New to the game - recap an amp (Rotel RA-840BX3)
- Solid State
- 1 Replies
Hey all,
I'm really new to the game, but have worked on pinball machines and some other odds and ends over the years, so working on circuit boards isn't outside of my wheelhouse. I currently have a Rotel RA-840BX3 that's on the fritz (has a broken bias potentiometer). As I'm ordering parts anyhow, I tested the readily available (read: huge 50V 10000uf) caps on the board, and one is dead, so I intend to replace those as well.
I have the schematic and service manual, but have never done a recap. I'm wondering a) if there's any particular way other than poring over the schematic to find the caps required, b) if there are any capacitors that are generally replaced and generally left alone, and c) how a guy goes about learning how that schematic hangs together i.e. what is each part of the circuit doing? I can easily test out a given component, but I don't entirely know how to say "oh, these components are the phono stage; this is the power supply" and so on. I've done some Googling and YouTube-ing and whatnot, but feel like I'm on a bit of a fishing expedition and not catching much. I figured as opposed to taking a shotgun approach, I'd try to learn to do it right. Thanks in advance for the help; really appreciate it.
I'm really new to the game, but have worked on pinball machines and some other odds and ends over the years, so working on circuit boards isn't outside of my wheelhouse. I currently have a Rotel RA-840BX3 that's on the fritz (has a broken bias potentiometer). As I'm ordering parts anyhow, I tested the readily available (read: huge 50V 10000uf) caps on the board, and one is dead, so I intend to replace those as well.
I have the schematic and service manual, but have never done a recap. I'm wondering a) if there's any particular way other than poring over the schematic to find the caps required, b) if there are any capacitors that are generally replaced and generally left alone, and c) how a guy goes about learning how that schematic hangs together i.e. what is each part of the circuit doing? I can easily test out a given component, but I don't entirely know how to say "oh, these components are the phono stage; this is the power supply" and so on. I've done some Googling and YouTube-ing and whatnot, but feel like I'm on a bit of a fishing expedition and not catching much. I figured as opposed to taking a shotgun approach, I'd try to learn to do it right. Thanks in advance for the help; really appreciate it.
Audiozen AZ OLED setup kits for Soekris dam1021/1121 (new unused)
As I am slimming out my DIY projects backlog due to the lack of time building all the stuff lying around I will post many undone projects and PCBs I collected over time.
For sale is a complete set of Audiozens AZOled Kit (SK lite). Kit has never been used and was meant to built for a friend who never responded so it was lying around for some time in my stock.
New and unsoldered transformer, protective film still over the oled display.
Asking what I originally paid 130 usd + shipping + PP fees.
For sale is a complete set of Audiozens AZOled Kit (SK lite). Kit has never been used and was meant to built for a friend who never responded so it was lying around for some time in my stock.
New and unsoldered transformer, protective film still over the oled display.
Asking what I originally paid 130 usd + shipping + PP fees.
DS-15 Class-D Modul with much Trouble and used in a lot of Models from Harman//JBL
- Class D
- 30 Replies
Attachments
Help with troubleshooting faint buzzing noise from headphones
- By Gula Melaka
- Headphone Systems
- 12 Replies
Attachments
Any opinions on EMI filtering between SMPS and TPA3255 amplifiers?
- By Audientid14
- Class D
- 12 Replies
This can be bought on Amazon.
https://www.amazon.se/lägesfilter-E..._tag=se&qid=1734382007&s=industrial&sr=1-1037
SMPS - DC in - EMI filter - DC out - tpa3255 module
Any opinions of the sound quality , measurements?
https://www.amazon.se/lägesfilter-E..._tag=se&qid=1734382007&s=industrial&sr=1-1037
SMPS - DC in - EMI filter - DC out - tpa3255 module
Any opinions of the sound quality , measurements?
Output Transformer Interleaving
- Tubes / Valves
- 23 Replies
Hey guys!
Having a bit of trouble settling on a final interleaving method , especially for the primary. Its a 4:5 primary to secondary interleave.
The primary is broken into 4 wound in series , but I'm having a bit of confusion as to how to get the best DCR balance for the B+ as well as longitudinal balance.
I separate the primaries into 4 : P1,P2,P3,P4 . Given P1 is wound first and so on , the length of wire per turn for each is P4>P3>P2>P1 . In my mind it makes intuitive sense that to get the best balance , connecting one end of P1 to P4 would give the best balance as shown in the 2 possible configurations above. Would you guys agree? then again , how do I do this connection ; each winding has 2 possible connection points.
Using the numbers above for each connection . is it best to go 1-2-7-8-3-4-5-6 ?
Its a Push Pull OT running off 4 EL84's. Building for -1db at 20Hz for lower frequency bound. The secondaries are 4,8,16 Ohm taps . Expected output of around 30W.
