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danilchap 6db6cdc604 [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs 
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1513303003

Cr-Commit-Position: refs/heads/master@{#11025}
2015-12-15 10:54:50 +00:00
..
api
Update API for Objective-C RTCIceServer
2015-12-14 03:58:19 +00:00
audio
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +00:00
base
Add "x"s in the end of a stripped IPv6 address string.
2015-12-14 10:07:08 +00:00
build
Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
2015-12-14 12:31:22 +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
Reland of Made EglBase an abstract class and cleaned up. (patchset #1 id:1 of https://codereview.webrtc.org/1522073002/ )
2015-12-15 10:48:13 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
p2p
Handle Turn error response to RefreshRequest, CreatePermissionRequest, and ChanelBindRequest
2015-12-11 23:16:58 +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
Remove unused and rarely used LOG_ macros.
2015-12-14 16:22:21 +00:00
test
Base webrtc fuzzers on a template.
2015-12-15 09:46:27 +00:00
tools
Enable cpplint for more webrtc subfolders and fix all uncovered cpplint errors.
2015-12-10 11:11:48 +00:00
video
Wire up bandwidth limitation info to GetStats and adapt_reason.
2015-12-14 10:08:19 +00:00
voice_engine
Correcting the check for the return code produced by
2015-12-14 07:02:40 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Support for unmixed remote audio into tracks.
2015-12-12 00:37:14 +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
…
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
…
common.h
…
config.cc
…
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
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
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
Update WARN_UNUSED_RESULT macro to match Chromium's version.
2015-12-12 13:47:24 +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
…
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
Wire up bandwidth limitation info to GetStats and adapt_reason.
2015-12-14 10:08:19 +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
…
webrtc_tests.gypi
Update API for Objective-C RTCIceServer
2015-12-14 03:58:19 +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.