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webrtc_m130/webrtc
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lliuu c4b126d70e Fix audio device excessive logging on Windows 
In the windows audio device implementation, play block size should be
number of samples per 10ms times the number of channels. Meanwhile
RequestPlayoutData is expecting number of samples per channel in a
block, and we should pass in the per channel number here to avoid debug
check.

BUG=7627

Review-Url: https://codereview.webrtc.org/2876593003
Cr-Commit-Position: refs/heads/master@{#18088}
2017-05-11 01:08:24 +00:00
..
api
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
audio
Fixing video loopback test with encoder factory.
2017-05-10 11:06:11 +00:00
base
Delete left-over declaration of AdjustCurrentProcessPrivilege.
2017-05-10 13:04:45 +00:00
call
Add untracked headers in modules/rtp_rtcp
2017-05-10 15:41:13 +00:00
common_audio
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
common_video
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
examples
Test for Gradle project generation.
2017-05-09 09:10:57 +00:00
logging
Create tool to print statistics about the file size usage of an RTC event log.
2017-04-28 14:48:17 +00:00
media
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
modules
Fix audio device excessive logging on Windows
2017-05-11 01:08:24 +00:00
ortc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
p2p
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
pc
Delete method MessageQueue::set_socketserver
2017-05-08 12:25:41 +00:00
sdk
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
2017-04-28 12:24:50 +00:00
test
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
tools
Moving scripts to download and build apprtc/collider.
2017-05-09 07:13:47 +00:00
video
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
voice_engine
Resolves race between Channel::ProcessAndEncodeAudio() and Channel::StopSend()
2017-05-03 15:30:15 +00:00
.gitignore
…
BUILD.gn
Remove config.h from //webrtc:webrtc target.
2017-05-03 12:28:46 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Add read support of RtpStreamId/RepairedRtpStreamId header extensions.
2017-04-19 09:59:48 +00:00
common_types.h
Delete RawVideoType enum, use the VideoType enum instead.
2017-04-28 14:18:05 +00:00
config.cc
Fix default extension id for video content type extension
2017-04-26 09:35:31 +00:00
config.h
Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )
2017-04-11 17:34:31 +00:00
DEPS
Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )
2017-04-05 10:02:20 +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_frame.h
Reland of Delete deprecated and transitional stuff related to video frame refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/2853383005/ )
2017-05-10 12:25:59 +00:00
video_receive_stream.h
Delete VieRemb class, move functionality to PacketRouter.
2017-04-19 06:38:35 +00:00
video_send_stream.h
Add number of quality adapt changes to VideoSendStream stats.
2017-04-04 12:51:49 +00:00
webrtc.gni
Remove the expat and jsoncpp dependencies to reduce binary size.
2017-04-24 19:05:06 +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.