Tracking at 24/96 vs 24/48

[James Price]

Do you think it's the norm for most studios to track at 24/96 or 24/48? Or is it a mixed bag?

Which one do you track at?

[Mike Rivers]

On 1/21/2017 11:19 AM, James Price wrote:
> Do you think it's the norm for most studios to track at 24/96 or 24/48? Or is it a mixed bag?

Mixed bag. Mostly, I track at 24-bit 44.1 kHz. I rarely track at 96 kHz
because I don't have any gear (or ears or sources, for that matter) that
needs to work on anything above 20 kHz. I can be more conservative with
level and leave more headroom when working at 24-bit than16-bit
resolution, so I waste my memory space on headroom that I might use
rather than frequency response that I won't use.

Some digital signal processors work better at 96 kHz, so if that's how
you work, there's an advantage to using that sample rate to start off
with. And if your paying client asks for a particular sample rate, you
should be prepared to use it.


--
"Today's production equipment is IT based and cannot be operated without
a passing knowledge of computing, although it seems that it can be
operated without a passing knowledge of audio" - John Watkinson

Drop by http://mikeriversaudio.wordpress.com now and then

[thekma...@gmail.com]

Mike Rivers wrote: "And if your paying client asks for a particular sample rate, you

should be prepared to use it. "

That is, if said paying client is even remotely
as concerned with sampling rate(or other
technical matters) as they are with laying
down a good effort in as few takes as possible.

[Mike Rivers]

It's no more difficult to record at 2x sample rate, but if that's what
the client asks for, that's what he'll get. Getting good recordings is
his job at whatever sample rate. I'll concede that some of them don't
know what sample rate means, but they may have read that 96 kHz is good,
or have been asked by the people who are paying the bills to get 96 kHz
recordings because it's company policy, or the recording you're making
will be part of a project with recordings coming in from other places,
and if they're all at the same sample rate, it's easier to combine them
into a CD than if the recordings were at multiple sample rates.

When we were using tape, doubling the tape speed meant using twice as
much tape, which could make a big dent in the project's budget.
Accommodating files at 2x sample rate on a disk drive may not add any
cost at all, or maybe only a few bucks. On the other hand, when you're
burning up tape twice as fast, you might not want to keep all of those
takes that might have a better word in them than your primary take. That
could save some time if you're not afraid to make a decision.

[Scott Dorsey]

James Price <mal...@cebridge.net> wrote:
>Do you think it's the norm for most studios to track at 24/96 or 24/48? Or is it a mixed bag?

If the converters are perfect, the only thing the higher sampling rate buys
you is added bandwidth.

So, there are three reasons to use the higher sampling rate:

1. You are trying to record ultrasonics for later analysis that may make them
audible or create audible effects from them. Say you are recording bird
song for analysis or machine vibrations or you're making dubs of LPs that
will later go through noise reduction which will benefit from the ultrasonic
content for pop detection.

2. Your customer demands it.

3. Your converters are imperfect and happen to sound better at the higher rate.

And there are three reasons to use the lower sampling rate:

1. You are trying to avoid recording ultrasonics for fear that they will later
mix down and form audible distortion products in the audible region.

2. Your customer demands it.

3. Your converters are imperfect and happen to sound better at the lower rate.

>
>Which one do you track at?

Having done A/B comparisons with my own converters, I think they sound better
at the lower rate. This might be because the clock stability is less crucial
and it may because of the ultrasonic filtering, or it might be some other
reason. But for that reason I will tend to record at 44.1 ksamp/sec.
This may not apply to anyone else's situation.

Now, that said, I have film customers who want everything at 48 ksamp/sec
and I'm happy to oblige them, as well as I am happy to oblige the folks who
want recordings at 96. They are paying the bill, if they demand that rate
for their own reasons that's fine by me.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[thekma...@gmail.com]

Mike Rivers wrote: "t's no more difficult to record at 2x sample rate, but if that's what

the client asks for, that's what he'll get. "

Of course it's no more difficult. My reply
was more concerned with client knowledge
of the process. Take for example a certain
young quartet from Liverpool: Do you think
John Lennon would be concerned about
sampling rated or bit depths if the Beatles
were in ascendency after the year 2000?

Not every artist has the same technical
prowess as a Don Fagan or Alan Parsons.
Most artists today probably are unaware of
the technical details, and are more wrapped
up in the creative process and just wanting to
give it their all.

