Not really audio related but I figured I will ask here. I have an AC motor that runs on 115V and draws 1.3A. I need to drop the voltage going to the motor (to slow it down) by using 10 Ohm wire wound resistor. What wattage of the resistor should I use so it is safe for a continuous use?

Not much discussions here lately.

What is the current status of the website?

I found a post on StereoNet referring to Melbourne Audio Club meeting end of May and that Hugh would be presenting a new amp? Newer than Maya v4 and Titan?

To keep things accurate what are the current prices for the range of amp kits?

Hope you are doing well, Hugh.

This product is a series noise filter for power mains for your audio system and started asa P2P on the bench and has been under development for some time now. I am pleased to say that the production prototype has been completed and passed CE and FCC testing and certification.

It is a 5th order Chebyshev filter built with premium name brand components including two massive 19.5mH 10A 1000v PCB mounted CMC chokes. It should provide about -50dB of noise attenuation at 100kHz and more at higher frequencies. This should significantly reduce SMPS noise in power lines. Good news is that it is rated for 10A and works from 85VAC to 230VAC and has combo NEMA/Schuko plug receptacle on the front.

Here is a render of the final product. I don’t have the final in my hands yet.



But here is a photo of the inside from the test lab:


This was my bench testing of an earlier prototype with 2.5mH inductors. We later moved to larger inductors and smaller Y2 capacitors to reduce parasitic dissipation to comply with CE requirements. This had a huge impact on the cost of components as you can imagine going from a 2.5mH to Kemet 19.5mH 10A CMC - but makes for a well engineered, safe, energy efficient, and robust electrical appliance that will clean up noise like a black hole.

Testing below verified -50dB at 100kHz noise removal.

Hi, I've been reading the "dome midrange thread" and keep getting that itch to try something new.
My current speakers are an active 3 way that I've made myself using using drivers that have been recommended from this awesome forum. They consist of a T25b tweeter(5in wave guide), NE149W-08 mid and an sb34nrxl75-8 woofer in a 65L ported box tuned at 29hz, Everything has been impedance checked for resonances and standing waves and rectified where needed.
I made these speakers modular so I could try different things.
They currently sound amazing and I'm blown away every time I turn them on, they have a large 3d/deep sound stage with great detail and an amazing easy listing experience but you know how it goes with us type of people!
These are run from a Minidsp flex ht, 6 channel home assembled Icepower amp(1200as2/400A2/125asx2 new conductor version for the 125asx2). Crossovers are currently 250hz/2700hz all LR2. I have a good bit of experience with DSP from my car audio setup also with a mindsp 8x12dl.


I have a spare set of NE149W-08 drivers and can sort out another two channels of amp. What i'm thinking is to make another box section that sits between the current woofer box and the midrange pod. Then put two NE149W's side by side in that box(that would be the same style as the woofer box) and then a scan speak D7608 3in dome in the mid pod(with the correct enclosure size of it's own) or make a new mid pod that gets the ctc spacing a little closer.
I did also think about using one NE180W for the midbass but if the two 149's per side would work then that would be better as I already have them.
Then I would aim for crossovers somewhere around 150-200ish/6-700/2700 or whatever works best.
Trying to manage tweeter height would also be of concern.
My current desire is lots of detail and good dynamics. I'm also not really wanting to spend Bliesma M74 type money on this experiment.

Any thoughts or problems you can see?

Thanks.

I am able to use the analogue outputs on the UMC1820 with camilladsp (on linux), but nothing seems to get through on either the electrical spdif or the optical toslink. Using an alsa loopback works fine to get audio to the UMC1820, but I really want to drive camilladsp with input from another computer, not the one running camilladsp.
The number of channels for analogue output also seems to vary, having worked with 8 channels, 10 channels and even 12 channels . Can't remember what the circumstances were at the time.
My current conclusion on this issue is that UMC1820 simply does not work properly with camilladsp / linux. Prior to this, I was using brutefir, mainly with other cards, but also with the UMC1820 - it fairs no better.
If someone could show me a config.yaml that will work to access the UMC1820 I would be very grateful. I have wasted literally weeks on trial and error tests.
None of these issues are helped by the complete lack of any detailed tutorials or instructions concerning any linux software. The instructions, such as they are, are always incomplete and seem to assume you are an expert in everything that the instructions do not cover.

Many thanks,

Does anyone have experience with KEF 105 series 2 speakers? Wondering if anyone has updated or upgraded and how it turned out?

I bought the flagship DAC project from Ian Canada. Gabster's YouTube videos were very helpful putting it together but I need help connecting power. I'm using two Hammond transformers and I've confirmed with a DVM that I'm getting 3.3 volts from one LinearPi Pro, +/- 5 volts from a dual LinearPi Pro and I have another LinearPi Pro that I'll use to power a Raspberry Pi with +5 volts once I get the DAC working.

For the default setup of this project, where do I need to connect the different power wires? Should I use the output from the separate LinearPi Pro for +5 volts somewhere or do the +/- 5 volts wires provide that voltage?

Thank you,

-Robert

Edit Nov 5, 2022: nice analysis and summary of DML materials and response with python script by @homeswinghome

https://www.diyaudio.com/community/...s-as-a-full-range-speaker.272576/post-7159983

Edit oct 27, 2021: a nice Google drive library of DML publications by vdljev:

DML - Google Drive

Edit April 1, 2020: nice summary of DML patents by Burntcoil

A Study of DML's as a Full Range Speaker

Edit Aug 12, 2019: Nice summary here by BurntCoil on how to maximize performance.



I had some cheap exciters that I got from PE a while back and tested them out a while ago with a full 20x30in FC panel here:

Foam Core Board Speaker Enclosures? - Page 225 - diyAudio

I found that a large 20in x 30in panel can sound quite good with nice bass extension and a snappy transient response:



Impedance:



Frequency Response & Harmonic Distortion:



Impulse Response:



The results were interesting in that it sounded nice - with surprising bass and good midrange. I thought nothing more about it until recently prodded by the master of DML, CLS. He has worked on this extensively and is a treasure trove of good info. I since have discovered that you can really do some cool things with them by playing with driver placement, cutouts, adding mass, adding felt, adding ribs, making them huge, making them multi-way FAST, etc. the options seemed almost limitless. Take for example, a large center channel and a super 40Hz capable multiway that CLS built here: PIEZO NXT type panel - Page 60 - diyAudio and PIEZO NXT type panel - Page 61 - diyAudio

I feel like this technology just isn't getting enough attention. There are several very large threads on this on the web. It might be tough for a newbie to comb through. I am a newbie at this so thought I would document my journey for the Full Range forum to follow. I think it really could be a great full range speaker with some careful experiments. This will be really useful with some modeling using CAD and FEA modal analysis - such as available in many CAD packages like SolidWorks. One can play with shapes, cutouts, mass loading, variation in thickness and materials, boundary clamping conditions, etc. The idea is to spread the modes evenly with not any one mode dominating and causing a spike.

First thing is to play with it to get a feel of what we are dealing with. What is nice is that it is relatively cheap to play with. Exciters cost $3 to $20 ea and can be made by removing or cutting out large holes in the cone from an old driver and leaving the spider and VC to attach to the panel. Foam core or corrugated cardboard doped with shellac or PVA seems to be the materials of choice.

Some interesting facts:

1. Although it has no baffle, it is not an open baffle (OB) dipole in behavior - that is, there is no huge bass rolloff and it hits surprisingly low (40 to 50Hz is easy) for a zero baffle driver.

2. It is not a dipole but behaves more like a bipole or an omni.

3. It has very quick snappy transient response - nice drum sounds.

4. It is sensitive to how you mount it or frame it or hold it.

5. You are building a driver in reality - a driver and zero enclosure.

6. You want to avoid symmetric shapes and symmetric exciter placement in order to reduce the effects of the main symmetric transverse drum head eignmodes. Think reflection anti-symmetric shapes like uneven trapezoids, pentagons, blobs, etc.

7. It operates more by having high velocities and large areas for good efficiency vs large displacements - thus small drivers and large panels can be surprisingly loud.

8. The impedance is essentially "flat" relative to normal drivers in that there are modal peaks (many of them) but they range from nominal Re value (say 6 ohms to maybe 8 or 9 ohms throughout the 40Hz to 20kHz range - this presents a very flat load to an amplifier.

Here is a photo of a basic panel I was testing (right before I cut off some edges to form a trapezoid) - panels is about 1 square ft in size and made of standard dollar store foam core board. I suspended it with two pieces of twine from the top corners between a ladder to reduce the effects of edge boundary clamping:



Then I started to play with mass loading by adding blobs of modeling clay (8, 5, 4, 2, 1, 0) so you can see the effect on the resonance modes and the harmonic distortion. What is neat with this test is you can do it live while playing music and immediately hear what sounds more pleasing to the ear. Note the 50Hz bass extension. It is also surprisingly efficient with 85dB and a puny little voice coil. These are 0.5m and 0.71v for equivalent SPL at 2.83v and 1m.

Frequency Response and Harmonic Distortion for...

No added mass:



8 pieces of added mass:



5 pieces of added mass:



4 pieces of added mass:



2 pieces of added mass:



1 piece of added mass:



Impulse Response of 1 piece of added mass, note the sharp transient capability with relatively low after pulse or ringing:



I liked the sound of the 5 and 1 mass the best and for a FAST with a 200Hz XO, the 5 mass might actually do fairly well. Of course, sound clips to follow once I have more time to work on this. Just to demonstrate how quick and easy this is, I did all these experiments in less than an hour including making the DML.

last weekend I have heard two different power amplifiers by direct comparsion:
1) Pass "X 600" vs.
2) Yamaha "MX 10.000" (MX10000, MX-10000)
I was surpriced about the low sonic differences between this two devices and I guess, that the HCA technology is also a good solution, especially if the loss power must be lower by the same output power.

Are there HCA diy projects respective other commercial amplifier brands, where is HCA technology inside?

Hello everybody,

The last months I read through this threads to help my in my repair of a NAD C372.
Since september last year I decided I wanted to learn to repair on a deeper level and so I use this case as a learning study.

