Copied from an online ad for what I thought was one of the worlds most respectable amplifier manufactures

unique three-point support foot "Stress-less Foot" is employed to enable stable installation without being affected by even the slightest distortion of the floor surface. Unlike conventional foot, the Stress-less foot is designed not to fix to the chassis directly, allowing it to vibrate freely to achieve a more natural sound.
The foot, made of machined steel, is not firmly fixed, but is mounted hanging from the bottom, enabling a more natural and rich sound with a sense of sound source deployment similar to that of a conventional pinpoint spike foot.

Yes they are talking about the amplifier not speakers

Opinions???🙂

Hoping I'm not opening a Pandora's box but here goes 😛

I'm still in the process of designing my Rega Elicit phono clone and been wondering about this. In one of my books by a reputable author, he describes how to achieve perfect network component values for the resistors and capacitors by paralleling each component with a triming one, with the requirement that the ratio of the main and trim values be greater than the ratio of their tolerances. Of course the value measurements have to be made with a precision bridge. This is particularly helpful if one has come up with "awkward" values outside of the standard series, which is not my case.

I don't have a bridge and honestly I think it might be overkill to do this. Reason is, one of the components of the network is the output impedance of the gain stage which I believe will have worse tolerance than the 0.1% resistors and 1% capacitors the author suggests unless output transistors and their associated circuit components are also selected to be within these tolerances. And I seriously doubt manufacturers go to these lengths.

But out of curiosity I simulated my design with 1% variation factors for the passive components of both passive and active eq filters; see frequency response output plot below for 27 seven different combinations. Yes the frequency response is less flat but the maximum deviation from the theorical values is on the order of 0.3dB peak-peak for the extremes. Can this difference really be heard?



Just wondering if I need to go to the trouble of adding space for extra trim components on my PCB. I might do it for resistors as this is not that complicated but for the polystyrene caps I'm not sure as there are large variations in footprints of the parts I can get my hands on off ebay.

Thanks in advance for any insights.
-Joris

Just trying to understand what would happen to a Pre-amp or a Dac which have a short at their output RCA jacks due to a stray wire or bad solder. Would the circuit in the preamp/Dac get burnt or just that there will be no output at all from them? Also is the power amp or a powered speaker at risk if connected to those outputs ?

Thanks

After finalizing the overall design, simulations are used to select specific components and determine resistor and capacitor values, followed by circuit board manufacturing for validation and iteration to achieve the initial design goals.

The Σ-Δ (Sigma-Delta) modulation method was chosen for the following reasons:

1，A fixed-frequency PWM modulator has almost zero Power Supply Rejection Ratio(PSRR) in its output stage, meaning it has no ability to suppress power noise. This creates significant challenges for power supply design, as any noise from the power supply directly converts into audio noise and reaches the headphones. In contrast, the Σ-Δ modulator offers some level of power supply noise rejection, greatly reducing the burden on power supply design.

2，Fixed-frequency PWM modulators use a standard triangular wave (blue waveform in Figure 1(a)), which contains a lot of high-frequency components. However, the integrator-generated waveforms from multi-stage integrators (green waveform in Figure 1(a)) resemble a triangular shape but with fewer high-frequency components and faster high-frequency roll-off (as shown in Figure 1(b), with blue representing the standard triangular wave's spectrum and green representing the multi-stage integrator waveform's spectrum). In an amplifier with the same bandwidth, the multi-stage integrator waveform results in lower distortion, and the PDM modulation signal's edge jitter (rise and fall times) is smaller, leading to overall lower noise and distortion of audio signal.


(a) (b)
Figure 1, Triangle Wave, Higher-Order Integral Wave, and Their Spectrum

3，The Σ-Δ modulator can use a differential integrator circuit, which outputs a pair of equal-amplitude, opposite-polarity integrator waveforms (see Figure 2(b)) to the comparator. In contrast, the fixed-frequency PWM modulator sends a triangular wave and input signal to the comparator (Figure 2(a) shows the comparator input waveform of the fixed-frequency PWM modulator). It's clear that the slope (rate of change) at the intersection points of these two sets of input waveforms differs significantly. The differential integrator waveforms have a much steeper slope at the intersection points compared to the PWM modulator’s waveforms. This means that for the same noise level, the edge jitter of the comparator output is smaller, thus improving the overall amplifier performance. Attentive readers might have figured out that the slope at the signal crossover points of the higher-order integrator waveform should be much steeper than that of the standard triangular wave, which could be one of the reasons why the higher-order integrator performs better than the standard triangular wave.

(a) (b)
Figure 2， Comparator Input Signals of PWM and PDM Modulators

4，Compared to fixed-frequency PWM modulators, Σ-Δ modulators offer greater flexibility and room for improvement in design. For example, a Schmitt trigger mechanism can be added to the comparator in a Σ-Δ modulator to improve the stability of the PDM pulse waveform edges.

The discussion so far has focused on reducing edge jitter in the PDM pulse signal. In Class D amplifiers, the pulse signal is central to the entire system; only precise pulses can result in low noise and low THD distortion. The edges of these pulses are crucial. If edge jitter is unrelated to the input signal, it generates noise, which, while having a smaller effect on distortion, can increase the amplifier’s background noise. If edge jitter is related to the input signal—such as when the triangular wave lacks linearity—the jitter (early or late edges) will correlate with the input signal amplitude, causing nonlinear distortion and resulting in higher THD.

Through the improvements discussed above, Σ-Δ modulation can potentially achieve lower distortion. The next step is to compare performance through simulation to select different components and parameters. This is a complex and tedious process, as components interact with each other, and changes to one typically require adjustments to related parameters. I'll use a key example to explain this further.

The precision of the modulated pulse signal comes from the accuracy of the multi-order integrator signals. The amplitude and frequency of these integrator waveforms are affected by many factors, such as the zero and pole distribution in the integrator network, the depth of the Schmitt trigger in the comparator, and the oscillation control resistor. For simplicity, Figure 3 shows a basic schematic of a multi-order half-bridge Σ-Δ modulator with a Schmitt trigger. The ratio of the Schmitt trigger resistor influences the modulation frequency, and the equivalent capacitance of the high-order integrator also affects the modulation frequency. These factors further shape the integrator waveform and its amplitude.

The accuracy of the integrator waveform has already been analyzed, but its amplitude is equally important, requiring a balance between the output capability of the integrator and the performance of the comparator. In general, a larger signal fed into the comparator results in more precise outputs, but an overly large output from the integrator can cause signal distortion.


Figure 3， Multistage Half-Bridge Sigma-Delta Modulator with Schmitt Trigger Mechanism

The second important aspect is the selection of the output stage, where the industry has reached a consensus: when cost requirements are not stringent, a full-bridge configuration is typically used. The advantages of the full-bridge mode include the ability to partially cancel even-order harmonics and suppress the back current of the output filter inductor, effectively reducing the power supply fluctuations in the output stage. This is particularly beneficial for Class D amplifiers with poor power supply rejection ratio (PSRR).

The output filter is also critical, especially for Class D amplifiers using PDM modulation. Since this design uses a modulation frequency above 2 MHz, there is more flexibility in choosing the low-pass filter’s cutoff frequency. This brings significant benefits: first, a higher cutoff frequency can be chosen, so the frequency response changes with different loads won't significantly impact the amplifier’s effective bandwidth. Second, since smaller inductors are used, they provide better linearity at the same size, reducing the distortion impact on the amplifier. During simulation, it was also observed that a multi-order output filter could reduce in-band noise, although there is no theoretical explanation for this yet. I welcome any interested parties to discuss this further.

The power supply solution is equally important. This design uses a single 3000mAh lithium battery, so a step-up boost converter is required. A major issue is that the amplifier operates at over 2 MHz, which is much higher than the frequency of commonly available boost converters. During a single PWM cycle, the supply voltage may drop due to the amplifier’s high current draw, which even the best boost chips struggle to handle in such cases.

After multiple experiments, both the pre-amp and output stages ended up using a two-stage single power supply solution. The pre-amp power (for the integrator and comparator) uses a combination of a boost circuit and a Class A power supply. The Class A supply uses a high-precision reference regulator with voltage accuracy of 0.1% and power supply noise as low as a few microvolts. Additionally, the midpoint reference voltage uses the same precision reference chip, not only improving voltage accuracy but also ensuring the integrator circuit operates symmetrically, optimizing open-loop performance.

Originally, I planned to use the same design for the output stage power supply, but due to the need to drive headphones with varying impedance and sensitivity, the Class A power supply’s efficiency was too low, which affected battery life. In the end, I chose a step-up circuit combined with a high-precision, high-current LDO and large capacitors for the output stage.

