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48 better than 88.2? Digital Converters
Old 4th April 2010
  #61
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Quote:
Originally Posted by Wavelength View Post
What's the deal with being rude, man? Can't we have a discussion without that nonsense? I don't mind being wrong, because it's often a good way to learn, but I don't care for rudeness - it's just not really very helpful... But thanks anyway, I suppose :(
It was ment as a joke ... i thought the smiley face would at least hint at that.,.,

I just wanted to indicate that you should not see your flaw in understanding this as something major.
I used to think exactly what you thought untill someone enlighted me!
Old 4th April 2010
  #62
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Seriously, this conversation comes around and around and around soooo many times.

It helps almost not at all to debate pictures of sine waves with sample points on them and to hypothesise about what is and isn't correctly reconstructed. Almost every time this conversation starts (and the same diagrams are drawn each time) it is never resolved to anyone's satisfaction.

Shannon and Nyquist are applications of Fourier. The mathematical theory is very very robust.

Shannon was mostly concerned with our ability to capture the information in a bandwidth limited noisy channel (i.e. any real channel) - this information is encoded in any way you like, ampltitude, frequency, phase; doesn't matter. He proved the limits to the information capacity of any channel (and refuted the old saw about analog having infinite resolution). Nyquist addd to this by placing limits on how we can capture that information with a sampled stream. Between them Shannon and Nyquist proved that it is possible to capture all the information present; with approriate constraints on the sample rate derived from the channel bandwidth specification, and a bit depth derived from the S/N ratio. If you can draw a diagram that seems to show some information that is missed you will be violating either the bandwidth or signal to noise specifications of the channel. Simple as that.

Reconstruction is identical - the mathematics works perfectly well with time running backwards. With the sampled stream you will get back all the information (no matter how encoded) from the original channel.

If you want to read further, and understand just how a 44.1kHz sample rate can exactly reproduce a 20kHz sine wave of arbitary amplitude and phase (subject to the S/N of the channel) you could do worse than study Dan Lavry's articles on the subject. Dan put far more care and effort into those than is humanly reasonable, so the least everyone could do is go and read them and rewards Dan's efforts at lifting the clouds of ignorance.
Old 4th April 2010
  #63
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Wavelength's Avatar
 

Quote:
Originally Posted by Francis Vaughan View Post
Seriously, this conversation comes around and around and around soooo many times.

It helps almost not at all to debate pictures of sine waves with sample points on them and to hypothesise about what is and isn't correctly reconstructed. Almost every time this conversation starts (and the same diagrams are drawn each time) it is never resolved to anyone's satisfaction.

If you want to read further, and understand just how a 44.1kHz sample rate can exactly reproduce a 20kHz sine wave of arbitary amplitude and phase (subject to the S/N of the channel) you could do worse than study Dan Lavry's articles on the subject. Dan put far more care and effort into those than is humanly reasonable, so the least everyone could do is go and read them and rewards Dan's efforts at lifting the clouds of ignorance.
Ok cool. I'm new here, so I wouldn't know if this comes up all the time without resolution, and I wouldn't know that people create the same crappy MS Word pictures (hehe!) every time - I just made those pictures to convey what I couldn't put into words.

I'll read Dan's book, thanks for the tip.

B.
Old 4th April 2010
  #64
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I have an Apogee Duet.
Old 4th April 2010
  #65
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Surbitone's Avatar
I find the Lynx aurora converters to balance out nicely between sound & performance @ 88.2k. However, 96k always seems to sound the best in ABX tests and in the end result - just ain't really practical to mix large projects at 96k, 88.2 is pushing it a bit really if truth be told. I've found different converters to sound better at different sample rates, even though there is apparently no 'big' difference.
Old 4th April 2010
  #66
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tuRnitUpsuM's Avatar
 

Somebody pinch me its groundhog day all over again. heh
Every year this topic springs from the shadows and every year the same rhetoric is regurgitated.

192khz is a bigger number therefore its has to be better.

Hey if 192khz is better than anything below it..why dont we all record at 352.8 khz? does protools go that high? is DSD good enough for Icelandic/Estonian pop music?

