Hi Folks,
Although I have soldered well over a hundred boards in my life (and got them all to work) I am totally clueless when it comes down to electronics design or how any of those boards actually work. As a result I have a hard time understanding a lot of the stuff in various threads here. But I can put components and modules together and make stuff work (eventually).
Hoping to learn and build a better, custom system and get better sound without breaking the bank.
Mike aka ZK
Although I have soldered well over a hundred boards in my life (and got them all to work) I am totally clueless when it comes down to electronics design or how any of those boards actually work. As a result I have a hard time understanding a lot of the stuff in various threads here. But I can put components and modules together and make stuff work (eventually).
Hoping to learn and build a better, custom system and get better sound without breaking the bank.
Mike aka ZK
Welcome aboard, I’m in a similar boat. I’ve built three power amps, one pre-amp and one headphone amp. I wouldn’t have been able to do it without the great people on this site.
Thanks for the welcome guys. I guess Most of the technical discussions here I don't understand one bit of. Could not design a circuit to safe my life.
But I can solder and follow instructions.
One of the problems is finding out what one needs before you can actually look for instructions or PCBs.
But I can solder and follow instructions.
One of the problems is finding out what one needs before you can actually look for instructions or PCBs.
Hi Rad-Cap!
I kind of feel lost in the wilderness. I am looking to build/I need balanced analog volume control for a RPI based streamer DAC but I still want to be able to control the volume from the app on a tablet. So far I am even in doubt which forum to post in to ask for directions....
I kind of feel lost in the wilderness. I am looking to build/I need balanced analog volume control for a RPI based streamer DAC but I still want to be able to control the volume from the app on a tablet. So far I am even in doubt which forum to post in to ask for directions....
Beginners should buy a used copy of "The Art of Electronics" second edition (not the third edition).
This will fill in your missing foundation at a suitable level.
This will fill in your missing foundation at a suitable level.
Hi Z.K.,
Simple. Old school. A motorized volume control.
Alternatively, a digital volume control IC using an optical encoder (rotary knob type control). Not what you are asking for, but this option will probably be presented to you. Some of these perform better than some rotary controls (classic volume control).
It doesn't matter what your source is. You want to control an analogue pot. I would think that "Analogue Source" might be your best bet.
Simple. Old school. A motorized volume control.
Alternatively, a digital volume control IC using an optical encoder (rotary knob type control). Not what you are asking for, but this option will probably be presented to you. Some of these perform better than some rotary controls (classic volume control).
It doesn't matter what your source is. You want to control an analogue pot. I would think that "Analogue Source" might be your best bet.
Hello Anatech,
Thanks. Maybe I should have been a bit more specific. When I say analog volume control I mean volume control without resampling.
Every RPI DAC HAT I have come across, including the ones from Ian Canada, do volume control in the digital domain. Unless I am misinterpreting what I found, but Ian's manual was pretty explicit about this as well as the need for a preamp to improve quality by means of analog volume control.
The idea of volume control by means of resampling goes completely against the audiophile grain if you ask me.
It took me a while before I realized what was meant by I/V stage but once I got that concept, there did not seem to be any high-end DIY solutions available (or any for that matter)
Motorized volume control or digital volume control IC, as long as it does the least damage to the signal/sound quality and there is no resampling, it is fine by me.
BUT how do I get that to work with whatever software is running on the RPI (Volumio, Moode etc) without having to learn to how to code/write drivers etc?
I have a pile of balanced amplifiers which is why I am looking for a balanced solution. I don't need source selection. A balanced RPI DAC HAT will be the only source
As long as there is no smd involved I believe I can successfully solder a board together. I have well above average computer skills, but coding isn't one of them.
I found a balanced preamp board here somewhere but not sure if the design is either suitable or sound quality wise sufficient.
Open to suggestions 🙂
ZK
Thanks. Maybe I should have been a bit more specific. When I say analog volume control I mean volume control without resampling.