What was happened?
The amp went into protection and the left channel was toast

What did I already do?
I went through the entire left amp channel pcb and tested every transistor out of circuit (with a Peak DCA75)
The following were gone and were replaced:
Q316, Q313, Q318, Q338, Q317, Q337 and Q328, the one that caused all this.
Q315 is also replaced.

3IC1 en 3IC2 were also replaced, as were R349 and R329 who are fusible resistors for the 64V line.
VR302 and VR301 were also changed.
At last IC301 was also toast, which I replaced.

All diodes were tested. All resistors were measured, some out of the board to make sure they value is correct.
All capacitors in the entire amp I changed to new ones, including the PSU rail.

So what is the problem now?
I can regulate the DC offset voltage between 0 and 3V als described in the manual.
But the big problem is the bias doesn’t go higher then 3mV.

The weird thing is that on the other channel I can’t even go beyond 2mV.
Here the DC offset opamp is also changed and also the 18V zener diode who was defective.

Since I checked both board completely as described above I can’t find the cause of this.
The only thing that stands out is that the supply voltage is 73V instead of 64V.
Is this the problem?

The amps plays music and works already but this bias problem indicates there is something wrong…

Can you guys help me?
Service manual is in the attachment

me.

I'm working on building an Ian Canada DAC. Other (prebuilt) DACs I have or have used: Emotiva Xda-3, Schiit Yggdrasil and the Oppo BDP-103. I returned the Yggdrasil since I couldn't hear an obvious difference when I compared it to the BDP-103. The BDP-103 is getting long in the tooth and can no longer access my NAS.

-Robert

hi there,

I know that some of our users here have pre-ordered some of those new Bliesna T34 tweeters which are shipping now.

It's a brand new company, the owner and inventor was former developer at Morel and later product-engineer at Accuton.

Now he came up with his very own product. The T34 tweeter.
A 34mm (1.3") aluminium/magnesium alloy dome tweeter with varaible thickness.

Have a look at the specs yourself:
http://www.bliesma.de/product.html

In this thread I'd like to hear experience with this tweeter and discuss this product.
The price is 580€ (ex. VAT) for a pair.

Hello All,

Fun to be back on a forum like this. I used to be way into audio gear, then kind of fell off for a bit. Got into in in high school when I bought a nigh club sound system, 2000 watt 4-way McIntosh & JBL custom setup. Also build a plasma tweeter back in those days (anyone remember Ulrich Haumann?) Wish I still had some of that stuff! Have been a nomad for a while but setting up a new system now that I am a homeowner. Looking forward to being part of the community and getting advice from the real experts...

Here is a PDF of of the Pearl 3 from Burning amp 2023 for those of you whom already have circuit boards.





EDIT: Video presentation on Pearl 3 for BAF '23 can be found here - https://burningampfestival.com/videos/


EDIT: Build documentation "Pearl 3 Phono Build Doc-1.0d" is attached below. Version 1.0d is the most current. Download and print.


EDIT: Interview with Wayne about Pearl 3 - #881

EDIT: Modushop/ HiFi200 Chassis set available here - https://www.diyaudio.com/community/threads/pearl-3-burning-amp-2023.404054/post-7538807 https://modushop.biz/site/index.php?route=product/product&product_id=971

EDIT: Online build guide - https://guides.diyaudio.com/Guide/Pearl+3/28

EDIT: SMD soldering tutorial using Pearl 3 - Login to view embedded media

CNC phonostage : One of the best phonostage designed by Hypnotoad, very easy to build and sounds great. Prices for each PCB $8 + shipping $6 by Registered post. This is a very highend phoostage a very long thread here Another Super High End Phono Stage! No expense spared... | Audiokarma Home Audio Stereo Discussion Forums
I can also arrange a kit. It includes all onboard parts minus any hardware like RCA, case etc,
Price $65 + $10 shipping
Fully soldered boards , Price $80+$10 shipping.



OPS_Rev1, Dual power supply is designed by Omishra. Its based on LM317/337. The output voltage is adjustable by using two onboard trimpots.
PCB $8
Module $25




Regards
Sachin

Hi
I have 6X DCB1 PCBs layout by my friend Omishra. This version can be built very cheap. There are very few parts on board. Here is a build thread DCB1 Build | HiFiVision.com

Price $8 each, shipping flat $6 by Registered post




Regards
Sachin

Let's return to the discussion of the wonderful amplifier.

I tried to improve it using a simulator.

I used field effect transistors in a voltage amplifier. I equalized the current of the upper transistor of the voltage amplifier by adding an additional current source, and disconnected the quiescent current control circuit from the voltage amplifier. The coil in the differential cascade increases the gain without feedback by 20 db

Hi all,

While thinking about ways to speed up the settling of a single-supply single-op-amp RIAA amplifier, see https://www.diyaudio.com/community/...upply-phono-preamp-design.413571/post-7702435 , I found a way to include a second- or third-order Butterworth high-pass filter. As it may be useful outside the context of single-supply circuits, I give it a separate thread.

When you just look at the topology and ignore the component values, this is a rather conventional RIAA amplifier (you could make it even more conventional by connecting R₇ in parallel with C₅, that doesn't matter much for the principle):



Normally, C₈ is used to cause roll-off in the subsonic region and the network R₇...R₉, C₅, C₆ realizes the RIAA poles and zero. In this case, however, I use C₈ to realize the lowest RIAA pole at -1/(3.18 ms) and R₇ to get subsonic roll-off.

Note that C₈/C₅ = 1000, meaning that without the subsonic roll-off, the DC gain would be 1001, a very ordinary value for a moving-magnet amplifier (1 kHz gain roughly 40 dB).

With everything ideal, at the value of the Laplace variable s where the impedance of C₈ cancels the impedance of R₁₂, the feedback disappears and the gain goes to infinity. This means that there is a pole at exactly -1/(R₁₂C₈), so if this has to be the first RIAA pole, one needs R₁₂C₈ = 3.18 ms. It's actually 3.196 ms in the schematic, pretty close.

The disadvantage of using C₈ for the first RIAA pole is that C₈, which has a relatively large value, needs to be accurate to get an accurate first RIAA pole. (C₈ has practically no effect on the gain at frequencies much greater than 50 Hz, so its tolerance affects deep bass, but not channel balance.) The advantage is that you can include better subsonic filtering in the loop by adding two more resistors and a capacitor.

As an intermediate step, suppose you could add an ideal inductor with a huge value between the output and the negative input of the op-amp, chosen such that it resonates with C₅ at the desired subsonic roll-off frequency, and that you chose R₇ such that it damps the LC circuit to a quality factor of 1/2 √2. The subsonic response would then be very close to second-order Butterworth. That's because the gain of the RIAA correction amplifier is one plus the ratio of the feedback impedance to the impedance from the negative op-amp input to ground, and that "one" is quite negligible at low frequencies. Mind you, R₈ and R₉ contribute to the damping of the LC circuit, but not by much. You could also choose a quality factor of 1 and design the AC coupling at the input for the same cut-off frequency. The combined response is then third-order Butterworth.

Such an ideal inductor is totally impractical, but it can be approximated with a T network consisting of two resistors with values much smaller than R₇ and a capacitor to ground at the point where they are connected, see this figure:



The transfer from the voltage going into R₁₁ to the current coming out of R₁₀ rolls off at a first-order rate from some very low frequency onwards, like would be the case with an inductor.

I haven't found any simple exact equations for any of the values except R₁₂C₈ = 3.18 ms. In fact, I've been very lazy and just calculated approximate values for the other components, and then used a pole-zero extraction program to fine-tune the values.

Regarding those approximate calculations:
R₁₂C₈ = 3.18 ms to get the first RIAA pole at the right spot.

The DC gain would be 1 + C₈/C₅ without subsonic roll-off, so C₈/C₅ = 1000 gives you a midband gain of roughly 40 dB.

At s = -1/((R₈ + R₉)(C₅ + C₆)), the impedance of the network R₈, R₉, C₅, C₆ goes to zero and the gain of the circuit becomes 1. As a gain of 1 is pretty close to 0, this must be close to the location of the RIAA zero. That is,

(R₈ + R₉)(C₅ + C₆) ≈ 318 μs

At s = -1/((R₈ + R₉)C₆), the impedance of the parallel connection of C₆ and R₈ & R₉ goes to infinity. The impedance of the whole feedback network remains finite due to the other branches R₇ and R₁₁, C₇, R₁₀, but it does get pretty large. That means the second RIAA pole must be close, so we get the extra criterion

(R₈ + R₉)C₆ ≈ 75 μs

The (theoretical) inductance L is chosen to resonate with C₅ at the required subsonic roll-off frequency and R₇ is chosen to get the desired quality factor. R₁₀ and R₁₁ get convenient values much smaller than R₇ with R₁₀ also much greater than R₁₂. We then have

C₇ = L/(R₁₀R₁₁)

Best regards,
Marcel

Edits:
Input RC coupling
The RCR T-network that approximates an inductor actually approximates an inductor with inductance L = R₁₀ R₁₁ C₇ and a series resistance of R₁₀ + R₁₁. At low frequencies, it stops behaving inductively, it just turns into the series connection of the two resistors.

As a result, one of the zeros of the high-pass filter that are supposed to lie at s = 0 actually lies somewhere around s = -(R₁₀ + R₁₁)/(R₁₀ R₁₁ C₇). For the second-order cases, I have used the first-order high-pass at the input to cover this zero by making the input RC time constant approximately equal to R₁₀ R₁₁ C₇/(R₁₀ + R₁₁), or actually to a more accurate value for the displaced zero found by the LINDA pole-zero extraction program.

For the third-order case, I have used the input RC coupling to make the real pole of the third-order Butterworth response, so I couldn't use it to cover the displaced zero. I used the output RC circuit in that case, or simply did not cover the zero. The effect of the zero not being in the origin is typically only seen below 1.something Hz anyway.

There is another zero not exactly in the origin, this is related to the + 1 term in the gain expression of a non-inverting op-amp amplifier. It is so close to 0 that I decided not to bother correcting for it.

