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1191 Commits

Author SHA1 Message Date
Randell Jesup
540b15e55f Bug 1038961: Patch 2 - Associate GMP plugin crash with a window and notify it r=bz,jib 2014-07-21 03:50:11 -04:00
Randell Jesup
57d4e61abc Bug 1038961: Patch 1 - Send GMP plugin crashes to observer, and implement PluginID system r=cpearce,jib 2014-07-21 03:50:09 -04:00
Jan-Ivar Bruaroey
41fc3f1755 Bug 1033833 - finish plumbing offerToReceiveAudio|Video to long. r=abr 2014-07-18 18:08:30 -04:00
Jan-Ivar Bruaroey
8e6e21ea2b Bug 1033833 - Remove signaling unittests for createAnswer options. r=abr 2014-07-18 17:58:55 -04:00
Jan-Ivar Bruaroey
f309bbf947 Bug 1033833 - Update CreateOffer/Answer API to spec - no longer takes constraints but a dictionary. r=smaug, r=abr 2014-07-10 14:31:25 -04:00
Jan Beich
493752226b Bug 1040168 - Unbreak WebRTC on more archs without SSE2 after bug 983504. r=gcp 2014-07-19 21:32:18 -04:00
Randell Jesup
28cc33c5b3 Bug 1040345: Fix shutdown design issues with Webrtc GMP interfaces and quash leaks r=gcp 2014-07-19 19:14:03 -04:00
Ryan VanderMeulen
6725688485 Merge m-c to inbound. a=merge 2014-07-18 10:25:14 -04:00
Mike Hommey
03ca84f432 Bug 1039897 - Don't tie webrtc-required X11 library requirements to in-tree cairo flags. r=ted,a=kwierso 2014-07-17 14:55:11 +09:00
Randell Jesup
e1e445ccda Bug 1037754: Must always use SyncRunnable for DISPATCH_SYNC on non-nsThreads or risk leaks r=drno 2014-07-17 03:08:38 -04:00
Chris Peterson
d433ecea90 Bug 1039917 - Fix clang and gcc warnings in webrtc/signaling. r=jesup 2014-07-15 20:28:57 -07:00
Matthew A. Miller
cf6bdfb1db Bug 1040124 - WebRTC Signaling tests fail to build on Ubuntu 12.04LTS r=ted,glandium 2014-07-17 22:23:00 -04:00
Randell Jesup
9543eb65d2 Bug 1038926 - implement window sharing in webrtc/getUserMedia r=jesup,gcp,smaug 2014-07-17 22:23:00 -04:00
Randell Jesup
11bdb873e7 Bug 1037754: Query GMPService to determine if H.264 is available r=cpearce 2014-07-16 22:59:17 -04:00
Randell Jesup
6295a6b90c Backed out changeset 6d976c67e926 (bug 1037754) 2014-07-16 23:50:10 -04:00
Randell Jesup
f4a7eaa5b8 Bug 1037754: Query GMPService to determine if H.264 is available r=cpearce 2014-07-16 22:59:17 -04:00
Gian-Carlo Pascutto
e0160eacdc Bug 1038799 - Properly wrap Xfixes.h header. r=glandium 2014-07-16 19:16:38 +02:00
Gian-Carlo Pascutto
4070c32981 Bug 983504 - Screensharing fix: B2G doesn't use X11. r=ted 2014-07-16 19:16:38 +02:00
Gian-Carlo Pascutto
c818c48e44 Bug 983504 - Enable SS2 flags for desktop capture during Mozilla build. r=ted 2014-07-16 19:16:38 +02:00
Matthew A. Miller
634daa255a Bug 983504 - Buildsystem changes for multimonitor support. r=ted 2014-07-16 19:16:38 +02:00
Gian-Carlo Pascutto
6fe9394267 Bug 983504 - Buildsystem changes for screen sharing. r=ted 2014-07-07 08:50:00 +02:00
Gian-Carlo Pascutto
861d824595 Bug 983504 - ViECapturer changes for screen sharing. r=jesup 2014-07-08 05:49:00 +02:00
Matthew A. Miller
4013f51303 Bug 983504 - ViEInputManager & config changes for screen sharing. r=gcp 2014-07-07 09:55:00 +02:00
Gian-Carlo Pascutto
18b8faebed Bug 983504 - Generic DesktopCaptureImpl implementation for screen sharing. r=jesup 2014-07-08 10:04:00 +02:00
Matthew A. Miller
88c0779dab Bug 983504 - Desktop capture code changes/updates for screen sharing. r=gcp 2014-07-08 10:00:00 +02:00
Gian-Carlo Pascutto
e6cc55d88e Bug 983504 - Add new files, null implementations for screen sharing. r=jesup 2014-07-08 10:06:00 +02:00
Randell Jesup
1f63fd310c Bug 1037910: Set H264 FMTP payload value even if max_fs/fr aren't set r=ehugg 2014-07-12 22:11:01 -04:00
Randell Jesup
d7d40e9c68 Bug 1037626: Support Webrtc H.264 offers with only packetization mode 1 r=ehugg 2014-07-11 16:35:36 -04:00
Martin Thomson
b7e17fcef6 Bug 1037205 - Initialize mPrivacyRequested. r=bwc 2014-07-10 15:48:00 -04:00
Chris Pearce
e7c5d218c2 Bug 1037317 - Move GMPBufferType to be a property of GMPVideoFrameEncoded. r=jesup 2014-07-11 10:39:10 -04:00
Jan Beich
7bb4d55796 Bug 1037363 - Unbreak WebRTC on BSDs after bug 1036049. r=jesup 2014-07-11 03:13:00 -04:00
Randell Jesup
e11b6fcb74 Bug 1036049: Support H.264 STAP-A depacketization in webrtc r=ehugg 2014-07-11 01:48:14 -04:00
Chris Pearce
2920e1c8f0 Bug 1020760 - Pass GMP codec specific info as a uint8_t[], and pass buffer type separately. r=jesup 2014-07-11 15:36:21 +12:00
Chris Pearce
9ede5114eb Bug 1020760 - Update GMP APIs to support EME plugins. r=jesup 2014-07-11 15:35:56 +12:00
Ryan VanderMeulen
d6e5175f96 Backed out 5 changesets (bug 1020760, bug 1035653, bug 1020090) for leaks on a CLOSED TREE. 
