I beg to differ.
Because it has been determined that having a different response at one point in the chain can have a benefit there while introducing a non-flat response that can amply be corrected elsewhere in the chain.
For example, imagine correcting RIAA in, as you say "in the tweeter feed" or between the amplifier terminals and/or crossovers?
So the argument here is no longer about whether this a is frequency response thing (that is just going backwards as that has been aired before), when the proposal is not about frequency response, since as you say, it can be corrected elsewhere.
Bottom line, if there is a pay-off in this, it has to be something else not frequency response related. That has now been taken out of the equation, indeed you are saying it too, tight?
Cheers, Joe
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Hi Pano
The point is that we can manipulate the frequency response anywhere along the path (including pre-DAC EQ) and test its audibility, even bring in the age thingy. That discussion will go on forever and nothing here will change that,
But in this instance it comes down to the very topic itself, what is the benefit of manipulating the post-DAC filter and then EQ it elsewhere.
What I am trying to say at the outset, that the topic here is not about "frequency response tailoring" as we can still point to a flat response in the end.
Cheers, Joe
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Yes, I understand that, thus my reply. Any simple EQ up there is inaudible to me, so if I can hear what this is doing, it's happening somewhere else.
But I'm pretty sure it can be seen in some sort of measurement, if we can hear it. Maybe noise, maybe IMD or HD. Worth a look.
I have a theory, and since I have little knowledge in this space it is quite likely off the wall.
A few things that I have read related to needing opamps with extremely high slew rate and low distortion into the MHz range in order to have clean sound out of a DAC. I think with my limited grasp that this is probably related to the harmonics present in the square wave output of the DAC.
Now putting a first order filter dirctly at the output is I would think going to start to shape that square wave (round off the corners?) and this in turn would reduce the harmonics present into the Mhz. Would this not reduce the requirements of the opamps downstream that are having to do the work to convert this from a square wave, back to sine wave based waveforms?
That is, it isn't what is happening in the audible range that is important, but it is reducing the stresses on the opamps in the non-audible range that allows them to perform better in the audible range???
As I said this is just my speculation, as I don't understand it well, but is it something worth considering?
Tony.
A few things that I have read related to needing opamps with extremely high slew rate and low distortion into the MHz range in order to have clean sound out of a DAC. I think with my limited grasp that this is probably related to the harmonics present in the square wave output of the DAC.
Now putting a first order filter dirctly at the output is I would think going to start to shape that square wave (round off the corners?) and this in turn would reduce the harmonics present into the Mhz. Would this not reduce the requirements of the opamps downstream that are having to do the work to convert this from a square wave, back to sine wave based waveforms?
That is, it isn't what is happening in the audible range that is important, but it is reducing the stresses on the opamps in the non-audible range that allows them to perform better in the audible range???
As I said this is just my speculation, as I don't understand it well, but is it something worth considering?
Tony.
What is this square wave you refer to?
So then why do the filtering in the audio band, rather than at, say, 50KHz?
This was of course explored and while I agree with the point made, slew rate induced distortion has been noted as a major cause of digital sound by Charlie Hansen and others, including myself.
In this instance we may have to look beyond that. Because it is also obvious with zero feedback post-DAC circuits and also the transformer scenario presented. The latter in particular raises an interesting question, since the Zobel has very little affect above 100KHz. This indicates something below 100KHz to be a clue as to what is going on. This is not a frequency bandwidth that opamps have problems with.
That this is not a slew problem is also indicated that I have actually done this to only a few opamps, that the Oppos we do are all zero feedback and based on OTA plus buffer, even the buffer is open circuit. So we cut our teeth with this whole topic on non-amps in place.
No, I believe the that this has an effect on the DAC itself.
But, having said that, it works a treat with opamps too. Quite a treat indeed.
Cheers, Joe
Cart. Horse. First determine if there's anything to speculate about. So far, no measurements, no listening tests.
We are way beyond speculation, even if there is nothing wrong with speculation. Hardly did any good idea come without speculation first. In fact, some of the greats did very well from 'thought experiments' and then the facts unfurled naturally. That is what we are hoping for here.
But we are now talking about hundreds who have heard what this does in the last two years - does that count for nothing?
True!
The topic of this thread was chosen very carefully and in consultation - keywords "practical" implies actions that are doable and "implementations" what can be done to test it.
I would have thought that to be a reasonable approach, since now anybody can tests it and see for themselves. No hocus pocus, no magic incantations nor New Age humbuckery. Just go for it!
Cheers, Joe
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Example? I don't think you have a good grasp of the history of science, nor of how "thought experiments" interact with theory and corroborating physical experiments. (1)
Really? What is the "thought experiment" here? Where is the natural unfurling of facts?
Pretty much, yeah.
(1) A great example of a "thought experiment" in physics was Galileo's famous "experiment" dropping balls of different weights off the leaning tower of Pisa. He never did that. He wrote about that experiment but never performed it -- that was the "thought experiment", which was informed by data and theory. The actual experiment he did was to roll balls of different weights down inclined planes and measure the time they took, slowly increasing the angle of the planes, then extrapolating the curves to 90 degrees. Note that there was a thought experiment couched in mathematical principles, which he wrote up for publication, and actual physical experiments to corroborate the result (no natural unfurling of anything, just actual work). We call that science.
For a slightly different take on eq and DAC's that may help
explore what Joe has found,
DBX type 1 and type 4 noise reduction circuits happen to combine
extremely successfully. Type 4 addresses the shortcomings of A/D
to enhance analog to digital conversion by using the top 4db to create
an overload region, similar to how tape also has a overload region
so recordings made are capable of greater dynamic range.
