| commit | 9356252bfb75c4f85abe8894ac5f86b20f22e302 | [log] [tgz] |
|---|---|---|
| author | Daniel Lee <dklee@google.com> | Fri May 03 14:40:13 2019 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri May 03 13:45:43 2019 +0000 |
| tree | 0a7190170c3145e2064d0d37001fd8aeee3ce356 | |
| parent | 87a92d087c0e170b3af96de9502b44169128c684 [diff] |
Ensure that we always set values for min and max audio bitrate. (Re-land reverted cr). Use (in order from lowest to highest precedence): -- fixed 32000bps -- fixed target bitrate from codec -- explicit values from the rtp encoding parameters -- Final precedence is given to field trial values from WebRTC-Audio-Allocation Bug: webrtc:10487 Change-Id: I573e996fa1f243e673785cdbe687e029fd5cbf4a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133483 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Daniel Lee <dklee@google.com> Cr-Commit-Position: refs/heads/master@{#27847}
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.