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846 Commits

Author SHA1 Message Date
Ethan Hugg
2258657fc5 Bug 901560 - Datachannel no longer make second ICE component r=abr 2013-08-30 12:51:05 -07:00
Mike Hommey
25972646a7 Bug 778236 - Treat gyp files as if their content was defined in moz.build files. r=gps 2013-12-12 23:41:05 +09:00
Martin Stransky
2b7ca13346 Bug 941595 - Build with system-cairo is broken. r=glandium 2013-12-10 10:37:03 -05:00
Ryan VanderMeulen
72c57be9ea Merge m-c to inbound. 2013-12-09 17:43:44 -05:00
Carsten "Tomcat" Book
4408dc08c5 merge b2g-inbound to mozilla-central 2013-12-09 09:32:02 +01:00
Jan Beich
98d71661b0 Bug 947862 - Initialize RealTimeClock on BSDs. r=jesup 2013-12-09 21:07:46 +01:00
Birunthan Mohanathas
759ab69b0a Bug 713082 - Part 2: Rename Util.h to ArrayUtils.h. r=Waldo 
--HG--
rename : mfbt/Util.h => mfbt/ArrayUtils.h
 2013-12-08 21:52:54 -05:00
Suhas
7b99f9f628 Bug 932845 - Support hints for non gUM mediastreams. r=jesup 2013-12-08 21:44:32 -05:00
Randall Barker
51715eb475 Bug 932295 - Android video capture should autofocus. r=gcp 2013-12-08 21:42:06 -05:00
Jan-Ivar Bruaroey
8511942534 Bug 908695: Add Packets/Bytes-sent/received to webrtc stats API r=jesup DONTBUILD 2013-12-07 20:31:56 -05:00
Randell Jesup
e7e3888f6f Backout 2bb7e7192ced for landing with wrong bug number rs=backout DONTBUILD 2013-12-07 23:45:21 -05:00
Jan-Ivar Bruaroey
944e3b35f2 Bug 902003: Add Packets/Bytes-sent/received to webrtc stats API r=jesup 2013-12-07 20:31:56 -05:00
Jon Coppeard
5bba15e3e0 Bug 942928 - Fix test code link error when build with GenerationalGC r=glandium 2013-12-07 12:49:02 +00:00
Randell Jesup
9831a6f2a3 Bug 947429: add prefs that let us disable gum video and peerconnection video r=jib 2013-12-06 18:43:02 -05:00
Phil Ringnalda
f199688512 Back out d5d23d937ffc (bug 947429) on suspicion of breaking gaia-ui-test 2013-12-06 19:50:54 -08:00
Randell Jesup
3f6b6af7ac Bug 947429: add prefs that let us disable gum video and peerconnection video r=jib 2013-12-06 18:43:02 -05:00
EKR
581ab6a60e Bug 906968 - Add support for TURN TCP. r=abr 
--HG--
rename : media/mtransport/third_party/nICEr/src/stun/nr_socket_turn.h => media/mtransport/third_party/nICEr/src/stun/nr_socket_buffered_stun.h
 2013-12-06 10:20:19 -08:00
Max Jonas Werner
d381473051 Bug 892148 - Make DTLS fingerprint accessible from JS r=abr 2013-12-04 15:36:00 -06:00
Jan-Ivar Bruaroey
06185186a7 Bug 944854 - Avoid triggering harmless assertion on PeerConnectionObserver weakref. r=abr 2013-12-03 09:02:07 -05:00
Shian-Yow Wu
5e3f46cbbe Bug 919979 - Crash when calling WebrtcVideoConduit::SendVideoFrame() with very small resolution. r=derf 2013-12-04 13:34:59 +08:00
Mike Hommey
bf60896454 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
dac5b53f4f 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
6b707d4378 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
caa77e8a92 merge fx-team to mozilla-central 2013-11-29 10:13:09 +01:00
Matthew Gregan
358b7e64f0 Bug 942657 - Devirtualize AudioStream. r=doublec,gps 2013-11-28 18:09:08 +13:00
Benjamin Peterson
