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Understanding Equipment / Mic Pre Bandwidth and how it affects the sound. Modular Synthesizers
Old 31st December 2010
  #1
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ionian's Avatar
Understanding Equipment / Mic Pre Bandwidth and how it affects the sound.

Hi,

At any point, feel free to correct my misinformation in addition to advice!

Starting with the Mic Pre, I've heard that having a wide bandwidth is good for keeping phase and distortion to a minimum.

Now, lets take an absurd example of wide bandwidth - say the new True 500 series mic pre which claims to have 1.5 Hz to 600 KHz. If I run that mic pre into a compressor or an EQ that has something like a bandwidth of 10 Hz to 50 KHz, did I just completely undo whatever good the True Systems pre did in maintaining the integrity of my source? Or is it OK because the True mic pre has already captured it and it will stay good regardless of what lower bandwidth stuff follows the pre?

For a converter I'm using the Steinberg 816 which at 96 kHz can capture 20 Hz to 40 Khz so is it all moot since the Steinberg will negate any positive effects that the wider bandwidth stuff before it imparted?

Or is this all simply a case of me worrying too much about what's going on outside the audible range affecting what's going on inside it?

Thank you all very much for your time and advice,

Frank
Old 31st December 2010
  #2
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JohnRoberts's Avatar
 

Quote:
Originally Posted by ionian View Post
Hi,

At any point, feel free to correct my misinformation in addition to advice!

Starting with the Mic Pre, I've heard that having a wide bandwidth is good for keeping phase and distortion to a minimum.
Half right. Having bandwidth or frequency response beyond both extremes of human hearing, does imply no phase shift from HP or LP filters in the transfer function. Bandwidth does not directly impact linearity, while there is a general trend of of falling gain bandwidth associated with stability compensation, so there could be elevated distortion in the top octave if the limited bandwidth is from running out of HF gain. OTOH if the 600khz preamp has a 20kHz rolloff incorporated into it's negative feedback path, there will not be any measurable increase in distortion associated with that reduced bandwidth.
Quote:
Now, lets take an absurd example of wide bandwidth - say the new True 500 series mic pre which claims to have 1.5 Hz to 600 KHz. If I run that mic pre into a compressor or an EQ that has something like a bandwidth of 10 Hz to 50 KHz, did I just completely undo whatever good the True Systems pre did in maintaining the integrity of my source? Or is it OK because the True mic pre has already captured it and it will stay good regardless of what lower bandwidth stuff follows the pre?
Good news bad news... The bad news is that the bandwidth of subsequent stages is additive, so if one stage has a LPF at 50 Khz, that's the best that will come out the other end. The good news is I can't hear signals up at 600kHz and suspect you can't either.

Quote:


For a converter I'm using the Steinberg 816 which at 96 kHz can capture 20 Hz to 40 Khz so is it all moot since the Steinberg will negate any positive effects that the wider bandwidth stuff before it imparted?

Or is this all simply a case of me worrying too much about what's going on outside the audible range affecting what's going on inside it?

Thank you all very much for your time and advice,

Frank
While some of this comes down to design philosophy, it is worthwhile worrying about out of band material. Your A/D convertor will have anti-alias filters built in, to prevent false frequency images reflected down from the sampling rate. Also with solid state electronics you want to either LPF the raw input, or pass it cleanly without slew limiting, to prevent rectification and spurious noise from RF interference. A 600kHz bandwidth preamp may pass some of the lower AM stations, but not everything.

In general when talking about much beyond 20kHz, it isn't how high it goes, but how cleanly it deals with above band signals.

If course this answer is a bit of a simplification.

JR
Old 31st December 2010
  #3
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ionian's Avatar
That post might have a been a bit simplified but it was still an education for me!

Thank you very much for taking the time to give such an informed post and clearing some stuff up for me.

