| commit | f66a9251424351ea6d631c54dd1feb64cc13d809 | [log] [tgz] |
|---|---|---|
| author | kwiberg <kwiberg@webrtc.org> | Thu Sep 24 03:18:40 2015 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Sep 24 10:18:48 2015 +0000 |
| tree | d6f21ac17b6bad95763c383264e86d00b7cd0b56 | |
| parent | 61e933eac7673feb2f8663c3e71e503b714b350f [diff] |
Don't link with audio codecs that we don't use We used to link with all audio codecs unconditionally (except Opus); this patch makes gyp and gn only link to the ones that are used. (This unfortunately fails to have a measurable impact on Chromium binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC fix were already being excluded from Chromium by some other means (likely just the linker omitting compilation units with no incoming references).) BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1349393003 Cr-Commit-Position: refs/heads/master@{#10046}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.