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Sebastian Jansson 9e3f1e4ca2 Fixed a miscalculation of sent bitrate caused by mixup of time units 
Bug: webrtc:7949
Change-Id: Ia57fdd3d1de0952b80e77c30b0a6cfe44515eff2
Reviewed-on: https://chromium-review.googlesource.com/561460
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18917}
2017-07-06 15:22:58 +00:00
..
api
Report timing frames info in GetStats.
2017-07-06 10:06:50 +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
Moving asm code out of common_audio_c sources list
2017-07-06 14:44:14 +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
Update screen simulcast config
2017-07-06 12:05:50 +00:00
modules
Fixed a miscalculation of sent bitrate caused by mixup of time units
2017-07-06 15:22:58 +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
Report timing frames info in GetStats.
2017-07-06 10:06:50 +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
Fix null ref in NetworkMonitorAutoDetect if Connectivity Manager service is unavailable
2017-07-05 16:55:09 +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
Remove MAIN_NIB_FILE from Info.plist because the substitution is broken
2017-07-06 14:09:57 +00:00
video
Implement RTP keepalive in native stack.
2017-07-06 11:38:06 +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
Implement RTP keepalive in native stack.
2017-07-06 11:38:06 +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
…
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
Report timing frames info in GetStats.
2017-07-06 10:06:50 +00:00
video_send_stream.h
Implement RTP keepalive in native stack.
2017-07-06 11:38:06 +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.