Henrik Boström 199e27bb63 RtpReceiverInterface::stream_ids() added.
This is the first step to removing streams from third_party/webrtc.
RtpReceiverInterface::streams() will have to be removed separately.
See https://crbug.com/webrtc/9480 for more information.

Bug: webrtc:9480
Change-Id: I6f9e6ddcda5e2245cc601d2cc6205b7b363f73ef
Reviewed-on: https://webrtc-review.googlesource.com/86840
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23929}
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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.

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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