commit | e47aee3b864fe5a4f964d405a7f6f3ac8c49f174 | [log] [tgz] |
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author | Daniel Lee <dklee@google.com> | Wed Apr 17 13:27:32 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Apr 17 14:40:23 2019 |
tree | 3e00ae3f6e33b383453f6fa2026f9d131714c5af | |
parent | d624c3903c036f6ce7686bf433e1f4749482a0f6 [diff] |
Ensure that we always set values for min and max audio bitrate. 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: I7e289f209a927785572058b6fbfdf60fa14edf05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126229 Reviewed-by: Minyue Li <minyue@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Daniel Lee <dklee@google.com> Cr-Commit-Position: refs/heads/master@{#27667}
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.