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webrtc_m130/webrtc
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tommi 31fc21f454 Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/ 
Also remove mischievous tab character!
This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler.

BUG=

Review URL: https://codereview.webrtc.org/1607353002

Cr-Commit-Position: refs/heads/master@{#11346}
2016-01-21 18:37:44 +00:00
..
api
Update API for Objective-C RTCPeerConnectionFactory.
2016-01-20 21:40:35 +00:00
audio
Enable transport seq num extension on receive channel to suppress log warning.
2016-01-21 14:32:48 +00:00
base
Make rtc::CriticalSection lockable from f() const.
2016-01-21 15:57:03 +00:00
build
rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni.
2016-01-21 11:34:47 +00:00
call
Fixes a bug which incorrectly logs incoming RTCP as outgoing.
2016-01-21 13:42:10 +00:00
common_audio
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common_video
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +00:00
examples
Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
2016-01-15 22:49:15 +00:00
libjingle
Remove DISABLED_ON_ macros.
2016-01-04 21:44:16 +00:00
modules
Fix probing breakage with send-side BWE introduced by r11322.
2016-01-21 16:55:08 +00:00
p2p
Remove two dead 'using' instances.
2016-01-18 23:24:36 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Remove ConditionVariableWrapper.
2016-01-20 12:36:42 +00:00
test
Add stefan@webrtc.org to webrtc/test/OWNERS.
2016-01-21 14:33:03 +00:00
tools
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
video
Fixes a bug which incorrectly logs incoming RTCP as outgoing.
2016-01-21 13:42:10 +00:00
voice_engine
Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/
2016-01-21 18:37:44 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Storing raw audio sink for default audio track.
2016-01-15 17:20:08 +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
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +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
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +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
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +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
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 attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +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
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +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
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +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.