Regarding the single-supply versions, the filter R₁, C₁ is meant to provide some power supply ripple and noise rejection. (Even though the single supply has to be regulated, some extra rejection is very useful at the input of an amplifier that amplifies hum frequencies hundreds of times.) R₂ and C₂ in series with the cartridge impedance also help to suppress supply ripple. The second-order versions have greater values of C₂ than the third-order versions and therefore have better ripple rejection at the lower audio frequencies (such as 100 Hz or 120 Hz).

16 Hz split-supply versions
This is a version for split supply and 16 Hz cut-off frequency, see post #58, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7927611



For the second-order version (values in parenthesis), the input coupling capacitor C₂ can be replaced with a short circuit if you don't mind when the roll-off reduces to first order below 1.3 Hz.

This is a variant with 46 dB rather than 40 dB midband gain:


Thanks to having R₆ split into R₆ and R₀, the time constant of the input RC coupling network can be set more accurately without needing awkward values for C₂. This was implicitly suggested by hbtaudio on another thread. Because of the high midband gain, the op-amp needs to have a fairly high gain-bandwidth product to get accurate RIAA correction (16 MHz gain-bandwidth product will give about -2 % error of the location of the second RIAA pole).

Finite gain-bandwidth product
See post #100, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7964481 , for some rough calculations on the effect of finite gain-bandwidth product of the op-amp.

From post #101 onward, Nick Sukhov points out that an amplifier with a high open-loop output impedance would result in a loop gain that depends much less on the RIAA correction circuit impedance. That's something to keep in mind when designing a discrete amplifier, you don't have the ability to choose a high open-loop output impedance when using op-amps.

Applying the subsonic filter to a discrete preamplifier based on the Hoeffelman and Meys configuration
The discussion with Nick and Chris about open-loop output impedances made me realize that the subsonic filter of this thread could be combined with a low-noise ("electrically cold") input termination resistance realized with a special feedback configuration that Dual already used in the late 1960's (CV40 phono section, see https://www.diyaudio.com/community/...o-input-load-modification.424717/post-7947176 ) and that was advocated by Hoeffelman and Meys in a 1978 AES article (Jean M. Hoeffelman and René P. Meys, "Improvements of the noise characteristics of amplifiers for magnetic transducers", Journal of the Audio Engineering Society, vol. 26, no. 12, December 1978, pages 935...939, see also Ernst H. Nordholt, "Comments on "Improvement of the noise characteristics of amplifiers for magnetic transducers"", Journal of the Audio Engineering Society, vol. 27, no. 9, September 1979, pages 680...681). The very first electrically cold resistance was made by William Spencer Percival and W. L. Horwood in 1939 as far as I know, but they used a different configuration and did not apply it to phono preamplifiers. See W. S. Percival, "An electrically "cold" resistance", The wireless engineer, vol. 16, May 1939, pages 237...240.

The schematics below show the resulting configurations. They are identical, but the left schematic is for people familiar with nullators and norators, the right schematic for people who feel more comfortable with high-gain twoports and op-amps. At frequencies well above 50 Hz, the input impedance approaches (R₁₃ + R₁₄)/(1 + R₁₃/R₁₂ + R₁₃/R₁₀). You can make this equal to 47 kΩ while using an R₁₄ that is much greater than 47 kΩ, thereby reducing the thermal noise current √(4kT∆f/R) that gets injected into the input.



You can't do this with op-amps (not without floating supplies anyway) because op-amps lack the negative output that conducts a (signal) current equal but opposite to the current through the positive output. That is, you can make electrically "cold" resistances with op-amps, but not as shown here.

Document about dimensioning the circuit
The attached zip file contains a pdf document that explains step-by-step how the component values were found (section 2) and that presents a more accurate method than I have used (one that doesn't need fine-tuning with a pole-zero extraction program, section 3). It also contains a spreadsheet for the more accurate way to calculate the component values.

Deriving the expressions was a nice exercise, but I'm not at all convinced that my more accurate calculation is of any practical use. It can very easily lead to negative or complex resistances.

I want to make an input selector, using Raspberry Zero and phono pre-amp, both jammed on a single board. Since I am IT guy, not an electrical engineer, can you PLEASE review my PCB design and let me know is this going to make it really bad and what should I change to make it work nicely?

This is a RIAA pre-amp part:


This one is an input selector, controlled with Raspberry, with connectors for switch and display:



Here is the entire board, top side first:


...and bottom side


THANK YOU IN ADVANCE!!!!

Marin

The past two weeks have been a black hole of DIY subwoofer obsession. With high-end audio gear costing more than a used car these days, I’ve started thinking—why not build my own

I’m toying with the idea of upgrading to a set of identical DIY subs in near future may be in a year. Is it a great idea or a potential money pit? No clue. But after walking through endless forum posts and YouTube rabbit holes (mostly featuring folks who think Dayton is the only driver on Earth), I’m officially ready to consult the real gurus—you guys.

The bass battleground:
13.5 ft wide x 21 ft long x 10 ft high
Its a Acoustically Treated Room (Thanks to Anthony Grimani Videos), I have M&K S150 LCRs, SVS Bookshelf Surround and Subwoofer i have two PB 2000 (Not Pro) at rear and single Rythmik FVX 15 in front

Here’s my current state of mind:

1. DIY subs: brilliant or bonkers?
2. My woodworking skills are… let’s just say Sharpening the pencil .
3. What good drivers are available in India? Dayton is everywhere, but I’ve got my eyes on Lavoce and BMS
4. 18-inch vs. 21-inch: is bigger really better?
5. WinISD is fun until it throws a “division by zero” tantrum. Can I really trust it to model things down to 20Hz?
6. Driver shortlist so far: Dayton UM1,Lavoce SAN214.50 (Refered fellow forum mate!) & BMS 18N862
7. Looks BMS is good.
8. Sealed or Ported which one to go.
9. And for designing Sealed do i need to depend Win ISD or i can go with the capacity details provided in OEM Site BMS says 150 Liter for Sealed Enclosure.

Goal: cinematic bass that punches you in the chest—just like the legendary JBL 4645C.

Oh, and fun fact—ChatGPT’s been throwing out suggestions like it’s got a subwoofer PhD. Honestly, it's like talking to an audio-obsessed buddy who never sleeps. 😄

I've been trying to figure out why there's car amplifiers that use 12-16vdc and produce 1,000+ watts rms easily but there's no easy solution to power class d amps boards in a car? I want to be able to power some class d amplifiers that require 32-36vdc in my car. If a car amplifier can do it why aren't there any power supplies available to boost 12v to 36v for this purpose? The only things I've seen even close to working are the cheap boost converters on eBay, Amazon, and AliExpress but the quality, lack of directions, and even worse the believability of the specs are less than ideal. I would have tried them anyway if I thought there was a chance they would work without introducing noise or other headaches.

I plan to use a Hypex SMPS400A400 to power a stereo AB100 integrated amplifier. My question is: has anyone added additional capactitance on the amplifier boards with this SMPS? I contacted Hypex and they just said "Our SMPS’s have sufficient capacitance for the power they can deliver. Adding extra capacitance may be possible but it is not required and not supported. Doing so will be at your own risk." Not very helpful. If you aren't familiar, the AB100 boards have places for four capacitors on the board. I have 4700uF 80V capacitors to put in those spots if it can handle it.

Hardly used. One has had a repair excellently done by the speaker manufacturer I bought them from.

£300 plus post costs to UK or collection from Guildford, Surrey.

Hello,

I am looking for a service manual for KEF TDM45B.

Any help?

Drivers for sale, most of them brand new in a box, some opened but still in new condition.
Notifications have been ticked but i dont receive them for some reason....

Pricing is per driver
Thank you

I have asked member ostripper to lay out a Class AB amp officially called the "diyAB amp" ( nicknamed "the Honey Badger" ) using many of the best characteristics of amps of this type that have appeared on our forums, so that we can feature it in the diyAudio Store. He has agreed to give it a go, is really interested in comments to help it be a paradigm of its type, and will be posting here very soon to describe where the design is at this point.

So pile on!

Mark
"Variac"

Build thread for diyAB Amp: http://www.diyaudio.com/forums/solid-state/211905-diyab-amp-build-thread.html

Simple amplifiers have always been a challenge. Here is the one, single supply operated (24V), 8W amplifier with only 4 transistors, singleton input and bootstrap, with coupling capacitors. The schematics follows:



The amp uses Vbe multiplier (Q10) to stabilize idle current with temperature. Idle current is set to about 60mA.

Frequency response is flat


Stability is very good and feedback factor is low and flat


Spectrum has mostly low harmonic order of distortion components


and THD+N vs. output power


If you asked me if I built it - I did, however lower power 2W version supplied from 15V PSU.

I got an idea that i can´t get loose.
What if we apply current dumping to reduce crossover distortion in an ordinary amplifier.
With an internal feedback of 850 ohms it puts some considerations of the input stage but that can be a later discussion.
Has someone seen a similar idea?

abandoning a project , i have a brand new pair of VOLT VM527 2inch dome driver to sell , never mounted , never used.
my loss is your gain .

wanting 450€ for the pair including shipping to E.U only
thanks for looking.

will not have longer use for them at this moment , i sell a brand new pair of Mundorf dipole AMT tweeter 17D2.2 same as used in the QUALIO speakers

new price 650€ asking 450€ shipping to E.U only

After visiting @Audio Elite in 2017 I knew I wanted to build speakers like that. Slowly but surely I gathered parts. 2nd hand or discounted. Since Stuart was not allowed to share the crossover schematic I had to find something else. But I saw something that was similar to the frequencies I "needed" to cover. I copied that and did a mod for the different tap on the Crites 3636 autoformer. I've posted the crossover pic but here is the complete system schematic..

PS...there is still some work to do on the crossover.

I also decided on open baffle mid-bass. Because I could go active/dsp. And because wood is very expensive.

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Hello! I have a BSR EQ-3000 that I bought back in college (REALLY long time ago 🙂 ) which I recently brought out to set up, as my son is getting into vintage eqpt and vinyl. So I set it up with my Denon and old Bose speakers and cassette deck. Everything worked fine... for about 30 minutes.