168ae22e9f backout dc9ebdf27e98 (bug 942657) for bustage 2013-11-28 20:50:16 -06:00
Matthew Gregan
647694fec9 Bug 942657 - Devirtualize AudioStream. r=doublec,gps 2013-11-28 18:09:08 +13:00
Chris Kitching
0c652ea063 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
9245936f8b Bug 874266 - Move all DEFINES that can be moved to moz.build. r=mshal 2013-11-27 22:55:07 +09:00
Mike Hommey
2b828323f2 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
fc98abb416 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
d210f8ff00 Bug 874266 - Move all DEFINES that can be moved to moz.build. r=mshal 2013-11-28 13:08:16 +09:00
Mike Hommey
1b90f90106 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]
9cadd5383f 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]
4267967009 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
d1e46b95a6 Bug 942399 - Fix -Wunused-private-field warnings in media/webrtc/signaling. r=rjesup 2013-11-25 21:48:46 -08:00
Birunthan Mohanathas
dc8784b476 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
2ab6caa2dc Merge m-c to inbound. 2013-11-22 15:37:03 -05:00
Ryan VanderMeulen
d7e49cc136 Merge b-i to m-c. 2013-11-22 15:35:31 -05:00
Adrian Cruceru
7172d47344 Bug 876876: Fix race condition in DeviceInfoDS::GetDeviceInfo()/GetDeviceFilter() r=bas,jesup 2013-11-22 15:29:52 -05:00
Ethan Hugg
51287137ee 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
8bdf89552d 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
468c9672d8 Bug 937119 - Increase default Android audio buffers to 40ms. r=jesup 2013-11-19 15:02:23 -05:00
Gian-Carlo Pascutto
3bff0e8c5a Bug 937119 - Fix stereo setting errors on Android/OpenSLES. r=jesup 2013-11-19 15:01:58 -05:00
Mike Hommey
bb6779efe3 Bug 939044 - Remove most definitions of MODULE. r=mshal 2013-11-19 11:47:39 +09:00
Jan Beich
46a17fccf2 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]
ad62c8db72 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
43f13f1f8e 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
9829841e7c 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
7ee547334b 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
62c5789f79 Bug 932112: Add a non-ARM MemoryBarrier function. r=glandium 2013-11-07 20:07:48 -05:00
Gian-Carlo Pascutto
0430169a6c Bug 932112: Initialize both JNI and OpenSLES so fallback can work. r=jesup 2013-11-07 20:07:48 -05:00
Randell Jesup
8c4c3f9d55 Bug 932112: JB reflect for low-latency params r=mfinkle 2013-11-07 20:07:47 -05:00
Gian-Carlo Pascutto
a830153b02 Bug 932112: Use the non-main-thread FindClass implementation r=blassey 2013-11-07 20:07:47 -05:00
Randell Jesup
52771d4abf 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
836b549082 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]
c4e9698bbd Bug 936031 - Attempted fix. r=ehugg 2013-11-07 15:03:06 -08:00
Byron Campen [:bwc]
70ded53d09 Bug 936031 - Test case for bug. r=abr 2013-11-07 14:48:43 -08:00
Carsten "Tomcat" Book
f10da167db merge b2g-inbound to mozilla-central 2013-11-04 13:52:18 +01:00
Chris Pearce
e8d0d063b3 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
1c1e844b44 Bug 837563 - Enable libcubeb's PulseAudio backend. r=glandium 2013-10-31 11:37:28 +13:00
