c2dfe0a8c9
Bug 1341285: Webrtc updated to branch 57 rev 52b6562a10b495; initial pull made Feb 2 2017 14:38 EST
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Pull updated from 71394677e4dc343ca5c0f996037207a9bd9616c9 to 52b6562a10b495 in late May
--HG--
rename : media/webrtc/trunk/webrtc/base/iosfilesystem.mm => media/webrtc/trunk/webrtc/base/applefilesystem.mm
rename : media/webrtc/trunk/webrtc/test/testsupport/gtest_prod_util.h => media/webrtc/trunk/webrtc/base/gtest_prod_util.h
rename : media/webrtc/trunk/webrtc/base/exp_filter.cc => media/webrtc/trunk/webrtc/base/numerics/exp_filter.cc
rename : media/webrtc/trunk/webrtc/base/exp_filter.h => media/webrtc/trunk/webrtc/base/numerics/exp_filter.h
rename : media/webrtc/trunk/webrtc/base/exp_filter_unittest.cc => media/webrtc/trunk/webrtc/base/numerics/exp_filter_unittest.cc
rename : media/webrtc/trunk/webrtc/base/rtccertificate_unittests.cc => media/webrtc/trunk/webrtc/base/rtccertificate_unittest.cc
rename : media/webrtc/trunk/webrtc/common_audio/swap_queue.h => media/webrtc/trunk/webrtc/base/swap_queue.h
rename : media/webrtc/trunk/webrtc/common_audio/swap_queue_unittest.cc => media/webrtc/trunk/webrtc/base/swap_queue_unittest.cc
rename : media/webrtc/trunk/webrtc/audio_receive_stream.h => media/webrtc/trunk/webrtc/call/audio_receive_stream.h
rename : media/webrtc/trunk/webrtc/audio_state.h => media/webrtc/trunk/webrtc/call/audio_state.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_bitstream_parser_unittest.cc => media/webrtc/trunk/webrtc/common_video/h264/h264_bitstream_parser_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/h264_sps_parser_unittest.cc => media/webrtc/trunk/webrtc/common_video/h264/sps_parser_unittest.cc
rename : media/webrtc/trunk/webrtc/frame_callback.h => media/webrtc/trunk/webrtc/common_video/include/frame_callback.h
rename : media/webrtc/trunk/webrtc/call/rtc_event_log.proto => media/webrtc/trunk/webrtc/logging/rtc_event_log/rtc_event_log.proto
rename : media/webrtc/trunk/webrtc/video/video_decoder.cc => media/webrtc/trunk/webrtc/media/engine/videodecodersoftwarefallbackwrapper.cc
rename : media/webrtc/trunk/webrtc/video/video_encoder_unittest.cc => media/webrtc/trunk/webrtc/media/engine/videoencodersoftwarefallbackwrapper_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receive_test.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/acm_send_test.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc => media/webrtc/trunk/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_tracker.cc
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack.h => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_tracker.h
rename : media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.c => media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_core_neon.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.c => media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_resampler.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.c => media/webrtc/trunk/webrtc/modules/audio_processing/aec/echo_cancellation.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.c => media/webrtc/trunk/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/agc/histogram.h => media/webrtc/trunk/webrtc/modules/audio_processing/agc/loudness_histogram.h
rename : media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/audio_processing_unittest.cc
rename : media/webrtc/trunk/webrtc/test/common_unittest.cc => media/webrtc/trunk/webrtc/modules/audio_processing/config_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft_mips.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/ooura_fft_mips.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft_neon.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/ooura_fft_neon.cc
rename : media/webrtc/trunk/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c => media/webrtc/trunk/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios.mm => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info.mm
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info_objc.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/device_info_ios_objc.mm => media/webrtc/trunk/webrtc/modules/video_capture/objc/device_info_objc.mm
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/rtc_video_capture_objc.h
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm => media/webrtc/trunk/webrtc/modules/video_capture/objc/rtc_video_capture_objc.mm
rename : media/webrtc/trunk/webrtc/modules/video_capture/ios/video_capture_ios.h => media/webrtc/trunk/webrtc/modules/video_capture/objc/video_capture.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/screenshare_layers_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp9/vp9_screenshare_layers_unittest.cc
rename : media/webrtc/trunk/webrtc/p2p/base/constants.cc => media/webrtc/trunk/webrtc/p2p/base/p2pconstants.cc
rename : media/webrtc/trunk/webrtc/p2p/base/constants.h => media/webrtc/trunk/webrtc/p2p/base/p2pconstants.h
rename : media/webrtc/trunk/webrtc/base/objc/NSString+StdString.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/NSString+StdString.h
rename : media/webrtc/trunk/webrtc/base/objc/NSString+StdString.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/NSString+StdString.mm
rename : media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCCameraPreviewView.m
rename : media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCDispatcher.m
rename : media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCEAGLVideoView.m
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceCandidate.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceServer+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceServer+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceServer.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCIceServer.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCStatsReport+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCLegacyStatsReport+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCLegacyStatsReport.mm
