Mixing In The Box / Outside The Box

[aarowsmith]

Ooh a three year old thread risen from the dead. I'll play

To me the biggest difference when I turn on my analog processors is I can no longer see the audio and I have to really start using my ears (Imagine that for audio). It seems like with DAWs there are an awful lot of threads of "The wave form doesn't look big enough", "how should a spectrograph of my audio look", "what should the peak meter read", "does this look right", "what numbers should I be seeing in the gain reduction field", etc etc etc

While I have to use a computer to record and mix to, I try and do as much as possible mapped to controllers or through analog processing with the screen turned off.

A blank screen takes away all the distraction of "Does this look right" and just like using tape, where a piece of unrecorded tape looks just like a piece of recorded tape, all that matters is "do I like the sound?"

Both digital and analog are both actually an analog of the original sound, in neither case does the actual sound travel through the processors, they just use different mediums and signal types to represent the sound

If I might add to your excellent observations of using digital verses analog audio;

With digital you also have to worry allot about the way you dither down to lower sample rates, such as going from your recorded 24 bit audio down to say 16 bit audio for CD mastering. If you do not do this process correctly it can sound badly muffled. And or if you have say recorded sampled loops in 16 bit and also recorded audio samples at 24 bit, then processing them all in the same way digitally can result in changes to the original audio quality that you may not like to hear. For example a 24 bit master mix not sounding any where close to the CD 16 bit master mix. This does happen allot not only to me and but many of my friends in mastering albums. But both sample rates should sound close right, because on their own it is hard to tell a difference in sound quality between a good 16 bit recording versus a 24 bit one, if done correctly. So then why does the CD master mix sound so different than the same song and same mix played inside the box ??? Mmmmmmm......

With analog audio you don't have to worry about sampling rates and dithering down sample rates to match lower quality sample rates.

With analog audio you don't have to worry about clock's not lining up correctly and timing errors resulting in audible digital jitter, and hence crappy audio.

Although with analog if you use too much EQ processing you can end up with audible frequency harmonic time distortion that causes certain frequencies to resonate and ring. Which may or may not sound bad to the listener. Too much processing of anything can be bad sounding, depending on what you are trying to shoot for.

With analog gear all the audio signals are brought in to the same highest fullest resolution. With analog audio there is no digital loss due to lower resolution digital stair stepping sampling of the original audio wave.




Also may I argue a point here:

In your observation of "Both digital and analog are both actually an analog of the original sound", well that is just not the case with analog audio going into digital. There is a major digital conversion sampling process going on that changes the original analog audio into one's and zero's, this is both a big physical change and an audible change.

However with a analog signal going into analog gear (unless it is being recorded into a analog tape machine) it does not change the original analog signal, it does not physically change it or sample it, unless of course you want it to. For example, changing the original analog audio through some form of analog processing.

That is the huge difference and the point that I am trying to make very clear here, about the differences between your pure analog sgnal going into digital versus it going into analog gear.

To simplify, with digital there is always a drastic sampling conversion change going on, with analog gear there is not such a drastic conversion taking place, and hence the audio stays much true'er to life, if I may put it that way. And no matter how good the software programmer's math is, there is something that gets lost in the digital translation.

I know what most people say about digital audio, it is just a computer doing the math in measuring and recording and sampling the numerous analog audio voltages in analog audio, and then hopefully accurately restoring them back to a analog audio wave again, hopefully sounding the same as when the audio came in. But computers are not always perfect in measuring and in error correction, and at times they also throw away bits of useful information as well.

Whether or not you can actually can hear this taking place and are not be able to tolerate it, is well maybe a mute point........or maybe not ?

 

[Farview]

With digital you also have to worry allot about the way you dither down to lower sample rates, such as going from your recorded 24 bit audio down to say 16 bit audio for CD mastering.

Dithering is for changing bit depth, not sample rates.


If you do not do this process correctly it can sound badly muffled. And or if you have say recorded sampled loops in 16 bit and also recorded audio samples at 24 bit, then processing them all in the same way digitally can result in changes to the original audio quality that you may not like to hear. [/quote]That might have been a problem in 1998, but all daws run at 32 bit floating point so the bit depth of the actual audio clips is irrelevant.

