Hi, I have a Yamaha A-S701 amplifier that recently has develop an annoying problem. If the power is off then the relay of the mains st-by module is starting to click on and then off after exactly 30 seconds.
That is happening sporadically, you cannot predict when and I cannot find why.
First I thought that there is some remote control in the home that is doing that but that is not the case.
Started to search the net and I found that this fault has appeared on many A-S series, 301, 501.
I've tried to change the typical "faulty" PP capacitors in the st-by module but it does the same.
Found a similar thread but it seems that no one solved the problem of the relay that seems that it has a mind of it's own :
Yamaha A-S501 relay issue | Audiokarma Home Audio Stereo Discussion Forums
Does anyone have the same issue or can someone help me isolate and find the problem ?
Thank you !

Hello, I'm looking for some help with a simple opamp circuit. I play bass and in the end of my pedal chain there is one which has two outputs: a DI (goes to front of house) and an unbalanced output (goes to stage amplifier). This DI is already too hot for some mixers, but the unbalanced output has not enough gain to fully drive the stage amplifier. Being just one gain control to both outputs, you see the dilema.

I'm looking for building a device with an opamp to put after the unbalanced output and give some gain before the stage amp. As I'm not any expert, I read this article: https://sound-au.com/dwopa.htm and draw the following circuit from it. Assuming the opamp I'm going to use is capable of that gain, could someone point if this should work, or give some directions?



Some information:
-> Unbalanced output is 2kOhms
-> Stage amp input is 10kOhms (Crown XLS2500)
-> A bass guitar average output is around 100mV so, in my head, 30x gain would give me 3V, which would be more than enough
-> If this gain would be too much, would just tweak R1 until I'm happy
-> I have an isolated PSU which provides +18V 250mA. My idea would be to use a converter module to get +-18V from it
-> Bass signal is already buffered
-> This building must be as small as possible

Which opamp to use is another dilema. Thing is that, where I live, any TI, AD or BB brand opamp to sell will be fake. I just expect the opamp to do what it's meant to: amplify with no coloration, distortion or noise, which I can't expect from these fake ones.
I've found some discrete opamps with good reviews and good price to get over here, like this one: https://www.aliexpress.com/i/3256805273980370.html?gatewayAdapt=bra2glo4itemAdapt

So, my questions are:
1. Am I being too naive expecting it to simple work as I described?
2. If it's okay, should I add something into circuit, expecting a better behaviour?
3. Input/output GND and PSU GND should be wired together as draw in the circuit?
4. Having +-18V is right to assume it won't need a capacitor in the output to block DC?
5. If all of this just doesn't make sense, do you have some other suggestion for my situation?

Thanks for any input

For sale are used pair JE-11P-1 input transformers from Jensen! The best!

Price: 120 EUR

Shipping worldwide.

Item location: Slovenia EU.

House clearing.

Found one more Pacific Microsonics PMD100 chip. $100 + shipping.

R

I would like to build from scratch (not a kit) a solid state stereo amplifier and would like some suggestions for a tried & true design (if one exists). This will be a learning project for me.

Ideally the amp has the following attributes:

1) all discrete components, no opamps or integrated circuits
2) all components are readily available from reputable sources
3) can be wired point-to-point, circuit boards are not required
4) power output should be around 20 watts (or more) per channel
5) capacitor coupled output is preferable
6) class AB, I'm aware of the JLH class A but would like something more conventional

The above requirements are based on my interest in early solid state amplifiers, ca. 1960's. I own several from that era and would like to build something similar with modern components.

I'd like to get a cermet pot that has an audio log law (Vishay L taper). Because the value is pretty high, these are not common (typically linear taper).

I found a reasonably priced one that has a Vishay 'F' taper. Below I copy a snip out of the datasheet. Vishay's F taper looks like it is the exact inverse of the L or audio log taper - pretty uncommon on audio amplifiers I assume. From this here old thread I understand that the F-tapered pot can be used just like an L tapered pot but you turn counterclockwise for increasing the volume. (So where the proverbial '10' on the volume dial is would be zero and vice versa.) So far so good.

Can I also turn the F-tapered pot into an L-tapered pot by reverse wiring it? By reverse-wiring I mean connecting the incoming signal to the 'c' terminal of the pot (see Vishay datasheet image below) and GND to the 'a' terminal? Which then, assuming the arrow at terminal 'b' for 'cw' (clockwise), causes the F and L labels on the log tapers to be swapped. (And the S and W curves to be inverted.) Is that correct?

The old post cited above appears to have contradictory advice on this point. (This post confidently says 'sorry but no', whereas the following one implies it would be no problem. Who among these posters is confused? Or is it just me who is confused?

Your help is much appreciated.

For a while, I’ve been stuck on repairing a channel of my Accuphase P-266. Initially, I tracked the fault to a shorted dual transistor (Q3: uPA75V) in the complementary differential input stage. I replaced both dual transistors with matched KSC1845 and KSA992 transistors. Additionally, I discovered that the R23 68-ohm fusible resistor was open, so I replaced it with a regular metal film resistor.

