how hard is to model electric circuits by software?
what is so inherant in analogue and even digital hardware that makes the sound hard to recreate by software?
recently i've been trying my hand at circuit bending, and the guts of these machines are just a bunch of resistors, capacitors and transistors (etc)
surely it isn't too hard to model these by software?
obviously "rebirth" was the most famous analogue modelling software. but why isn't there software that just lets you join a bunch of resistors, capacitors and transistors to make pretty much real analogue synths???
why isn't every analogue synth ever made already coverted to a coded equivalent?
On Apr 24, 2006, at 6:50 PM, hard off wrote:
how hard is to model electric circuits by software?
what is so inherant in analogue and even digital hardware that makes the sound hard to recreate by software?
Its not really that hard to model analogue sound in digital. But there is a cult of analog sound which believes that it is. I mean, you could measure the difference between the real analog and a high quality digital recreation, but you couldn't hear it.
recently i've been trying my hand at circuit bending, and the guts of these machines are just a bunch of resistors, capacitors and transistors (etc)
surely it isn't too hard to model these by software?
obviously "rebirth" was the most famous analogue modelling software. but why isn't there software that just lets you join a bunch of resistors, capacitors and transistors to make pretty much real analogue synths???
There is, its expensive software used by electronic engineers. It even models things like how the heat will be radiated from each component and how that heat will affect the circuit. There is simpler software which you can use for small circuits for free. Its called Eagle.
why isn't every analogue synth ever made already coverted to a coded equivalent?
They are many times over, but they are mostly proprietary software.
.hc
________________________________________________________________________ ____
I spent 33 years and four months in active military service and during that period I spent most of my time as a high class muscle man for Big Business, for Wall Street and the bankers. - General Smedley Butler
On Tue, 25 Apr 2006, Hans-Christoph Steiner wrote:
Its not really that hard to model analogue sound in digital. But there is a cult of analog sound which believes that it is. I mean, you could measure the difference between the real analog and a high quality digital recreation, but you couldn't hear it.
And chances are that the difference is due to bugs in the original analog hardware, such as unplanned auto-inductions and crosstalks and...
For the analog cult, analog gear is perfect, it's equated with perfect, and anything else is only seen as a poor imitation of analog. It's like how, for patriotic Americans, a Camembert is a poor imitation of Cheez-Whiz...
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Tue, Apr 25, 2006 at 11:25:26AM +0200, Hans-Christoph Steiner wrote:
On Apr 24, 2006, at 6:50 PM, hard off wrote:
how hard is to model electric circuits by software?
what is so inherant in analogue and even digital hardware that makes the sound hard to recreate by software?
Its not really that hard to model analogue sound in digital. But there is a cult of analog sound which believes that it is. I mean, you could measure the difference between the real analog and a high quality digital recreation, but you couldn't hear it.
Do you have empirical data to back up this statement? Has neuroscience really progressed to the point that we can claim absolutely that the human brain is incapable of the task of high frequency signal analysis?
component and how that heat will affect the circuit. There is simpler software which you can use for small circuits for free. Its called Eagle.
Also, `apt-cache search spice`.
Best regards,
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
On Apr 25, 2006, at 6:42 PM, Chris McCormick wrote:
On Tue, Apr 25, 2006 at 11:25:26AM +0200, Hans-Christoph Steiner wrote:
On Apr 24, 2006, at 6:50 PM, hard off wrote:
how hard is to model electric circuits by software?
what is so inherant in analogue and even digital hardware that makes the sound hard to recreate by software?
Its not really that hard to model analogue sound in digital. But there is a cult of analog sound which believes that it is. I mean, you could measure the difference between the real analog and a high quality digital recreation, but you couldn't hear it.
Do you have empirical data to back up this statement? Has neuroscience really progressed to the point that we can claim absolutely that the human brain is incapable of the task of high frequency signal analysis?
Neuroscience probably has not gotten to that point, but I don't know that much about neuroscience. But I do know enough about science to tell you that humans cannot tell the difference between such things in double-blind tests. So you don't need to know anything about neuroscience to make a useful judgement about this.
