[RadioHamster]

Hi,

I've read a comment by a JDS labs person saying DACs performed the best with windows volume at -0.1dB or 99% because higher volume creates less error, but somehow full volume resulted in more error. However, as I was reading other posts, I've also discovered this thing about low-volume channel imbalance caused by potentiometers in amps. Apparently this happens even in amps of the highest quality with high spec pots.

While I was listening to my DT770, I detected a quite noticeable channel imbalance, but I found out it's mostly the headphone by testing mono sound on different sources with side swapping too. Even though I think my magni 3 doesn't have much sound imbalance at low volume, I want the optimal settings that would eliminate potential errors the most so that I get the best overall SQ.

I know I'm being a bit nitpicky, but, does anyone know the golden rule considering both the potential channel imbalance and the benefit from the higher source volume, etc? I'm wondering if there's any noticeable difference in SQ of very quiet (or ambient) sounds when source volume is higher. If not, I'm thinking of perhaps reducing my source volume and increasing my amp volume to be relatively free of the channel imbalance.

Thanks!

 

[tomb]

First, the two are not related … at all.

1. Software volume - never heard the 99% story, but it doesn't makes sense. Less than 100% volume does make sense, however, because unless you are using ASIO or WASAPI drivers, every bit of the sound goes through the Windows engine. That means it interpolates the bits in the sound stream to reduce the volume. Thus, the "purity" of the sound stream has been degraded. WASAPI is available since Windows 7 and this should be used if you want the absolutely purist music stream coming from the PC.

2. Volume pot channel matching - this is a physical phenomenon related to the pot structure itself. Manufacturing resistors is a tolerance process. Some resistors have smaller tolerance than others. The very best have resistance values controlled down to 1% or even 0.1%. Potentiometers, especially ones made for stereo volume control, have much wider tolerances. The manufacture of the resistance components in volume potentiometers is much more difficult, because there are more factors in play than just simple resistance values. A volume pot has to have a logarithmic response, in order to match the loudness capability of the human ear. It must also change this volume consistently and evenly throughout a ~300 deg range of motion. They are most decidedly different than simple resistors.

In any event, it's very difficult - even with the highest quality volume pots - to have absolutely perfect matching throughout the travel of the pot. In most cases, the matching between channels is worst at the very bottom end - the points of highest resistance. This makes sense, because the greatest concentration of resistance is at the lowest volume settings. In our culture, the ALPS RK27 or TKD pots probably have the best reputation for volume matching, even at the low end. That said, even the ALPS RK097 - one used almost ubiquitously in small portable amps, does not have good volume matching when compared to the RK27. Here's a great resource where Tangent measured and recorded the resistance matching of several popular stereo volume pots over their travel: http://tangentsoft.net/audio/atten.html

 

[RadioHamster]

First, the two are not related … at all.

1. Software volume - never heard the 99% story, but it doesn't makes sense. Less than 100% volume does make sense, however, because unless you are using ASIO or WASAPI drivers, every bit of the sound goes through the Windows engine. That means it interpolates the bits in the sound stream to reduce the volume. Thus, the "purity" of the sound stream has been degraded. WASAPI is available since Windows 7 and this should be used if you want the absolutely purist music stream coming from the PC.

2. Volume pot channel matching - this is a physical phenomenon related to the pot structure itself. Manufacturing resistors is a tolerance process. Some resistors have smaller tolerance than others. The very best have resistance values controlled down to 1% or even 0.1%. Potentiometers, especially ones made for stereo volume control, have much wider tolerances. The manufacture of the resistance components in volume potentiometers is much more difficult, because there are more factors in play than just simple resistance values. A volume pot has to have a logarithmic response, in order to match the loudness capability of the human ear. It must also change this volume consistently and evenly throughout a ~300 deg range of motion. They are most decidedly different than simple resistors.

