/webrtc-audio-processing

A packaging-friendly copy of the WebRTC AudioProcessing module

Primary LanguageC++BSD 3-Clause "New" or "Revised" LicenseBSD-3-Clause

About

This is meant to be a more Linux packaging friendly copy of the AudioProcessing module from the WebRTC[1][2] project. The ideal case is that we make no changes to the code to make tracking upstream code easy.

This package currently only includes the AudioProcessing bits, but I am very open to collaborating with other projects that wish to distribute other bits of the code and hopefully eventually have a single point of packaging all the WebRTC code to help people reuse the code and avoid keeping private copies in several different projects.

[1] http://code.google.com/p/webrtc/ [2] https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git

Feedback

Patches, suggestions welcome. You can send them to the PulseAudio mailing list[3] or to me at the address below.

-- Arun Raghavan mail@arunraghavan.net

[3] http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss

Notes

  1. Some files need to be patch to avoid pulling in the gtest framework. This should ideally be pushed upstream in some way so we're able to just pull in what we need without changing anything.

  2. It might be nice to try LTO on the library. We build a lot of code as part of the main AudioProcessing module deps, and it's possible that this could provide significant space savings.