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Comment by rectang | Hacker News Reader

> Video conferencing platforms have opted to deliver audio that arrives quickly but is low in quality. Platforms aim for a lag time of less than 150 milliseconds. Yet that is long enough to violate the no-overlap/no-gap convention to which speakers are accustomed.

EXACTLY. Latency is a killer. Even on cell phones.

Latency does not get enough attention. It's not as in-your-face as dropouts, but the levels of latency which we experience in modern communication channels are crazy bad.

And even if platforms are aiming for 150 ms, in practice the latency is often much higher than that.

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Furthermore, a lot of users have substantial latency introduced by their setups - for example, earlier Bluetooth headphones like AirPods 1 add 300ms of latency (126ms for AirPods 2 Pro) https://stephencoyle.net/airpods-pro-2 . So even if you have a great network backbone and a great protocol, someone's crappy home setup might make it all for naught.

I've long held the opinion that many audio compression codecs have outlived their usefulness, and now simply introduce latency and reduce audio fidelity to preserve bandwidth that is no longer in short supply.

Every internet connection I've had over the past 10 years in the US has been fast enough to send and receive at least a dozen uncompressed CD-quality PCM streams simultaneously. On paper, my current fiber connection should be able to handle 1,400 of them. This insistence on audibly mangling the audio to keep it at ~96 kbps makes less and less sense every day -- especially considering how the audio is usually carrying all the important information in the call anyway!

(I find the same thing happens on streaming services too. 98% of the bits are for video, 2% for sound.)

While less compression is necessary than before, I'd argue that some basic, ultra-fast compression would be useful to allow for some basic bandwidth savings and error correction codes.

Honestly, my bluetooth headset probably adds almost as much latency as my fiber connection and wifi combined. That's another issue most people don't even consider: bluetooth is orders of magnitude slower than a wire, even in the best case scenarios.

it's more likely that these companies would rather save a few pennies by introducing a new 64 kbps codec that reduces quality by only 10%, while offering the old codec's bitrate as a “fidelity mode” for premium+ subscribers.

You've also got to consider that the higher the bandwidth a stream is, the harder it is to deliver it with a low latency.

If every packet has a risk of being dropped or of arriving out of order then increasing either the size or the sheer quantity of those packets will also increase how often they get mishandled or buffered.

Plus you run into compatibility problems between different kinds of devices such as mobile users trying to connect over spotty 3G connections.

Yes latency in practice is often higher than a second. Even in-person I must be careful not to barge in conversations and cutting off people, but latency makes it much worse. If you are in a larger group, one tends to not say anything to the group spontaneously because it always feels like cutting in.

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