Byron Campen [:bwc]
5cc4d97f6f 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]
e1af51f0e7 Bug 906990 - Part 7: Populate candidate pairs in RTCStatsReport. r=jib 2013-10-28 16:02:00 -07:00
Wes Kocher
39d8f615ff Backed out changeset 00f838879263 (bug 906990) 2013-11-01 17:14:59 -07:00
Wes Kocher
73ea42404f Backed out changeset 57a7a785a964 (bug 906990) 2013-11-01 17:14:54 -07:00
Byron Campen [:bwc]
b65db06520 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]
64e352b730 Bug 906990 - Part 7: Populate candidate pairs in RTCStatsReport. r=jib 2013-10-28 16:02:00 -07:00
Randell Jesup
cd3e87b2df Bug 932215 - Lazily allocate log buffers for webrtc (4MB saving). r=jib 2013-10-30 12:13:07 -04:00
Nathan Froyd
3879d8a3ee Bug 933071 - add --with-macos-private-frameworks to support cross-compiling; r=mshal 2013-10-31 09:50:26 -04:00
Ethan Hugg
f0456f6520 Bug 901560 - Backout of compatibility-breaking datachannel ice component fix r=jesup 2013-10-29 08:52:04 -07:00
Gregory Szorc
9ada834d56 Bug 927837 - Don't manage generated files in configure; r=glandium 
--HG--
extra : rebase_source : b502ce209de6a0ae10e130644e424687e4fae85e
 2013-10-23 14:43:32 -07:00
Phil Ringnalda
11e739f7c6 Merge m-c to m-i 2013-10-27 19:25:15 -07:00
Jaroslav Kopecký
8dd49a58dd Bug 931590 - Pass proper directory when building --with-system-nspr r=jesup 2013-10-27 19:43:04 -04:00
Nicholas Nethercote
ba1e9bce90 Bug 925584 - Remove some unnecessary jsapi.h inclusions from .cpp files. r=Ms2ger. 
--HG--
extra : rebase_source : 41fcb0e922a519ef679c1c1b6293c2b638e83a48
 2013-10-10 15:22:35 -07:00
Phil Ringnalda
a1f80ad10b Back out f872d288480b:9b86b4e60b29 (bug 929513) for failing to build on Windows 
CLOSED TREE
 2013-10-27 15:38:40 -07:00
David Zbarsky
8ce46e2762 Bug 929513 Part 3: Use some LayerIntSize in gfx/layers r=nical 2013-10-27 17:53:27 -04:00
David Zbarsky
b28c18df90 Bug 929513 Part 1: Use gfx::IntSize for image layer sizes r=nical 2013-10-27 17:53:26 -04:00
Peter Van der Beken
cbf7a0c800 Bug 918345 - Turn on WebIDL binding generation for Window and hook it up to quickstubs. r=bz. 
--HG--
extra : rebase_source : 7bde7ddfe297e189ffa678ca1d9c34000bc904ec
 2013-10-08 17:51:42 +02:00
Ms2ger
34f7a76bb1 Backout changeset 2e466ccc7bd0 for devtools test failures. 2013-10-26 17:02:20 +02:00
Peter Van der Beken
a521d7eace Bug 918345 - Turn on WebIDL binding generation for Window and hook it up to quickstubs. r=bz. 
--HG--
extra : rebase_source : 673c08ef093339e6bfb1418366af5cc5fabe7c4d
 2013-10-08 17:51:42 +02:00
Randell Jesup
f551a29a6a Bug 920325: ntohl() isn't defined on Windows unless you include winsock/winsock2.h r=tbsaunde 2013-10-25 20:46:35 -04:00
Randell Jesup
ee8e35ca44 Bug 920325: Add WebRTC latency logging from capture to RTP and from RTP to speakers r=padenot 2013-10-25 18:13:42 -04:00
Randell Jesup
345ac3892d backout 5f38b1bd3358 for bustage CLOSED TREE 2013-10-25 19:25:54 -04:00
Randell Jesup
60b12a2e89 Bug 930603: Ensure AEC known delay doesn't go negative (rev 4886 at webrtc.org) r=jib 2013-10-25 18:21:33 -04:00
Randell Jesup