Using a DBX type 1 noise reduction unit that has eq and compression whilst
recording and eq that must be opposite but expansion on playback
usually improves playback immensely, the A/D or D/D and DAC receiving a eq
and compressed version of audio, very much relieving them of having
to process transients. and in the process becoming capable as
assisting storage mediums, if a storage medium like a hard disk is used.
Results just using Type 1 are great but now as well, insert a type
4 noise reduction, so the eq version from the first stage of type 1
is further disallowed from overloading, and record that. with two
models of type 1 DBX you can monitor at the same time.
The recordings made and played back or just real time monitored
are very analogue in presentation, inferring eq and companding
in a compressed state at the classic ratio's 2.1 are I think what DAC's should be
processing, leaving expansion circuits in analogue stages to do the
mirror image 1.2 and at the same time correct eq that was placed
during recording in other words the famous emphasis and de-emphasis
invented by Murray G Crosby.
I think if we purchased CD's in a held state like this awaiting
conversion in playback equipment, we would benefit from
the work of many pioneers in this field, such as David Blackmer
Equipment used was a type 1 DBX 150x , a Yamaha CDRHD1500
and a DBX Quantum - with LM4562 and LME49710 where appropriate
in all pieces of gear.
explore what Joe has found,
DBX type 1 and type 4 noise reduction circuits happen to combine
extremely successfully. Type 4 addresses the shortcomings of A/D
to enhance analog to digital conversion by using the top 4db to create
an overload region, similar to how tape also has a overload region
so recordings made are capable of greater dynamic range.
Using a DBX type 1 noise reduction unit that has eq and compression whilst
recording and eq that must be opposite but expansion on playback
usually improves playback immensely, the A/D or D/D and DAC receiving a eq
and compressed version of audio, very much relieving them of having
to process transients. and in the process becoming capable as
assisting storage mediums, if a storage medium like a hard disk is used.
Results just using Type 1 are great but now as well, insert a type
4 noise reduction, so the eq version from the first stage of type 1
is further disallowed from overloading, and record that. with two
models of type 1 DBX you can monitor at the same time.
The recordings made and played back or just real time monitored
are very analogue in presentation, inferring eq and companding
in a compressed state at the classic ratio's 2.1 are I think what DAC's should be
processing, leaving expansion circuits in analogue stages to do the
mirror image 1.2 and at the same time correct eq that was placed
during recording in other words the famous emphasis and de-emphasis
invented by Murray G Crosby.
I think if we purchased CD's in a held state like this awaiting
conversion in playback equipment, we would benefit from
the work of many pioneers in this field, such as David Blackmer
Equipment used was a type 1 DBX 150x , a Yamaha CDRHD1500
and a DBX Quantum - with LM4562 and LME49710 where appropriate
in all pieces of gear.
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They for sure don't alert their readers to it.
Still, the main question was about the ill-defined lpf due to the DAC's output impedance. Do we agree that the cap after the DAC must be adapted for each DAC model to get a consistent roll-of?
The big challenge with this filtering approach and its impact (improvement) over the soundscene quality is to measure it.
The results of applying this filtering method is very audible, as is very obvious for those who decided to use it. The big questions are how to measure this, and what to measure. How to correlate what is to be heard with a measurement approach?
I will want to suggest to we direct the discussion into this filed, instead of pure speculations about it, or sceptical meanings about if this filtering it really have any impact at all over the audio signal.
I were very well presented lot variants (scenarios) for implementations of this filtering on many platforms, in many configurations. The ones interested in this subject may have now many practical informations to try it by themselves this approach.
There is not more productive to focus the discussion on the practical filed of implementation of this filtering, coming out with own findings, or exchange own experiences (eventual about some measurements), and trying so to shape a main measurement approach?
The results of applying this filtering method is very audible, as is very obvious for those who decided to use it. The big questions are how to measure this, and what to measure. How to correlate what is to be heard with a measurement approach?
I will want to suggest to we direct the discussion into this filed, instead of pure speculations about it, or sceptical meanings about if this filtering it really have any impact at all over the audio signal.
I were very well presented lot variants (scenarios) for implementations of this filtering on many platforms, in many configurations. The ones interested in this subject may have now many practical informations to try it by themselves this approach.
There is not more productive to focus the discussion on the practical filed of implementation of this filtering, coming out with own findings, or exchange own experiences (eventual about some measurements), and trying so to shape a main measurement approach?
I only want to mention that I had/have the same positive results when applying this filtering (same cap value) to very different DACs devices (ES9018, PCM1792). I used also the same post DAC processing circuits (module based on OPA1632).
One more thing: the improvement effect of this filtering it become more accentuate if correlated with a large capacities decoupling for the involved circuits/devices.
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Yes, indeed, is also my opinion that the filter it not necessary need a DAC output to work, creating or revealing the same effect/improvement. I used the same cap on a differential input of an different device (preamp) and I could observe the same improvement, or increasing the improvement if the preamp it was connected to a DAC system with this filtering already implemented.
My conclusion is that the effect amount it can be controlled/adjusted somehow if one apply this filtering in two different stages of the system, by adjusting the caps value to a optimum.
Well, one can manipulate the bandwidth of a circuit by other kind of filtering methods, as the phase of the signals it can be manipulated through different methods. Only the overall result is not the same. This cap it just "fix" something or many things in the same time, only by its presence over the differential lines, in a very simple end efficient approach...
My conclusion is that the effect amount it can be controlled/adjusted somehow if one apply this filtering in two different stages of the system, by adjusting the caps value to a optimum.
Well, one can manipulate the bandwidth of a circuit by other kind of filtering methods, as the phase of the signals it can be manipulated through different methods. Only the overall result is not the same. This cap it just "fix" something or many things in the same time, only by its presence over the differential lines, in a very simple end efficient approach...
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Then it's not really a post-DAC filter, but just a tone control, even if it is 'post'.
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