Backed out changeset f0b20e3db93c (bug 1020760)
Backed out changeset 412b654e5cd2 (bug 1035653)
Backed out changeset 01ba0892af29 (bug 1020760)
Backed out changeset c7de1f4b078f (bug 1020760)
Backed out changeset 96aa9d33a1f5 (bug 1020090)
 2014-07-10 21:43:04 -04:00
Chris Pearce
ae2830d64c Bug 1020760 - Remove assertion that doesn't compile on Linux Debug on TBPL. r=bustage CLOSED TREE 2014-07-11 13:21:12 +12:00
Chris Pearce
d4a63d9c19 Bug 1020760 - Pass GMP codec specific info as a uint8_t[], and pass buffer type separately. r=jesup 2014-07-11 12:21:13 +12:00
Chris Pearce
8c996fc76f Bug 1020760 - Update GMP APIs to support EME plugins. r=jesup 2014-07-11 12:20:51 +12:00
Randell Jesup
cf095091a1 Bug 1022008: Hook up SDP negotiation for H.264 GMP codecs r=ehugg 2014-07-08 15:28:56 -04:00
Randell Jesup
6c9637ba4a Bug 1035067: Don't hint we expect a track if we're not going to receive it r=ehugg 2014-07-07 14:45:36 -04:00
Randell Jesup
11047083d3 Bug 989944: Increase decode timestamp map to handle delayed decode on 8x10 r=jesup 2014-07-03 12:46:28 -04:00
Wes Kocher
2c188e3374 Merge m-c to inbound 2014-07-02 17:44:20 -07:00
Changbin Park
4c8f4fab91 Bug 1029983 - H.264 codec is working on B2G ignoring preference 'media.peerconnection.video.h264_enabled'. r=ehugg 2014-07-01 16:09:20 -07:00
Martin Thomson
c5c3855cbb Bug 1032525 - Making isolation dependent on peerIdentity property r=abr 2014-07-02 13:56:10 -07:00
Randell Jesup
22997cd9a3 Bug 979716: drop opus bitrate to 16000bps to reduce mobile cpu use r=jmspeex 2014-07-01 05:10:49 -04:00
Randell Jesup
0f90121c45 Bug 979716: Make Opus complexity configurable in WebRTC; default Gonk to complexity 1 r=jmspeex 2014-07-01 05:10:44 -04:00
Randell Jesup
00669b380e Bug 1022008: Support max-fs & max-fr in SDP for H.264; clean up video codec fmtp generation r=ehugg 2014-07-01 04:19:32 -04:00
Chris Pearce
771af733aa Bug 1024300 - Allow GMPs to be segregated by origin. r=josh 2014-06-30 11:02:39 +12:00
Randell Jesup
ba88c5d5e1 Bug 1031500: Increase number of buffers for webrtc OMX H.264 decode r=sotaro 2014-06-27 21:49:24 -04:00
Randell Jesup
1f5537d9c4 Bug 1030338: Don't assert or generate bad stats if we go from ICE New->Closed directly r=bwc 2014-06-27 13:55:40 -04:00
Gian-Carlo Pascutto
f4ca624100 Bug 1018928 - Work around Camera focus mode bug in some Android devices. r=blassey 2014-06-27 12:13:50 +02:00
Benoit Jacob
2088c4eef4 Bug 1028588 - Fix dangerous public destructors in media/webrtc/ - r=rjesup 2014-06-26 09:31:20 -04:00
Chris Pearce
3e97757bd4 Bug 1024300 - Backout 72040861741d. r=burninator. 2014-06-26 16:00:28 +12:00
Chris Pearce
418fcf0ab2 Bug 1024300 - Allow GMPs to be segregated by origin. r=josh 2014-06-26 15:44:54 +12:00
Paul Kerr
ce59bfde23 Bug 1027100: visual distortion work-around by re-initializing the vp8 encoder on frame size changes r=jesup 2014-06-25 13:40:18 -07:00
Ethan Hugg
15c6e24190 Bug 1028962 - Fix for setting maxFramerate with Gecko Media Plugins. r=jesup 2014-06-25 09:08:41 -07:00
Byron Campen [:bwc]
7af124268c Bug 1028408 - Expose candidate pair stats to content. r=drno 2014-06-20 14:47:14 -07:00
Chris Peterson
e71b9b477d Bug 1026336 - Fix warnings in content/media/webrtc and mark FAIL_ON_WARNINGS. r=jesup 2014-06-15 11:57:30 -07:00
Benoit Jacob
817cdfbfe9 Bug 1027251 - Fix or whitelist dangerous public destructors in media/webrtc - r=rjesup 2014-06-20 07:08:23 -04:00
Birunthan Mohanathas
bc0233fe47 Bug 1026535 - Fix mismatched class/struct tags. r=ehsan 2014-06-18 17:57:51 -07:00
