What I want to know is whether Lyra takes any longer to encode than the alternatives. Because as far as I can tell, nobody cares in the slightest about latency. Phone calls are getting to be like writing postcards to each other. Speak in a whole paragraph. Wait several seconds for the latency to clear. Then the other party responds with a whole paragraph, waits several seconds for the latency to clear... Improvements…
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-...
[1] https://tools.ietf.org/html/rfc7587#section-6.1
[2] https://chromium.googlesource.com/external/webrtc/+/HEAD/mod...