This is a fantastic question. I agree with you that we're slowing boiling the frog (and the frog is ourselves) in accepting more and more latency in our real-ti

This is a fantastic question. I agree with you that we're slowing boiling the frog (and the frog is ourselves) in accepting more and more latency in our real-time communications. I think the answer for Lyra is that latency is a concern, but maybe at this stage not as much of a concern as it could be. I'm only guessing, though based on this [0]: > The basic architecture of the Lyra codec is quite simple. Features, or distinctive speech attributes, are extracted from speech every 40ms and are then compressed for transmission. That sounds like the minimum frame size for Lyra is 40ms. For Opus (the audio codec used for most WebRTC applications), the default frame size is 20ms [1], and most implementations support frame sizes of 10ms [2]. Of course, your favorite web browser might not default to 20ms frames for Opus. And by "most implementations" I meant Google Chrome. :-) [0] https://ai.googleblog.com/2021/02/lyra-new-very-low-bitrate-codec-for.html https://ai.googleblog.com/2021/02/lyra-new-very-low-bitrate-... [1] https://tools.ietf.org/html/rfc7587#section-6.1 https://tools.ietf.org/html/rfc7587#section-6.1 [2] https://chromium.googlesource.com/external/webrtc/+/HEAD/modules/audio_coding/codecs/opus/audio_encoder_opus.cc#63 https://chromium.googlesource.com/external/webrtc/+/HEAD/mod...