Thank you so much !
MA Audio HK2000D
Can someone help me identify these output drive IC's (SounDigital EVO X 800.1)
Hello all, I am trying trying to identify the 4 driver IC's on the output card of my SounDigital 800.1. SoundDigital has taken the liberty of sanding the part numbers off to make them unidentifiable 🙄. Thank you in advance!
Attachments
Easy way to bridge an amplifier
- By vilfort
- Analog Line Level
- 6 Replies
A while back there was a discussion on this forum about an easy way to bidge a dual channel amplifier. No need for special phase splitter/turner boards. No need for changes to the internal wiring of the amp to be bridged. If you are like me, you likely have several amps laying around that you would like to bridge - just to hear how it sounds. If so, here is some good news: all you need is a pre-amp with a balanced output. Then create a quick interconnect that takes the in-phase signal (+) and the out of phase signal (-) and feed that into the Right and Left input RCA's. We are assuming that the preamp here has a volume control and that we have 2 identical amps that we can bridge for each channel.
As it turned out, I have a DAC with balanced output and 2 HAFLER 200's that would work for this experiment. All I needed was to create the interconnect:
That's it. Here Pin 1 is common ground. Pin 2 is (+) and Pin 3 is (-). (+) to one channel and (-) to the other. The speaker is connected bridged between the two positive output terminals of the power amp. Essentially the amp is running with each channel amplifying each leg of the balanced signal.
So how does it sound? - Well that is the most surprising: the old HAFLER 200's sound more alive than ever! Trumpets and saxophones have more bite. And I hear details on old familiar tracks that only the best amps manage to bring out. The Haflers already have good specs even by today's standards. Re-capping helps as well. But this inexpensive bridge interconnect brings them to a new level. I cannot explain it. Why? Common mode rejection of some left-over distortion?
If you have any idea why this is happening, please post your thoughts. Or if you just want a fun and easy project for the holidays, this might be what you have been looking for.
Have fun.
As it turned out, I have a DAC with balanced output and 2 HAFLER 200's that would work for this experiment. All I needed was to create the interconnect:
That's it. Here Pin 1 is common ground. Pin 2 is (+) and Pin 3 is (-). (+) to one channel and (-) to the other. The speaker is connected bridged between the two positive output terminals of the power amp. Essentially the amp is running with each channel amplifying each leg of the balanced signal.
So how does it sound? - Well that is the most surprising: the old HAFLER 200's sound more alive than ever! Trumpets and saxophones have more bite. And I hear details on old familiar tracks that only the best amps manage to bring out. The Haflers already have good specs even by today's standards. Re-capping helps as well. But this inexpensive bridge interconnect brings them to a new level. I cannot explain it. Why? Common mode rejection of some left-over distortion?
If you have any idea why this is happening, please post your thoughts. Or if you just want a fun and easy project for the holidays, this might be what you have been looking for.
Have fun.
Crossover design: How low is too low impedance?
I'm wrapping up my first 3-way crossover design and am looking at the impedance curve of the system.
How low is too low impedance?
And what is the significance of phase angle?
Any other issues I should be concerned with on this chart?
How low is too low impedance?
And what is the significance of phase angle?
Any other issues I should be concerned with on this chart?
RCA interconnects - your collection and making your own
- By Miller-8
- Everything Else
- 8 Replies
Attachments
Narrow Subwoofer Driver Location
- Subwoofers
- 49 Replies
I'm looking at making a subwoofer to pair with a set of small full-range speakers, but the catch is that the only place I can really fit it is underneath the TV cabinet, which gives me maximum ~100mm. I've found what appears to be a decent driver, albeit small, in the Peerless SLS-85S25CP04-04. My thoughts were to use a pair, and have modelled them in a ported enclosure around 4.5 - 5L, tuned to 38Hz. The plan would then be to add a LPF somewhere in the region of 80-100Hz in order to cross effectively with the full-range set.
Anyway, that's the background, but the main question I have is around the placement and orientation of the two drivers. Originally, I was thinking facing opposite walls, horizontal in terms of listening position, but this brings the magnets mighty close to the top and bottom walls (using 12mm ply). The important parts seem to be around 68-72mm on the diagrams, and the enclosure would have an internal height of 78mm max. Would this be a major issue for this application?
Second thought was having them facing up, but this introduces the associated problems with upward-firing speakers (and fires directly into the TV cabinet). It gives them a lot more room to breathe, however, though the magnet is then about 15-20mm from the base.
I've attached some images to illustrate the two options (I'm aware I've probably over-engineered the bracing, but I was aiming to have it CNC milled). The grey cylinders are an approximation of the speaker at it's widest points. The black tube is the port.
Option 1 - the "lid" would be inset, so sitting on top of the cross-beams.
Option 2
Any thoughts or obvious errors? Does it even matter if it's going to be booming straight off the TV cabinet above?