In a carefully controlled environment, with very nice equipment, something like 99.5% of people cannot tell the difference between things like tube and transistor amps, digital or analog audio, or electronic circuits or digital recreations. Try it yourself sometime.
Double-blind is the key. Humans are extremely sensitive to nuances that we give off. So to do it right, both the listener and the controller of the experiment cannot know about which source is which.
.hc
component and how that heat will affect the circuit. There is simpler software which you can use for small circuits for free. Its called Eagle.
Also, `apt-cache search spice`.
Best regards,
Chris.
chris@mccormick.cx http://mccormick.cx
________________________________________________________________________ ____
"[W]e have invented the technology to eliminate scarcity, but we are deliberately throwing it away to benefit those who profit from scarcity." -John Gilmore
On Tue, Apr 25, 2006 at 07:05:21PM +0200, Hans-Christoph Steiner wrote:
On Apr 25, 2006, at 6:42 PM, Chris McCormick wrote:
On Tue, Apr 25, 2006 at 11:25:26AM +0200, Hans-Christoph Steiner wrote:
On Apr 24, 2006, at 6:50 PM, hard off wrote:
what is so inherant in analogue and even digital hardware that makes the sound hard to recreate by software?
Its not really that hard to model analogue sound in digital. But there is a cult of analog sound which believes that it is. I mean, you could measure the difference between the real analog and a high quality digital recreation, but you couldn't hear it.
Do you have empirical data to back up this statement? Has neuroscience really progressed to the point that we can claim absolutely that the human brain is incapable of the task of high frequency signal analysis?
that much about neuroscience. But I do know enough about science to tell you that humans cannot tell the difference between such things in double-blind tests. So you don't need to know anything about
In a carefully controlled environment, with very nice equipment, something like 99.5% of people cannot tell the difference between things like tube and transistor amps, digital or analog audio, or electronic circuits or digital recreations. Try it yourself sometime.
I believe you, but could you point me to any articles that back this up? I've often wondered whether such research has been conducted, and have never managed to find it, but maybe I'm reading the wrong journals. Performing a double-blind test on myself would result in purely anecdotal evidence, which I'm not that interested in.
Best,
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
On Apr 26, 2006, at 4:27 AM, Chris McCormick wrote:
On Tue, Apr 25, 2006 at 07:05:21PM +0200, Hans-Christoph Steiner wrote:
On Apr 25, 2006, at 6:42 PM, Chris McCormick wrote:
On Tue, Apr 25, 2006 at 11:25:26AM +0200, Hans-Christoph Steiner wrote:
On Apr 24, 2006, at 6:50 PM, hard off wrote:
what is so inherant in analogue and even digital hardware that makes the sound hard to recreate by software?
Its not really that hard to model analogue sound in digital. But there is a cult of analog sound which believes that it is. I mean, you could measure the difference between the real analog and a high quality digital recreation, but you couldn't hear it.
Do you have empirical data to back up this statement? Has neuroscience really progressed to the point that we can claim absolutely that the human brain is incapable of the task of high frequency signal analysis?
that much about neuroscience. But I do know enough about science to tell you that humans cannot tell the difference between such things in double-blind tests. So you don't need to know anything about
In a carefully controlled environment, with very nice equipment, something like 99.5% of people cannot tell the difference between things like tube and transistor amps, digital or analog audio, or electronic circuits or digital recreations. Try it yourself sometime.
I believe you, but could you point me to any articles that back this up? I've often wondered whether such research has been conducted, and have never managed to find it, but maybe I'm reading the wrong journals. Performing a double-blind test on myself would result in purely anecdotal evidence, which I'm not that interested in.
About 10-15 years ago, I used to build my own speakers and amps, so I was in the "audiophile" world. There is some discussion about this kind of thing in audiophile magazines, but then there are many audiophiles that say things like "double-blind tests hide the differences in equipment". But basically none of them will argue that its hard to tell the difference between such things in double- blind tests.