When I turned the amplifier on, nothing seemed out of the ordinary. However, when I began measuring the bias on T.P.1, I noticed the reading was very unstable. According to the schematic, the correct reading for T.P.1 in class AB mode is 88mV, but the value fluctuated wildly. It would jump well above 100mV and sometimes drop all the way down to 14mV.

I tested all the semiconductors in diode mode with my multimeter and didn’t find any anomalies. I’ve also replaced all the electrolytic capacitors. At this point, I’m stumped and unsure of what could be causing the issue.

Here are a few possibilities I’m considering:

1. Could the trimmpotentiometers be faulty? I’ve cleaned them with IPA and lubricated them with contact cleaner, but could they still be causing the instability?
2. Could the new complementary differential input transistors I installed be contributing to the problem?
3. Is it possible that a diode or a transistor, while testing well, is failing under load?

I’ve attached a photo of the repaired board for reference. I would appreciate any insights or advice!

I am selling a KIT for MM Phono Preamplifier with PCF802 tubes. Is a very simple tube phono stage with amazing sound. Very appreciated by all users.

PCB+socket pair+ PCF802 NOS tubes - 50 Euro+shipping. I can send in Europe only.

With the KIT I will send also the details to make it work.

Hello all. Occasional "look-in"er, but now need seek advice from anyone cleverer than me (that's probably all of you!) Interested in HiFi generally & have done some basic selector switch / volume control construction.

If you want the ultimate in ripple rejection and ease of use, this regulator is for you.

Key features:

• True floating regulator design
• Phenomenal ripple rejection (20 uV output ripple in my setup!)
• Soft start
• Stable with capacitive load
• No need for expensive and bulky high-power resistors
• 2x2 inch (50x50 mm) board footprint

It's now 32 years since Mike Maida authored National Semiconductor Linear Brief 47, describing a high voltage regulator based on the LM317. Lots have happened since then. National Semiconductor was acquired by Texas Instruments for one... And semiconductors have improved tremendously since the 1970'ies. So given that it's been 32 years to the month since LB-47 was published, I figured I'd do an update of Mike Maida's original regulator.

The main drawback of the original Maida regulator is that it requires at least 5 mA (10 mA worst case) to flow in the regulator for it to regulate properly. Typically, this current flows in the feedback network. For lower output voltages, this is no big deal. But for higher output voltages - such as the ones typically used in tube circuits - the power dissipated in the feedback network becomes quite significant, necessitating the use of 5~10 W rated resistors.
In addition, implementing soft start on the original Maida regulator is actually a bit of a challenge as it requires the use of high-voltage PNP or PMOS devices. These are becoming increasingly hard to source.
Modern voltage regulators also have much lower drop-out voltages than the LM317, hence, less power is dissipated in the regulator. As a result, the only heatsink needed is for the cascode device.

My "21st Century Maida Regulator" is based on the same topology as the original Maida Regulator; a low voltage regulator with a cascode in front to drop the voltage. I chose the LT3080 as it has a low drop-out voltage and needs only 300 uA (typ; 500 uA worst case) to operate. As described above, this minimizes the amount of power dissipated in the feedback network. Hence, only 2~3 W rated resistor types are needed.
The LT3080 is a low dropout regulator and only needs 1.4 V (worst case) across it to regulate. This minimizes the power dissipated in the LT3080. It doesn't even need a heat sink.
For the cascode I use a beefy NMOS - STW12NK95 (10 A, 950 V). I've used these in my other regulators and they work well. They're also capable of surviving the conditions present at regulator start-up without running into SOA limits.

My prototype regulator was adjusted to 420 V out @ 200 mA. There is no measurable ripple on its output. With 16 V RMS (50 Vpp) ripple in, I measure 20 uV (yes, micro volts) RMS of ripple and noise on the output of the regulator. Attached pictures show the transient response as function of load current and load capacitance. It looks rock solid to me...
The start-up time comes in at about 10 seconds. This does, however, require a resistive load. Without load, the start-up time is about one second as the output capacitor is charged through zener diode D2. The start-up is smooth without tendency to overshoot.

Using the values in the schematic, I only get about 1 mA running in the feedback network. In order for the LT3080 to regulate properly, at least 300 uA must flow in the LT3080. Hence, with Iout = 0 A, the current flowing in zener diode D2 must not exceed 700 uA. With R1 = 68 kOhm, I get 700 uA when Vin-Vout > 48 V. This isn't enough to guarantee reliable start-up across worst case mains variation, hence, the 330 kOhm "minimum load" seen across the output terminals in the image of the regulator prototype. I'll probably end up burning 1.5~2 mA in the feedback path to ensure that the regulator will start up and regulate properly. I'll also increase R1 from the value in the schematic.

After running for a few hours feeding 200~210 mA into my 300B amplifier, the LT3080 has reached 35 deg C. Clearly, no heat sink is needed for the LT3080 - the cascode still needs a heat sink, obviously. I have no audible hum in the speakers and the amp is dead quiet. I like it....

