[nickbatz]

Hey, why (if true) does oversampling not accomplish the same thing as using a SR high enough to keep the brick wall filter ringing out of the audible spectrum?

[Don Pearce[_3_]]

On Fri, 8 Feb 2019 11:11:11 -0800 (PST), nickbatz
wrote:

Hey, why (if true) does oversampling not accomplish the same thing as using a SR high enough to keep the brick wall filter ringing out of the audible spectrum?

Oversampling IS using a high sampling rate followed by a brickwall
filter. That is followed by decimation and the result is burned to a
CD.

d

[nickbatz]

Oversampling IS using a high sampling rate followed by a brickwall
filter. That is followed by decimation and the result is burned to a
CD.

I'm talking about oversampling converters being used on 44.1/48 audio vs. just recording at 96 or whatever.

[Don Pearce[_3_]]

On Fri, 8 Feb 2019 12:30:24 -0800 (PST), nickbatz
wrote:

Oversampling IS using a high sampling rate followed by a brickwall
filter. That is followed by decimation and the result is burned to a
CD.

I'm talking about oversampling converters being used on 44.1/48 audio vs. just recording at 96 or whatever.

Nobody just records at 44.1 or 96. All audio A/D converters record at
probably at least 1.5MHz. This is internally processed with brickwall
filters to less than the final Nyquist rate, whatever that may be.

d

[nickbatz]

Don, if you read my last post in response to Scott, you'll probably understand my lack of understanding.

I don't really know how else to phrase the question: what is the argument for storing audio files post-converter at anything over 44.1/48 if the filtering is being done way above the audible spectrum anyway?

[Don Pearce[_3_]]

On Fri, 8 Feb 2019 14:00:50 -0800 (PST), nickbatz
wrote:

Don, if you read my last post in response to Scott, you'll probably understand my lack of understanding.

I don't really know how else to phrase the question: what is the argument for storing audio files post-converter at anything over 44.1/48 if the filtering is being done way above the audible spectrum anyway?

OK, now I get you. Very little point at all, unless you are doing
something other than reproduce sound. The response of even the best
microphones will go horribly peaky above 20kHz, and it is as well to
get rid of it. All that extra spectrum can possibly do is waste power,
and possibly cause intermodulation distortion down into the actual
audible range.

But - although an A/D that works at 192kS/sec may be wasted digits,
the fact that it has been designed that way is usually a good
indication that its performance at 44.1 will be exemplary. Not always,
but usually.

d

[Les Cargill[_4_]]

nickbatz wrote:
Don, if you read my last post in response to Scott, you'll probably
understand my lack of understanding.

I don't really know how else to phrase the question: what is the
argument for storing audio files post-converter at anything over
44.1/48 if the filtering is being done way above the audible spectrum
anyway?


Some plugins work better at higher SR. Higher SR may make lower round
trip latency possible. It may emerge that for significant recordings,
future generations will find a use for the additional bandwidth.

The last bit seems unlikely but you never know.

--
Les Cargill

[Scott Dorsey]

In article ,
nickbatz wrote:
Hey, why (if true) does oversampling not accomplish the same thing as using a SR high enough to keep the brick wall filter ringing out of the audible spectrum?

It saves storage space. Why sample at some crazy high rate and store a lot
of data when you can just downsample it for storage and lose only bandwidth
you don't need?

Of course, nobody really uses oversampling any more, now that we live in the
age of sigma-delta converters. Unless you think of the sigma-delta method
as sort of the extreme end of oversampling.

I haven't seen a brick wall filter in more than twenty years now.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[nickbatz]

Thanks Scott.

Is that the same thing as 1-bit conversion? I thought that was different from PCM?

And in that case (sigma-delta) why does anyone use 96K these days?

My knowledge is clearly out of date...

[Mike Rivers[_2_]]

On 2/8/2019 3:38 PM, nickbatz wrote:
why does anyone use 96K these days?

One reason is that we've been brainwashed. Same for 192 kHz which some
people known for their good work insist that things recorded at 192 kHz
sample rate really sound better.

A practical reason to use higher than standard sample rate is that for a
given portion of the cycle, you have more samples to work with. When
doing signal processing in the digital domain (software plug-ins, for
instance), having more samples allows for better resolution within the
process. So applying the same EQ to both a standard sample rate
recording and a 2x or 4x sample rate recording may indeed sound better,
even when converted back to standard sample rate.



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

[nickbatz]

Well, processing 32-bit floating penis makes sense - and I'm almost convinced I kind of heard the difference one time during a blood moon while squinting and wearing lederhosen.

But I don't understand why having more samples to describe a sine wave that can only go one direction or the other makes any sense whatsoever.

I work mostly with instrument plug-ins and sample libraries these days, with occasional live overdubs, so it's not practical to sacrifice half my computer horsepower for such a small improvement - especially when the samples are recorded at 44.1 or 48 anyway.

[Mike Rivers[_2_]]

On 2/8/2019 8:12 PM, nickbatz wrote:
Well, processing 32-bit floating penis makes sense - and I'm almost convinced I kind of heard the difference one time during a blood moon while squinting and wearing lederhosen.

You need more virgins.

But I don't understand why having more samples to describe a sine wave that can only go one direction or the other makes any sense whatsoever.

It's more of a mastering argument. Everything you do to a waveform adds
more bits and usually creates harmonics. 24-bits leaves room for the
bits, and a doubling the sample rate keeps those harmonics that you can
deal with more gracefully if you leave them there until it's time to
bring the project back to a standard format.

I work mostly with instrument plug-ins and sample libraries these days, with occasional live overdubs, so it's not practical to sacrifice half my computer horsepower for such a small improvement - especially when the samples are recorded at 44.1 or 48 anyway.

You won't double your computer usage, but you'll increase it some. I'm
not arguing for 2x or 4x sample rates all the time. I record everything
at 24-bit 44.1 kHz because just about all the recording I do is live, in
the field. The less crap I record, the less I have to worry about what
to get rid of. If you like what you're doing, then just keep doing it.

On the rare occasions that I have a paying customer, if he asks for 96
kHz, I just push that button. The customer may not always be right, but
doing what he wants makes getting paid more certain.


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