An accurate likeness, as long as you ignore the facts. If we look at the actual reality, it is more like 13 bits / 88 kHz. Very good for an analog system, and way better than vinyl, but nowhere near a modern digital system.
By the way, here is a good paper on digital recording, and from an audiophile hifi vendor at that:
Sanders Sound Systems: Digital Recording White Paper.
That one doesn't use the bias frequency but the music itself (which may be an advantage or a disadvantage; in any case it is no reason to record the bias frequency).
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Not much.
Here is a comparison of the Studer A820 at 15 ips (light green, -2 dB @ 20 kHz) and 30 ips (dark green, -2 dB @ 32 kHz), compared to the Apogee PSX-100 digital system at various sample rates. As we can see, 48k (black) covers the 15 ips case, and 96k (red) more than covers the 30 ips case.
As to number of bits needed, the Studer has a A-weighted SNR of 77 dB, equivalent to 13 bits.
Interesting graph, pity it only goes to -6 dB.
The thread starter uses High Com and DBX Noise reduction, so unless you want to put an expander in the digital signal processing or after the DAC, you will need a bit more than 16 bits. 24 should be plenty.
Well, the direction is clear
What SNR do you think he can achieve even with DBX? It is basically a 2:1 compander, so you gain 6 dB. Still pretty far from the > 90 dB 16 bits is capable of.
My suggestion is that when the OP gets a 192/24 audio interface, he should download Audacity or some other free audio software, and post spectrograms of both one of his recordings and of a recording of pure silence recorded off the tape. That will once and for all show how many bits and what sample rate is really needed.
2:1 refers to the slope on a dB scale, so theoretically, 77 dB without compander becomes 154 dB with compander. In reality you don't want to compress very low levels because of breathing effects and rectifier inaccuracies and in any case, practical voltage controlled amplifiers aren't even close to a 154 dB signal to noise ratio anyway.
My guess is that his signal to noise ratio will be about 100 dB to 110 dB referred to the maximum level, that is at least what is specified in typical DBX compander manuals. If you want to keep the degradation of the noise floor below 1 dB, the ADC noise floor (quantization + circuit noise) must be at least 5.87 dB lower. This is certainly not feasible with 16 bits, at least not without special tricks such as noise shaping.
OK, even 100 dB would be great from tape, and might justify the risk of the dreaded DBX pumping effects, but do you think the dynamic range of any source material would be more than the 90-96 dB 16 bits gives us? Background noise in any studio will be around 30 dB, and how many mics can take 120 dB?
That is a different discussion altogether. The microphones need not be the limiting factor; I have two AKG C900 microphones, which are relatively inexpensive professional condenser stage microphones, and they have a specified maximum SPL of 139 dB at 1 % distortion and a noise level of 17.5 dB(A), equivalent to a dynamic range of 121.5 dB(A). Large-diaphragm microphones used in studios are often some 10 dB quieter than mine. Considering the super-professional equipment used by GeeVee, I expect his microphones are much better than my microphones.
Regarding studio background noise: there is no law of physics forbidding values below 30 dB(A) and in any case, the background noise is not a hiss but rather a combination of normal sounds.
Still, I think you have a good case if you would want to argue that 16 bits is enough for normal domestic listening. At least I've never heard the noise coming out of my CD player.
Regarding studio background noise: there is no law of physics forbidding values below 30 dB(A) and in any case, the background noise is not a hiss but rather a combination of normal sounds.
Still, I think you have a good case if you would want to argue that 16 bits is enough for normal domestic listening. At least I've never heard the noise coming out of my CD player.
Right - I am actually arguing that 16 bits is even enough for any professional recording once the recording has been normalized. You do want to do the actual record in 24 bits just to have headroom.
Hi All
I must admit that I did back out form the thread once the subject started to turn a little towards the technical pros and cons of analogue tape.
Suffice to say that I know what I hear, and just how rich it sounds.
Anyway I thought now would be a good time to give an update on my USB interface quest.
After much research, I settled on a demo unit of the Steinberg UR28M.
https://www.steinberg.net/en/products/audio_interfaces/ur_serie/modelle/ur28m.html
I managed to get this for a great price, and based on the reviews, I was certain that it would give a good idea of how well analogue recording would transfer to digital, despite the units max resolution of 24bit / 96Khz. (quite low it would seem, when compared to some other stratospheric units.)
The first thing that I found was how careful you have to be, to not drive the unit into clipping, which was quite easy.
