See bug for more info. In this case, the offset of the byteBuffer was observed to be 4 bytes when testing, meaning that the first 4 bytes sent to the AudioSamples callback were empty, and the last 4 bytes that should have been sent were not sent. This CL adjusts the range copied from the backing array to match the offset. Bug: webrtc:9175 Change-Id: I40ac6e10c6d7058ead7eff1c9fa2f342920cf2a4 Reviewed-on: https://webrtc-review.googlesource.com/75123 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23172}
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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