Picture yourself on an assembly line.... watching all those bits to a product hurry past you and are being told what needs to be done at the position you are in. Now this is your first day.... and the day starts off well.(44.1-48khz) But later on in the day... you notice more and more of these bits are travelling down the line and its getting really tough to keep up but you try (88.2-96khz) when all of a sudden a foremen decides to stand behind you and consistently "encourages" you to speed up, get faster, get a move on, lets go go go....(176.4-192khz). what happens when your work load increases as well as the time it takes to do it in decreases ? you start producing errors!! right? until the absolute break point at which you just cannot carry on any longer.(352.8khz- 5.6 Mhz)

With each Increase in your sampling rate, you are increasing the chances your signal will contain errors. Its very simple logic.... speed things up and increase workload.... bad stuff may happen (not always).

cheers
Old 4th April 2010
  #67
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Bob Olhsson's Avatar
 

Using real world filters, there's lots of research showing a need for sample rates of at least 50kHz. or 60kHz. to avoid audible artifacts below 20kHz.

We got 44.1 x 16 bits for CDs because it allowed digital recordings to be edited using video tape machines and the use of an existing video disk system for mass replication in existing vinyl pressing plants. It was a compromise in audio quality that saved the record industry a lot of money. No knowledgeable scientist or engineer ever believed the marketing hype about it being perfect. It wasn't and it isn't.

We got 48kHz x 20 bits as a professional production minimum because the electronics industry in Japan imposed it as the standard for video production. Many knowledgeable people felt it should have been at least 50 kHz. but they didn't have enough financial clout to fight Sony, JVC and Panasonic.
Old 4th April 2010
  #68
urumita
 
7rojo7's Avatar
 

so it goes
went
Old 4th April 2010
  #69
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swafford's Avatar
 

Quote:
Originally Posted by Mike Brown View Post
Not sure what there is to disagree with.

I think its pretty straightforward logic...

We want our conversions between analog and digital to be accurate. The more accurate the better.
I'd have to disagree. I bought purple convertors, because I'm not interested in accurate, I want my sound to be purple.

YMMV.
Old 4th April 2010
  #70
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Dog_Chao_Chao's Avatar
 

interesting...
Just started a rock album 2 days ago.

First day we spent it tuning drums, choosing mics, placement, preamps, comps, eqs...etc.
In the end we made a test at 44.1 and 88.2. Aurora 16. I asked the band witch one they prefer, in a blind test. Everybody could tell them apart. I found that interesting too. So its not just me and a couple of other geeks. But then it was much harder to choose the one we liked better for that sound. We could all hear that the drumbass and bass sounded best at 88.2. It had better low freq information. It sounded closer to what we were listening when in the recording room.
But on the other side of the spectrum, not so sure. Cymbals and dist. guitars sounded warmer at 44.1 ( perhaps due to the lack of better high freq resolution). In the end we chose 88.2 as we thought it would be better to attenuate some freq if we dont like them in the end than not having them there in the first place.
I still have our test....

However, I do not think that better resolution equals better sound. Music is not a science contest. There are much more complex things to consider. As emotion and individual taste.
If not, how can anyone still be recording to tape? Its the sound that matters
Old 4th April 2010
  #71
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Well, see, back in the '50s at Bell Labs they didn't have any electric gee-tars to try out all these newfangled theorems. That's why they got it all wrong
Old 4th April 2010
  #72
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Quote:
I asked the band witch one they prefer, in a blind test. Everybody could tell them apart.
If you told them "I'm going to play you 'A', and now I'm going to play you 'B'" and they said they could tell a difference between "A" and "B", that doesn't tell you much.
Old 4th April 2010
  #73
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Quote:
Originally Posted by Cellotron View Post
The problem is that particular metaphor doesn't actually make sense when discussing PCM. What you're describing is a bitmap system, where higher resolution directly correlates to a more accurate representation. In a bitmap you never actually have a smooth curve - only the illusion of one based on being zoomed out from some stair steps.