The first transfers were good, but overdriving was very nasty and apparent.
After a few hours of experimenting, using the supplied software, and careful adjustment of the tape output levels, and backing off the dbx expansion to a ratio of 1: 1.2, I finally found the correct levels.
All I can say that the transfer to an uncompressed WAV file is very good, and I am well and truly advanced in transferring many of my master recording archives.
Whilst I have found the WAV file to be very good, I does seem to lack the richness when compared to the analogue master.
It is very hard to describe, but it sound like the mid frequencies are not rolling off at their natural decay. Almost like there’s a filter cropping off the last harmonics.
I am still experimenting, and it may very well be that this reduced richness that I am feeling could be to do with settings rather that the recorded material or transfer system.
Anyway, this has been, and will continue to be a lot of fun.
I will report back as soon as I have undertaken more transfers. – Next will be some vinyl!!
Kind Regards
George.
I must admit that I did back out form the thread once the subject started to turn a little towards the technical pros and cons of analogue tape.
Suffice to say that I know what I hear, and just how rich it sounds.
Anyway I thought now would be a good time to give an update on my USB interface quest.
After much research, I settled on a demo unit of the Steinberg UR28M.
https://www.steinberg.net/en/products/audio_interfaces/ur_serie/modelle/ur28m.html
I managed to get this for a great price, and based on the reviews, I was certain that it would give a good idea of how well analogue recording would transfer to digital, despite the units max resolution of 24bit / 96Khz. (quite low it would seem, when compared to some other stratospheric units.)
The first thing that I found was how careful you have to be, to not drive the unit into clipping, which was quite easy.
The first transfers were good, but overdriving was very nasty and apparent.
After a few hours of experimenting, using the supplied software, and careful adjustment of the tape output levels, and backing off the dbx expansion to a ratio of 1: 1.2, I finally found the correct levels.
All I can say that the transfer to an uncompressed WAV file is very good, and I am well and truly advanced in transferring many of my master recording archives.
Whilst I have found the WAV file to be very good, I does seem to lack the richness when compared to the analogue master.
It is very hard to describe, but it sound like the mid frequencies are not rolling off at their natural decay. Almost like there’s a filter cropping off the last harmonics.
I am still experimenting, and it may very well be that this reduced richness that I am feeling could be to do with settings rather that the recorded material or transfer system.
Anyway, this has been, and will continue to be a lot of fun.
I will report back as soon as I have undertaken more transfers. – Next will be some vinyl!!
Kind Regards
George.
Sure, but we might still debate what "know" means.
Do You Hear What I Hear? Amazing Auditory Illusions Explained.
Also, hearing is one thing, the laws of physics are another thing. Digital SNR and resolution is not an issue of opinion, but information theory - the stuff that makes it possible for you and me to have this discussion across thousands of miles.
That sounds like a gain issue. Are you sure you aren't using mic input levels?
Now you see why pro studios like to use 24 bits - you have a lot of headroom for avoiding the problem. Adjust your gain structure so that the loudest possible signal from your tape deck comes in at something like -10 dB FS. That takes care of any possible clipping, and allows you to normalize the gain in the digital domain after recording.
By the way, if you are honestly interested, I can make you a 16 bit FLAC file with > 100 dB of dynamic range (digitally synthesized, as that sort of dynamic range doesn't exist on real recordings), so that you can try recording it onto tape and seeing if your tape system can reproduce it. But it is a bit of work to do, so I won't do it unless you really are prepared to try...
Hi Julf
Thanks for your comments, and my apologies for the delay in responding. My work can take me to the outback of Australia at a moment’s notice. The life of a civil engineer working in the mining industry!!
I have seen similar auditory illusions previously, and did take a look at the one you noted. Unfortunately, in my opinion, they all suffer from one common failing. They rely on visual cues to predict an outcome; hence one must watch the video simultaneously.
It is nowhere near as effective if you just listen, but don’t watch.
I’m not quite sure as to what parallel you are attempting to draw here, but I suspect that you are trying to emphasise that visual cues play a role in our perceived quality of sound, regardless if the visual is static or dynamic.
For example, an upward moving picture of a stair case, coupled with a 2 tones, 1 being an octave below the other, adds to the illusion of a rising tone, even though it’s not, is no different to a technophile looking at some exotic cable, and convincing himself that he hears more.
Please correct me if I am wrong.