Carsten "Tomcat" Book
61dfe39e65 Merge mozilla-central to b2g-inbound 2014-06-17 14:40:36 +02:00
Ehsan Akhgari
47286f9f0b Bug 950676 - Enable unified builds for b2g by default; r=glandium 2014-06-17 08:35:19 -04:00
Ehsan Akhgari
8d64d57e4f Bug 1025393 - Enable building webrtc with clang-cl; r=jesup 
--HG--
extra : rebase_source : 16c3846d3a31b71e4ba3f9e4214c1ef8ff6a03e4
 2014-06-16 18:17:47 -04:00
Randell Jesup
2697000e13 Bug 1025176: Save AEC dumps in a specified directory depending on platform/pref r=pkerr 2014-06-16 15:51:45 -04:00
Randell Jesup
29b465940b Bug 1025349: fix error in ccsnap line label indexes r=ehugg 2014-06-16 15:10:16 -04:00
Randell Jesup
586275c808 Bug 1025354: fix out-of-sync name array for SIPCC logs r=ehugg 2014-06-16 15:10:05 -04:00
Randell Jesup
182834f226 Bug 1025343: fix issues with overlong codec names in AudioConduit r=pkerr 2014-06-16 01:00:33 -04:00
Randell Jesup
ff19ae9907 Bug 1025106: if someone passes us a bogus videocodec config, say it's 'unknown' r=pkerr 2014-06-16 01:00:25 -04:00
Randell Jesup
9424944a6a Bug 1022235: Make the webrtc LoadManager/LoadMonitor a singleton r=bsmedberg,pkerr 2014-06-13 15:50:51 -04:00
Randell Jesup
e1780c8d5c Bug 1024288: Add a button to about:webrtc to turn on/off AEC logging r=jib,smaug,unfocused 2014-06-12 12:21:38 -04:00
Randell Jesup
9734e5889c Bug 1024288: Allow aec debug data to be dumped on the fly, with max size r=pkerr 2014-06-12 12:20:10 -04:00
Ed Morley
f1343cd304 Backed out changeset 7b4feb3d3a39 (bug 1024288) for compilation errors; CLOSED TREE 2014-06-12 17:41:12 +01:00
Ed Morley
226523e5a8 Backed out changeset 5d63a1316981 (bug 1024288) 2014-06-12 17:40:44 +01:00
Randell Jesup
06c4824015 Bug 1024288: Add a button to about:webrtc to turn on/off AEC logging r=jib,smaug,unfocused 2014-06-12 12:21:38 -04:00
Randell Jesup
c611dcc32d Bug 1024288: Allow aec debug data to be dumped on the fly, with max size r=pkerr 2014-06-12 12:20:10 -04:00
Randell Jesup
86210c1be1 Bug 1017332: log WebRTC SDP parse errors due to no \n r=ehugg 2014-06-12 12:03:42 -04:00
Byron Campen [:bwc]
77f21e56c7 Bug 1022776 - Bump max transmit count by 1 and modify unit-tests to compensate. r=ekr 2014-06-09 17:31:44 -07:00
Karl Tomlinson
6bb9b6ad65 b=1023697 use MediaStream to convert between stream time and seconds/ticks in MediaPipeline r=roc 
The fake graph needs an implementation of the conversion methods.

The real graph will change to use audio ticks for time in a subsequent patch,
but the fake graph doesn't know about MEDIA_TIME_FRAC_BITS, so that change
can be made now in the fake graph.

--HG--
extra : transplant_source : %22%C4%01Yh%5D%F0%A6%11%40%CD%B5%89%0A%8C%8A%C2%19%5E%CC
 2014-06-12 16:44:58 +12:00
Chris Peterson
9349d93d76 Bug 1023075 - Fix more clang warnings in webrtc/signaling. r=jesup 2014-06-09 22:42:11 -07:00
Randell Jesup
97ee4f6627 Bug 970713: Adjust webrtc trace buffering for about:webrtc changes r=pkerr 2014-06-09 04:34:37 -04:00
Jan-Ivar Bruaroey
2ab2b54f0d Bug 970713 - Add 'Start Debug Mode' button to about:webrtc. r=smaug, r=Unfocused, r=jesup 2014-06-08 21:00:12 -04:00
Paul Kerr [:pkerr]
9d6e8e5434 Bug 970713 - Part 1: Control webrtc logging from about:config settings r=jesup 2014-06-08 18:54:47 -07:00
Randell Jesup
376a8cd458 Bug 999704: Implement GMP codec interface to webrtc (not enabled yet) r=joshmoz,ehugg,jesup,pkerr 2014-06-08 17:25:18 -04:00
Ryan VanderMeulen
cbc7d5d8db Backed out changeset 2af237fa2079 (bug 999704) for bustage. 
