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They need to be understood without some ancient tape jockey babbling about headroom and frequency response.
Hi, all. I think I just recognized myself, there!
Actually, I completely agree with the 16/24 argument, but I think that, sometimes, we assume that better equipment makes better recordings, without acknowledging the difference made by the skill of the operator.
In my view, the differences between 16/24 are hard to detect, whilst the skill of the operator makes a huge difference. Some of us who remember working with tape, when you had a noise floor you could measure on the VUs and nothing was tracked without a compressor to preserve the s/n ratio without switching in the Dolbys, are quite happy to stick with 16 bit.
Just sayin....
ROG.
I've worked in both worlds, tape and digital - in many formats. I don't have measurement equipment at home that is precise enough to show the difference between 16 and 24 bit. What I do know is that since switching over to 24 bit, I've not had to wring my hands about trying to maximize the input range of my A/D for every track. I can be quite sloppy with it, and when I then go back and boost those low-level recorded signals, they are still doggoned clean sounding. This is not the case when doing the same practice with 16 bit. I can indeed start to hear hiss if I take a recording that I made perhaps 25 dB down in 16 bit, then boost it up 20 dB. Try that experiment for yourself.
-Scott
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Quote:
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They need to be understood without some ancient tape jockey babbling about headroom and frequency response.
Hi, all. I think I just recognized myself, there!
Actually, I completely agree with the 16/24 argument, but I think that, sometimes, we assume that better equipment makes better recordings, without acknowledging the difference made by the skill of the operator.
In my view, the differences between 16/24 are hard to detect, whilst the skill of the operator makes a huge difference. Some of us who remember working with tape, when you had a noise floor you could measure on the VUs and nothing was tracked without a compressor to preserve the s/n ratio without switching in the Dolbys, are quite happy to stick with 16 bit.
Just sayin....
ROG.
I always find these 16/24 discussions amusing. I started out with a Tascam Portastudio 144 in 1979 (4-track/cassette). You want to talk about noise floor? 
I agree with ROG - the gear will not make you a better operator. It will, however, remove a lot of the sonic obstacles that plagued the "good old days".
Regards,
Bob
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And don't forget that 0 on your digital meters is about 18dB hotter then 0 on an analog desk.
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I can indeed start to hear hiss if I take a recording that I made perhaps 25 dB down in 16 bit, then boost it up 20 dB. Try that experiment for yourself.
Scott.
The first thing I ask myself is - why would someone with your obvious knowledge and experience want to record anything 25db down?
One thing you must remember is that you have a theoretical dynamic range of your digital recording medium, but an actual one based on the noise floor from the inherent noise in the signal chain. If, for example, you put a cheap mic in front of an old Marshall tube amp, your dynamic range is, I think the technical term is "shot-to-hell". Even just a good mic through an expensive pre-amp will have a noise floor higher than the digital medium. The experienced engineer will will know how to minimize these problems and get the best results out of the available gear.
You'll notice that I agreed with the main 16/24 argument in my earlier post. I was just making the point that we encourage people to up-grade more often than we tell them to get some specialized training.
ROG.
Last edited by ROG; 06/08/12 10:16 AM.
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Journeyman
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I remember my home studio. I would mix down to a DAT Recorder. This was around 1990? I'd send the audio synced with midi from my Atari ST right to the DAT machine. It took some time to get used to setting levels. Many of my first mixes were full of distortion. What an ugly sound it was. I remember it sounding like blown speakers. I started to use a compressor/limiter to help me keep it even.
Wayne,
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Quote:
I remember my home studio. I would mix down to a DAT Recorder. This was around 1990? I'd send the audio synced with midi from my Atari ST right to the DAT machine. It took some time to get used to setting levels. Many of my first mixes were full of distortion. What an ugly sound it was. I remember it sounding like blown speakers. I started to use a compressor/limiter to help me keep it even.
Wayne,
That was the old 16 bit Audio Engine.
Today, all audio recording programs are using the 32 bit engine. No more digital "thwack" from overruns, which is another way of saying digital clipping.
--Mac
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Quote:
... I think I just recognized myself, there!
Actually, I completely agree with the 16/24 argument, but I think that, sometimes, we assume that better equipment makes better recordings, without acknowledging the difference made by the skill of the operator.
In my view, the differences between 16/24 are hard to detect, whilst the skill of the operator makes a huge difference. Some of us who remember working with tape, when you had a noise floor you could measure on the VUs and nothing was tracked without a compressor to preserve the s/n ratio without switching in the Dolbys, are quite happy to stick with 16 bit.
Rog,
Hoping that the analog chops we worked so hard to acquire could be translated to digital audio processing is common to most of us oldsters. 
______________________________________________________________
In Linux, audio information is transmitted from one application to another for special-purpose processing, and the bit depth of the transmitted files can be 32 bit, or 64 bit. Files can even be encoded as 32 or 64 bit floating point.
I had not grasped the significance of this until reading Scott's description of "quantization noise" :
"What 24 bit recording does do is give you a better signal to quanitization noise ratio than lower bit depths.
For every bit depth there is an additional 6 dB of SQNR. You can go off and google and wikipedia this to get the complex math but I'll try to keep it simple here.
Here's the way to think about this. Let's pretend you have a pure sine-wave at a fixed peak-to-peak voltage. Let's pretend that we take this signal and we amplify it so that the the peak-to-peak voltage is the same as the peak-to-peak voltage that the A/D converter can handle. You can quantize this into 16 bits, or 2^16 values, or one of 65,536 possible values. The sine wave will have very tiny little stairstepped voltage values into which the wave is encoded.
Now, let's take the same signal, and quantize it into a 24 bit A/D converter, which has one of 16,777,216 possible values. As a result, the stair-steps are WAY smaller with 24 bit A/D conversion than with 16 bit conversion.