Interestingly, switching from a Class A supply to an LDO + large capacitors for the output stage resulted in virtually no subjective change in sound quality. However, changing the pre-amp from an LDO to a Class A power supply led to a significant improvement in sound. The high frequencies became more transparent, the low frequencies were stronger and more extended, and the soundstage was more stable. This change was quite noticeable, without the need for careful listening. Could this be due to the higher precision in both supply and bias voltages? I would love to hear thoughts from those interested in discussing this. The first version of the pre-amp, which was discarded, used a voltage divider for the bias reference, as shown in Figure 4, with resistor accuracy of 0.1%.


Figure 4， Circuit for Generating the Bias Voltage in the Integrator

There are many other crucial details in the amplifier design and debugging process that cannot be overlooked, and I will discuss them further in appropriate contexts.

Figure 5 shows the Spice simulation results for the background noise. Within an effective bandwidth of 35 kHz, the highest background noise is as low as -170 dBV, with the low-frequency section reaching nearly -180 dBV.

Figure 5，Zero Input Circuit Configuration (Background Noise) Simulation Results

Figure 6 shows the simulation results with a 1kHz input signal. No harmonic distortion related to the 1kHz frequency was observed. However, compared to the zero-input results in Figure 5, the background noise increased by about 10dB, reaching approximately -170dBV. This frequency-independent distortion is a typical occurrence in such circuits.

Figure 6， Single Sine Wave Input Circuit Configuration Simulation Results

Figure 7 presents the amplifier simulation results based on the previous discussions. The modulation frequency is around 2.3 MHz, with an input signal peak-to-peak value of about 196 mV and an output peak-to-peak value of around 4000 mV, resulting in an overall gain of approximately 26 dB. The input signal consists of three frequencies: 500 Hz, 5 kHz, and 20 kHz. The amplifier's background noise level is about -167 dBV, which is 3 dB higher than the -170 dBV observed in the 1 kHz sine wave test shown in Figure 6, likely due to frequency-independent distortion causing this noise floor increase.

During the simulation, it was observed that different multi-stage integrator network configurations affect the spectral characteristics of frequency-independent distortion. By adjusting the integrator network, it is possible to reduce low-frequency distortion at the cost of increasing high-frequency distortion. In this design, I chose to balance the frequency-independent distortion across the spectrum, leading to a flat noise floor.

In reality, there are other approaches. One could adjust the integrator network to minimize the total or weighted power of frequency-independent noise within the effective bandwidth, without prioritizing flatness. My choice for a flat noise floor was based on the fact that amplifiers typically exhibit more distortion in high frequencies compared to low frequencies, whether it is frequency-dependent or independent. A DIY amplifier with a flat noise floor might offer a different subjective listening experience.

Given the multi-frequency input, the amplifier's total distortion is primarily composed of harmonic and intermodulation distortion. The simulation results show that the highest harmonic distortion reaches about -156 dBV, with the minimum signal-to-harmonic ratio being 152 dB. These results demonstrate excellent noise suppression and distortion control, achieving the expected high-performance standards, at least based on the simulation outcomes.


Figure 7，Multi-Sine Wave Input Circuit Configuration Simulation Results

The following lists the different distortion frequencies and their amplitudes generated from the simulation, for reference by those interested:

• 1.5 kHz, -164.3 dBV
• 4 kHz, -159.5 dBV
• 9.5 kHz, -159.1 dBV
• 10 kHz, -160.7 dBV
• 10.5 kHz, -160.2 dBV
• 14.5 kHz, -157 dBV
• 15.5 kHz, -157.0 dBV
• 19 kHz, -157.1 dBV

Overall, the simulation results show extremely low distortion, with distortion in the high-frequency range being about 5-7 dB higher than in the low-frequency range. It's noteworthy that there is a distortion point at 10 kHz, which corresponds to the second harmonic of the 5 kHz signal. However, the distortion values at these points are so low that the specific values are not very meaningful for further analysis —they are all well beyond the range of human hearing.

For those who might not be familiar with these simulation figures and find them hard to interpret, it's worth noting that if you were to input a true 24-bit audio source into the amplifier, the output would be virtually undistorted. The harmonic distortion is more than 7 dB lower than the 24-bit quantization noise, making it effectively distortion-free.

In addition, I added negative feedback to the circuit from Figure 7 and adjusted the overall gain to 20 dB, 6 dB lower than without feedback. Observing the results in Figure 8, no frequency-dependent harmonics were observed, but the overall noise floor increased by about 1 dB compared to the no-feedback case.

This phenomenon may differ from traditional Class B amplifiers. Class D amplifiers, due to their modulation frequency, generate out-of-band noise. When negative feedback is introduced, this out-of-band noise can also be fed back to the input. While negative feedback effectively suppresses harmonic distortion caused by nonlinearity, the feedback of out-of-band noise may increase frequency-independent distortion, thus raising the overall noise floor.

Although this analysis partially explains the rise in noise floor, it has not been rigorously validated and is provided for reference only. If I gather further experimental data or validation, I will share it separately in the future.


Figure 8， Simulation Results of Multi-Sine Wave Input Circuit Scheme with Global Feedback

Additionally, when observing the operating frequency of the amplifier, it becomes apparent that as the peak value of the input signal increases, the divergence of the main frequency enhances, while the peak itself becomes lower. This phenomenon is a characteristic of the sigma-delta self-oscillation architecture, providing significant advantages in electromagnetic compatibility (EMC). Since the self-oscillation frequency is not fixed, it dynamically changes under different load and signal conditions, dispersing EMI (electromagnetic interference) energy rather than concentrating it at a specific frequency. This reduces interference with other devices, making sigma-delta amplifiers relatively superior in EMC performance in practical applications.

The following text is excerpted from "Audio Power AMP Design Handbook 4th edition" by Douglas Self, which discusses the increasing distortion characteristics of traditional Class B amplifiers at higher frequencies. Although this book was published in 2006, and advancements in technology may have significantly improved the high-frequency distortion characteristics of current amplifiers, this fundamental principle still holds unless there are breakthrough developments.



From the simulation results, Class D amplifiers also adhere to this pattern, but their high-frequency distortion is only a few dB higher than the low-frequency distortion. Given the overall low distortion rates, this difference can generally be ignored.

Another noteworthy aspect is that the amplifier's phase response tends to be linear. As we know, the human ear is not very sensitive to phase response in sound. Therefore, it raises an interesting question: do linear-phase amplifiers produce different subjective listening experiences? If anyone has personal experiences or insights on this, it would be great to share and discuss!

Based on a simple calculation of the simulated distortion effects, the amplifier's THD value can reach approximately 0.0000025% with the minimal SNR is 152dB. Considering that the harmonic components are several dB above the noise floor, the THD+N value should be in the same range. This simulation result seems almost unrealistically good, so I conducted multiple verification simulations and cross-checked with actual test results, ultimately confirming that the simulation results have a certain degree of credibility. The verification approach was as follows:

1. Device SPICE Model Verification: Each device's SPICE model was individually verified by building its recommended test circuit using the model and comparing the results with those provided by the manufacturer. The simulation results matched the manufacturer's data closely, confirming the accuracy of the models used.
2. Comparison of Simulation and Measured Parameters: Key parameters of the fabricated circuit (such as the amplitude and shape of the integral waveform, the amplifier's oscillation frequency, and gain) were measured and compared with simulation results, with discrepancies generally kept within 10%. This directly demonstrates the reliability of the simulated circuit.

However, due to the lack of sufficiently high-precision instruments, I was unable to directly verify the noise floor and THD distortion values through measurement. This limitation makes the validation of these performance metrics in the simulation somewhat lacking, but overall, the verification results still hold reference value.

Of course, the performance results from the simulation represent only a theoretical upper limit; the actual amplifier's performance may degrade to varying degrees due to various factors. A good simulation result is merely a starting point, and the process of building the amplifier is essentially a struggle against these degradation factors.

Writing the threads has been particularly challenging for me. I’ve tried to clearly describe the simulation conditions at the time, but the reality is far more complex than what I've expressed here. If there are any unclear or incorrect points, I welcome further discussion. For now, I will share the results of my two major version of DIY AMP and some subjective listening experiences, so stay tuned!

HI all,

I'm flying into Houston for the weekend and was wondering if there are any places or members that sell vacuum tubes around there?
In particular looking for 245/345 tube pairs or singles!

SOLD, thank you.

Excellent condition pair of Tektronix P6111B 200mHz 10X probes, complete.

Shipping is included (only to CONUS).

$46 for both, shipped, including PP G&S fee, or $44 if paid by F&F.

Thank you.

I’m new to DIY speakers but have a good understanding of basic electronics and I’d say I’m an advanced DIY’r with all the necessary tools and equipment to build a very nice system. I just don’t know how to design the speakers for the sound I’m looking for. I’ll try to explain.

