<p><strong>Summary:</strong><br>
<strong>1.</strong> Is it normal among lossey codecs?<br>
<strong>2.</strong> what are the effects of this specifically when handling this 'cutoff' audio file?<br>
<strong>3.</strong> Why is it still saving the file as 48khz if it doesn't contain that data anymore?
+++++++++++++++++++++++++++++++++++++++++++++++++++++<br>
I was reading the wiki for Opus. And the bottom notes caught my eye, mainly:
<a href="https://en.wikipedia.org/wiki/Opus_(audio_format)" rel="nofollow noreferrer">https://en.wikipedia.org/wiki/Opus_(audio_format)</a></p>

<blockquote>
  <p>Opus cuts audio above 20 kHz, the generally accepted upper limit of
  the human hearing range.</p>
</blockquote>

<p>It lists very specific sample rates that are supported as well for opus on that page: 8khz, 12khz, 16khz, 24khz, 48khz  </p>

<p>So if I intend to use opus as my audio format does it mean I need only record at a sample rate of 24khz? Anything above 20khz is just extra data for transforming the waveform before export? So no intent to transform would mean you might as well just record at 24khz?</p>

<p>I'm not overly familiar with the specifics of audio codecs, but is this normal for codecs to cutoff such a large range of sounds? That just seemed concerning to me, given everyone records at ~44.1khz minimum and in general 48khz is well used. Is it better to use a different 'lossey' codec than opus if I intend on possibly editing/re-encoding the audio again at some point? Or would there be no real difference between the loss experienced when editing and re-encoding an opus vs the loss when editing an aac/etc?</p>

<p>I'm also confused why when I re-encode from a 48khz f32 .wav to opus it saves itself as a 48khz sample rate still. Isn't all data above 20khz gone...? That's confusing to me. edit: well I guess when you merge different sample rates they sound of different speeds so that's why it would keep the original format..?</p>

<p>And yes I'm aware that lossless is the ideal format if I intend to re-encode/edit at some point, but for the sake of argument let's pretend we have to choose between lossey formats (at least for the initial export, we can trans-code later to lossless)</p>

<audio><codec><audio-recording><audio-quality>
"Opus cuts audio above 20kHz" - Is this normal for lossey audio codecs?
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# "Opus cuts audio above 20kHz" - Is this normal for lossey audio codecs?

(Asked by: username_0 on 2019-01-04)

**Summary:**  
**1.** Is it normal among lossey codecs?  
**2.** what are the effects of this specifically when handling this 'cutoff' audio file?  
**3.** Why is it still saving the file as 48khz if it doesn't contain that data anymore?
+++++++++++++++++++++++++++++++++++++++++++++++++++++  
I was reading the wiki for Opus. And the bottom notes caught my eye, mainly:
<https://en.wikipedia.org/wiki/Opus_(audio_format)>

> Opus cuts audio above 20 kHz, the generally accepted upper limit of
> the human hearing range.

It lists very specific sample rates that are supported as well for opus on that page: 8khz, 12khz, 16khz, 24khz, 48khz

So if I intend to use opus as my audio format does it mean I need only record at a sample rate of 24khz? Anything above 20khz is just extra data for transforming the waveform before export? So no intent to transform would mean you might as well just record at 24khz?

I'm not overly familiar with the specifics of audio codecs, but is this normal for codecs to cutoff such a large range of sounds? That just seemed concerning to me, given everyone records at ~44.1khz minimum and in general 48khz is well used. Is it better to use a different 'lossey' codec than opus if I intend on possibly editing/re-encoding the audio again at some point? Or would there be no real difference between the loss experienced when editing and re-encoding an opus vs the loss when editing an aac/etc?

I'm also confused why when I re-encode from a 48khz f32 .wav to opus it saves itself as a 48khz sample rate still. Isn't all data above 20khz gone...? That's confusing to me. edit: well I guess when you merge different sample rates they sound of different speeds so that's why it would keep the original format..?

And yes I'm aware that lossless is the ideal format if I intend to re-encode/edit at some point, but for the sake of argument let's pretend we have to choose between lossey formats (at least for the initial export, we can trans-code later to lossless)

**Tags:** audio, codec, audio-recording, audio-quality


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## 0 answers