For example a 24 bit master mix not sounding any where close to the CD 16 bit master mix. The both should sound close because it is hard to tell a difference in sound quality between a good 16 bit recording versus a 24 bit one, if done correctly.

Usually the difference in sound between the 24 bit and 16 bit masters have more to do with the processing during mastering than anything that happened with dithering. Of course, all of that is on purpose. In your defense, there were some pretty bad down-sampling programs out there 10-15 years ago, but that is a thing of the past now.

With analog you don't have to worry about sampling rates and dithering down sample rates to match lower quality sample rates, and or timing errors with digital sampling clocks that audibly click and are not matching up in digital sync, and hence destroying the audio quality.

Nope, just wow, flutter, alignment issues, tape calibration, etc...

Again, the digital clocking problems really don't come into play much anymore, since most people don't have racks and racks of individual pieces of digital gear that all needs to be synched. That was a problem in the early 1990's, but your computer will always be in synch with itself and the interface that's attached to it.

With analog gear all the audio signals are brought in to the same highest fullest resolution. With analog there is no digital loss due to stair stepping sampling of the original audio wave and then math errors getting rid of some of the one's and zero's later on resulting in dull sound and audio dynamics, errors in harmonics, less stereo imaging. All of this from lost digital translation.

There is not stair stepping. That graphic that you saw depicting how digital sampling works is over-simplified and basically wrong. If you take an oscilloscope and connect it to the output of a digital converter, you won't see any stair steps. It will look just like the waveform that was fed into it.

Think about it. If the stair steps were there, they would exist at the sample rate. The sample rate is much higher than we can hear AND is filtered out. (much like the bias signal used to get audio onto tape) So they can't be there.

I have to say that in your observation of "Both digital and analog are both actually an analog of the original sound", well that is just not the case with analog audio going into digital. There is a digital sampling process going on that changes the original analog audio into one's and zero's, this is both a physical change and a audible change.

Definition of analog: A person or thing seen as comparable to another.

Again, the analog tape recording process adds the audio signal to a super high frequency bias signal which then is used to magnetize some rust. It is both a physical and an audible change.

 

[aarowsmith]

That is why I said this:

However with a analog signal going into analog gear (unless it is being recorded into a analog tape machine) it does not change the original analog signal, it does not physically change it or sample it, unless of course you want it to. For example, changing the original analog audio through some form of analog processing.

That is the huge difference and the point that I am trying to make very clear here, about the differences between your pure analog sgnal going into digital versus it going into analog gear.

To simplify, with digital there is always a drastic sampling conversion change going on, with analog gear there is not such a drastic conversion taking place, and hence the audio stays much true'er to life, if I may put it that way. And no matter how good the software programmer's math is, there is something that gets lost in the digital translation.

Again that is why I said (unless it is being recorded into a analog tape machine) and I agree with you on what you said about analog tape recordings:

"Again, the analog tape recording process adds the audio signal to a super high frequency bias signal which then is used to magnetize some rust. It is both a physical and an audible change."


Sorry I did mean bit depth instead of sampling rate when dithering down, but to be clear you also have to sample rate convert to the same sample rate as well so that there is not a audible speed difference. Usually this all done for you in software and or hardware, but not always is this the case.

Not everyone today uses just new digital gear and software, people still use some older digital gear and software, so I was applying it to all home studio users.

Yes I know analog gear and analog tape has its flaws too, and digital does as well. That is why the discussion here. It's all good......

 

[aarowsmith]

You said "Definition of analog: A person or thing seen as comparable to another."

Your definition above is the wrong terminology of what we are talking about here, what the term "analog" means here is: it means analog signal capturing of the original pure audio wave signal, as in "analog wave form. " As in your ears hear the "analog" sound wave.


So my definition here of a analog audio signal is: the original analog wave form captured by a microphone through analog gear, analog in the same way your ear captures sound waves through air,

And I am not talking about analog coming from a digital synthesizer either.

Sorry if I did not make that clear, That is what I thought that this thread is talking about here, not "A person or thing seen as comparable to another."

 

[Bristol Posse]

Well strictly speaking the sound is presure wave through air which is then converted into a voltage that is "analogous" to that sound pressure wave. Hence the term analog. The voltages are used as an analog of the sound wave

That is a pretty drastic conversion

Every time you add a circuit to the chain you add thermal noise that is inherent in any electrical circuit. very quiet but if you have multiple processors in a chain it can add up.
each processor adds noise and alters the original signal so that it is no longer strictly analogous to the original sound.

