commit | d5512429740a0ec7e91ddf07341952457e303ecf | [log] [tgz] |
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author | Tommi <tommi@webrtc.org> | Wed May 28 11:47:27 2025 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed May 28 14:39:12 2025 |
tree | 4a876d28679b3b8c92e783ad0e2bf4d7b70d918c | |
parent | 3dfa6f80d7c32ccb1ab3018040abf0a21b0e817e [diff] |
Add AudioMultiVector::ReadInterleavedFromIndex() The new method differs from the existing method with the same name in two respects: * It reads into an InterleavedView<> instead of pointer. * It either reads exactly the requested number of samples or fails After examining the behavior in NetEq, where this method is called from, it seems that what we need is a method that reads exactly the required number of samples, not partially. A follow-up change to NetEqImpl and SyncBuffer will change SyncBuffer::GetNextAudioInterleaved() to read the requested buffer size or fail, as well as update NetEqImpl::GetAudioInternal accordingly. Bug: chromium:335805780 Change-Id: Ia78f888d52c9acb4b53b7042b31b8b0c53199e7b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392143 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44787}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.