In any event, it's very difficult - even with the highest quality volume pots - to have absolutely perfect matching throughout the travel of the pot. In most cases, the matching between channels is worst at the very bottom end - the points of highest resistance. This makes sense, because the greatest concentration of resistance is at the lowest volume settings. In our culture, the ALPS RK27 or TKD pots probably have the best reputation for volume matching, even at the low end. That said, even the ALPS RK097 - one used almost ubiquitously in small portable amps, does not have good volume matching when compared to the RK27. Here's a great resource where Tangent measured and recorded the resistance matching of several popular stereo volume pots over their travel: http://tangentsoft.net/audio/atten.html

Here's the link to the post I've read it from.
I'm aware that ASIO/WASAPI bypasses DirectSound, but when I use ASIO/WASAPI on foobar, I notice that software volume control is still enabled through foobar's volume control, which means there's still bit manipulation going on before sending out sound stream. Since less volume means less purity, I'd like to keep the software volume as high as possible. But then if I keep it high, I gotta turn the volume low on my amplifier, which then introduces the potential channel imbalance problem. That's why I was wondering if there'd be a golden balance between the two.

I didn't know that a simple pot design could be that difficult, but if it is, then it makes sense why people find channel imbalance on a wide variety of amps on the market. Thanks for your insight into this issue

 

[MindsMirror]

It will depend on where and how bad the imbalance is on your particular pot, how loud that position is with your headphone and source, and how tolerant you are of it, so there's not really a universal golden rule. I'd start with your software volume at 100% and set the amp volume pot to the range where you would be listening. If you notice any imbalance in that range, then just lower the software volume a bit and repeat, until you find a level where the balance is acceptable to you.

 

[RadioHamster]

It will depend on where and how bad the imbalance is on your particular pot, how loud that position is with your headphone and source, and how tolerant you are of it, so there's not really a universal golden rule. I'd start with your software volume at 100% and set the amp volume pot to the range where you would be listening. If you notice any imbalance in that range, then just lower the software volume a bit and repeat, until you find a level where the balance is acceptable to you.

I didn't wanna do the trial and error to balance it myself as I could get caught up in it for days... but you're right, that sounds like the only way to go about this

Thanks for your input!

 

[Rhamnetin]

It will depend on where and how bad the imbalance is on your particular pot, how loud that position is with your headphone and source, and how tolerant you are of it, so there's not really a universal golden rule. I'd start with your software volume at 100% and set the amp volume pot to the range where you would be listening. If you notice any imbalance in that range, then just lower the software volume a bit and repeat, until you find a level where the balance is acceptable to you.

This. I used to have an amp that had horrible issues with its volume pot, but it was actually matched well at 100%. So I was limited to using DACs with variable out and adjusting volume there. Always kept Windows at 100% though.

 

[tomb]

Here's the link to the post I've read it from.
I'm aware that ASIO/WASAPI bypasses DirectSound, but when I use ASIO/WASAPI on foobar, I notice that software volume control is still enabled through foobar's volume control, which means there's still bit manipulation going on before sending out sound stream. Since less volume means less purity, I'd like to keep the software volume as high as possible. But then if I keep it high, I gotta turn the volume low on my amplifier, which then introduces the potential channel imbalance problem. That's why I was wondering if there'd be a golden balance between the two.

I didn't know that a simple pot design could be that difficult, but if it is, then it makes sense why people find channel imbalance on a wide variety of amps on the market. Thanks for your insight into this issue

Well, that thread you reference is full of crap, excusing my language. It's all a bunch of gobbledy-goop to make up for the glaring design choice (BAD) that NWAVGUY made with designing the O2. The O2 places the volume pot after the signal gain portion of the amplifier. That means it's highly susceptible to clipping, because there's nothing attenuating the signal prior to the gain stage. If the signal is too high - on high hat transients, for instance - then the O2 clips. So yeah, set your digital volume down a skosh so that doesn't happen. Or … buy/build any other headphone amplifier in existence that places the pot as the first thing the source signal sees. Then never, ever worry about clipping again. The volume pot (placed correctly in the circuit) can attenuate any signal, no matter how strong.