At that point the BSR went dark. Turning it off and on produced a flash from the LEDs on the eq sliders, and a mild crackling noise but nothing else (the LEDs flash then go off, everything stays dead). I verified that everything else still works by physically removing the eq from the loop, and tried letting it sit for 30 mins and plugging it in by itself -- same behaviour (see the video).

Any ideas where to start troubleshooting? I haven't taken it apart yet but I fear that's the next step. Thoughts?

edit: can't get the video to show up properly on my post 🙁

I'm selling off some speaker parts that I'll never use. I hope someone else can use them. Brand new condition, open box but never used or mounted together or mounted on a baffle. I never did anything with them.

1. Pair of EOS-8 waveguides. 2-bolt pattern mounting holes. 1-inch (2.54 cm) diameter throat opening. $30 for the pair, plus shipping (they're light and not very large).

2. Pair of DNA-150 HF compression drivers. 1-3/8" T-18 thread screw-on mount, as for Eminence 10CX, etc. $35 for the pair, plus shipping (small and don't weigh much).

3. Pair of 1-3/8" screw mount to 2/3 bolt mount horn throat adapters. $15 plus shipping.

Prices are US dollars.

no signal, bridge.


5hz.


100hz.


2khz.

10khz.



20klhz.


50khz.


100khz.

300khz.

senoidal 20khz.


the preamplifier.

1.177 / 5.000


Hello, first of all I want to make it clear that, as almost everyone knows, Chinese advertisements and clones often do not even look like what they advertise. In this case, I want to make known what I consider a bargain. Leaving aside the title we have a preamplifier that I consider excellent, at a ridiculous price of 10$ with shipping (at least to Spain), small size (easy to connect if you want an integrated amplifier), you only need a 15-0-15 v. transformer, or a switched source for example of equal value for another 7$, it has its regulators, with its capacitors that really work very well, at the input it has an OP JRC-5532D, then BC550 and BC560, everything quite normal, I ordered 2 units, assembled it and passed it through the oscilloscope, the result is excellent (in my opinion and the price) I put captures of up to 400khz in square wave, the sine wave logically better, it responds from 5hz without problems, I have tried other clones and other originals that I have, and this one on paper gives better measurements, something that measures well can't sound bad. The sound, the little I have tried so far, is quite good, better than the Denon 1500pra that I have.
and other clones except the PASS 1.7, for me it is better for more detail, dynamics and width, but for the price it is gold, now here come the more technical forum members (I am not) who could suggest improvements, for the already excellent preamp, I attach captures from the oscilloscope, greetings.

Hi, I have a Mark Levinson 31.5 cd transport which works well for a while and then appears on the display: SERVO RESET. the reader does not obey to stop or change the track, it has to be turned off completely. Has anyone had a similar situation that could repair it? Thank you very much.

I am looking for offers on this last devices I have from Toshiba, I can supply 40 pairs of 2SC4793/2SA1837 and 50 units of 2SC1815GR
minimun quantity to buy is 4 for any part reference.
4 pieces 2SC1815GR 3€/lot
4 pieces 2SA1837 9€/lot
4 pieces 2SC4793 9€/lot
All supplied transistor are from the same lot
Payment by PP
I have more parts from Toshiba that you could check here:
https://www.ebay.de/usr/rocksandsound
Regards

I have for sale the following evaluation boards from Tripath.
RB-TA3020-96€
RB-TA0105-160€
I do have more Tripath amplifier boards that you can check here:
https://www.ebay.de/usr/rocksandsound
Payment by Paypal tracked shipment,shipping to US on request.
For more information PM

I have here for sale severall capacitors from Nichicon used.
1-Raw
Used Nichicon 15000uF 71V 14€/unit or all 8 for 100€
2-Raw
Used Nichicon 12000uF 71V 7€/unit or all 6 for 35€
All used capacitors here for sale have been measured and are within 10% of its rated capacitance
Regards

I recently rebuilt a pair of ESL-63s, and printed some 5 degree tilt back feet, but I don't like having the sound beamed up at my ears from close to the floor, so I designed and built some stands using 1"x1" and 1"x2" t-slot. The stands lift the speaker about 350 mm off the floor, and they are sturdy and center the weight so they aren't easy to knock over.



Details and link to CAD file here. Extra bonus: they are high enough that my cat will probably stop using them as a scratching post.

Yeah, looks like a clickbait but I had such a design target.
Analog Processing Unit contains two devices in the same housing. THAT1510 based preamp-LNA 34/60db gain, 10Vrms max output level, and 48VDC phantom power, just in case if need to work with a condenser mic. The input impedance is 47kOhm 100pF in case it needs to use as a MM phono-preamp without RIAA(Cosmos ADC has internal DSP and may handle RIAA EQ online accurately). The input noise of the LNA is about 130nVrms(A) i.e.
5-6x times fewer than APx555 which, for instance, is not capable to measure the dynamic range of $13 Meizu HiFi DAC, APx555 result is 123-124db(A). Cirrus promises AES17 DR 130db(A) for CS43131, Cosmos APU+Cosmos ADC result is 130.5db(A) with 12db room to APU's residual AES17 DR for 2mV. The LNA's balanced inputs are clamped at 5-6V by TVS thru the 10ohm 0603 resistors in serial to each input i.e. an overvoltage with a good enough current rather may burn the resistors than TVS but that's safe for THAT1510.
The second and biggest half of Cosmos APU is the 1/10kHz active notch filter with -30db ratio(Q about 50) at the fundamental frequency and nearly 0db at 2nd and 3rd harmonics i.e. you can simply divide by 30(or subtract 30db) the THD/THD+N result to get a normal value. Also, with REW or Arta, you can use a calibration file to normalize the notch's frequency response and observe FFT results directly. The max balanced input level is 10Vrms, it is also clamped but by the active clamp with opamp's rail voltage-1V, hence, 10ohm 0603 again may be burned like a fuse. The residual harmonics of the distortions are <-150db@1kHz or <-130db@10kHz, that's probably isn't too spectacular but still better than APx555. The residual THD+N, practically achieved one, -132db@1kHz@10Vrms(DAC+LPF approach), calculations based one is -134db but I can't confirm that so far because of can't find a perfect-zero-noise sine for that 😉 It is clear that APU is kinda a microscope for your ADC, even any laptop audio input with APU gets the ability to measure a tiny noise of DACs or LDOs, and distortion levels are perhaps lower than AP SYS2722 or even Cosmos ADC.
The best way to power Cosmos APU with a power-bank(or a good smartphone charger), is to minimize GND loops issues.
The device is quite simple and inexpensive, strange that's not on the market already. Especially inexpensive(~$70) would be a version without the case as a PCBA which any DIYer can adjust by 4pcs trim-pots(2pcs for the 1kHz Twin-T -30db or -40db, another pair for 10kHz -30db).


REW with the calibration file DAC+LPF->CosmosAPU->CosmosADC


The residual THD+N of 5 preproduction samples, -101db of reading need to subtract 30db i.e. THD+N -131db@9.5Vrms the same about harmonics -127db means -157db







APU Notch frequency responses:





AES17 Dynamic Range test performs at -60dbfs level by measuring the THD+N A-weighted. That plot shows the residual THD+N (A) of the LNA at 1-10mV level sweep. 2mV corresponds 2V 0dbfs DAC like CS43131, and -82db needs to subtract 60db to get AES17 DR = 142db. A typical high-performance DAC with 5Vrms 0dbfs could be measured down to 150db and so on.




The residual noise of 5 preproduction samples(the S2 using SSM2019 instead of THAT1510), the reading need to divide by 1000 due to 60db of LNA i.e. a uV is actually nV.




The gain of Cosmos APU preamp is +60db(1000x times), hence the scale units are nVrms(A). After the APU unit turns On, you can see 5s of settling,
from 6s the input noise reaches 130nV, about 9s 48V power was turned On, and after 1s reached 142nS. Hence, APU's phantom power adds just 12nV to the preamp noise, that's a nearly ideal result as I think 😉 The test was performed with a shorted preamp's inputs(In+ to In-) and biased to the GND by 2x6.8kOhm resistor as a dummy condenser mic.



and the spectrum 20Hz-130kHz of the same test, units need to divide by 1000 as well, so uV becomes nV. This is a good illustration to killing the myth that SMPS is always dirty 😉 Cosmos APU contains no linear voltage regulators at all, only SMPS with Fswitching > 1MHz.



Finally, I got working Cosmos ADC with internal DSP and RIAA EQ implemented for all Fs from 44.1 to 384. A pair of Cosmos APU was used as a dual-mono phono preamp +34db gain with no analog RIAA, which was applied in Cosmos ADC digitally. The video is a comparison of LP vs CD sound, recorded this way.
The formal specs of such a combo:
THD+N@5mV-40ohm(AP output) -83db, and suddenly it places Cosmos APU to the 2nd position of the phono-preamps ASR rating(and yes, I know his tests are silly) 😉


I remeasured Cosmos APU+Cosmos ADC+RIAA max accurate, AC line hum virtually zero.
5mV 10ohm source APU gain 34db Cosmos ADC 1.7V sensitivity, THD+N -89db or -97.7db(A). Due to H2 and H3 being very low, SNR is also 97db(A).


250uV 10ohm source APU gain 60db Cosmos ADC 1.7V sensitivity, THD+N -66db or -78db(A). Due to H2 and H3 being lower than the noise floor, SNR is also 78db(A).


The real SNR i.e. APU source is a cheap MM cartridge AT3600L, the LP record is digitized by Cosmos ADC@32/384, normalized to fit max peak to the -6dbfs, next, the stylus is up to the air, Pioneer PL-300(made in Japan 1980) turntable keeps 33.33rpm. This way the SNR = 91,2dbfs-6db=85.2db(A), that's looks decent if the LP "silence" between the tracks shows SNR = 65db(A) and less.


