Henrik Lundin 1ff41eb784 Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"
This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1.

Reason for revert: Breaks downstream tests.

Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
> 
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
> 
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
> 
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
>   no longer be reached.
> 
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}

TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org

Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:44 +00:00
2018-06-19 14:00:39 +00:00
2018-06-20 07:04:09 +00:00
2018-05-30 08:30:00 +00:00
2018-06-19 14:00:39 +00:00
2018-05-15 16:41:02 +00:00
2018-06-19 14:00:39 +00:00
2018-06-19 14:00:39 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2018-06-20 12:39:11 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2018-06-08 10:31:38 +00:00
2018-02-23 10:34:16 +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.

Development

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

Authoritative list of directories that contain the native API header files.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%