897f2d691d
Bug 1341285 - Restore number of CPU cores detection on BSDs. r=jesup
2017-06-17 13:43:31 -04:00
7bf7ce4954
Bug 1341285 - Sync sndio with WebRTC 57 and fix warnings. r=jesup
2017-06-17 13:43:11 -04:00
b0560565d7
Merge m-c to inbound, a=merge
...
MozReview-Commit-ID: 8k4A4tEOtIT
2017-06-16 18:17:38 -07:00
6cdb6b786c
Bug 1373450: report if MaxMessageSize was set in SDP. r=jesup
...
MozReview-Commit-ID: OqspJsw1Bw
--HG--
extra : rebase_source : ff9c5883e6d9f284c7f0771f937d48f48910b508
2017-06-15 16:12:25 -07:00
44063ab8cf
Bug 1349539 - Use CheckedInt in MediaPipeline.cpp; r=jesup
...
MozReview-Commit-ID: Lj9FlT3J42M
--HG--
extra : rebase_source : 31b4e88aaee819a3b1d083c2cc6d7c2564567f99
2017-05-31 15:54:32 -04:00
9b6197930c
merge mozilla-central to autoland. r=merge a=merge
2017-06-15 20:13:40 +02:00
1b26da1b2f
merge mozilla-central to mozilla-inbound. r=merge a=merge
2017-06-15 11:17:07 +02:00
4252852455
Bug 1373144: turn multiple msid's error into warning. r=bwc
...
MozReview-Commit-ID: 5jxHFAsbvRf
--HG--
extra : rebase_source : 5ac29c3dc12f4b8f24846a9d50cccaf2df66bb7c
2017-06-14 22:25:36 -07:00
8e114807a7
Bug 1341285 - Fix lint errors in WebRtcAudioTrack.java; r=drno
...
--HG--
extra : rebase_source : 84981127cde93ca9de53d56ee29a4d8b00aebfeb
2017-06-14 08:38:07 -04:00
ccc649dbc7
Bug 1341285 - Add missing BSD bits lost during the rebase. r=jesup
2017-06-14 20:58:52 -04:00
66305cc2e0
Bug 1371161: port SDP file parser to LibFuzzer r=decoder
...
MozReview-Commit-ID: FJhOdy2ZVqf
--HG--
rename : media/webrtc/signaling/fuzztest/sdp_file_parser.cpp => media/webrtc/signaling/fuzztest/sdp_parser_libfuzz.cpp
extra : rebase_source : f12c0e593212e077b694ef5732568e19c0a7fbed
2017-06-09 16:36:38 -07:00
bd7620cf36
Merge mozilla-central to mozilla-inbound
2017-06-13 12:11:42 +02:00
101940382d
Merge m-c to autoland, a=merge CLOSED TREE
...
MozReview-Commit-ID: K0rvhhOLins
2017-06-12 17:13:12 -07:00
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
8837b303c7
Bug 1371841: Only send PeerConnectionImpl::RecordEndOfCallTelemetry telemetry when connection information is exchanged. r=bwc
...
MozReview-Commit-ID: Dw6HMtdngD5
--HG--
extra : rebase_source : af9c6165c34afbddd1dd429c151a835acd36d742
2017-06-09 14:21:33 -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
6b92cb978e
Bug 1341285: Fix bustage on android due to merge failure r=bustage
2017-06-13 02:33:20 -04:00
450c4d90a1
Bug 1341285: rollup of changes for webrtc after applying webrtc.org v57 update r=ng,jesup,pehrsons,drno,dminor,cpearce,jya,glandium,dmajor
...
Includes re-importing gyp files removed from upstream in v56, and then
updating them to match the BUILD.gn file changes.
--HG--
rename : media/webrtc/trunk/webrtc/call/audio_send_stream.cc => media/webrtc/trunk/webrtc/call/audio_send_stream_call.cc
2017-06-13 01:54:13 -04:00
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
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
da7254ecbf
Bug 1339906 - pt 5 - add writable field to webidl for RTCIceCandidatePairStats and implement readable and writeable fields. r=drno,qdot
...
MozReview-Commit-ID: 6IODhX5mtnP
--HG--
extra : rebase_source : 30245be09b3b4bf057672cceb2d90d3393da035c
2017-06-06 17:33:02 -05:00
1b76f106fd
Bug 1339906 - pt 4 - add last sent and received timestamps to RTCIceCandidatePairStats. r=drno,qdot
...
MozReview-Commit-ID: GE23lS7qs9n
--HG--
extra : rebase_source : 5b39e4232258eca1807d3c962a2ed40c2724822b
2017-06-06 17:36:40 -05:00
eb58727687
Bug 1339906 - pt 3 - change componentId to transportId to match RTCIceCandidatePairStats spec. r=drno,qdot
...
MozReview-Commit-ID: Jfc2BOMt98v
--HG--
extra : rebase_source : db7bca1a23cc2f46cce778dd0638182cc1143791
2017-06-06 16:59:18 -05:00
69f85fff44
Bug 1339906 - pt 1 - Add bytesSent and bytesReceived to RTCIceCandidatePairStats. r=drno,qdot
...
MozReview-Commit-ID: BQGPTUzRCB3
--HG--
extra : rebase_source : cd2abb970a07479c8425a13d12fc0fbaa487b00e
2017-06-06 16:30:56 -05:00
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
Jan Beich
Wes Kocher
Nils Ohlmeier [:drno]
Dan Minor
Sebastian Hengst
Carsten "Tomcat" Book
Kershaw Chang
Paul Ellenbogen
Paul Adenot
Masatoshi Kimura
Randell Jesup
Jim Chen
Nico Grunbaum
Michael Froman
Nathan Froyd