So I am trying to build my first real DIY project and of course I have a bunch of questions popping up as the work progresses.
This is the project so far:
It started when I came across Alexw88's website and his blanced pre amp:
DIY audio balanced preamp
I began copying his design, but on one huge PCB instead of multiple PCBs with cables connecting them. This was purely estethic/visual, I just think it looks better.
I have since scraped the idea of one large PCB and have sourced the exact same input selector kit, and I also the bought the two PCBs made by Alexw88, the actual line stage (I think that's the correct term, otherwise, the green PCBs in his finnished build I linked to above)
Since I want to have a subwoofer (or two) I have found a summing circuit on this forum that a Danish guy posted (I am at work now but will post the schematics of it when I get home). My plan is to make two small identical PCBs for this SUB-out so that I can have two SUB-outputs that outputs exactly the same thing.
I thought I could just use the filter/crossover adjustment on the active subwoofer(s) to dial in what frequency the subwoofers should play. Fortunately I realized that this isn't a good idea.
This brings me to the question that made me start this thread (finally 😛 )
I want either one rotary switch/encoder/potentiometer that controls what frequencies the outputs to the speakers receives and what frequencies the SUB-outputs receives.
OR
Two rotary switches/encoders/potentiometers, one that sets the cutoff frequency for the regular speaker outputs and one that does the same for the SUB-outputs.
Either way, I need a button that lets me bypass this LP/HP filter/crossover (If I end up using two dials, one for LP and one for HP, then I need to be able to bypass them individually/seperatly)
When I get home I will put together a couple of pictures, showing exactly how I intend to wire everything and schematics etc.
I really want as much input as possible to make sure I don't screw up everything and end up with a pre amp that doesn't work or even worse, damage components in the pre amp and/or things connected to it.
This is my first DIY project and I am not an electronics engineer.
That being said, I have good imagination, strong will, know my way around electronics (soldering, even SMD), etc.
So either wait until I post the pictures etc tonight or have at it right now, give me feedback, how would you do this? What would you change?
Don't hold back, but please be constructive. I might have done everything completely wrong, but I am not an idiot 🙂 I see life as a learning experience and I'm well aware that I'm a complete novice when it comes to DIY audio gear
I'll update this thread as I go along, like a build thread.
Thanks!
/Daniel S
This is the project so far:
It started when I came across Alexw88's website and his blanced pre amp:
DIY audio balanced preamp
I began copying his design, but on one huge PCB instead of multiple PCBs with cables connecting them. This was purely estethic/visual, I just think it looks better.
I have since scraped the idea of one large PCB and have sourced the exact same input selector kit, and I also the bought the two PCBs made by Alexw88, the actual line stage (I think that's the correct term, otherwise, the green PCBs in his finnished build I linked to above)
Since I want to have a subwoofer (or two) I have found a summing circuit on this forum that a Danish guy posted (I am at work now but will post the schematics of it when I get home). My plan is to make two small identical PCBs for this SUB-out so that I can have two SUB-outputs that outputs exactly the same thing.
I thought I could just use the filter/crossover adjustment on the active subwoofer(s) to dial in what frequency the subwoofers should play. Fortunately I realized that this isn't a good idea.
This brings me to the question that made me start this thread (finally 😛 )
I want either one rotary switch/encoder/potentiometer that controls what frequencies the outputs to the speakers receives and what frequencies the SUB-outputs receives.
OR
Two rotary switches/encoders/potentiometers, one that sets the cutoff frequency for the regular speaker outputs and one that does the same for the SUB-outputs.
Either way, I need a button that lets me bypass this LP/HP filter/crossover (If I end up using two dials, one for LP and one for HP, then I need to be able to bypass them individually/seperatly)
When I get home I will put together a couple of pictures, showing exactly how I intend to wire everything and schematics etc.
I really want as much input as possible to make sure I don't screw up everything and end up with a pre amp that doesn't work or even worse, damage components in the pre amp and/or things connected to it.
This is my first DIY project and I am not an electronics engineer.
That being said, I have good imagination, strong will, know my way around electronics (soldering, even SMD), etc.
So either wait until I post the pictures etc tonight or have at it right now, give me feedback, how would you do this? What would you change?
Don't hold back, but please be constructive. I might have done everything completely wrong, but I am not an idiot 🙂 I see life as a learning experience and I'm well aware that I'm a complete novice when it comes to DIY audio gear
I'll update this thread as I go along, like a build thread.
Thanks!
/Daniel S
So, I promised I would upload a simple drawing/schematics of what I am trying to do.
In this drawing, I have not showed the input selector nor power supply, that is all done already.
So, I would like to get input from fellow forum members. Especially regarding the LP/HP crossover since I really don't know how to do that.
I hope I get some feedback at least. Even if it might not be such an interesting project for all of you experts that are on this site.
Thanks in advance.
/Daniel S
In this drawing, I have not showed the input selector nor power supply, that is all done already.
So, I would like to get input from fellow forum members. Especially regarding the LP/HP crossover since I really don't know how to do that.
I hope I get some feedback at least. Even if it might not be such an interesting project for all of you experts that are on this site.
Thanks in advance.
/Daniel S
Attachments
What are the gain limits for changes in R6?
What are the gain limits for changes in R13?
When combined, what is the total gain range without resorting to the switched attenuator?
Prompting the final question:
If the switched attenuators and the following Buffers were omitted, could sufficient gain range be implemented with R6 alone?
You are using a summing circuit for the sub outputs. The output from the 5534, running on +-15Vdc, is the sum of the hot outputs from the balanced stereo feeds. That may overload the 5534s. Reducing the 4k02 Rfeedback will reduce the 5534 output if this becomes a problem. Turning down the sub gain cannot prevent clipping in the summer circuit.
