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336 315 коммитов 615 Ветки 98 Теги 5,7 GiB
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679 Коммитов

Автор SHA1 Сообщение Дата
Max Jonas Werner d0f3485699 Bug 892148 - Make DTLS fingerprint accessible from JS r=abr 2013-12-04 15:36:00 -06:00
Jan-Ivar Bruaroey ba6d226e8f Bug 944854 - Avoid triggering harmless assertion on PeerConnectionObserver weakref. r=abr 2013-12-03 09:02:07 -05:00
Shian-Yow Wu 0508c47feb Bug 919979 - Crash when calling WebrtcVideoConduit::SendVideoFrame() with very small resolution. r=derf 2013-12-04 13:34:59 +08:00
Mike Hommey 881a7c1554 Bug 943728 - Replace double quotes with single quotes in Makefiles (or remove them when it makes sense). r=mshal 2013-12-03 06:34:21 +09:00
Mike Hommey 206b516fca Bug 944634 - mozmake.py (for gyp) handling of escaping (or lack thereof) for defines is wrong. r=ted 2013-12-03 06:34:21 +09:00
Ajitesh Gupta a3e2d917e1 Bug 917047: Remove the security.enable_md5_signatures pref, r=briansmith 
--HG--
extra : rebase_source : ae721bc7f91b399e61c2efb869e15ab902c810e0
 2013-11-19 16:05:26 -08:00
Carsten "Tomcat" Book 2bd99eb2d7 merge fx-team to mozilla-central 2013-11-29 10:13:09 +01:00
Matthew Gregan 2ca1434c82 Bug 942657 - Devirtualize AudioStream. r=doublec,gps 2013-11-28 18:09:08 +13:00
Benjamin Peterson 02f058e57a backout dc9ebdf27e98 (bug 942657) for bustage 2013-11-28 20:50:16 -06:00
Matthew Gregan 93237d768e Bug 942657 - Devirtualize AudioStream. r=doublec,gps 2013-11-28 18:09:08 +13:00
Chris Kitching 01f91bd119 Bug 709230 - Part 4: annotate members referenced by WebRTC's JNI code. r=gcp,rnewman 2013-11-18 17:27:00 -08:00
Mike Hommey 8fd06cf41b Bug 874266 - Move all DEFINES that can be moved to moz.build. r=mshal 2013-11-27 22:55:07 +09:00
Mike Hommey b32a4ed166 Backout changeset 3fd4b546eed4 (bug 874266) and changeset a35d2e3a872f (bug 942043) for ASAN build bustage and Windows test bustage 
--HG--
extra : amend_source : f20d09aeff1c8b5cbd0f1d24c7ce04e86f3aed1d
 2013-11-28 14:24:05 +09:00
Mike Hommey a1486ce449 Bug 942043 - Straighten up zlib linkage wrt shared js and system zlib. r=gps,r=ted 2013-11-22 23:03:21 +09:00
Mike Hommey 682364d535 Bug 874266 - Move all DEFINES that can be moved to moz.build. r=mshal 2013-11-28 13:08:16 +09:00
Mike Hommey 0a1fb8d11c Bug 942043 - Straighten up zlib linkage wrt shared js and system zlib. r=gps,r=ted 2013-11-28 12:49:43 +09:00
Byron Campen [:bwc] 67f0c411ba Bug 935723. Part 1. Decouple ICE state with ICE gathering state r=ekr,abr,jesup 2013-11-13 14:53:30 -08:00
Byron Campen [:bwc] aebc089f65 Bug 906990: Part 10. Webidl and implementation for WebrtcGlobal. Encompasses things like global stats and logging. r=jib,bz 2013-11-25 11:01:03 -08:00
Chris Peterson 1a33509be0 Bug 942399 - Fix -Wunused-private-field warnings in media/webrtc/signaling. r=rjesup 2013-11-25 21:48:46 -08:00
Birunthan Mohanathas a36b98a14d Bug 784739 - Switch from NULL to nullptr in webrtc/signaling/; r=ehsan 
--HG--
extra : rebase_source : 47841196d7805fd8d69749d554afcc31eff18826
 2013-11-25 14:05:03 -05:00
Ryan VanderMeulen 6777d0901e Merge m-c to inbound. 2013-11-22 15:37:03 -05:00
Ryan VanderMeulen 6789e60ca8 Merge b-i to m-c. 2013-11-22 15:35:31 -05:00
Adrian Cruceru ca8f75f22d Bug 876876: Fix race condition in DeviceInfoDS::GetDeviceInfo()/GetDeviceFilter() r=bas,jesup 2013-11-22 15:29:52 -05:00
Ethan Hugg 3b77750899 Bug 940819 - Signaling - use IPC_PRIVATE instead of generating unique key for IPC r=jesup 2013-11-21 11:08:15 -08:00
Jan-Ivar Bruaroey 5e7893b429 Bug 933447 - Use new weakRef to JSImpl feature. r=smaug, r=mccr8, r=abr 2013-11-08 19:45:59 -05:00
Gian-Carlo Pascutto 91f86bbdfa Bug 937119 - Increase default Android audio buffers to 40ms. r=jesup 2013-11-19 15:02:23 -05:00
Gian-Carlo Pascutto a12ad88339 Bug 937119 - Fix stereo setting errors on Android/OpenSLES. r=jesup 2013-11-19 15:01:58 -05:00
Mike Hommey e80e877ab7 Bug 939044 - Remove most definitions of MODULE. r=mshal 2013-11-19 11:47:39 +09:00
Jan Beich 4e080ef7cd Bug 939532 - Re-apply lost hunk from bug 807492, forgotten in bd8f1571937f. r=jesup 2013-11-17 18:56:17 +01:00
Byron Campen [:bwc] 2bc61c1c1d Bug 933841. Add event handler to dump RLogRingBuffer on test failure, and clear RLogRingBuffer on test start. r=ekr 2013-11-01 13:50:49 -07:00
Randell Jesup 397b3c64ac Bug 938070: Fix misplaced #ifdef for GONK in webrtc audio_device_impl from 3.43 merge r=jesup 2013-11-15 11:33:18 -05:00
