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[2] or to me at the address below.

-- Arun Raghavan <mail@arunraghavan.net>

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

Notes
=====

Assembling some quick notes on maintaining this tree vs. the original WebRTC
project source code.

1. Running meld on a pristine tree's webrtc/ vs. the same directory in the
   Chromium third_party/webrtc/should produce a fairly easy-to-parse set of
   differences that can be merged. I've kept the test code out for now, but
   this might get merged in the future. The .gn files are included to make
   allow tracking changes to the build system easier.

2. 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.

3. 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.