Of course, both these functions, mic and phono preamp, are features just for the feature list, but I think both are decently implemented and virtually for free so I couldn't call that a marketing 😉

https://e1dashz.wixsite.com/index/cosmos-apu
https://archimago.blogspot.com/2022/05/early-look-cosmos-apu-high-performance.html#more
https://www.l7audiolab.com/f/e1da-cosmos-apu-2/ APx555b measurements
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Some developers of tube amps claim, that for HV power supplies (for anode voltage) only passive filtering (maybe with zeners for slightly shunt regulation) are ensures the best possible sound quality resp best sonic performance. I. e. only one or two inductors and a chain consist of several RC filters. From my experience this is right, as long only top quality elcaps are in use and in the last RC filter additional a MKP version with 50-100uF.
An example are several pre-amp devices from Klimo.

Schematics of first five attached images show such passive versions - i. e. without serial or parallel voltage regulation (except one version in inage 2 with the zeners).
The later attachments show schematics of serial regulators in several variations.

At first glance the benefit of serial and parallel regulators are a very stable voltage, that is additional adjustable in a wide range.
But is it in real live really a fact, that a stable anode voltage provide a better audio than a passive filtered voltage ?
In general I don't think so.

Thank you for comments.

P.S.: some URL's:

HV Serial Regulator
https://tubecad.com/2021/10/blog0546.htm
https://pearl-hifi.com/06_Lit_Archive/14_Books_Tech_Papers/Broskie_John/HV_Regulator_OP27_IRF820.pdf
https://www.bartola.co.uk/valves/2015/01/10/popping-the-shunt-voltage-regulators-clogs/
https://www.bartola.co.uk/valves/2012/07/15/new-ht-bench-power-supply-600v/
https://www.bartola.co.uk/valves/page/2/
https://www.tubecad.com/2021/10/blog0547.htm
https://www.tubecad.com/2020/02/blog0493.htm
https://www.tubecad.com/2021/09/blog0545.htm
A commodity-based HV regulator: FlexHV
High Voltage Jung Regulator

HV Parallel (Shunt-) Regulator
https://www.tubecad.com/2006/11/blog0087.htm
https://www.tubecad.com/2021/11/blog0548.htm
https://www.tubecad.com/2021/01/blog0527.htm
https://www.tubecad.com/2007/06/blog0109.htm
Very low parts count shunt regulator

Passive Filter (RC/LRC)
https://www.jogis-roehrenbude.de/Leserbriefe/Carsten-RIAA/Carsten-RIAA.htm
https://www.audionist.de/2015/11/27/netzteil-fuer-einen-roehrenverstaerker-2/
https://www.turneraudio.com.au/audiofilterchokes.html
https://www.turneraudio.com.au/audiofilterchokes-page2.html
Choke ratings and resistor values in choke input supply?

TUBE AMP POWER SUPPLIES (basics to all supply variants)
https://www.turneraudio.com.au/powersupplies.html

croft series 7 HV regulator
https://www.tubecad.com/2021/10/blog0547.htm

Hi there. I have enjoyed a fine Rega Elicit mk2 amplifier for several years. However, recently I noticed that at a certain position of the volume control (or rather the LED track, as the knob has no fixed position) the sound from the left channel disappears, if I turn the control knob (or remote control) a few clicks clockwise the sound reappears, although it is slightly muffled. I checked different inputs, different sound sources - the problem does not disappear, the sound from the left channel disappears exactly in this position of the LEDs. I need your help.

I have twelve (12) brand new 22,000 uF, 80V capacitors. Ordering error from Mouser and I'm stuck with them. I'm trying to recover some funds. Asking $35 each, plus USPS shipping from zip 17402.

4.125 inches tall and 2.5 inches in diameter. Great for all your power amp projects! Send me a PM if you're interested.

Mouser part number: 598-CGS223U080W4C

Hello Friends

Geneva Audio Classic/XL great simple looking speaker and config. 2 tweeters 2 5.25woofers and 2 - 8" woofers ported​

Wanting to build a powerful All in one box for parties and moe around the house. I want more bass pressure I dont get from Marshall Woburn or Bose S1 speakers at home.

Can you suggest the combination Thinking of using the Dayton KAB 4x100 bluetooth amps that also have DSP. Unfortunately they Dont have AirPlay 2
I was also wishing if TPA 3255 based amps had some DSP built in

In Drivers please suggest Peerless or SB or last option Dayton

regards
anand

This amp uses only 2 transistors, one J113 JFET and one BD139.
And only 4 resistors.
What is different is there is no feedback.
Because of this there is some distortion, THD 0.5% at +10dB into 32 Ohm.
The amp should probably work into headphones 16-600 Ohm.
But the design was only developed using 32 Ohm.

PCB available in the diyAudio store here -

M2X – diyAudio Store



A First Watt "M2" amplifier without (or with!) 2SJ74 / 2SK170 JFETs

After Burning Amp 2017, member 6L6 and I drove to San Francisco airport and naturally we talked about diyAudio the whole way. We imagined that many people would love to build a DIY version of the First Watt "M2" amplifier -- except that the M2 requires JFETs which are painfully difficult to source. Toshiba has stopped making them, and the Linear Systems replacements are hard to find too. (The diyAudio Store has been sold out of the most sought-after Idss grade for months and months).

HOWEVER, we noticed that the JFETs were used only in the "input stage" of the M2 amplifier, to implement a unity gain buffer that drives the primary of an autotransformer. The remainder of the M2 amplifier uses components that are in active production, available from multiple different distributors, today.

So I proposed that we lay out a new PCB for the First Watt M2 amplifier, with the input stage and its Unobtanium JFETs contained on a removable, replaceable, interchangeable Daughter Card. Now we can build input stage (unity gain buffer) daughter cards, with whatever components we wish. Of course one option would be (should be!) Nelson Pass's original design using Toshiba JFETs, for those who are willing and able to obtain them. And there will be other options too. Options that implement a unity gain buffer without Unobtanium parts. Fortunately it is not an impossible task to design a unity gain buffer which fits on a small daughter card.

My feeling was: let's make it possible for people to build and listen to an M2 amplifier, at the same time they search the globe for JFETs. Months later, if and when they succeed in finding and purchasing JFETs, they can stuff and solder the JFETs onto a daughter card, swap it in, and keep listening. Or maybe, just maybe, they'll fall in love with the sound of the M2 without JFETs, and quietly stop searching.

As we drove along and discussed it more, both 6L6 and I came to the conclusion that this sounded like a pretty exciting idea, and I began to work on the PCB layout and the input stage circuit design(s) during hobby hours.

I'm now pleased to announce that revision V1 of the M2x amplifier has arrived. PCBs for the amplifier left and right channels (the "mother" boards) have been built. PCBs for five different input stage circuit designs (the "daughter" boards) have been built. I've shipped all PCBs and all daughter boards and all electronic components to 6L6's electronics lab, for one of his beautiful Build Guide jobs: Assembly and Step By Step Photography. I'm very pleased to say that he has built the amplifier and a few of the input stages. It works. It sounds good.

But 6L6 has a full time job and must fit diyAudio into his spare time. The Build Guide is not yet complete and won't be complete for a while longer. Similarly, my own build of my own M2x amplifier has begun but it is also not complete. That's how hobby projects work.

While we wait for the Build Guide to arrive, I thought readers might enjoy looking at the circuit schematics of the M2x amplifier and its five different input stage options.

I've given each of the input stages a code name, derived from the city where its input devices were manufactured. (As best as I was able to learn which fab made which devices). The five M2x input stages we have today are

1. ISHIKAWA - Nelson Pass's original M2 input stage circuit design, using Toshiba JFETs made in Ishikawa, Japan. The PCB naturally also accepts Linear Systems JFETs.

2. MOUNTAIN_VIEW - A single ended, class A unity gain buffer circuit design, using the J112 JFET made by Fairchild in Mountain View, California

3. TUCSON - An unity gain buffer circuit using a high performance Burr Brown opamp, built in their fab in Tucson, Arizona

4. NORWOOD - A two-chip unity gain buffer using an extremely high current video driver IC, plus a front end JFET opamp made by Analog Devices in Norwood, Massachusetts.

5. AUSTIN - A BJT complementary "diamond buffer" using Motorola (now ON Semiconductor, owner of Fairchild) high Beta bipolars, built in Austin Texas

How might someone choose which of these to build? (Or, which of these to build first?) Here are some thoughts.

ISHIKAWA (using Toshiba or LS JFETs) has the absolutely lowest parts count and is the easiest to solder together. But those FETs are hard to find.

TUCSON using the thru-hole OPA604 chip, is the lowest parts count board that doesn't use unobtainium devices. And the OPA604 opamp has JFET inputs, so if you love JFETs, it's one of the opamps you probably know and adore.

TUCSON using the SMD OPA1611 chip, is the lowest distortion input stage of them all. Even lower than the Nelson Pass original. It only has one surface mount part; the rest of the board is thru-hole.

NORWOOD is all-SMD, just like the entire electronics industry has been for the last 20 years. Norwood includes a 200 mA driver IC, and a JFET input opamp from Analog Devices. Its output current capability is at least a factor of five greater than any of the other M2x input stages. If you're at all concerned about the input impedance of the output devices at HF, reflected from the transformer secondary back into the primary, Norwood can drive it better than all the others.

MOUNTAIN_VIEW uses single-ended, class A, discrete circuitry and no feedback. If you are a big believer in single ended class A, if you love this quote from a famous article by Nelson Pass: "For reproducing music as naturally as possible, push-pull operation is not the best approach. Air is not symmetric and does not have a push-pull characteristic. ... Only one linear circuit topology delivers the appropriate characteristic, and that is the single-ended amplifier. Single ended amplification only comes in pure class A," then you probably are interested in listening to the Mountain View input stage. Parts note: please see post #3318 in this thread if you're having trouble finding a few MV5075C LED parts. That post suggests alternatives.

AUSTIN is a thru-hole board with 4X more components than the other boards; its resistors are mounted vertically to save PCB area. This permits a Diamond Buffer circuit with high performance, precision current source loads. Builders who love parts substitutions / optimizations will appreciate that Austin's PCB silkscreen doesn't indicate transistor orientation. Instead, transistor pins are labeled B, C, E in a perfect circle; allowing you to drop in Japanese 2SA/2SC transistors (pinout ECB) or European BC transistors (pinout CBE) or American 2N transistors (pinout EBC) however you please. Austin is recommended ONLY for experienced builders. If you're a newbie, hire someone to solder your Austin board!!