Every RPI DAC HAT I have come across, including the ones from Ian Canada, do volume control in the digital domain. Unless I am misinterpreting what I found, but Ian's manual was pretty explicit about this as well as the need for a preamp to improve quality by means of analog volume control.
The idea of volume control by means of resampling goes completely against the audiophile grain if you ask me.
It took me a while before I realized what was meant by I/V stage but once I got that concept, there did not seem to be any high-end DIY solutions available (or any for that matter)
Motorized volume control or digital volume control IC, as long as it does the least damage to the signal/sound quality and there is no resampling, it is fine by me.
BUT how do I get that to work with whatever software is running on the RPI (Volumio, Moode etc) without having to learn to how to code/write drivers etc?
I have a pile of balanced amplifiers which is why I am looking for a balanced solution. I don't need source selection. A balanced RPI DAC HAT will be the only source
As long as there is no smd involved I believe I can successfully solder a board together. I have well above average computer skills, but coding isn't one of them.
I found a balanced preamp board here somewhere but not sure if the design is either suitable or sound quality wise sufficient.
Open to suggestions 🙂
ZK
Hi ZK,
Well, you absolutely do want any DAC (digital volume control) to be oversampled. Same for CD DACs. This is one area I am extremely familiar with. A straight variable resistor (manual volume control) can be had with a motor on it as well. It has a clutch for the end stops.
Non OS DACS must use a 7th or 9th order low pass filter. Those do not sound good, and the reason we went to oversampling is so we can use a filter with a more gentle slope far above your hearing. If you exclude the filter as so many poorly informed people do, the sampling frequency goes out the front door into your equipment. Typically high amplitude 44.1 KHz. You really do not want to do that.
Balanced is not a benefit. Most equipment is single path inside for very good reasons. Most equipment with balanced has the single ended to balanced single converter hung on the ends. Balanced circuitry is 1.414 x noisier than single ended. The only one reason we use balanced is to solve specific problems. Example, microphone running distance as in recording studios, live performances and the like. Balanced is a lower impedance standard. At normal signal levels, the only reason you would run balanced is for a long run. You shouldn't be running signals that far.
Manual control sections do not track well. A quad control has the same issues x 2. That means our signal amplitudes on each phase will not be equal and you don't get the CMRR benefit people like to talk about. You can with a transformer, but you run into distortion issues and phase shifts. So you pick the least damaging path, balanced throughout is not it.
This is very simplified, but the main points are there.
I would highly recommend you go with a non-balanced circuit. I'm sure there is probably a project with motor drive circuits you can copy. You can also buy premade motorized controls with relay switches from China. Some are pretty good. You get the remote and everything.
The electronics field is vast. It becomes easier the more you learn. Use common sense, logic and the rules of physics. Don't listen to "high end" audio folks as they are typically badly misinformed. I've been doing this over 50 years and still learn, professionally for nearly 50 years now. It's okay to be wrong, keep an open mind and learn. There is so much to know, it is scary at first. I still remember looking at a simple circuit in "Popular Electronics" thinking, I'll never understand it. You can't read it once and understand. So as rayma said, "The Art of Electronics" is an excellent text, but can be pretty advanced. Learn how the parts work and behave. This is the bedrock. Then you build on that understanding. It takes years, but it is extremely rewarding.
Well, you absolutely do want any DAC (digital volume control) to be oversampled. Same for CD DACs. This is one area I am extremely familiar with. A straight variable resistor (manual volume control) can be had with a motor on it as well. It has a clutch for the end stops.
Non OS DACS must use a 7th or 9th order low pass filter. Those do not sound good, and the reason we went to oversampling is so we can use a filter with a more gentle slope far above your hearing. If you exclude the filter as so many poorly informed people do, the sampling frequency goes out the front door into your equipment. Typically high amplitude 44.1 KHz. You really do not want to do that.
Balanced is not a benefit. Most equipment is single path inside for very good reasons. Most equipment with balanced has the single ended to balanced single converter hung on the ends. Balanced circuitry is 1.414 x noisier than single ended. The only one reason we use balanced is to solve specific problems. Example, microphone running distance as in recording studios, live performances and the like. Balanced is a lower impedance standard. At normal signal levels, the only reason you would run balanced is for a long run. You shouldn't be running signals that far.
