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By default the streaming mic capture uses buffer/chunk size of 1 second. This adds a long latency in real time applications. Can the chunk size be made configurable/smaller?
Is your feature request related to a problem? Please describe.
Large buffer increases audio latency and makes application sluggish to use.
Describe the solution you'd like
Provide a parameter to configure chunk size when using streaming mic capture
Additional context
Add any other context or screenshots about the feature request here.
gaborvecsei, mcorroyer, Pastells and GuillaumeTong