While the extra quality is lost on listeners, from experience I've found that super-HQ source material can change your results (for the better) when fed through distortion/compression/warming effects processors
I also have some gear that aliases at a volume you can't hear, but when you plug it into an analog distortion pedal, the aliasing in the high frequencies becomes apparent. This would be avoided if it had a higher sample rate so the aliasing was far out of the audible range.
For other sorts of effects like spectral processing, pitch shifting, the extra detail above 22khz really does make a difference, especially if pitching down.
• Less quantisation noise, so your noise floor is a bit lower and therefore, when you're mixing stuff together, you accumulate less noise
• More numerical room to play with, you can scale or shift the volume up and down more without clipping and with less loss of accuracy
With 16-bit CD audio, you can't just convert to 24-bit and suddenly have less noise. You might get more room, though.
As for higher sampling rates (more kHz, if you will), I think Monty mentioned some benefit regarding less sharp aliasing filters (can have a larger transition band from 20–48kHz, say, rather than from 20—22.1kHz), but it's not something I understand the benefit of well.
Specifically, if 24/192 AAC is worth it compared to 16/44.1 AAC (and the answer to that is yes, although the answer to 24/192 WAV is no)
The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering.
Nyquist is fine in theory, and if you've never actually tried to implement a clean filter you'll likely watch the xiph video and think "Well that makes sense."
If actually know something about practical DSP and the constraints and challenges of real filter design you're not going to be quite so easily impressed.
Likewise with higher bit depths. "Common sense" suggests that no one should be able to hear a noise signal at -90dB.
Common sense is dead wrong, because the effects of a single bit of dither are absolutely audible.
And if you can hear the effects of noise added at -90dB, you can certainly hear the effects of quantisation noise artefacts on reverb tails and long decaying notes at -30 to -40dB, added by recording at 16 bits instead of 24 bits.
Whether or not that level of detail is present in a typical pop or classical recording is a different issue. Realistically most music is heavily compressed and limited, so the answer is usually "no."
And not all sources have 24-bits of detail [1]. (Recordings made on the typical digital multitrack machines used in the 80s and 90s certainly don't.)
That doesn't mean that a clean unprocessed recording of music with a wide dynamic range made on no-compromise equipment won't show the difference clearly.
Speaking from experience, it certainly does.
[1] Technically no sources have 24-bits of detail. The best you'll get from a real world converter is around 22-bits.
What video? This thread is about an article.
> The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering.
I just said that, I think.
As Monty demonstrates, it's a fraudulent waste to try to sell the result as a product to the end listener.
The article is highly technical. Does anyone have a way to describe this phenomenon intuitively?
Therefore, rather than just being useless extra data, it can be actively harmful to the listening experience.
- 480p vs 2160p is measuring the resolution of your cell phone propped up on a pillow at the other end of your living room
- experimental evidence shows that your eyesight is not good enough to pick up on the increased resolution, you've maxed out your sensory perception
- Your phone stutters trying to stream at 4k so the playback might actually be worse.
When the high resolution image is downscaled poorly, some high (spatial) frequencies are aliased down into lower (spatial) frequencies, manifesting as blocky/jagged lines. The images are 2D data in the spatial domain, while the audio is 1D data in the temporal domain, but the aliasing of high frequency signal to lower frequency artifact is similar.
Viewing a high resolution image on a panel that has enough pixels to display it without scaling is analogous to listening to 192 kHz audio on a fantastic speaker system that can reproduce those high frequencies accurately, instead of causing distortion by aliasing them to lower frequencies. On the other side, viewing a high resolution image which has been downscaled poorly is analogous to listening to that 192 kHz audio on a realistic speaker system that cannot reproduce high frequencies, which results in those signals aliasing down into the audible range.
And as you say, there is a point where, for the viewer/listener's sake, it doesn't make sense to push for higher frequencies because even if you can build a panel/speaker that will faithfully reproduce those frequencies without aliasing, the eye/ear will not be able to perceive the additional detail.
[1] http://www.maxrev.de/files/2012/08/screenshot_interpolation_...
Yet people have been reliably _unable_ to do this.
The gold standard in listening tests for this is an ABX where you are simply trying to show that you can discern a difference.
When properly setup and calibrated people are unable to show that they can distinguish 48kHz and 192kHz.
Moreover, by the numbers, audio hardware tends to work less well at higher rates if they're different, because running them at higher rates makes their own internal noise shaping less powerful. (Though for anything done well the distinction should still be inaudible).
Finally, if you do have cruddy converters that sound significantly different at different rates nothing stops you from using a transparent software resampler (SSRC is well respected for good reason) to put the audio at whatever rate you want.. until you get better hardware. :)
That would have to be the noise. The math doesn't lie...
When you play something stored at 44 khz, there are no ultrasonic sounds recorded. At 192 khz instead, there are, and the speakers may push some of the ultrasonic sounds down to the audible spectrum, causing distortion that the human ear can hear.
Imagine the normal operation of a speaker as a swing, except you are pushing and pulling the swing all throughout the cycle as it goes up and down. Now, you can technically move the swing at a variety of frequencies if you’re holding onto it the whole time. However imagine as you push it back and forth (low frequencies), you also vigorously shake the swing at the same time (high frequencies). This would probably result in the chains rattling, similar to the unwanted distortions in the speakers caused by ultrasonic frequencies.
Another way to look at is that digital audio is not like digital imaging. There aren't pixels. Increasing the data rate does not continue to make the waveform more and more detailed to human auditory perception in the way that raising the pixel density does for human visual perception.
To describe it intuitively, forget your intuition that audio is like visual and start from "there is no metaphor between these two things."
https://en.wikipedia.org/wiki/Aliasing
Even analog imaging has concerns that are better described in the frequency domain than the spatial domain:
The difference is audio is "actually" bandlimited and frequency-based, but images are "actually" spatial. When you try using frequency math on images you get artifacts called ringing, which are easy to see with JPEGs or sharp scalers like Lanczos.
Of course audio isn't really frequency-based either, or else it would just be one sound repeated forever. So there's still ringing artifacts (called "pre-echo") and cymbals are the first to go in an MP3.
I.e. audio is sampled along one dimension, while images are sampled along two dimensions. Note that frequency-domain considerations play a crucial role in all optical design, including imaging sensors.
For the mathematically inclined, this would probably be a good time to repeat: pixels are not little squares[1].
Early low data rate codecs - such as the one used for GSM mobiles - are obviously inferior, but still functional. I think a better analogy is that an iPhone 7 has a 1 megapixel screen, so there's no difference between a 1 megapixel image and a 5 megapixel image, except one is much larger. Of course visually you can zoom in (or move closer in real life), but audibly you can't.
Digital distortion basically simulates high gain with soft clipping, which basically takes a narrow slice of the amplitude domain and magnifies it. The extra resolution has to be there for that not to turn into ugly sounding shit with artifacts.
1 / (2 * pi * bandwidth * 2 ^ bitdepth)
So for full scale 44/16 signal it is about 0.1 nanosecond.
More here [1]. There are also some sample files for ABX tests there.
[1] - https://hydrogenaud.io/index.php/topic,108987.msg896449.html...
I haven't had time to check the linked article but if it's the one I'm thinking of I'm pretty sure it goes into this.
In the end though, double blind testing (again, I think mentioned in the article) shows that there's a threshold (a bit below 48kHz, probably why 44.1kHz was chosen for CD audio) after which people cant distinguish between higher sample rates at any more accuracy than randomly guessing.
So no.