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Sample Rate Roll Call


Sample Rate Roll Call  

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  1. 1. Sample Rate Roll Call

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I'm wondering if there's still a move toward higher sample rates, or whether 44.1 kHz still works for you. No judgement here - I'm just curious, given that virtually all interfaces, computers, and software programs can handle whatever sample rate you choose. So...which one do YOU choose?

 

You can choose more than 1 option, if for example you favor one sample rate, but also do projects for clients who want a different sample rate. Vote away!

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My answer would be complicated. I never paid much attention to sample rates until more recently as I was more concerned with bit depth as an amateur, but my recording computer iMac is dead so I haven't been able to mess with much. Anything other than 24/44.1 on Windows causes a mess with my Yamaha MG10XU usb mixer, even though it's supposed to go up to 192. The iPad limits me to 44.1 as well.

 

My fancy audio interface goes to 96khz. As mentioned though I've just used 44.1 as I didn't pay too much attention...still stuck in the CD era I guess. When things are running again, I'll probably work at 96; there's a limit to how large of files I want to deal with and how much strain on the processor I want to do. Then again I have to get a new interface, which will probably do 192, so I'm going to mess with that a little.

Yamaha: Motif XF8, MODX7, YS200, CVP-305, CLP-130, YPG-235, PSR-295, PSS-470 | Roland: Fantom 7, JV-1000

Kurzweil: PC3-76, PC4 (88) | Hammond: SK Pro 73 | Korg: Triton LE 76, N1R, X5DR | Emu: Proteus/1 | Casio: CT-370 | Novation: Launchkey 37 MK3 | Technics: WSA1R

Former: Emu Proformance Plus & Mo'Phatt, Korg Krome 61, Roland Fantom XR & JV-1010, Yamaha MX61, Behringer CAT

Assorted electric & acoustic guitars and electric basses | Roland TD-17 KVX | Alesis SamplePad Pro | Assorted organs, accordions, other instruments

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48Khz here. Orchestral sample libraries are all at 48Khz, so this is one rate for the whole studio. I sometimes record solo piano at 96Khz, but it's just because "I can". NARAS recommends 96Khz for acoustic recording. People can't hear it (our ears literally don't have structures of the right size to resonate above 20Khz), but they argue that it is the best choice for master recordings since processing and other things will be done to the signals before release. Its a "safety margin" argument, and an archival one. It does allow for shallow filters to be used in the future if desired.

 

At this point computers are so powerful that it hardly matters, and storage is cheap. But when working with my samples, everything is 48Khz and I don't think about it at all. All digital release platforms accept 48Khz & 96Khz, I don't produce CD's, so 44.1 is never used here.

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Guess I'll toss my 2 cents in, I voted yesterday.

When I started using a computer based recording system I had a 2008 Mac Pro with 6 gigs of ram and Firewire 400 was my fastest throughput.

I had an early Mackie Onyx interface and at 44.1 I could get latency down to a still audible but workable level without glitching.

 

Using a MacBook Pro with 16 gigs of RAM and a Thunderbolt 2 interface I can easily use 48k with no noticeable latency. As some have mentioned, that is now standard for video production.

I am able to run at 96k with no problems as well. Besides the potential (if seemingly inaudible) benefits of having plenty to work with when processing, the issue of two notes creating a lower subharmonic have made interesting reading. So I am playing with using 96k but I am very open to the possibility that the subharmonics created by any truncated notes are probably inaudible even at 44.1k.

 

I'll probaby not use 96k often and maybe never but I haven't decided yet so I added it to 44.1 and 48.

It took a chunk of my life to get here and I am still not sure where "here" is.
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I remember reading once that the conversion rate math worked better at 48khz over 44.1 and I don't recall the details. It's something I should know but I don't keep up the way I used to. I've been sticking with that more or less for quite a few years now. I wouldn't want my live project files to be any larger.
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I remember reading once that the conversion rate math worked better at 48khz over 44.1 and I don't recall the details. It's something I should know but I don't keep up the way I used to. I've been sticking with that more or less for quite a few years now.

 

I believe that the tale of conversion to double or half the sample rate came from the believe that all you need to do is take every other sample to convert down and double every sample to convert up. That actually isn't a very good way of doing it. The proper way to do sample rate conversion, and these days there's enough computer horsepower to do it so everyone does, is to virtually reconstruct the waveform, then re-sample it at whatever sample rate you want, up or down. Since you're, in essence, restoring the analog signal as accurately as possible, that's the best starting point. Then you can sample it at the desired rate.

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I use 44.1 kHz because it stresses out the computer less, and I don't hear any difference between 44.1 and 48 kHz. For soft synths and amp sims, I'll upsample and render at 96 kHz and then convert the rendered audio down to 44.1 kHz. This gives the advantage of 96 kHz sound quality (i.e., no foldover distortion) at 44.1 kHz.

