mozilla
/
gecko-dev
зеркало из https://github.com/mozilla/gecko-dev.git
1
0
Форкнуть 0
Код Задачи Пакеты Проекты Релизы Вики Активность
gecko-dev/media
История
Randell Jesup c2dfe0a8c9 Bug 1341285: Webrtc updated to branch 57 rev 52b6562a10b495; initial pull made Feb 2 2017 14:38 EST 
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
..
ffvpx Bug 1370062 - turn off more warnings in ffvpx. r=rillian 2017-06-04 17:42:50 +09:00
gmp-clearkey/0.1 Bug 1352556 - Validate output of MFGetAttributeSize. r=gerald 2017-05-01 16:12:27 +12:00
kiss_fft Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
libaom Bug 1370975 - aom: Use Visual Studio 14 configuration. r=gerald 2017-06-06 14:07:11 -07:00
libav Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
libcubeb Bug 1372057 - Limit pulse-rust to nightly builds. r=kinetik 2017-06-12 11:26:33 +10:00
libjpeg Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
libmkv Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
libnestegg Bug 1360396 - Update nestegg from upstream. r=kinetik 2017-04-27 17:14:25 -07:00
libogg Bug 1349467 - Fix some Audio/Video BUG_COMPONENTs in moz.build files r=jmaher 2017-03-22 17:23:12 -07:00
libopus Bug 1367208 - Update to libopus 1.1.5. r=kinetik 2017-05-23 13:38:45 -07:00
libpng Bug 1371266 - libpng should be built with -std=c89. r=glennrp 2017-06-08 06:29:03 -07:00
libsoundtouch Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
libspeex_resampler Bug 1371536 - Suppress more MSVC warnings in media/libspeex_resampler. r=padenot 2017-06-09 14:54:47 +09:00
libstagefright Bug 1367128: P2. Add methods to trim index. r=gerald 2017-06-02 21:15:10 +02:00
libtheora Bug 1349467 - Fix some Audio/Video BUG_COMPONENTs in moz.build files r=jmaher 2017-03-22 17:23:12 -07:00
libtremor Bug 1349467 - Fix some Audio/Video BUG_COMPONENTs in moz.build files r=jmaher 2017-03-22 17:23:12 -07:00
libvorbis Bug 1349467 - Fix some Audio/Video BUG_COMPONENTs in moz.build files r=jmaher 2017-03-22 17:23:12 -07:00
libvpx Bug 1279593 - Add /arch:AVX or /arch:AVX2 to compile some files under media/libvpx. r=ted.mielczarek 2017-06-06 20:31:34 +09:00
libyuv Bug 1354350 - libyuv should not build with NVALGRIND under MOZ_VALGRIND builds; r=drno 2017-05-30 15:33:55 -04:00
mtransport Bug 1343743 - Part6: Pass a labeled event target to StunAddrsRequestChild's constructor., r=bwc 2017-06-12 00:25:00 +02:00
omx-plugin Bug 1357873 - Fix format warning in omx-plugin; r=cpearce 2017-05-01 14:46:01 -04:00
openmax_dl Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
openmax_il/il112
…
pocketsphinx Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
psshparser Bug 1349467 - Fix some Audio/Video BUG_COMPONENTs in moz.build files r=jmaher 2017-03-22 17:23:12 -07:00
sphinxbase Bug 1292534 - Patch lexical parser files generated by flex. r=jgilbert 2017-03-20 18:20:19 -04:00
webrtc Bug 1341285: Webrtc updated to branch 57 rev 52b6562a10b495; initial pull made Feb 2 2017 14:38 EST 2017-06-13 01:52:22 -04:00
moz.build Bug 1344470 - add BUG_COMPONENT to media/* files. r=padenot 2017-03-07 13:06:45 -05:00
update-libjpeg.sh Bug 1278648 - Update libjpeg-turbo to version 1.5.0. r=jrmuizel 2016-06-26 15:41:09 -04:00