commit | 4a68fb9ab43d8f12a70319a30d601436a354535a | [log] [tgz] |
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author | Ruslan Burakov <kuddai@google.com> | Wed Feb 13 13:25:39 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Feb 13 15:54:43 2019 |
tree | 755dd0ca159c5d9be433f18df2af4c796df93f9f | |
parent | 69bb3afd53d8a32c1678ec8ed484d3f7e6b132d0 [diff] |
Separate base minimum delay and minimum delay. On NetEq level latency corresponds to delay and two terms can be used interchangeably here. In order to implement latency constraint we need to provide a range of possible values which should be constant. See getCapabilities() here: https://www.w3.org/TR/mediacapture-streams/#dfn-applyconstraints-algorithm Lowest possible value accepted value is constant and equals 0. But because |packet_len_ms_| and |maximum_delay_ms_| may be updated during live of DelayManager upper bound is not constant. Moreover, due to change in |packet_len_ms_| the |minimum_delay_ms_| which was valid when its was set may be considered invalid later on. To circumvent that and provide constant range for capabilities we separate base minimum delay and minimum delay. ApplyConstraints algorithm will set base minimum delay. Base minimum delay will act as recommendation for lower bound of minimum delay and will be used to limit target delay. If user sets base minimum delay through ApplyConstraints which is bigger than currently possible maximum (e.g. bigger than NetEq maximum buffer size in milliseconds) then base minimum delay will be clamped to currently possible maximum to match user's intentions as best as possible. Note, that we keep original behavior when minimum_delay_ms_ (effective_minimum_delay_ms after this CL) in LimitTargetLevel method may be above upper bound due to changing packet audio length. Bug: webrtc:10287 Change-Id: I06b8f5cd3fd1bc36800af0447f91f7c4dc21a766 Reviewed-on: https://webrtc-review.googlesource.com/c/121700 Commit-Queue: Ruslan Burakov <kuddai@google.com> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26666}
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.