Looking for music with meaningful ultrasonic content

[Scytales]

I plan to do compare sample-rates blind later (when i have better speakers) but i am already looking for good test tracks.

I want to see how high in frequency i can detect a lowpass filter once i have good equipment.

I am especially interested in testing if ultrasonics above say 24 Khz can make a difference (when there is basically zero chance of me hearing them directly).

Try to spot the Denon Audio Check SACD (COGQ-28) : https://www.discogs.com/fr/release/10440800-Various-Denon-Audio-Check-SACD

As far as I have been able to measured it with an old HP3581A, the DSD pink noise test tracks may go as far as 50 kHz high and possibly higher.

Using a relatively stable noise test signal instead of music might make it easier to hear anything, if they are anything to hear, at least at the begining of an experiment.

ASR forum member GXAlan also possesses this Denon SACD and have the means to digitally analyzed its content to confirm or not the pink noise bandwidth and, perhaps, digitize these test tracks in high sample rate PCM in order make experiments.

Some of the music demo tracks on this Denon SACD have frequency contents that go higher than 20 kHz, also not much (25 to 30 kHz max, if I remember correctly the last time I checked with the HP3581A).

 

[Grandzoltar]

I've posted this before try this I'm curious.

A HyperSonic Sound system – or HSS – does not use physical speakers. HSS pulsates quartz crystals at a frequency thousands of times faster than the vibrations in a normal speaker – creating ultrasonic waves at frequencies far beyond human hearing. Unlike lower-frequency sound, these waves travel in a tight path – a beam. Two beams can be focused to intersect each other, and where they interact they produce a third sonic wave whose frequency is exactly the difference between the two original sounds. In HSS that difference will fall within the range of human hearing – and will appear to come from thin air. This is known as a Tartini Tone – in honor of Giuseppe Tartini, the eighteenth-century Italian composer who first discovered this principle.

 

[ChrisG]

How much resolution do you feel you need before the playback is 100% transparent to the mic feeds?

Depends on the mics.

To take a couple of frequently-used examples:

First up, the Shure SM58. As a live sound engineer, I have to deal with these pretty often. They've been around almost forever, and musicians are comfortable seeing them. They're the minimum standard of any serious venue. They're far from my favourite mic, but that's another story. Typical usage is a rock band vocalist screaming into it at point-blank range.

Info here: http://recordinghacks.com/microphones/Shure/SM58

Frequency response drops very quickly at 10kHz. Having measured a couple, I can say that it falls off a cliff there. You can't EQ it back. Given that, you could argue for a 20kHz sample rate, and you wouldn't miss much from one of the most popular mics on the planet.
Bit depth is more tricky. In theory, dynamic mics (mylar diaphragm with a coil attached. Coil moves in a magnetic field, generating voltage) have virtually unlimited maximum SPL, so a very high bit depth is called for. Microphones do get connected to a preamp, though, and those do have a limited dynamic range - at the low level, they have intrinsic noise relating to the Johnson Noise of the input resistors. At a high level, they clip. The required bit depth can thus be calculated.


At the opposite end of the microphone spectrum, we have the Schoeps MK4: https://schoeps.de/en/products/colette/capsules/cardioids/mk-4.html
This is one of the mics that will get chosen for high-end classical recording, picking up an entire orchestra. With the latest CMC1 preamp attached, it's good for 135dBSPL at the high end, and has an Equivalent Noise Level of 14dB(A). ie, 121dB of dynamic range.
Unfortunately, Schoeps doesn't publish the frequency response in the ultrasonic region. I can say, though, that my MK41 (supercardioid version) can pick up ultrasonic pest repellers working, putting out 25kHz + 50kHz. The frequency response of those repellers is unknown, but the spectral analysis showed 50kHz being around 10dB down compared to 25kHz. I used a MOTU M4 running at 192kHz, feeding into REW's RTA.
Given that these mics can clearly capture something well into the ultrasonic region, a high sample rate could be called for, if one wishes to capture 100% of the mic's signal.