Anyway, that's the background, but the main question I have is around the placement and orientation of the two drivers. Originally, I was thinking facing opposite walls, horizontal in terms of listening position, but this brings the magnets mighty close to the top and bottom walls (using 12mm ply). The important parts seem to be around 68-72mm on the diagrams, and the enclosure would have an internal height of 78mm max. Would this be a major issue for this application?
Second thought was having them facing up, but this introduces the associated problems with upward-firing speakers (and fires directly into the TV cabinet). It gives them a lot more room to breathe, however, though the magnet is then about 15-20mm from the base.
I've attached some images to illustrate the two options (I'm aware I've probably over-engineered the bracing, but I was aiming to have it CNC milled). The grey cylinders are an approximation of the speaker at it's widest points. The black tube is the port.
Option 1 - the "lid" would be inset, so sitting on top of the cross-beams.
Option 2
Any thoughts or obvious errors? Does it even matter if it's going to be booming straight off the TV cabinet above?
What to do with this power supply?
Hi All,
I am toying around with the idea of building a class D amplifier using ICE modules. I have (3) power supplies from some old Denon 250 watt class A mono block amps that died many years ago. The power supplies look very beefy, huge tansformer and capacitors.
Would it be worth trying to do something like this with them?
Thanks,
Axis
I am toying around with the idea of building a class D amplifier using ICE modules. I have (3) power supplies from some old Denon 250 watt class A mono block amps that died many years ago. The power supplies look very beefy, huge tansformer and capacitors.
Would it be worth trying to do something like this with them?
Thanks,
Axis
different length speaker cable question
- By henrylrjr
- Full Range
- 41 Replies
My setup requires 15 feet of speaker cable for the right side speaker to amp and the left side needs about 25 feet. I've read that speaker cables should be the same length, but don't know if coiling 10 feet of cable, on itself, in the bottom of a cabinet, might have its own problems. What would be the best, trouble free, setup?
Thanks
Thanks
Dynaudio "ARBITER"/Horch Elektroakustik "POWER STAGE" internal Images wanted
- Solid State
- 0 Replies
Attachments
Identify Power Transformers
- By tonechaser59
- Parts
- 28 Replies
Does anyone know what brand and type of power transformer this is? I have 2 New in the Box and I don't have any idea what
they are? Can you please help me identify a power transformer? The code is: 007018-6066709
SnubWay Noise Defender
I have been discussing this new product for some time now during its development stages in the What’s On the Bench Thread. But I think it is time for its own dedicated thread. This usually happens when we reach a point of critical discovery or progress. Today is that point. We have conclusive data showing that a parallel plug-in filter (not a flow through filter) can reduce SMPS noise that contaminates AC mains power lines and outlets strips. The SnubWay Noise Defender is a compact module with a 120vac NEMA plug for the U.S. and Euro compatible Schuko plug for the 230VAC EU market. The 230VAC version remains to be tested.
I initially showed that the SnubWay can filter out 60kHz to 400kHz signal from a function generator imposed on an outlet strip. This was not plugged into a wall outlet or any real equipment.
Recently, I have made tests using a spectrum analyzer to look at the noise floor of the audio output of real devices to see if the noise floor can be improved by plugging in the SnubWay. I had a small SE Class A headphone amp (LuFo Lite) repaired and so I thought why not look at its noise floor. I was pleased to see that there was some 3kHz to 5kHz “grass” on the noise floor that was subsequently cut down by the SnubWay.
Above is a photo of the test in progress. You will notice the funny lampshade on the LuFo Lite (needed to cover the bright halogen bulb acting as resistive loads in the signal path). Here is the closeup of the noise floor cleanup:
Here is a photo of the MC phono line stage (Jhofland’s interpretation of a Pearl 3). It has a separate linear PSU connected by an umbilical to keep the noise low.
Here is a photo of the bench setup used to measure the phono line stage:
The noise floor of a phono line stage is one of the most critical low noise audio devices that we use in our systems. Given that the gain in these systems is around 60dB (that’s 1000 x) any sort of noise can be amplified and heard. Which is why it’s important to have as clean of a power source and power supply as possible for MC phono stages. Here is the measurement of the impact of the SnubWay for the output of the phono line stage:
Here are technical details of the testing:
Spectra were acquired using 256k length data sets acquired at 192kHz and FFTs processed with Blackman Harris 7 window function with 16 averages. Focusrite 2i4 USB audio interface and REW software. 100kHz 8th order passive brick wall filter on FFT input to prevent data aliasing from high frequency content. FFT scale: 0dB FS is 2.2vrms.