I plan to offer boards for sale on my website. I haven't done the cost calculations yet, but I figure I'll land around $10.

~Tom

Gentlemen,
i do not understand the search function here. Today I wanted to look up Hasling, a Danish name.
Even if put in parenthesis, it will turn up lot of results which have nothing to do with the original
phrase. The results are for HASL, an abbreviation I do not know and has no connection with the
name I look for. This is not the same as with a Ggl search, so what is going on here, how can I
adapt to this strange routine ?
Thanks.

Hey there,

I want to replace the LM358 in this circuit with an AD8656. As I am a beginner: Can you please help me and say whether this will possibly work? The LM358 is hissing.
It's SOIC8 right?

See a picture of the circuit a bit farther down.

Best regards

Hi!
I'm having a strange issue with davis acoustics 3-way speakers made in France. They're not mine and came to me with 2 blown mid drivers (I think they are 17KLV6A). I sent them to davis acoustics for repair because I don't have the voice coils for these speakers. In the end they changed everything (spider, coil, kevlar cone and suspension) on both and they worked perfectly for about three months but now one is back, again with a blown mid driver (voice coil is open). The amp is a Vincent SV-233 that works perfectly, the volume stayed below 1W of output and there was no noise prior to the failing, both woofer and tweeter are fine. I've heard from someone with the same problem that Davis recommended placing a cap in parallel with the driver to solve this but he sold the speakers and doesn't remember what the value was. When I contacted Davis Acoustics they had no trace of this and could not tell me anything more. Here's a basic drawing of the crossover (I measured the inductor with a simple component tester so I don't know how close this is to the actual value). All other component measure within 5% of their announced value with 0.5% voltage loss on the electrolytic (47µF and 0.1% on the film cap (6.8µF) and very low esr. Does this filter seem adequate and would there be some kind of mod that could protect these drivers without altering the frequency response ?

Anyone know of a nice ADC module like this one sold in the U.S.?

https://www.audiophonics.fr/en/inte...cm1804-rca-to-spdif-24bit-192khz-p-17047.html

hello,
I am designing a recoring device based on CM6535 from C-Media.

I purchased the IC from:
CM6535 - CM6535 USB 2.0 Audio Chip with Tri-Colors PWM LED driver

after making a small test PCB - i connected it to my linux device, and it does get detected by lsusb commmand, but no audio i/o avaliable (checked using aplay -L).

I have another dev board for a similar IC CM6510 - and it does work well.

my thughts are that a firmware is needed, but i have no idea how to even begin.

can anyone please direct me to the right direction?

thanks allot.

Have an interst in fix my sounding gears, such as CDP , ampliefier to upgrade my sounding sytem,also enjoyed the fun and happy to learn more from diyAudio always!

thanks

fc2fc

Is it generally safe to leave FW amps powered on without being connected to an input or a load? I am comparing some amps back to back and find they sound fairly different after 15 min or so of warming up. The idea is to let them heat up, switch power off momentarily to plug in the source and speakers, then power back on to get straight to a "warmed up" amp sound signature.

I have had the following motor kit for quite a few years now.

https://simplemotor.com/shop/motor-kits/kit-6/

I modded it to use an actual shaft, bearings and early on I learned the magnets tended to fly off the rotor at real high speeds so I used some neodymium magnets and epoxied them in place to where the epoxy is flush with the ends of the rotor.

Was a pain to properly balance the rotor so that it doesn't vibrate when running fast.

I used two 12 volt coils instead of the single coil and I used a 7805 to regulate the voltage to the hall effect sensor.

Here's the schematic.



Here's the motor.




The motor will increase speed until about 23Vdc and above that the waveform at the coil starts to have a higher duty cycle which tends to not allow any real speed increase.

Torque is not that great either, but it's fun to experiment with.

Currently I need to replace the hall sensor and transistor for it to operate. I could simply replace the parts with like parts, but there might be parts better suited for this application.

Getting a little side tracked for a brief moment.

Two things that I've been kicking around in my mind.

1. Get another rotor and build it the exact same and put two on one shaft with one clocked so the magnets are 45 degrees offset from the other rotor's magnets then use another separate transistor, sensor and coils. That would increase torque while likely keeping the higher speed,

2. Get a larger pipe so that I can use more magnets and possibly more coils. That will increase torque, but lower speed. Only reason I don't do that is I am not good at making stuff like that and I'd likely wind up with a rotor that is not balanced at all. What I'd need is a small pipe or other plastic thing that already has 8 or more flat sides on it. I've even thought of making a really large rotor maybe 5" and fill it with equally spaced magnets.

Now if I really wanted to get adventurous I could build a larger rotor and locate coils and sensors 120 degrees apart from each other. I'd then have a three phase brushless DC motor.

That said back on the topic at hand.

Four things I have questions about.

1. Is there a better hall effect sensor than the 21E for this application?
2. Is there a better transistor than the TIP-107 for this application? Perhaps a power MOSFET?
3. Is the 12 volt coil ok or would a 6 volt coil be better and would the coil type I used be best or is there a better type?
4. Is it better to have the coils in series or parallel?

