henrika 5641fbb5ec Add support for saving local audio input to file in AppRTCMobile
Uses new WebRtcAudioRecordSamplesReadyCallback which was added recently in
https://webrtc-review.googlesource.com/c/src/+/49981.

This CL:
- Serves as a test of new WebRtcAudioRecordSamplesReadyCallback.
- Useful for debugging purposes since it records the most native raw audio.

Bug: None
Change-Id: I57375cbf237c171e045b0bdb05f7ae1401930fbc
Reviewed-on: https://webrtc-review.googlesource.com/53120
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22128}
2018-02-21 14:09:56 +00:00
.gn
2018-02-19 15:07:45 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2018-01-29 11:18:18 +00:00
2017-09-15 04:25:06 +00:00
2018-02-19 14:35:45 +00:00
2018-02-17 22:47:22 +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
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