| commit | d36df89d40bde3c62ee5cbff841933e50b3c007b | [log] [tgz] |
|---|---|---|
| author | tommi <tommi@webrtc.org> | Tue May 24 12:49:04 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue May 24 12:49:10 2016 |
| tree | 00dbf46a78cb23b56e6bdba122f920a25237dd1c | |
| parent | 32e80e4b2ec2df9449bb8b98fa1428d6e5c4aaa9 [diff] |
Adding a some checks and switching out a few assert for RTC_[D]CHECK. These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled. I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state. BUG=chromium:613482 NOTRY=true (using notry due to offline android_arm64_rel bot) Review-Url: https://codereview.webrtc.org/2007563002 Cr-Commit-Position: refs/heads/master@{#12870}
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.