At the beginning of Sarah McLachlan’s “Angel”, on one of the best systems I’ve heard (the one I’d like to build) the grand piano gives a massive “heart thumping” feeling. The remake of Sound of Silence by Disturbed is another song that my system can’t seem to “dig” into enough.

I have Kef R300s and two large subs (SVS PB-13 Ultra and Definitive Tech Supercube Reference 111lbs ). These are run with a Denon AVR-X3800H in my home theatre/music room. I also have a KEF R200C with R800DS surrounds some DT surrounds all running in 7.2.4 for now. The room has also been treated with 6 very large hanging panels and a large floor rug.

But no matter what I do I can’t seem to tune my system to give me that intense grand piano bass. Thomas and Stereo (the YouTuber) describes his Earthquake Sound Tigris here as doing exactly that.

How would I go about building and designing a set of towers to give me this “sound” that would make it different than any other pair? Here’s where I have zero experience but it seems this frequency I’m looking for is around 300-400hz?

Is it the cabinets? Woofer size? Crossover?

I understand no one is going to be able to answer this with a single simple response, I get that it’s a combination of the parts. Maybe that’s what I’m asking, what combination of parts is more likely to give me this finished product that can give results? What general direction should I be looking? Or is it even a realistic expectation?

Thanks all in advance!

For the capacitor, I have both 0.1uF in ceramic and polypro. Can I use either of them? If yes, which one is preferrable?

Thanks in advance.

I'm selling my 6 channel Parasound HCA-806A if anyone is interested. This model has an issue with the output relays, I replaced all 6 when I bought it a few years ago. I also replaced and upgraded the power supply caps. I'm asking $375. It's a well built amp weighing 50lbs. I have it listed on eBay and Audiogon, but if anyone is interested here, just PM me. 4 channels are bridgeable, essentially turning it into a 4 channel amp. I used it to power my mains and center. The center I bi-amped. I replaced it with a Proceed HCA-3 amp. Which was made by Mark Levinson, so that's why I'm not using it. I actually don't want to sell it, but my business has slowed down, so I'm planning on using the funds to build an Ian Canada streamer.


http://www.audioreview.com/product/amplification/amplifiers/parasound/hca-806.html
https://www.diyaudio.com/community/threads/fs-parasound-hca-806-6ch-x-80w-sf-bay-area-ca.191917/

https://www.bigazspeakers.com/produ...high-current-amplifier-80-120-180-wpc-bridged

I'm taking offers also...it is a used amp. This is also the "A" version,. which includes a remote turn on for your pre-amp, which I will include.
Here's the listing on Audiogon, so I don't have to post pictures here. Cosmetically it is in very good condition. It came with the "rack mount" face plate, but if anyone wants it and doesn't like the rack mount, I do paint and body and I can cutoff the "rack portion" and blend it with OEM blackout finish. The only issue with that is it may not be a "perfect" blend because I would have to stay away from the "Parasound' emblem, but in natural lighting inside, you'd never see it. I'm not cutting it off unless someone asks me too however.

Audiogon listing...

https://www.audiogon.com/listings/lisbb646-parasound-hca-806a-multi-channel-amps

Hi there,

I am selling a new power transformer type Antek 800VA 45V ( AN-8445) and unused boards for the DIY 150W class AB amplifier "Honey Badger".
Too many projects waiting, and I just gave up on this one.

Original cost was $39 for the PCBs and $98 for the PT.
I am asking for $80 and shipping for 18lbs from 06851.

Ultralow distortion LM3886 chipamp

Hi everyone! It is my first message on this forum and I hope it will be interesting for you.
So, my power amp is based on LM3886. It has a very deep feedback - about 70dB at 20kHz, and much more on lower freq. It considerably reduces distortions compare to the standard schematic. I started this project almost one year ago, and now I have a very good working one. It has about -120dB THD at 11kHz, 4Ohms, 60W. IMD 19+20 is less, and I could not measure it.
Here is the schematic, parts list, PCB layout.
Some measurements.
That's it for today, I hope I'll post more soon.
Nick.

I am a professional electronics repairer. I actually did it last time in the early 90s. Currently, I collect Technics equipment and repair them if necessary. But with CD players, your knowledge and help is needed because I don't have much of it myself.

I am looking to build a 3 way pa top with 60x40 dispersion so two cabs can be splayed side by side. I already have a few ME60 horns and some 15” drivers from a couple old carvin Trx 215n and friend gave me. Basically I would get two more ME60 horns and build two more cabs to split the woofers between a them and use an 8” midrange with a wave guide. I cannot find much information for designing a waveguide. Is it as simple as angling the sides out from the cone at the desired angle or is there more to it such as the length of the guide relative to the bandwidth?

Recently i made a simple common emitter stage on a breadboard that was an altered version of this amplifier from "John audio tech" with and without the constant current source,
the version with the constant current source made around 1.1W of clean power into 8 ohms, and the version without the constant current source made around 0.45W.

The alternations included an additional constant current source version, diffrent resistor values and an single mje3055 npn instead of an darlington, i was really suprised and wanted to expand on this idea and make it higher power and more practical but i soon realised that the bjt common emitter stage is pretty limited and inpractical for anyting highrt power so i made this but now im having trouble with integrating a gain stage into this circuit.

Hi to everyone.
Classic material is a MDF.
Also HDF, and their variations and types, depending on the type of wood, glues, additional materials used.
Plywood is also increasingly popular. Baltic birch is a classic, but lately I've been reading about bamboo plywood. There are also different varieties.
Artificial stone and its extensive proprietary variations. Many praise, but still not so popular.
Kevlar concrete. I met it rarely, I'm not sure that it is more effective than the previous ones, I don't know what was more evident here - acoustic properties or marketing.
Maybe best - aluminum?
Either just the front panel, 1 cm thick, 0.4 inches, as in Technics SB-1000, or the entire case as in Krell MRS, the front panel is 2.5 inches 6.3 cm, and the sides are 1 inch 2.5 cm, also found the deep sea sound Mariana 18SA is made of 6061-T6 aluminum 3/8 inches thick. Many expensive companies make aluminum bodie - Magico, Goldmund, Preсide.
Zinc alloy - used in Realistic speakers.
When I was searching, this topic seemed interesting.
https://www.diyaudio.com/community/threads/aluminium-speaker-cabinets.373719/
Finally, lead, it is ideal if it were not for the toxicity. But it can be painted. In any case, despite its perfection, I have not encountered lead boxes in acoustics.

Thank you.

I'm looking for a schematic for a second order high pass filter with the ability to vary the Q from say 0.707 to 2.8 or so. Playing around with an update to the JBL BX63/BX63A. I know I can do it with rotary switches and fixed resistors, but I'd prefer to use a dual gang potentiometer.

From my calculations with C1=C2= 0.22uF, I'd need R1 to vary from 20k to 5.1k while R2 varies from 39k to 150k for Qs from 0.707 to 2.75.

I've searched and not found anything, maybe I'm not using the correct keywords?

Hi Everyone,

Just wanted to connect a little bit and ask you what new technologies, ideas have caught your eye or sparked new projects. Have you found something worth the hype or things which you realized just weren't worth the money? I find myself in kind of a lull. After decades of being an audiophile and some, ahem, expensive lessons in speaker building I've become a little jaded. The one thing I still like is composite mid's and mid-woofers. That is, drivers made of multiple layers, as well as fiber/resin composite designs.

My other big discovery is the joy of doing DSP/Active speakers. OMG it's so much easier to do just what I want. Not wanting to say active or passive is better, but as a designer it's so much easier I can't see myself doing a passive speaker anymore.

What are your lessons and happy discoveries that have stood the test of time lately?

Best,

Erik

SOLD

New in bag Tektronix P6112 100mHz, 10x. Proper probe for TDS 210 or TDS 220.

Price includes shipping only to CONUS.

$25 if using PP G&S, $24 if using F&F.

Thank you.

I was recently looking at writing a book (to be published on-demand/online--like Kindle Direct Publishing, etc.), and was thinking about how I would afford to enlist the services of a copyeditor to clean up the manuscript. It turns out that independent copyeditors serving micro-published authors are apparently just about to be "out of business", to be replaced by ChatGPT:

https://thejohnfox.com/2023/04/how-to-use-chatgpt-to-copyedit-your-book/

I have to say I'm glad I'm not an independent copyeditor. C'est la vie!

But for those that have been harboring secret thoughts of publishing a book, the costs and logistics are now much lower/easier...and interactive.

Chris

Has anyone built one of these? Elektor magazine designed PCBs and a layout back in 2002 for the Claus Byrith modified Mullard 5-20. I'm about to start building one.