Finally, despite what your reference described, every DAC test that I've ever built or seen or read about is done with zero volume control on the DAC. A typical DAC has no volume control. Further, if you're testing one, there is no attenuation of the test signal - that's controlled by the testing software. Whatever you are using for input in the testing is used to attenuate the signal. However, that has nothing to do with the level output of the DAC. Again IMHO, this is all a crutch to allow the O2 to perform its best under the varying conditions of different DAC output levels.​

The other guys have given you great advice on your pot channel mis-matching. It's essentially a gain issue. Your amplifier has too much gain for the strength of the source you have. What I said above is true - a volume pot can attenuate any signal if it's the first thing in the amplifier circuit. However, with a strong source, that may mean the attenuation desired is in the channel mis-match range of the pot. If you were a DIY-er, you could do one of two things: 1) attenuate the signal with a resistor(s) ahead of the pot, throwing the signal into the higher range of the pot, where channel mis-matching is minimal or 2) buy a better pot and install it, instead. I sell kits for the Starving Student Millett Hybrid. We give specific instructions on adding resistors so as to push the attenuation of the signal into the "good" range of a somewhat inexpensive volume pot (as in Starving Student costs). Otherwise as I said, the other guys have given you great advice.

 

[ProtegeManiac]

I didn't know that a simple pot design could be that difficult, but if it is, then it makes sense why people find channel imbalance on a wide variety of amps on the market.

Analogue pots natively have imbalance, even the best ones, it's just a matter of how wide or narrow the range from the lowest setting up to where it gets balanced is.

Amp manufacturers get around that in two ways. First, set a very low low gain setting. I have a Meier Cantate.2 and its low gain setting is actually -10dB or something, usable only with IEMs (not that full volume on the HD600 won't damage your hearing) and high sensitivity headphones like Grados. This amp already has an otherwise solid ALPS potentiometer.

Second is to use a digital pot, problem is usually it requires buttons to control, which a lot of people don't like. Meier then got around that on the current generation amps by using a knob with an ADC on it that, in simple terms, translates the pot movement for the digital controller (think of how A/V receivers' DSP chips controls the volume with their knobs). Now the low gain setting is +4dB or thereabouts.

In your case, short of replacing the amp or the DAC (assuming your DAC might be outputting higher than 2V; some DACs like the Modi2 only have 1.5V on the output), just use Windows volume control. Just note though that on some DACs Windows control can be disabled depending on the drivers used.

 

[RadioHamster]

Well, that thread you reference is full of crap, excusing my language. It's all a bunch of gobbledy-goop to make up for the glaring design choice (BAD) that NWAVGUY made with designing the O2. The O2 places the volume pot after the signal gain portion of the amplifier. That means it's highly susceptible to clipping, because there's nothing attenuating the signal prior to the gain stage. If the signal is too high - on high hat transients, for instance - then the O2 clips. So yeah, set your digital volume down a skosh so that doesn't happen. Or … buy/build any other headphone amplifier in existence that places the pot as the first thing the source signal sees. Then never, ever worry about clipping again. The volume pot (placed correctly in the circuit) can attenuate any signal, no matter how strong.

Finally, despite what your reference described, every DAC test that I've ever built or seen or read about is done with zero volume control on the DAC. A typical DAC has no volume control. Further, if you're testing one, there is no attenuation of the test signal - that's controlled by the testing software. Whatever you are using for input in the testing is used to attenuate the signal. However, that has nothing to do with the level output of the DAC. Again IMHO, this is all a crutch to allow the O2 to perform its best under the varying conditions of different DAC output levels.​

I thought a comment from a reputable trade would be trustworthy, but I guess I should take everything with a grain of salt.