8777f9a0f5 Bug 930603: Increase WebRTC AEC tail from 48ms to 128ms (rev 4837 at webrtc.org) r=jib 2013-10-25 18:21:23 -04:00
Randell Jesup
2e3491f74c Bug 920325: Add WebRTC latency logging from capture to RTP and from RTP to speakers r=padenot 2013-10-25 18:13:42 -04:00
Jan-Ivar Bruaroey
de7feceb63 Bug 929534 r=jesup 2013-10-25 10:52:17 -04:00
EKR
595e343db4 Bug 930651 - Increase logging level for periodic A/V counters. r= abr 2013-10-24 19:23:24 -07:00
Patrick Wang
4afedddb6b Bug 881761 - Part 3: Initialize NSS when initializing PeerConnection. r=ekr,bsmith 2013-07-22 10:16:13 +08:00
Patrick Wang
6fd870fe12 Bug 932881: Explicitly cast currentSipccState to uint32_t. r=abr 2013-10-30 18:00:28 +08:00
Brian O'Keefe
4c98f61956 Bug 928709 - Convert chromium-config.mk to mozbuild, r=mshal 2013-10-02 13:17:55 -04:00
Randell Jesup
3ad7a7e6d1 Bug 864654: cleanup AudioConduit r=ekr 2013-10-23 06:20:55 -04:00
Randell Jesup
50c7d4e360 Bug 864654: merge backend for send and receive VideoConduits to match AudioConduits & cleanup r=ekr 2013-10-23 06:20:54 -04:00
Mark Banner
7433db8bcc Bug 920991 - Default stun server ip address should be changed to a domain name. r=abr 2013-10-23 09:59:37 +01:00
Ethan Hugg
fdecaf06b3 Bug 925896 - Signaling - Addref when adding sessiondata_t to hash r=abr 2013-10-22 13:14:43 -07:00
Ehsan Akhgari
3b751c0f92 Bug 928712 - Remove the rest of the unneeded prtypes.h inclusions; rs=bsmedberg 2013-10-20 22:59:48 -04:00
Ethan Hugg
7177d480ae Bug 928537 - Datachannel streams should be kept in the range 1 to MAX_NUM_STREAMS r=jesup 2013-10-18 15:01:46 -07:00
Ehsan Akhgari
0259ccb7f4 Backed out changeset 0ddbf9b3b20c (bug 928712) because of build bustage 2013-10-20 09:42:51 -04:00
Ehsan Akhgari
050f402290 Bug 928712 = Remove the rest of the unneeded prtypes.h inclusions; rs=bsmedberg 2013-10-20 09:10:07 -04:00
Jan-Ivar Bruaroey
5e7781f43a Bug 928060: Parse ?transport=[udp|tcp] in TURN uri. r=ehsan 2013-10-18 18:14:21 -04:00
Randell Jesup
c0930623a7 Bug 928221: reland (backed out due to bug 924992: webidl changes sometimes fail in incremental builds) r=jesup,abr 2013-10-19 12:21:06 -04:00
Ehsan Akhgari
2b0a6b40b4 Backed out changeset dc2b71e57211 (bug 928221) because it calls a non-existing GetWeakReferent function 2013-10-19 10:48:41 -04:00
Jan-Ivar Bruaroey
1281c48717 Bug 928221 r=jesup, abr 2013-10-18 17:22:05 -04:00
Byron Campen [:bwc]
0a8df8473f Bug 902003: Dispatch getStats to STS thread and back. r=jesup 2013-10-11 17:13:09 -07:00
Jan-Ivar Bruaroey
cd0eb58d0e Bug 902003: getStats API skeleton. r=jesup, smaug 2013-10-17 18:00:05 -04:00
Randell Jesup
42e193ce99 Bug 926598: fix some this-in-initializer warnings and remove tabs in MediaPipeline r=ekr 2013-10-16 16:12:09 -04:00
Ehsan Akhgari
f1166cb601 Bug 924107 - Make dist/include available in all of the WebRTC code; r=jesup,glandium 2013-10-15 15:08:43 -04:00
Randell Jesup
291fa9a013 Bug 910810: don't read prefs off "main" thread in unittests r=abr 2013-10-14 14:32:08 -04:00
Jan-Ivar Bruaroey
c81e140873 Bug 917328: Second, convert PeerConnectionImpl and PeerConnectionObserver to webidl. r=bz, rjesup 2013-10-14 12:53:56 -04:00