In an effort to bring some sort of meaning back into the lives of Audiophiles in the Pacific Northwest, the "Sound DIY Club" along with the "Pacific North West Audio Society" (PNWAS) are joining up in order to hold "The Run What 'Cha Brung Audio Fest". on Saturday September 9th It's open to any DIY'ers that would like to show their home-built gear (in my case inserting the word "attempted" would be a worthwhile additional note!)
This is not a contest, or any sort of competition... it's merely to exhibit or demo any stereo gear that has been modified or built!

Now for the best part: It's free to attend, as well as free parking.

Mercer Island Congregational Church
4545 Island Crest Way
Mercer Island, WA 98040

The event will take place in the basement and is near the lower parking lot of the Church. Right now, it appears that it will start around Noon and start shutting down at around 4 to 5 O'Clock, although we would like to begin a bit earlier (Check back to see if the start time has changed).

Hope to see you there!

Best Regards,
Terry Olson

I noticed that the newer incarnations of the Excel drivers are advertised to have a newly designed motor. Most impressive is the 5x reduction in Le, e.g. W22NY001 vs NY003:
http://www.troelsgravesen.dk/W22NY003.htm

Does anyone know of a white paper or other publications that show the motor design?

I am building a center channel channel for our living room Home Theater. I had several drivers already I decided to use, 10 3” FR drivers and a Morel 1” dome. I decided to add the 1” dome above 10k as it has pretty smooth off axis response and the FRs get a little ragged that high.

I plan on wiring each set of 5 in a Bessel alignment to keep horizontal beaming to a minimum.

My question is horizontal spacing. I have no limit up to 60”, but I was wondering if anyone had an ideal of optimum spacing with them being in a Bessel, I really don’t know where the acoustic centers will be and how they will sum. I really do not feel building test baffles and measuring and was hoping someone on here would have a tip.

Thank you!

For the first time i see cap on i2s lines, what's purpose? Thanks

I've got this RCD-951 which is in good condition and seems to play fine however one channel is faulty. If I turn the volume up that channel goes from silent to DC popping and pushing the cone of the test speaker.Just finished reflowing all joints.

Does anyone have any ideas to help me out with this one?
Thanks!

Hi all.

Recently bought a pair of these speakers. It became very clear that one tweeter was outputting at a small fraction of the other - mono white noise signals sent to the HF inputs showed this very clearly. Tested both tweeters by removing them and - carefully - wiring them straight to the amp's output, and they performed identically. Refitted them, and the same as before. Took the amp's speaker cable direct to the crossover's output to this dodgy tweeter, and it tweeted loudly as it should. So, a fault in the crossover.

No obvious dry or bad joints, all resistors tested to be correct values, and inductors tested for continuity. I have no way to test the caps, but none are looking 'faulty' - no leaks, no bulges. So, most likely a capacitor has gone faulty. No biggie - I will replace them all.

Q's - what type of poly capacitors should I choose? Any benefits from staying with what's there (round polys) or going flat-rectangle!
And, can anyone explain what's going on with the circuit? Why have two caps in parallel when one (larger) one would do? And, why have an electrolytic in parallel with a poly?

The schematic is mine, but I've rechecked it a few times. Thanks!

Hi,

I have a couple of XMOS USB based DAC's that have given up the ghost.

I'll focus on the the ifi micro idsd bl as it's the nicest (the other one is a minidsp u-dac8). The issue with the ifi is definitely the XMOS USB daughtercard because SPDIF works fine.

So here's the DIY Q, how easy is it to fix the ifi with a DIYINHK or MCHStreamer USB to I2S card?

Let's assume it's I2S and I have figured out the pins and made a connection "bridge".

So its more to do with the XMOS firmware, the ifi does PCM 32/768, DSD512 and double speed DXD, I'm assuming multiple I2S channels, maybe four and not sure if the DSD just uses I2S as well?

Say I was "happy" with 32/384 and no DSD, would a "stock" firmware work?

I can solder but I'm no programmer and so I'm just assessing how feasible this is.

Thanks

Hello everyone,

I'm an audio enthusiast, but I also appreciate convenience (read lazy). Currently, I control my streaming audio through a voice assistant to select songs, adjust the volume, etc., and it works great. In addition to streaming, I also have a turntable and a vinyl collection. While I still need to get up to change records, I enjoy controlling the volume of my turntable through my voice assistant as well. To achieve this, I hacked together a simple device using an Allo R-Attenuator and an ESP32, which has worked well so far. However, I'm now considering upgrading some of my vinyl gear, and the new setup will have a balanced output. This brings me back to my volume control challenge…

Does anyone know of a device I can use in place of the R-Attenuator to control the volume of a balanced (XLR) connection via I²C? I looked online for ready-made relay attenuators with XLR input/outputs, but they all seem to require a potentiometer for volume adjustment.

Thanks in advance!

- Justin

I read this forum a lot. Solid info here. Recently designed and 3d printed synergy horns loaded with silverflutes and peerless tweeters all base off info from this forum. Joined to enquire about roar15 driver selection. Thanks.