 

The biggest problem with 44.1 kHz is that I have quite a few pieces of hardware that run at 48 kHz natively, so I can't do a digital-to-digital transfer without real-time sample rate conversion. I never should have sold my Alesis AI-1, I think it can do that...

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Hmmm...it appears that hardware sample rate converters are few and far between. All I've seen that's not super-expensive is the Behringer SRC2496 and Ultramatch, both discontinued and found in places like reverb.com and eBay. Anyone have any experience with these guys?
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The biggest problem with 44.1 kHz is that I have quite a few pieces of hardware that run at 48 kHz natively, so I can't do a digital-to-digital transfer without real-time sample rate conversion. I never should have sold my Alesis AI-1, I think it can do that...

Yes, it can...and I still have one, and I don't really use it. :)

 

dB

:snax:

 

:keys:==> David Bryce Music • Funky Young Monks <==:rawk:

 

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I'm surprised at how many people are using 48 kHz. Do you hear a difference compared to 44.1? Do you do a lot of video work? Is it the Spinal Tap effect, where 48 kHz goes up to 11, but 44.1 kHz only goes up to 10?
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I'm surprised at how many people are using 48 kHz. Do you hear a difference compared to 44.1? Do you do a lot of video work? Is it the Spinal Tap effect, where 48 kHz goes up to 11, but 44.1 kHz only goes up to 10?

 

 

I see video as being an important way to communicate musical productions.

I am also a photographer, visual arts are part of my creativity.

 

I have a great friend who spent 35 years in local television, he performed all duties - from cameraman to producer/director to writer, etc.

 

Here is a piece of his work, his song (copyrighted so I can share), recorded at a friend's studio (great singer on backups) and created/directed the video concept.

It takes the song to another level in my opinion.

 

https://www.facebook.com/permalink.php?story_fbid=2769977883288462&id=100008288330364&notif_id=1589490889302673&notif_t=comment_mention

 

And 48khz is the standard for video. Latency is not a problem, I see no reason NOT to use it.

It took a chunk of my life to get here and I am still not sure where "here" is.
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All my tracks and masters are 24bit 44.1K (I record at 32bit floating, but with 24 bit AD/DA, I can't see how it's anything more than 24bit).

 

I'll dither and bounce down to 16 bit 44.1K, but I've never ventured into other sample rates.

 

Todd

Sundown

 

Working on: The Jupiter Bluff; Driven Away

Main axes: Kawai MP11 and Kurz PC361

DAW Platform: Cubase

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Hmmm...it appears that hardware sample rate converters are few and far between. All I've seen that's not super-expensive is the Behringer SRC2496 and Ultramatch, both discontinued and found in places like reverb.com and eBay.

 

Hardware sample rate converters go back a long time. Roland had one, so did Weiss, and Prism made a couple of A/D/A converters with input A/D and D/A clocks so that in the digital-in-to-digital-out mode you could send out a different sample rate than what came in. The good ones were very expensive because they actually converted digital to analog and then back to digital, and you wanted the highest quality converters as you could get as to not introduce distortion in the intermediate step. One of the most common applications in the early days wasn't to go between 44.1 and 48 kHz, but rather go from 44.056 kHz to 44.1 kHz. Early digital recording systems utilized a PCM adapter to a VCR as the storage medium. The Sony PCM-F1 was the first common one. Since it was standard video, and they wanted to synchronize the sample rate with the frame rate, they needed this odd rate. But when that was imported into a DAW or other digital playback system, there was a small change in pitch, as I recall around 0.1 cent. Some people just ignored it but those who could afford it or didn't want to get caught used an expensive SRC.

 

Of course once computers got powerful enough to do the calculations that gave the best conversions (and there were several algorithms floating around, and probably still are) the dedicated hardware faded into the sunset. For the majority of users, just "save as" whatever sample rate you wanted works just fine. And a lot more goes out the window when you convert to MP3 or other data-reduced format.

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I remember reading once that the conversion rate math worked better at 48khz over 44.1 and I don't recall the details. It's something I should know but I don't keep up the way I used to. I've been sticking with that more or less for quite a few years now.

 

I believe that the tale of conversion to double or half the sample rate came from the believe that all you need to do is take every other sample to convert down and double every sample to convert up. That actually isn't a very good way of doing it. The proper way to do sample rate conversion, and these days there's enough computer horsepower to do it so everyone does, is to virtually reconstruct the waveform, then re-sample it at whatever sample rate you want, up or down. Since you're, in essence, restoring the analog signal as accurately as possible, that's the best starting point. Then you can sample it at the desired rate.

 

That may have been it, been so long ago I don't really remember. In any case 48k seems to be a default with some of the gear I'm using, my hardware/software seems to work great with it so I just go with it. I'm actually pretty easy to get along with.

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