webrtc/examples/androidtests/src/org: Removed. |
| * Source/webrtc/examples/androidtests/src/org/appspot: Removed. |
| * Source/webrtc/examples/androidtests/src/org/appspot/apprtc: Removed. |
| * Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test: Removed. |
| * Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/FileVideoCapturerTest.java: Removed. |
| * Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java: Removed. |
| * Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/VideoFileRendererTest.java: Removed. |
| * Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/capturetestvideo.y4m: Removed. |
| * Source/webrtc/modules/audio_device/android: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_common.h: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_device_template.h: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_device_unittest.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_manager.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_manager.h: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_manager_unittest.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_record_jni.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_record_jni.h: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_track_jni.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/audio_track_jni.h: Removed. |
| * Source/webrtc/modules/audio_device/android/build_info.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/build_info.h: Removed. |
| * Source/webrtc/modules/audio_device/android/ensure_initialized.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/ensure_initialized.h: Removed. |
| * Source/webrtc/modules/audio_device/android/java: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/BuildInfo.java: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java: Removed. |
| * Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java: Removed. |
| * Source/webrtc/modules/audio_device/android/opensles_common.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/opensles_common.h: Removed. |
| * Source/webrtc/modules/audio_device/android/opensles_player.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/opensles_player.h: Removed. |
| * Source/webrtc/modules/audio_device/android/opensles_recorder.cc: Removed. |
| * Source/webrtc/modules/audio_device/android/opensles_recorder.h: Removed. |
| * Source/webrtc/modules/audio_processing/test/android: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest/AndroidManifest.xml: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest/default.properties: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest/jni: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest/jni/main.c: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest/res: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest/res/values: Removed. |
| * Source/webrtc/modules/audio_processing/test/android/apmtest/res/values/strings.xml: Removed. |
| * Source/webrtc/modules/utility/include/helpers_android.h: Removed. |
| * Source/webrtc/modules/utility/include/jvm_android.h: Removed. |
| * Source/webrtc/modules/utility/source/helpers_android.cc: Removed. |
| * Source/webrtc/modules/utility/source/jvm_android.cc: Removed. |
| * Source/webrtc/system_wrappers/source/cpu_features_android.c: Removed. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-10 Alex Christensen <achristensen@webkit.org> |
| |
| Fix iOS libwebrtc build after r212127 |
| https://bugs.webkit.org/show_bug.cgi?id=168134 |
| |
| * Configurations/libwebrtc.xcconfig: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| I got a little carried away removing ObjC sources. |
| We still need RTCLogging.mm and RTCUIApplication.mm on iOS. |
| Also sorted the project file. |
| |
| 2017-02-10 Alex Christensen <achristensen@webkit.org> |
| |
| Fix iOS libwebrtc build after r212127 |
| https://bugs.webkit.org/show_bug.cgi?id=168134 |
| |
| * libwebrtc.xcodeproj/project.pbxproj: |
| I got a little carried away removing -fobjc-arc. These files need it. |
| It was originally added in r211902 and these files are in the |
| EXCLUDED_SOURCE_FILE_NAMES[sdk=macosx*] list in libwebrtc.xcconfig |
| so adding this flag won't break the 32-bit El Capitan build. |
| |
| 2017-02-10 Alex Christensen <achristensen@webkit.org> |
| |
| Remove unnecessary automatic reference counting in libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=168134 |
| |
| Reviewed by Youenn Fablet. |