Hello,

New member here. Been reading a lot about t-line boxes and learned some simulation in Hornreesp. Have a few more questions about the same which I will eventually post.

Thanks and great community here with loads of knowledge.

I'm a 300B SET enthusiast and electronics hobbyist looking to expand my knowledge on circuit design and system integration. I recently bought a calibrated mic and have driven myself nearly insane trying fifty different speaker/listening spot positions and room treatment options! I must have done at least 350 frequency response sweeps in the last month alone. Now I'm working on my first attempt to integrate a subwoofer into my 2ch only system. Impressed with the Rythmik F12 SE's ability to disappear so far. Unfortunately I'm also working on restoring the piano black finish as one of the corners was chipped during shipping. Stabilized the chip with super glue, filled with baking soda cemented with more super glue. Sanded flat and painted above flush. Just waiting for the paint to cure so I can wet sand and polish it to match. So far so good.

Hello there,

I've just completed my take on a 6AF11 champ type guitar amp (see attached.)

I thought it sounded like muddy garbage until I plugged in a guitar pedal with a buffered output.

I thought "huh, the cable im using must have too much parasitic capacitance!" So I tried using a short 6" jumper cable straight from my guitar to the input jack.
No dice.

I have a pedal testing rig with a switchable buffer, and no matter how much cable length I added, turning on the buffer would make it sound good again.

Then I tried removing the capacitor between the grid and cathode of the input triode. Nothing changed. I thought for sure it was the interelectrode capacitance shunting high frequencies. Nope.

I'm now planning on adding a built in mosfet buffer circuit to go in front of the first triode, but I can't help but feel that I'm missing something.

Any ideas?

1 piece shunt reg from Guido Tent for -15V supply.Iv used this years back for my tda1541 dac chip.
Price 20 euro plus 4,50 euro shipping inside EU.

For sale a quad of BB Opa 627AU single opamps on adapters.This are original chips!I buy this way back from Audiotuning Germany.Price is 60 euro plus 5,90 euro shipping inside EU!

Any one knoe what I can use these for? they go between amp and speaker rated for 15 "audio watts", what ever that means.

I am mostly building Pass projects.

Regards, Vinay

The three-stage amplifier (sometimes referred as Lin topology or Blameless by D.Self) is a fundamental design in the realm of analog electronics, particularly in audio and power amplification. By dividing the amplification process into three distinct stages, each responsible for a specific part of the signal processing chain, this design offers a robust solution for various applications. The stages include the input stage, the voltage amplifier stage (VAS), and the output stage, each presenting its own advantages and disadvantages.

In the input stage, a differential amplifier is typically employed, which plays a crucial role in preparing the incoming signal. This stage is essential for rejecting unwanted noise, especially common-mode noise, which refers to interference affecting both input terminals similarly, such as electrical noise picked up along signal cables. The input stage often operates as a transconductance amplifier, converting the input voltage signal into a proportional current that will be further processed in the subsequent stage. By maintaining the signal's integrity, the input stage sets a solid foundation for the amplification process.

One of the primary benefits of the input stage is its noise rejection capability. The differential amplifier excels at filtering out common-mode noise, making it particularly effective for sensitive applications where clean signals are critical. Additionally, the input stage typically presents a high input impedance to the source, minimizing loading effects and preserving the original signal's quality. However, this stage does come with challenges. It requires precise matching of its components to ensure optimal performance, which can increase design complexity and manufacturing costs. Furthermore, the complexity of the differential amplifier adds a layer of difficulty to the overall circuit design, making it harder to tune and optimize.

The voltage amplifier stage (VAS) is where the majority of the signal's voltage gain occurs. Once the input stage has prepared the signal, the VAS amplifies it to the desired voltage level. This stage functions as a transimpedance amplifier, converting the current output from the input stage back into a voltage signal. The VAS ensures that the signal possesses sufficient voltage swing to drive the output stage, which ultimately handles the current demands of the load.

The VAS provides significant advantages, primarily in its ability to deliver high voltage gain. This stage is essential for amplifying low-level input signals to the required amplitude, ensuring that the original audio quality is preserved during the process. Moreover, the separation of voltage gain into a dedicated stage allows for easier fine-tuning and optimization of this critical aspect of the amplifier's performance. However, the VAS is not without its drawbacks. If improperly designed, it can introduce harmonic distortion or other artifacts, particularly when handling high-voltage signals. Additionally, voltage amplifiers can be prone to thermal instability, necessitating careful thermal management and compensation.

The output stage is responsible for driving the final load, usually a speaker or other low-impedance device. This stage provides current amplification, ensuring that the amplified voltage signal from the VAS is delivered with enough current to effectively power the load. Typically, the output stage is designed to operate efficiently while minimizing distortion, particularly in Class AB configurations, which offer a balance between efficiency and linearity.

