erikvarga@webrtc.org 9a0a17fb7a Make it possible to change the amplitude of the pulses generated by PulsedNoiseCapturer.
This adds a SetCapturer function to testing::FakeAudioDevice::PulsedNoiseCapturer
that can be used to update the volume of the generated audio mid-call. It also modifies
CreatePulsedNoiseCapturer to use PulsedNoiseCapturer's type directly so that its new
function is visible for the callers.

Bug: webrtc:8666
Change-Id: I47726e242ccf221f5511e2797b2954ce035ba371
Reviewed-on: https://webrtc-review.googlesource.com/34650
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21521}
2018-01-08 15:50:02 +00:00
2018-01-08 08:35:41 +00:00
2018-01-03 08:37:12 +00:00
2018-01-04 11:41:08 +00:00
2018-01-08 08:30:31 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +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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