CLOSED TREE DONTBUILD
 2014-06-08 14:39:44 -04:00
Randell Jesup
11db644e91 Bug 999704: Implement GMP codec interface to webrtc (not enabled yet) r=joshmoz,ehugg,jesup 2014-06-08 14:07:53 -04:00
Randell Jesup
4084b370e3 Bug 970742: Add receive state monitoring to webrtc CodecStatistics r=jib 2014-06-08 11:06:30 -04:00
Randell Jesup
8fad7dd25d Bug 970742: Monitor decoder error state to enable recording errors and error recovery times r=jib 2014-06-08 10:33:02 -04:00
Jan-Ivar Bruaroey
ed8fb59254 Bug 951496 - Codec telemetry. r=jesup 2014-06-07 17:33:39 -04:00
Jan-Ivar Bruaroey
72df921a1d Bug 951496 - Codec getStats. r=smaug, r=jesup 2014-06-07 17:27:26 -04:00
Steven Lee
d63ac551ec Bug 951496 - Statistics data for checking the status of codec. r=jesup 2014-06-04 23:56:30 -04:00
Jan-Ivar Bruaroey
4a0ae13401 Bug 951496 - Fix Stastistics typo in vie_codec. r=jesup 2014-06-04 23:57:02 -04:00
Adam Roach [:abr]
c092c70c00 Bug 1018372 - Check aThread against mThread in PeerConnectionImpl constructor r=jesup 2014-06-06 15:56:47 -05:00
Karl Tomlinson
2167e5f1d0 b=1015828 match Fake_MediaStreamListener::NotifyPull time advances to timer period and Fake_AudioStreamSource::Periodic buffer size r=rjesup 
Also, increment Fake_SourceMediaStream::mDesiredTime each period,
instead of each listener notification.

--HG--
extra : rebase_source : 723a2a3b126adca486154d0b686746c21dbac37e
 2014-06-05 10:11:51 +12:00
Randell Jesup
a02f87eea0 Bug 1003712: Codec availability support and prioritization r=ehugg 2014-06-04 14:52:32 -04:00
Randell Jesup
e324737c53 Bug 1003712: Use Media Resource Manager to reserve OMX Codecs r=jhlin 2014-06-04 14:52:31 -04:00
Byron Campen [:bwc]
74c49d3d46 Bug 998989 - Part 1: ChromeOnly API for getting notifications when PCs are initted, or change ICE connection/gathering state. Also, expose the PC id, and allow getAllStats to be filtered by the same. r=jib, r=bz 2014-05-22 14:14:56 -07:00
Robert O'Callahan
2a92625af7 Bug 1015664. Part 2: Remove some NS_HIDDEN usage. r=bsmedberg 2014-06-03 00:08:24 +12:00
EKR
1ea7cf9b40 Bug 1018473. Unit test for duplicate trickle candidates. r=bwc 2014-05-31 12:06:45 -07:00
Byron Campen [:bwc]
01ccd3683c Bug 1018473: Detect when vcmRxAllocICE has already been called for a given stream, and suppress the (duplicate) connection to SignalCandidate. r=ekr 2014-05-31 19:41:53 -07:00
Byron Campen [:bwc]
3d1bd46584 Bug 1017291 - Keep track of the number of errors in AddIceCandidate before ICE completes, and record this number in telemetry in the success and failure cases separately. r=ekr 2014-05-29 08:40:31 -07:00
Mike Hommey
bcfae34d17 Fix non-unified build bustage from bug 987979 on a CLOSED TREE. r=me 2014-05-30 09:32:08 +09:00
Randell Jesup
b5ac06a0e7 Bug 987979: Patch 12 - Add webrtc JNI target annotations to stop ProGuard from removing too much code. r=blassey 2014-05-29 17:05:16 -04:00
Randell Jesup
9f738ae94f Bug 987979: Patch 11 - Add webrtc 3.50 support for Froyo/Gingerbread/Ice Cream Sandwich. r=blassey 2014-05-29 17:05:16 -04:00
Randell Jesup
7d91d878c8 Bug 987979: Patch 10 - Support building with older Android SDKs. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
07bd430f23 Bug 987979: Patch 9 - Use Android JNI Wrappers for off-thread FindClass and Global Context. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
3b05d7cae8 Bug 987979: Patch 8 - Support rotating/suspending/resuming an ongoing WebRTC call. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
4812c3eb03 Bug 987979: Patch 7 - Remove JSON/UCI requirements for Camera capture capability. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
66ce6ff1ad Bug 987979: Patch 6 - Include CPU feature detection source directly. r=blassey 2014-05-29 17:05:15 -04:00
Randell Jesup
2f742c010a Bug 987979: Patch 5 - Enable switching between OpenSLES and JNI backends, dummy OpenSLES output. r=rjesup 2014-05-29 17:05:14 -04:00
Randell Jesup
5c562e73d6 Bug 987979: Patch 4 - Rework WebRTC.org audio code for Mozilla integration. r=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
964601c191 Bug 987979: Patch 3 - Fix various build issues in webrtc.org/Mozilla integration. r=rjesup 2014-05-29 17:05:14 -04:00
Randell Jesup
21318d2311 Bug 987979: Patch 2 - Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=jesup 2014-05-29 17:05:14 -04:00
Randell Jesup
0654f9ad2b Bug 987979: Patch 1 - Webrtc updated to branch 3.50 rev 5764, pull made Mon Mar 24 15:36:34 EDT 2014 rs=jesup 
--HG--
rename : media/webrtc/trunk/webrtc/video_engine/new_include/config.h => media/webrtc/trunk/webrtc/config.h
rename : media/webrtc/trunk/webrtc/video_engine/new_include/frame_callback.h => media/webrtc/trunk/webrtc/frame_callback.h