Hi all. New member here, attempting to repair a BP7002 amp for a set of towers i purchased locally for next to nothing knowing one amp didn't work but the other worked perfectly.

i read through hundreds of threads on the subject and being familiar with fixing a couple vintage receivers, i thought i'd give it a try. i bought another non-working 7002 amp off ebay to use instead of mine (in case i mess up and for spare parts) where the led comes on, but no sound through the sub- although after gettin git and hooking it up, it has random hissing/static, kind of like the sound you get when scanning a tuner in FM between stations. Glad i bought one though because when i removed the bad one from the tower, it had been tinkered with and had some burn marks on it and the led didn't even come on - so i put it aside and focused on the ebay one. i compared the ebay one to my good one, measuring about every voltage possible and everything checks out. there are a few that are "slightly" off but in looking at the boards and the schematics i found, there was a rev B to rev C change that added/changed a few resistor values and Q types, so the small variances seem to make sense. Regardless, nothing is grossly off, all very, very close.

The ebay amp also suffered from the led light going out sometimes and voltages skewing when the light went out, and a weird noise when plugged in....i assumed that was the power supply and was planning to replace all the electrolytic caps anyway as they are known failure points (except the huge ones), which i did, and now both those issues have been fixed (4 of the caps actually tested bad) HOWEVER the scratchy/static sound from the sub is still there with no input, just plugged in, and no sound. I tried on both towers as well and different outlets. It seems this just cannot simply be voltages, or caps.

Moving to the amp board, I checked the three big output transistors on the heat sinks, and the several small ones, as well as all the diodes, several resistors, and the two caps (in circuit) and it all checks out nearly identical to the good one i have.

At this point, i'm thinking it's one of the IC's....or I'm missing something, otherwise I'm at a loss. Any thoughts on anything else i can/should try?

Any help is appreciated.

Hey all. New member here, interested in learning more about audio repair. I frequent audiokarma for vintage stuff but want to broaden my horizons with general repair and most recently, a class D amp repair that's giving me issues - will post in the class D section. Thanks.

Hi all,

I registered as there seems to be a great community here to learn from and learn with. Over time I hope to have fun and safely explore some concepts as I dive into the hobby. Long term goals are to build a tube tester, a tube preamp, and then a couple power amps (SS and tube).

I already have really nice gear at home, but things do break and I want to be able to fix it or replace it with like circuits. In the past I had Audiolab 8000p amps that developed the same fault in the second amplification stage. Clearly the transistors there were at a failure point due to design or choice of part. I was able to rebuild the traces and put in replacement transistors, but I abandoned the project as I lacked the equipment to bias the circuit and the building and testing knowledge to ensure the rest of the circuit would operate safely and within spec.

So I'll lurk and learn for a while until I can ask good questions in the future.

Cheers,
Waffle

I would like to interface one of the Behringer: POWERPLAY 16 P16-I

with an ARM based single board computer with an SPI input (pretty much all of them do - from the Raspberry Pi to newer, more powerful ones).

The "Ultranet" output is apparently just 2x AES3 data streams over Cat5 cable (running 8 chans x 48kHz sampling rate on each stream): X32 User Net • View topic - The pm16-m protocol

So they are using 2x AK4114 digital interfaces, which seem to be obsolete (there is an AK4118 which maybe is a successor part, but seems hard to find).

Transformer isolation is used in this system, so thought I should do the same. Given the Cat5 interconnects used, the architecture I have in mind goes like this:

RJ45 "PoE" compatible MagJack --> 2x clock recovery --> 2x SPI inputs on ARM board

the MagJack could be something like:
SI-52008-F - STEWART CONNECTOR - SOCKET, 1 PORT POE GIGABIT MAGJACK | CPC

I would appreciate any comments, suggestions. Particularly if there are projects already going on that I could re-use, e.g. for the clock recovery.

Hey everyone.

I'm interested in building all sorts of audio equipment. Currently working on 2 projects: a modelling amp head (based on NAM) and an ultranet personal mixer to interface with my XR18.

While trolling the bay I found this:

http://cgi.ebay.com/2-X-NAIM-CLONE-...ryZ12050QQssPageNameZWDVWQQrdZ1QQcmdZViewItem

Would this be a worthwhile first project for a kit newbie? Not much published on it, though I did find this:

http://hk.myblog.yahoo.com/ricky_t7/article?mid=338&prev=340&next=335

Highly suspect source, references to Kenny G and such...

Does anyone have a schematic/Service Manual for these PA speakers?

I am building myself a new CD transport based on a NOS CDpro2 drive i have been saving. I have been looking at various schematics looking for a high quality output driver/buffer/converter call it what you will. interface from the Cdpro2 unit to a rca/bnc SPDIF jack. IDEALLY I would like to have AES/EBU, RCA, Toslink and ST. But I will settle with just rca/bnc.
I see that ARC uses a AM26LS31, Wadia uses various 74 series parts like the 74HCT86D or the 74LS00 etc etc.
.
I am looking for Info on the thought process for these devices. I would imagine you need a high current FAST chip capable of driving a capacitive load into 75 ohms. I tried googling various versions of SPDIF output board. etc and did not find much that was what i was really looking for. are there any really good designs I should look at?

Greetings. I just carefully designed and built a beautiful set of speakers, but when I measure the woofer low-end response, it's very far from the calculated response. Here are the details.

1. I'm using pairs of CSS WR125 drivers. I got their T/S parameters from the pdf file included in the documents. (see CSS-FR125S-wr125-dimensions)
2. I enter these parameters into VituixCAD. and modeled a closed box with Qtc of 0.8. It said 18.6 liters for 2 drivers in parallel. (see VituixCAD showing 18.6 liters)
3. So I build a cabinet with 18.1 liters and stuff it with the recommended amount of Polyfill - 1lb/ft^3 I think. I mount the 2 drivers in parallel.
4. I made close-mic (nearfield) measurements with the Arta software. I chose one of the two woofers, but kept them both in parallel. I've got a calibrated microphone from Dayton Audio, a nice external power supply with synchronized clocks for the input and output, a nice LM3886 amplifier etc . . Not sure of where to put the microphone, I made several measurements of one woofer from 0.5cm to 5.0 cm from the phase plug, as well as an off-axis measurement. They were all very similar after normalizing absolute SPL levels, so microphone placement is not an issue.
5. The FRD data shows a big peak above 100Hz and a 14dB/octave slope! (see VituixCAD FRD)
6. I also measure the impedance (of both drivers in parallel) with the Dayton DATS V3. The result says my Qtc=1.18! (see snip of DATS data)

So I'm kinda' stumped, and hoping there's something wrong with my measurements. One thing I'm considering is that my cabinet is too tall and skinny. the cross-section is a trapezoid with a 5" baffle, an 11.75" back and 5.75" deep for a cross section area of 48in^2. Then it's 21.25" tall. Did I accidentally make a transmission line without an opening?
I'm confused and frustrated. Any advice would be helpful.
View attachment 1358224

Hi all,
Measuring TAD TL-1601, I found its frequency response missing 100Hz to 300Hz range. it has a very deep "abyss" in that range (~15dB).
Pls check this pic.

I need advice on how to cross this bass driver, i have tried to cross it with third-order, fourth order crossover (500Hz) but that "abyss" is still there.
Thanks all.

Suppose I want to find all the threads started by, say Minek, including the word "linearity"?
How can I proceed using a single, simple step? It cannot be that difficult

I have several pairs of Dunlavy speakers.
I am a huge fan, my budget finally allows what I drooled over in the ‘90’s
So I own a pair of SM-1’s and built boxes with downfiring Scanspeak 25w8565-00 to emulate Dunlavy models SM-IIIA, Corinthian, Aletha, Athena etc… all similar. I have read John Dunlavy state that there is no crossover between the Viva 6.5’s and the 10” I was under the assumption that there was only an inductor on the 10” downfiring Scanspeak. But recently a gentleman I met
On FB showed me a drawing of his crosssover for only the Scanspeak 10”
This is what I was given as it’s crossover

The DCR OF THE Inductor seems high.

hey, not long ago I acquired a dac from the 90s called the muse model two. it uses pcm63p-k grade chips, a cs8412 input receiver, df1700 digital filter and ad846 op amps.
I want to upgrade this thing, I want to switch the cs8412 in favor of a wm8804 which should be easy with a wm8804->cs8412 adapter such as fetaudio's offering.
but what should I do with the op amps? is there any upgrades that can be done? I read somewhere that the ad846 is used as a passive I/V. what does that mean? It could be wrong, but there are no schematics to verify that.
(side question: is it possible that the dac board uses all 16 pins of the IDC cable that is connected from the digital input section? eventually I want to drive the D/A board with a amanero usb and a 12s input switch so that I could keep the current digital input section AND have a separate usb input that I could just switch between. problem is that the only ones I have seen use a 5pin input (I uploaded the picture of the module), will it work? I used my amazing drawing skills to draw an illustration)
I am pretty ignorant when it comes to audio engineering, so a lot of what I said is probably wrong. so any help regarding this would be very much appreciated!
(other dac upgrade or alternatives to what I said are welcome!)