This is NOT how PCM works! To take a different visual metaphor from Photoshop and similar graphics programs - PCM works much more like a vector system making Bezier curves - Bézier curve - Wikipedia, the free encyclopedia - where all you need is two sample points and a couple control points to construct any perfectly smooth curve shape imaginable. In these cases adding more sample points to describe the curve that is to be reconstructed doesn't make it represented any more accurately!



Best regards,
Steve Berson
Good post Brotha.
That makes sense, and the example curve betwwen the points is kina hypnotic. lol
Old 4th April 2010
  #74
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Quote:
Originally Posted by swafford View Post
I'd have to disagree. I bought purple convertors, because I'm not interested in accurate, I want my sound to be purple.

YMMV.

hehhehheh

I just write the word "accurate" on mine in sharpie... seems to be working.

Everyone says they WANT "accurate" when really most people want converters that "sound nice".....

They are not the same thing in most cases.
Old 4th April 2010
  #75
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Dog_Chao_Chao's Avatar
 

Quote:
Originally Posted by chris319 View Post
If you told them "I'm going to play you 'A', and now I'm going to play you 'B'" and they said they could tell a difference between "A" and "B", that doesn't tell you much.
hum...thats why I wrote that it was a blind test. I just asked if they prefer n1 or n2. There was no way to know...They are not beginners though. Really good musicians can tell subtle differences apart.
Old 4th April 2010
  #76
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Dog_Chao_Chao's Avatar
 

Quote:
Originally Posted by Mike Brown View Post
hehhehheh

I just write the word "accurate" on mine in sharpie... seems to be working.

Everyone says they WANT "accurate" when really most people want converters that "sound nice".....

They are not the same thing in most cases.
exactly
Old 4th April 2010
  #77
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Quote:
Originally Posted by Dog_Chao_Chao View Post
interesting...
Just started a rock album 2 days ago.

First day we spent it tuning drums, choosing mics, placement, preamps, comps, eqs...etc.
In the end we made a test at 44.1 and 88.2. Aurora 16. I asked the band witch one they prefer, in a blind test. Everybody could tell them apart. I found that interesting too. So its not just me and a couple of other geeks. But then it was much harder to choose the one we liked better for that sound. We could all hear that the drumbass and bass sounded best at 88.2. It had better low freq information. It sounded closer to what we were listening when in the recording room.
But on the other side of the spectrum, not so sure. Cymbals and dist. guitars sounded warmer at 44.1 ( perhaps due to the lack of better high freq resolution). In the end we chose 88.2 as we thought it would be better to attenuate some freq if we dont like them in the end than not having them there in the first place.
I still have our test....

However, I do not think that better resolution equals better sound. Music is not a science contest. There are much more complex things to consider. As emotion and individual taste.
If not, how can anyone still be recording to tape? Its the sound that matters
Even though this has got kinda of side tracked...THIS IS EXACTLY WHY I STARTED THIS THREAD...do yourself a favour and try 48...for rock drums it's weird but I think it really works well...it won't take you long as you seem to have the time (which is great) really try it..it's a lot different than 44 IMHO of course.
Old 4th April 2010
  #78
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Dog_Chao_Chao's Avatar
 

Quote:
Originally Posted by glissando View Post
Even though this has got kinda of side tracked...THIS IS EXACTLY WHY I STARTED THIS THREAD...do yourself a favour and try 48...for rock drums it's weird but I think it really works well...it won't take you long as you seem to have the time (which is great) really try it..it's a lot different than 44 IMHO of course.
Im always willing to try different approaches to my work. So I will. Although I dont have that much time as you suggest! Actually I dont have a single free day the last 4 month and the 3 upcoming! I just prefer to work really hard before pressing rec...Thanks.
Old 4th April 2010
  #79
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Quote:
Originally Posted by Cellotron View Post
The problem is that particular metaphor doesn't actually make sense when discussing PCM. What you're describing is a bitmap system, where higher resolution directly correlates to a more accurate representation. In a bitmap you never actually have a smooth curve - only the illusion of one based on being zoomed out from some stair steps.

This is NOT how PCM works! To take a different visual metaphor from Photoshop and similar graphics programs - PCM works much more like a vector system making Bezier curves - Bézier curve - Wikipedia, the free encyclopedia - where all you need is two sample points and a couple control points to construct any perfectly smooth curve shape imaginable. In these cases adding more sample points to describe the curve that is to be reconstructed doesn't make it represented any more accurately!



