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db7543809e3e8a52c80a81aa49d0bc34a9670c00
platform-external-webrtc/webrtc
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stefan db7543809e Tune BWE to be more sensitive on low capacity networks. 
Also lowering the min bitrate for simulations to be able to better capture this issue in the BweFeedbackTest.Choke200kbps30kbps200kbps performance test.

BUG=webrtc:6105
NOTRY=true
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2201093006
Cr-Commit-Position: refs/heads/master@{#13639}
2016-08-04 13:42:14 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
audio
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 12:28:28 +00:00
base
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
build
Don't build VP9 for iOS framework.
2016-08-02 16:38:40 +00:00
call
Log current time if we stop logging before stop_time_.
2016-07-29 19:38:45 +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
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.
2016-08-04 12:28:28 +00:00
modules
Tune BWE to be more sensitive on low capacity networks.
2016-08-04 13:42:14 +00:00
p2p
Prepare for ICE renomination.
2016-08-04 02:50:53 +00:00
pc
Add support for GCM cipher suites from RFC 7714.
2016-08-04 12:20:38 +00:00
sdk
Disable encoder scaling on iPhone4S.
2016-08-03 19:57:18 +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
Remove a few meaningless TODOs
2016-08-03 13:38:20 +00:00
tools
Add plot of network delay change computed based on transport feedback.
2016-08-02 14:22:23 +00:00
video
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
voice_engine
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +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
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
GN: Create target for video_capture_tests on Linux.
2016-08-04 13:29:31 +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
…
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
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
…
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
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
Enable socketservers unittests on windows.
2016-08-03 13:14:48 +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.