Hi,

I am starting to design new 3-way speakers with Scan Speak 26W/8534G00 woofer inside a box of about 76 liters.

I would like to increase the depth of the bass and therefore the simplest solutions that come to mind are two: bass reflex or passive radiator.

I have carried out simulations with the bass box that I have attached.


As you can easily see, opting for a simple bass reflex results in significantly higher performance than a passive radiator such as a Scan Speak 26W/0-00-00.

I also did some tests with the SB Acoustics SB29NRX2-00 and the result is more or less similar.



The best result is with dayton audio passive radiator:




If the simulation is correct from a sonic point of view there should be no reason to opt for a passive radiator also considering the non-negligible additional cost.

The reasons why they made me evaluate a passive radiator instead of the bass reflex are two:

1) The crossover is positioned on the bottom of the speaker, to access it I would have to position the woofer at the bottom and therefore far from the midrange and this is not positive. If I used a passive radiator I could position the woofer near the midrange and the passive radiator at the bottom so that I could access the crossover by removing the passive radiator instead of the woofer.

2) Aesthetic reasons, the front passive radiator is nice to look at.

Now I wonder from a predominantly sonic point of view could the passive radiator have advantages over the bass reflex? It seems to me not.

What do you think?

Update

I have corrected the Scan Speak 26W/8534G00 parameters and compared to Dayton Audio DSA315-PR.




much better now and is close to reflex. What you think ?

I have this thread on another (ModWiggler) forum and it was suggested that I place it on here (although they vanish quickly if I don't bump it, which it now has).

I still have tons of components for sale. Any price I quote will be in * USD * as most reference this. I would prefer Wise to Paypal as a guy just paid me direct in AUD and it cost him very little to use Wise in comparison to Paypal.

Here is the list if you want some let me know which ones you are after. Postage as a reference is $15 USD for 250g to the USA ($20 USD to Germany for 250g) / $20 for 500g total to USA.

If you can find a good price for any you are interested in on a 'legitimate' site (NOT EBay as there is so much junk on there), I will do better by 20% at least. EBay is okay if it's for a reasonable price on there, showing me an IC worth $20 going for $1 on EBay is not realistic. The numbers are approximate, but I would check and count them exactly if interested.

If someone wants to do a bulk buy for a project on here; for example a Moog Filter and needs TIS97 transistors or needs transistors for 303 / 606 /808 /909 like 2603s and 1115s etc, I can do them a good deal. Point me to a price even if no longer stocked and I will do better.

Using EBay as an example of say the THAT4301P Analogue Engine, the price = 50 GBP ~ $63 USD - I can sell them for $30 USD each, less than half. I don't know about the origin of the ones on EBay, but my 4301s came directly from THAT Corporation years ago. Just like ALL of the audio components on my other thread on here came direct from TI & National before the were taken over by TI. So the National parts I have had for at least about 14 years in storage.

Here are the Lists of Components:

Audio Semiconductors.txt ----------- AUDIO Specific Components DIRECT from TI & National

Like High Quality Audio and Instrumentation OpAmps & Power Amps.

Electronics and Music Components.txt ------------ Other Components for General Electronics & Electronic Music.

Like Rare Transistor Arrays and Synthesizer Transistors.

https://drive.google.com/drive/folders/1biIKbagCNJVCQzVrnarfznPFaL4qDL4P?usp=drive_link

Here are the details of the changes I made to my utracer6 to improve low voltage at high current accuracy and to reduce failures of the MOSFET switch and driver transistors.

For more context see google group "https://groups.google.com/g/utracer" and threads "Improving the basic accuracy of the uTracer6" and "Utracer6 waveforms"

I invite comments and suggestions.
https://groups.google.com/g/utracer/c/gatTKyiHGLI

https://groups.google.com/g/utracer/c/gatTKyiHGLI

Hi All,

As always I should start with a big thanks for all the knowledge provided within these threads - such an amazing resource.

I'm in the process of designing a set of 3-Way active speakers and am trying to get my head around the best way to power these, previously I've only built passive speakers with an active cross over so this is all very new to me. I'm trying to keep the build relatively cheap - just to see what's possible on a bit more of a budget - but can stretch the budget if necessary, I've come up with a couple of ideas, explored below and would like some input as to whether my approach is feasible or if I should explore some other options.

Driver ratings are roughly 100W sub, 50W mid, 20W tweeter.

Option 1:
Each speaker has a 100W per Channel amp board. One channel for mid / high (this channel has a cross over and a high pass filter on mid), 2nd channel is exclusively for the bass driver with a low pass filter. I've attached a schematic of what I think this would like.

Thinking to use either of the following amp boards:
100W Amp board 1
100W Amp board 2

With maybe this power supply or something similar
Power Supply

Option 2:
I use a plate amp, installed into one of the speakers, I'd then just connect the other speaker to this plate amp via the 'Link' port.
500W Plate Amp

This is of course a slightly cheaper option coming in at maybe half the price of option 1, however I'm unsure as to the quality of this amp, I'm unable to find detailed specifications on it and haven't used the website previously, so who knows what that'd be like. I also think that designing the cross over for this option would be far more involved, although perhaps I'm wrong about this.
The only other thing I think I'd need to take into account is the difference in internal volume betw. the speaker with the amp and the one without.

So, are there any issues with the above approaches? Which would you go for? I'm also open to others (budget allowing) - I guess there's things like mini DSP or just making them passive (although would really like self contained units on this build).

Many thanks,
Lawrence

For some reason suddenly the graph displays are black and also prints black. The cursor
still blinks and tracks curves, but all else is pitch black... Any suggestions?

I have three ONE (two are now sold) Tektronix P3010 100mHz 10X probes, new old stock in unopened packaging ready for new homes.

Price includes shipping only to CONUS.

Cost:
1 probe = $31 including PP G&S fees and shipping/$30 if you use F&F

Thank you.

Good day all.
Does someone perhaps have the service manual or a schematic of the Polk psw110 subwoofer.
Much ppreciated !

Appreciate the opportunity.

Up for sale two pieces of the excellent Steg QK200.4
Both in excellent condition, boxed with all their stuff inside.
Price 600 euros FOR EACH, shipped within European Union


The QK200.4 from STEG is a 4-channel amplifier with outstanding performance. The amplifier has an impressive 4 channels with an output of 200 watts RMS each at 4 ohms and can be switched to 2 ohms for an output of 4 x 300 watts. It also offers a bridged output of 2 x 600W RMS at 4 ohms and includes HP and LP filters for different frequency ranges. With its compact dimensions of 394 x 64 x 207 mm, this reliable device fits easily into any facility.

Data

• 4-channel power amplifier Class AB
• Terminals current: 35mm²
• Power: 4 x 200 Watt RMS @ 4 Ohm
• Power: 4 x 300 Watt RMS @ 2 Ohm
• Power bridged: 2 x 600W RMS @ 4 Ohm
• HP filter 12 dB/oct. 20 - 300Hz or 400 - 6000Hz
• LP filter 12 dB/oct. 30 - 600Hz or 300 - 6000Hz
• Full, high-pressure or low-pressure filter.
• Dimensions: 394 x 64 x 207 mm

Hey everyone,

looking forward to interact with you.
I recently changed from my Marantz SR7010 to a self-build Mac mini M2, USB DAC, CamillaDSP solution.
Looking forward to improve this to be a kind of diy receiver.
Let's see 🙂

I am the middle of building my own speakers, too.

I hope I can learn a lot from experienced people in this forum and am excited about the discussion that may happen over time.

Cheers!
Marius

As the title says, here is the list:

- 2SK240-Y (2x 2SK170Y inside) -12 pcs - $12 ea or take them all for $120. SOLD

- 2SK150A-BL - 110 pcs - $4.5 ea or $4 ea for 10 pcs or more

- uPA68H-N - 38 pcs - $4 ea

- uPA63HI-L - 16 pcs - $3.5 ea or take them all for $40 SOLD

- 2SK58 - 5 pcs - $6.5 ea or $25 for all 5 pcs SOLD

-2SK246 & 2SJ103-GR - 24 pairs - $2.5/pair SOLD

-2SK117-BL - 120 pcs - $1.75 ea or $1.5 ea for 20 pcs or more SOLD

-2SK30A-Y - 240 pcs - $1 ea or $0.75 ea for 20 pcs or more

-2SK373-GR (high voltage jfet) - 28 pcs - $1.75 ea SOLD

-2SK301-R - 50 pcs - $1 ea

-2SK40-B - 100 pcs - $1 ea (equivalent to 2SK30A-O)


For NEC Jfets below only sold as matched (Pair/Quad)

-2SK68A-N - $5 for MP (only 16 sets available)

-2SK163-L - $4 for MP & $10 for MQ

-2SJ45-L (equivalent to 2SJ44) - $5 for MP & 12 for MQ

-2SK162-K&N - Low en (0.65nV/sqHz) & high GM (45 mS) - $6.5 for MP & $15 for MQ.
*due to the high GM of this jfet, I can only guarantee matched at 0.2mA.