Finally, it's worth noting that virtually every recording you've ever heard will have some dynamic processing. Compressors, limiters, gates and expanders. A compressor might take a 24-bit microphone feed and reduce the dynamic range to 20-bit. A gate will increase the dynamic range infinitely, but aren't used where you might care. Limiters are a sub-set of compressors. Expanders do occasionally get used in classical mixing, and can increase the dynamic range of the mic feed beyond what was there originally. Used carefully, they can produce excellent results.
Even so, the final mix will be subject to compressors/limiting to get the ratio of quiet/loud sections "correct", which puts a final upper limit on the theoretical dynamic range of a piece of music.


The mixing desk that I use for multi-track recording (I can post some clips if anyone's interested in what real, uncompressed instruments sound like) is 48kHz/32-bit-float, and I've found it hasn't been a limiting factor on the quality of my recordings, even when I bring out the Schoeps mics.

Chris

 

[radix]

My understanding of a digital mixing board (I only ever used analog back in the day) is that one needs to keep the analog input to the ADC close to 0dBFS. Otherwise, you are losing bits of resolution. If I have an input that peaks around -6 dBFS, then I've lost 1 bit right there. If I have a quiet passage thats -24 dBFS, then I've lost 4 bits. If you're recording something with a peak of 120 dB SPL and an average of 60-80 dB SPL, that's 40-60 dB (7 to 10 bits) that don't have any data most of the time.

 

[Robin L]

My understanding of a digital mixing board (I only ever used analog back in the day) is that one needs to keep the analog input to the ADC close to 0dBFS. Otherwise, you are losing bits of resolution. If I have an input that peaks around -6 dBFS, then I've lost 1 bit right there. If I have a quiet passage thats -24 dBFS, then I've lost 4 bits. If you're recording something with a peak of 120 dB SPL and an average of 60-80 dB SPL, that's 40-60 dB (7 to 10 bits) that don't have any data most of the time.

My understanding is that the additional bit depth of hi-rez recorders allows for more headroom. With 16 bits one wants to keep close to the upper dynamic limit, with 24 bits not so much.

 

[radix]

My understanding is that the additional bit depth of hi-rez recorders allows for more headroom. With 16 bits one wants to keep close to the upper dynamic limit, with 24 bits not so much.

Well, whatever your bit depth, the if you want high dynamic range, just remember the soft parts are only using a fraction of the bits. Anyway, I'm all for recording at high bit depth, but I don't think it's too helpful for playback by the majority of systems. I mean, how many can really play back even 80 dB of DR with a 30-50 dB home background noise floor? I guess this is getting OT a bit, and it's been discussed at length in other threads.

 

[theREALdotnet]

Does anyone have a HiRes/DSD version of Bat out of Hell, and can take a spectrogram? I feel that’s one thing we should check.

 

[IAtaman]

Depends on the mics.

To take a couple of frequently-used examples:

First up, the Shure SM58. As a live sound engineer, I have to deal with these pretty often. They've been around almost forever, and musicians are comfortable seeing them. They're the minimum standard of any serious venue. They're far from my favourite mic, but that's another story. Typical usage is a rock band vocalist screaming into it at point-blank range.

Info here: http://recordinghacks.com/microphones/Shure/SM58

Frequency response drops very quickly at 10kHz. Having measured a couple, I can say that it falls off a cliff there. You can't EQ it back. Given that, you could argue for a 20kHz sample rate, and you wouldn't miss much from one of the most popular mics on the planet.
Bit depth is more tricky. In theory, dynamic mics (mylar diaphragm with a coil attached. Coil moves in a magnetic field, generating voltage) have virtually unlimited maximum SPL, so a very high bit depth is called for. Microphones do get connected to a preamp, though, and those do have a limited dynamic range - at the low level, they have intrinsic noise relating to the Johnson Noise of the input resistors. At a high level, they clip. The required bit depth can thus be calculated.