rename : media/webrtc/trunk/webrtc/base/objc/RTCLogging.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCLogging.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaConstraints.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaSource.mm
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCMediaStreamTrack+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.m => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCNSGLVideoView.m
rename : media/webrtc/trunk/webrtc/api/objc/RTCOpenGLVideoRenderer.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCOpenGLVideoRenderer.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription+Private.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription+Private.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/RTCSessionDescription.mm
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.cc => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_decoder.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_decoder.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_decoder.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.cc => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu_unittest.cc => media/webrtc/trunk/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu_unittest.cc
rename : media/webrtc/trunk/webrtc/base/objc/RTCCameraPreviewView.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
rename : media/webrtc/trunk/webrtc/base/objc/RTCDispatcher.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCEAGLVideoView.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceCandidate.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCIceServer.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCStatsReport.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
rename : media/webrtc/trunk/webrtc/base/objc/RTCLogging.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaConstraints.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaStreamTrack.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCNSGLVideoView.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCSessionDescription.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCVideoFrame.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCVideoRenderer.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
rename : media/webrtc/trunk/webrtc/api/objc/RTCMediaSource.h => media/webrtc/trunk/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
rename : media/webrtc/trunk/webrtc/api/objctests/RTCIceCandidateTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCIceCandidateTest.mm
rename : media/webrtc/trunk/webrtc/api/objctests/RTCIceServerTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCIceServerTest.mm
rename : media/webrtc/trunk/webrtc/api/objctests/RTCMediaConstraintsTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCMediaConstraintsTest.mm
rename : media/webrtc/trunk/webrtc/api/objctests/RTCSessionDescriptionTest.mm => media/webrtc/trunk/webrtc/sdk/objc/Framework/UnitTests/RTCSessionDescriptionTest.mm
rename : media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_mac.cc => media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_darwin.cc
rename : media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_posix.cc => media/webrtc/trunk/webrtc/system_wrappers/source/atomic32_non_darwin_unix.cc
rename : media/webrtc/trunk/webrtc/video/full_stack_plot.py => media/webrtc/trunk/webrtc/video/full_stack_tests_plot.py
rename : media/webrtc/trunk/webrtc/call/transport_adapter.cc => media/webrtc/trunk/webrtc/video/transport_adapter.cc
rename : media/webrtc/trunk/webrtc/call/transport_adapter.h => media/webrtc/trunk/webrtc/video/transport_adapter.h
rename : media/webrtc/trunk/webrtc/modules/utility/source/file_player_unittests.cc => media/webrtc/trunk/webrtc/voice_engine/file_player_unittests.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/channel_transport.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/channel_transport.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/channel_transport.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/traffic_control_win.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/traffic_control_win.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_manager_win.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_manager_win.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_win.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket2_win.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_posix.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_posix.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_unittest.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_unittest.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_wrapper.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_wrapper.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_manager_wrapper.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_posix.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_posix.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_wrapper.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_wrapper.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper_unittest.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_socket_wrapper_unittest.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport_impl.cc
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.h => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport_impl.h
rename : media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_unittest.cc => media/webrtc/trunk/webrtc/voice_engine/test/channel_transport/udp_transport_unittest.cc
2017-06-13 01:52:22 -04:00
f8412cddff
Merge inbound to central, a=merge CLOSED TREE
...