Manual control sections do not track well. A quad control has the same issues x 2. That means our signal amplitudes on each phase will not be equal and you don't get the CMRR benefit people like to talk about. You can with a transformer, but you run into distortion issues and phase shifts. So you pick the least damaging path, balanced throughout is not it.
This is very simplified, but the main points are there.
I would highly recommend you go with a non-balanced circuit. I'm sure there is probably a project with motor drive circuits you can copy. You can also buy premade motorized controls with relay switches from China. Some are pretty good. You get the remote and everything.
The electronics field is vast. It becomes easier the more you learn. Use common sense, logic and the rules of physics. Don't listen to "high end" audio folks as they are typically badly misinformed. I've been doing this over 50 years and still learn, professionally for nearly 50 years now. It's okay to be wrong, keep an open mind and learn. There is so much to know, it is scary at first. I still remember looking at a simple circuit in "Popular Electronics" thinking, I'll never understand it. You can't read it once and understand. So as rayma said, "The Art of Electronics" is an excellent text, but can be pretty advanced. Learn how the parts work and behave. This is the bedrock. Then you build on that understanding. It takes years, but it is extremely rewarding.
Hi Rayma,
I know what resistors, capacitors and LEDs are, what AC and DC is. I can solder as long as no smd is involved. Did not fail physics but I am more mechanically inclined than electronically.
Unfortunately I am also a perfectionist. Don't mind putting work in but taking speakers as an example, I would be more inclined to figure out how to for example to make a mold (maybe 3D printed) to cast a speaker cabinet out of epoxy resin and quartz than to design a cross over circuit.
By the time I get to the level where the result of a circuit design of my own, starting from almost below zero knowledge, would meet my rather perfectionist standards, I am afraid I might need a hearing aid by then....
When it comes down to speakers, if I was to embark on such a project, I would want to construct something that would play in the same league as e.g B&W 802 D2/D3 but without spending anywhere near the amount of what those speakers cost..
Translated into electronics, I would imagine it would probably take 10-15 years before, starting from zero knowledge, I could design anything near that level of stuff. While DIY is fun, we all have a limited lifespan and I dont think I have that amount of time (or that many years of determination).
BTW what is wrong with the third edition?
Hello Anatech,
Thanks for your elaborate reply. I read it twice and quite a few things made my head spin.
Regarding Balanced/Unbalanced. I was aware that most get converted to unbalanced first but all I could do was read the manual and my amplifier modules do actually expect a balanced signal. Long runs in my case may not even be that unlikely. (Depending on the definition of long)
Buying a ready made circuit from china would roughly be the last thing I am interested in. I tend to avoid buying made in china products.
Regarding learning circuit design, please read my comments in my reply to Rayma.
I bought various boards from Ian. The idea of swapping boards, experimenting with various clocks, I/V stages, upgrading power supplies and messing around with a RPI seems more attractive and fun to me than the very daunting task of learning how to design a preamp from scratch.
In Ian's DAC HAT manual I found this:
"Use a decent pre-amplifier with balanced XLR inputs for the volume control
The ES9038Q2MP Dual Mono II DAC has very good digital volume control. However, a decent pre-amplifier with
balanced XLR inputs would be still highly recommended to keep the best sound quality. You must set the controller
volume to 0dB in this case.
1. Can make good use of the higher quality balanced outputs
2. Keep the DAC running at full scale
3. Without break the bit-perfect"
Particularly #2 and #3 is why I am looking for what I asked for
Streamer [RPI, Linear power supply, DAC HAT, I/V stage, preamp/volume control] => Power Amp
2 boxes total.
You said: "Manual control sections do not track well."
and "Alternatively, a digital volume control IC using an optical encoder (rotary knob type control)."
The second remark is a solution for the problem in the first remark?