If I were to record myself through a condenser mic into a preamp, through a compressor into an EQ and finaly to tape I've added at least four circuits worth of thermal noise, I've limited trasnsient response to what the mic diaphragm is capable of reacting to as well as adding and taking away some ot the pure frequency of my vocal based on the frequency response of the mic, I've altered the dynamic range in the compressor and depending on the gain used possibly added some harmonics in saturation, caused a phase shift and change in the frequency balance with the EQ, and then possibly further compressed, phase shifted and rolled off frequency in the tape machine along with added wow and flutter
Once you've done that you have to convert the voltage into movement of a speaker to be analogous to the voltage to create a new pressure wave in the air to be heard as sound
this pressure wave will be effected by the size and material of the speaker and cone
Each time you play the tape, you loose some of the fidelity of the original recorded sound usually percieved as a "dulling" of the sound

None of that was present in the actual vocal that I sang

Don't get me wrong I love hardware and what it does to the sound and don't want to talk anyone out of it but is far, far, far away from being a perfect anaolg of the sound fed into it and it's all those compunding variations from the true sound that make it so pleasing to me and many others.

Heck even the purely digital guys are using VSTs that are an analog of the analog processes

 

[Farview]

You said "Definition of analog: A person or thing seen as comparable to another."

Your definition above is the wrong terminology of what we are talking about here, what the term "analog" means here is: it means analog signal capturing of the original pure audio wave signal, as in "analog wave form. " As in your ears hear the "analog" sound wave.

That was a response to your response

I have to say that in your observation of "Both digital and analog are both actually an analog of the original sound", well that is just not the case with analog audio going into digital. There is a digital sampling process going on that changes the original analog audio into one's and zero's, this is both a physical change and a audible change.

to this sentence

Both digital and analog are both actually an analog of the original sound, in neither case does the actual sound travel through the processors, they just use different mediums and signal types to represent the sound

In that sentence, the definition I posted was the one being used.

 

[Mo Facta]

While Bristol Posse is pretty much on the money, this day and age analogue audio refers more to the fact that, unlike digital audio, it is not broken down into discreet steps (samples and bits) but is continuous. This is said to be analogous to the original pressure wave.

To be a bit silly for a second (no pun intended)...

My question is, was analogue called analogue before digital audio was invented, or simply "audio"? Or was the term coined out of contrast to digital?

Is a mechanical watch analogue?

Cheers

 

[mindsound]

At this time, really, a great mixing engineer will do a great mix, be it analog or digital. Digital got so much improvements (DAC, plugs, DAW) in the past years, that there is no reason for a mixing engineer to fail at getting a great -warm, punchy, 3D whatever- mix.

I heard some all-analog mixes that sounds thin and bland and really fat punchy and 3d digital mixes. And the opposite is also true.

Now, I think that one will choose a format more for the working process that differ a lot..... excepting if it's a matter of belief and faith, what is a lot of times...a big part of the sound.

 

[aarowsmith]

Great discussion everyone here',

Thank you I have opened my mind up more to digital now, I'm not saying I just use analogue gear, or the American spelling of "analog".

I do use digital allot now a days because you can't solicit music without going into a digital format. And you can not collaberate or do business without using Pro Tools or an equavelent DAW. I use Reaper DAW and love it'. Reaper is pretty much unlimited to what you can do with audio and MIDI tracks.

I guess what I am saying here and those that use analog equipment also have said many times, there is something about the sound that you don't get with digital. I could be wrong here, I have not tried all the new plug-in software and VST's out there, so there probably is a very close sound to a analog gear using software only.

I just heard from my friends, and from what I have experienced that most software does not sound as good as the original analog gear, but really what does ? I would imagine that digital converters would be very hard to emulate too in software, and do they even make plug-in emulators for different digital converters audio sound quality ?

Anyways, it is all good, digital and analog can live together happily ever after'.......

Now back to making more music'.

 

[Farview]

The plugins don't sound exactly like any specific piece of hardware. They do do the same job as the real thing. in my experience, no two hardware units actually sound the same, so getting a plugin to sound like the one or two units that you happen to be familiar with is impossible.

