admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
asapersson 076c0118c5 Change unit of logged bitrate stats in bytes/s to bits/s. 
Multiplier added to ToString method in AggregatedStats.

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2535323003
Cr-Commit-Position: refs/heads/master@{#15330}
2016-11-30 13:17:21 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Disabled flaky P2PTestConductor tests on ASAN and MSAN.
2016-11-30 12:23:39 +00:00
audio
Relanding "Pass time constant to bwe smoothing filter."
2016-11-30 12:47:47 +00:00
base
Make sure GetLastError on a PlatformThread return an error that is relevant to the thread.
2016-11-30 12:53:14 +00:00
build
MB: Add swarming bots in the FYI waterfall and remove memcheck swarming.
2016-11-30 10:35:47 +00:00
call
Change unit of logged bitrate stats in bytes/s to bits/s.
2016-11-30 13:17:21 +00:00
common_audio
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
common_video
Delete deprecated versions of Copy, ScaleFrom and CropAndScaleFrom.
2016-11-30 09:52:12 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
RtpDataEngine, FindCodecByName: Don't reassign codecs
2016-11-30 12:52:10 +00:00
modules
Greatly reduce number of level controller tests.
2016-11-30 13:04:47 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
sdk
Create webrtc/sdk/android folder
2016-11-30 12:51:05 +00:00
stats
Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
2016-11-30 09:50:36 +00:00
system_wrappers
Use swap instead of copy in RtcHistogram::GetAndReset.
2016-11-30 08:29:17 +00:00
test
Relanding "Pass time constant to bwe smoothing filter."
2016-11-30 12:47:47 +00:00
tools
Relanding "Pass time constant to bwe smoothing filter."
2016-11-30 12:47:47 +00:00
video
Change unit of logged bitrate stats in bytes/s to bits/s.
2016-11-30 13:17:21 +00:00
voice_engine
Relanding "Pass time constant to bwe smoothing filter."
2016-11-30 12:47:47 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
RTC_[D]CHECK_op: Remove superfluous casts
2016-11-28 23:59:03 +00:00
common_types.h
Replace some asserts with DCHECKs
2016-11-29 13:30:47 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Implement periodic bandwidth probing in application-limited region.
2016-11-28 21:11:24 +00:00
voice_engine_configurations.h
Remove 3 defines in voice_engine_configurations.h
2016-11-30 08:12:57 +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.