I also selling 2SK170-BL matched (quad & octet) here

Shipping is a flat $10 Worldwide (no matter how many you buy)

PM me if you're interested

Thanks for looking!

Collector of vintage components in particular transistors and capacitors.
Also have some vintage HiFi stuff, B&W speakers, about x4 different sets and a strange pre- Amp probably built in the 70's or 80's.

Always interested to see stuff people build and collect etc.

Noddy

This is probably a very elementary question but I was unable to find a definitive answer so here goes......

Does the lining of the enclosure count against the enclosure space?

I know polyfill "expands" the enclosure space by making the "air denser"

One more elementary question since I probably already sound stupid......

Would it be ideal to keep any sort of lining away from the woofer port entrance on the inside of the enclosure?


I plan to use this SAFB material and it seems from the charts that the thicker, the better. I am pondering why anyone doesn't just go with the thickest material they can find. That is, unless it counts against the enclosure air space.

hello chevin amplifier psu its little aluminium box with 12 IRF740 and small transformer permannt running @100Khz
https://www.diyaudio.com/community/...-a-series-schematic.55780/page-2#post-7789674
i think to build two FSFR2100XS board , 400W max its OK , for home music,no need 1500Watts output
original psu no optocoupler so i want do same , in CV/CC
so my qustion its: to substitute tho optocoupler with two rsistors its insan idea???
thanks

I haven´t found anything of substance to this topic. Obviously, different manufacturers set their speakers differently, different design choices are made for different reasons, but it would be nice to have some words about that.

For example, repeatedly I have found that RCF drivers work great at, or even past Xmax. Haven´t seen an issue yet. Though looking at the magnetic gap, I am worried how can they sustain the centering at these excursions. Feels ridiculous by looking at it with "common sense" eye.
On the other hand, although okay by design, I saw with many B&C speakers, that they have stiffer and steeper onset of suspension breaking at Xvar (B&C speakers Xmax marking), and the speaker soon stops responding to more stimuli in greater excursions. Also more power is burnt on suspension at Xmax/Xvar than usually. What would be the reasoning behind this? How it translates to real world usage? (Well, I burned my 21DS115 a bit, working it to Xvar, but there might be more to it). I guess that compared to RCF, additional ~120-150Watts in power compression, does not going to change that much if the speaker is going to burn or not. or would it?

How about long term longetivity of the suspension at, or past Xmax? Do rentals and fully exploited installs/rigs take recones as consumables to be replaced within 3-5 years of use?



Working at 12mm with negligible loss is awesome to see.
Not planning to go past 15mm, but it is nice to see that the driver handles even 3dB momentary overload with grace. No signs of rubbing or too stressed noises, which in turn can be dangerous, because if the speaker does not protest either with noise, sound, smell or anything, then one cannot do anything about it dying.

A Nixie bargraph tube IN-13 with driver PCB which includes high-voltage DC-DC and control circuit. All it needs is 5v and control signal: it can be controlled by analog voltage (0 to 5v) or by I2C. Lot includes 1 nixie tube and one driver PCB.

It is ready to be connected to Arduino or any other MCU. No need in additional circuits

The driver can work be controlled in two ways:

• by analog voltage on pin Vctrl from 0 to 5 volts.
• digitally thru I2C: The module address is defined by A0 pin. If it is 0 (connected to GND), the address is 0x13 (7 bit address mode), if A0 is high (connected to +5V), the address is 0x14. So up to two modules can be connected to one bus.

Price is 25 USD/piece (includes the In-13 tube and driver board)

Also available in my Tindie store:
https://www.tindie.com/products/eclipsevl/in-13-bargraph-nixie-tube-with-driver-and-dc-dc/

Hello people,

I have a Quad 303 amp ( ser# 4890 - so an early one!).
I was given it by a friend as a non worker - so after a couple of weeks of tinkering I have finally got it back into some form of life.

I have replaced the electrolytics and a couple of burnt resistors and the trim pots - all the transistors and diodes are original.

The amp turns on and i am measuring 67v between pins 1&9 and 33.5v between pins 5&9. The quiescent on the RIGHT channel is at 9mA via RV101 but i cant get the quiescent on LEFT channel to change from 85mA irrelevant of which way i turn the POT.

I have methodically tried tracing the issue but I have run out of ideas! - I could 'cheat ' and just replace components for the sake of it but that takes the fun away - I will get to the bottom of this but I just need help!

Electronics is not my job - I just enjoy it!!!

any help/pointers greatly appreciated!

regards

Mark

I have bought this dac from an German fellow who told me the dac was perfect. But sadly it's not. He does not answer anymore messages so i am now stuck with a nice defective DAC. Thanks for that!

The unit does not lock on any input, so no sound on both outputs. I have tested multiple sources and cables. Also i replaced the CS8412 but that did not helped. The regs and PSU are al oke. There are no scematics online and i have no scope. What will be the next "usual suspect"; the DF1700?

I don't have any at hand for testing unfortunately. If somebody have any other sugestions, i would be greatful.

Picked up a 10 Band EQ called EQ210 from Ali Express.
I am looking for the schematic or even better if anybody has entered this up into a spice / Tina TI / Micro-Cap file.
The idea is to play with the Band Values. But all I really would like to do is just eliminate the need for a pre amp. The thought process is if I could use the first OP amp to boost the signal a bit. I wouldn't need a pre-amp. Would the idea work ?. Or would it fall flat i.e. the second I use the first opamp to boost signal Ill mess up the signal ?.

This is the link for the item.
https://www.aliexpress.com/item/1005003536983742.html?
There is a very low res image of the schematic I guess its better than nothing. But maybe somebody will recognise it and or have a clearer image. Ive given up with playing with Tone Control Boards. Plan to give this a shot.

Hi everyone.
I have a big problem with this Mark Levinson cd player. After a storm it was turned on but is in this position, the display goes haywire. Checked the various components, I found the 5v voltage regulator (7805) shorted, two transistors 2n6730 and 2n6718 shorted, the multiple resistor 51ohm damaged and what's worse, the microcontroller ALTERA inserted in the socket is very hot, in practice even this looks burnt to me. In fact, i took the chip ALTERA out of the socket, with a multimeter i tested the power and ground pins result in short circuit
Now after checking all the components and replacing the damaged ones the problem is in the microcontroller. Does anyone know where it can be found.
Unfortunately in Italy the Mark Levinson assistance centers can be counted on the palm of your hand and when contacted nobody can give me an answer.
I imagine that in the ALERA chip there is also the firmware as well as the CD control.

Can anyone experienced help me?

Hello,
i bought this amp a few years ago with a broken PS and Audio. Fixed this a few years ago, tested, but had a power off bump in the sub. I decided a few weeks later, too work again on it, but it stays in protect from the beginning.
Now, 2 years later, i had time to work on it again. Connected it to a sub, and a multimeter.
Powered up, dc offset up to 4VDC, slowly decreasing to 0.6VDC. Amp works fine in this state, without protection . But with this little offset, you can hear a little bump.
Anyway, i think it wont work reliable.

Removed the Pre amp board, no dc offset. LED isnt lit, but draws the normal current 0f 1.4Amps.

Any suggestions?

link ebay : https://www.ebay.it/itm/365109524414

Hi All

Got yamaha ca-2000 for repair. It seems secondary winding for service voltages got shorted inside the trafo. So rather then to repair big trafo I would install small one for all those service - lamps, relays etc. BUT - on schematic available in the net there are no voltages. Just b+/b-, b1+/b1- etc. No single voltage marked on the scheme. I would appreciate if any happy owner measure ac voltage marked as Be/Be (source for B3 system voltage) - these are two blue wires connected to the power supply board. I assume should be about 34-40 volt AC. B3 would also helpful. Thanks in advance.

Hey, newbie here. Love vintage stereo and restoring/maintaining. My skill set is low, trying to improve, but I especially like seeing what people are doing to the things that I own. My audio bucket list includes wanting to get to where I can build a system myself (valve amp and pre, speakers) just for fun. Love the system I have.

Hello everybody!

I am looking for a vector network analyzer that can measure audio signals (20kHz down to 20Hz).
I know that the "best" solution would be just using the soundcard from the computer and some software but I want a standalone device.

I already searched online but I only either found audio analyzers that would just analyze a incoming signal (no built in signal generator) or I would find vector network analyzers that worked from 9kHz to a few GHz.
But I want an actual VNA for audio frequencies.

Can anybody recommend me anything?

