admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
minyue-webrtc fae474c9cd Implement packet discard rate in NetEq. 
BUG=webrtc:7903

Change-Id: I819c9362671ca0b02c602d53e4dc39afdd8ec465
Reviewed-on: https://chromium-review.googlesource.com/555311
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18899}
2017-07-05 10:18:00 +00:00
..
api
Remove unused static VideoEncoder functions
2017-07-04 13:03:41 +00:00
audio
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
base
Delete SignalThread class.
2017-06-29 13:21:20 +00:00
call
Reland of Add received audio/video call duration metrics based on packets.
2017-07-04 11:05:06 +00:00
common_audio
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
common_video
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
examples
Fixes build issue based on usage of Android O specific API
2017-07-04 14:02:35 +00:00
logging
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
media
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
modules
Implement packet discard rate in NetEq.
2017-07-05 10:18:00 +00:00
ortc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
p2p
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
pc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
rtc_base
Reland "Update includes for webrtc/{base => rtc_base} rename (3/3)"
2017-06-30 17:45:21 +00:00
rtc_tools
fix comment length
2017-07-03 07:47:44 +00:00
sdk
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
stats
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
system_wrappers
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
test
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
tools
Revert of Remove webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2970743003/ )
2017-07-04 09:02:49 +00:00
video
Fix issue with zero rtt reports when using FlexFEC and add perf test.
2017-07-05 10:03:02 +00:00
voice_engine
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
.gitignore
…
BUILD.gn
Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )
2017-06-30 15:11:44 +00:00
codereview.settings
…
common_types.cc
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
common_types.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
config_unittest.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.cc
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
config.h
Support encrypted RTP extensions (RFC 6904)
2017-06-29 19:31:36 +00:00
DEPS
Move webrtc/{tools => rtc_tools}
2017-06-30 10:04:59 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
OWNERS
Don't add or rename files in webrtc/ and webrtc/api/ without a proper review
2017-05-08 13:01:54 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Make WARN_UNUSED_RESULT a no-op on gcc
2017-04-06 11:32:27 +00:00
video_receive_stream.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
video_send_stream.h
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
2017-07-01 14:48:18 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.