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The existing restriction of max 48k seems old and outdated. I am unable to see any issues by simply extending the support to 96 and utilize the existing resampler in WebRTC. There are no memory limitations involved either. It is a rather common case today in Chrome that users need 96k/192k input; hence this simple change will have a positive impact for many WebRTC clients using gUM. Bug: webrtc:10958 Test: https://webrtc.github.io/samples/src/content/peerconnection/audio/ using mic @96k Change-Id: I8123da886ef7d48cbec9482795ec837ec1f61d81 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152162 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29135}