I don't remember any specific studies off hand, but I do remember reading the Journal of the Audio Engineering Society (aka JAES) and the journal Speaker Builder a fair amount, and things like Stereophile magazine.
.hc
________________________________________________________________________ ____
If you are not part of the solution, you are part of the problem. - Eldridge Cleaver
On Wed, 26 Apr 2006, Hans-Christoph Steiner wrote:
but then there are many audiophiles that say things like "double-blind tests hide the differences in equipment".
So, basically, they are acknowledging that the differences in equipment are due to prejudice, branding, advertisements, colour of the case, gadgetism, and corporate/academic equipment procuration policies ;-)
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Wed, Apr 26, 2006 at 11:30:34PM -0400, Mathieu Bouchard wrote:
On Wed, 26 Apr 2006, Hans-Christoph Steiner wrote:
but then there are many audiophiles that say things like "double-blind tests hide the differences in equipment".
So, basically, they are acknowledging that the differences in equipment are due to prejudice, branding, advertisements, colour of the case, gadgetism, and corporate/academic equipment procuration policies ;-)
Heheh. One thing I've always wondered about (and I could be about to highlight my total lack of signal progressing understanding) is the representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
Best,
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
hi,
Am Donnerstag, 27. April 2006 05:39 schrieb Chris McCormick:
On Wed, Apr 26, 2006 at 11:30:34PM -0400, Mathieu Bouchard wrote:
On Wed, 26 Apr 2006, Hans-Christoph Steiner wrote:
but then there are many audiophiles that say things like "double-blind tests hide the differences in equipment".
So, basically, they are acknowledging that the differences in equipment are due to prejudice, branding, advertisements, colour of the case, gadgetism, and corporate/academic equipment procuration policies ;-)
Heheh. One thing I've always wondered about (and I could be about to highlight my total lack of signal progressing understanding) is the representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these
yes. but in analog audio adc's, you usually have an digital and/or an analogue antialiasing filter. also, frequencies close to nyquist are always problematic to reconstruct, due to the nature of the sampling.
often, the output is modulated when you have frequencies very close to sr/2.
two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
on analogue gear you have no samplerate at all. its continous.
when played as digital source, a waveform at sr/2 will always look sine-like, regardles of the osc source beeing a sine or rectangle osc. (at least on my scope ....)
depending on the used dac (=quality, circuitry), even a 10 khz wave looks more sine like, even if the source is meant to be a rectangle....
Best,
Chris.
greets,
chris
chris@mccormick.cx http://mccormick.cx
PD-ot mailing list PD-ot@iem.at http://lists.puredata.info/listinfo/pd-ot
On Thu, Apr 27, 2006 at 07:55:50AM +0200, Christian Klippel wrote:
Am Donnerstag, 27. April 2006 05:39 schrieb Chris McCormick:
On Wed, Apr 26, 2006 at 11:30:34PM -0400, Mathieu Bouchard wrote:
two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
on analogue gear you have no samplerate at all. its continous.
Yeh, 22050 was referring to the frequency of the wave. I guess my point is that when the come out of the speaker, there is often going to be some difference between something generated from an analogue source and something generated from a digital source, depending on many complicating factors like the cables used, rise/fall time of the TTL chips, etc. Then it comes down to whether or not we can hear that difference.
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
On Thu, 27 Apr 2006, Christian Klippel wrote:
on analogue gear you have no samplerate at all. its continous.
Continuity is a human-made concept that approximates reality. Since Quantum Mechanics, it is no longer believed that continuity can describe the world exactly, because it's been observed that all energy changes happen in sudden very small jumps. For example, the theory of incandescence made no sense at all until the assumption of continuity got dropped.
