commit | d14525eb59b94ad10df093cddd43ba543832075a | [log] [tgz] |
---|---|---|
author | Jakob Ivarsson <jakobi@webrtc.org> | Fri Mar 06 09:49:29 2020 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Mar 06 14:49:37 2020 +0000 |
tree | 36bcfa424318ff6405ac5591818e726a6c2c3a0a | |
parent | b0f2e0ced4cd7ff23a54d0c494a9264cd62e0b1a [diff] |
Make sure that the audio stream is allocated with the correct overhead. This fixes two cases when the allocation is not updated correctly: - The frame length range is not updated when audio network adaptor is enabled. - The per-packet overhead is not updated unless the bitrate observer has been reconfigured. Bug: webrtc:11001 Change-Id: I2ee25f956741a4be08661f874556582dd60a3bd0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169848 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30709}
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.