rename : media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java => media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
rename : media/webrtc/trunk/webrtc/common_unittest.cc => media/webrtc/trunk/webrtc/test/common_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/direct_transport.h => media/webrtc/trunk/webrtc/test/direct_transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_decoder.cc => media/webrtc/trunk/webrtc/test/fake_decoder.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_decoder.h => media/webrtc/trunk/webrtc/test/fake_decoder.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_encoder.cc => media/webrtc/trunk/webrtc/test/fake_encoder.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/fake_encoder.h => media/webrtc/trunk/webrtc/test/fake_encoder.h
rename : media/webrtc/trunk/webrtc/video_engine/test/libvietest/testbed/fake_network_pipe_unittest.cc => media/webrtc/trunk/webrtc/test/fake_network_pipe_unittest.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/flags.cc => media/webrtc/trunk/webrtc/test/flags.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/flags.h => media/webrtc/trunk/webrtc/test/flags.h
rename : media/webrtc/trunk/webrtc/common_video/test/frame_generator.h => media/webrtc/trunk/webrtc/test/frame_generator.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/frame_generator_capturer.cc => media/webrtc/trunk/webrtc/test/frame_generator_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/frame_generator_capturer.h => media/webrtc/trunk/webrtc/test/frame_generator_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/gl/gl_renderer.cc => media/webrtc/trunk/webrtc/test/gl/gl_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/gl/gl_renderer.h => media/webrtc/trunk/webrtc/test/gl/gl_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/glx_renderer.cc => media/webrtc/trunk/webrtc/test/linux/glx_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/glx_renderer.h => media/webrtc/trunk/webrtc/test/linux/glx_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/linux/video_renderer_linux.cc => media/webrtc/trunk/webrtc/test/linux/video_renderer_linux.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/run_tests.mm => media/webrtc/trunk/webrtc/test/mac/run_tests.mm
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/video_renderer_mac.h => media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/mac/video_renderer_mac.mm => media/webrtc/trunk/webrtc/test/mac/video_renderer_mac.mm
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_platform_renderer.cc => media/webrtc/trunk/webrtc/test/null_platform_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_transport.cc => media/webrtc/trunk/webrtc/test/null_transport.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/null_transport.h => media/webrtc/trunk/webrtc/test/null_transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/rtp_rtcp_observer.h => media/webrtc/trunk/webrtc/test/rtp_rtcp_observer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/run_loop.h => media/webrtc/trunk/webrtc/test/run_loop.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/run_tests.h => media/webrtc/trunk/webrtc/test/run_tests.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/statistics.cc => media/webrtc/trunk/webrtc/test/statistics.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/statistics.h => media/webrtc/trunk/webrtc/test/statistics.h
rename : media/webrtc/trunk/webrtc/video_engine/test/test_main.cc => media/webrtc/trunk/webrtc/test/test_main.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/vcm_capturer.cc => media/webrtc/trunk/webrtc/test/vcm_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/vcm_capturer.h => media/webrtc/trunk/webrtc/test/vcm_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_capturer.cc => media/webrtc/trunk/webrtc/test/video_capturer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_capturer.h => media/webrtc/trunk/webrtc/test/video_capturer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_renderer.cc => media/webrtc/trunk/webrtc/test/video_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/video_renderer.h => media/webrtc/trunk/webrtc/test/video_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/d3d_renderer.cc => media/webrtc/trunk/webrtc/test/win/d3d_renderer.cc
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/d3d_renderer.h => media/webrtc/trunk/webrtc/test/win/d3d_renderer.h
rename : media/webrtc/trunk/webrtc/video_engine/test/common/win/run_loop_win.cc => media/webrtc/trunk/webrtc/test/win/run_loop_win.cc
rename : media/webrtc/trunk/webrtc/video_engine/new_include/transport.h => media/webrtc/trunk/webrtc/transport.h
rename : media/webrtc/trunk/webrtc/video_engine/test/loopback.cc => media/webrtc/trunk/webrtc/video/loopback.cc