There's a samplerate for analogue gear, but it's both fuzzy and hidden. The signal-to-noise ratio of a digital transmission at 0 dB is 2^-n if you transmit n bits/sample (it's the rounding errors). If you turn this idea backwards, the bits/sample of an analogue wire would be the log2 of the signal-to-noise ratio of the wire. You could compute the bitrate of an analogue wire by combining the signal-to-noise ratios for frequencies all over the spectrum, to find a kind of "effective Nyquist frequency" related to the way the wire doesn't handle high frequencies very well...
This concept can be extended to any sequence of wires/connectors/filters. The bitrate of two things together may be smaller than both bitrates of individual things.
What do you think?
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Thu, 27 Apr 2006, Chris McCormick wrote:
representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
yeah, it would be more accurate, but are ears capable of hearing the difference? If the resonance of each sensor of the ear is linear, then each sensor can only react to one frequency band, and so if the smallest sensor's frequency is about 20000 Hz, then the ear is incapable of discerning.
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Apr 27, 2006, at 8:13 AM, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Chris McCormick wrote:
representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
yeah, it would be more accurate, but are ears capable of hearing the difference? If the resonance of each sensor of the ear is linear, then each sensor can only react to one frequency band, and so if the smallest sensor's frequency is about 20000 Hz, then the ear is incapable of discerning.
And looking one step before the ear: there are very, very few loudspeakers or headphones that can accurately reproduce 15,000 Hz and above. So even if you are generating them in analog or digital, they are getting filtered by the speakers long before reaching your ears.
.hc
________________________________________________________________________ ____
"If nature has made any one thing less susceptible than all others of exclusive property, it is the action of the thinking power called an idea, which an individual may exclusively possess as long as he keeps it to himself; but the moment it is divulged, it forces itself into the possession of everyone, and the receiver cannot dispossess himself of it." - Thomas Jefferson
On Thu, Apr 27, 2006 at 04:20:27PM +0200, Hans-Christoph Steiner wrote:
On Apr 27, 2006, at 8:13 AM, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Chris McCormick wrote:
And looking one step before the ear: there are very, very few loudspeakers or headphones that can accurately reproduce 15,000 Hz and above. So even if you are generating them in analog or digital, they are getting filtered by the speakers long before reaching your ears.
Good point. A quick survey of heaphones available at a local electronics store reveals that the top end varies between 13kHz and 30kHz. I guess speakers are going to have an even more varied response.
Best,
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
On Apr 28, 2006, at 3:30 AM, Chris McCormick wrote:
On Thu, Apr 27, 2006 at 04:20:27PM +0200, Hans-Christoph Steiner wrote:
On Apr 27, 2006, at 8:13 AM, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Chris McCormick wrote:
And looking one step before the ear: there are very, very few loudspeakers or headphones that can accurately reproduce 15,000 Hz and above. So even if you are generating them in analog or digital, they are getting filtered by the speakers long before reaching your ears.
Good point. A quick survey of heaphones available at a local electronics store reveals that the top end varies between 13kHz and 30kHz. I guess speakers are going to have an even more varied response.
Did they tell you in dB range of the frequency response? My laptop speakers do 1-1,000,000 Hz (+-100dB). Obviously, the important part there is the +-100dB. Generally, a decent frequency response is +-3dB. Many manufacturers use +-6dB because it makes them look much better. And many just make up the numbers.
If they don't tell you what the +-dB are, they are probably lying.
.hc
________________________________________________________________________ ____
"Information wants to be free." -Stewart Brand
On Fri, 28 Apr 2006, Hans-Christoph Steiner wrote:
Did they tell you in dB range of the frequency response? My laptop speakers do 1-1,000,000 Hz (+-100dB). Obviously, the important part there is the +-100dB. Generally, a decent frequency response is +-3dB. Many manufacturers use +-6dB because it makes them look much better. And many just make up the numbers. If they don't tell you what the +-dB are, they are probably lying.
It's like statistics in which you aren't told the standard deviation and the confidence level.
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Thu, Apr 27, 2006 at 02:13:17AM -0400, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Chris McCormick wrote:
representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
yeah, it would be more accurate, but are ears capable of hearing the difference? If the resonance of each sensor of the ear is linear, then each sensor can only react to one frequency band, and so if the smallest sensor's frequency is about 20000 Hz, then the ear is incapable of discerning.