The rest, they leave up to the engineers! :)

[John Williamson]

On 22/01/2017 13:32, thekma...@gmail.com wrote:
> Mike Rivers wrote: "t's no more difficult to record at 2x sample rate, but if that's what
> the client asks for, that's what he'll get. "
>
> Of course it's no more difficult. My reply
> was more concerned with client knowledge
> of the process. Take for example a certain
> young quartet from Liverpool: Do you think
> John Lennon would be concerned about
> sampling rated or bit depths if the Beatles
> were in ascendency after the year 2000?
>

He might be, if he were given the chance to compare two takes using
different sample depths and rates. He was definitely interested in the
latest technology and how to get the best out of it when he was recording.

The difference between 16 and 24 bit isn't always obvious on the first
generation, but when mixing more than one or two tracks, while applying
effects, the difference can become noticeable. Whenever possible, I
always record at 24 bit, at a rate to either match the expected output
format, or double that rate. Okay, I may only be getting 10 to 14 bits
of dynamic range in the room, but I can always throw bits away, and with
storage now being as cheap as it is, there's no real financial penalty
apart from the 24 bit converters costing more to buy.

> Not every artist has the same technical
> prowess as a Don Fagan or Alan Parsons.
> Most artists today probably are unaware of
> the technical details, and are more wrapped
> up in the creative process and just wanting to
> give it their all.
>
> The rest, they leave up to the engineers! :)
>

Until they listen to the final result, and start moaning about the poor
quality...

To be fair to the performers, though, a lot of artistes now are are more
aware of the technical processes than they were in even the 1990s, as
many of them now start by doing recordings in their own rooms, only
progressing to a real studio as they realise the limitations of their
room and equipment. This trend started with the introduction of the
Fostex And other makes of) 4 track, cassette based Portastudios.
--
Tciao for Now!

John.

[thekma...@gmail.com]

John Williamson wrote:

"The difference between 16 and 24 bit isn't always obvious on the first
generation, but when mixing more than one or two tracks, while applying
effects, the difference can become noticeable. Whenever possible, I

always record at 24 bit."

Absolutely. Use the highest spec one
can in production. Akin to giving the
dog the 'biggest backyard to play
around in' in terms of bit depth
and spectrum. Lets the artist be
themself during the creative process.

[James Price]

On Sunday, January 22, 2017 at 7:18:19 AM UTC-6, Scott Dorsey wrote:

> James Price wrote:
> >Do you think it's the norm for most studios to track at 24/96 or 24/48?
> >Or is it a mixed bag?
>
> If the converters are perfect, the only thing the higher sampling rate buys
> you is added bandwidth.
>
> So, there are three reasons to use the higher sampling rate:
>
> 1. You are trying to record ultrasonics for later analysis that may make them
> audible or create audible effects from them. Say you are recording bird
> song for analysis or machine vibrations or you're making dubs of LPs that
> will later go through noise reduction which will benefit from the
> ultrasonic content for pop detection.

On a side note, is the ability to record at 24/96 limited by the frequency response of the microphone? Wouldn't you need a mic that can capture audio above 22kHz?

[PStamler]

Juat about any microphone can capture audio above 22kHz. That audio may be reduced in level, and the response may be dropping at 12dB per octave, but there is some audio there.

Meanwhile, I try to record at the sampling rate that will be used in the final product. Recently I did a remastering job, turning a 2-track analog tape from 1980 into a CD. I did the transfer at 96kHz, and did all the editing at that rate (including topping and tailing, level adjustments, and the like). It sounded excellent.

Then I converted the sample rate to 44.1, which it needed to be for CD. And all that wonderful sound...went away. The result sounded as grubby as an early-years digital recording.

I'd done the SRC in Audition, which had tested well in the Infinite Wave comparison (http://src.infinitewave.ca/). But I wasn't happy. So I took the files to the university, which had iZotope's RX5 on one of their machines -- iZotope's SRC had tested extremely well in the Infinite Wave comparison. Same result.

So I redid the transfer, this time at 44.1kHz from the beginning. Luckily, I'd kept detailed notes of all my remastering work, so I could redo it without too much hassle. And I finally had a result I could live with.

Bottom line: Sample rate conversion is still a can of worms, and I for one will continue to transfer at the sample rate which will be used for the final product, to avoid having to do SRC in the process.

