commit | 1d037ae704b0941769623eb6a4a9787e40070615 | [log] [tgz] |
---|---|---|
author | Ilya Nikolaevskiy <ilnik@webrtc.org> | Thu Mar 15 14:46:17 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 15 15:42:57 2018 |
tree | 79901a81dc65a2bb293977b7bb8ee453bfcf7df0 | |
parent | 4425b055f92c3bbbf242ed9ae62676e498229977 [diff] |
Don't crash in SingleNalu packetization for h264 if no space in packet Also, pass correct max payload data size to encoders: now accounting for rtp headers. Bug: chromium:819259 Change-Id: I586924e9246218fab6072e05eca894925cfe556e Reviewed-on: https://webrtc-review.googlesource.com/61425 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22460}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.