With respect to what I hear, I am convinced that the original master tape sound somewhat richer. I am not talking about wholesale differences, just very subtle ones.
I don’t want to make this a digital vs analogue debate, as both camps have their opinions firmly set in stone.
I have had friends comment that I am attracted to the “romance” of analogue: i.e. the whirring of tapes, the lowering of a tone arm onto vinyl. Being a more tactile medium, they believe that one feels more involved with the music because of this strange connection.
I have never known what to make of this, but it does make me smile.
Anyway, with regard to levels, my issue was definitely one of front end overload on peaks, and this has now been rectified. I also got to hear how nasty clipping in the digital world is.
The recordings are getting better and better, and I am delighted with the results to date.
Kind Regards
George.
VK5ZG
Thanks for your comments, and my apologies for the delay in responding. My work can take me to the outback of Australia at a moment’s notice. The life of a civil engineer working in the mining industry!!
I have seen similar auditory illusions previously, and did take a look at the one you noted. Unfortunately, in my opinion, they all suffer from one common failing. They rely on visual cues to predict an outcome; hence one must watch the video simultaneously.
It is nowhere near as effective if you just listen, but don’t watch.
I’m not quite sure as to what parallel you are attempting to draw here, but I suspect that you are trying to emphasise that visual cues play a role in our perceived quality of sound, regardless if the visual is static or dynamic.
For example, an upward moving picture of a stair case, coupled with a 2 tones, 1 being an octave below the other, adds to the illusion of a rising tone, even though it’s not, is no different to a technophile looking at some exotic cable, and convincing himself that he hears more.
Please correct me if I am wrong.
With respect to what I hear, I am convinced that the original master tape sound somewhat richer. I am not talking about wholesale differences, just very subtle ones.
I don’t want to make this a digital vs analogue debate, as both camps have their opinions firmly set in stone.
I have had friends comment that I am attracted to the “romance” of analogue: i.e. the whirring of tapes, the lowering of a tone arm onto vinyl. Being a more tactile medium, they believe that one feels more involved with the music because of this strange connection.
I have never known what to make of this, but it does make me smile.
Anyway, with regard to levels, my issue was definitely one of front end overload on peaks, and this has now been rectified. I also got to hear how nasty clipping in the digital world is.
The recordings are getting better and better, and I am delighted with the results to date.
Kind Regards
George.
VK5ZG
Hi Julf
Now this I am very interested in, and will be a very worthwhile exercise.
I give you my word that I will be honest in the evaluation.
You can email me the file to this address:
george.vlahos@aecom.com
Thanks
Regards
George.
a possibly helpful test would be ABX of digital recording of the analog tape output vs the analog tape out direct
if you hear all the analog tape recording "niceness" in the digitized tape output then it is simply what the analog recorder electronics, tape saturation, other behaviors are adding - not what digital is missing
I have read that bouncing to tape is considered a deliberately added sound effect in today's digital studios
if you hear all the analog tape recording "niceness" in the digitized tape output then it is simply what the analog recorder electronics, tape saturation, other behaviors are adding - not what digital is missing
I have read that bouncing to tape is considered a deliberately added sound effect in today's digital studios
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You might try setting the "average" signal level about 10dB MORE than the level you've found below clipping. Peaks on as well-done recording can be even higher than you expect. Also, see if you can hear any difference in the 'normal' recording by recording at such a lower level. The noise floor of the recording will still be well above that of the ADC and digital reproduction circuitry.
Another thing is it's probably not just be the "numbers" you're hearing - if you can afford it, try several different models of interfaces at the same sample rate and bit depth, and see if you can tell the difference between them.
Just because a device records and plays at a certain resolution doesn't mean it actually performs perfectly at that level, nor that higher numbers automatically translate to better sound. Otherwise, all 24-bit interfaces would have 144dB S/N ratio...
Great! I am travelling this week, but should have time to prepare the files during the weekend.
Actually, I realized that Monty of Xiph.org (developers of ogg vorbis and opus audio codecs) has already done exactly that.
1 kHz tone at 0 dBFS (16 bit / 48 kHz WAV)
1 kHz tone at -105 dBFS (16 bit / 48 kHz WAV)
So the way to do the test is to take the 0 dBFS file, and record it at 0 dB (or whatever gain setting that is just below clipping), and then record the -105 dBFS one at exactly the same gain setting.
Can you still hear the second tone on playback from the tape deck?
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