Shian-Yow Wu
0f1c51d063 Bug 881935 - Part 4: Signaling unit test for max-fs and max-fr. r=abr 2013-10-13 09:44:55 +08:00
Shian-Yow Wu
6fbcddd672 Bug 881935 - Part 3: Video conduit unit test for max-fs. r=derf 2013-10-13 09:44:53 +08:00
Shian-Yow Wu
51a96191ff Bug 881935 - Part 2: Device configuration for max-fs and max-fr. r=derf 2013-10-13 09:44:50 +08:00
Shian-Yow Wu
13b9d0d463 Bug 881935 - Part 1: SDP parsing/building for max-fs and max-fr parameters. r=abr 2013-10-13 09:43:00 +08:00
EKR
abb8ab509e Bug 925960 - Change the environment variable to enable mediaconduit_unittests. r=ehugg 2013-10-11 15:44:02 -07:00
Mike Hommey
c2fbbcbbda Bug 922460 - Kill media/webrtc/shared_libs.mk. r=ted 
--HG--
rename : media/webrtc/shared_libs.mk => layout/media/webrtc/Makefile.in
 2013-10-11 08:15:24 +09:00
Eric Rescorla
7bd3e5e22f Bug 925226 - Fix incorrect downcast in signaling unittest. r=abr 2013-10-09 20:07:00 -07:00
EKR
1123965f12 Bug 922068 - Move ICE candidate retrieval to the STS thread. r=abr 2013-10-08 15:58:13 -07:00
Ethan Hugg
f32fa0a678 Bug 916429 - use sctpmap line for datachannels r=jesup 2013-09-23 15:20:18 -07:00
Steven Michaud
4574e97d4b Bug 918943 - Duplicate symbol errors linking WebRTC when using the 10.9 SDK with --disable-optimize. r=ethanhugg 2013-10-01 20:29:50 -05:00
Nicolas Silva
8605f47406 Bug 922202 - Make PlanarYCbCrImage::Data forward-declarable and remove some header includes. r=bjacob 2013-10-01 17:57:50 -07:00
Adam Roach [:abr]
3641c7bf24 Bug 922245 - Make SDP buffer allocation dynamic in feature message r=ehugg 2013-09-30 22:28:38 -05:00
Ryan VanderMeulen
4dd576fb2f Merge m-c to inbound. 2013-09-30 16:51:06 -04:00
Ryan VanderMeulen
439f7d7d01 Merge m-c to b2g-inbound. 2013-09-30 16:30:26 -04:00
Jason Smith
d4709491c3 Bug 918186 - Add null pointer check in onPreviewFrame to prevent NullPointerException. r=gcp 2013-09-28 21:47:41 -07:00
Vladimir Vukicevic
498d8e53fa Bug 919815 - cpr_win_ipc.c not 64-bit safe. r=ehugg 2013-09-24 10:47:00 -04:00
Ethan Hugg
4e04b1c2e2 Bug 921604 - Fix trickle unittests for machines with multiple addresses r=abr 2013-09-27 13:23:15 -07:00
Adam Roach [:abr]
de9fad68d9 Bug 842549 - Part 5: Fix slots cleanup when VcmSIPCCBinding is destroyed r=ekr 2013-09-25 19:58:16 -05:00
EKR
c47e32317f Bug 842549 - Part 2: Plumb candidates up to signaling r=abr 2013-09-16 17:21:33 -07:00
Adam Roach [:abr]
d1f5ffb9c0 Bug 919767 - Clean up Call and CallInfo when finished r=ehugg 2013-09-25 19:58:15 -05:00
Gian-Carlo Pascutto
81622d3e3f Bug 918372 - Use RAII and JNI Frames for when we cannot attach+detach the JVM. r=blassey 2013-09-25 08:08:37 +02:00
Gian-Carlo Pascutto
00c747b094 Bug 918372 - Allow debugging early Android WebRTC functionality. r=blassey 2013-09-25 08:08:28 +02:00
Gian-Carlo Pascutto
4b6f20b8c6 Bug 918372 - Allocate a single GlobalRef for the Android Context. r=blassey 2013-09-25 08:06:21 +02:00
Gian-Carlo Pascutto
958950efd6 Bug 918372 - Add some debugging assertions for Android WebRTC. r=blassey 2013-09-25 08:03:40 +02:00
Jacek Caban
822115bfb3 Bug 919513 - content/media/directshow fails to compile on GCC. r=cpearce 2013-09-24 10:41:00 +02:00