This sounds a bit speculative to me - too many assumptions. It's probably not wise to be so quick to dismiss the viewpoint of "analogue cultism" until we have concrete research that confirms that humans are unable to perceive such differences.
Best,
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
hi,
Am Freitag, 28. April 2006 03:21 schrieb Chris McCormick:
On Thu, Apr 27, 2006 at 02:13:17AM -0400, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Chris McCormick wrote:
representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
yeah, it would be more accurate, but are ears capable of hearing the difference? If the resonance of each sensor of the ear is linear, then each sensor can only react to one frequency band, and so if the smallest sensor's frequency is about 20000 Hz, then the ear is incapable of discerning.
This sounds a bit speculative to me - too many assumptions. It's probably not wise to be so quick to dismiss the viewpoint of "analogue cultism" until we have concrete research that confirms that humans are unable to perceive such differences.
here are some attempts to explain why analog sounds analog ..... lets take the example of a square wave ..... you have two frequencies at once here: for one, the fundamental frequency of the waveform, and then the frequency resulting from the rise/fall time of the wave. since the latter is probably much higher than the fundamental one, you can never ever reproduce that with a dac (unless, of course, you are sampling in the mhz range).
on analog equipment, to the contrary, that compnent is preserved, maybe even shaped by the filters.
another example, the sine wave .... on analog equipment, you have an infinite number of different levels forming the sine. on a dac, you only have discrete values. the filtering after the dac shapes the waveform into something that should closely resemble the original wave, but only close (like, a square comming out as sine at all). since you have to jump between discrete levels, that again adds some harmonics, which need to get filtered out, effectively modifying the overall frequency response.
if you have purely digital generated sounds (for example, oscillators), they are far too perfect. in fact, each osc~ sounds the same. this is never the case in the analog world. you can have 100 osc circuits, all built the same, but you will never ever find two osc's that have an exact identical output. just 1 cm more wire makes a difference. each and every connection between analog equipment is some kind of a filter: you have the capacitance of the wire-pair, plus its resistance, plus the source's or sink's impedance.
of course all these differencies are very, very subtle, and you may not be able to hear the difference between two single elements. but if you now mix & combine all that together, the many non-linearities add up as well, giving a much wider spectrum (number of active frequencies) than what you would have with digital stuff.
oh, and last but not least: noise and distortion. every analog source has some noise, and maybe a small level of distortion. each part/wire adds some more noise. of course, the noise is part of the final sound as we hear it. many people complain that digital sound is too "sterile" ... thats why ... something is missing.
you can do a quick test yourself: create an osc~ object (copy the existing one, for example). inside the dsp routine, at the end, add some noise to the signal (sample[x] = sample[x] + ((1000 / random(1000)) * noise_level) or the like (thats probably non-working code above ;)... create some synth with these osc~'s and compare the generated sound against the standard osc~'s .... try with higher/lower noise_level settings. take it even further and add the same stuff to the used *~, +~, etc ..... if you multiply somewhere with a fixed number, make that number fluctuate a little as well. same for filter parameters, etc..... do that sample-by-sample, and not just reusing one single random value over the whole block ....
you will be surprised by the difference! (and of course, by the increase of cpu load as well ;-)
Best,
Chris.
greets,
chris
chris@mccormick.cx http://mccormick.cx
PD-ot mailing list PD-ot@iem.at http://lists.puredata.info/listinfo/pd-ot
On Apr 28, 2006, at 4:39 AM, Christian Klippel wrote:
hi,
Am Freitag, 28. April 2006 03:21 schrieb Chris McCormick:
On Thu, Apr 27, 2006 at 02:13:17AM -0400, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Chris McCormick wrote:
representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
yeah, it would be more accurate, but are ears capable of hearing the difference? If the resonance of each sensor of the ear is linear, then each sensor can only react to one frequency band, and so if the smallest sensor's frequency is about 20000 Hz, then the ear is incapable of discerning.