Anybody got anything beyond the mfg. measurements?

https://www.parts-express.com/Celestion-TFX0515-5-Coaxial-Full-Range-Professional-Driver-294-1974

I have read that when people talk about good/great tube amps, among all the other things important, it seems one of the things at the top of the list is the quality of the “iron.” And of that “iron” (transformers), the output transformers seem to impart the most difference in sound-quality. I would like to have a discussion on what makes a good/great OT. I know many of the top names: Hashimoto, Tango, Tamura, Lundahl, old Peerless, et.al., but this post is about what is inside those that make them so good/great. I am definitely not an expert in this field, so I am open to your information, and also your corrections. Hopefully, I can bring in a few Tube afficionados with experiential knowledge, that have owned different types of OT’s.

I’ll start with what I have:

1. There is something about the nickel content of the iron core. Interesting, since nickel is undesirable on speaker binding post, but of course, this is a different application.

2, The quality of the copper in the windings, (or the silver winding in Steve McCormack’s half a million dollar amps.)

3. The particular ways of winding.

Ok, that’s all I know. Feel free to elaborate on any of these, and what I might have left out. Sound quality observations are appreciated, and so are part and SQ comparisons between brands. Thanks!

Hi folks, I've been out of the speakers from raw drivers + ideas hobby for about 10 years, no particular reason, just other responsibilities (and some space issues). Now I'm looking to design a few things and hopefully share some good designs too. I regret not putting some quality designs out there after having spent so many years learning the craft in audio forums. Anyway, 10 years is a century for software and I'd really appreciate anyone willing to skim the following and chime in with their suggestions.

This is how I work / what I know how to do; I'm wanting to hear what people with similar processes like to use for software:

1. concept/purpose - usually I just gather candidates for driver(s) based on what is needed for the idea, attempt to simulate bass and sometimes full in-cabinet response with diffraction/etc. all from factory data (so have to trace curves unless I have measurement files). I would also be interested in simulations of generic sources at this stage for horns/waveguides, rooms, crazy acoustic filter ideas, etc., but didn't get much further than partially learning Hornresp back in the day.

2. get drivers (and take my own driver measurements if suspected necessary), verify cabinet design works physically, build cabinet

3. take in-cabinet measurements of individual drivers: impedance/electrical, nearfield response, gated 1M response 0-90°, confirm acoustic centers of drivers, confirm max. SPL in bass.

4. repeat steps 1-3 until fatal flaws in cabinet design and/or driver selection are fixed.

5. simulate complete speaker and xover using in-cabinet measurements (preferably with polar/directivity in the sim) until satisfied with xover, verify results, adjust and repeat.

P.S. if you're curious, I was mostly using PCD (the simulator that only worked with outdated Excel) for design and ARTA for measurements. I also used Boxsim pretty often for general messing around with ideas. I can't remember the bass cabinet sim I used most the time. I had loads of other software but these were 99% of the time.

I am in the process of assembling a 3-way PA system intended for events of 50-150 people. As someone who’s relatively new to this (background being more DJing and promoting), I’ve still got plenty to learn, hopefully through forums like diyAudio.

Specifically, I’m looking to get advice or assistance with amp selection, driver selection/installation, and rewiring for two vintage JBL 4560 cabinets (which need 15s) and new tweeters (see attached). I had purchased both secondhand last spring from the original owner who had them custom built by Banquox Sound owner, Henry Kreindler. These builds are to complement two nearly finished dual-loaded JBL 218s subs, which are pending restoration and aesthetic detailing and will handle the low ends of the 3-way system.

I’m really happy to join this forum because it helps me get useful advice and support about my interests.

I have a MTX 7402 that came in after someone else attempted a repair on it. I corrected a bunch of issues and got it operational, with one channel performing flawlessly, and the other channel with the pictured interference. This interference exists whether or not any signal is being fed into the RCA's. This interference is not present at the input sides of the NE5532. The NE5532 has been replaced three times just to make sure I don't have a bad one. And the frequency of the interference suggests that it is from the power supply switching. This made me want to investigate grounds on the channel with the fault however have not been able to locate anything specific. Diodes also appear fine. Any help or suggestions would be very much welcome.

I want to tap the heater secondaries for a voltage doubler. (to power relays)
Problem is that the heater secondaries use a CT or a simulated CT with the two 100ohm resistors.
Since the doubler does not want a CT used, how to I deal with this?

Recently picked up a Velodyne spl-800i and it’s been running fine under testing. Decided to plug it in tonight and watch a film but noticed if the volume is very low the sub has a slight crackle and cuts in and out. If the volume is higher the sub works as expected and it can be turned right up without any issues.

I have read posts of issues at normal or high volumes but can’t seem to find issues at low volumes. I will pull the amp panel off tomorrow to have a look if it’s anything obvious but has anyone else encountered anything similar?

As New / open box original packaging. Can demo (near Cambridge / St Neots) in Auro 3D 11.2 active system... Superb in stereo or running the FL and FL as three way active monitors.
Multiple inputs AES / EBU digital in plus Fully balanced XLR analog inputs. 6 fully balanced XLR outputs.
New cost £900 will sell for £595. Pay Pal payment with Royal Mail Special Delivery (£25 extra) or cash on collection.
Legendary spec / features performance... See multiple You Tube tutorials / videos / reviews.
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I've just come across the Faital Pro 15PR300. I'm not sure of their naming convention but assume it's a little brother to the well liked 15PR400. I'd just like to get some feedback on the specs. It has a lovely flat response graph where it plays, down low. It seems happy in a sealed cabinet too which is why I caught my eye.

Using Speaker box lite (less than ideal but I'm on Mac) the plots are:

Red - 15PR400 in preferred ported cabinet 103.8l tuned to 44.4Hz as recommended by app
Blue - 15PR400 in 45l sealed cabinet recommended by app
Green - 15PR300 in 130.7l sealed cabinet recommended by app

Here's the 15PR400 for comparison https://faitalpro.com/en/products/LF_Loudspeakers/product_details/index.php?id=101060100

What do you think as a sealed alternative to the 15PR400?

Edit 2023/01/22: Complete kits are now available! View the sale thread here, or go directly to the order form.
Edit 2023/09/03: All kits have sold! Thank you to everyone involved
Edit 2024/04/18: Kits are available again, now on my website: reduxkits.com
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Background:
In 2012 I helped put on an event for newbies who were interested in electronics or audio build their first amplifier. We called it Amp Camp. My largest contribution to the camp was to design and machine 50 (!) chassis for our 25 attendees. We were lucky enough to work with the venerable Nelson Pass, who supplied the circuit design and many of the parts for that camp.

Many of the design decisions he made then had to do with accessibility for new builders. Among them:
Switching PSU to prevent exposure to mains voltage
Low parts count for simplicity
No transistor matching
Single potentiometer for easy biasing
Low parts cost to entice the the uninitiated

Objectives:
A return to form for the Amp Camp Amp. Above all, ease of assembly and low cost. Accessibility is the name of the game for this variation. My intent is not to make the best performing ACA, but a design further optimized for someone who is interested in our hobby with little to no build experience.

Results to date:
An even easier to build ACA (1.6) for about $80 per channel in low quantities, if using non-premium parts. The cost comes down when building more as a batch. I built 2 channels and had an unexperienced friend build 2 channels for a first prototype run about 2 years ago and have been using them in my desk stereo system since. I have sat on this design for a while now and figured it was time to finally write it up for sharing. The amp you see below is a single channel.



The enclosure presents the biggest opportunity to cut cost in this amplifier (and most other DIY amps). There a few critical functions that an enclosure performs.
1. Heatsink
2. Finger-proofing
3. Aesthetic

I selected a heatsink meant for LED lights. Typical "grow light" heatsinks are now produced in large quantities and are one of the more economical large passive heatsinks available. The one I've selected is a circular forged pin heatsink 120mm diameter and 70mm tall (4.7" × 2.75"). I estimate its thermal resistance at around 0.9°C/Watt. It cost $19 with free shipping.

Another advantage of the LED heatsink is that the base plate is meant to be horizontal with the pins sticking up (and LEDs pointing down). This is what makes the "caseless" design a bit more practical. By placing the PCB below the heatsink it provides a bit of protection for the components underneath and IMO, some visual interest. Standoffs hold the whole thing up off the table and no other enclosure parts are needed since the amp operates at a touch safe voltage.