What are the gain limits for changes in R13?
When combined, what is the total gain range without resorting to the switched attenuator?
Prompting the final question:
If the switched attenuators and the following Buffers were omitted, could sufficient gain range be implemented with R6 alone?
You are using a summing circuit for the sub outputs. The output from the 5534, running on +-15Vdc, is the sum of the hot outputs from the balanced stereo feeds. That may overload the 5534s. Reducing the 4k02 Rfeedback will reduce the 5534 output if this becomes a problem. Turning down the sub gain cannot prevent clipping in the summer circuit.
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Hi AndrewT.
These are great questions. Unfortunately, I have no clue 🙁
I was planning on using Alps pot with four "gangs?" for the volume attenuator. Is that not going to work?
I just thought that since this seemed to work for Alexw88, it can't sound bad. I haven't worried about gain etc. Frankly, I need to do some reading on the subject.
Ok so you think there might be a problem with the summing circuit? Could you give me any alternative that you think will work much better with regards to what I am trying to do?
Thanks for your input. I'll try to read up on gain, what is considered to much, how to calculate, etc.
These are great questions. Unfortunately, I have no clue 🙁
I was planning on using Alps pot with four "gangs?" for the volume attenuator. Is that not going to work?
I just thought that since this seemed to work for Alexw88, it can't sound bad. I haven't worried about gain etc. Frankly, I need to do some reading on the subject.
Ok so you think there might be a problem with the summing circuit? Could you give me any alternative that you think will work much better with regards to what I am trying to do?
Thanks for your input. I'll try to read up on gain, what is considered to much, how to calculate, etc.
if you reduce the 4k02 (Rfeedback) to half that value you reduce the gain through the diff buffer by half (-6dB).
That is probably all that is needed to ensure the 5534 does not overload on high signals. But you may find the 5534 has sufficient overhead that it won't overload on sensible signal levels.
That is probably all that is needed to ensure the 5534 does not overload on high signals. But you may find the 5534 has sufficient overhead that it won't overload on sensible signal levels.
If I have understood it correctly, the total gain from the pre amp stages is 4. I still don't know what the gain limits are or how to calculate them 🙂
There's probably a better summing circuit that I could use so if anyone can point one out to me that would be great.
AndrewT, any ideas on what could be done with the HP/LP filter? Anyone? I can't seem to find a crossover solution that can do what I want it to do.
There's probably a better summing circuit that I could use so if anyone can point one out to me that would be great.
AndrewT, any ideas on what could be done with the HP/LP filter? Anyone? I can't seem to find a crossover solution that can do what I want it to do.
Nothing wrong with your bass only summer circuit.
The output is Rfeedback/Rsource * Vleft + Rfeedback/Rsource * Vright = VLeft+Vright * Rfeedback/Rsource.
Feed in a CDP signal of 2.2Vac and the peak of that signal is ~3.1Vp
Now send those Vleft & Vright to the summer and it will output 6.2Vpk
That is a high signal in comparison to the +-15Vdc supply rails.
If you have added some bass boost earlier in the system then you could have even higher outputs from the summer.
That is why I'm suggesting that you may need to look at reducing the Rfeedback to half value to get the nominal output of the summer to a similar level to what the stereo opamps are seeing.
You can use two 5534 and a supply of +-20Vdc for more bass channel headroom.
The output is Rfeedback/Rsource * Vleft + Rfeedback/Rsource * Vright = VLeft+Vright * Rfeedback/Rsource.
Feed in a CDP signal of 2.2Vac and the peak of that signal is ~3.1Vp
Now send those Vleft & Vright to the summer and it will output 6.2Vpk
That is a high signal in comparison to the +-15Vdc supply rails.
If you have added some bass boost earlier in the system then you could have even higher outputs from the summer.
That is why I'm suggesting that you may need to look at reducing the Rfeedback to half value to get the nominal output of the summer to a similar level to what the stereo opamps are seeing.
You can use two 5534 and a supply of +-20Vdc for more bass channel headroom.
Ok, thanks AndrewT. I will take your advice and reducing the Rfeedback to half. I have been told that the feed signal from a typical source, like a CD player is around 0.75v at 0db.
But you are talking about 2.2V?
Do you think the Alps 10k pot will work good as a volume attenuator? Alexw88 is using a stepped attenuator in his original design that I am trying to emulate.
But you are talking about 2.2V?
Do you think the Alps 10k pot will work good as a volume attenuator? Alexw88 is using a stepped attenuator in his original design that I am trying to emulate.
0dBfs is usually between 2.0Vac and 2.3Vac for CDP and similar.
If the average signal is measured at 0.75Vac then maybe the average signal is not at 0dBfs and is more likely at ~ -9dB ref maximum.
Some/many recordings have a typical average level that is only 6 to 12dB below maximum. Good dynamic recordings have typical average levels of 20 to 30dB below maximum.
Have you looked at what the middle resistor (R6) does for stage gain yet?
I have not used it, it may have a noise consequence, but at least simulate or calculate it's effect on gain.
If the average signal is measured at 0.75Vac then maybe the average signal is not at 0dBfs and is more likely at ~ -9dB ref maximum.
Some/many recordings have a typical average level that is only 6 to 12dB below maximum. Good dynamic recordings have typical average levels of 20 to 30dB below maximum.
Have you looked at what the middle resistor (R6) does for stage gain yet?
I have not used it, it may have a noise consequence, but at least simulate or calculate it's effect on gain.
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