The other guys have given you great advice on your pot channel mis-matching. It's essentially a gain issue. Your amplifier has too much gain for the strength of the source you have. What I said above is true - a volume pot can attenuate any signal if it's the first thing in the amplifier circuit. However, with a strong source, that may mean the attenuation desired is in the channel mis-match range of the pot. If you were a DIY-er, you could do one of two things: 1) attenuate the signal with a resistor(s) ahead of the pot, throwing the signal into the higher range of the pot, where channel mis-matching is minimal or 2) buy a better pot and install it, instead. I sell kits for the Starving Student Millett Hybrid. We give specific instructions on adding resistors so as to push the attenuation of the signal into the "good" range of a somewhat inexpensive volume pot (as in Starving Student costs). Otherwise as I said, the other guys have given you great advice.

Attenuating signal with resistors/getting a better pot indeed sound interesting and more like a proper fix, but I haven't gone the DIY route yet, so I will keep that in mind for the future Thanks for your suggestions!

 

[RadioHamster]

Analogue pots natively have imbalance, even the best ones, it's just a matter of how wide or narrow the range from the lowest setting up to where it gets balanced is.

Amp manufacturers get around that in two ways. First, set a very low low gain setting. I have a Meier Cantate.2 and its low gain setting is actually -10dB or something, usable only with IEMs (not that full volume on the HD600 won't damage your hearing) and high sensitivity headphones like Grados. This amp already has an otherwise solid ALPS potentiometer.

Second is to use a digital pot, problem is usually it requires buttons to control, which a lot of people don't like. Meier then got around that on the current generation amps by using a knob with an ADC on it that, in simple terms, translates the pot movement for the digital controller (think of how A/V receivers' DSP chips controls the volume with their knobs). Now the low gain setting is +4dB or thereabouts.

In your case, short of replacing the amp or the DAC (assuming your DAC might be outputting higher than 2V; some DACs like the Modi2 only have 1.5V on the output), just use Windows volume control. Just note though that on some DACs Windows control can be disabled depending on the drivers used.

I don't have much knowledge in this matter, so I'd like to learn a few things from you:
1. If an ADC is used for the knob, then it later uses DAC to convert it back to analog right? Wouldn't that decrease signal purity and increase latency from going through an extra cycle of conversion?
2. Why do they use ADC in the first place? Can't they just use the digital pot directly on the input which comes from from a DAC?
3. To my knowledge, digital volume control always introduces loss of signal purity, unless it's very high resolution and hence not very noticeable. So doesn't using a digital pot again, compromise the fidelity of the sound?
4. I'm actually using modi2U. Why do you suggest replacing DACs with less than 2V output voltage?

Thanks!

 

[Rhamnetin]

I thought a comment from a reputable trade would be trustworthy, but I guess I should take everything with a grain of salt.

Yeah, engineers will have different takes and opinions on how to approach different problems, and some engineers are far more knowledgeable than others. What tomb says makes a lot of sense to me.

 

[castleofargh]

Well, that thread you reference is full of crap, excusing my language. It's all a bunch of gobbledy-goop to make up for the glaring design choice (BAD) that NWAVGUY made with designing the O2. The O2 places the volume pot after the signal gain portion of the amplifier. That means it's highly susceptible to clipping, because there's nothing attenuating the signal prior to the gain stage. If the signal is too high - on high hat transients, for instance - then the O2 clips. So yeah, set your digital volume down a skosh so that doesn't happen. Or … buy/build any other headphone amplifier in existence that places the pot as the first thing the source signal sees. Then never, ever worry about clipping again. The volume pot (placed correctly in the circuit) can attenuate any signal, no matter how strong.

Finally, despite what your reference described, every DAC test that I've ever built or seen or read about is done with zero volume control on the DAC. A typical DAC has no volume control. Further, if you're testing one, there is no attenuation of the test signal - that's controlled by the testing software. Whatever you are using for input in the testing is used to attenuate the signal. However, that has nothing to do with the level output of the DAC. Again IMHO, this is all a crutch to allow the O2 to perform its best under the varying conditions of different DAC output levels.​

