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Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

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Re: Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

#201

Post by dang »

drudu wrote: post #188
vortico wrote: post #160

Why in the world do we need 3 kbps audio for voice? It's so hard to hear people speaking over mobile phones. Why don't we use 32-64 kbps 48 kHz for all voice communication? GSM CSD offers 9-14 kbps down/up, 3G offers 384 kbps, and EDGE offers 473 kbps. Why limit to 3 kbps?

if you can't answer the question yourself, you wouldn't ever be able to understand the answer. maybe i want to have two calls at once / maybe i have a shitty gsm/csd link / maybe we are transmitting over a link that has less than 10kbps bandwidth. asking this question implies you have no imagination or awareness beyond what you already know. i'll get banned from hn for this but goddamn you're a dumb person

We've banned this account. Please do not create accounts to break HN's rules with.

https://news.ycombinator.com/newsguidelines.html

Re: Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

#202

Post by Jowsey »

perryizgr8 wrote: post #157
walrus01 wrote: post #31

Microsoft also just announced a 6kbps advanced audio codec: https://techcommunity.microsoft.com/t5/microsoft-teams-blog/...

And it sounds significantly better to me, which is expected since they use double the bitrate.

Who could've possibly guessed that doubling the bitrate would increase the quality? :P

Re: Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

#203

Post by p1esk »

ChuckMcM wrote: post #199
p1esk wrote: post #198

Earlier quoted context omitted.

I think you would agree that a GAN, or any generative model can only generate something in the same domain as what it was trained on. If you trained on mostly on human faces with a little bit of rabbits, it's not going to generate rabbits well. If you trained it on mostly on English text and a little bit on Mandarin, it's not going to generate good text in Mandarin. Same with sounds. Different languages use different…

I agree with you 100%! This is where I am wondering: > If they use any generative model in their codec, they had to train it first, offline, on some dataset. One thing I'm wondering if they have a model that can be "retrained" on the fly. Let's assume for this discussion that you've got a model with 1024 weights in it. You train it on spoken text, all languages, just throw anything at it that is speech. That gets a y…

What you described could work in principle, but in practice, "mini" training on a single sample is not likely to produce good results, unless the sample is very large. Also, this finetuning would most likely be quite resource intensive. I recall older speech recognition systems where they would ask you to read a specific text sample to adapt the model to your voice, so yes, this can work.

If you can fit a large generative model (e.g. an rnn or a transformer) in the codec, you might be able to offer something like "prompt engineering" [1], where the weights of the model don't change, but the hidden state vectors are adjusted using the current input. So, using your analogy, weights would be DNA, and the hidden state vectors would be the "mental state". By talking to this person you adjust their mental state to hopefully steer the conversation in the right direction.

[1] https://www.gwern.net/GPT-3#prompts-as-programming

Re: Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

#204

Post by StavrosK »

bscphil wrote: post #98
domoritz wrote: post #95

Earlier quoted context omitted.

Off topic but how do you put images on ipfs and what’s the advantage over e.g. Imgur?

Cloudflare are kindly hosting [1] a free HTTP gateway for the IPFS [2] network. So I can host an image myself on a server with IPFS, and Cloudflare will cache it for me. This is better than Imgur because the latter has been redirecting users to annoying "social" pages with ads instead of showing them the actual image, at least in some cases. I also can't be sure whether Imgur recompresses your uploads or not - I assu…

There's also https://imgz.org, that doesn't have annoying social stuff (I made it specifically for that!).

Re: Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

#205

Post by dwild »

bscphil wrote: post #79

I've just taken a minute to confirm what my ears told me in Audacity. Please have a look at this screenshot: https://cloudflare-ipfs.com/ipfs/Qma41RMzieQ6ZGdGem9rLxnxEL1... The Lyra version is clearly much louder. This is a serious problem and it borders on being reasonable to call it "cheating". It's well known in the audio biz that if you ask people to compare two experiences, and one of them is a bit louder than t…

I lowered the volume on the Lyra one and the sound is still clearly WAY MORE clearer than the other two.

Re: Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

#206

Post by dwild »

toomim wrote: post #69
kwindla wrote: post #61

Earlier quoted context omitted.

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…

Google Chrome has a latency of 20ms to just repeat back audio on the local device . That is, with no networking, and no processing, it takes 20ms for any information to from microphone back out to speakers.

I can't click a button I see flash faster than 200 ms, and that's when 100% of my attention is looking closely at a light change color and clicking. If 20ms is an issue for a conversation, where there so much more processing and understanding from both side, I'm clearly a subhuman...

Re: Lyra audio codec enables high-quality voice calls at 3 kbps bitrate

#207

Post by xellisx »

kwindla wrote: post #67
Aloha wrote: post #55

Earlier quoted context omitted.

A cell phone tbh, is about the same latency as a landline in most end to end call circumstances. Latency only really is noticeable when better than 600ms. (And only a real problem over 1000)

I disagree with these numbers, in general. Though of course "noticeable" is subjective and varies by use case as well as by person. For many people, end-to-end audio latency in a 1:1 conversation becomes noticeable/annoying at 200ms. And in a multi-participant conversation, talking over each other becomes noticeably more common even at 100ms compared to 50ms.

IIRC the Bell standard points out no more than 50ms.
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