Regards,
Frank
Old 31st December 2010
  #4
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Hi
John has (as usual) covered it in that having a very wide bandwidth capability can be a double edged sword and may be problematic in studio environments with large quantities of particularly HF 'rubbish' which can be demodulated. Switchmode supplies, mobile phones and computers which although may not be noticable individually can possibly profuce other frequencies when combined.
Super LF response can also be a drawback if you use aircon as even if it appears 'silent' to your ears may have considerable VLF rumble which would be captured by a good mic.
Having effective filters for HF and LF at an early stage in the chain is usually the best way to prevent problems as gear does not really like being 'blasted' with high levels of out of band signal.
Matt S
Old 31st December 2010
  #5
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Quote:
Originally Posted by Matt Syson View Post
Having effective filters for HF and LF at an early stage in the chain is usually the best way to prevent problems as gear does not really like being 'blasted' with high levels of out of band signal.
Matt S
+1.. and this gets into design philosophy that some from the "DC to light" brigade may disagree with.

I have always designed raw input products, like phono preamps and mic preamps with definite attention to band passing the audio so it can be handled cleanly by even less than well designed subsequent stages. Another problem from the not so good old days, is that LF energy (like from a non-centered record holes or unfiltered plosive sounds) could wreak havoc with tape recording and companding NR.

Since roll offs are additive and modern electronics cheap and fast, 50k-100kHz LPF seems reasonable for most audio paths. I'll leave it to others to argue the merits of sampling digital at rates only bats will enjoy.

JR
Old 31st December 2010
  #6
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Hi
Probably of greater importance is the understanding of what is going on in each of the units in your chain and try not to force wideband material through to the detriment of the overall chain.
Matt S
Old 31st December 2010
  #7
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Quote:
Originally Posted by JohnRoberts View Post
Another problem from the not so good old days, is that LF energy (like from a non-centered record holes or unfiltered plosive sounds) could wreak havoc with tape recording and companding NR.
those of us that sample from vinyl still have to deal with this, and its amazing the effect of non-centered holes, or too-large holes have.
Old 31st December 2010
  #8
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Quote:
Originally Posted by oxide54 View Post
those of us that sample from vinyl still have to deal with this, and its amazing the effect of non-centered holes, or too-large holes have.
One solution I came up with back in the not so good old analog days, was to make an adaptive HPF that would slide the one pole filter cut off up higher in frequency, the lower the total amplitude of the signal. This was to cure a phantom modulation that would occur with encode/decode NR and tape paths. The very LF not saved to tape would not be present during playback to properly decode so the audible audio was modulated by this now missing vLF signal. My trick filter rolled off the vLF only at low level when it could become a significant part of the total signal envelope controlling encode/decode gain.

Most digital paths should harmlessly pass vLF. What is the nature of your problem? Surely you're not saving vinyl to old school tape?

JR
Old 31st December 2010
  #9
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Quote:
Originally Posted by JohnRoberts View Post
Most digital paths should harmlessly pass vLF. What is the nature of your problem? Surely you're not saving vinyl to old school tape?

JR
no i was talking very generally about the non-centered hole thing, thinking more on the wide variance of vinyl. forgive me I am sitting in a room full of vinyl!

tbh I find HPF filter 39/40hz works for a lot things, really clears up the mud sometimes.

i thought I typed this but obviously didn't: luckily we have 20/21st century samplers and computers to record into,
Old 31st December 2010
  #10
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Quote:
Originally Posted by ionian View Post
Hi,

At any point, feel free to correct my misinformation in addition to advice!

Starting with the Mic Pre, I've heard that having a wide bandwidth is good for keeping phase and distortion to a minimum.

Now, lets take an absurd example of wide bandwidth - say the new True 500 series mic pre which claims to have 1.5 Hz to 600 KHz. If I run that mic pre into a compressor or an EQ that has something like a bandwidth of 10 Hz to 50 KHz, did I just completely undo whatever good the True Systems pre did in maintaining the integrity of my source? Or is it OK because the True mic pre has already captured it and it will stay good regardless of what lower bandwidth stuff follows the pre?

