sprang 982bf89444 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ )
Reason for revert:
Speculative revert.
Intermittent memory access errors suspected to be caused by this cl.

See for instance https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/8018

UNADDRESSABLE ACCESS of freed memory: reading 0x0331d330-0x0331d334 4 byte(s)
# 0 webrtc::voe::RtcpRttStatsProxy::LastProcessedRtt
# 1 webrtc::ModuleRtpRtcpImpl::Process

Original issue's description:
> Add RtcpRttStats to AudioStream
>
> BUG=webrtc:6508
>
> Committed: https://crrev.com/e0729c56d35acfaf9738fdb32c6508cd78eaf089
> Cr-Commit-Position: refs/heads/master@{#14595}

TBR=stefan@webrtc.org,minyue@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6508

Review-Url: https://codereview.webrtc.org/2415943002
Cr-Commit-Position: refs/heads/master@{#14631}
2016-10-13 13:23:18 +00:00
2016-08-11 14:01:03 +00:00
2016-06-14 09:39:40 +00:00
2015-09-11 09:04:09 +00:00
2016-10-05 12:06:41 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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