Note that this was an in-the-box remastering job from a 2-track master tape. If I were doing a multitrack job from scratch, and mixing down in the analog domain, I might be tempted to record at 96k, then (if the result is to be a CD) mix down to a 44.1k 2-track, or 48k if it'a going to be used for video. But I'll avoid SRC as much as possible.

YMMV, of course.

Peace,
Paul

[Scott Dorsey]

Yes, and not only that you need a microphone that can capture _significant_
and _useful_ audio above 22 KHz. If you play a fiddle into an SM-57 you'll
get get some output above 22 KHz, but it will be nasty junk that isn't adding
anything.

The wider your bandwidth is, the more noise you have to deal with, and if you
don't think the noise at 25 KHz is a problem, wait until you play it back
through a dome tweeter and get intermodulation products resulting from it
that turn up in the midrange. Wider bandwidth means nonlinear distortion
effects become more significant.

Of course, you can record at 96 ksamp/sec and then low-pass at 20 KHz anyway,
which is what I often do when customers request high sampling rates.

[James Price]

If you ultimately low-pass at 20 KHz, what's the advantage (if any) of recording at 24/96 vs. 24/48, other than to appease the client?

[Scott Dorsey]

James Price <mal...@cebridge.net> wrote:
>On Sunday, January 22, 2017 at 12:31:55 PM UTC-6, Scott Dorsey wrote:
>
>If you ultimately low-pass at 20 KHz, what's the advantage (if any) of recording at 24/96 vs. 24/48, other than to appease the client?

There is none, but the client is happy and it sounds good. In the end,
those are really the only two important things ever.

[Mike Rivers]

On 1/22/2017 10:48 AM, James Price wrote:
> If you ultimately low-pass at 20 KHz, what's the advantage (if any)
> of recording at 24/96 vs. 24/48, other than to appease the client?

There may be none, but don't belittle appeasing the client, unless it's
a non-paying client.

But if you're working in a DAW, which you almost certainly will be, at
least for some part of the project, and you use a processing plug-in
that works better at high sample rates, that might be to your advantage.

[thekma...@gmail.com]

PStamler wrote: "Meanwhile, I try to record at the sampling rate that will be used in the final product.

Recently I did a remastering job, turning a 2-track analog tape from 1980 into a CD. I did the transfer at
96kHz, and did all the editing at that rate (including topping and tailing, level adjustments, and the like).
It sounded excellent.

Then I converted the sample rate to 44.1, which it needed to be for CD. And all that wonderful sound...
went away. The result sounded as grubby as an early-years digital recording. "

Anyone wanna venture what happened in his
second paragraph? I'm guessing he should
have used a sampling rate that was a multiple
of that used for the deliverable. I.E. 88.2kHz
for a 44.1kHz(CD) product. Is 88.2 avail. as
a rate?

[Scott Dorsey]

Sadly, most SRC programs won't filter-and-decimate when they encounter a
multiple, they just use the same algorithm they use for non-multiples.
So having sample rates that are multiples doesn't actually help you.

Good SRC has become very good these days, but bad SRC remains very very bad.
But there is good free SRC available if you don't need it to work in realtime.

[geoff]

On 22/01/2017 5:53 PM, Mike Rivers wrote:
.
>
> It's no more difficult to record at 2x sample rate, but if that's what
> the client asks for, that's what he'll get. Getting good recordings is
> his job at whatever sample rate. I'll concede that some of them don't
> know what sample rate means, but they may have read that 96 kHz is good,
> or have been asked by the people who are paying the bills to get 96 kHz
> recordings because it's company policy, or the recording you're making
> will be part of a project with recordings coming in from other places,
> and if they're all at the same sample rate, it's easier to combine them
> into a CD than if the recordings were at multiple sample rates.

Higher sample rates can be demanded by those with little understanding,
such as with 'loudness'.

>
> When we were using tape, doubling the tape speed meant using twice as
> much tape, which could make a big dent in the project's budget.
> Accommodating files at 2x sample rate on a disk drive may not add any
> cost at all, or maybe only a few bucks. On the other hand, when you're
> burning up tape twice as fast, you might not want to keep all of those
> takes that might have a better word in them than your primary take. That
> could save some time if you're not afraid to make a decision.
>

Surely no reason for anybody to ever track with less than 24-bit bitdepth ?