The output stage presents its own set of advantages. It supplies the necessary current to drive low-impedance loads, ensuring that the amplifier can handle high-power applications. Class AB output stages are particularly efficient, striking a compromise between the linearity of Class A amplifiers and the efficiency of Class B designs. This reduces heat generation while maintaining high-quality output. Nonetheless, the output stage also generates significant heat, especially in high-power applications, which requires careful thermal management. Additionally, the output stage must be accurately matched to the load to avoid instability and distortion, complicating the design process.

When considering the three-stage amplifier design as a whole, several compelling advantages emerge, alongside some challenges that designers must address. The division of amplification into three distinct stages allows for performance optimization, enabling engineers to tailor each stage for its specific function. The input stage focuses on noise rejection and signal conditioning, the VAS provides critical voltage gain, and the output stage delivers the necessary current to the load. This modularity enhances flexibility, making it easier to adapt the design for various power levels, load types, and performance requirements. Moreover, the inherent noise immunity provided by the differential input stage is vital for high-fidelity applications, ensuring that the amplifier remains resilient to interference.

However, the three-stage amplifier approach does have its complexities. The intricacies of tuning and matching each stage require meticulous attention to detail, potentially extending development time and increasing costs. Managing heat becomes a common challenge, particularly in the output stage, as excessive heat can lead to performance degradation or component failure if not properly addressed. Furthermore, variations in the characteristics of transistors or other active components can affect overall performance, particularly in the input and VAS stages, necessitating careful design and testing.


Inspired by the three-stage topology, I set out to create opamp schematics consisting of discrete components that offer distinct design advantages over commercially available opamps. Main goal of my design was achieving a high open-loop gain within audio band frequencies. To accomplish this, I employed an advanced type of voltage amplifier stage (VAS) compensation known as two-pole compensation. This technique creates two poles on the Bode plot, effectively expanding the open-loop gain at higher audio frequencies. As a result, the loop gain plot exhibits a steeper roll-off of 12 dB per octave, compared to the simpler Miller compensation method, which rolls off at a rate of 6 dB per decade.

To optimize LTP, PMP5201 and PMP4201 matched double transistors were used. These transistors are essential for ensuring precision in the input differential pair, current mirror, and constant current sources. Using just two matched devices also simplifies the bill of materials.

Update: newly redesigned opamp uses all 2N3904/3906 transistors as those offer higher bandwidth resulting in better performance. See post #17

Opamp specs:
THD (G=+1, 600R, Vin=1.41Vrms, 1kHz): 0.000004% or -147dB
THD (G=+1, 600R, Vin=9Vrms, 1kHz): 0.00005% or -126dB
THD (G=+1, 300R, Vin=1.41Vrms, 1kHz): 0.000005% or -146dB
THD (G=+1, 300R, Vin=9Vrms, 1kHz): 0.000754% or -102dB

BW (G=1): 4.25MHz
Noise: 5.77 nV/rtHz
Voffset: 0.1mA
AOL (100Hz): 118dB
AOL (20kHz): 72dB

This is posted as a concept and does not perform correctly in frequency response simulation yet.

I would like to convert existing 3-way loudspeaker boxes (currently with passive crossovers) into an active system. I am looking for 3-channel amplifiers with 3-way active crossovers, all together assembled ready on one single circuit board for each loudspeaker box and ready for connection. (power AC or DC; input; 3x output to speakers; nothing else). It is sufficient if the three amplifiers for one box together produce around 70 watts @ 4 ohm. It does not have to be "high end".
I have not found anything suitable online so far, only 2.1, 5.1 amps etc., plate amps only as 2-way, perhaps I am not using the right keywords.
Can anyone give me a hint, perhaps links to corresponding offers?

I have some boards that are in need of a good home. Actually two good homes.

Anybody has these cabinets for sale?

Payment in parts or money-combination of both.

R

Does anyone have a circuit or even a block diagram ? Or a PCB photo or any real technical information ?

And one for sale ??

Thanks Dave

Here's something I've always wondered. I use Monster Power conditioners, a 1600 and 2500. I paid next to nothing for them and use them mainly for the surge protection. I did notice a slight improvement in sound quality from my equipment but nothing earth shattering. If I were to add a mains filter to each device after the power conditioner would it make any positive difference? Or is the filter in the power conditioners do all the smoothing that can be done so to speak?

I have a new office and I think that dictates I build a new sound system for my desk.
Budget is under $400. Considering mostly low power drivers as I don't need the power

Here are the 4 different styles I have been going back on

1. MTM with the two woofers angled directly at me so it acts a single point source
   - I could do this with some cheaper woofers ($20-50) and a decent tweeter.
   • This allows for just a 2 way keeping the crossover decently cheap and allowing more cash for the drivers
   • I can always just slap a sub under the desk if I don't want to port this which would keep the foot print very small
2. 3 way - tweeter, mid dome, woofer
   • I do think a mid dome would be good for near field as they are supposed to be very delicate sounding and are all quite low power
   • This requires a 3 way crossover so a bit more money on the XO
   • Most dome mids seem to have issues unless you drop a lot of cash (considering dayton's fabric dome here)
   • I don't know if a 3 way is worth it for near field
   • The box size required for a woofer that would hit low enough to have a mid dome make sense might defeat the purpose
   • I have considered sticking a low hitting woofer on the side of the enclosure. This keeps it slim but I can always go very tall to get volume
3. Traditional 2 way
   • Cheap XO and only 2 drivers let me spends more money on drivers
   • This opens up driver selection a bit though I notice drivers usually go from around $50, then to $100, then its a HIGH jump to $300-500
   • This is boring, this is what I already have at my desk, but if this is best then let me know.
4. Full range
   -Considering a Mark Audio 3" in an oblong enclosure so I don't need to notch the baffle.
   • Seems a lot of the full range stuff needs a really large box which I feel defeats the purpose a little bit. I do have some W5-2143 in my pool room and they are awesome for what they are.
   • I sometimes find full range drivers harsh when listening near field
   