| |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-10 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] Activate libwebrtc G711/G722 audio codecs |
| https://bugs.webkit.org/show_bug.cgi?id=168123 |
| |
| Reviewed by Alex Christensen. |
| |
| Adding G711/G722 missing codec files. |
| Activating use of these in the build system. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Source/webrtc/modules/audio_coding/codecs/g711/g711.c: Added. |
| (ulaw_to_alaw): |
| * Source/webrtc/modules/audio_coding/codecs/g711/g711.h: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h: Added. |
| * Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-10 Alex Christensen <achristensen@webkit.org> |
| |
| Fix ASAN build. |
| |
| * Source/webrtc/base/sanitizer.h: |
| SANITIZER_UNUSED3 wasn't defined if we are using address_sanitizer but not memory_sanitizer. |
| |
| 2017-02-09 Alex Christensen <achristensen@webkit.org> |
| |
| Fix El Capitan build. |
| |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Remove more SSE4 code. |
| |
| 2017-02-09 Alex Christensen <achristensen@webkit.org> |
| |
| Fix El Capitan build. |
| |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Remove more SSE4 code. |
| |
| 2017-02-09 Alex Christensen <achristensen@webkit.org> |
| |
| Fix iOS and El Capitan builds of libwebrtc. |
| |
| * Configurations/libwebrtc.xcconfig: |
| Skip building audio_mixer_manager_mac.cc on iOS. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| El Capitan doesn't like the SSE4 optimizations in opus. |
| Just don't include them for now. |
| |
| 2017-02-09 Alex Christensen <achristensen@webkit.org> |
| |
| Fix warnings in libwebrtc build |
| https://bugs.webkit.org/show_bug.cgi?id=168088 |
| |
| Reviewed by Youenn Fablet. |
| |
| * Source/third_party/opus/src/src/opus_decoder.c: |
| Silence a warning. Debug builds of opus can be slow. No big deal. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| More sdk files need ARC. |
| |
| 2017-02-09 Alex Christensen <achristensen@webkit.org> |
| |
| Fix iOS libwebrtc build after r211960 |
| https://bugs.webkit.org/show_bug.cgi?id=168038 |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.h: |
| GlContextType declaration needs to be platform specific here like it is in RTCOpenGLDefines.h |
| |
| 2017-02-09 Alex Christensen <achristensen@webkit.org> |
| |
| Fix i386 libwebrtc build |
| https://bugs.webkit.org/show_bug.cgi?id=168038 |
| |
| Reviewed by Geoffrey Garen. |
| |
| Unfortunately, 32-bit ObjC can't use all the coolest new features of ObjC. |
| Fortunately, we can move things around a bit to become valid old ObjC. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCAVFoundationVideoSource.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCAudioSource.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCFileLogger.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCI420Shader.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCMediaSource+Private.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCMediaSource.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCMediaStream.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCPeerConnectionFactory.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCRtpReceiver.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCRtpSender.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCShader.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCVideoFrame.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter.h: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCVideoRendererAdapter.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCVideoSource.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/RTCVideoTrack.mm: |
| * Source/webrtc/sdk/objc/Framework/Classes/avfoundationvideocapturer.mm: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h: |
| * Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h: |
| Make code compile for i386. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Added missing headers so Xcode can search them. |
| |
| 2017-02-09 Alex Christensen <achristensen@webkit.org> |
| |
| Remove svn:executable property from headers. |
| |
| * Source/webrtc/base/sigslottester.h: Removed property svn:executable. |
| * Source/webrtc/modules/video_processing/util/skin_detection.h: Removed property svn:executable. |
| |
| 2017-02-08 Alex Christensen <achristensen@webkit.org> |
| |
| Fix libwebrtc build. |
| https://bugs.webkit.org/show_bug.cgi?id=168017 |
| |
| * Configurations/libwebrtc.xcconfig: |