The potential 'cost' of higher sample rates (why stop at 88 or 96, what
about 192 !!!) is degradation of performance of a DAW wrt CPU loading,
or capability of the transfer method between A-D and DAW ...

geoff

[geoff]

88k2 instead of 96K sprung immediate to my mind too, for potentially
fewer SRC issues.

geoff

[Les Cargill]

PStamler wrote:
> On Sunday, January 22, 2017 at 11:40:58 AM UTC-6, James Price wrote:

<snip>

>
> Bottom line: Sample rate conversion is still a can of worms, and I
> for one will continue to transfer at the sample rate which will be
> used for the final product, to avoid having to do SRC in the process.
>

<snip>

You'd need two full setups of course, but it seems like simply
transferring in real time via analog would be preferrable to
SRC.

> YMMV, of course.
>
> Peace,
> Paul
>

--
Les Cargill

[Mike Rivers]

On 1/23/2017 4:47 AM, Les Cargill wrote:
> You'd need two full setups of course, but it seems like simply
> transferring in real time via analog would be preferrable to
> SRC.

I agree with Paul Stamler that the best approach to sample rate
conversion is to avoid it. It seems that there are good SRC processes
and not-so-good ones, just like there are good and not-so-good D/A and
A/D converters, and good sample rate conversion requires a lot of
computer horsepower. You have to, in essence, reconstruct the original
waveform of the source, then re-sample the (virtually) reconstructed
waveform at your desired rate. If the reconstruction isn't accurate,
you'll end up re-sampling interpolated values, and that's what doesn't
sound right.

A while back, someone, maybe it was Mytek, made a converter that could
use separate clocks for the D/A and A/D sections, so you could just pipe
the output into the input and convert the sample rate in a single
hardware box. It sounded better than any of the software SRC programs of
its time. But then, Mytek's converters were among the best sounding ones
in the industry, so they had a good start.

[James Price]

On Sunday, January 22, 2017 at 12:54:22 PM UTC-6, Scott Dorsey wrote:

> >James Price wrote:
> >On Sunday, January 22, 2017 at 12:31:55 PM UTC-6, Scott Dorsey wrote:
> >
> >If you ultimately low-pass at 20 KHz, what's the advantage (if any) of
> >recording at 24/96 vs. 24/48, other than to appease the client?
>
> There is none, but the client is happy and it sounds good. In the end,
> those are really the only two important things ever.

In Mastering Audio, Bob Katz indicated that one benefit of higher sampling
rates (eg. 96 kHz) would be moving unwanted converter noise above the
audible frequency range, some of which is filtered out upon downsampling.
The main idea being that moving the filter cutoff to 48 kHz (for 96 kHz SR)
relaxes the filter requirement and makes it easier to design filters with less
ripple in the passband and less phase shift near the upper frequency limit.


Any thoughts?

[Scott Dorsey]

This was absolutely critical back in 1985, and it was why you saw higher
rates used on things like the Mitsubishi recorders. But then we got
oversampling everywhere which did the same thing without having to record
at the higher rate. And then the whole world changed with sigma-delta
converters where the noise issues are totally different.

>Any thoughts?

Thank God we don't deal with ladder converters that are barely able to keep
up with the sample rate and drift all over the place today. The sigma-delta
technology has made getting real linearity across a wide range far less
expensive and made brickwall filters a thing of the past. It was a real and
actual revolution.

[Tatonik]

Scott Dorsey <klu...@panix.com> wrote:

> Sadly, most SRC programs won't filter-and-decimate when they encounter a
> multiple, they just use the same algorithm they use for non-multiples.
> So having sample rates that are multiples doesn't actually help you.
>
> Good SRC has become very good these days, but bad SRC remains very very bad.
> But there is good free SRC available if you don't need it to work in realtime.
> --scott

What is the good SRC? Paul Stamler mentioned being dissatisfied with
iZotope, which I thought was supposed to be one of the good ones. I've
also heard mention that Weiss Saracon is good, but it's far from free.

[Don Pearce]

Have a look here. Someone has done a lot of work comparing the quality
of various SRCs.

http://src.infinitewave.ca/

d

[PStamler]

That was my resource in choosing the SRCs from Audition and iZotope. Both did very well in the site's test, but I didn't like the results of either as well as I did my redub at 44.1, with no SRC. How come? I don't know.

