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2181078ca8d4130bc44c7f49b1ca37f684b72c4d
platform-external-webrtc/webrtc
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zijiehe 2181078ca8 [Chromoting] Remove screen saver logic out of ScreenCapturer implementations 
After change https://codereview.chromium.org/2080723008/, chromoting hosts are
using device::PowerSaveBlocker to block screen saver and suspend. So similar
logic in ScreenCapturer are not useful, and should be removed.

BUG=626839

Review-Url: https://codereview.webrtc.org/2155813003
Cr-Commit-Position: refs/heads/master@{#13491}
2016-07-16 00:05:17 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Fix VideoFrame inclusion in mediastreaminterface.h
2016-07-14 14:14:31 +00:00
audio
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
base
Use correct case for including a file.
2016-07-15 14:21:28 +00:00
build
Revert of Fix inconsistent setting of the rtc_include_unittests flag. (patchset #1 id:1 of https://codereview.webrtc.org/2154693002/ )
2016-07-15 11:13:28 +00:00
call
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_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
Camera2 in AppRTC Android Demo.
2016-07-06 07:58:35 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed.
2016-07-15 06:36:00 +00:00
modules
[Chromoting] Remove screen saver logic out of ScreenCapturer implementations
2016-07-16 00:05:17 +00:00
p2p
Adding more logging to BasicPortAllocator.
2016-07-15 21:42:27 +00:00
pc
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
sdk
AVFoundationVideoCapturer: Output native frames instead of I420 frames
2016-07-14 15:12:23 +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
Add a fuzzer for the pseudoTCP implementation used by chromoting. This
2016-07-07 12:52:29 +00:00
tools
Initial version of the local visualization tool for WebrtcEventLogs.
2016-07-13 13:44:48 +00:00
video
Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed.
2016-07-15 06:36:00 +00:00
voice_engine
Added voice_engine_unittests and voe_auto_test targets to GN.
2016-07-08 09:13:02 +00:00
.gitignore
…
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
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
Move WEBRTC_BUILD_LIBEVENT definition to base/BUILD.gn and base/base.gyp. This use all_dependent_configs (GN) / all_dependent_settings (GYP) to export WEBRTC_BUILD_LIBEVENT to all dependent targets transitively. This fixes a problem with dependent projects in Chromium that indirectly include task_queue.h
2016-07-15 06:25:54 +00:00
call.h
Fix bug where min transmit bitrate wasn't working
2016-07-06 07:54:35 +00:00
codereview.settings
…
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
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +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
…
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +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
…
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 of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +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.