Gian-Carlo Pascutto
4d2ec16675 Bug 902431 - Don't clean up references to global Android WebRTC objects. r=blassey 2013-09-23 14:41:41 +02:00
Ethan Hugg
c15070152a Bug 901560 - Datachannel no longer make second ICE component r=abr 2013-08-30 12:51:05 -07:00
Randell Jesup
f1dc6db9a7 Bug 886052: Turn on audio webrtc_trace logs for getUserMedia r=gcp 2013-09-18 17:12:38 -04:00
EKR
63298ca29b Bug 917619 - Fix setup direction when a=setup is missing r=ehugg 2013-09-17 17:43:05 -07:00
Adam Roach [:abr]
2ce5571f5d Bug 880067 - Part 5: rtcp-fb unit tests r=ekr 2013-09-05 17:00:37 -05:00
Shih-Chiang Chien
68e8048a99 Bug 918523 - Prevent rec_queue overrun. r=jesup 2013-09-28 09:12:39 +08:00
Randell Jesup
8a78d90df8 Bug 916426: Remove increment of SCTP port number when building SDP answers r=ehugg 2013-09-14 11:41:04 -04:00
Jan Beich
750262169a Bug 916216 - Add missing platforms (NetBSD, DragonFly, GNU/kFreeBSD) support to webrtc from ipc/chromium (bugs 753046 & 901414) r=jesup 2013-09-14 09:28:02 +02:00
Gian-Carlo Pascutto
229d4fe877 Bug 932692 - Check for uncaught exceptions after JNI calls followed by JNI calls. r=blassey 2013-11-22 09:54:45 +01:00
Gian-Carlo Pascutto
da7ba4559a Bug 932692 - Remove unused rotation variables in Android WebRTC driver. r=blassey 2013-11-22 09:54:44 +01:00
Steve Singer
ef58ae6716 Bug 913556 - Add exotic cpu archs to the list of platforms in webrtc (from bug #654056). r=jesup 2013-09-13 17:17:33 +02:00
Adam Roach [:abr]
a480ae37d8 Bug 880067 - Part 4: Video Conduit configuration for RTCP feedback r=ekr 2013-09-05 15:11:47 -05:00
Daniel Holbert
56ec1c6d46 Bug 915344: Make variables 'SAMPLES' and 'numSamplesReadFromInput' unsigned, to fix build warning in mediaconduit_unittests.cpp. r=jesup 2013-09-11 13:54:45 -07:00
Ethan Hugg
4fe9eed4e9 Bug 901560 - Interim fix of datachannel ICE components to be compatible with old and new versions r=abr 2013-09-04 13:13:16 -07:00
Wes Kocher
08a3746138 Backed out changeset bdcd192bda52 (bug 880067) for bustage 2013-09-10 16:51:50 -07:00
Wes Kocher
c79012984b Backed out changeset d0a0127e099e (bug 880067) 2013-09-10 16:49:26 -07:00
Adam Roach [:abr]
82f9486ac5 Bug 880067: Fix bustage r=me 2013-09-10 16:10:22 -07:00
Adam Roach [:abr]
0e0acfbaee Bug 880067 - Part 4: Video Conduit configuration for RTCP feedback r=ekr 2013-09-05 15:11:47 -05:00
Randell Jesup
5c73c402f3 Bug 904784: use a separate critical section for the recording callback r=mwu 2013-09-07 23:42:01 -04:00
Ethan Hugg
cb3f5861e5 Bug 844071 - Patch 3 - DTLS role negotiation unit test r=ekr 2013-08-26 21:55:43 -07:00
Ethan Hugg
28ed975a6c Bug 844071 - Patch 2 - Reset DTLS role on SDP negotiation r=ekr 2013-08-27 07:53:24 -07:00
Ethan Hugg
68498357a0 Bug 844071 - Patch 1 - handle building and parsing of setup and connection attributes r=abr 2013-08-27 07:40:22 -07:00
Ethan Hugg
2e43a12567 Bug 907353 - Disable second component when rtcp-mux r=ekr 2013-08-27 12:34:44 -07:00
Adam Roach [:abr]
684eb9b88f Bug 906843 - Shorten sleep period for ASSERT_TRUE_WAIT from 200ms to 10ms; change traffic checks to count packets rather than wait 10 seconds r=ekr 2013-09-05 17:11:37 -05:00