The other guys have given you great advice on your pot channel mis-matching. It's essentially a gain issue. Your amplifier has too much gain for the strength of the source you have. What I said above is true - a volume pot can attenuate any signal if it's the first thing in the amplifier circuit. However, with a strong source, that may mean the attenuation desired is in the channel mis-match range of the pot. If you were a DIY-er, you could do one of two things: 1) attenuate the signal with a resistor(s) ahead of the pot, throwing the signal into the higher range of the pot, where channel mis-matching is minimal or 2) buy a better pot and install it, instead. I sell kits for the Starving Student Millett Hybrid. We give specific instructions on adding resistors so as to push the attenuation of the signal into the "good" range of a somewhat inexpensive volume pot (as in Starving Student costs). Otherwise as I said, the other guys have given you great advice.

the link was about the DAC, not the amp section. I remember that because it was driving me crazy at the time not to understand why I sometimes was hearing crappy clipped like sound. my own issue turned out to be intersample clipping, which can be seen as another reason not to stay at 100% digital gain on the computer.
but I've since talked to a few engineers and DIYers who seemed to agree that 0dB could show more distortions. although I never understood what they were saying as a possible cause, so maybe it's also a matter of intersample clipping? IDK. nowadays I can't live without replaygain anyway so I've given up on actual bit perfect concerns.


@RadioHamster by default, analog attenuation is the best choice, which is why we tend to leave the heavy work to the amplifier's pot. but as digital attenuation isn't that bad and channel mismatch from some knobs can be fairly significant, I don't think it's a bad idea to mix both options. you should always try to avoid attenuating digitally by 50dB or some crazy value like that, but to fine tune a little and maybe to get out of some massive imbalance area on the knob, IMO digital gain is great. just set the output to 24 or 32bit(whatever your DAC can handle), to further reduce the impact of digital gain and you're good. changing the gain on the amp like @tomb suggests is best when possible, so you end up with the usually nominal area for balance on the knob(way up), but not so high that it doesn't get loud enough for your use. which is why IMO people should pay a lot more attention to the gain values of their amps when purchasing a new one(unless you have like 10 headphones).
about checking for a golden area on the knob while reading the output, it could be good if you have the wiring ability to invert the channels on your input, just to check that the input of your soundcard or cellphone doesn't have its own imbalance. and if it does, that way you can quantify it.
it doesn't take that long just a few very annoying minutes handling the knob like you're trying to disarm a bomb. I usually get a RTA of sort, or even did it with a voltmeter switching back and forth between channels(now that's a real bother. it took me at least 2.7 eternities one afternoon). and before getting mad because one side gives 0.5V and the other side gives 0.51V, remember to look how much that means in dB.

for the headphone itself showing imbalance, the issue usually is that it's rarely a global imbalance. often the biggest mismatch can be at a few local frequencies. it can be dealt with using per channel EQ(just have to check the filter type to avoid creating even more subjective imbalance with time delay but it's obvious when you get that wrong so no big deal). the real difficulty is to properly measure the headphone. that can be much trickier than we imagine and small errors in placement relatively to a microphone can result a several dB changed in the measurement.
there is no perfect world out there, only dissatisfied paranoiacs like myself.

 

[ProtegeManiac]

2. Why do they use ADC in the first place? Can't they just use the digital pot directly on the input which comes from from a DAC?

Like I said in my prior post, it's there to interpret the movement of an analogue knob into something that a digital potentiometer will understand. Normally the control for digital volume are + and - buttons. What Meier did, basically, was to have that ADC transmit counterclockwise motion as - button movement and clockwise motion as + movement. This isn't all that simple either because the most common application of this are more along the lines of the stick controls on console/PC control pads, which at least initially practically worked the same way as up/forward, down/back, left/strafe left, right/strafe right buttons on the left side of the control pad, but with zero attenuation like for example how the "throttle" controls on a Gundam cockpit as implied by the shows transmit the speed of how the throttle is moved (and four buttons on the controller as in SEED and Destiny) so that the arm either reaches out with its palm so a pilot can walk out to the hand and bury a body in a lake or grab another mobile suit's hand, or a quick thrust forward with all buttons pressed will punch another mobile suit in the face or stab it with a beam saber deployed on the wrist.