   
   Drivers I am considering
   
   MTM
   -Hivi Q1R tweeter (15w) - Seems very well behaved
   https://www.parts-express.com/HiVi-Q1R-1-1-8-Textile-Dome-Tweeter-297-417?quantity=1
   
   -Dayton Aluminum 4" woofer - Seems like a good bang for the buck
   https://www.parts-express.com/Dayton-Audio-DA115-8-4-Aluminum-Cone-Woofer-295-328?quantity=1
   
   -Peerless 830656 5.25 Woofer - I do very much like my Peerless 8" so I imagine this would be good
   https://www.madisoundspeakerstore.com/approx-5-woofers/peerless-830656-5.25-woofer/
   
   3 Way
   
   -Hivi Q1R again
   
   -Dayton Dome - Cheap and might be a good entry into domes
   https://www.parts-express.com/Dayto...Fabric-Dome-Midrange-8-Ohm-285-022?quantity=1
   
   -Tang Band Dome - This is going to take a huge chunk out of the budget
   https://www.parts-express.com/Tang-Band-50-1426SE-2-Fabric-Dome-Midrange-264-855?quantity=1
   
   -Dayton LW150 6" low profile - this could be side mounted and would hit pretty low
   https://www.parts-express.com/Dayton-Audio-LW150-4-6-Low-Profile-Woofer-4-Ohms-295-255?quantity=1
   
   - Dayton Reference RS150 6" - This would likely need to be forward mounted so this would take a lot more space but I might be able to get away with 1st order XO
   https://www.parts-express.com/Dayto...ference-Paper-Woofer-8-Ohm-295-573?quantity=1
   
   2 Way
   
   - Morel MDT12 Tweeter - I don't know a ton about this but it looks good and I could mount it very close to the woofer
   https://www.madisoundspeakerstore.com/morel-soft-dome-tweeters/morel-mdt12-1-textile-dome-tweeter/
   
   -Hivi Q1R again
   
   - SB SB21RDCN Ring Dome Tweeter - This thing looks cool and it would fit the bill. Very linear response
   https://www.madisoundspeakerstore.c...-sb21rdcn-c000-4-ring-dome-tweeter-neodymium/
   
   -Hivi L6-4R Kevlar Woofer - very well behaved FR and it looks cool. Would be easy to XO
   https://www.madisoundspeakerstore.com/hi-vi-woofers-6-7/hi-vi-l6-4r-6-woven-kevlar-cone-4-ohm/
   
   - SB SB12CACS25 Ceramic 4" Woofer - Well behaved, will hit decently low if I port it. Ceramic looks cool
   https://www.madisoundspeakerstore.c...ustics-sb12cacs25-08-4-ceramic-woofer-8-ohms/
   
   -SEAS Prestige CA12RCY 4.5" Woofer - I hear very good things about this woofer
   https://www.madisoundspeakerstore.c...e-ca12rcy-h1152-4.5-coated-paper-cone-woofer/
   
   - Scanspeak 15W 5" Woofer - Boring looking but I'll bet it sounds great
   https://www.madisoundspeakerstore.c...anspeak-15w/8434g00-discovery-5.25-midwoofer/
   
   Full Range in Circular enclosure
   - MarkAudio Alpair 5 3.5" Full Range - I hear the mid range of this is magical.
   https://www.madisoundspeakerstore.com/approx-3-fullrange/markaudio-alpair-5-gen.3-gold-3-full-range/
   
   -Scanspeak Discovery 10F 4" Full range - Again, I hear really good things
   https://www.madisoundspeakerstore.c...canspeak-discovery-10f/8414g-10-4-full-range/
   
   -Fostex FE108-SOL - I have heard nothing about these but Madisound says good for a horn. Before my recent 2 way I listened to a folded horn design of my own making for about a year. That build now lives in my kitchen and still sounds pretty great for a $10 driver. Amazing what a proper folded horn can do
   https://www.madisoundspeakerstore.c...ex-fe108-sol-limited-production-4-full-range/
   
   I honestly cannot decide what to do here so let me know your opinion. Thanks!

I have been using a couple of 12v lead acid batteries to power a modified Quad 33 pre amp and a QED Digit which has been modified for 2 separate power supplies. The batteries are automatically switched over to smart chargers when not in use, and one powers the Quad while the other powers both sections of the Digit. The Quad battery is 3.2ah and the other is 2.8ah, and is getting on a bit - my question is, should I have separate batteries for each section of the Digit, or, if not necessary, what would be the recommended size for a single?