Best regards,
Steve Berson
I love you Wikipedia guys. bezier ? worse analogy than mine. What do vector graphics have to do with audio. We are talking about resolution. Beziers are parametric they are dependent from resolution and are based on a mathematical formula.

wav file is stored in a data structure so therefore they are bit-mapped not bitmapped like a graphic but stored in an array (matrix) of a certain size and length..... or resolution and depth as more commonly used terminology. When stored, it is a map of bits and then a stream of bits or a bit stream when played back. When bits are stored to an array or a table or whatever they are mapped. generic terminology

PCM refers to a signal moving between 2 extremes a high and a low. It is not a direct correlation to what the actual stored data being transferred is. It is only a stream of bits that could be any depth or resolution. As far as PCM is concerned it is some voltage or no voltage not a resolution or bit depth or your irrelevant analogy, a bezier curve ? which I still don't understand where you pulled that one from. Are you saying the visual representation of these two extremes resembles a curve? Well I guess it does sort of? I think? Though when it is re-clocked it
looks more like _| |___| |___| |___| | __

PCM is a vehicle...... resolution and depth are the different animals, s-to-n ratio and dynamic range are synonymous with bit depth and range of frequency would be the sample rate. Computer networks ultimately use PCM as its vehicle is that data represented in bit depth and resolution? or packets? or bezier curves? lol

With raster graphics it's very similar conceptually. There is a bit depth (color) and a resolution (graph/matrix). Both data structures for a wav file and say a .bmp are very similar as far as the way they are stored. Sure the way they are displayed/reproduced are obviously way different. A .wav file is static... to 'modulate'? isn't that a dynamic? In computer science resolution is resolution and bit depth is bit depth, it is universal. You are talking about something totally different. I am not directly comparing a raster graphics file to an audio file. I am simply comparing resolution to resolution. Maybe directly comparing a picture to an audio file was a bad idea, but if a picture has 2x the resolution as and other picture and a music file has 2x the resolution as another music file, the 2x one looks/sounds better since it is clearer and has more data representing it. That was all I was trying to convey. I realize on the grand scale wav and raster graphics are not compatible.

bezier ....
Old 4th April 2010
  #80
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Quote:
Originally Posted by Dog_Chao_Chao View Post
hum...thats why I wrote that it was a blind test. I just asked if they prefer n1 or n2. There was no way to know...They are not beginners though. Really good musicians can tell subtle differences apart.
If they know there is a difference between n1 and n2, they are more likely to say they hear a difference between them. If you play n1 and then play n1 again and they say they hear a difference, that tells us something right there.
Old 4th April 2010
  #81
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Quote:
It is only a stream of bits that could be any depth or resolution.
PCM is a stream of samples in which the bits are in parallel. If you're thinking of "a stream of bits" you're thinking of DSD or PDM like those little Korg pocket recorders use.
Old 4th April 2010
  #82
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Cursed Lemon's Avatar
Quote:
Originally Posted by monomer View Post
Shannon/Niquist specificly state that you can only perfectly reconstruct signals BELOW niquist frequency.
Your 22050Hz sine is not part of it.

And you confuse the data that is captured (which looks square) with what the DAC reconstructs from it (which is a sine).

A 22k square, when sampled, will come out of the DAC looking like a sine.
This is because all upper harmonics of the square (which make up the square part of the signal) will be filtered away beforehand, leaving only the fundamental (sine) to be sampled.
The sample representation of this sine will LOOK like a square, but don't confuse this with the square you wanted to sample.
It is the sample representation of a sine wave, not a square.
Once these square samples will hit the DAC they will become a sine.

heh
Okay, I just tested that and you're correct, but that only applies to my extreme example.

