admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
peah f3669661bd Removed the issue with the leading semicolon in the audio 
processing module experiment description that was present
when AEC3 was not activated and when RefinedAdaptiveFilter
was activated.

BUG=webrtc:5778, webrtc:5777

Review URL: https://codereview.webrtc.org/1899123002

Cr-Commit-Position: refs/heads/master@{#12424}
2016-04-19 10:40:15 +00:00
..
api
Fix crash when receiving a texture frame with rotation bit set.
2016-04-18 19:40:28 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Roll chromium_revision 212f976fef..61ed337cfe (387882:388120)
2016-04-19 10:03:31 +00:00
build
Fix coverage build.
2016-04-19 06:08:15 +00:00
call
Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot.
2016-04-18 23:19:02 +00:00
common_audio
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
common_video
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
examples
Use the new appr.tc URL
2016-04-15 14:16:19 +00:00
libjingle
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
media
Remove use_openssl from webrtc
2016-04-18 18:13:17 +00:00
modules
Removed the issue with the leading semicolon in the audio
2016-04-19 10:40:15 +00:00
p2p
Make ReliableQuicStream::Write use base::StringPiece, not std::string
2016-04-14 18:12:51 +00:00
pc
Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ )
2016-04-13 17:48:32 +00:00
system_wrappers
Add histogram stats for average QP per frame for VP9 (for sent video streams):
2016-04-18 09:58:52 +00:00
test
Added new VideoFrameBuffer methods Data[YUV]() etc.
2016-04-18 12:34:45 +00:00
tools
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
voice_engine
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +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
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +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
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Added support in the AEC for refined filter adaptation.
2016-04-15 18:23:36 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +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
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +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
Delete method webrtc::VideoFrame::native_handle.
2016-04-15 10:43:45 +00:00
video_receive_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
video_send_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
webrtc_examples.gyp
Add test runner scripts for instrumentation tests
2016-04-08 13:54:32 +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
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc.gyp
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +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.