rename : media/webrtc/trunk/webrtc/video_engine/internal/transport_adapter.cc => media/webrtc/trunk/webrtc/video/transport_adapter.cc
rename : media/webrtc/trunk/webrtc/video_engine/internal/transport_adapter.h => media/webrtc/trunk/webrtc/video/transport_adapter.h
rename : media/webrtc/trunk/webrtc/video_engine/new_include/video_renderer.h => media/webrtc/trunk/webrtc/video_renderer.h
 2014-05-29 17:05:13 -04:00
Nils Ohlmeier [:drno]
24c7206622 Bug 1014304 - Remove onconnection and onclosedconnection from RTCPeerConnection. r=jib, r=jesup, r=smaug 2014-05-28 09:36:00 -04:00
Jan-Ivar Bruaroey
16cde275f8 Bug 859565 - Remove legacy PeerConnectionImpl.readyState. r=bholley, r=abr 2014-05-17 17:11:27 -04:00
Byron Campen [:bwc]
6e411ef3aa Bug 1016724 - Make sure the word "gathering" appears in the timecard stamp for complete. r=jesup 2014-05-27 17:19:45 -07:00
Randell Jesup
3e5a393123 Bug 743703: allow mirroring of trace logs to NSPR; fix backwards lazy allocation defines r=pkerr 2014-05-28 03:18:33 -04:00
Enda Mannion
944f1623f0 Bug 1003994 - H.246 and multiple video codec tests. r=jesup 2014-05-26 10:07:19 +01:00
Jan-Ivar Bruaroey
fd6c40896d Bug 970685, telemetry for WebRTC bandwidth, stats-tweak approach. r=jesup 2014-05-27 14:41:17 -04:00
Jan-Ivar Bruaroey
68e827df7d Bug 970685 - tweak internal RTCStatsQuery to use nsAutoPtr for report, so it can be stolen 2014-05-27 12:58:03 -04:00
EKR
017a4114a2 Bug 1015409 - Fix trickle between CreateOffer() and SetLocal(). r=bwc 2014-05-27 13:13:43 -07:00
Randell Jesup
f4bb1be0ae Bug 1014819: Replace OMX GetCodecConfig with straight caching of H.264 SPS/PPS r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
0f38f52f24 Bug 985254: Modify H264 OMX code to deal with upstream code inserting start codes r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
2dc6a27a54 Bug 1014921: Wallpaper 8x10 OMX H264 encode/decode mismatch by forcing IDRs r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
d6733b246b Bug 997567: Send iframes for HW H264 encoder when bitrate changes with long GOP r=jhlin 2014-05-24 18:28:03 -04:00
Randell Jesup
8370108c2b Bug 997567: Support reconfiguration for frame-rate changes on OMX H.264 encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
05874298f3 Bug 1015029: Use OMX_VIDEO_ControlRateConstantSkipFrames mode for H.264 OMX encoder r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
28f4f67da7 Bug 989945: add a bit more logging to H264 OMX codec r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
3c43eb9170 Bug 989945: Use configureDirect to set OMX HW H264 encoder config correctly r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
3cb67e15cc Bug 989945: increase logging for H264 OMX code r=jhlin 2014-05-24 18:28:02 -04:00
Randell Jesup
6ddd2f12d7 Bug 985253: Support H.264 RTP mode 1 support in webrtc signaling r=ehugg 2014-05-24 18:28:02 -04:00
Randell Jesup
f8592cfe05 Bug 985254: Add H264 codec-specific structure to carry negotiated data r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
146015cbf8 Bug 985254: Cleaup H264 packetization and jitter buffer r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
fef0238f25 Bug 985254: modify upstream h264 packetization patch to make it work r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
4a993a5cdc Bug 985254: review cleanups from H264 packetization patch r=pkerr 2014-05-24 18:28:01 -04:00
Randell Jesup
80e5a12d40 Bug 985254: H264 RTP packetization imported from upstream patchset #10 r=jesup 
https://webrtc-codereview.appspot.com/13399004/
 2014-05-24 18:28:00 -04:00
Randell Jesup
5d71bfa862 Bug 1004396: Make video codec default bitrates configurable for WebRTC r=ekr 2014-05-24 18:28:00 -04:00
Kyle Huey
8c5cca136c Bug 996133: Remove unnecessary NS_DISPATCH_NORMAL arguments to NS_DispatchToMainThread. r=ehsan 2014-05-23 12:53:17 -07:00
Jan-Ivar Bruaroey
98ea0e204d Bug 1013238 - Fix timer event crash on shutdown in recent PeerConnectionCtx change. r=jesup 2014-05-21 22:32:03 -04:00
Anders Lund
efc993ca04 Bug 942188 - Added parsing of ice-lite attribute and start ice checks as controlling if peer is ice-lite. r=abr 2014-05-16 01:32:00 -05:00
Byron Campen [:bwc]
743284e6c7 Bug 1013729 - Null check in case PushLayers failed when registering for the DTLS connection signal. r=jesup 2014-05-21 08:49:03 -07:00
Carsten "Tomcat" Book
a79c1ec7ff Backed out changeset 9b2588d41e3a (bug 969395) for bustage 2014-05-21 11:29:21 +02:00
Qiang Lu
c9bd29e3a3 Bug 969395 - Add stub library for accesing VP8 HW codec through android native mediacodec interface. r=rjesup 2014-05-21 10:14:31 +08:00
EKR
b9c43f7c18 Bug 1012999: When STUN global rate limit is exceeded, record this in telemetry. r=ekr 2014-05-19 19:16:38 -07:00
Jan-Ivar Bruaroey