 

[aarowsmith]

My question is, was analogue called analogue before digital audio was invented, or simply "audio"? Or was the term coined out of contrast to digital?

Is a mechanical watch analogue?

Cheers

Well to answer your question:" Is a mechanical watch analogue?" Yes it most certainly is.

Kind of funny that some in society have coined that term "analogue" and or "analog" as meaning something mechanical in nature and with no digital processing chips. I guess the real term for analogue is something in relationship to something else. Which does not fully describe the several other meanings and uses of the word analog.

I don't really know how long they have been using the terminology of "analogue", for thousands of years ?

But this is where it gets real anal and maybe really stupid,

Do you call something with a digital Op-Amp chip(they are a form of digital chip circuitry in essence) being a digital device rather than an analog device ? For example: Many times there are digital chips( like Op-Amps etc.) inside even fully analog mixers, that I guess could be considered digital processing ? Also the same here applies to modern speaker amplifiers, analog stomp boxes, and so called analog effects devices that also may have digital chip processors inside them.

I would say, atleast in the last 30 years or so, you can't get away from digital chips being used inside most analog audio gear. So I guess you can't get away from digital circuitry entirely, and why would you even care ?

After all, if you like the sound, even from internal digital processing chips, then what's not to like

 

[aarowsmith]

The plugins don't sound exactly like any specific piece of hardware. They do do the same job as the real thing. in my experience, no two hardware units actually sound the same, so getting a plugin to sound like the one or two units that you happen to be familiar with is impossible.

Very true indeed.

 

[mindsound]

We touch something here: digital emulation vs analog. First, a piece of gear is a piece of gear....analog or digital. It is what you make with that count a lot more.

But more important to me is that a lot of people work with digital technology the same way people worked with analog gear for the last 6 decades. I think that what we love most about analog sound is something (a colour, a warm tone and so on...) our ears are used to.
But, of course how can we imagine that we'll get the same results (the sound our ears "want") using the same technique with different tools, i.e. DAC in place of a tape recorder?
In example, what are we doing with all these 7-10 kh boosted on most condenser mics that were intended to record on tape that were literally eating high freq. as the session progressed (what is the opposite problem with digital....a lot of highs but not as low end as tape)? You will have at least to change your micing techniques...
And it is the same with the mixing process (that begin anyway well before the micing stage).
So, we love the qualities of the analog sound? Let's use our ears and therefore, let's treat the sound source according to the completely different format that is digital.

 

[aarowsmith]

So, we love the qualities of the analog sound? Let's use our ears and therefore, let's treat the sound source according to the completely different format that is digital.

Exactly, I do agree also that digital can sound like anything, even analog gear, so in that respect if digital can have the same character and same sound quality of any audio signal it samples(records), then it does seem to represent the audio signal very accurately.

For example if you record a cassette tape to CDR, using 16 bit, 44 k sampling rate, it does yeild a fairly accurate A and B comparison, identical to the original cassette tape sound quality. So I am thinking that (this of course is not new news to anyone here) it proves that analog audio tape must be coloring the original audio a little bit. Where as digital does not add anything to the original analog audio, and nor does it take away anything from it, for the most part digital can be very accurate.

And so the only way you can really hear a bad difference on digital audio quality is to do many, many analog to digital transfers or vice versa. Then the D/A and A/D converters are perhaps doing something a little different each time(losing some information I guess) to the original audio. You can hear this if you do allot of analog and digital conversions. The sound gets duller and less dynamic each time. But you can hear this difference allot less when using higher sampling rates and bit depths. But you still should limit the digital conversions as much as possible.

There is yet another example of bad digital audio quality, especially when recording in lower bit rates. For example when using 16 bit depth to record with you can hear a much rougher sampling audio quality(and you will notice more background noise levels too) when recording the audio at very low sound levels. That is why you should treat all digital audio, especially on 16 bit recordings, always record using the highest audio levels possible.

So to get healthy sounding audio, you must occasionally peg the red meters, especially on softer sounds. But be careful not to over due the audio signal level, and do not ever go above 0 db because you can easily clip(clipping is audio levels going over digital 0 db, and it sounds very bad) digital audio on a dynamic part of the recording. So it is best to lower the audio level a little bit so it only hits 0 db at the highest signal peak levels. You can use also a compressor/limiter to achieve peaks at near 0 db signal levels.

