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Jonathan Yu 36344a0c9b Fix incorrect memset on muted frames. 
Broken by https://codereview.webrtc.org/2750783004/. Since samples are
two bytes each, only half of the buffer was being zeroed, leading to
garbage noise.

BUG=webrtc:7885,webrtc:7343

Change-Id: I46ecf90258b681ccdebbcfadd2e84ac6abadc9fe
Reviewed-on: https://chromium-review.googlesource.com/593092
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19194}
2017-07-31 22:18:41 +00:00
..
api
G711 implementation of the Audio{En,De}coderFactoryTemplate APIs
2017-07-31 18:34:57 +00:00
audio
Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
2017-07-26 14:28:51 +00:00
base
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
call
Only one implementation of MockRtpPacketSink once
2017-07-28 17:05:45 +00:00
common_audio
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
common_video
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
examples
Update native plugin dll for turn servers and video.
2017-07-25 23:04:31 +00:00
logging
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
media
Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
2017-07-26 09:09:44 +00:00
modules
Fix incorrect memset on muted frames.
2017-07-31 22:18:41 +00:00
ortc
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
p2p
Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
2017-07-26 23:09:33 +00:00
pc
Make RTCStatsReport::ToString() return JSON-parseable string.
2017-07-28 14:29:12 +00:00
rtc_base
If adapter type is unknown and interface name is "ipsec", treat as VPN.
2017-07-27 22:05:29 +00:00
rtc_tools
Print general usage information for event_log_analyzer
2017-07-20 15:05:09 +00:00
sdk
ObjC RTCEAGLVideoVideo: Check GL context is non-nil in constructor
2017-07-31 16:11:46 +00:00
stats
Make RTCStatsReport::ToString() return JSON-parseable string.
2017-07-28 14:29:12 +00:00
system_wrappers
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
test
Rename isolated_output to test_output and add a method to get the test_output directory.
2017-07-25 14:31:18 +00:00
video
Only one implementation of MockRtpPacketSink once
2017-07-28 17:05:45 +00:00
voice_engine
Move total audio energy and duration tracking to AudioLevel and protect with existing critial section.
2017-07-20 16:57:42 +00:00
.gitignore
…
BUILD.gn
Remove remains of webrtc/base
2017-07-19 17:40:47 +00:00
codereview.settings
…
common_types.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
common_types.h
Add parsing/serializing for MID RTP header extension.
2017-07-21 17:33:25 +00:00
config_unittest.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.cc
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
config.h
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +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
Report interframe delay sum in old GetStats
2017-07-07 08:26:24 +00:00
video_send_stream.h
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
2017-07-11 10:44:17 +00:00
webrtc.gni
Turning on Opus 120ms frame length switch.
2017-07-27 17:23:35 +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.