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webrtc_m130/webrtc
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Fredrik Solenberg ea07373a2e Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors. 
BUG=webrtc:5268,webrtc:5273
TESTED=Fixed issues reported by:
find webrtc/audio -type f -name *.cc -o -name *.h | xargs cpplint.py
find webrtc/call -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1483323002 .

Cr-Commit-Position: refs/heads/master@{#10853}
2015-12-01 10:26:46 +00:00
..
audio
Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
2015-12-01 10:26:46 +00:00
base
Add helper KeepRefUntilDone.
2015-12-01 06:15:53 +00:00
build
iOS: Set enable_protobuf=1 by default.
2015-11-20 08:37:50 +00:00
call
Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
2015-12-01 10:26:46 +00:00
common_audio
Add aecdump support to audioproc_f
2015-11-20 08:11:58 +00:00
common_video
Add helper KeepRefUntilDone.
2015-12-01 06:15:53 +00:00
examples
Add simple end to end test for video capture and encode using textures.
2015-12-01 06:59:42 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
Revert of Add _decoder CHECK to VCMGenericDecoder constructor. (patchset #2 id:20001 of https://codereview.webrtc.org/1485713002/ )
2015-12-01 07:14:37 +00:00
p2p
Initialize type_preference_ in TestPort.
2015-11-30 17:17:07 +00:00
sound
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
system_wrappers
Introduce helper class NtpTime
2015-11-26 17:01:18 +00:00
test
Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.
2015-11-27 18:46:47 +00:00
tools
Remove webrtc/test/channel_transport/include
2015-11-11 09:34:35 +00:00
video
Remove duplicate decoders in BitrateEstimatorTest.
2015-11-27 15:35:14 +00:00
video_engine
Remove RegisterExternal{De,En}coder error codes.
2015-11-27 13:09:14 +00:00
voice_engine
Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream.
2015-11-27 18:46:47 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Re-Land: Implement AudioReceiveStream::GetStats().
2015-10-22 08:49:39 +00:00
audio_send_stream.h
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
audio_state.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
BUILD.gn
Add fuzzing of VP8 QP parsing.
2015-11-26 13:52:28 +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
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +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
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
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
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +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 DecodedImageCallback::Decoded() function with custom decode time value.
2015-11-10 13:00:45 +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
Require negotiation to send transport cc feedback over RTCP.
2015-11-21 02:05:53 +00:00
video_renderer.h
Remove frame time scheduing in IncomingVideoStream
2015-11-25 02:08:03 +00:00
video_send_stream.h
Reland of Collecting encode_time_ms for each frame (patchset #1 id:1 of https://codereview.webrtc.org/1383283005/ )
2015-10-07 09:43:25 +00:00
webrtc_examples.gyp
Remove webrtc/examples/android/media_demo.
2015-11-11 12:11:21 +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
Revert of Created a test that reports the statistics for the duration of APM stream processing API calls. (patchset #15 id:280001 of https://codereview.webrtc.org/1436553004/ )
2015-11-25 20:27:46 +00:00
webrtc.gyp
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +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.