Gustaf Ullberg 8c51f2e9cd AnalyzeReverseStream with StreamConfig
Adding a version of AnalyzeReverseStream with audio parameters
described by StreamConfig. This is part of preparations for
multichannel APM in Chromium.

Bug: webrtc:10913
Change-Id: I7c4650eab8bd7fcdec970a7e4a8fa203f09bed9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157897
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29573}
2019-10-22 14:27:14 +00:00
2018-10-05 14:40:21 +00:00
2019-09-10 10:03:50 +00:00
2019-07-08 13:45:15 +00:00
2017-09-15 04:25:06 +00:00
2018-12-18 12:30:58 +00:00
2019-10-08 12:20:39 +00:00
2017-09-15 04:25:06 +00:00
2018-07-23 15:28:48 +00:00
2017-09-15 04:25:06 +00:00
2019-09-03 14:55:43 +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

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