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153,018 Commits 1 Branch 166 Tags 1.2 GiB
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163 Commits

Author SHA1 Message Date
Randell Jesup
cd3e87b2df Bug 932215 - Lazily allocate log buffers for webrtc (4MB saving). r=jib 2013-10-30 12:13:07 -04:00
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
Jaroslav Kopecký
8dd49a58dd Bug 931590 - Pass proper directory when building --with-system-nspr r=jesup 2013-10-27 19:43:04 -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
Brian O'Keefe
4c98f61956 Bug 928709 - Convert chromium-config.mk to mozbuild, r=mshal 2013-10-02 13:17:55 -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
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
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
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
Shih-Chiang Chien
68e8048a99 Bug 918523 - Prevent rec_queue overrun. r=jesup 2013-09-28 09:12:39 +08:00
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
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
Randell Jesup
5c73c402f3 Bug 904784: use a separate critical section for the recording callback r=mwu 2013-09-07 23:42:01 -04:00
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
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
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
Mike Hommey
474ed6071f Bug 907473 - Handle generator_flags gracefully in gyp. r=gps 2013-08-21 09:37:45 +09:00
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
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
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
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
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
Ehsan Akhgari
9854ac6166 Bug 872127 - Part 2: Replace mozilla/StandardInteger.h with stdint.h; r=Waldo,ted 2013-07-30 10:25:31 -04:00
Daniel Holbert
a07d665490 Bug 899240: Fix typo in close-comment syntax, for commented-out line in neteq_defines.h. r=jesup 2013-07-29 14:21:20 -07:00
Randell Jesup
1a83b7de43 Bug 876878: Avoid null deref if camera doesn't update framelist ptr r=bas 2013-07-25 15:30:46 -04:00
Randell Jesup
da96a161fb Bug 880879: re-land changes lost in the original merge of bug 880879 rs=jesup,derf 
Bug 832579 (RTCP NACK doesn't work) plus one small mis-applied diff in alsa that lost the GUID
values for recording devices
 2013-07-25 07:52:58 -04:00
Randell Jesup
8648333afd Bug 886886: Remove 44100->44000 kludges r=derf 2013-07-21 03:47:40 -04:00
Randell Jesup
b4d4d18ecf Bug 886886: replace fixed-ratio capture resampler in webrtc with speex resample r=derf,jmspeex 2013-07-21 03:47:24 -04:00
Gian-Carlo Pascutto
07c7eb48ea Bug 885031 - Don't try to get information about the camera on Froyo. r=blassey 2013-07-15 11:21:15 +02:00
Gian-Carlo Pascutto
4e0a616387 Bug 891158 - Listen to onOrientationChanged instead of onConfigurationChanged. r=blassey 2013-07-11 17:17:37 +02:00
Jan Beich
96518f0ec3 Bug 892102 - Explicitly include stdlib.h for abs(). r=jesup 2013-07-11 10:43:35 -04:00
Randell Jesup
5ba61d4466 bug 880879: Rollup of changes previously applied to media/webrtc/trunk/webrtc rs=derf f=gcp r=jesup 2013-07-10 03:12:59 -04:00
Randell Jesup
10378680df bug 880879: Webrtc updated to 4180; pull made on Wed Jan 05 04:11:00 EDT 2013 rs=derf 
--HG--
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/OWNERS => media/webrtc/trunk/webrtc/modules/video_coding/OWNERS
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/temporal_layers.cc => media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/temporal_layers_unittest.cc => media/webrtc/trunk/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/Android.mk => media/webrtc/trunk/webrtc/modules/video_coding/utility/Android.mk
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/exp_filter.cc => media/webrtc/trunk/webrtc/modules/video_coding/utility/exp_filter.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_dropper.cc => media/webrtc/trunk/webrtc/modules/video_coding/utility/frame_dropper.cc
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/exp_filter.h => media/webrtc/trunk/webrtc/modules/video_coding/utility/include/exp_filter.h
rename : media/webrtc/trunk/webrtc/modules/video_coding/main/source/frame_dropper.h => media/webrtc/trunk/webrtc/modules/video_coding/utility/include/frame_dropper.h
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/traffic_control_windows.cc => media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.cc
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/traffic_control_windows.h => media/webrtc/trunk/webrtc/test/channel_transport/traffic_control_win.h
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/udp_socket2_manager_windows.cc => media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.cc
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/udp_socket2_manager_windows.h => media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_manager_win.h
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/udp_socket2_windows.cc => media/webrtc/trunk/webrtc/test/channel_transport/udp_socket2_win.cc
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/udp_socket_manager_posix.cc => media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_manager_posix.cc
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/udp_socket_posix.cc => media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_posix.cc
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/udp_socket_wrapper.cc => media/webrtc/trunk/webrtc/test/channel_transport/udp_socket_wrapper.cc
rename : media/webrtc/trunk/webrtc/modules/udp_transport/source/udp_transport_impl.cc => media/webrtc/trunk/webrtc/test/channel_transport/udp_transport_impl.cc
 2013-06-11 21:08:23 -04:00