I think I'm still correct regarding lower frequencies with room for harmonics.
Old 4th April 2010
  #83
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Quote:
Originally Posted by chris319 View Post
PCM is a stream of samples in which the bits are in parallel. If you're thinking of "a stream of bits" you're thinking of DSD or PDM like those little Korg pocket recorders use.
well samples are represented in bits at least in/on/ or about a computer (scratch head) or any DIGTITAL device
hence the word digit, which is made of of 2 what? if you go to the paint store, color samples are represented on paper?
At the candy store samples, are in a jar. pretty cool. Do you know if you have 8 bits it's then a bytestream?
Now... If i have 8 pieces of candy, what do they call that? Over-sampling? you may get a cavity
from too much sugar or a big belly.


stop it please?
Old 4th April 2010
  #84
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monomer's Avatar
 

Quote:
Originally Posted by robertshaw View Post
I love you Wikipedia guys. bezier ? worse analogy than mine. What do vector graphics have to do with audio. We are talking about resolution. Beziers are parametric they are dependent from resolution and are based on a mathematical formula.

wav file is stored in a data structure so therefore they are bit-mapped not bitmapped like a graphic but stored in an array (matrix) of a certain size and length..... or resolution and depth as more commonly used terminology. When stored, it is a map of bits and then a stream of bits or a bit stream when played back. When bits are stored to an array or a table or whatever they are mapped. generic terminology

PCM refers to a signal moving between 2 extremes a high and a low. It is not a direct correlation to what the actual stored data being transferred is. It is only a stream of bits that could be any depth or resolution. As far as PCM is concerned it is some voltage or no voltage not a resolution or bit depth or your irrelevant analogy, a bezier curve ? which I still don't understand where you pulled that one from. Are you saying the visual representation of these two extremes resembles a curve? Well I guess it does sort of? I think? Though when it is re-clocked it
looks more like _| |___| |___| |___| | __

PCM is a vehicle...... resolution and depth are the different animals, s-to-n ratio and dynamic range are synonymous with bit depth and range of frequency would be the sample rate. Computer networks ultimately use PCM as its vehicle is that data represented in bit depth and resolution? or packets? or bezier curves? lol

With raster graphics it's very similar conceptually. There is a bit depth (color) and a resolution (graph/matrix). Both data structures for a wav file and say a .bmp are very similar as far as the way they are stored. Sure the way they are displayed/reproduced are obviously way different. A .wav file is static... to 'modulate'? isn't that a dynamic? In computer science resolution is resolution and bit depth is bit depth, it is universal. You are talking about something totally different. I am not directly comparing a raster graphics file to an audio file. I am simply comparing resolution to resolution. Maybe directly comparing a picture to an audio file was a bad idea, but if a picture has 2x the resolution as and other picture and a music file has 2x the resolution as another music file, the 2x one looks/sounds better since it is clearer and has more data representing it. That was all I was trying to convey. I realize on the grand scale wav and raster graphics are not compatible.

bezier ....
He ment that when the samples exit the DA converter they get reconstructed into an analog signal.
This works by using a special filter.
Samples are -not- the real waveform.
They are just numbers that help the filter produce the right waveform.
The filter produces a continuous analog signal, so has a pretty high resolution.
If the frequency is low enough, the samples will seem to follow the waveform that will be reconstructed by the filter.

So samples are very much like the points on the curve in the example.
They are not raster points.
The data of the samples is just a number and can be put in an array, transferred, etc, etc..
But they represent information that drives the reconstruction filter.
Old 4th April 2010
  #85
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Cellotron's Avatar
Quote:
Originally Posted by robertshaw View Post
I love you Wikipedia guys. bezier ? worse analogy than mine. What do vector graphics have to do with audio. We are talking about resolution. Beziers are parametric they are dependent from resolution and are based on a mathematical formula.

wav file is stored in a data structure so therefore they are bit-mapped not bitmapped like a graphic but stored in an array (matrix) of a certain size and length..... or resolution and depth as more commonly used terminology. When stored, it is a map of bits and then a stream of bits or a bit stream when played back. When bits are stored to an array or a table or whatever they are mapped. generic terminology

PCM refers to a signal moving between 2 extremes a high and a low. It is not a direct correlation to what the actual stored data being transferred is. It is only a stream of bits that could be any depth or resolution. As far as PCM is concerned it is some voltage or no voltage not a resolution or bit depth or your irrelevant analogy, a bezier curve ? which I still don't understand where you pulled that one from. Are you saying the visual representation of these two extremes resembles a curve? Well I guess it does sort of? I think? Though when it is re-clocked it
looks more like _| |___| |___| |___| | __