| Trying to compile audio_device_not_implemented_ios.mm on Mac doesn't work. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Add some neon files. They are nicely protected by macros at the top, so their contents are only compiled if necessary. |
| |
| 2017-02-08 Alex Christensen <achristensen@webkit.org> |
| |
| Fix libwebrtc build on iOS simulator |
| https://bugs.webkit.org/show_bug.cgi?id=168017 |
| |
| Reviewed by Tim Horton. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| Use $(inherited) |
| * Source/webrtc/modules/audio_device/ios/audio_device_ios.h: |
| * Source/webrtc/modules/audio_device/ios/audio_device_ios.mm: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Configuration.mm: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSession.h: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSession.mm: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.h: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionConfiguration.m: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h: |
| * Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.mm: |
| Renamed RTCAudioSession* to WebRTCAudioSession* so that all ObjC classes in WebCore start with Web prefix. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Add necessary files. Some iOS-specific files need ARC, |
| and this matches the Build.gn in Source/webrtc/modules/audio_device |
| |
| 2017-02-08 Alex Christensen <achristensen@webkit.org> |
| |
| Fix iOS libwebrtc build |
| https://bugs.webkit.org/show_bug.cgi?id=168004 |
| |
| Reviewed by Youenn Fablet. |
| |
| We might still need to add some neon files. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| * Configurations/opus.xcconfig: |
| Don't build sse-specific files for iOS. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| Don't include the sse4 optimization for now. |
| We can add the optimization for CPUs that support it later. |
| |
| 2017-02-08 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] Fix libwebrtc build system |
| https://bugs.webkit.org/show_bug.cgi?id=167978 |
| |
| Reviewed by Alex Christensen. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| * Configurations/usrsctp.xcconfig: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-07 Youenn Fablet <youenn@apple.com> |
| |
| Fix libwebrtcpcrtc target include path |
| https://bugs.webkit.org/show_bug.cgi?id=167971 |
| |
| Reviewed by Alex Christensen. |
| |
| * Configurations/libwebrtcpcrtc.xcconfig: |
| |
| 2017-02-07 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] usrsctp (libwebrtc third party library) is not compiling |
| https://bugs.webkit.org/show_bug.cgi?id=167969 |
| |
| Reviewed by Alex Christensen. |
| |
| Also removing .gitignore files in libwebrtc directory. |
| |
| * Source/.gitignore: Removed. |
| * Source/third_party/boringssl/src/.gitignore: Removed. |
| * Source/third_party/gflags/src/.gitignore: Removed. |
| * Source/third_party/jsoncpp/source/.gitignore: Removed. |
| * Source/third_party/libyuv/.gitignore: Removed. |
| * Source/third_party/protobuf/.gitignore: Removed. |
| * Source/third_party/protobuf/csharp/.gitignore: Removed. |
| * Source/third_party/protobuf/ruby/.gitignore: Removed. |
| * Source/third_party/usrsctp/usrsctplib/.gitignore: Removed. |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_input.c: |
| (sctp_process_cookie_existing): |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_output.c: |
| * Source/tools/.gitignore: Removed. |
| * Source/webrtc/.gitignore: Removed. |
| |
| 2017-02-07 Alex Christensen <achristensen@webkit.org> |
| |
| Move webrtc/pc to own Xcode target |
| https://bugs.webkit.org/show_bug.cgi?id=167970 |
| |
| Reviewed by Youenn Fablet. |
| |
| It needs to include different directories than the rest of libwebrtc. |
| Also moved some target names so liblibsrtp.a is changed to libsrtp.a, etc. |
| |
| * Configurations/libwebrtcpcrtc.xcconfig: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-07 Alex Christensen <achristensen@webkit.org> |
| |
| [libwebrtc] Move libsrtp and libyuv to own Xcode targets |
| https://bugs.webkit.org/show_bug.cgi?id=167966 |
| |
| Reviewed by Youenn Fablet. |
| |
| * Configurations/libsrtp.xcconfig: Added. |
| * Configurations/libyuv.xcconfig: Added. |
| * Configurations/usrsctp.xcconfig: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-07 Alex Christensen <achristensen@webkit.org> |
| |
| Fix libwebrtc build after r211817 |
| https://bugs.webkit.org/show_bug.cgi?id=167944 |
| |