What happens next?

First, 6L6 and I work on finishing our respective M2x amplifiers in our respective spare time. This includes building and listening to all versions of the input stage daughterboards.

Second, 6L6 posts his photographic Build Guide here on diyAudio.

Third, if the amplifiers work well and sound good, I'll offer my approximately 30 sets of PCBoards (2xAmp plus 8xInputStage per set), if anyone wants to try the M2x.

Fourth, if I happen to sell out, 6L6 and I will put our heads together to figure out the next move.

Schematics and BOMs attached below in .zip file



edit- if you are wondering how a correctly-built and properly function M2x will appear during a dim bulb test, see (a post written in Sept 2019) and also see (another post written in Oct 2023). They show you what to look for.

~

Am I missing something? I can't seem to save the filter settings for the forum so I end up re-entering them every time.
Any thoughts? Problem Exists Between Keyboard And Chair?

Background – I’ve built speakers in the past but my most recent build was in 1999. So I am trying to read and catch up with current speaker building trends.

A lot more attention is being paid to off-axis performance these days. I gather that consistent and as wide as possible directivity are said to give a more realistic sound. That said, I am seeing two different approaches to off axis in tweeters. I have borrowed a couple of normalized off-axis curves from hificompass.com to demonstrate. Many thanks to them for their fine measurements.

Some folks are trying to keep the widest directivity to as high a frequency as possible followed by a smooth narrowing at higher frequencies. An example of this is the BlieSma T25S-6 where there is no narrowing of the off-axis curves until about 6.5k Hz. Then it narrows linearly until a break-up in the 17k – 20k region.

Others prefer an early but smooth narrowing, often accomplished with a wave guide. An example of this philosophy is the Satori TW29BNWG-4 which uses a wave guide and starts narrowing about 1300 Hz. As it happens, hificompass.com offers 3D printer plans for a the BlieSma T25S-6 which changes the T25S-6 into something like the Satori.

My question is which kind of directivity pattern is preferred and why? If it’s just preference, what are the advantages and disadvantages of each.


Thanks.

I'm adapting an existing ps to supply +/- 15-20 vdc to preamp boards that draw zbout 55ma per rail. I'm planning to use an existing transformer and have a simple PSU board to adapt with spare parts to provide the vdc to the boards I've built. Once the AC from the transformer (23-0-23vac) has been converted to VDC via a bridge rectifier it then has some bulk caps. The output voltage is, however, 31.8vdc (unloaded) and I'm looking for a simple way of reducing this to 15-20vdc bearing in mind the 55ma draw of each board. What would you good folk recommend to achieve this? I have a simple option on the existing board to insert a through hole component... The boards have regulators so I just need stable raw power supply that can handle the current draw without smoking...

Building the Pass/Firstwatt F4

This is a fantastically good sounding amp - read more about it here before staring;

https://www.firstwatt.com/wp-content/uploads/2023/08/prod_f4_man.pdf

The thread at DIY audio -

http://www.diyaudio.com/forums/pass-labs/97540-f4-power-amplifier.html

And this the the corrected schematic (The schematic in the Firstwatt article has a typo, also this one agrees with the PCB)



Here you will find a build guide for the Pass / Firstwatt F4 power amplifier using PCBs and chassis from the DIYaudio store.