MozReview-Commit-ID: 4j8ngmg8FAD
2017-06-12 17:02:56 -07:00
104a4b2e9c
Bug 1343743 - Part6: Pass a labeled event target to StunAddrsRequestChild's constructor., r=bwc
...
Pass an event target argument in constructor, so we can call gNeckoChild->SetEventTargetForActor before sending constructor message to parent.
2017-06-12 00:25:00 +02:00
ad7b0fb2a3
Bug 1371362: Remove unused audio and video stream counters of PeerConnectionImpl r=bwc
...
MozReview-Commit-ID: AtZRVqCyF4A
--HG--
extra : rebase_source : 5c395ecd15a8403de8181675a61f2545a5a14299
2017-06-08 11:17:08 -07:00
cf31c442c8
Bug 1370601 - Make it possible for offerer and answerer to switch roles in jsep_session_unittest.cpp; r=bwc
...
MozReview-Commit-ID: A34A5ER92oP
--HG--
extra : rebase_source : 05ebb3889b8eed8c4074f6d15f3c2d2e39d48507
2017-04-07 17:29:43 -07:00
08dd28ea34
Bug 1369967 - Remove one use of a stack-allocated buffer and fix another buffer's size. r=jesup
...
Two things here:
- The default stack size of the thread pool is not very big, it's better to
stick the buffer we need on the object.
- There was a unit mismatch between bytes and samples. This changes the name to
make the unit more obvious, and fixes its usage by dividing by the sample size.
MozReview-Commit-ID: 19bbS6iGvTw
--HG--
extra : rebase_source : bb5c2c074b8c1c3d69e002c8d82f4f72cc57582d
2017-06-05 11:42:43 +02:00
a39099bca1
Bug 1166955 - Stop including nsAutoPtr.h from BasePin.cpp. r=jesup
...
MozReview-Commit-ID: CCvJyVmH1JI
--HG--
extra : rebase_source : 72cf22b38f7c4d3ef5c315351f0eca942345b752
2017-06-03 17:59:29 +09:00
269f7e9f5c
Merge m-c to inbound, a=merge
...
MozReview-Commit-ID: 9wTctDOsPpO
2017-06-02 17:31:31 -07:00
ae21d19935
Bug 1353030: document use of WrapRunnable(this) r=cpearce
...
MozReview-Commit-ID: Fb3KjsI9tE3
2017-06-02 16:36:34 -04:00
0d038d6513
Merge autoland to m-c a=merge
...
MozReview-Commit-ID: Fjt5XIDd0p6
2017-06-02 17:21:39 -07:00
858e777c78
Bug 1369108 - 3. Implement new device permission code path for Fennec; r=esawin
...
Instead of asking for permission in VideoCaptureDeviceInfoAndroid.java,
we now merely check for permission there. The actual permission prompt
now happens in WebrtcUI.js, using the new
"getUserMedia:ask-device-permission" and
"getUserMedia:got-device-permission" notifications.
MozReview-Commit-ID: DSVPjjW2JNR
2017-06-02 16:11:53 -04:00
9798dcedbb
Bug 1369108 - 2. Refresh Android camera list when necessary; r=jesup
...
Currently, if permission is first denied, the list of cameras is empty.
However, if permission is later granted, the list stays empty because we
never try to refresh the list. This patch causes us to refresh the list
when necessary.
MozReview-Commit-ID: 5eodPCWVyaP
2017-06-02 16:11:53 -04:00
9406837246
Bug 1367930: adjust ssrc's and encondings as simulcast answerer. r=bwc
...
MozReview-Commit-ID: EPdIWF5nn7u
--HG--
extra : rebase_source : 5301c7541f1f2d2eeac37f5b0545ad83e71d8e46
2017-05-27 00:04:55 -07:00
f8f118d912
Bug 1335262: read and emit datachannel max-message-size. r=jesup
...
MozReview-Commit-ID: HwaoshovZIS
--HG--
extra : rebase_source : c1c171c53e5433607083dcd88fc9de0820e3382f
2017-05-31 20:58:53 -07:00
712586b20b
Bug 1369724: Fix setting size for fake H264 codec r=jya
...
From bug 1341285
MozReview-Commit-ID: E6g6phiQC5H
2017-06-02 09:59:36 -04:00
8cb88eaf14
Bug 1176415: verify absence of SSRC's in data channel m-sections. r=bwc
...