The 'stair-steppy-ness' of the signal is the quantization noise. Yet another way to think about this is to think of the smallest possible signal that can be encoded, a signal so quiet that the A/D converter switches between the lowest possible value at zero and the next highest value. It switches back and forth between 0000000000000000 and 0000000000000001. When those values eventually get sent back to D/A conversion, that twitching between the two values tweaks the output D/A and generates an unintended analog output noise: Quantization noise. (For those of you that know binary formats, and the difference between signed and unsigned stuff - bear with me here, just trying to make a point)
For a 24 bit recording, if we adjust the signal down so that the switching back and forth occurs between 000000000000000000000000 and 000000000000000000000001, the lowest two values, and as it goes back to D/A the tweak is much smaller and hence the quantization noise is much smaller. In fact the quantization noise will be roughly 48 dB lower than the 16 bit quantization noise.
Note - no magic with the dynamic range occurs.
The big benefit to recording with 24 bit recording over 16 bit is that one doesn't really have to worry nearly as much about using the full dynamic range of the A/D converter in order to get a nice signal to quantization noise ratio, and the little stair steps that occur in the digitized data are peanuts in comparison to 16 bit.
In other words, you can be quite a bit less careful about it, and just get on with recording."
Thank you, Scott! 
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In other words, you can be quite a bit less careful about it, and just get on with recording.
Oren.
You don't think that good practice once learnt, is something which becomes second nature? Once we start to tell people it's ok to be careless about one thing, we then have to specify HOW careless and to differentiate between that and the things that DO need attention.
Start of the slippery slope?
ROG.
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I can indeed start to hear hiss if I take a recording that I made perhaps 25 dB down in 16 bit, then boost it up 20 dB. Try that experiment for yourself.
Scott.
The first thing I ask myself is - why would someone with your obvious knowledge and experience want to record anything 25db down?
One thing you must remember is that you have a theoretical dynamic range of your digital recording medium, but an actual one based on the noise floor from the inherent noise in the signal chain. If, for example, you put a cheap mic in front of an old Marshall tube amp, your dynamic range is, I think the technical term is "shot-to-hell". Even just a good mic through an expensive pre-amp will have a noise floor higher than the digital medium. The experienced engineer will will know how to minimize these problems and get the best results out of the available gear.
You'll notice that I agreed with the main 16/24 argument in my earlier post. I was just making the point that we encourage people to up-grade more often than we tell them to get some specialized training.
ROG.
Rog,
It's not that I make recording at low levels a matter of practice, but allowing plenty of room does make for less hassles during mixing, not having to apply attenuation in the box to each track, so that when I sum the tracks, I'm not summing to something that's overdriving.
It's just a matter of convenience, I suppose. When I think back to when I had only 16 bit available, I spent a lot of time worrying about maximizing the A/D for each recorded track, and then dialing down from there inside the box.
I do almost no worry of that now and just get on with recording the next track and on to mixing. It's been a very rare occurrence when my songs have summed to the point where I have to go in and adjust gains of individual tracks down - and my 2 channel mixdowns sound very clean.
That wasn't the case in my previous 16 bit life.
I will say that my mixdown 'process' is much less structured than most here, as I don't really have a mixer paradigm in the DAW that I use and I'm fine with it being that way.
Hopefully this is making sense.
-Scott
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This has been a very informative thread.
Thanks guys.
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Scott.
Forgive me if I appeared to be critical of your recording technique. In threads like this, I tend to court controversy, which is a character flaw I find difficult to throw off!
At the end of the day. it's the final mix which we're judged on, not the way we got there.
I learnt my recording in the 60s and 70s, when attention to detail such as levels throughout the signal chain was essential, bearing in mind the equipment we had. Old habits die hard. I have to say, though, there's no way I would want to go back to tape and enormous mixers with no automation.
BTW, if you like maths, which you seem to have a good grasp of, try factoring in the fact that quantization noise is frequency sensitive - it gets scary!
ROG.
Last edited by ROG; 06/09/12 01:30 AM.
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Rog,
No harm no foul. Your question was valid. Please understand that I still care about gain staging and so forth and I do still use most of the A/D available range when I record - I just worry less about it now. It used to make a difference. Now, with the limited time I have for this hobby, I spend less time worrying about it and it's none for the worse as a result.
Regarding maths, I learned this stuff about 10-12 years ago in the EE-638 Digital Signal Processing class at Purdue University. I was the only mechanical engineer in the course. The professor was Michael Zoltowski - great guy who knows a shed load of information about DSP. His specialty is spread spectrum communications. I bugged him a great deal with my questions about applications to audio for music purposes.
My favorite moment in the class was the exercise we went through to show how sending this stair-stepped digitized signal through the D/A filter actually DOES result in a smoothed analog signal (up to a certain frequency) through the superposition of each bit's impulse response through the D/A filter stage.
That was an eye opener for me.
Last edited by rockstar_not; 06/09/12 06:32 AM.
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Quote:
You don't think that good practice once learnt, is something which becomes second nature? Once we start to tell people it's ok to be careless about one thing, we then have to specify HOW careless and to differentiate between that and the things that DO need attention.
Start of the slippery slope?
ROG,
Music as art has intrigued me, and the science of music is fascinating, but the craft of making music - from composition to performance to recording to post production - is what really puts the frost on my pumpkin.
In that context, the care with which a craftsperson approaches an audio project is paramount; the defining quality of the experience, for me. (not that you'ld automatically know that from some of the doo-doo I've generated - but we're all entitled to our mistakes... )
So "yes" to good practice as second nature, and "no" to slippery slopes, analog or digital! 
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