I noticed when I did this on all my digital recordings it sounded much fuller, tones were richer, and much more clear sounding, especially on 16 bit recordings. And when doing 24 bit recordings, this works equally well.

Allot of people suggest much lower audio levels on 24 bit recordings, and recording at lower volume levels with 24 bit now is not such a big issue anymore like it was with 16 bit recordings. But I don't agree because you are still not getting your full audio sampling dynamic punch, and nor are you getting the highest digital sampling resolution you can get with 16 bit or 24 bit. Simply put when you record at low audio levels you are not getting the top performance in both your digital and analog gear.

And remember in the playback mix you can always adjust the audio levels up or down if you need to, and then adjust them again on the final master mix as well. You can have all loud dynamic sounding recordings and still be able to lower the volume levels later on playback if you need to on each track(adjusting your audio levels in the mix on playback as needed). By recording loud you will not only maintane your full 16 bit and 24 bit audio resolution, but at the same time you can then easily add even more punch and dynamics if you need to each track in the final mix/master. But when you record at lower volume levels your options become far less, it is will be very hard to add more punch and or dynamics to your low volume recorded tracks, without adding allot more background noise as well.

This is the big diifference too between analog tape versus digital audio sampling. With digital it sounds the best when audio levels are at or near 0 db. With analog tape the opposite is true, distortion is less at lower recording levels.

So remember if you record loud in the first place, you will also get lower background noise, and smoother bigger more dynamic sound. And in the end when you do finally crank up the final master mix to listen to it, the sound will be again not only smoother and more dynamic sounding but the background noise will be at a much lower level too, even on 24 bit recordings.

Not pegging the audio meters almost to the red line, is kind of like buying a expensive sports car and never going above 3,000 RPM and staying in 1st gear only, when really its made to go all day long at 6,000 RPM and all the way through 5th gear and get you there really fast !
Hopefully you can get it out of 1st gear ??

As long as you don't hit over 0 db with the digital audio peaks, and or don't go over + 9 db with analog audio tape, I say go for it when you record digital audio and Push it to the limit to get your money's worth.

 

[Farview]

But be careful not to over due the audio level above 0 db because you can easily clip(going over digital 0 db is very bad audio quality) digital audio on a dynamic part of the recording. So it is best in that case to probably lower the audio level a little bit, and or just use a compressor/limiter enough to achieve descent peaks at near 0 db signal levels. I noticed when I did this on my digital sampling(recording) it sounded much fuller, tones were richer and more accurate sounding especially on 16 bit recordings. And when doing 24 bit recordings this works even better.

So you're saying that when you compress audio it sounds fuller? The hell you say! The compression is changing the audio much more than sampling something a few Db hotter.

Allot of people suggest lower audio levels on 24 bit recordings, but I don't, because you are not getting your full audio quality, dynamic punch and or highest digital sampling resolution. You are not getting the top performance of what you really paid for in your digital and analog recording gear.

We suggest recording at normal levels...line level, just like you would going to tape. Depending on your converter's calibration, line level is probably around -18dbfs. The main reason for this is so you don't over-stress your preamps and the analog signal path on the way to the converters. A signal with small transients peaking at -1dbfs on the digital side will be sitting at +17dbVU on the analog side. That's not the sweet spot for any preamps I know of. Anything even moderately budget friendly will start to sound bad long before that.

With digital, once you are sufficiently above the noise floor it sounds the same all the way up to the point where it clips. With 24 bit, the digital noise floor is so low that it is irrelevant. In fact, if you record something that peaks at -48dbfs in 24 bit, you have the same resolution as a 16 bit recording at full scale. Making sure that all your recording levels are as close to zero as possible is pointless, leftover advice from the 90's when everything was 16 bit and line level was as high as -9dbfs.

In the end you can always lower your audio levels when you do the rough mix and then the final master.

But your recordings will sound better when close to pegging the meters almost to the red line. It's kind of like buying a expensive sports racing car and never going above 3,000 RPM. When its made to go all day long at 6,000 RPM ! Of course hopefully you get it out of 1st gear first ???

Actually, it's more like burning a steak, then putting it on ice trying to make it medium rare. It's too late once you've recorded it too hot.