Peace,
Paul

[Scott Dorsey]

Wow, this is an interesting site. It doesn't look like the detail is enough
so you can tell the differences between the best converters, but it sure makes
it obvious what the differences between the best and not-so-best are.

I wish this had some of the hardware sample rate converters on there, just to
show how bad most of them are.

[James Price]

On Monday, January 23, 2017 at 3:41:11 PM UTC-6, Don Pearce wrote:

> Have a look here. Someone has done a lot of work comparing the quality
> of various SRCs.
>
> http://src.infinitewave.ca/

The graphs for Adobe Audition 5.5 are pretty impressive.

[thekma...@gmail.com]

So what should we see with those graphs?

The fewer cross-hatch lines the better?

[JackA]

On Tuesday, January 24, 2017 at 1:36:08 PM UTC-5, thekma...@gmail.com wrote:
> So what should we see with those graphs?
>
> The fewer cross-hatch lines the better?

Too many bean counters. Forget graphs, use God given ears, and you too, will receive compliments like this....

"I’ve read for years (on BSN) that the double tracked vocal was missing from the multi track, and was dubbed in mono onto the final master. Your version blows that theory out of the water!"

Jack

[geoff]

On 23/01/2017 7:31 a.m., PStamler wrote:
>
>> On a side note, is the ability to record at 24/96 limited by the frequency response of the microphone? Wouldn't you need a mic that can capture audio above 22kHz?
> Juat about any microphone can capture audio above 22kHz. That audio may be reduced in level, and the response may be dropping at 12dB per octave, but there is some audio there.

All tracking should be done at 3kHz.

geoff

[Les Cargill]

Audacity comes out well in that. Interesting.

---
Les Cargill

[Trevor]

On 24/01/2017 1:32 AM, Mike Rivers wrote:
> On 1/23/2017 4:47 AM, Les Cargill wrote:
>> You'd need two full setups of course, but it seems like simply
>> transferring in real time via analog would be preferrable to
>> SRC.
>
> I agree with Paul Stamler that the best approach to sample rate
> conversion is to avoid it. It seems that there are good SRC processes
> and not-so-good ones, just like there are good and not-so-good D/A and
> A/D converters, and good sample rate conversion requires a lot of
> computer horsepower.

Which any computer made in the last decade has plenty of for any audio
process.

> You have to, in essence, reconstruct the original
> waveform of the source, then re-sample the (virtually) reconstructed
> waveform at your desired rate. If the reconstruction isn't accurate,
> you'll end up re-sampling interpolated values, and that's what doesn't
> sound right.

And the only reason for that is bad programming. Which sadly does abound
of course.

>
> A while back, someone, maybe it was Mytek, made a converter that could
> use separate clocks for the D/A and A/D sections, so you could just pipe
> the output into the input and convert the sample rate in a single
> hardware box. It sounded better than any of the software SRC programs of
> its time. But then, Mytek's converters were among the best sounding ones
> in the industry, so they had a good start.

Yep, good DA-AD is going to beat bad software, and vice versa.

Trevor.

[JackA]

Still can't understand how you can remaster a master. Digitally enhance, yes, but no remastering. Allow me to explain. This Asia '80's hit was remixed from (3) stereo .ogg files. Obviously, more than (6) tracks were actually used, someone premixed the multiple tracks to minimize file trading size...
http://www.angelfire.com/empire/abpsp/images/heatofmoment.mp3

Note: No 3kHz were harmed during the remixing of this fine tune.

Jack

[Scott Dorsey]

Les Cargill <lcarg...@comcast.com> wrote:
>Don Pearce wrote:
>> On Mon, 23 Jan 2017 15:29:07 -0600,
>>

>> Have a look here. Someone has done a lot of work comparing the quality
>> of various SRCs.
>>
>> http://src.infinitewave.ca/
>

>Audacity comes out well in that. Interesting.

Not surprising, seeing that it's basically designed by mathematicians and
dsp people, and because it does the conversion as batch it has all the time in
the world and doesn't need to take any shortcuts.

The Audacity UI is just horrible, but the compute engine is hard to beat
inside.

Contrast this with say the old Sonic Studios systems that tried to do the
conversion in realtime on a 68k.

[Les Cargill]

Tree-fiddy.

--
Les Cargill

[JackA]

You might tell Mr. G', I've been receiving some nice site compliments. No, not just on audio content, but I explain the REAL WORLD of (re)mastering.