At the opposite end of the microphone spectrum, we have the Schoeps MK4: https://schoeps.de/en/products/colette/capsules/cardioids/mk-4.html
This is one of the mics that will get chosen for high-end classical recording, picking up an entire orchestra. With the latest CMC1 preamp attached, it's good for 135dBSPL at the high end, and has an Equivalent Noise Level of 14dB(A). ie, 121dB of dynamic range.
Unfortunately, Schoeps doesn't publish the frequency response in the ultrasonic region. I can say, though, that my MK41 (supercardioid version) can pick up ultrasonic pest repellers working, putting out 25kHz + 50kHz. The frequency response of those repellers is unknown, but the spectral analysis showed 50kHz being around 10dB down compared to 25kHz. I used a MOTU M4 running at 192kHz, feeding into REW's RTA.
Given that these mics can clearly capture something well into the ultrasonic region, a high sample rate could be called for, if one wishes to capture 100% of the mic's signal.


Finally, it's worth noting that virtually every recording you've ever heard will have some dynamic processing. Compressors, limiters, gates and expanders. A compressor might take a 24-bit microphone feed and reduce the dynamic range to 20-bit. A gate will increase the dynamic range infinitely, but aren't used where you might care. Limiters are a sub-set of compressors. Expanders do occasionally get used in classical mixing, and can increase the dynamic range of the mic feed beyond what was there originally. Used carefully, they can produce excellent results.
Even so, the final mix will be subject to compressors/limiting to get the ratio of quiet/loud sections "correct", which puts a final upper limit on the theoretical dynamic range of a piece of music.


The mixing desk that I use for multi-track recording (I can post some clips if anyone's interested in what real, uncompressed instruments sound like) is 48kHz/32-bit-float, and I've found it hasn't been a limiting factor on the quality of my recordings, even when I bring out the Schoeps mics.

Chris

Thank you for the insights.

I for one would be interested to hear what uncompressed instruments sound like.

 

[Philbo King]

Thank you for the insights.

I for one would be interested to hear what uncompressed instruments sound like.

Any musician would be happy to demonstrate... Most of them are broke.

Here is a track I made completely uncompressed of 6 string acoustic guitar & singing recorded with a mid/side mic pair. If you listen in a quiet environment you can hear everything. Listen to it while driving in a car, and a lot of musical bits are inaudible.

It's track #14 ('Buzzard Bag Rag') at this link:

https://www.soundclick.com/philboking

 

[IAtaman]

Any musician would be happy to demonstrate... Most of them are broke.

Here is a track I made completely uncompressed of 6 string acoustic guitar & singing recorded with a mid/side mic pair. If you listen in a quiet environment you can hear everything. Listen to it while driving in a car, and a lot of musical bits are inaudible.

It's track #14 ('Buzzard Bag Rag') at this link:

https://www.soundclick.com/philboking

Thanks for that. I think I can hear you breathing in the intro

I was also hoping to get a sample of how instruments sound when recorded with Schoeps MK4 before being compressed in the studio.

 

[ChrisG]

My understanding of a digital mixing board (I only ever used analog back in the day) is that one needs to keep the analog input to the ADC close to 0dBFS. Otherwise, you are losing bits of resolution. If I have an input that peaks around -6 dBFS, then I've lost 1 bit right there. If I have a quiet passage thats -24 dBFS, then I've lost 4 bits. If you're recording something with a peak of 120 dB SPL and an average of 60-80 dB SPL, that's 40-60 dB (7 to 10 bits) that don't have any data most of the time.

This is true. Bit depth is often thrown away at the start of the signal chain. Sound engineers do that because the alternative is much worse:

Apologies if this is condescending. I don't know how much everyone knows about live sound, so I'll try to cover everything.

Soundcheck happens before a show. It's time spent with the musicians, sound engineer, and equipment, used for making sure that everything sounds good before the audience arrives. During soundcheck, I'll ask each musician to play their instrument (and/or sing) in turn, so that I can set the preamp (pre-ADC) gain set correctly. I'll also cue each channel in my headphones (Shure SRH1540, if you're interested) and make sure I'm happy with what I'm hearing. I'll probably throw a bit of EQ around as well. At the very least, a highpass filter to remove unwanted stage rumble.
Next, I'll unmute the stage monitors and talk to the musicians about what they need to hear to keep in time/tune. After dialling that in and making sure they're happy with the on-stage sound, I'll dial up the front-of-house mix (the one the audience will hear).