I say go for it when you record audio , especially when using analog audio tape. As long as you follow common sense loud audio levels and don't hit over 0 db with digital peaks, and or go over + 9 db with audio peaks using analog tape.

Common sense would be recording at line level, since that is the level that all the gear was designed to work best at.

 

[aarowsmith]

Jay, like I have been saying all along, I never said record too hot ! I guess you did not really comprehend what I said.

Do you understand how digital audio converters work ? All digital audio converters are made to record at their highest possible audio resolution when audio levels are closest to digital 0 db. Do you understand this fact ? Or do you know something that is not in the engineer manuals ?


A very good analogy is: the more light that a digital camera CCD can pick up, the brighter the sunlight or artificial light, the better the picture resolution becomes. Better picture quality, brighter better colors, higher contrast ratio and more digital pixel recorded information. And the same factual analogy applies to digital audio sampling converters when exposed to higher recorded audio signal levels, gives it more audio information and thus much higher audio quality.


Getting Closer to digital converters 0 db recording audio level is always better recording quality, and it still applies today, not just in the 1990's, and or whatever time period you think it only applies. This is measurable, provable and you can perform your own test to hear this.


All I am saying is why record at lower signal levels just because you have a full 24 bit recording ? You then are not getting the full resolution or dynamics of a digital recording as you possibly can get, that your digital converters are capable of. Utilize them to the full and why not ? What are you scared of ?

And Why would you be satisfied with getting only 16 bits of audio resolution out of a 24 bit recording ?


And I never said using compressor/limiters improves sound quality immensely, but sometimes it may improve the sound if you have a good quality compressor/limiter and use it sparingly to get audio levels more consistent and hence louder. And yes it can make the sound much fuller, have you ever recorded vocals or bass using a good compressor/limiter ?


Recording with your digital audio gear whether it is 16 bit CDR, DAT, ADAT and or 24 bit DAW or whatever, line level is not the level you record at because Line level can be any where on the audio level signal meter, its relative so line level has nothing to do with getting the loudest possible recorded audio level.

And what is line level really ? Since it can be -10 db with consumer gear and at other times it can be +4 db levels with pro gear. Line level is not usually the recorded level anyways. And you may need to raise the line level on your gear using its volume control to get a louder recording. That is why you need to trust your digital audio recording meters and then also listen back to your digital recordings to make sure it isn't audio clipping, going over 0 db, the digital limit. And then you can also adjust your analog mixer with a little headroom safety margin accordingly to get your safe maximum digital audio recording level.


How you maintane your headroom in your digital mixer or analog mixer applies to how you set your audio levels at playback. So that it does not overload your master stereo buss on your mixer, usually engineers like to keep everything playing back at 0 db on the stereo master buss meter, and always keep the working mix at that level even on the highest audio peaks. That will keep you well within all your audio gear headroom limit.


All digital and analog audio gear meters need to be universally reading the same audio signal levels together, but not all do so thus again you need to do your own testing/caliberating from your digital recording levels then adjust your other analog audio gear to your digital gear. Usually your computer DAW is accurate on its audio level meters, so you can use it to adjust all audio gear.


On analog mixers you always have more headroom than digital 0 db would give you, so I have never encountered headroom issues when mxiing with digital audio gear, again as long as I keep all audio levels well within 0 db even on audio level peaks, using a analog and or DAW mixer stereo buss meters.


But the only time I have run out of headroom on mixing is when using reel to reel analog audio tape, you have to be careful it can exceed 120 db audio levels very quickly ! Good 2" analog reel tape can blow away most digital audio converters when it comes to higher dynamics, higher recording levels and higher frequency response. Because Analog tape is so dynamic it can damage audio gear fairly quickly. I never had digital audio damage gear, although it can happen.


Most audio tracks are not going to be at full 0 db anyways, some tracks are going to be much lower in volume, and some may be adjusted higher or lower in the final master mix.

But by getting all your recordings as hot as possible when you mix on playback you then have the option of raising the volume a bit on the mixer track without adding a huge amount of background hiss and or noise, and without losing any audio quality. Or by lowering the tracks audio level you lose less of its dynamics the hotter the original audio recording is. This applies to 24 bit audio as well. It's a provable audible fact.