And here is what it looks like to use the SnubWay - simply plug it into the power strip:
Here is what it looks like inside:
It is a parallel network of 5 LRC filters designed using LTSpice circuit analysis and laid out on a stack of 3 two sided PCBs using premium SMT devices such as carbon film high energy pulse resistors, X2 rated safety film caps and CoilCraft shielded inductors. There is also an MOV/GDI for lightning suppression and protection from high voltage transients.
The 3D printed prototype looks like this:
The final product will be in satin anodized CNC’d aluminum with a premium Viborg acrylic base and copper wall outlet pins/blades.
We are now working on CAD models to produce the SnubWay that will look something like this:
We are still debating on final finish and color choice.
Here was an early test of the prototype as a P2P circuit and tested with a function generator noise source.
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Acoustical and mathematical advice requested
- By Vermond
- Subwoofers
- 9 Replies
Hi all,
Simulating speaker response gives me great joy and my personal 'challenge' is to be able to achieve the same as software like WinISD and Hornresp. I use Matlab for generating the plots, and if the acoustic part cannot be represented by lumped elements I use ANSYS Mechanical to generate the acoustic impedance by applying a velocity to the diaphragm and export the average pressure over the diaphragm. The complete circuit is solved for all frequencies in Matlab, by applying the 2-port network approach, which I got from Leo. L. Beranek & Tim J. Mellow's "Acoustics".
All of the above is fine, but there is a more complex problem, for which I am looking for some help (or motivation 🙂). 2 loudspeaker diaphragms enter a horn, where they are not loaded equally. My current workflow cannot deal with this. Below is a fictional example.
Summary:
- 1 speaker diagram, see below. This Matlab Simulink model is not used for calculation, only to show how my 2-port network Matlab script works. This is solved fine, and if required the acoustic impedance from ANSYS Mechanical is used for Zafront and/or Zarear.
- 2 speaker diagram (wired parallel), see below. The electrical and mechanical parts are put in subsystems for readability.
And a generic design in which these speakers are located, see below. 2 10" speakers, where both diaphragms have a velocity of 1mm/s. The 2nd speaker has a round obstruction in its way. The model is 1/2 symmetry. All boundaries are rigid, only the sphere's surface has absorption elements on it to allow wave propagation outside the mesh domain.
At 10hz the obstruction does not really affect the pressure on both diaphragms (or at least equal to both):
At higher frequencies we see the diaphragms are loaded differently (219Hz):
My parallel speaker diagram is probably not correct, because both diaphragms are affecting each other, so there must be a link between them somewhere. Beranek's book shows an impedance matrix which (I think) can be used to solve this problem. It needs 4 elements (z11, z12, z21, z22) which describe the relation between the pressure at both diaghrams, as functions of both velocities of the diaphragms.
Maybe some acoustical or mathematical wizards can help me solve this, maybe by elaborating on the impedance matrix for example. In the meantime I will try to do my homework by solving it myself 🙂.
Simulating speaker response gives me great joy and my personal 'challenge' is to be able to achieve the same as software like WinISD and Hornresp. I use Matlab for generating the plots, and if the acoustic part cannot be represented by lumped elements I use ANSYS Mechanical to generate the acoustic impedance by applying a velocity to the diaphragm and export the average pressure over the diaphragm. The complete circuit is solved for all frequencies in Matlab, by applying the 2-port network approach, which I got from Leo. L. Beranek & Tim J. Mellow's "Acoustics".
All of the above is fine, but there is a more complex problem, for which I am looking for some help (or motivation 🙂). 2 loudspeaker diaphragms enter a horn, where they are not loaded equally. My current workflow cannot deal with this. Below is a fictional example.
Summary:
- 1 speaker diagram, see below. This Matlab Simulink model is not used for calculation, only to show how my 2-port network Matlab script works. This is solved fine, and if required the acoustic impedance from ANSYS Mechanical is used for Zafront and/or Zarear.
- 2 speaker diagram (wired parallel), see below. The electrical and mechanical parts are put in subsystems for readability.
And a generic design in which these speakers are located, see below. 2 10" speakers, where both diaphragms have a velocity of 1mm/s. The 2nd speaker has a round obstruction in its way. The model is 1/2 symmetry. All boundaries are rigid, only the sphere's surface has absorption elements on it to allow wave propagation outside the mesh domain.
At 10hz the obstruction does not really affect the pressure on both diaphragms (or at least equal to both):
At higher frequencies we see the diaphragms are loaded differently (219Hz):
My parallel speaker diagram is probably not correct, because both diaphragms are affecting each other, so there must be a link between them somewhere. Beranek's book shows an impedance matrix which (I think) can be used to solve this problem. It needs 4 elements (z11, z12, z21, z22) which describe the relation between the pressure at both diaghrams, as functions of both velocities of the diaphragms.
Maybe some acoustical or mathematical wizards can help me solve this, maybe by elaborating on the impedance matrix for example. In the meantime I will try to do my homework by solving it myself 🙂.