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webrtc_m130/webrtc
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Peter Boström 43e4e23eeb Remove thread-id wraparounds in event tracing. 
Prints tid as a DWORD on Windows and an int on other systems.

BUG=
R=thakis@chromium.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1522483002 .

Cr-Commit-Position: refs/heads/master@{#10992}
2015-12-11 19:29:47 +00:00
..
audio
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
base
Remove thread-id wraparounds in event tracing.
2015-12-11 19:29:47 +00:00
build
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +00:00
call
Replace EventWrapper in video/, test/ and call/.
2015-12-10 12:03:00 +00:00
common_audio
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
common_video
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
examples
Add aecdump support to AppRTCDemo
2015-12-11 07:28:50 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
Add FEC producer fuzzing and a unittest for one of the issues found.
2015-12-11 17:25:56 +00:00
p2p
Add logs when stun or turn host lookup is completed.
2015-12-07 20:06:27 +00:00
sound
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
system_wrappers
Nuke TickTime::UseFakeClock.
2015-12-09 10:21:09 +00:00
test
Add FEC producer fuzzing and a unittest for one of the issues found.
2015-12-11 17:25:56 +00:00
tools
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
video
modules/rtp_rtcp/include folder cleared of lint warnings
2015-12-10 17:52:01 +00:00
voice_engine
modules/rtp_rtcp/include folder cleared of lint warnings
2015-12-10 17:52:01 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Create fuzzer tests for audio decoders
2015-12-08 10:27:34 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Revert of Make overuse estimator one dimensional. (patchset #5 id:80001 of https://codereview.webrtc.org/1376423002/ )
2015-11-27 09:02:35 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Add RTC_ prefix to contructormagic macros.
2015-09-16 12:37:52 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
…
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Use suffixed {uint,int}{8,16,32,64}_t types.
2015-10-07 10:23:32 +00:00
video_decoder.h
Add VideoCodec::PreferDecodeLate
2015-12-10 17:27:45 +00:00
video_encoder.h
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
video_receive_stream.h
Add VideoCodec::PreferDecodeLate
2015-12-10 17:27:45 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Adding reduced size RTCP configuration down to the video stream level.
2015-12-09 20:37:59 +00:00
webrtc_examples.gyp
Fix Mac ObjC PeerConnection API compilation.
2015-12-03 00:42:41 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.