This sounds a bit speculative to me - too many assumptions. It's probably not wise to be so quick to dismiss the viewpoint of "analogue cultism" until we have concrete research that confirms that humans are unable to perceive such differences.
here are some attempts to explain why analog sounds analog ..... lets take the example of a square wave ..... you have two frequencies at once here: for one, the fundamental frequency of the waveform, and then the frequency resulting from the rise/fall time of the wave. since the latter is probably much higher than the fundamental one, you can never ever reproduce that with a dac (unless, of course, you are sampling in the mhz range).
on analog equipment, to the contrary, that compnent is preserved, maybe even shaped by the filters.
another example, the sine wave .... on analog equipment, you have an infinite number of different levels forming the sine. on a dac, you only have discrete values. the filtering after the dac shapes the waveform into something that should closely resemble the original wave, but only close (like, a square comming out as sine at all). since you have to jump between discrete levels, that again adds some harmonics, which need to get filtered out, effectively modifying the overall frequency response.
I think you mean the opposite. With decent quality digital vs. analog, you'll have a much more perfect sine wave from a digital source than an analog. And that's why things "sound" analog, because analog is imperfect.
.hc
if you have purely digital generated sounds (for example, oscillators), they are far too perfect. in fact, each osc~ sounds the same. this is never the case in the analog world. you can have 100 osc circuits, all built the same, but you will never ever find two osc's that have an exact identical output. just 1 cm more wire makes a difference. each and every connection between analog equipment is some kind of a filter: you have the capacitance of the wire-pair, plus its resistance, plus the source's or sink's impedance.
of course all these differencies are very, very subtle, and you may not be able to hear the difference between two single elements. but if you now mix & combine all that together, the many non-linearities add up as well, giving a much wider spectrum (number of active frequencies) than what you would have with digital stuff.
oh, and last but not least: noise and distortion. every analog source has some noise, and maybe a small level of distortion. each part/wire adds some more noise. of course, the noise is part of the final sound as we hear it. many people complain that digital sound is too "sterile" ... thats why ... something is missing.
you can do a quick test yourself: create an osc~ object (copy the existing one, for example). inside the dsp routine, at the end, add some noise to the signal (sample[x] = sample[x] + ((1000 / random(1000)) * noise_level) or the like (thats probably non-working code above ;)... create some synth with these osc~'s and compare the generated sound against the standard osc~'s .... try with higher/lower noise_level settings. take it even further and add the same stuff to the used *~, +~, etc ..... if you multiply somewhere with a fixed number, make that number fluctuate a little as well. same for filter parameters, etc..... do that sample-by-sample, and not just reusing one single random value over the whole block ....
you will be surprised by the difference! (and of course, by the increase of cpu load as well ;-)
Best,
Chris.
greets,
chris
chris@mccormick.cx http://mccormick.cx
PD-ot mailing list PD-ot@iem.at http://lists.puredata.info/listinfo/pd-ot
PD-ot mailing list PD-ot@iem.at http://lists.puredata.info/listinfo/pd-ot
________________________________________________________________________ ____
Man has survived hitherto because he was too ignorant to know how to realize his wishes. Now that he can realize them, he must either change them, or perish. -William Carlos Williams
On Fri, 28 Apr 2006, Chris McCormick wrote:
On Thu, Apr 27, 2006 at 02:13:17AM -0400, Mathieu Bouchard wrote: This sounds a bit speculative to me - too many assumptions. It's probably not wise to be so quick to dismiss the viewpoint of "analogue cultism" until we have concrete research that confirms that humans are unable to perceive such differences.
Why wouldn't it be wise? I'm happy with my bitrate and I don't care whether analogue cultists are right or wrong. I can happily trim expenses by buying medium-bitrate gear knowing that the people that I won't please are the analogue cultists.