For a converter I'm using the Steinberg 816 which at 96 kHz can capture 20 Hz to 40 Khz so is it all moot since the Steinberg will negate any positive effects that the wider bandwidth stuff before it imparted?

Or is this all simply a case of me worrying too much about what's going on outside the audible range affecting what's going on inside it?

Thank you all very much for your time and advice,

Frank

True pre and Earthworks mic have a very wide bandwith...
but there are better sounding "musical" equipments.
Old 1st January 2011
  #11
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I really appreciate the answer and advice given here.

As to what end this all is for me and what I'm looking for is simply clarity and openness.

I've went the traditional gearslut route in the past and had a good amount of transformer equipment, mic pres, compressors and eqs and believe it or not I've come to strongly dislike transformers and the effect they have on the overall sound which for me was manifested in very reduced clarity on top, a "closed" sound and a narrowing of the stereo field, which someone else had told me was due to distortions that the transformers introduce to the signal.

For me, the main part was to remove all transformers from my path and to use strictly transformerless mics, mic pres, compressors and eqs as outboard equipment but I'm wondering if that's all there is to clear and open sound.

That's what led me to my curiosity about bandwidth and how it affects the sound.

For example, is a mic pre that has bandwidth from 1 HZ to 600 kHz really more clearer and open sounding then a mic pre that ranges from 5 Hz to 150 kHz? Or is even a mic pre who's bandwidth just simply 10 hZ to 50 kHz more then enough as long as there's no transformers involved? Or are there other factors involved? Then after capturing this signal, is it fine regardless of the bandwidth of the gear following or can taking a signal from a mic pre that captures up to 600 kHz and shoving it through a compressor that tops out at 50 kHz actually harm the signal? In this case would it be less damaging to a signal to capture it with a mic pre that goes up to 200 kHz and use a compressor that tops out at 150 kHz as a lot less of the upper bandwidth is going to get filtered out at once on the way down?

Please forgive me if any of this makes me sound ignorant. I'm in very strange waters here and none of this is ever really addressed when people discuss gear and performance elsewhere on this site. I'm simply trying to understand what's happening to my overall signal on its way to my converter and how much the inaudible range can actually affect the audible range and how to attain the most clarity in my sound when recording.

Thank you all very much so far for your time and postings!

Regards,
Frank
Old 1st January 2011
  #12
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I fear you are looking for a simple answer to a more complex question.

Frequency response is just a measure of the amplitude response and nothing more.

Human hearing does not extend as high or low as the numbers you are throwing around. Since you are by definition only hearing differences inside the human hearing range, differences that are not reflected in the simple frequency response at these frequencies you can actually hear, are something else besides bandwidth.

Transformers can in fact be very wide band and very linear, but these "good" transformers will generally be expensive too.

So one generalization I will offer is that it cost less to deliver a clean flat path without transformers, but this doesn't mean that all transformer-less paths are clean and flat, or that all transformer paths are distorted. Deane Jensen (RIP) and a few others made some very well respected transformers so as always it comes down to execution, not the approach.

Linearity, or distortion may be a better specification to explore for sonic differences when frequency response is flat over the audio bandwidth.

JR

PS: There are some who embrace the coloration caused by less than high linearity transformer paths. So some popular or fashionable products may sound less transparent than others. This is all subjective with a dose of fashion so like hemlines does not always follow a rational pattern.
Old 1st January 2011
  #13
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Hi
As John has just commented the questions you asked originally are only the tip of a massive iceberg involving impedance and signal level compatability between bits of gear, cable capacitance (if using long cables), correct termination of any transformers in the system, clipping points for all pieces of gear in the chain and so on. ALL of these and more will have some impact on the results you manage to record.
I built a 'clone' 'Pultec' EQ a few years ago and one of the users commented that the LF is a bit lacking even with the EQ switched out (still using the amplifier path). The LF is not lacking as it is 'flat' from around 7 Hz to 80 KHz but the input transformers I have used do not saturate at LF unless fed a very high level compared to other units.
Again back to learning your gear thoroughly.
Matt S
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