MozReview-Commit-ID: ADstE36mzzu
--HG--
extra : rebase_source : 4f94e662bb931ed7d5b87a7658a78d19ad450976
2017-06-01 11:29:07 -07:00
b922e0c78a
Bug 1176415: stop adding ssrc's to data channel m-sections. r=bwc
...
MozReview-Commit-ID: H7tiJ9YRThQ
--HG--
extra : rebase_source : ac638a15838cef16d9c614ca7eb677960e38df2f
2017-05-31 22:13:04 -07:00
e45ddffb1e
Bug 1359775 - Part 1 - add RTCRtpContributingSourceStats;r=jib,smaug
...
Still left TODO:
* add an aboutWebrtc.js section
* write tests
MozReview-Commit-ID: DwFxq19KWeu
--HG--
extra : rebase_source : fad3018d851316af83df48c62db16028a1a84b5c
2017-04-26 04:27:13 -07:00
e7f79e098e
Bug 1363885 - 2. Remove ViERenderer dependency on orientation listener; r=snorp
...
ViERenderer is not used anywhere but has a couple calls to the obsolete
GeckoAppShell orientation listener. The entire ViERenderer.java file is
getting removed in the upcoming WebRTC update, so we should just go
ahead and remove those lines.
MozReview-Commit-ID: AwG7dBg5MV8
2017-05-25 18:33:30 -04:00
411dd4acd1
Bug 1366581: offer bundle support for single m-section. r=bwc
...
MozReview-Commit-ID: DYmjCmV1fKF
--HG--
extra : rebase_source : 1d9e60663e11261f4ef8fd578de149f9b9517910
2017-05-22 18:42:12 -07:00
fd474519c7
Bug 1365291 - capture 'this' with RefPtr on dispatch to STS thread. r=jib
...
- Avoid any potential for this going away from underneath the dispatch
to STS thread.
- Added notes in PeerConnectionImpl on the test-only nature of the
AddRIDExtension, AddRIDFilter, and GetMediaPipelineForTrack methods.
PeerConnectionImpl::GetMediaPipelineForTrack by returning a reference
to the RefPtr instead of a copy.
MozReview-Commit-ID: EwMr9ulKtm8
--HG--
extra : rebase_source : 55c8b14f63020feda57accd2b4b331de708866c4
2017-05-16 16:07:33 -05:00
117651c749
Bug 1325991 - sections with bundle-only should have port set to 0. r=drno
...
Now that Chrome release is bundle-aware, let's reapply the patch to
properly emit port 0 for m-lines in sections with the bundle-only
attribute.
MozReview-Commit-ID: 8RftSXIzIpD
--HG--
extra : rebase_source : 6f9c4cb6b322aec7c00060827e1f5e7852f8acfc
2017-05-08 14:06:54 -05:00
edfc95cb55
Bug 1338521: don't over-write remote SSRC with random value r=jesup
...
MozReview-Commit-ID: CEZnuzxeHkz
--HG--
extra : rebase_source : 6541a71ca473af08075895e047218460e8407a2f
2017-04-28 16:00:06 -07:00
652aa12a75
Bug 1361206: warn about non-matching RTP header extension IDs. r=bwc
...
MozReview-Commit-ID: DLG5ICBydAK
--HG--
extra : rebase_source : dfc81395d47ddd25c9347e94e6cf630d727cdd43
2017-05-01 21:02:53 -07:00
4dd7f2dc11
Bug 1365090: use target bitrate instead of max for simulcast. r=bwc
...
MozReview-Commit-ID: GThcXHHnoCV
--HG--
extra : rebase_source : 352a82ad81858782898a10440ff77b4891af6a60
2017-05-16 16:15:04 -07:00
4d8185fc04
Bug 1365090: clear RID vector when reconfiguring send media codec. r=jesup
...
MozReview-Commit-ID: Bs5Cihjt8fV
--HG--
extra : rebase_source : 0e423cdf7b82ecca29aecc2c72601005b56786cd
2017-05-15 21:47:34 -07:00
c1d1748428
Bug 1359490 - add an event loop spinning abstraction function; r=gerald
...