If I understand the concept of that digital volume control correctly, it would be a solution because it does not mess with #2 and #3
I found this:
https://www.diyaudio.com/community/threads/balanced-volume-controller-line-stage.325268/
and I was wondering if the non-smd digital version (PGA2310) would be an appropriate high quality solution?
But the main issue there remains, how can I control the volume from the streamer software running on the RPI without having to program drivers etc? I can edit config files etc but coding is a different story.
Thanks,
ZK
Thanks for your elaborate reply. I read it twice and quite a few things made my head spin.
Regarding Balanced/Unbalanced. I was aware that most get converted to unbalanced first but all I could do was read the manual and my amplifier modules do actually expect a balanced signal. Long runs in my case may not even be that unlikely. (Depending on the definition of long)
Buying a ready made circuit from china would roughly be the last thing I am interested in. I tend to avoid buying made in china products.
Regarding learning circuit design, please read my comments in my reply to Rayma.
I bought various boards from Ian. The idea of swapping boards, experimenting with various clocks, I/V stages, upgrading power supplies and messing around with a RPI seems more attractive and fun to me than the very daunting task of learning how to design a preamp from scratch.
In Ian's DAC HAT manual I found this:
"Use a decent pre-amplifier with balanced XLR inputs for the volume control
The ES9038Q2MP Dual Mono II DAC has very good digital volume control. However, a decent pre-amplifier with
balanced XLR inputs would be still highly recommended to keep the best sound quality. You must set the controller
volume to 0dB in this case.
1. Can make good use of the higher quality balanced outputs
2. Keep the DAC running at full scale
3. Without break the bit-perfect"
Particularly #2 and #3 is why I am looking for what I asked for
Streamer [RPI, Linear power supply, DAC HAT, I/V stage, preamp/volume control] => Power Amp
2 boxes total.
You said: "Manual control sections do not track well."
and "Alternatively, a digital volume control IC using an optical encoder (rotary knob type control)."
The second remark is a solution for the problem in the first remark?
If I understand the concept of that digital volume control correctly, it would be a solution because it does not mess with #2 and #3
I found this:
https://www.diyaudio.com/community/threads/balanced-volume-controller-line-stage.325268/
and I was wondering if the non-smd digital version (PGA2310) would be an appropriate high quality solution?
But the main issue there remains, how can I control the volume from the streamer software running on the RPI without having to program drivers etc? I can edit config files etc but coding is a different story.
Thanks,
ZK
"what is wrong with the third edition?"
The third edition is great, but is too advanced for a beginner, and is more of a successor than just a new edition.
The second edition will be just right. If you ever outgrow the second, give it to a friend and get the third.
The third edition is great, but is too advanced for a beginner, and is more of a successor than just a new edition.
The second edition will be just right. If you ever outgrow the second, give it to a friend and get the third.
Hi ZK,
rayma is dead on. Volume three is very thick and chock full of information. Get through Vol two, that's a lot.
1.) Avoid balanced in unless you really don't have a choice. In that case, they make balanced to single ended adapter ICs that are extremely high quality. Single ended to balanced output ICs are also readily available. I've used them, no problem. I'd have to look up the current part number. Internally you are way further ahead with single ended. One balanced preamp was the Carver Lightstar Direct. Worked great, sounded great. Horribly complicated for it's simple functions - and very expensive to make.
2.) Yes, that's how you do it. Some volume controls reduce the number of bits as the volume drops - a lot.
3.) "bit perfect" can be a myth depending on the context. Streaming services are SIP, no error correction at all. Just like VoIP or cell communication. So if you're streaming, do not expect anything close to bit perfect. However you probably will not notice, that's the other side of things. If your system is connected to a music server via ethernet you can reasonably expect bit perfect compared to the file on the music server. That is the cold hard truth.
All my systems are single ended and they are very good systems.
The PGA2310 is very good. Much more difficult to implement than a "pot". However, some folks here may find it very easy. At least channel balance would be better than a rotary control, I don't like switched resistor controls very much, but they are fine if you don't mind steps in level. To be honest, normal pots are fine if they are good ones.