Nathan Froyd 888032c78b Bug 933320 - part 1 - make find_sdk.py silently comply if we're not running on a Mac host; r=ted 2013-10-31 13:34:02 -04:00
Brad Lassey 0c23983f47 bug 936549 - Tab sharing capture device won't stream, add rgb image support to media pipeline r=jesup 2013-11-10 16:24:37 -05:00
Gian-Carlo Pascutto a048c6417e Bug 932112: Add a non-ARM MemoryBarrier function. r=glandium 2013-11-07 20:07:48 -05:00
Gian-Carlo Pascutto 9515f68410 Bug 932112: Initialize both JNI and OpenSLES so fallback can work. r=jesup 2013-11-07 20:07:48 -05:00
Randell Jesup 6aad2e8cb1 Bug 932112: JB reflect for low-latency params r=mfinkle 2013-11-07 20:07:47 -05:00
Gian-Carlo Pascutto 205e94be74 Bug 932112: Use the non-main-thread FindClass implementation r=blassey 2013-11-07 20:07:47 -05:00
Randell Jesup 53daa692c8 Bug 932112: Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup 
* * *
* * *
Add AndroidAudioManager to the moz.build files.
 2013-11-07 20:07:47 -05:00
Randell Jesup 8338325b64 Bug 932112: Webrtc updated to 5041, pull made Mon Oct 28 12:17:00 EDT 2013 rs=jesup 
--HG--
rename : media/webrtc/trunk/webrtc/modules/audio_coding/main/source/acm_common_defs.h => media/webrtc/trunk/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/org/webrtc/voiceengine/WebRTCAudioDevice.java => media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java
rename : media/webrtc/trunk/webrtc/modules/audio_processing/test/unit_test.cc => media/webrtc/trunk/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.h => media/webrtc/trunk/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
rename : media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/receiver_fec_unittest.cc => media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_capture/android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java => media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/CaptureCapabilityAndroid.java
rename : media/webrtc/trunk/webrtc/modules/video_capture/android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java => media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/stream_generator.cc => media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/stream_generator.h => media/webrtc/trunk/webrtc/modules/video_coding/main/source/test/stream_generator.h
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/unit_test.cc => media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/unit_test.h => media/webrtc/trunk/webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h
rename : media/webrtc/trunk/webrtc/modules/video_render/android/java/org/webrtc/videoengine/ViEAndroidGLES20.java => media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java
rename : media/webrtc/trunk/webrtc/modules/video_render/android/java/org/webrtc/videoengine/ViERenderer.java => media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViERenderer.java
rename : media/webrtc/trunk/webrtc/modules/video_render/android/java/org/webrtc/videoengine/ViESurfaceRenderer.java => media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java
 2013-11-07 20:07:47 -05:00
Byron Campen [:bwc] aa20a25d38 Bug 936031 - Attempted fix. r=ehugg 2013-11-07 15:03:06 -08:00
Byron Campen [:bwc] 18be1240fd Bug 936031 - Test case for bug. r=abr 2013-11-07 14:48:43 -08:00
Carsten "Tomcat" Book 651904135a merge b2g-inbound to mozilla-central 2013-11-04 13:52:18 +01:00
Chris Pearce a4d939b3c4 Bug 933579 - Export IsVideoContentType() to VideoUtils, so that it can be used elsewhere, and move all of VideoUtils into namespace mozilla. r=kinetik 2013-11-04 11:45:19 +13:00
Matthew Gregan 36a1e81bf8 Bug 837563 - Enable libcubeb's PulseAudio backend. r=glandium 2013-10-31 11:37:28 +13:00
Byron Campen [:bwc] d3b8da349d Bug 906990 - Part 8: Create a chrome-only stats interface, and only expose the candidate pair stats there. r=jib 2013-10-29 10:29:43 -07:00
Byron Campen [:bwc] 3bca483297 Bug 906990 - Part 7: Populate candidate pairs in RTCStatsReport. r=jib 2013-10-28 16:02:00 -07:00
Wes Kocher ee011e95f8 Backed out changeset 00f838879263 (bug 906990) 2013-11-01 17:14:59 -07:00
Wes Kocher cf41f0e098 Backed out changeset 57a7a785a964 (bug 906990) 2013-11-01 17:14:54 -07:00
Byron Campen [:bwc] 09e2c59a35 Bug 906990 - Part 8: Create a chrome-only stats interface, and only expose the candidate pair stats there. r=jib 2013-10-29 10:29:43 -07:00
Byron Campen [:bwc] 5d64fba8c5 Bug 906990 - Part 7: Populate candidate pairs in RTCStatsReport. r=jib 2013-10-28 16:02:00 -07:00