Randell Jesup
47228d99c8 Bug 899159: clean up record issues in webrtc OpenSLES code + wallpaper r=padenot,derf,mwu 
More to be done upstream and then will replace this
 2013-09-05 15:34:05 -04:00
Randell Jesup
9b423ad2fe Bug 897981: access ViEReceiver::receiving_/receiving_rtcp_ under lock (in upstream r=mflodman) 2013-09-05 15:34:05 -04:00
Randell Jesup
ac2e6b8d42 bug 912613: remove last vestige of WebRTC_Word* types in big-endian builds only r=padenot DONTBUILD 2013-09-05 15:29:36 -04:00
Mike Hommey
f5d048db5e Bug 912292 - Always traverse sub-directories after executing rules in the current directory. r=gps 2013-09-05 15:08:43 +09:00
Mike Hommey
f1cf3b4238 Bug 912293 - Remove now redundant boilerplate from Makefile.in. r=gps 2013-09-05 09:01:46 +09:00
Adam Roach [:abr]
b971650eeb Bug 906843 - Instrument signaling for isolation of system delays r=ehugg 2013-09-04 18:50:28 -05:00
Randell Jesup
07d67e84f5 Bug 912450: remove WEBRTC_EXPORT to avoid exporting webrtc symbols from xul.dll r=ted 2013-09-04 17:01:48 -04:00
Jan Beich
1a2e41945a Bug 910875 - Add missing ifdefs to make audio_device work on BSDs. r=jesup 2013-08-30 22:13:55 +02:00
Ben Brittain
8b090f4750 Bug 875097 - Telemetry for number of calls per session. r=derf 2013-08-27 19:22:19 -04:00
Randell Jesup
5946f2b5b2 Bug 901583: Reapply mozilla patches on top of webrtc.org 3.34, use NEON detection rs=jesup 
--HG--
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.cc => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.cc
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.h => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.h
 2013-08-30 02:08:57 -04:00
Randell Jesup
05d8c5f266 Bug 901583: Webrtc updated to 4563; pull made Sat Aug 17 11:00:00 EDT 2013 rs=jesup 
--HG--
rename : media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.cc => media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.cc
rename : media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.h => media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.h
 2013-08-30 02:08:04 -04:00
Michael Wu
222ae37fd1 Bug 895531 - Add support for webrtc pulseaudio backend on gonk, r=rjesup 2013-08-28 15:43:47 -04:00
Makoto Kato
03cfadff33 Bug 908523 - Build Skia on GTK3 widget. r=karlt,gps 2013-08-28 20:14:47 +09:00
Ryan VanderMeulen
3972d92239 Merge m-c to inbound on a CLOSED TREE. 2013-08-27 22:40:49 -04:00
Ben Brittain
8aed309d71 Bug 874670 - Telemetry for call duration. r=derf 2013-08-26 14:54:55 -04:00
Randell Jesup
58feac255e Bug 884365: Deliver gUM data directly to PeerConnection to avoid delay buildup and resampling r=roc 2013-08-24 09:53:11 -04:00
Adam Roach [:abr]
3ea57d4082 Bug 880067 - Part 3.1: Fix harmless copy-and-paste error r=ehugg 2013-08-27 16:15:42 -05:00
Adam Roach [:abr]
69b7aa1c8b Bug 880067 - Part 3: SDP negotiation of rtcp-fb r=ehugg 2013-08-22 13:18:38 -05:00
Adam Roach [:abr]
129f09e047 Bug 880067 - Part 2: Finish SDP Unit Tests r=ehugg 2013-08-22 13:18:38 -05:00
Nicholas Nethercote
42ccf38dcd Bug 905017 (part 1) - Minimize inclusions of JS engine headers in .h and .idl files. r=billm. 