Anyway, don't hold Mr. G' up, I'm sure he has to rush to buy some new headphones so he can appreciate a song with, ahem, great (re)mastering.

Jack
>
> --
> Les Cargill

[alan.b...@rentrakmail.com]

On Saturday, January 21, 2017 at 2:19:43 PM UTC-5, James Price wrote:
> Do you think it's the norm for most studios to track at 24/96 or 24/48? Or is it a mixed bag?
>

> Which one do you track at?

the reason to do all first / best need quality recording of raw tracks w/ at least 24/48 or 24/96 is not just individual track dynamic range - which is way beyond the dynamics of human music. The main reason in the digital age to use 24/48 at least (and 24/96 is ez to achieve these days in A/D/A and DSP systems and also allows very gentle slop anti-aliasing filters that start above human hearing to be used, which essentially have no phase effect on the 20-20 khz audio range music signal) is more the 24 bit over 16 bit part than the 48 versus 96 sample rate. When all the multitrack mix down and EQ is done digitally with DSP (in the box or with outboard digital gear) as opposed to being processed with outboard analog gear (like neve and or SSL quote current and or vintage super analog stuff) hooked to the multichannel A/D/A systems - is due to DSP math processing and acuumulated and compounded round off errors. All the DAW mixer and EQ engines these days (pro tools, cubase, sonar, etc) are 64 bit - and when processing 24 bit wav files -- the round off error all stay way above impacting the 24 accuracies of the individual track wav files...... i have listened to 16 and 24 bit wav files processed with 40 bit and 64 bit DSP - the 16 with 40 bit is horrid, the 24 bit is better but the 24 bit processed with DAWs and digital outboard gear using 64 bit math - are VERY audibly better overall, and ez to hear the difference even when dithered down to 16/44.1 CD and or compressed with a high rate MP3.

[jjaj...@netscape.net]

Regardless of technology and even the highest bit and sample rates, man will always be the weakest link when it comes to music.

Jack

[mako...@yahoo.com]

i have listened to 16 and 24 bit wav files processed with 40 bit and 64 bit DSP - the 16 with 40 bit is horrid, the 24 bit is better but the 24 bit processed with DAWs and digital outboard gear using 64 bit math - are VERY audibly better overall, and ez to hear the difference even when dithered down to 16/44.1 CD and or compressed with a high rate MP3.

....

can you make a quantitative measurement to demonstrate this.

m

[Scott Dorsey]

<mako...@yahoo.com> wrote:
>i have listened to 16 and 24 bit wav files processed with 40 bit and 64 bit=
> DSP - the 16 with 40 bit is horrid, the 24 bit is better but the 24 bit p=
>rocessed with DAWs and digital outboard gear using 64 bit math - are VERY a=
>udibly better overall, and ez to hear the difference even when dithered dow=

>n to 16/44.1 CD and or compressed with a high rate MP3.
>
>....
>
>can you make a quantitative measurement to demonstrate this.

Probably not anymore. I think you'll be hard-pressed to find anything
still doing fixed-point math for audio applications any longer. Maybe some
cheaper standalone gear.

But then, I tend to avoid doing any processing in the digital domain at all.

[geoff]

You should give it a try.

geoff

[Scott Dorsey]

geoff <ge...@nospamgeoffwood.org> wrote:
>On 8/11/2017 8:10 AM, Scott Dorsey wrote:
>>
>> But then, I tend to avoid doing any processing in the digital domain at all.
>

>You should give it a try.

Well, sometimes you can't avoid it. For the most part, I should say that I
tend to avoid doing processing of any sort if it can be avoided.

[Geoff]

On 8/11/2017 8:17 AM, Scott Dorsey wrote:
> geoff <ge...@nospamgeoffwood.org> wrote:
>> On 8/11/2017 8:10 AM, Scott Dorsey wrote:
>>>
>>> But then, I tend to avoid doing any processing in the digital domain at all.
>>
>> You should give it a try.
>
> Well, sometimes you can't avoid it. For the most part, I should say that I
> tend to avoid doing processing of any sort if it can be avoided.
> --scott
>

Not even the universally applicable 3K-boost ?!!! (oops 3k)

geoff

[jjaj...@netscape.net]

3k at 24/96. Might give it a try, think it deserve better!!!

I am taking that (3k) self discovery to my grave!!