In short, if it was programmed like a first gen Dual Shock control pad, it will work like a button: twist left/right, hit hard stop, volume goes down/up until you let go.

1. If an ADC is used for the knob, then it later uses DAC to convert it back to analog right? Wouldn't that decrease signal purity and increase latency from going through an extra cycle of conversion?

The Meier Jazz and Classic are amplifiers only, they do not have any DACs.

The specifically programmed ADC is only there to interpret knob movement for the digital potentiometer so it doesn't work like a button where you twist to one side, hard stop, volume progressively gets louder or softer until you release, ie, like mashing + and - buttons. The knob works just like a normal analogue knob without the imbalance of analogue potentiometers. It does not handle the audio signal directly, it just interprets an input into the potentiometer, same way that if you were to lose a hand the prosthetic robot hand needs to have a chip and properly programmed firmware in it that will interpret the electric pulses from your brain down to your nerves so that it doesn't end up either taking several seconds trying to just get a proper grip on a soda can or crush each soda can (and anything else) you try to hold with that prosthetic hand, so imagine Luke Skywalker ending up killing Kylo Ren when he was a baby when he goes, "hi, I'm Uncle Luke! awwwwww!" and the hand works like a Playstation controller so he accidentally drops him or mutilates him.

3. To my knowledge, digital volume control always introduces loss of signal purity, unless it's very high resolution and hence not very noticeable. So doesn't using a digital pot again, compromise the fidelity of the sound?

That's software volume control that messes with the bit length instead of just the digital gain level of a digital audio file, which is not the same as a digital potentiometer that just has a digital control and is actually controlling an analogue signal.

4. I'm actually using modi2U. Why do you suggest replacing DACs with less than 2V output voltage?

That's already at 1.5V. Any lower and you'd be using the line out on a portable DAC-HPamp or some DAPs.

Given the low output voltage of your DAC, what you need is lower gain on the amp.

 

[RadioHamster]

the link was about the DAC, not the amp section. I remember that because it was driving me crazy at the time not to understand why I sometimes was hearing crappy clipped like sound. my own issue turned out to be intersample clipping, which can be seen as another reason not to stay at 100% digital gain on the computer.
but I've since talked to a few engineers and DIYers who seemed to agree that 0dB could show more distortions. although I never understood what they were saying as a possible cause, so maybe it's also a matter of intersample clipping? IDK. nowadays I can't live without replaygain anyway so I've given up on actual bit perfect concerns.


@RadioHamster by default, analog attenuation is the best choice, which is why we tend to leave the heavy work to the amplifier's pot. but as digital attenuation isn't that bad and channel mismatch from some knobs can be fairly significant, I don't think it's a bad idea to mix both options. you should always try to avoid attenuating digitally by 50dB or some crazy value like that, but to fine tune a little and maybe to get out of some massive imbalance area on the knob, IMO digital gain is great. just set the output to 24 or 32bit(whatever your DAC can handle), to further reduce the impact of digital gain and you're good. changing the gain on the amp like @tomb suggests is best when possible, so you end up with the usually nominal area for balance on the knob(way up), but not so high that it doesn't get loud enough for your use. which is why IMO people should pay a lot more attention to the gain values of their amps when purchasing a new one(unless you have like 10 headphones).
about checking for a golden area on the knob while reading the output, it could be good if you have the wiring ability to invert the channels on your input, just to check that the input of your soundcard or cellphone doesn't have its own imbalance. and if it does, that way you can quantify it.
it doesn't take that long just a few very annoying minutes handling the knob like you're trying to disarm a bomb. I usually get a RTA of sort, or even did it with a voltmeter switching back and forth between channels(now that's a real bother. it took me at least 2.7 eternities one afternoon). and before getting mad because one side gives 0.5V and the other side gives 0.51V, remember to look how much that means in dB.

for the headphone itself showing imbalance, the issue usually is that it's rarely a global imbalance. often the biggest mismatch can be at a few local frequencies. it can be dealt with using per channel EQ(just have to check the filter type to avoid creating even more subjective imbalance with time delay but it's obvious when you get that wrong so no big deal). the real difficulty is to properly measure the headphone. that can be much trickier than we imagine and small errors in placement relatively to a microphone can result a several dB changed in the measurement.
there is no perfect world out there, only dissatisfied paranoiacs like myself.

