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Sampling 24 bits 96khz..


Bachus

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Can you hear the difference?

 

I tried both 16 bit/48khz and 24 bit/96 khz samples and dacs, but could not hear the difference..

 

Might be i am getting old.... But considering the fact that 24bits/96 khz uncompressed samples are 3 times as big, it is something to comsider..

Korg Kronos 88, Yamaha Tyros5 (76), Integra 7, macbook pro/mainstage
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Do you mean recording tracks or sampling a single sound for a rompler?

 

I myself cannot hear the difference between 16/48k vs. 24/96k when listening to s single sound or track. Some people claim to hear a difference with a higher sampling rate, but I do not.

 

That said, there are good reasons to do musical recordings at 24 bits - the extended dynamic range is a big plus. This is especially true when multiple tracks and effects will be digitally summed together.

 

My default recording set-up is 24 bit/44.1kHz. I don't use 48kHz sampling rate unless required to do so for media reasons.

 

If I was doing a sampling of an instrument for a rompler, I would use 16/44.1k because of the smaller file size. If I want the very best instrument sound, I would record it directly and not use a sampler.

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Sampling for what use? Live performance usage? Go with small files and faster load times. Live they sound fine.

"It doesn't have to be difficult to be cool" - Mitch Towne

 

"A great musician can bring tears to your eyes!!!

So can a auto Mechanic." - Stokes Hunt

 

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What you're referring to when you talk about sampling and bit rates is how accurately you're attempting to capture the frequencies present and their dynamic range.

 

I guess to compare it, it's sort of (though not exactly) like getting an new TV. What is the resolution in pixels and how accurately does it reproduce the colors. The questions are, how good are our eyes, how much does it even matter the further back you get from the TV, what are the lighting conditions in the room, how much does it affect wether or not you enjoyed the movie, etc. etc.

 

In your case... how good is your signal chain to the DA, how good is your DA, how good is your AD, how good is your signal chain out of the PC - the amp, the speakers or headphones, and how good are your ears, how good is the song, and at what point would a lesser quality recording affect your desire to listen to the song? Would it ruin it for you if it was recorded at 16bit/44.1? Probably not. I know some of those rips from the early days of MP3 encoding didn't please my ears. But you do get to a point where it's indiscernable so, at that point, what's the difference? If storage is a concern, don't feel bad about going with a "lower quality" recording.

 

When Mike Martin says it depends on what is being recorded, I THINK he is referring to the frequency and dynamic range of the source material being recorded. Sometimes aspects of the recording are flat out lost or misrepresented (even false) if the mathematics involved are not capable of capturing and reproducing what's there. The wink also makes me feel like he's teasing a bit, implying that if what you're recording isn't any good then it doesn't matter at all.

 

Anyway, don't fall in love with numbers. Your ears don't lie (well they do - but they're your only way of getting audio information so what does it matter if they lie, it's all you have to work with). All things considered, 16bit/48k is capable of capturing more than anything your ears can likely discern. It's only the lower grade of recordings where you might say, "this recording sounds like ass".

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It depends on what is being recorded. ;)

 

Interesting...

 

Could you add some examples to that statement please and shed some light into my blind eyes..

 

Well you haven't answered the question about what you're really trying to do. But the extra bit depth going from 16bit to 24bit is going to give you a much wider dynamic range. So if you need to record soft elements in addition to very loud ones, or something with a very long decay (fade out), then 24bit is desirable. If the material is pretty static in amplitude then there is no reason for it. As for frequency response, it probably isn't worth it in terms of sample size and storage, certainly not if you can't hear it.

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If you are doing risers, drops or one shots it doesn't matter. Sometimes 8-bit is what you want. If you are trying to build a virtual piano or some instrument where you need a wide range of dynamic response you want the highest audio resolution available. I don't do such things.

 

A Mirage can be really cool when you WANT grainy sounds.

"It doesn't have to be difficult to be cool" - Mitch Towne

 

"A great musician can bring tears to your eyes!!!

So can a auto Mechanic." - Stokes Hunt

 

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For sure it depends on the source you use to play back the sampled signal. You could take a 1950s microphone and a 12 bit emulator, and put the output on a guitar amp: would you think the sound would improve with more bits or sampling frequency ?

 

Let's look at the technological/scientific basis for sampling. The theory (my theory book was from the 60s, reissued in the 80s IIRC) states that a if a (analog) signal has no frequency components higher than half the sampling frequency, sampling it can be completely turned back to a analog signal, with a perfection only limited by the number of bits being used . So if a decent signal is put on a Analog to Digital Converter, the sampled signal can be turned back almost perfectly to the original analog signal, provided you use enough "vertical" resolution (e.g. 16 or 24 bits).