Board layout and parts selection:
In the interest of beginner friendliness, I've elected to print component values on the board and line up all the small resistors according to their numbered designation. The large capacitors lay down flat to minimize the overall height (currently 96mm / 3.8" tall). The heatsink is "grounded" through brass standoffs to the PCB mounting holes. Nylon standoffs screw into the brass standoffs so they may act as somewhat less scratchy feet for the amp to sit on. I added a bit of camp themed artwork to the back side of the PCB along with the simplified circuit from Pass' original article. I chose white soldermask because I think it makes the art look good, but it's quite difficult to see the traces ¯\(ツ)/¯





In an attempt to make assembly as easy as possible, all the connectors and the power switch are mounted to the PCB. Look ma, no wires! PCB mount power jacks and and RCA inputs are commonly available. Binding posts are less common. This design uses a right angle screw terminal with a 5 way binding post screwed into it. The threads are secured with a bit of solder. Because the binding posts and screw terminals have a high thermal mass, this is the most difficult part to solder. But even a beginner with patience and a higher powered soldering iron should be able to make it work.



These LED heatsinks often come with pre-tapped holes with spacing for the most common LED modules and lenses. I aligned the MOSFETs with the furthest apart M3 holes and used the M4 holes near the edges for the standoffs.





Future improvements and changes for the ACA Redux:

• Add more info to silkscreen, e.g. bias voltage. Possibly sacrifice rear side art for complete build instructions.
• Add more output capacitance (this version has 2000uF).
• Replace LSK170 label with something a bit more generic.
• Consider repositioning the LED to shine up through the heatsink pins.
• Consider squaring up the back edge of the board so power and RCA connectors aren't angled.
• Move to a larger 140 x 70mm heatsink for more cooling capability and board area. This design doesn't need it, but I want to continue exploring this physical design while standardizing the heatsink size.

I'd love to incorporate feedback and improvements from the hive mind on diyAudio and gauge interest in boards / kits, no promises though!

Further developments:
A slow moving, inaudible fan above the heatsink will more than double the cooling performance, opening the door for higher power designs in the same form factor. I've started tinkering with some old F5 boards I had laying around and I've started thinking about a 2 channel, single board ACA for bridge / parallel operation. These future projects are the primary reason for moving to a larger PCB / heatsink.

Hello everyone. I have a problem with a small Fatman Itube MK1 amplifier. A resistor has burned out, in R17 (see the photos) and I cannot find the diagram of the printed circuit, in order to replace it. Do you have any idea of the resistance value? Thanks in advance Pierre

Looking for a few pairs of LME49720HA TO-99 can opamps from an authorized source such as Digikey, Mouser, etc. No Chinese sources please. Each pair should have matching production lot codes. Please pm me. Thanks in advance!

Rick

Has anybody designed a teeny (as in 10cm/4" midwoof) 2-way passive with a nice accurate sealed-box response from 200Hz and up or so? I tried searching around but didn't find much except a few DSP ones. No particular budget, just looking to see what's out there in general that matches the goal.

Some years ago I built 801/801a/10/10Y preamp.
I'm a fan of thoriated tungsten (TT) filament tubes (was made in the first half of the last century), collected its for decades.

I've been using it daily for six years, satisfied.

It's two box design, HT and LT supply, and preamp box.

I use 801 graphite anode, ceramic base Radiotron.
It's (801) has a little lower power than later developed 801a, but as preamplifier it's irrelevant.

The design can work with 801/801a (military sign VT62) or 10/10Y (military sign VT25). The VT25A also can work, but it's not TT tube (oxide coated filament).

The tube is "gyrator" loaded, filament biased. The filament regulator designed by Rod Coleman (V4, later changed to V7).

The gyrator is early prototype of Ale Moglia design.



The HT PSU working with AZ1 mesh anode rectifier, c-L-C-L-C design.


The LT raw supply C-r-CMC-C type.


The PSU box is simple, I put it at the bottom of the rack.
It's powered up with common switch of HiFi rack.
The ramp of AZ1 enough slow for heating the DHT preamp tubes.

The amplifier box made using a prefabricated chinese box.
I usually use these heatsinked boxes for build. The quality is enough good and price (was) enough cheap.

The side heatsink cooling the filament bias resistor, the R.C. regulator and the gyrator.




The inner space is a little narrow, so the wiring is "neurotic" too. 🙂

I thought I'd share this for anyone interested. I am building open baffle subwoofers and I did some testing to find the better design. I was considering Ripole, SLOB, and H-frame. I figured that ripole and SLOB are both similar in that they constrict the front-firing side of the woofer so I built a ripole-like box with an adjustable gap and one 12" woofer to test it:



The cover slides left and right and fits pretty firmly for a decent seal. The back side of the box has a fixed opening of 13.5" tall x 6" wide and I tested the front various widths, all with fixed height of 14.75". Widths: 8.5", 6", 4", 2", 1". I tested with the enclosure and mic on the floor near the middle of my living room. I tested at 3ft and at 6ft mic distances, but turns out 6ft doesn't really matter because 3ft shows plenty. I also measured the back side at 3ft but that didn't show anything interesting either.




After doing this test I realized the ~90hz and ~110hz dips were because I tested with the mic on the wrong side of the room. When I 'flipped' the mic side those disappeared, all looked smooth like the green nude line. So ignore those and only compare the differences rather than absolutes. Here's a couple more for comparison:



I already had 8" boards attached so that's why I tested the 15x15x8 'sealed' box (open side face down into carpet). I also tested a 2x4 wide-face across center of the 15" square OB and it made very little difference. I also 3d-printed a conical insert that I tape to some wood to make a ring-gap but the test was even worse down low than the 2x4. I tried a couple depths with that and none were beneficial.



So basically the largest 'ripole' gap (which is the largest slot if it were a SLOB) performed the best. This isn't what I expected, I thought something like a 2" gap would be better, but I was wrong.

I'm going with the H-frame because it looks the best overall and it has no weirdness after 300hz, less coloration. The ripole-style box doesn't really save me space anyway because for this build I want them narrow and tall. My goal is to get 20hz-40hz as high as possible with 12" open baffle while fitting reasonably well in my space.

I know this test isn't exactly the same as ripole because it is just one-sided, but theoretically if this 1 woofer was playing twice as loud as a ripole pair would then the character of the output would be about the same, excluding the distortion difference from the offset & excursion distance. This test was just to see how the restriction of the front in this style would perform.

My conclusion: don't restrict woofers.

I was given a Sony SAWP780 sub today by a co-worker.

Said it didn't work so I plugged it in and nothing.

Replaced the missing fuse and still nothing. Power transformer got warm, but the power LED didn't light.

Here's the measurement of the sub. Not gonna try to disassemble yet to get a driver only measurement as I am not sure how it comes apart and don't want to damage the cabinet.



I decided to test it with a Dayton Audio SA100 plate amp and connected it to my main system.

The sub actually sounded decent. It seems to have a decent low frequency response.

Not sure what this means but under a certain frequency the driver seems to move more and push more air, although it still sounds decent.

What's y'alls opinion of the sub?

Is it something actually good enough to use for music or is it more of a home theatre sub?

Also is there any benefit to putting the driver in a different enclosure?

I used to read somewhere that there might be a problem if the crossover point of the tweeter was set too high. Firstly, is it true? Secondly, what is that symptom/phenomenon called?

In addition, from my experience with experiments of the combination of 2” dome midranges and 3/4” tweeters, I preferred the upper midrange or lower treble coming from the mid rather than the tweeter. Finally, so how high is too high?

In fact, I used to have an idea to design the mid-to-high crossover for the 2” mid and 3/4” tweeter combo by using mechanical low-pass filter of the 2” mid and electrical high-pass filter at somewhere in the upper range of the tweeter, or somebody would recognize the 3/4” tweeter as a “super tweeter“. But I’m not sure about the possibility of this concept.

Hello,
I' d be absolutely happy, if anyone could help:

Which secondary tranny voltage would you prefer 18 or 20v when going the standard psu route?

Many thanks

Rents

Hi all,
I'm just in the process of designing a one-off scratch built belt drive turntable, however the current hurdle is selecting a suitable motor system. I have plenty of mechanical/machining knowledge but little motor control experience so hopefully this will be a good learning opportunity.

Initially I was looking at the SG4 with MA-3D driving a BLWS231S-24V-2000 as suggested by Pyramid, and was hoping to order from Radwell ( https://www.radwell.co.uk/Buy/ANAHEIM AUTOMATION/ANAHEIM AUTOMATION/BLWS231S-24V-2000 ). However from speaking to them it looks like even though they list it on the website they no longer sell them, and there doesn't seem to be any other source for Anaheim Automation this side of the pond.

I would greatly appreciate any opinions on what the best route would be for me to go - has anyone got a UK source/spares for reasonably beefy motor which would drive a ~12kg platter? Ideally I would stay with AC but could go the DC route if that's the consensus (I have looked at the Maxon offerings, but am slightly put-off by the price tag and a few mentions of QC issues). The Papst motors sound enticing but there don't seem to be many lying around on ebay etc.

Any help would be massively appreciated!
Herbie

Ravi de vous rencontrer ! Je suis de Lille (France) intéressé par la musique et le son, mais assez nul en électronique ....je pense avoir besoin de vos conseils.