The 5U 'BIG Amp Chassis' is shown, because that's the one I have. It will fit comfortably in a 4U 'Jack of all chassis' and have enough heatsink as well.

~~~~~

There are plenty of places that you could start, but for the sake of illustration let's begin with the heatsink assemblies -



This is the heatsink(s) from the 5U 'BIG Amp Chassis' It has a mirror-imaged set of pre-drilled heatsinks and brackets to hole them together and make a mounting point for the rest of the enclosure. The 4U is similar, but the heatsink is a single piece.




This build will also utilize the 'DIY friendly' baseplate, here shown with the feet and hardware, and also the heatsink's brackets.


There is a hardware package available for the pre-drilled back and heatsinks, including input and output jacks, IEC module, and hardware for the PCB and heatsinks.


The contents of the hardware bag.


Using the brass PCB standoffs, install them into the PCB mount holes as shown, to get the following pattern;


Now there is a place to mount the amplifier PCB


Speaking of PCB, it's a very nice layout, plenty of room, and the ability to use many sizes of resistors and caps. This is the front.


Here is the back.


Stuffing the PCB should be done in the usual order, from smallest device to biggest - so that would be diodes and resistors first.

Of note, I got the bigger (Dale/Vishay RN60) resistors to see how they would fit on the PCB. They are the size of the PRP resistors that are quite popular amongst the fancy parts crowd. They are great everywhere except the row flanking the small transistors right in the middle. They don't fit there side by side. You could mount them soldier style, or just mix in a few smaller resistors like I did. Or just get RN55's. They are the smaller size.


Pots and transistors next. (yes, I didn't stuff the input pair when the photo was taken…)


Ah, there they are.
The ziptie is just to help their thermal tracking.


And finally the capacitors.


There is a top and bottom to the Universal Mounting Spec holes, the board mounts as shown


I find it helpful to bend the leads of the transistors first, and mount them (a little bit loose) to the heatsink.


Like this


And then mount the PCB. You can snug all the screws down and then solder and trim.

Ok, now lets move on to the Power Supply.


Here is a photo of the PSU board, I am going to use integrated bridge rectifier blocks, so you need to remove the part of the PCB that mounts the diodes. The new PCB, not quite yet available at the time of this writing, will have a similar feature with the diodes, as well as room for more/bigger capacitors.


As always, stuff the small components first - the light blue resistors are the filter resistors, the darker ones with the teflon are the bleeder resistors, and the small ones are for the LEDs.

This PSU board is the exact same DIYaudio PSU board, just without the top blue soldermask.


This shows the INPUT edge (from the diode bridges)

The capacitors are Panasonic T-UP 33,000uf 35V


Connecting the bridges to the PCB




This is the OUTPUT edge of the PSU - the colors are
Red V+
White GND
Green V-
The black connects the PSU GND to the CL-60 to the chassis.


The wiring from the PSU to the amp PCB is clearly shown.


Here you can see the bridges with the wires attached from the transformer secondary. Remember that the green attached to a bridge must have continuity with the blue attached to the same bridge. (As it's the 2 ends of the same piece of wire)



As long as were are tailing about the transformer, here is a photo of the terminal block shown wired for 120v. The Blur lead is the AC Live, and the clear the AC neutral. The reds and blacks are the transformer primaries.

Transformer is an Antek 400VA 18v+18v, part number AN-4218.



The last bit of the PSU wiring is the chassis connection, the black comes from the PSU GND, and the green is the AC safety earth.

The AC to primary wiring confuses everybody, so;

Let's look at the PSU schematic just to make sure everything is OK... Remember that I am wiring it for 120v operation, so the transformer primaries are in parallel. People wiring for 240 with a transformer like this, please ignore.



Notes in red are mine.

Look at the connections of the transformer primary, through the thermistors and line cap, to the mains.

Hot AC is connected to the "120" (which in my case is the red leads on the primaries) One red primary is connected to AC hot through a thermistor.

Neutral AC is connected to the black "0" leads, one of which is connected to the AC through a thermistor.

AC Hot and Neutral have a cap across the leads.


So, yes, the AC will be connected to the center 2 posts, which is across the cap.


(This photo lifted from my F5 thread, but it's the same PSU…)

Left to right we have post 1, 2, 3, 4

POST 1 - Transformer primary 'B 0' which will be connected to AC Neutral at post 2, through the thermistor between post 1 and 2.

POST 2 - AC Neutral in (not shown in photo), connected to Transformer primary 'A 0" , a thermistor to post 1, and a line cap to post 3

POST 3 - AC Hot in, connected to transformer primary "B 120", thermistor to post 4, and line cap to post 2

POST 4 - Transformer primary "A 120", connected to AC Hot through the thermistor to post 3

If you look at the red and black wires in the photo you will see that the Mains AC must to pass through a thermistor to connect to each of the 2 primaries. And that is the point of them, to keep inrush under control during powerup.

Ok!

Now we need to put everything together -

But first a bit more mechanical assembly.


This is the pre-cut back plate and the thick front plate.


Gather and mount the IEC plug.




The speaker posts.




And the RCA jacks.

Note that the shoulder washer goes on the inside, so the metal of the chassis doesn't touch the metal of the jack. There is a similar washer on the speaker posts.




The inside of the back panel.


And the outside. Looks good, yes?


The IEC module is wired as shown. This will switch both the Live and Neutral. The blue (live) and clear (neutral) go the the wiring block with the thermistors, cap and transformer primaries.


The amp PCB completely wired.


A bit closer.


The top connections labeled.


This happens to be the other channel, but the connections are all the same.

Remember that V- / GND / V+ is always left to right as you are looking at the format of the PCB


I'm not entirely sure what I was trying to show here, other than the screw and washer. It looks cool. I will keep the photo in the guide.





A few notes on bias -
P1 controls the bias, measured across any of the 3W source resistors. Adjust for 0.13v when it's cold, and watch that it doesn't get higher than 0.2v once it's up to temperature in about an hour. Adjust for 0.2v when hot.

P2 is used to adjust the DC offset on the output to zero.


Attach a DC voltmeter across the speaker outputs to measure offset.




If you find that the P1 doesn't have enough range, I.E., you can't turn it up enough, replace R9 with a smaller resistor, I used 4.75K and it works well.



Connect a voltmeter across any of the source resistors. The outboard ones are easier to clip across.

Adjust for about 0.13v cold, and once the amp is up to operating temperature, trim for 0.20v - It takes a long time to warm up, take your time.

Adjust P2 for zero offset, then re-trim P1

Here is a photo of it all connected and working -



I'm driving the F4 with an O2 Headphone amp sourced from an iPod; Driving 85.5db speakers. Although it is a small room, it gets louder than I want to listen. It still can't drive it to clipping, but it does get really, really loud.

One thing worth mentioning, and it speaks very highly to the quality of the amp, is that it is completely non-fatiguing, and more interestingly, very easy to listen to turned up too loud… I don't realize how loud it actually is sometimes. Complete transparency is a word used a lot when describing this amp - but I have to agree. It's fantastic!





Please comment away if you desire.

Also please feel free to ask any F4 questions here, and if you would like to post photos of your F4 completions, old or new, please do!

dear all,

Please allow me to anounce:

http://www.tentlabs.com/Components/Tubeamp/Tubeheater/index.html

also the backgrounds might be of interest

http://www.tentlabs.com/InfoSupport/page35/files/Heatingmethods.pdf

enjoy

Guido

I bought A old Fisher MT-6330 DD turntable, I would like to bring it back to spec performance. At best is has a peak 2S wow at 0.093% giving a DIN W&F at 0.037%, but more often the peak is 0.14 and DIN 0.05 . It is very sensitive to user adjusting speed pots, and speed oscillates several seconds if platter is disturbed.
Almost like the servo gain is too high .

spec is 0.035% presuming DIN as source is the german manual.
I can find settings for reference voltage and DC balance, but now setting for servo gain.

I am going to clean potmerers for speed setting both internally and the user speed pots, and set the motor drive DC balance according to service manual .
But What about adjusting the servo gain when there is no variable resistor for i?
What else can I do?

I have a rather large stock of NOS MOSFETs. New and used.

New are $20/pcs, used $15/pcs.

Ships from Sweden.

Thanks for looking.

R Malmin

I have a 8 channel active setup. When I want to take a sweep in Rew I only have the option to playback one channel at a time. output is 1 to 8 but I want to measure the response on all of them.

I use Audiolense to create correction files. Playback in Roon.

CEC (C.E.C) claim by their mostly amplifier models, it's class A (here in Germany even "Pure Class A") The name of circuit topology is "LEF (Load Effect Free). The developer is Mr. Carlos Candeias. Have a look to this URLs: Amp 3300R:
CEC AMP3300R
and Amp 5300R (Amp5300R Amp 5300R Amp5300)
CEC EUROPE WEBSITE
The last model delivers 120W/8 ohms and 135W/4 ohms, as to read in the CEC datasheet
If anybody can posted the schematic of one of both models, I will make an evaluation. I guess, the term "Pure Class A" is concerning the pre-driver stage and driver stage of a push pull complementary CFP resp. Sziklai pair, but I don't know this exactly.
If there really pure Class A by the output power devices, where is a heatsink with the appropriate sizes?
Thank you very much for your posted schematics in advance

Hi everyone,


I’m wondering if anyone could help me identify the capacitors used in the Canton Fonum PC 210 speakers.
Unfortunately, I can’t read all the values on the original capacitors, and I haven’t been able to find any information online about the crossover specifications or component values.


Any help, photos, or schematics would be greatly appreciated!


Thanks in advance!

Hi everyone! I'm passionate about vintage audio and enjoy modifying and upgrading old amplifiers to achieve a rich, warm analog sound. My current setup includes a JVC RX-308 amplifier upgraded with OPA2134PA opamps and a Yyaudio P-037 power cable. I'm running Canton Fonum PC 210 sealed 3-way speakers, which I’ve internally rewired with Neotech STDSPCT-22 silver-plated OFC copper cables using lead-free solder with 3.7% silver content. External speaker wiring is done with 2×2 meters of KáCsa KCE-LS25 cable. For vinyl playback, I use a JVC AL-A151 turntable with an Audio-Technica AT85EP cartridge, an acrylic mat, and Sommer Cable RCA interconnects. I'm always striving to extract the most musical, natural and analog character from these classic components.

I have two Thor subs, each with its analogue crossover. With peerless xls 12" drivers. At present crossover set at 50hz I think. I am not guaranteeing the crossover is working, but they were working when made up.

I've never used these, and so they need to go to someone who will.

Each individual subwoofer with its crossover £150.

Collection from Guildford Surrey. No posting for obvious reasons... 🙂



Thanks.

My Topping DAC D50 does not power on. I tried different USB charger 5V-1.0A and 2.0A.

Did anyone experience the same problem?

I hear it "click" on, but indicator light (display) does not light up. Has anyone any ideas how to get it working?

Thanks.

Hi everyone,

I’m completely new to the world of hi-fi and DIY audio, and I’m really excited to start this journey. I’ve recently developed a fascination with high-quality sound and have been exploring.

While I don’t have a technical background, I’m keen to learn — whether it’s about amps, speakers, or the basics of audio electronics. I joined diyaudio.com to connect with others, ask lots of (probably beginner) questions, and slowly build my understanding of this amazing hobby.


Looking forward to learning from all of you and sharing my progress as I go. Thanks for having me here!

Has anyone gotten a Raspberry Pi 5 to work successfully as a DSP crossover music streamier?

I have a working DSP crossover streamer configuration on my Raspberry Pi 2B using Charlie Laub's ACDf filters. The OS is Bullseye 5.10.103-v7+. The 2-channel (stereo) audio stream from mplayer (using -ao alsa) is successfully split into multiple channels and processed by the filters in /etc/asound.conf and this outputs multi-channel PCM 96khz to my AV receiver via HDMI and the resulting sound quality is excellent in my 2.5 way speaker setup. The tweeters go tweet, the squawkers go squawk and the woofers go woof!.

With my new Raspberry Pi 5 running Bookworm 6.6.20+rpt-rpi-27172 I have not been as successful. With sudo raspi-config I have selected 1 PulseAudio (only other choice is 2 Pipewire) and audio output 0 vc4-hdmi-0. On the GUI desktop, I have the options for Stereo, 5.1 Surround, and 7.1 Surround. I have installed the ACDf filters in the (I believe) correct folders /usr/lib/ladspa and for good measure also /usr/local/lib/ladspa, and I have alsa configuration in /etc/asound.conf.

With 7.1 Surround selected on the Desktop:

In terminal
$ speaker-test -t wav -Dpulse -c 8

speaker-test 1.2.8

Playback device is pulse
Stream parameters are 48000Hz, S16_LE, 8 channels
WAV file(s)
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 24 to 262144
Period size range from 8 to 87382
Using max buffer size 262144
Periods = 4
was set period_size = 65536
was set buffer_size = 262144
0 - Front Left
4 - Center
giving sequential 8 channel audio output Front Left, Front Center, etc. to all 6 drivers in my system. The audio is mismatched with the drivers, but I had the same issue on the Raspberry Pi2 and corrected everything using the t-table in /etc/asound.conf.