This function is arguably nicer than calling NS_ProcessNextEvent
manually, is slightly more efficient, and will enable better auditing
for NS_ProcessNextEvent when we do Quantum DOM scheduling changes.
2017-05-15 09:34:19 -04:00
0f4a3fb4f5
Bug 1364325: replace AddLocalRTPExtensions with SetLocalRTPExtensions. r=bwc
...
MozReview-Commit-ID: G98AVhWA5FU
--HG--
extra : rebase_source : 4d24e3814adf60fc188e050760fc07a4010a9b15
2017-05-11 23:08:37 -07:00
9b6b4c7751
Bug 1363563: remove and erase existing header extensions. r=mjf
...
MozReview-Commit-ID: IzVEaOhLNwR
--HG--
extra : rebase_source : 0574353c3f0be051fe766090802a5d91d41077e9
2017-05-10 16:19:40 -07:00
d13f3ad75f
Bug 1361139 - pt 2 - remove test-related SSRC filtering from MediaPipeline. r=drno
...
Now that RID filtering (Bug 1358224) has fixed the intermittant oranges
from Bug 1351531 and 1351590, remove the functionality from MediaPipeline.
MozReview-Commit-ID: 1rED3iaHRCK
--HG--
extra : rebase_source : 5539f9badc99a8abfcf5419b436718233e9ab567
2017-05-05 17:32:01 -05:00
6856ba9b3d
Bug 1363259 - set min and max fps through AVCaptureConnection. r=jib
...
MozReview-Commit-ID: 4GY1gOICLqU
--HG--
extra : rebase_source : 97b50fa34186f9c92f0d01e1d486137b5159a8bd
2017-05-09 10:28:31 +08:00
667a1e29ae
Bug 1363395 - nsGlobalWindow::GetLocation should support to be called on the outer window, r=smaug
2017-05-09 22:41:19 +02:00
11d4a9bf51
Bug 1363102 - Add a error check of Location::ToString in PeerConnectionImpl.cpp, r=mystor
2017-05-08 19:48:03 +02:00
a67a0a31e8
Bug 1362003 - nsGlobalWindow::GetLocation doesn't need to receive an ErrorResult param, r=bz
...
Renaming nsGlobalWindow::GetLocation() to Location().
2017-05-08 15:49:31 +02:00
5b467557ab
Bug 1357873 - Fix format warning in webrtc; r=jesup
...
Use printf macros to fix format warnings on AArch64.
2017-05-01 14:46:01 -04:00
4e635ff251
Bug 1358224 - pt 3 - fix leak in RTPHeaderExtension's rid char buffer. r=drno
...
Turns out since Firefox doesn't receive simulcast streams, we never
noticed this leak. Convert RTPHeaderExtension.rid from a char* to
rtc::scoped_ptr<char[]> so it gets deleted properly. This also
requires a new copy constructor and assignment operator.
MozReview-Commit-ID: Jh4Gp4dAl9g
--HG--
extra : rebase_source : 8c1081fecd6e56a8f932af54fbd294adb85866f5
2017-04-27 12:27:02 -05:00
efa82b57bc
Bug 1358224 - pt 2 - change to RTP stream id filtering on simulcast mochitests. r=drno
...
The simulcast mochitests setup the receiving PeerConnection to receive
simulcast video streams which Firefox doesn't really support. Without
a test media server, this is about the best we can do and still test
simulcast.
Unfortunately the two simulcast streams arriving with different ssrcs
(as expect) exercises code we have to deal with some services switching
ssrcs midstream. In the tests, this causes intermittent failures
because the test is waiting to receive a certain ssrc, and the receiving
VideoConduit has switched to the other ssrc.
This change adds the ability to filter on RID at the MediaPipeline level,
which we can setup prior to media flowing. This avoids the ssrc switching
issue since the VideoConduit only receives one ssrc until we change the
RID filter to the second RID. At that point, the VideoConduit sees a new
ssrc and the switching code works as intended.
The modified mochitests setup the RTP stream id header extension, and then
filter on each of the RTP stream ids in turn.
MozReview-Commit-ID: KApfaxMX8rl
--HG--
extra : rebase_source : d7ae88d9675acd7b3700f342ca6a68d0bbb0ced5
2017-04-26 10:51:00 -05:00
c526c139fc
Bug 1358224 - pt 1 - addRIDExtension and addRIDFilter chrome-only API for RID (RTP Stream Id) filtering of receive tracks. r=qdot
...
