| commit | 98ab3a46d6b98bd6626ab741092f7cbf104d127b | [log] [tgz] |
|---|---|---|
| author | kwiberg <kwiberg@webrtc.org> | Wed Sep 30 21:54:21 2015 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Oct 01 04:54:29 2015 +0000 |
| tree | aeb35ee96119d07d88d32cf67eec873174fa60df | |
| parent | 456696a9c1bbd586701dcca3e4b2695e419a10ba [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. (This was previously landed as revisions 10046 and 10060, and got reverted because it broke several of the Chromium FYI bots.) BUG=webrtc:4557 Review URL: https://codereview.webrtc.org/1368843003 Cr-Commit-Position: refs/heads/master@{#10127}
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.