 

There's a big "but" in this theory: to get the *original* analog signal back from the samples, you need a very long and compute intensive filter called the reconstruction filter. In theory, such filter takes infinitely long, and a lot of sine(t)/t computations, for which there is no algebraic simplification or something. It is interesting through, that a mathematically perfect theory exists which states that under the given conditions, or wonderful digital recordings can be perfectly put back into an analog signal.

 

Now most Digital to Analog Converters, usually chips of a couple of known brands (for electronicists) do only a very easy and short version of the required reconstruction filtering, with as result that anything digital sounds, well, uh, digital, maybe chilly, cold, maybe a bit clarity challenged in the mid range, grungy in the high frequencies, blares and distorts too much, etc.

 

So apart from making sure you make no aliasing distortion at the input of the ADC, and refraining from digital processing (which always mangles the signal unnaturally and incorrectly compared with analog equipment, as it is), you DAC is going to be more of a problem for achieving digital High Fidelity than the sampling frequency and the exact bit resolution.

 

With computers, you often can play a bit with how you play back digital signals, which output sampling frequency and bit accuracy you use, and in some cases (I know it's possible on windows 8pro) whether you upsample or not. Clearly if you take a high definition audio track (say 24bit/96kHz which is some sort of studio norm) like I enjoy from "HDTracks", and you play it back from a CD player or a computer sound card that is set to 16 bits / 44.1 kHz, you're wasting bits. So don't do that, make sure a high res file is actually played in high resolution.

 

Some CDs are made with certain CD players or media players in mind. This from good music sources can entail a certain combination of deep studio processing that somewhat compensates for the errors certain standard DACs make. Playing a random high res recording may easily be less satisfactory to listen to, because using more bits is beneficial mostly with large dynamic range materials and accurate speakers, and using a higher sampling frequency is actually disproportionally better for the high frequencies (at 96 or even 192kHz, the highest audio frequencies are a lot further away from half the sampling frequency than just a factor 2 or 4), but even going to 192kHz isn't a complete solution for the reconstruction errors. Given the dimensions of high frequency audio waves, probably about a Mega Hertz (!) sampling frequency will not anymore sound "digital" and similar to a good quality analog record being played, also in a reflecting room (where the reconstruction errors soon sound terrible, in spite of all kinds of attempts of speaker builders to filter the signal: the correct reconstruction filter cannot be approximated ny proper speaker design).

 

So in short, there's the DAC's reconstruction error that contributes the most to the digital signal sound incorrect, even though a simple *correctly executed* comparison between the same recording in 16 bits / 44.1 kHz and a properly made 24 bit / 96 kHz on a decent audio system most likely will make the difference transparently clear within seconds, even for audio laymen.

 

T.V.

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It depends on what is being recorded. ;)

 

Interesting...

 

Could you add some examples to that statement please and shed some light into my blind eyes..

 

Well you haven't answered the question about what you're really trying to do. But the extra bit depth going from 16bit to 24bit is going to give you a much wider dynamic range. So if you need to record soft elements in addition to very loud ones, or something with a very long decay (fade out), then 24bit is desirable. If the material is pretty static in amplitude then there is no reason for it. As for frequency response, it probably isn't worth it in terms of sample size and storage, certainly not if you can't hear it.

 

Thanks for the answer mike...

 

I am in the process of buying some sample libraries to create my own multisampled sounds with in Kontakt..

 

And i was thinking with 7 bits (0-127) midi depth.. How much would the extra depth add for me? I can see how it matters in the mix. But for samples?

 

Korg Kronos 88, Yamaha Tyros5 (76), Integra 7, macbook pro/mainstage
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It seems like only yesterday.

Indeed, it wasn't that long ago when many of us were satisfied with 8-bit performance.

 

Have we gotten greedy?  ;-)

 

 

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On anything except headphones the number of bits and all kinds of distortion related to timing issues with the samples are going to affect the reverberation in the listening space. A lot of waves come back from the the room or hall, and there are only so many 441/16 signals tricks that will sound a bit nice.

 

Some people try low bitrates, usually in conjunction with a signal path that doesn't just play a sample, but affects the signal, like various forms of FFT based processing. Playing withe errors connected with aliasing errors, timing (and transient) rounding because of sampling (like a square wave being rounded to fit samples (a 1 kHz wave consists of 44.1 samples, which somehow is a rounding error of a few percents), and the DACs reconstruction wave preferences (looping a live signal back through the ADC-DAC) all isn't a pretty sound.

 

Once you make a sounds which form a very small sub-class of all possible CD (or high res) sounds, it is possible that before a certain averaging effect you need a few bits, and you might want to play with moderately small signal deviations to see what your setup will do. That's not the same strictly plugging in a 16 bit rounder effect in a neutral signal chain.

 

T

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