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henrik.lundin bf7c620280 Fix lint errors in audio_coding_module_unittest.cc 
BUG=None
NOTRY=True

Review-Url: https://codereview.webrtc.org/2389263003
Cr-Commit-Position: refs/heads/master@{#14544}
2016-10-06 08:53:39 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Add onCameraDisconnected camera event.
2016-10-06 07:59:18 +00:00
audio
Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.
2016-10-04 09:45:54 +00:00
base
Revert of Delete transformadapter.cc and transformadapter.h. (patchset #2 id:20001 of https://codereview.webrtc.org/2368243002/ )
2016-10-06 08:37:42 +00:00
build
Add iOS static library GN build script.
2016-10-05 20:16:07 +00:00
call
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
common_audio
Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere
2016-10-04 20:47:02 +00:00
common_video
Support for parsing CABAC coded bitstreams
2016-10-05 09:16:28 +00:00
examples
Speculatively increase timeouts for DirectRTCClientTest to reduce flakiness.
2016-10-06 07:34:47 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
logging
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
media
Delete unused code, cpuinfo.{h,cc}.
2016-10-06 06:44:45 +00:00
modules
Fix lint errors in audio_coding_module_unittest.cc
2016-10-06 08:53:39 +00:00
p2p
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
pc
Delete unused class cricket::MediaSinkInterface, and mediasink.h.
2016-10-04 12:33:53 +00:00
sdk
Add iOS static library GN build script.
2016-10-05 20:16:07 +00:00
stats
RTCStatsMember<bool> and RTCStatsMember<std::vector<bool>> added.
2016-10-04 21:37:15 +00:00
system_wrappers
Fixed flaky clock_unittest by using relative comparison.
2016-10-05 08:55:10 +00:00
test
Harmonize using-declarations in FEC header formatter fuzzers.
2016-10-05 14:35:01 +00:00
tools
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
video
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
voice_engine
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
.gitignore
…
BUILD.gn
Add UMA metrics for ICE regathering reasons.
2016-10-05 18:47:39 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
…
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
config.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
DEPS
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Delete webrtc::VideoFrame::CreateEmptyFrame.
2016-10-05 06:27:37 +00:00
video_receive_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
video_send_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +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.