And if you don't like to go that way and you find recording at lower levels is better for you, to avoid maybe audio level errors and for headroom safety reasons or whatever then fine, do that.


Get your money's worth out of your gear', that is all I am saying here.
And if you don't like the sound of using a compressor/limiter to get a little more fuller sound and more punch from your audio tracks, then don't.


The choice is yours, as always, I am just suggesting some provable ways of getting better audio quality in recording your digital audio tracks. If you don't want to try it then that is Ok too.


Like I said before not every person works with only new gear and 24 bit audio, some use higher and lower bit depths and sampling rates, and or they use older digital audio technologies, so yes getting recordings closer to the digital 0 db audio level whether using new or old digital gear, these concepts very much apply today and tomorrow.

 

[Mo Facta]

Do you call something with a digital Op-Amp chip(they are a form of digital chip circuitry in essence) being a digital device rather than an analog device ? For example: Many times there are digital chips( like Op-Amps etc.) inside even fully analog mixers, that I guess could be considered digital processing ? Also the same here applies to modern speaker amplifiers, analog stomp boxes, and so called analog effects devices that also may have digital chip processors inside them.

I would say, atleast in the last 30 years or so, you can't get away from digital chips being used inside most analog audio gear. So I guess you can't get away from digital circuitry entirely, and why would you even care ?

Um, no. You are assuming things that are not correct.

Firstly, an op-amp is not digital. Not one little bit. It is an OPERATIONAL AMPLIFIER, which is a completely ANALOGUE device. It works exclusively in the voltage domain and has no digital component whatsoever. If an analogue device has a digital chip inside of it, there has to be some sort of analogue to digital (AD) conversion to interface the chip and then, once the DSP has been applied, there has to be digital to analogue (DA) conversion before being fed to the analogue outputs. Any device that applies DSP in between conversion steps is generally said to be a "digital device", like a digital reverb. Granted, a digital reverb has got both analogue and digital components, but, for all intents and purposes, it is a long way from being a true analogue device.

Secondly, completely analogue processors still exist and they are legion. Countless. And most of them are still available and being manufactured. That little 4-channel Mackie mixer? 100% analogue. That UA 1176LN compressor? 100% analogue. That Oram large format console? 100% analogue. I could go on...

And to address your link, guitar stomp boxes have used digital technology for a long time but not the ones you posted in your link. Most of those are also fully analogue. What was your point there?

And to address the level issue...(again)

Yes, digital capture is pretty true to the original and that is actually the caveat. Once a signal has been encoded to digital, it does not exhibit the same non-linearities that is does in the analogue domain, in a strictly digital sense. Analogue circuits SATURATE and overload and [strict] digital does not (well, 16 and 24-bit audio can clip). 32-bit FP digital audio is completely immune to this phenomenon and has thousand of dB of headroom.

HOWEVER. Digital audio, AT SOME POINT, must interface with an analogue circuit. This means that if you're feeding, say, the analogue input of a digital to analogue converter a super hot, full scale signal, the converter is going to capture that SATURATED AND OVERLOADED ANALOGUE signal. In other words, it is going to capture and encode any and all distortion present in the signal. Once it's been digitized, there's no turning back. The distortion is there forever. It is well known that modern analogue circuitry saturates and produces distortion several dB below CLIP POINT. CLIP POINT is the point at which an analogue device will overload and 'clip' off the top of the waveform because it is outside of its operational range, voltage-wise. And PLEASE NOTE. MOST DIGITAL DEVICES HAVE GOT ANALOGUE SECTIONS. What does that mean? It means that if you're slamming a microphone preamp towards its clip point (which might produce distortion) and then you feed that signal into your converter/interface, you will saturate that circuit too, and that will result in production DOUBLE FOLD distortion. Then, if you play that back through the analogue outputs, it will most likely saturate that circuit too and further distortion will be picked up. This makes the distortion factor TRIPLE FOLD. It is a cumulative effect that piles up if you're working with these kinds of levels across the board.

SO, how do we combat this? CONSERVATIVE LEVELS. All gear was designed to work within an optimal range. That optimal range is NOT between +12dBu and clip point (for example). That is why we have long standing standards like +4dBu and 0VU. That is why meters have a green range, an orange range, and a red range. To scoff at these guidelines is asking for distortion.

