henrika 1d4db392c7 Revert of If SRTP sessions exist, don't create new ones when applying answer. (patchset #1 id:1 of https://codereview.webrtc.org/3019443002/ )
Reason for revert:
Speculative revert since all Android bots on WebRTC FYI started to fail when this CL landed.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29

Original issue's description:
> If SRTP sessions exist, don't create new ones when applying answer.
>
> Instead, call the "Update" methods of SrtpSession, which will just call
> srtp_update, instead of wiping out the session state completely.
>
> This was causing decryption to stop working when subsequent
> offers/answers are applied. We don't know enough about SRTP to
> understand the root cause, and I wasn't able to write an integration
> test that reproduces the issue... But at least this fixes the bug that
> can be reproduced reliably using Hangouts.
>
> BUG=webrtc:8251
>
> Review-Url: https://codereview.webrtc.org/3019443002
> Cr-Commit-Position: refs/heads/master@{#19874}
> Committed: https://webrtc.googlesource.com/src/+/5ada7acf603e90e71990e9d4ff8f49389f24958c

TBR=zhihuang@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:8251
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/3017543002
Cr-Commit-Position: refs/heads/master@{#19882}
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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.

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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