Jack

[John Williamson]

On 07/11/2017 21:38, jjaj...@netscape.net wrote:

> I am taking that (3k) self discovery to my grave!!
>

Best place for it.


--
Tciao for Now!

John.

[jjaj...@netscape.net]

On Tuesday, November 7, 2017 at 4:41:54 PM UTC-5, John Williamson wrote:
> On 07/11/2017 21:38, jjaj...@netscape.net wrote:
>
> > I am taking that (3k) self discovery to my grave!!
> >
> Best place for it.

I THOUGHT we were beyond this. Let's be friends.
Actually, when I have a MINUTE to spare, both you and Geoff can teach me everything you know about impressive sounding audio.

Peace.

Jack

[thekma...@gmail.com]

To be fair, a 3k boost or cut with EQ will be far more
audible than the differences between 24/96 or
24/48.

[Mike Rivers]

To be accurate, such a boost or cut will be equally noticeable at either
sample rate. Whether it's good or bad depends on the source.


--

For a good time, call http://mikeriversaudio.wordpress.com

[thekma...@gmail.com]

Mike Rivers wrote: "
To be accurate, such a boost or cut will be equally noticeable at either
sample rate. Whether it's good or bad depends on the source.


--

For a good time, call http://mikeriversaudio.wordpress.com "

That's more or less what I said. So much stink
is made about high res, especially in delivery,
that no one gives a second thought to
decsions made at the mastering level, or
back in the recording process.

[None]

theccchhhhhkkkkhhhhmaaah:

> So much stink is made about high res, especially in delivery,
> that no one gives a second thought to decsions made at the
> mastering level, or back in the recording process.

Yes, by now everyone knows that you don't have the slightest idea how
decisions are made at the mastering level or in the recording process. And
you never will. You're just too stupid. KSFNJU. FCKWAFA.

[Trevor]

On 8/11/2017 6:10 AM, Scott Dorsey wrote:
> <mako...@yahoo.com> wrote:
>> i have listened to 16 and 24 bit wav files processed with 40 bit and 64 bit=
>> DSP - the 16 with 40 bit is horrid, the 24 bit is better but the 24 bit p=
>> rocessed with DAWs and digital outboard gear using 64 bit math - are VERY a=
>> udibly better overall, and ez to hear the difference even when dithered dow=
>> n to 16/44.1 CD and or compressed with a high rate MP3.
>>
>> ....
>>
>> can you make a quantitative measurement to demonstrate this.

Far easier to make a quantitative measurement to show minute differences
than it is to hear them! And far easier to claim to hear minute
differences than to prove it. :-)

> Probably not anymore. I think you'll be hard-pressed to find anything
> still doing fixed-point math for audio applications any longer. Maybe some
> cheaper standalone gear.
>
> But then, I tend to avoid doing any processing in the digital domain at all.

Not enough added distortion for you?

Trevor.

[geoff]

On 8/11/2017 10:36 PM, Trevor wrote:

>>
>> But then, I tend to avoid doing any processing in the digital domain
>> at all.
>
> Not enough added distortion for you?

There was some plugin (VST !) version '500-series' module or
channel-strip not long ago that claimed to have modeled the tolerance
variations and nonlinearities of the individual discrete components of
the hardware version, randomly between instances.

That should do it.

geoff

[John Williamson]

Not really, as the real thing is pretty consistent each time you turn it
on, while differing between units. To minimise day to day differences,
you leave the unit turned on overnight.

The randomness you are referring to will give a different result every
time you turn it on as if you had bought a new unit overnight.

[Trevor]

:-)
Nah, just like tape emulation, or actually bouncing to tape and back,
the purists won't have a bar of it whether they can tell the difference
or not. You must do it the hard way, like playing vinyl, when digital
can easily capture all the wow, flutter, rumble, mis-tracking, pinch
effect, limited channel separation, frequency non linearities, THD, IMD
etc. forever in a more easily accessible format. :-)

Trevor.

[Trevor]

Er, leave the computer on then, just like you would the real unit. :-)
There is probably a "random" option setting anyway if you really care,
although I'm pretty sure you don't. :-)

Trevor.

[John Williamson]

It's software, so each time you call the program, it initialises
according to the settings. This means the only way to ensure continuity
is to either put the computer to sleep or hibernate it with the DAW and
plugins all open. Close the DAW program, and you get a newly randomised,
set of plugins when you re-open it, unless you can disable the
randomisation which sort of defeats the object.

