| commit | 0505115b5c80162d84b1c024c31515457a4438fc | [log] [tgz] |
|---|---|---|
| author | Palak Agarwal <agpalak@google.com> | Wed Sep 20 08:01:18 2023 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Sep 26 15:57:52 2023 |
| tree | 26fa6c09cee5c5aae8020cf4e327a9a0c69e9120 | |
| parent | 3218d743be73982f90632310c3e3bca5cee87b3e [diff] |
Pass the correct abs_capture_timestamp while cloning audio frame This change replaces type of absolute_capture_timestamp_ms_ in TransformableOutgoingAudioFrame from int to optional uint and makes the function AbsoluteCaptureTimestamp() inside TransformableAudioFrameInterface pure virtual. Bug: webrtc:14949 Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000 Commit-Queue: Palak Agarwal <agpalak@google.com> Reviewed-by: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40814}
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.