The thread you linked to is addressing non-issues for you. Don't worry about it.
The system you propose is fine. I would go separate preamp and separate sources. It allows you to keep digital noise at bay, and that is a problem. You can also easily upgrade each part or experiment. Up to you, but multi-box is more flexible and you can get much higher quality. Also your power supplies can be optimized for each circuit with isolation from each.
I am not very strong on programming or getting stuff like that to work. I'm pretty good at troubleshooting and repairing it, I go for analogue design.
rayma is dead on. Volume three is very thick and chock full of information. Get through Vol two, that's a lot.
1.) Avoid balanced in unless you really don't have a choice. In that case, they make balanced to single ended adapter ICs that are extremely high quality. Single ended to balanced output ICs are also readily available. I've used them, no problem. I'd have to look up the current part number. Internally you are way further ahead with single ended. One balanced preamp was the Carver Lightstar Direct. Worked great, sounded great. Horribly complicated for it's simple functions - and very expensive to make.
2.) Yes, that's how you do it. Some volume controls reduce the number of bits as the volume drops - a lot.
3.) "bit perfect" can be a myth depending on the context. Streaming services are SIP, no error correction at all. Just like VoIP or cell communication. So if you're streaming, do not expect anything close to bit perfect. However you probably will not notice, that's the other side of things. If your system is connected to a music server via ethernet you can reasonably expect bit perfect compared to the file on the music server. That is the cold hard truth.
All my systems are single ended and they are very good systems.
The PGA2310 is very good. Much more difficult to implement than a "pot". However, some folks here may find it very easy. At least channel balance would be better than a rotary control, I don't like switched resistor controls very much, but they are fine if you don't mind steps in level. To be honest, normal pots are fine if they are good ones.
The thread you linked to is addressing non-issues for you. Don't worry about it.
The system you propose is fine. I would go separate preamp and separate sources. It allows you to keep digital noise at bay, and that is a problem. You can also easily upgrade each part or experiment. Up to you, but multi-box is more flexible and you can get much higher quality. Also your power supplies can be optimized for each circuit with isolation from each.
I am not very strong on programming or getting stuff like that to work. I'm pretty good at troubleshooting and repairing it, I go for analogue design.
Hi Anatech,
2) thats why I want to avoid it.
3) I think there is a terminology issue. I will be playing FLAC or WAV from an SSD aybe also from the local network but definitely not from the Internet..
This is actually for a second system. My main system will use an external ladder type dac with built in analog volume control. The "streamer" will also be RPI based but it will be entirely based on Ian Canada's various HATs. Isolator, reclocker, linear power supply. There I can experiment with clocks, power supplies etc without the need for designing or understanding circuits.
You said:
"The thread you linked to is addressing non-issues for you. Don't worry about it."
Is that because it is balanced or because it will not do what I need?
2) thats why I want to avoid it.
3) I think there is a terminology issue. I will be playing FLAC or WAV from an SSD aybe also from the local network but definitely not from the Internet..
This is actually for a second system. My main system will use an external ladder type dac with built in analog volume control. The "streamer" will also be RPI based but it will be entirely based on Ian Canada's various HATs. Isolator, reclocker, linear power supply. There I can experiment with clocks, power supplies etc without the need for designing or understanding circuits.
You said:
"The thread you linked to is addressing non-issues for you. Don't worry about it."
Is that because it is balanced or because it will not do what I need?
I better go find myself a copy then. Hopefully I manage to actually understand what is in it.
Hi ZK,
The link is addressing concerns you don't have to an extent they will affect anything. Some folks worry about things that either are not a problem, or you may have more significant problems.
Balanced is not an intelligent signal format to follow in any small system. The negatives outweigh the imagined benefits. I worked in the test and measurements field, the telecom field in addition to audio. In audio I worked in several recording studios, and one major (still existing) one. The reasons to go balanced are:
Understand?
The link is addressing concerns you don't have to an extent they will affect anything. Some folks worry about things that either are not a problem, or you may have more significant problems.