The simulcast mochitests exhibit an intermittent failure due to ssrc-based
filtering that can be solved by filtering by RID. The RTP header parser
used in MediaPipeline also needs to have the RID RTP header extension
specified in order for it to properly parse the RTP header and allow
filtering on RID.
MozReview-Commit-ID: E54HCGLVYDk
--HG--
extra : rebase_source : b53085f23cb6558611aa7622f55637e19439c9c3
2017-04-26 10:01:07 -05:00
4b1556a5f2
Bug 1355056 - replace (function(args) { /* do stuff using this */ }).bind(this) with arrow functions, r=jaws.
2017-04-27 00:25:45 +02:00
64a9d5c37f
Bug 1343143: enable 3 temporal layers in simulcast mode, at all resolutions r=bwc
...
MozReview-Commit-ID: 3xf6qTMZPks
2017-04-21 11:03:45 -04:00
0f453b4ff8
Bug 1357328 - Remove media/webrtc/signaling/test/moz.build. r=jesup
...
Its content is a no-op since bug 1322707.
The code in the same directory, though, is meant to move to gtests
(bug 1316611).
--HG--
extra : rebase_source : fa269a034fd327856fde8d0673de58eba9b02d8e
2017-04-18 17:37:58 +09:00
20dd5f52c2
Merge m-c to autoland, a=merge
2017-04-13 17:35:34 -07:00
514e230373
Merge inbound to central, a=merge
2017-04-13 17:24:01 -07:00
a9c52a60b1
Bug 1241066 - fix mozRtt always 0 or 1;r=jib
...
My shortest patch to date.
MozReview-Commit-ID: 8r3ZrGUk40D
--HG--
extra : rebase_source : 38cc51ce85e03c03f46e063bf92f594927d1365f
2017-03-20 16:58:53 -07:00
75a0220f53
Bug 1325513: Check RTP extension header length. r=jesup
...
MozReview-Commit-ID: 6sUVQjUh8bF
--HG--
extra : rebase_source : 296cb8688a9c27b437380e5f70fd3cf9d43629f2
2017-04-12 15:09:18 -07:00
ab9060d531
Bug 1349883 - part 3: resolve decode promise according to buffer status. r=jya
...
MozReview-Commit-ID: JwOOi56t30Y
--HG--
extra : rebase_source : 48f0cc0bfde3f84cc0574c5a5da6c738112c843f
2017-04-07 17:07:02 +08:00
37ee2e02bb
Bug 1355259: only filter out udp candidates if force_tcp is set. r=mjf
...
MozReview-Commit-ID: AKv0N74epZ1
--HG--
extra : rebase_source : 9a77174afca8fd45b614ec49f4eb86f04ced888a
2017-04-10 16:52:09 -07:00
d61aa73f60
Bug 1354993 - Add PictureID into VP8 CodecSpecificInfo. r=jolin
...
MozReview-Commit-ID: 7l2WygPuHRX
--HG--
extra : rebase_source : 1594349dbce9140041e0a976534327f5fd4ed8be
2017-04-10 14:49:36 +08:00
9352ee95aa
Bug 1344970 - rename mozRtt to roundTripTime r=jib,smaug
...
MozReview-Commit-ID: 3kES8JUPd3n
--HG--
extra : rebase_source : e49846845d7cbd43f96d13cb1881e0383517f197
2017-03-06 15:50:10 -08:00
7c01ecff24
Bug 1337294 - Removed the unnecessary call to c_str() r=jesup
...
MozReview-Commit-ID: CpDEC7kRSXs
--HG--
extra : rebase_source : 7d44fe50c35abc6ee643a35821c9b3dea9560f10
2017-03-02 11:09:08 +05:30
Randell Jesup
Wes Kocher
Kershaw Chang
Paul Ellenbogen
Nils Ohlmeier [:drno]
Paul Adenot
Masatoshi Kimura
Jim Chen
Nico Grunbaum
Michael Froman
Nathan Froyd
Munro Mengjue Chiang
Andrea Marchesini
Florian Queze
Mike Hommey
John Lin
Vineet