| commit | c3eb9fd49f7343ab7ea2ea49ae1fa576aae5231d | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Wed Jan 29 16:42:52 2020 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 29 18:42:34 2020 |
| tree | 4aa1a4077b92f4e4441d6b27cf5b46a243f0ae20 | |
| parent | 0cda7b832a9e86b7fb5d48d00b94a8d321602cdb [diff] |
Reland "Reland "Only include overhead if using send side bandwidth estimation."" This is a reland of 086055d0fd9b9b9efe8bcf85884324a019e9bd33 ANA was accitendly disabled even when transport sequence numbers were negotiated due to a bug in how the audio send stream is configured. To solve this we simply continue to always allow enabling ANA and leave it up to the application to ensure that it's not used together with receive side estimation. Original change's description: > Reland "Only include overhead if using send side bandwidth estimation." > > This is a reland of 8c79c6e1af354c526497082c79ccbe12af03a33e > > Original change's description: > > Only include overhead if using send side bandwidth estimation. > > > > Bug: webrtc:11298 > > Change-Id: Ia2daf690461b55d394c1b964d6a7977a98be8be2 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166820 > > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Reviewed-by: Ali Tofigh <alito@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30382} > > Bug: webrtc:11298 > Change-Id: I33205e869a8ae27c15ffe991f6d985973ed6d15a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167524 > Reviewed-by: Ali Tofigh <alito@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30390} Bug: webrtc:11298 Change-Id: If2ad91e17ebfc85dc51edcd9607996e18c5d1f13 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167883 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30413}
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.