[John Williamson]

Yes, but you lose the *warmth* of the analogue experience, as the
digital stuff robs it of all reality. ;-)

Or so the Audiophools think. Personally I would be processing the
digitisation to remove the imperfections as far as possible.

[mako...@yahoo.com]

snip
..... when digital

> can easily capture all the wow, flutter, rumble, mis-tracking, pinch

> effect, .....snip


what is the pinch effect?

m

[Geoff]

Or just use a different plugin, without the designed-in flaws, and enjoy
the accuracy instead.

geoff

[Geoff]

Naa - just turn the treble down a tad. Or move your head a few inches !

geoff

[Geoff]

"Ouch !" ?

geoff

[Trevor]

This is the first URL Google turns up.
https://www.vinylengine.com/turntable_forum/viewtopic.php?t=31865

I'll leave it up to you to search for others.

Trevor.

[mako...@yahoo.com]

ok thanks

thankfully most of us no longer have to worry about it

m

[Scott Dorsey]

<mako...@yahoo.com> wrote:
>On Thursday, November 9, 2017 at 1:50:23 AM UTC-5, Trevor wrote:
>> > what is the pinch effect?
>>
>> This is the first URL Google turns up.
>> https://www.vinylengine.com/turntable_forum/viewtopic.php?t=31865
>>
>> I'll leave it up to you to search for others.
>>
>

>ok thanks
>
>thankfully most of us no longer have to worry about it

Because we use modern fineline (van den Hul) styli.

[Trevor]

On 10/11/2017 1:55 AM, mako...@yahoo.com wrote:
> On Thursday, November 9, 2017 at 1:50:23 AM UTC-5, Trevor wrote:
>> On 9/11/2017 12:54 AM, mako...@yahoo.com wrote:
>>> snip
>>> ..... when digital
>>>> can easily capture all the wow, flutter, rumble, mis-tracking, pinch
>>>> effect, .....snip
>>>
>>>
>>> what is the pinch effect?
>>
>> This is the first URL Google turns up.
>> https://www.vinylengine.com/turntable_forum/viewtopic.php?t=31865
>>
>> I'll leave it up to you to search for others.
>>

> ok thanks
>
> thankfully most of us no longer have to worry about it

Yes, but strange many want to go back. I wonder how long it will last?

Trevor.

[Trevor]

On 10/11/2017 2:10 AM, Scott Dorsey wrote:
> <mako...@yahoo.com> wrote:
>> On Thursday, November 9, 2017 at 1:50:23 AM UTC-5, Trevor wrote:
>>>> what is the pinch effect?
>>>
>>> This is the first URL Google turns up.
>>> https://www.vinylengine.com/turntable_forum/viewtopic.php?t=31865
>>>
>>> I'll leave it up to you to search for others.
>>>
>>
>> ok thanks
>>
>> thankfully most of us no longer have to worry about it
>
> Because we use modern fineline (van den Hul) styli.
> --scott

VdH and Shibata styli were not commonly used in the glory days of vinyl,
even less so now by the masses. I'm glad I don't have to replace my
V15VMR anyway. Once copied to digital I can now play forever without
wearing out the stylus thank god!


Trevor.

[Les Cargill]

Scott Dorsey wrote:
> geoff <ge...@nospamgeoffwood.org> wrote:
>> On 8/11/2017 8:10 AM, Scott Dorsey wrote:
>>>
>>> But then, I tend to avoid doing any processing in the digital
>>> domain at all.
>>
>> You should give it a try.
>
> Well, sometimes you can't avoid it. For the most part, I should say
> that I tend to avoid doing processing of any sort if it can be
> avoided. --scott
>

I went for a long time thinking simple biquads were a solved problem.

Nope. There are good compromises. But they're hardly cookbook items.

--
Les Cargill

[geoff]

On 10/11/2017 5:12 PM, Les Cargill wrote:

>
> I went for a long time thinking simple biquads were a solved problem.
>
> Nope. There are good compromises. But they're hardly cookbook items.
>

Naa. Lots of work at the gym for them.

geoff

[jjaj...@netscape.net]

Yeah, high res' crap. Putting low res' content on high res' media or format outputs low res'.

Great point!!

Jack

[jjaj...@netscape.net]

This "warmth" they talk about is the vacuum tube equipment they use!

Jack