It's only unwise if my grant request gets evaluated by a cultist, but then, in this case, it doesn't matter either whether analogue cultists are right or wrong: all that matters is that I speak in consonance with their beliefs.
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Fri, Apr 28, 2006 at 09:09:16PM -0400, Mathieu Bouchard wrote:
On Fri, 28 Apr 2006, Chris McCormick wrote:
On Thu, Apr 27, 2006 at 02:13:17AM -0400, Mathieu Bouchard wrote: This sounds a bit speculative to me - too many assumptions. It's probably not wise to be so quick to dismiss the viewpoint of "analogue cultism" until we have concrete research that confirms that humans are unable to perceive such differences.
Why wouldn't it be wise? I'm happy with my bitrate and I don't care whether analogue cultists are right or wrong. I can happily trim expenses
I'm not talking about right or wrong; I'm talking about acknowledging that there possibly are audible differences in the analogue and digital "sound" and that a preference for one or the other is totally subjective personal preference. I myself love the smell of aliased digital filters in the morning, but I will defend the preference of those with the squidy analogue love.
Best,
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
On Sat, 29 Apr 2006, Chris McCormick wrote:
Why wouldn't it be wise? I'm happy with my bitrate and I don't care whether analogue cultists are right or wrong. I can happily trim expenses
I'm not talking about right or wrong; I'm talking about acknowledging that there possibly are audible differences in the analogue
We're talking about right or wrong to assume that there are audible differences.
and digital "sound" and that a preference for one or the other is totally subjective personal preference. I myself love the smell of aliased digital filters in the morning, but I will defend the preference of those with the squidy analogue love.
We were not talking about which one of analogue or digital is best, but only of whether there is a difference at all. If there is no difference then the rest is moot (or a matter of delusions).
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
Qui, 2006-04-27 às 11:39 +0800, Chris McCormick escreveu:
Heheh. One thing I've always wondered about (and I could be about to highlight my total lack of signal progressing understanding) is the representation of two waveforms at Nyquist; if you have a squarewave and a sawtooth wave, both sampled at 44100 and playing at a frequency of 22050, these waves will both be represented by two samples each, correct? So when they are sent back into the analogue domain, won't these two waveforms look identical? Wouldn't they look like exactly the same sound? On analogue gear, wouldn't it be the case that the exact shape of the waveforms would be more accurately represented (e.g. a sawtooth and a squarewave at 22050 would look like such)?
Best,
Chris.
Your right that both waves will be identical. When converted to analog, both will be converted more or less to a sine wave at 22050, since the DAC has a lowpass filter set to this frequency to avoid aliasing to higher frequencies.
With analog hardware, when passing through a system that will have less response to higher frequencies, the same will happen, both will be more or less sinewaves. Actually, if you put a lowpass filter at 22k, the result will be the same.
Basically, the squarewave and sawtooth you mentioned, with a fundamental of 22050, will only differ in their partials, and these will have a frequency which is >= 44100.
So, there's no way to turn it around... You can faithfully represent digitally anything with frequency <= 22050.
Never forget that if you put an analog square wave on one end of a long cable, on the other end what you will receive won't be a square wave, but a rounded version of it. Moreover, a squarewave is a digital concept mainly. There's really no way to do it in analog.
-- Miguel Ramos
On Thu, Apr 27, 2006 at 02:27:29PM +0100, Miguel Ramos wrote:
Never forget that if you put an analog square wave on one end of a long cable, on the other end what you will receive won't be a square wave, but a rounded version of it.
Yeah, and the rounding as a result of an analogue signal coming through a cable will probably be different to the rounding generated by a dac from a digital signal. I guess this is my point. There is going to be a difference in the wave forms, but it comes down to whether we can hear a generic difference in a waveform generated from analogue gear, or from digital gear.
Moreover, a squarewave is a digital concept mainly. There's really no way to do it in analog.
And all digital signals are actually analogue on the wire.