5a14073109 Bug 970685 - Thread approach for WebRTC telemetry for jitter, packet-loss and RTT. r=jesup 2014-05-10 08:54:41 -04:00
John Lin
32e9769fb9 Bug 1011422 - Clear mOMXConfigured flag to correctly restart OMX H.264 encoder. r=jesup 2014-05-18 19:30:00 +02:00
Carsten "Tomcat" Book
fef547ad4e Backed out changeset 426b187eae45 (bug 1001422) wrong bugnumber in commit message 2014-05-19 11:44:00 +02:00
John Lin
3ba9c2eabc Bug 1001422 - Clear mOMXConfigured flag to correctly restart OMX H.264 encoder. r=jesup 2014-05-18 19:30:00 +02:00
John Lin
33b91e228f Bug 1010841 - Handle on-demand key frame request in OMX H.264 encoder. r=jesup 2014-05-16 01:56:00 -04:00
Randell Jesup
f884ae7641 Bug 1011214: Release OMX monitor when shutting down Encoder output drain thread r=jhlin 2014-05-16 04:37:08 -04:00
Randell Jesup
71910ce4dc Bug 981780: fix disable-webrtc r=glandium 2014-05-09 14:40:32 -04:00
Martin Thomson
5acda6875e Bug 966066 - Add principal observer to RTCPeerConnection. r=jib 2014-04-25 10:34:00 -04:00
Neil Rashbrook
5b3f3e053a Bug 514280 Only use nsCOMPtr for interfaces r=bsmedberg 2014-05-11 10:47:11 +01:00
Ryan VanderMeulen
893e7c7ecb Backed out changeset 047f98eef5cf (bug 1007196) for intermittent failures. 2014-05-09 14:13:21 -04:00
Ethan Hugg
6b867e1cb5 Bug 1007196 - Re-enable the Signaling unittests for Linux and OSX. r=ted 2014-05-07 13:04:34 -07:00
Chris Peterson
475d4e8367 Bug 990764 - Replace MOZ_ASSUME_UNREACHABLE in webrtc/signaling. r=jesup 2014-04-19 11:05:10 -07:00
Neil Rashbrook
fac8c73779 Backout of bug 514280 changeset c738f7348dea for build failure on a CLOSED TREE 2014-05-08 20:35:09 +01:00
Neil Rashbrook
5b1f7b4a77 Bug 514280 Only use nsCOMPtr for interfaces r=bsmedberg 2014-05-08 20:08:38 +01:00
Chris Peterson
b65637a65e Bug 1005784 - Fix -Wuninitialized warnings in webrtc/modules/audio_device/linux/. r=jesup 2014-05-05 23:38:04 -07:00
Byron Campen [:bwc]
2c84d5f0bc Bug 1002831 - Display remote and local SDP on about:webrtc. r=smaug, r=jib 2014-05-05 11:13:24 -07:00
Byron Campen [:bwc]
a622f4666a Bug 970734 - Part 2: Record final ICE/media stats when PeerConnections are closed, so they show up in about:webrtc. r=smaug, r=jib 2014-05-05 09:35:57 -07:00
Robert O'Callahan
e1bb2e935f Bug 1006248. Part 4: Use better #include paths for libstagefright headers in a couple of places. r=glandium 
--HG--
extra : rebase_source : e8c7e019b0bc5bf60081aad158a7d89fbb261e29
 2014-05-06 17:40:59 +12:00
Martin Thomson
5471fc97e1 Bug 1006112 - Fixing regressions in signaling_unittests. r=ekr 2014-05-05 14:19:00 +02:00
Martin Thomson
c3c2709899 Bug 942367 - Stream isolation for WebRTC r=bholley 2014-05-01 12:51:00 +02:00
Ethan Hugg
3ae788c1d5 Bug 1002890 - Signaling unittests no longer need exceptions to mainthread checks. r=jesup 2014-04-28 19:45:40 -07:00
Ethan Hugg
2e714cf592 Bug 819549 - Signaling unittests should dispatch to main thread when calling PC. r=jesup 2014-04-28 15:00:19 -07:00
Randell Jesup
95437d211a Bug 985253: Send rtcp-fb for all video codecs, and fix answer generation for H.264 for rtcp-fb r=ehugg 2014-04-30 18:18:35 -04:00
John Lin
7e66846d66 Bug 1002402: typo fix for adjusting SPS/PPS timestamps r=jesup 2014-04-30 01:20:41 -04:00
John Lin
4fd3ee35e5 Bug 1002402: (temporary) change SPS/PPS timestamps so webrtc jitter buffer won't drop them r=jesup 2014-04-29 13:25:40 -04:00
Ed Morley
e41a3e1c8a Merge mozilla-central and inbound 2014-04-29 18:23:29 +01:00
Randell Jesup
4cfc4d0e45 Bug 1002306: don't accidentally disable non-H264 codecs in the OMX code r=ekr 2014-04-28 19:52:16 -04:00
John Lin
b9eab230af Bug 911046 - Get graphic buffers of decoded frames through gonk native window callback. r=jesup 2014-04-27 21:07:00 -04:00
John Lin
4bf83010dc Bug 1002402 - Support RTP H.264 input data in WebRTC OMX decoder. r=jesup 2014-04-28 23:37:00 +02:00
Byron Campen [:bwc]
ad61a0ec58 Bug 1001959 - Give up references to NrIceMediaStream on STS instead of main. r=jib 2014-04-28 09:01:29 -07:00
Birunthan Mohanathas
5f1fde8824 Bug 900908 - Part 3: Change uses of numbered macros in nsIClassInfoImpl.h/nsISupportsImpl.h to the variadic variants. r=froydnj 2014-04-27 03:06:00 -04:00
Garvan Keeley
40aeac4872 Bug 1001708: Don't use ternary operator with class const statics r=jesup 2014-04-27 00:02:17 -04:00
Byron Campen [:bwc]
799148596a Bug 970690 - Part 2: Add basic telemetry for ICE. r=mt 2014-03-03 10:58:35 -08:00
Martin Thomson
6c6647cf1a Bug 1001539 - Fix compilation warning in ccsip_pmh.c. r=bwc 2014-04-25 10:58:00 -04:00
Paul Kerr [:pkerr]
8dc11ae04a Bug 970691 - Part 2: Restore digit stamping function to YuvStamper. r=jesup 
Refactor digit writing method to use the new internals. Allows digit string
to wrap through multiple lines in a small frame.