| * Configurations/usrsctp.xcconfig: |
| Disable more warnings. |
| |
| 2017-02-07 Alex Christensen <achristensen@webkit.org> |
| |
| build usrsctp with Xcode |
| https://bugs.webkit.org/show_bug.cgi?id=167944 |
| |
| Reviewed by Youenn Fablet. |
| |
| * Configurations/usrsctp.xcconfig: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/netinet/sctp_pcb.c: |
| * Source/third_party/usrsctp/usrsctplib/usrsctplib/user_atomic.h: |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-06 Alex Christensen <achristensen@webkit.org> |
| |
| [WebRTC] Remove unneeded build directory accidentally checked in with libwebrtc source. |
| |
| Reviewed by Youenn Fablet. |
| |
| * third_party/usrsctp/build: Removed. |
| |
| 2017-02-03 Alex Christensen <achristensen@webkit.org> |
| |
| [WebRTC] Add more files to libwebrtc build |
| https://bugs.webkit.org/show_bug.cgi?id=167824 |
| |
| Reviewed by Youenn Fablet. |
| |
| * Configurations/libwebrtc.xcconfig: |
| * Configurations/usrsctp.xcconfig: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-02-02 Alex Christensen <achristensen@webkit.org> |
| |
| Build libwebrtc and dependencies with Xcode |
| https://bugs.webkit.org/show_bug.cgi?id=167758 |
| |
| Reviewed by Dean Jackson. |
| |
| * Configurations: Added. |
| * Configurations/Base.xcconfig: Added. |
| * Configurations/DebugRelease.xcconfig: Added. |
| * Configurations/boringssl.xcconfig: Added. |
| * Configurations/libwebrtc.xcconfig: Added. |
| * Configurations/opus.xcconfig: Added. |
| * libwebrtc.xcodeproj/project.pbxproj: |
| |
| 2017-01-30 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] Upload a diff of WebKit libwebrtc code and original libwebrtc code |
| https://bugs.webkit.org/show_bug.cgi?id=167573 |
| |
| Reviewed by Alex Christensen. |
| |
| * WebKit/patch-libwebrtc: Added. |
| |
| 2017-01-27 Dan Bernstein <mitz@apple.com> |
| |
| Ignore Xcode’s project.xcworkspace and userdata directories in this new project like we do |
| in other projects. |
| |
| * libwebrtc.xcodeproj: Added property svn:ignore. |
| |
| 2017-01-24 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Use HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE for libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=167353 |
| |
| Reviewed by Alex Christensen. |
| |
| * CMakeLists.txt: |
| |
| 2017-01-23 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Filter libwebrtc link flags |
| https://bugs.webkit.org/show_bug.cgi?id=167287 |
| |
| Reviewed by Alex Christensen. |
| |
| * CMakeLists.txt: |
| |
| 2017-01-23 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] Make VP8 optional in libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=167257 |
| |
| Reviewed by Darin Adler. |
| |
| Reusing strategy used to have VP9 optional for VP8 codec. |
| |
| * CMakeLists.txt: Updated tocompile and link vp8_noop.cc |
| * Source/webrtc/media/engine/webrtcvideoengine2.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h: |
| * Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc: |
| * Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc: Added. |
| * Source/webrtc/video/video_encoder.cc: |
| |
| 2017-01-20 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] Update build system to make G711 optional in libwebrtc |
| https://bugs.webkit.org/show_bug.cgi?id=167256 |
| |
| Reviewed by Alex Christensen. |
| |
| * CMakeLists.txt: Updating to add compilation of generic pcm encoder functions. |
| |
| 2017-01-20 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] Update libwertc AudioRtpSender::SetAudioSend |
| https://bugs.webkit.org/show_bug.cgi?id=167243 |
| |
| Reviewed by Alex Christensen. |
| |
| Introducing WEBRTC_WEBKIT_BUILD macro to match existing WEBRTC_CHROMIUM_BUILD. |
| WEBRTC_WEBKIT_BUILD is defined by current WebKit libwebrtc build system. |
| |
| * Source/webrtc/api/rtpsender.cc: |
| |
| 2017-01-20 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] libwebrtc NO_RETURN is conflicting with WebKit one |
| https://bugs.webkit.org/show_bug.cgi?id=167244 |
| |
| Reviewed by Alex Christensen. |
| |
| * Source/webrtc/typedefs.h: Defining NO_RETURN only if not already defined. |
| |
| 2017-01-20 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] libwebrtc headers are incompatible with WebKit compilation flags |
| https://bugs.webkit.org/show_bug.cgi?id=167242 |
| |
| Reviewed by Alex Christensen. |
| |
| WebKit is enforcing -Wunused-parameter and -Wunused-variable which conflict with some included libwertc headers. |