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
Sex, 2006-04-28 às 09:19 +0800, Chris McCormick escreveu:
On Thu, Apr 27, 2006 at 02:27:29PM +0100, Miguel Ramos wrote:
Never forget that if you put an analog square wave on one end of a long cable, on the other end what you will receive won't be a square wave, but a rounded version of it.
Yeah, and the rounding as a result of an analogue signal coming through a cable will probably be different to the rounding generated by a dac from a digital signal. I guess this is my point. There is going to be a difference in the wave forms, but it comes down to whether we can hear a generic difference in a waveform generated from analogue gear, or from digital gear.
Yes, it *may* be different, if nothing else is working as a low pass filter. Each subsequent stage of an amplifier also has a frequency response, and will also work as a low pass or band pass filter. But the square wave and saw wave at 22.05kHz you mentioned, only differ at frequencies above 44.1kHz! It's possible to accept that 22.05kHz is an audible frequency, for the sake of argument, but above 44.1kHz?? And if that would be audible, do you think there's any chance this difference of >=44.1kHz frequencies will pass on to the speakers and from then to spl?
Moreover, a squarewave is a digital concept mainly. There's really no way to do it in analog.
And all digital signals are actually analogue on the wire.
Well, but there's a big difference, don't you see? A signal processing chain has many elements. With a digital square or saw wave, your processing is working on an as-close-as-possible approximation of a square or saw wave. When converted to analog, the low pass filter does the best job possible of approximating the audible part of the timbre of a square or saw wave to analog. In analog, you start working with a crude approximation of a square or saw wave, full of peaks; as the signal goes through more and more processing elements, the crudeness increases...
Miguel
On Thu, 27 Apr 2006, Miguel Ramos wrote:
Never forget that if you put an analog square wave on one end of a long cable, on the other end what you will receive won't be a square wave, but a rounded version of it. Moreover, a squarewave is a digital concept mainly. There's really no way to do it in analog.
This is the cause of the limited "bitrate" of analog cables that I mention in the previous mail.
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Fri, Apr 28, 2006 at 08:53:21PM -0400, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Miguel Ramos wrote:
Never forget that if you put an analog square wave on one end of a long cable, on the other end what you will receive won't be a square wave, but a rounded version of it. Moreover, a squarewave is a digital concept mainly. There's really no way to do it in analog.
This is the cause of the limited "bitrate" of analog cables that I mention in the previous mail.
I would say it's more due to the resistance, impedance, capacitance of each individual wire that has a filtering effect on an electrical signal that moves through it.
Chris.
------------------- chris@mccormick.cx http://mccormick.cx
On Sat, 29 Apr 2006, Chris McCormick wrote:
On Fri, Apr 28, 2006 at 08:53:21PM -0400, Mathieu Bouchard wrote:
On Thu, 27 Apr 2006, Miguel Ramos wrote:
Never forget that if you put an analog square wave on one end of a long cable, on the other end what you will receive won't be a square wave, but a rounded version of it. Moreover, a squarewave is a digital concept mainly. There's really no way to do it in analog.
This is the cause of the limited "bitrate" of analog cables that I mention in the previous mail.
I would say it's more due to the resistance, impedance, capacitance of each individual wire that has a filtering effect on an electrical signal that moves through it.
That's the same thing, really. Capacitor effects and coil effects act as lowpass/highpass filters (though I never really recall which is which). Lowpass-filtering of a squarewave makes the Gibbs phenomenon appear: a wavy pattern appears at each discontinuity.
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada
On Apr 27, 2006, at 5:30 AM, Mathieu Bouchard wrote:
On Wed, 26 Apr 2006, Hans-Christoph Steiner wrote:
but then there are many audiophiles that say things like "double- blind tests hide the differences in equipment".
So, basically, they are acknowledging that the differences in equipment are due to prejudice, branding, advertisements, colour of the case, gadgetism, and corporate/academic equipment procuration policies ;-)
Ssshh, don't tell them that! :)
.hc
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As we enjoy great advantages from inventions of others, we should be glad of an opportunity to serve others by any invention of ours; and this we should do freely and generously. - Benjamin Franklin