PCM is a vehicle...... resolution and depth are the different animals, s-to-n ratio and dynamic range are synonymous with bit depth and range of frequency would be the sample rate. Computer networks ultimately use PCM as its vehicle is that data represented in bit depth and resolution? or packets? or bezier curves? lol
Very good points. Again - I was NOT trying to say that PCM equivalent to Bezier curves in the actual way it works at all. Instead I was saying Bezier is a better METAPHOR from the visual world than a bitmap is simply in terms of grasping that you can get a perfectly smooth reconstructed wave from only 2 sample points, and that additional samples in between these points wouldn't necessarily lead to this wave being more accurately represented. So again - the point I was trying to show from the metaphor is:
once necessary minimums are met to represent the desired bandwidth and dynamic range desired, more samples in a PCM file does not increase the actual accuracy of the representation of the sound in the same that more pixels in a bitmap does for a visual representation.

My apologies if I caused any confusion because by trying to find a better metaphor from the visual world to illustrate this single point made it seem as if I was making assertions regarding the mechanics of PCM that simply are not true.

Best regards,
Steve Berson
Old 4th April 2010
  #86
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everyone seems to dismiss the advantages of running plugins (especially software synths) at higher sample rates....?
Old 4th April 2010
  #87
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monomer's Avatar
 

Quote:
Originally Posted by Cursed Lemon View Post
Okay, I just tested that and you're correct, but that only applies to my extreme example.

I think I'm still correct regarding lower frequencies with room for harmonics.
Hmm.,,. nope heh
You never sample anything that is above nyquist rate unless your ADC is broken.
So ALL possible sound you can represent with samples will be lower than your extreme example!
In fact, you can try it with lower squares.
If you sample (or generate) a 10kHz square in 44.1 then you will have captures a 10kHz fundamental and the first harmonic of the square.
No other harmonics are invloved as these are above niquist rate and are cut off.
Everything will just be reconstructed as intended.

Of course, if you sample a much lower square wave then more harmonics will fall inside the bandwidth and you will capture those as well.
Old 4th April 2010
  #88
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Quote:
Originally Posted by monomer View Post
He ment that when the samples exit the DA converter they get reconstructed into an analog signal
well an analog signal isn't represented in bits or resolution is it? since it is (cool symbol huh?)Again let me SIMPLIFY.................my point

If you have a picture with 100 pixels and another picture with 200 which represents whatever the picture is of...... which has more detail?

If I have a painting of an object with 500 brush strokes and a painting (by the same artist) of the same object with 1000 brush strokes, which is clearer? or rather which has more detail?

it really is that simple I am sorry I compared screen resolution to audio resolution. But then again I was comparing an ear to and eye and it is pretty tough to eat an orange with your eye or hear an apple with your ear

honestly just stop the semantics pretty please? I get it, I made a bad analogy. Since I am not talking about PCM I am talking about clarity and depth as it relates to the ear not a textbook or Wikipedia

But I apologize for my ignorance. please forgive me. I am a condemned man, I have been humiliated. I was wrong
Old 4th April 2010
  #89
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Cellotron's Avatar
Quote:
Originally Posted by The Beatsmith View Post
everyone seems to dismiss the advantages of running plugins (especially software synths) at higher sample rates....?
There's definitely way less anti-aliasing when upsampling prior to processing and then downsampling after particularly for eq's and limiters. So for intermediary processes upsampling definitely makes a lot of sense. The thing is whether you need to keep the end result stored at the upsampled rate. I'd say most likely you don't.

Best regards,
Steve Berson
Old 4th April 2010
  #90
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monomer's Avatar
 

Quote:
Originally Posted by robertshaw View Post
PCM refers to a signal moving between 2 extremes a high and a low.
What you describe is sound.
PCM refers to an arbitrary continuous bandlimited signal...
"moving between 2 extremes, a high and a low" is meaningless in PCM.
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