My 2c (ZAR)

Cheers

 

[gecko zzed]

For example if you record a cassette tape to CDR, using 16 bit, 44 k sampling rate, it does yeild a fairly accurate A and B comparison, identical to the original cassette tape sound quality. So I am thinking that (this of course is not new news to anyone here) it proves that analog audio tape must be coloring the original audio a little bit. Where as digital does not add anything to the original analog audio, and nor does it take away anything from it, for the most part digital can be very accurate.

Whether tape colours the sound or not is often debated here. There are some who assert that it doesn't; that it faithfully reproduces the sound of the original source. On the other hand, there is a school that acknowledges the physics of tape saturation and its effect on the sound. Similarly, there are those who argue as you declare, that digital neither adds nor subtracts and that it is invisible in the recording process. Others argue the opposite; that the process of sampling influences the sound negatively.

And so the only way you can really hear a bad difference on digital audio quality is to do many, many analog to digital transfers or vice versa. Then the D/A and A/D converters are perhaps doing something a little different each time(losing some information I guess) to the original audio. You can hear this if you do allot of analog and digital conversions. The sound gets duller and less dynamic each time. But you can hear this difference allot less when using higher sampling rates and bit depths. But you still should limit the digital conversions as much as possible.

As a piece of general advice, minimizing A/D conversions is sound. In fact, minimizing the signal path as much as possible reduces the opportunity for introducing unwanted noise and distortion. The way I do things, there is only one A/D conversion: the signal on the way into the computer. The next time there is an A/D conversion is when someone goes from digital to analog on their CD player. There is, of course, a D/A conversion when I listen to playbacks and so on. However, that doesn't feed back into the recording chain.

This advice is even more applicable to tape operations. Tape to tape or track to track transfers yield greater quality losses per transfer than digital. I remember the days of operating a 4-track cassette recorder and having to plan very carefully to minimise track bouncing.

 

[aarowsmith]

SO, how do we combat this? CONSERVATIVE LEVELS. All gear was designed to work within an optimal range. That optimal range is NOT between +12dBu and clip point (for example). That is why we have long standing standards like +4dBu and 0VU. That is why meters have a green range, an orange range, and a red range. To scoff at these guidelines is asking for distortion.

My 2c (ZAR)

Cheers

Yes thank you for that information, I did not know that op-amps are not digital, and 32 bit audio has really high headroom, even higher recording levels than analog tape. Very interesting good stuff to add to anyones knowledge.

As far as conservative audio levels, I agree with you. My example is kind of like a digital camera captures more information when exposed to a higher light environment, so it is with digital audio converters when they are exposed to reasonable high audio levels, as long as not to exceed the 0 db digital limit and to make sure the highest peak levels do not ever go over 0 db.

Digital audio does not distort even all the way up to 0 db audio levels, you should of course leave a safety margin and get close enough to 0 db if you can. in fact with digital it sounds better the higher the recording level, within reason.

So when a digital camera is exposed to too much light, it then can over saturate or clip and the picture quality is drowned out, much like digital audio. But I am not saying over saturate your digital audio recordings, just make sure you are getting enough audio signal information recorded, that's all and it will result in more audio information thus quality. You then will have more audio information to work with, in the mix, and or for digital audio processing in or outside the box.

Unless of course you are going for that really distant and low volume sounding recording ?

And then when you mix at playback you can always adjust audio levels to be at a reasonable level, on analog mixers this is not to exceed 0 db on its meters. So whatever real World audio level measurement it is, I don't know, because every audio level meter is caliberated differently, depending on +4 db or -10 db pro level versus consumer line levels. And some audio gear have variable volume line level controls so line level can be anywhere pretty much.

I say trust your computer sound card and DAW metering. And as long as you don't exceed your audio gears maximum recording level and or op-amp when recording. And then if also you don't exceed your playback mixer's head room either, try to keep your maximum audio levels at around 0 db on its meters, then you should not have any distortion problems at all. If you do then check for other problems in your connections, mics or audio gear.

Best regards,

Aaron

 

[aarowsmith]

Very good advice on digital audio quality, only one A/D conversion when recorded, and then only one D/A conversion when played back from CD master, thanks I will try to do that and see how my sound improves.

Best regards,

Aaron