Thanks for mentioning intersample clipping, that sounds like a legitimate reason to not use 100% volume during this era of loudness war. Although I've read that some audiophile devices reserve headroom for this issue, you never know until you actually measure it yourself

Given that digital attenuation gets rid of the channel imbalance problem, it does sound nice to mix both attenuation methods as well. I would love to measure both the channels and try to perfectly match them with EQ, but it seems to be another challenge that I'd have to take my time to learn on this forum. I'm hoping to try it out when I get the budget to take the DIY route and get some nice measurement equipment.

Indeed there is no perfect world, and all we can do is force ourselves to forget that the sound we're hearing isn't perfect

 

[RadioHamster]

Like I said in my prior post, it's there to interpret the movement of an analogue knob into something that a digital potentiometer will understand. Normally the control for digital volume are + and - buttons. What Meier did, basically, was to have that ADC transmit counterclockwise motion as - button movement and clockwise motion as + movement. This isn't all that simple either because the most common application of this are more along the lines of the stick controls on console/PC control pads, which at least initially practically worked the same way as up/forward, down/back, left/strafe left, right/strafe right buttons on the left side of the control pad, but with zero attenuation like for example how the "throttle" controls on a Gundam cockpit as implied by the shows transmit the speed of how the throttle is moved (and four buttons on the controller as in SEED and Destiny) so that the arm either reaches out with its palm so a pilot can walk out to the hand and bury a body in a lake or grab another mobile suit's hand, or a quick thrust forward with all buttons pressed will punch another mobile suit in the face or stab it with a beam saber deployed on the wrist.

In short, if it was programmed like a first gen Dual Shock control pad, it will work like a button: twist left/right, hit hard stop, volume goes down/up until you let go.




The Meier Jazz and Classic are amplifiers only, they do not have any DACs.

The specifically programmed ADC is only there to interpret knob movement for the digital potentiometer so it doesn't work like a button where you twist to one side, hard stop, volume progressively gets louder or softer until you release, ie, like mashing + and - buttons. The knob works just like a normal analogue knob without the imbalance of analogue potentiometers. It does not handle the audio signal directly, it just interprets an input into the potentiometer, same way that if you were to lose a hand the prosthetic robot hand needs to have a chip and properly programmed firmware in it that will interpret the electric pulses from your brain down to your nerves so that it doesn't end up either taking several seconds trying to just get a proper grip on a soda can or crush each soda can (and anything else) you try to hold with that prosthetic hand, so imagine Luke Skywalker ending up killing Kylo Ren when he was a baby when he goes, "hi, I'm Uncle Luke! awwwwww!" and the hand works like a Playstation controller so he accidentally drops him or mutilates him.




That's software volume control that messes with the bit length instead of just the digital gain level of a digital audio file, which is not the same as a digital potentiometer that just has a digital control and is actually controlling an analogue signal.




That's already at 1.5V. Any lower and you'd be using the line out on a portable DAC-HPamp or some DAPs.

Given the low output voltage of your DAC, what you need is lower gain on the amp.

Sorry, I feel so silly that I asked those questions without even knowing what a digital pot is. Little did I know, I was thinking it accepts digital input, but indeed the digital part is only used for changing steps from the mechanical input of the user. Now I get why the ADC was used, and I think that's pretty clever . Though I wish I could clearly picture your Gundam analogy, I can only imagine the outer shell of a generic Gundam for now I watched Gundam 00 long time ago by my friend's recommendation, and I thought it was decent. I'll probably give seed and destiny a try when I get the time

I still don't really understand question 4 though, how does lower-than-optimal? voltage on a DAC make lower gain on an amp better?

Thanks for your reply again!