--HG--
extra : rebase_source : 984c61ab12f46be0509b1ce0d458d9a6e5841c64
 2013-08-17 15:50:18 -07:00
Ethan Hugg
a424b3c6bd Bug 863306 - Turn off rtcp-mux in config 2013-08-20 13:21:27 -07:00
Mike Hommey
474ed6071f Bug 907473 - Handle generator_flags gracefully in gyp. r=gps 2013-08-21 09:37:45 +09:00
Wes Kocher
5af64d9ade Backed out changeset 36a2061cff79 (bug 863306) 2013-08-20 15:03:50 -07:00
Ethan Hugg
a1b1b1a394 Bug 863306 - Turn off rtcp-mux in config r=abr 2013-08-20 13:21:27 -07:00
Landry Breuil
c2c78fdefe Bug 807492 Part X - Allow gyp mozmake generator to handle various BSD flavors r=ted 2013-08-20 22:59:28 +02:00
David Zbarsky
04ff4c97fa Bug 903819 - Don't include Layers.h everywhere, part 4 r=nrc 2013-08-20 15:45:30 -04:00
Suhas Nandakumar
17835f971e Bug 863306: Propagate RTCP_MUX Status to pipeline via VCM. r=abr 2013-06-27 18:08:20 -07:00
Ms2ger
a04009b5b7 Merge latest PGO-green inbounc changeset to m-c. 2013-08-14 14:45:47 +02:00
Ms2ger
df8525f77d Bug 901323 - Don't include nsContentUtils.h unnecessarily; r=jlebar 2013-08-14 08:56:21 +02:00
Landry Breuil
dc54d7e485 Bug 807492 Part 12 - Rename _P to _pp in timestamp_extrapolator, it's a #define in ctype.h on OpenBSD, and the C99/C++ standard forbids identifiers starting with an underscode followed by a capital. r=jesup 2013-08-14 00:00:07 +02:00
Landry Breuil
d1dfb16d5d Bug 807492 Part 2 - Allow to build media/webrtc/signaling on BSD r=ehugg 2013-08-14 00:00:03 +02:00
Chris Pearce
d4df168752 Bug 861693 - Make DirectShow BaseFilter's destructor virtual, and move some code around to make our DirectShow BaseClass replacement easier to useoutside of webrtc module. r=jesup 2013-08-13 16:49:25 +12:00
Mike Hommey
99638b2be1 Bug 903341 - Avoid gyp overwriting Makefiles when they wouldn't be modified. r=gps 2013-08-10 15:55:21 +09:00
Suhas Nandakumar
b10a1e2005 Bug 786307: Implement RTCP MUX in MediaPipeline r=ekr 2013-06-27 09:13:09 -07:00
Phil Ringnalda
8c6d971e13 Back out e3483fe77b6d (bug 786307) on suspicion of causing OS X make check crashes 
CLOSED TREE
 2013-08-07 22:33:59 -07:00
Suhas Nandakumar
b2be68a5d0 Bug 786307: Implement RTCP MUX in MediaPipeline r=ekr 2013-06-27 09:13:09 -07:00
Randell Jesup
96ed08b655 Bug 901527: null pointer when resetting a resampler r=roc 2013-08-07 01:36:03 -04:00
Randell Jesup
9931376437 Bug 901527: reset the resampler on rate change r=jmspeex 2013-08-06 23:05:15 -04:00
Randell Jesup
2a8f055a74 Bug 825112: Remove jni.h from opensles per review r=mwu 2013-08-06 14:01:16 -04:00
Jon Coppeard
71a6b47923 Bug 900986 - Convert JS_*Element API to use MutableHandleValue for out params r=terrence r=bholley r=smaug 2013-08-05 14:02:47 +01:00
Randell Jesup
3f6b4f213a Bug 825112: Enable opensles webrtc backend on gonk r=mwu,jesup,ted 
--HG--
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.cc => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.cc
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.h => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.h
 2013-07-17 20:00:43 -04:00
Mike Hommey
d01b5df996 Bug 881323 - Re-implement CycleCollectorParticipant with actual vtables, with constexpr to avoid static initializers. r=mccr8 2013-08-02 10:29:05 +09:00