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kjellander 0ae7878bbc MB: Add Windows configurations 
BUG=589510, 636414
NOTRY=True

Review-Url: https://codereview.webrtc.org/2235833002
Cr-Commit-Position: refs/heads/master@{#13721}
2016-08-11 07:49:55 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Add libjingle_peerconnection_unittest to GN targets.
2016-08-10 10:10:53 +00:00
audio
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 12:28:28 +00:00
base
Add magjed@webrtc.org as owner of webrtc/base/java/
2016-08-10 11:49:35 +00:00
build
MB: Add Windows configurations
2016-08-11 07:49:55 +00:00
call
Generate random rtp packets with RtpPacketToSend instead of RtpSender
2016-08-10 11:23:38 +00:00
common_audio
Pull out the PostFilter to its own NonlinearBeamformer API
2016-07-02 00:19:32 +00:00
common_video
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed.
2016-07-15 06:36:00 +00:00
examples
Remove LooperExecutor and replace it with built-in HandlerThread.
2016-08-09 09:03:41 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
2016-08-05 18:14:54 +00:00
modules
Probe bitrate estimator correction.
2016-08-10 17:00:52 +00:00
p2p
Do not create incompatible TurnPort if the server address family is known.
2016-08-10 17:33:17 +00:00
pc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
sdk
iOS render: Handle frame rotation in OpenGL
2016-08-10 14:58:35 +00:00
system_wrappers
UMA log for audio_device Init and Start(Playout|Recording). Make Init return a more specific error code, if possible.
2016-07-01 11:35:31 +00:00
test
Style updates to ProducerFec/FecReceiver.
2016-08-10 07:51:59 +00:00
tools
Fix tools_unittests in GN.
2016-08-10 14:47:20 +00:00
video
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
voice_engine
Broke out 'level_indicator' of the voice_engine GN target. This is
2016-08-10 12:01:59 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +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
GN: Add targets for AppRTCDemo on Mac+iOS
2016-08-08 04:37:56 +00:00
call.h
Fix bug where min transmit bitrate wasn't working
2016-07-06 07:54:35 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +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
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +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
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Add decoder-specific settings with proper lifetime.
2016-08-05 16:19:33 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
Reland Enable socketservers unittests on windows.
2016-08-05 12:55:49 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.