At the start, when I'm setting my input gains, I have a level meter on each channel. I keep an eye on this as I adjust, and typically aim for "just nudging into the yellow". That's peak levels just over 0dBU. The maximum input level is +20dBU, so I'm throwing away 3 bits, in HiFi terms.

The reason is this: headroom. I've seen drummers hit 12dB harder, going from soundcheck to the show. Adrenaline kicks in and people hit harder and sing louder. They're human.

If I'd set everything for as close to 0dBFS (+20dBU, for my desk) as possible, then the vocals, drums, (and probably guitar/bass amps, which were turned up to match) would be 12dB into clipping. Even an amateur would probably spot that it didn't sound good.


Chris

 

[vintologi]

I found this album, 192 KS/s and the ultrasonics correlate with the loudness of the music:

Katie Curley, Solo Harp: Leonard Cohen's Songs of Love and Hate in High-Resolution Audio

Download Solo Harp: Leonard Cohen's Songs of Love and Hate by Katie Curley in high-resolution audio at ProStudioMasters.com - Available in 192 kHz / 24-bit, 96 kHz / 24-bit AIFF, FLAC high resolution…

www.prostudiomasters.com

 

[egellings]

Good luck with that - both finding test tracks with meaningful ultrasonic content, AND hearing a difference if you do.

By ultrasonic, do you mean above audibility? If so, why bother?

 

[vintologi]

By ultrasonic, do you mean above audibility? If so, why bother?

I think people are just using it to refer to frequencies above 20Khz even in cases where it's still audible.

We can also have a scenario where the ultrasonic conternt by itself is not audible but it becomes audible due to interacting with lower frequencies.

 

[IAtaman]

I think people are just using it to refer to frequencies above 20Khz even in cases where it's still audible.

We can also have a scenario where the ultrasonic conternt by itself is not audible but it becomes audible due to interacting with lower frequencies.

How would ultrasonic content interact with sonic content and become audible? Does that interaction happen in the audible range as well, would, say 4KHz content, interact with 400Hz content and affect its audibility?

 

[antcollinet]

and the ultrasonics correlate with the loudness of the music:

Not sure what you mean by this.

 

[vintologi]

How would ultrasonic content interact with sonic content and become audible? Does that interaction happen in the audible range as well, would, say 4KHz content, interact with 400Hz content and affect its audibility?

If there is something very non-linear with regard to sound in the human body that might allow us to somehow detect ultrasonics when they interact with lower frequencies. There has been some studies indicating that this is indeed the case so i am going to do some more tests myself.

This might allow sound that is otherwise inaudible to make a noticeable difference with some music.

Not sure what you mean by this.

The ultrasonics are louder when when the music itself is louder indicating that it's not merely noise.

 

[ChrisG]

FWIW, I'm pretty happy accept the premise that ultrasound might be worth reproducing, and I like discussing the technical side of it.

I think that chimes might be a good way to test the audibility. They involve lots of little bits of metal hitting each other, so there should be plenty of VHF content there. If I had a set (of chimes!), I could do the following setup:

- Schoeps MK41
- Motu M4 @ 192kHz/24-bit

Then, to test audibility:
- Take the 192kHz file, down-sample to 44.1kHz, 48kHz, whatever.
- Then, up-sample back to 192kHz.
- ABX test the resulting files.

Up-sampling the file back to 192kHz eliminates the possibility that the DAC or audio software deals with files of different sample rates in different ways.


Chris

 

[MarnixM]

In another thread Zaphod stated:

Like I said: My Also Sprach Zarathustra (under Zubin Mehta) recording has several seconds of low organ bass at very high level.

 

[vintologi]

Schoeps MK41

Wouldn't earthworks M50 be a better option?

This does not look particularly great:

Or is something i am missing here?