Balanced is not an intelligent signal format to follow in any small system. The negatives outweigh the imagined benefits. I worked in the test and measurements field, the telecom field in addition to audio. In audio I worked in several recording studios, and one major (still existing) one. The reasons to go balanced are:
- Long runs (leading to high cable capacitance)
- Electrically noisy environments, cable shields often have some % of effectiveness. Balanced cables allow you to use the CMRR of the input circuit to cancel most
- Lower impedance also forms a voltage divider with radiated noise to reduce the induced noise level
- Original balanced lines used signal transformers which theoretically have perfect CMRR performance and no added noise. Transformers add distortion and they are not perfect.
- Electronic balanced converters have lower CMRR, better frequency and phase performance and a touch of noise. Better than transformers sometimes.
Understand?
Reclockers.
I have a $30,000 instrument I measure clock performance with (HP 5372A). It is locked to the GPS giving me a maximum error of approx 5 exp -12, so way more accurate than the clocks used in creating the music. However, when I need to characterize a clock I have to disconnect the GPS reference clock. That's because the internal double oven crystal oscillator is more stable in the short term.
So what is the most important in music decoding? Short term stability. Long term doesn't matter at all. The most accurate and stable reference is a crystal oscillator, period. The normal crystals can be down to 10 -6 error, and SC cut more accurate. Ovenized oscillators get down to 10 -9 for example. So what do you need? A normal crystal oscillator, that's all and it is far better than you'd think. Now consider your encoded musics will have embedded all jitter from the source, that's in there as sample interval errors. Reclocking will not change anything. So all this noise about the best clock totally fails to consider the source. On top of that, even normal crystal variations cannot be detected by a human being. I don't care who you are, the human brain isn't capable.
So when considering how to make your system better, you have to consider the entire system and source and that includes what your own body limits are.
Why not stay locked to the GPS signal? Because the disciplining methods makes occasional corrections to the crystal frequency, causing tiny jumps in frequency. You would never hear this, but for testing it shows up like a sore thumb.
I have a $30,000 instrument I measure clock performance with (HP 5372A). It is locked to the GPS giving me a maximum error of approx 5 exp -12, so way more accurate than the clocks used in creating the music. However, when I need to characterize a clock I have to disconnect the GPS reference clock. That's because the internal double oven crystal oscillator is more stable in the short term.
So what is the most important in music decoding? Short term stability. Long term doesn't matter at all. The most accurate and stable reference is a crystal oscillator, period. The normal crystals can be down to 10 -6 error, and SC cut more accurate. Ovenized oscillators get down to 10 -9 for example. So what do you need? A normal crystal oscillator, that's all and it is far better than you'd think. Now consider your encoded musics will have embedded all jitter from the source, that's in there as sample interval errors. Reclocking will not change anything. So all this noise about the best clock totally fails to consider the source. On top of that, even normal crystal variations cannot be detected by a human being. I don't care who you are, the human brain isn't capable.
So when considering how to make your system better, you have to consider the entire system and source and that includes what your own body limits are.
Why not stay locked to the GPS signal? Because the disciplining methods makes occasional corrections to the crystal frequency, causing tiny jumps in frequency. You would never hear this, but for testing it shows up like a sore thumb.
Good morning/afternoon/evening, Z.K:
Pretty much in the same boat as you, Mulburg, Read-cap, and probably others. Can follow schematics a bit but understanding how it all works/what does what/etc. is a headbanger at times (aside from something maybe simple).
Found this online...The Art of Electronics - 2nd ed pdf. So, had at it. (A very) Long read for sure.
Thanks. (and thanks to the University of Wisconsin-Madison for providing the PDF.) 😎
Sincerely,
Kingsley.
You young guys are lucky.
I don't retain stuff off a monitor well. I like a book. Plus it doesn't take batteries and I can relax without emails or ads. lol!
I don't retain stuff off a monitor well. I like a book. Plus it doesn't take batteries and I can relax without emails or ads. lol!