 2014-04-24 19:58:21 -07:00
Paul Kerr [:pkerr]
07fe6406b7 Bug 970691 - Part 1: Add timestamp to fake video. r=jesup 
Update YuvStamper utility. Add a CRC32 to the encoded
payload and have the decode method us this to verify reception.
Wrap encoded values across multiple lines in the frame buffer
when necessary. Use YuvStamper to encode a timestamp in each fake video frame.
Extract the value in VideoConduit to calculate the video latency
and add this to a running average latency when enabled via config.
 2014-03-22 16:35:43 -07:00
John
356b84852a Bug 999902 - Enable WebRTC OMX codec only when Android version >= 18. r=jesup 2014-04-23 02:59:00 +02:00
Wes Kocher
50c5a26e84 Backed out 2 changesets (bug 970691) for build bustage on a CLOSED TREE 
Backed out changeset 83f7aec5a083 (bug 970691)
Backed out changeset 94348d189ed5 (bug 970691)
 2014-04-23 18:26:05 -07:00
Paul Kerr [:pkerr]
c45ebab36f Bug 970691 - Part2: Restore digit stamping function to YuvStamper. r=jesup 
Refactor digit writing method to use the new internals. Allows digit string
to wrap through multiple lines in a small frame.
 2014-04-23 10:03:18 -07:00
Paul Kerr [:pkerr]
7db94a6847 Bug 970691 - Part 1: Add timestamp to fake video. r=jesup 
Update YuvStamper utility. Add a CRC32 to the encoded
payload and have the decode method us this to verify reception.
Wrap encoded values across multiple lines in the frame buffer
when necessary. Use YuvStamper to encode a timestamp in each fake video frame.
Extract the value in VideoConduit to calculate the video latency
and add this to a running average latency when enabled via config.
 2014-03-22 16:35:43 -07:00
John Lin
14da65f440 Bug 911046 - Part 6: Make H.264 preferred video codec when enabled in preferences. r=jesup, ekr 2014-04-21 23:44:00 +02:00
John Lin
9153e591e1 Bug 911046 - Part 5: Register H.264 external codec for B2G. r=jesup, ekr 2014-04-21 23:43:00 +02:00
John Lin
2a88e07bb8 Bug 911046 - Part 4: Add external H.264 HW video codec implementation for B2G. r=jesup 2014-04-21 23:42:00 +02:00
John Lin
704708f61b Bug 911046 - Part 2: Support 'handle-using' video frames for WebRTC on B2G. r=jesup, ekr 2014-04-21 23:41:00 +02:00
John Lin
7df65a157b Bug 911046 - Part 1: Support external codec in VideoConduit. r=jesup 2014-04-21 23:40:00 +02:00
Ethan Hugg
95043ad5a4 Bug 995380 - Signaling unittests should use the real main thread. r=jesup 2014-04-21 19:37:22 -07:00
Ryan VanderMeulen
ecb85b74fb Backed out changesets 1e581e74878d, 7d2138e87ca0, and 7cc66aee4341 (bug 942367) for B2G mochitest failures. 
CLOSED TREE
 2014-04-17 22:26:07 -04:00
Randell Jesup
dd18e038e2 Bug 996853: handle AUDIO_FORMAT_SILENCE in MediaPipeline and AudioSegment::WriteTo r=roc 2014-04-17 17:45:25 -04:00
Martin Thomson
33ae3b1f29 Bug 942367 - Part 3: Stream isolation for WebRTC. r=jib, r=bholley 2014-04-10 11:52:08 -07:00
Nils Ohlmeier [:drno]
969d5ff514 Bug 989936 - fire the onsignalingstatechanged event if close was called locally. r=jesup 2014-04-16 18:02:00 +02:00
Carsten "Tomcat" Book
e285a213e7 Backed out changeset e6c72bcaa64c (bug 942367) 2014-04-16 09:54:31 +02:00
Martin Thomson
d27d0a86fc Bug 942367 - Stream isolation for WebRTC. r=jib,bholley 2014-04-15 14:36:00 +02:00
Jonathan Watt
200e95e9eb Bug 996901 - Remove lots of gfxASurface.h and gfxImageSurface.h includes and forward declarations that are no longer needed. r=mattwoodrow 2014-04-16 01:41:40 +01:00
Randell Jesup
a4d6a74a90 Bug 996329: remove trailing space from m=application SDP lines r=ehugg 2014-04-15 14:00:59 -04:00
Nils Ohlmeier [:drno]
10db806555 Bug 993780 - Ignore calls to SetSignalingState_m once PC is in close. r=jib,rjesup 2014-04-10 14:55:00 +02:00
Nils Ohlmeier [:drno]
99bc6d8fc4 Bug 994999 - Rename IsClosed() to HasMedia() and let IsClosed() return SignalingState instead. r=jesup, r=bwc 2014-04-13 16:17:51 -04:00