Audio: Reduce max channels to 16 to prevent buffer overflow Lowers kMaxNumberOfAudioChannels from 24 to 16 to fit within the statically allocated AudioFrame buffer (7680 max samples) when resampling at 48kHz (7680 = 16 channels * 480 samples). Also implies safe channel capping in resampler to prevent buffer overflows. Bug: webrtc:495018167 Change-Id: I60b30832b34508deff036a3a63fcabac1a35c889 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/460680 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47284}
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.