In terminal
$ sudo speaker-test -t wav -Dpulse - c 8

speaker-test 1.2.8

Playback device is pulse
Stream parameters are 48000Hz, S16_LE, 1 channels
WAV file(s)
ALSA lib pulse.c:242🙁pulse_connect) PulseAudio: Unable to connect: Connection refused

Playback open error: -111,Connection refused

In terminal
$ sudo speaker-test -t wav -Ddefault -c 8

speaker-test 1.2.8

Playback device is default
Stream parameters are 48000Hz, S16_LE, 8 channels
WAV file(s)
ALSA lib pcm_params.c:2226🙁snd1_pcm_hw_refine_slave) Slave PCM not usable
ALSA lib pcm_params.c:2226🙁snd1_pcm_hw_refine_slave) Slave PCM not usable
Broken configuration for playback: no configurations available: Invalid argument
Setting of hwparams failed: Invalid argument

Using the sudo causes the action to fail.

mplayer (using -ao alsa) now gives audio output to Front Left and Front Right channels only at 44.1khz which is the same 44.1khz as the internet radio stream indicating the resampling of /etc/asound.conf is being ignored.

mplayer (using -ao pulse) gives audio output to all channels at 44.1khz which indicates the resampling of /etc/asound.conf is being ignored. This is confirmed by turning off all the outputs in the t-table.

aplay -l reports the following.
** List of PLAYBACK Hardware Devices **
card 0: vc4hdmi0 [vc4-hdmi-0], device 0: MAI PCM i2s-hifi-0 [MAI PCM i2s-hifi-0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: vc4hdmi1 [vc4-hdmi-1], device 0: MAI PCM i2s-hifi-0 [MAI PCM i2s-hifi-0]
Subdevices: 1/1
Subdevice #0: subdevice #0

I know that PulseAudio is looking at the configurationi in /etc/asound.conf because I can comment out a formatting character such as #} and this will throw an error with mplayer. It just appears that the filters are being ignored by PulseAudio. Maybe the Pi5 OS being 64bit and the filters were written for 32bit has something to do with it?

The Micro-Audio SMPS600-R2 is a high-quality, unregulated 600W switching power supply designed for audio applications. It offers a single-rail output of 46Vdc. SMPS was bought few months ago and it is in perfect working condition. Dimensions are 187x123x50 mm and weight is around 800g.

This SMPS is designed to power various TPA3255 amplifiers, like 3e modules and others. It provides significant sonic upgrade, compare to cheap SMPS.

I paid around 200 EUR (all shipping costs and custom fees included) to import it to Slovenia (EU). My price is 125 EUR + shipping + PayPal fee.

In recent exchange (by PM) I got asked if it's possible to do a tower version of the Open Source Monkey Coffin (OSMC). My gut feeling is that it shouldn't be too difficult to modify the OSMC design into a Open Source Monkey Tower. I am therefore starting this new thread to discuss the tower idea.

Sad to hear about Rick's passing. He designed some speakers back in 2007 for me. I was wondering if anyone knows who still has access or owns his designs? I'm looking to build some speakers (RS180 + 27TDFC) that he had designed that tonally match the ones he designed for me (RS150 + Peerless HDS). Does anyone still sell his designs or have access to them?

For sale a very rare pair of Titanium JBL tweeters perfect working condition.
Never been repaired.
250 Euros+shipping.

Came in with blown power supply. Replaced PS fets, gate resistors and ucc27524 driver.

When I apply power, no remote, the ucc27524 shorts its outputs(pins 5 & 7) to ground with 5.4 ohms resistance. This also happens with the fets removed. I tried another ucc27524 and no change. Parts are from mouser and work in another amp I have here with the same driver(stetsom ex6000eq).

With the ucc27524 removed I can see drive signal on both input pins(2 & 4). I get 12volts on the vdd (pin 6) and ground on pin 3.

After a few minutes disconnected from power the short goes away.

I am at a loss.

Is there anything fundamentally better about a discrete r2r DAC as opposed to an IC?

I need schematic diagram of Unison Research Unico to take pre-out amp.
Please help me?

Dear friends!
I am assembling a headphone amplifier. I have a question: how to correctly calculate the value of the potentiometer RV1? I heard that it should be about 10 times less than the input resistance of the amplifier. In my case, the input resistance is determined by the value of R12 and is equal to 33k. Then I need a potentiometer of 3.3k? But everywhere where I studied the circuits of similar amplifiers, they recommend a potentiometer value from 10k to 100k.
How to choose correctly?

PS. Input signal goes from PCM2704 DAC (recommended output impendance 10k or more).

I just downloaded a file from there; it's the first time is about a year I've been able to. Might want to check yours as well?

Anyone use their leftover / extra Car Audio stuff for home HiFi? One would think - by now - the name brand car audio stuff would have specs that far exceed what is considered good for noise, distortion, frequency response et al. 12V power shouldnt be a mountain to overcome...

I picked up a Rockford Fosgate R-150 / 2 at a gamble price; the blue light came on when I powered it up. Sounds good to my ears in my garage system. I'm going to try a FRFR for guitar / vocal with it, with two FR speakers and amp / cab models upstream.

Hello all,

Here are some Triplett VU Meters from an MCI tape machine

Fits 5000 series consoles and JH-100, JH16 and JH-24 open reel tape machines

In excellent condition & Working perfectly

Genuine & original part dates from late 70's- early 80's

Top quality meter, Made in USA

Includes built-in diode bridge for AC operation

Meter measures 70.3mm x 60.9mm x 46.7mm deep excluding terminating posts

Complete with bezel 79.3mm x 43.6mm and mounting nuts

But without the tabs to secure the bezel and meter assembly to a panel

Requires a 75.4mm x 39.8mm panel cutout for the bezel

Perfect for MCI replacement or other vintage gear restoration

8x of these meters and bezels are available

I'm thinking USD 45 each plus postage or open to offers for multiples

Cheers, Ralph

https://audiofile.net.au/

Does anyone know where I might find these or equivalent.
The amplifier is a Sundown Salt-6

Can TIP32C or MJE3055T be used for B631K?
and KTD600K-Y or 2SD600K) for D600K

What would be a good combo?

Thanks

Contracted manufacturer to install 1.0mm pin sockets in Magnoval bases. Tested on 12GT5 Novar tubes.
Money back guarantee if not satisfied with fit. Background link below.
$2.50 each + ship to US, Canada.
Thanks,
Jim

https://www.diyaudio.com/community/threads/real-novar-sockets-inside.397358/#post-7304388

A friend recently lent me a Dayton DATS LA to test. Dayton advertises that the DATS LA is capable of performing small- and large-signal analyses. This review focuses exclusively on large-signal analysis, or rather, the part of the program known as " Symmetry Test." Further details and the associated patent can be found on the Dayton website (https://www.daytonaudio.com/product/2090/dats-la-loudspeaker-analyzer). The patent—probably not entirely unintentionally—suggests a similarity to the Klippel analyzer. However, that would set the bar quite high.

Before we get into the measurement results, a little more information about the Dayton DATS LA. I believe the measurement principle was already introduced in 1992 by Blind, Phillips, and Geddes at the 93rd AES Convention, although they used a different measurement principle (DUMAX) in their publication.
Fig. 1

The patent specification, published on the Dayton Audio website, also refers to this source. According to the patent specification, the loudspeaker is to be excited with a very low-frequency alternating voltage and a superimposed 0.7-second sweep.
Fig. 2

The real measurement signal or a measurement run looks like this.
Fig. 3
Fig. 4

The operating point is set using a modified "square-wave signal" of approximately 0.33 Hz, and the impedance at the operating point is then determined using a 0.7-second sweep. Obviously, the measurement is not performed alternating between cone in and cone out as shown in Fig. 6, but rather continuously decreasing in 19 predefined steps, resulting in 19 impedance curves , as shown in the following figure.

Fig. 5

It is inevitable that the voice coil will heat up when exposed to direct current. DATS LA attempts to reduce this undesirable effect during measurements by varying pause or cooling times. For example, a measurement cycle with ± 18V DC takes approximately twice as long as with ± 9V DC due to the longer pause times.

Fig. 6: Wavecor WF182BD10 (9.0V DC)
Fig. 7: Faital 12PR320 (18.0V DC)

The following shows the effect of applying DC voltage to the voice coil on Re and the voice coil temperature. The measurement cycle is clearly visible in the progression.

Fig. 8: Wavecor WF182BD10

Fig. 9: Faital 12PR320

The picture of a measurement run also shows that the cone does not return to the zero position after each application of DC voltage, but rather an offset slowly builds up. Here's an evaluation example for the Faital 12PR320:
Fig. 10
Since only electrical measurements can be performed with the DATS LA, the cone excursion X-DC (red marked) must be calculated from the following data set:


Tab. 1

In the above-mentioned frequency range of 0.33 Hz, the following should apply in the linear range to a good approximation (Beranek & Mellow , Acoustics, page 286):

X(U) = BL(U) * U/Re(U) * Cms ( 0)

However, I still haven't figured out how Dayton DATS LA calculates the excursion in the nonlinear range. Unfortunately, my attempts at correction are only applicable to specific cases.

Below is a first measured example (Visaton AL130). The blue curve shows the DATS LA result for the excursion, while the red dashed curve shows the result calculated using the formula above. In the linear range, it fits quite well, as expected...
Fig. 11
Now let's do it all again with the excursion measured using a triangulation laser (dotted black line). The DATS result isn't quite identical, but it's pretty close.

Fig. 12
Is this the accuracy the system can offer in terms of displacement estimation? More on that later.

What impact do these inaccuracies in the excursion determination have on the parameter analysis? The following analysis is again from the Visaton AL130 (blue = DATS, red = laser-corrected). The linear excursion X(BL) is ± 6.1 mm for the DATS LA and ± 5.4 mm for the laser-corrected version.

Fig. 13
Since a full set of Klippel data is available for the AL130 used here, a direct comparison of the BL(x) analysis between the different measurement methods is a good option.

The following image shows the results of three - or four - measurement methods (red = Klippel, blue long dashed = DUMAX, green short dashed = DATS LA, black dotted = DATS LA laser-corrected). At least for the AL130, the results lie within a fairly narrow scatter band despite the very different measurement methods.

Fig. 14

To verify these results, two more examples are now shown ( Faital 12PR320, Wavecor WF182BD10).

Fig. 15: Faital 12PR320
Fig. 16: Wavecor WF182BD10


Obviously, the quality of the DATS LA excursion estimate in the first example cannot be transferred to other loudspeakers. Especially with the 12PR320, but also with the WF182BD10, there are significant discrepancies between the excursion estimate and the measurement. To illustrate the effects of these differences, the corresponding BL(x) diagrams are shown below (blue line = DATS LA, red line = laser-corrected).

Fig. 17: 12PR320

Fig. 18: WF182BD10

DATS LA shows ± 4.80 mm for the linear excursion X(BL,12PR320) and ± 5.95 mm for the laser-corrected variant, or ± 6.75 mm for X( BL,WF 182BD10) and ± 5.90 mm for the laser-corrected variant.

These deviations in the excursion determination naturally have an impact not only on the BL(x) curves but also on the other results such as Kms (x), Le(x), etc.

To further support the previous results, the X(BL) evaluation for all 22 measured loudspeakers is shown below:


Tab. 2

The difference between the DATS estimate and the laser-corrected measurement averages 16%, with a single standard deviation of ± 20%. This is probably not quite up to the quality standards of the Klippel Analyzer.

How do we evaluate these results?

Regardless of how relevant one considers the deviations to be, if one never knows how close the excursion calculated by DATS LA is to the measured reality without additional laser measuring equipment, there will always remain a feeling of uncertainty regarding the reliability of the measured data and this will inevitably lead to dissatisfaction in the long run.

Regards
Heinrich

This are my assumptions, and i am not an audio engineer.

This midrange is not designed for bass even if has lower fs. 30hz
The low fs come from the very lose suspension, believe it or not a loose suspension means eficiency, in a way, because it lowers the qes! Also making it good for ported, making it even more efficient.
Thats why they have such low qes qts.
This means that the coil doesnt have to be denser to achieve lower qes, instead the coil can have thinner wires to achieve a bigger bl (more total coil in the gap) better motor force controll. And with the loose suspension mechanical looses are very low.
This gives it very low distortions

Have a PPI A300 amp connected to the front speakers. I am noticing some slight static when the car is off. When the car is on, there is some slight wining as well. I ran both neg/pos 4 Gauge straight from the battery and have made good connections on both ends.

Can the amp itself be the issue? It seems the input is quite sensitive as well as I have a Xover 2XS in front of the input which I had to turn the gain almost all the way down on it or the volume is very loud at low volume levels on the head unit.

Thanks

Hiya,

Looking at SMD components to minimize area on PCBs, but kind of lost in what components to use.

In particular I want to add some caps/ferrites to stop noise/oscillations and wanted to know if I should go for PPS caps for the 10nF/100nF, and what to look for for ferrites. In through-hole components I am using MKP for the caps, ferrite beads on some wire and 150R tantanlum resistors but this takes up more space than I would like (even when using resistors/ferrites vertically.

Not sure what to look for in SMD equivalents. Anyone with experience in SMD components?

Working perfectly and in excellent cosmetic condition .

Bought to compare in my speakers

£60 + shipping ( I’m in the UK )

Who knows the exactly name of very high voltage booster capacitor manufacturer to the logo "db" (see attachement) ?