| Removed unused parameter names for inlined functions. |
| |
| * Source/webrtc/api/jsep.h: |
| (webrtc::SessionDescriptionInterface::RemoveCandidates): |
| * Source/webrtc/api/mediastreaminterface.h: |
| (webrtc::AudioSourceInterface::SetVolume): |
| (webrtc::AudioSourceInterface::RegisterAudioObserver): |
| (webrtc::AudioSourceInterface::UnregisterAudioObserver): |
| (webrtc::AudioSourceInterface::AddSink): |
| (webrtc::AudioSourceInterface::RemoveSink): |
| (webrtc::AudioTrackInterface::GetSignalLevel): |
| * Source/webrtc/api/peerconnectionfactory.h: |
| * Source/webrtc/api/peerconnectioninterface.h: |
| (webrtc::MetricsObserverInterface::IncrementEnumCounter): |
| (webrtc::PeerConnectionInterface::AddTrack): |
| (webrtc::PeerConnectionInterface::RemoveTrack): |
| (webrtc::PeerConnectionInterface::CreateSender): |
| (webrtc::PeerConnectionInterface::GetStats): |
| (webrtc::PeerConnectionInterface::CreateOffer): |
| (webrtc::PeerConnectionInterface::CreateAnswer): |
| (webrtc::PeerConnectionInterface::UpdateIce): |
| (webrtc::PeerConnectionInterface::SetConfiguration): |
| (webrtc::PeerConnectionInterface::RemoveIceCandidates): |
| (webrtc::PeerConnectionInterface::StartRtcEventLog): |
| (webrtc::PeerConnectionObserver::OnAddStream): |
| (webrtc::PeerConnectionObserver::OnRemoveStream): |
| (webrtc::PeerConnectionObserver::OnDataChannel): |
| (webrtc::PeerConnectionObserver::OnIceCandidatesRemoved): |
| (webrtc::PeerConnectionObserver::OnIceConnectionReceivingChange): |
| * Source/webrtc/api/rtpsender.cc: |
| * Source/webrtc/base/messagehandler.h: |
| (rtc::FunctorMessageHandler::OnMessage): |
| * Source/webrtc/base/sanitizer.h: |
| (rtc_AsanPoison): |
| (rtc_AsanUnpoison): |
| (rtc_MsanMarkUninitialized): |
| (rtc_MsanCheckInitialized): |
| * Source/webrtc/base/stream.h: |
| (rtc::StreamInterface::ConsumeReadData): |
| (rtc::StreamInterface::ConsumeWriteBuffer): |
| * Source/webrtc/media/base/mediachannel.h: |
| (cricket::DataMediaChannel::GetStats): |
| (cricket::DataMediaChannel::OnNetworkRouteChanged): |
| * Source/webrtc/media/engine/webrtcvideodecoderfactory.h: |
| (cricket::WebRtcVideoDecoderFactory::CreateVideoDecoderWithParams): |
| * Source/webrtc/media/engine/webrtcvideoencoderfactory.h: |
| (cricket::WebRtcVideoEncoderFactory::VideoCodec::VideoCodec): |
| (cricket::WebRtcVideoEncoderFactory::EncoderTypeHasInternalSource): |
| * Source/webrtc/media/engine/webrtcvideoengine2.cc: |
| * Source/webrtc/modules/include/module.h: |
| (webrtc::Module::ProcessThreadAttached): |
| * Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc: |
| * Source/webrtc/p2p/base/port.h: |
| (cricket::Port::HandleIncomingPacket): |
| (cricket::Port::HandleConnectionDestroyed): |
| (cricket::Connection::set_receiving_timeout): |
| * Source/webrtc/p2p/base/stun.h: |
| (cricket::StunAttribute::SetOwner): |
| * Source/webrtc/p2p/base/stunrequest.h: |
| (cricket::StunRequest::Prepare): |
| (cricket::StunRequest::OnResponse): |
| (cricket::StunRequest::OnErrorResponse): |
| * Source/webrtc/p2p/base/transport.h: |
| (cricket::Transport::SetLocalCertificate): |
| (cricket::Transport::GetLocalCertificate): |
| (cricket::Transport::GetSslRole): |
| (cricket::Transport::SetSslMaxProtocolVersion): |
| * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc: |
| * Source/webrtc/typedefs.h: |
| |
| 2017-01-20 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] Update libwertc AudioRtpSender::SetAudioSend |
| https://bugs.webkit.org/show_bug.cgi?id=167243 |
| |
| Reviewed by Alex Christensen. |
| |
| Introducing WEBRTC_WEBKIT_BUILD macro to match existing WEBRTC_CHROMIUM_BUILD. |
| WEBRTC_WEBKIT_BUILD is defined by current WebKit libwebrtc build system. |
| |
| * Source/webrtc/api/rtpsender.cc: |
| |
| 2017-01-20 Youenn Fablet <youennf@gmail.com> |
| |
| [WebRTC] libwebrtc H.264 codec is using VTB only for IOS |
| https://bugs.webkit.org/show_bug.cgi?id=167245 |
| |
| Reviewed by Alex Christensen. |
| |
| * Source/webrtc/sdk/objc/Framework/Classes/videotoolboxvideocodecfactory.cc: Removing WEBRTC_IOS flag. |
| |
| 2017-01-19 Youenn Fablet <youenn@apple.com> |
| |
| [WebRTC] Upload libwebrtc code base |
| https://bugs.webkit.org/show_bug.cgi?id=167205 |
| |
| Reviewed by Alex Christensen and Jon Lee. |
| |
| Add initial libwebrtc source from branch 56. Here's how to get what we committed: |
| git clone https://chromium.googlesource.com/external/webrtc.git && cd webrtc && git checkout 7bf536976366443ea59153ff3d22da0ec32badc1 |