Unusual to see three First Watt clones for sale at the same dealer: https://www.ebay.co.uk/str/emporium...cat=2581724017&_trksid=p4429486.m3561.l170197

I am very proud about my DIY backloaded horns based on the FOSTEX FE 108 EΣ sigma 4"inch fullrange speaker.

The cabinets construction plans were downloaded from Fostex.

I have Recently modified them in a two way speaker replacing the Fostex 4inch with an ETON 4 ohms midwoofer and adding a 4 ohm tweeter.

My crossover is divided in a 4th class for the woofer and a 3d class for the tweeter.Still expirimenting on these crossovers but the results up to now are excellent.

Fostex FE 108 EΣ sigma 4"inch are great speakers but a little bit harsh in the mid and higher frequencies.

Now the lower frequencies have more warmth and power and the mids are warm and sweet sounding.

When i am placing the speakers close to the back wall the bass production becomes huge and fills the room 40 (sq.meters)

Please show some consideration about the appearance of the units (marks and scratches) .I will update with photos of the final painting, veneer etc
Any questions or thoughts about these speakers are well accepted.

Thank you

OK, so this is my own SP-10 mk3. I didn’t have a lid for the plinth so had one made locally, it’s a light brown smoke, made of real Perspex.
As usual it would not stay up and I agonised over the various hinge modifications. I dismissed tapping out to M5 or a rivnut/captive-nut solution as both meant removing material and potentially making it weaker. So a good friend made two 3mm thick reinforcement squares. The existing M4 thread was kept and the thread on the addition pieces was tapped deliberately at the same wrong angle as the existing distorted thread, giving far more material and thread. Now this lid and a customers original one stay open really well.

The EPA-500 and EPA-A501 G shown here will be available when I finish the FR-64s rebuild.

God out of the past a Krell kid ....
Any body thinks is possible to convert this to a usb device ?
I think that usb pins exist in the contact
will this ever work ?
anybody look at this idea before ?

Is the maximum frequency of AMT driver limited by plead depth (1/4 wavelenght)?

I was thinking of AMT ribbon plead geometry and I become concerned how depth of pleads could affect the sound. Becose once the plead starts pumping the air, the sound from the opening part of the plead, the closest part to air of listener will arrive sooner than sound generated at the bottom, the deepest part of the plead.

As long as the frequency is long enough compared to fold depth, all is ok but when the fold depth becomes half as long as wavelenght it tries to reproduce, it will take so long for sound from the back of the plead to travell to the opening/front of the plead that at that time, the plead is moving at 180 degrees out of phase which I expect will cause significant phase cancelation, I expect the frequency reponse above 1/2 wavelenght to have comb filter like response with deep nulls due to phase cancelation.

Is my theory correct or does this not matter and you can make for example AMT tweeter with 1 inch deep folds and have it be flat to 20KHz?
I asked ChatGPT and it told me about the 1/4 wavelenght limit which is 4.2mm for 20KHz.

A member asked how to wire compound nude dipole woofers. After wiring was solved, vivid discussion about the sanity of compuond (opposed) arrangement benefits.

I happen to to have two SEAS 10" woofers 25F-EWX H108 from 1982, so here are my measurements and pics. I used serial connection for double woofers, naturally with opposed compound, the backside woofer was in inverted polarity. With parallel wiring sensitivity would increase or distortion for same spl would be better.

Test was performed indoors with a woodden ladder frame. the frame and the room both make artefacts in measurements, but they are similar in every series.
I use ODAC, Vincent SV-129 (volume at 9 o'clock) and UMIK-1 mk2 with REW.

Nearfield measurements of a single driver in this post.

Hello,

I'm about to build a CD Player out of an old Rom-Drive and a controller Board from Aliexpress.
So I need a DAC board next, to convert the SPDIF signal.

Mostly the whole thing is about building the player, so I look for a good solution without breaking the bank.
About 100€ would be great, and if I could upgrade it with a tube buffer, even better.

Do you have any suggestions for me?
I would prefer a kit over a complete module.

Even if it is a finished board, is the Lampucera still a good go?

Thank you and with kind regards,
Felix

For repair amplifier LUXMAN L-435 i need original pair transisors 2SC3182N but may be two pair 2sc3182n/2sa1265. One moment - I live in Belarus

Hi,
I recently bought a second hand Cyrus III with PSX-R which works okay on all inputs except the phono. I only get sound from the left channel when I connect my turntable. The same turntable and cables work fine into another amp.
I've cleaned the input sockets but still no right channel output.
Any ideas what could be causing this and any other tips for trying to fix the problem would be gratefully received.
Mark.

Hello, my name is Dan. I like to repair vintage audio equipment. It’s my hobby.

Hello everyone,

My name is Vincent. I really like listening to music on my self build speakers. I have build a 2 way bluetooth speaker with a b&c de 250 and a Dayton Audio RS270p-4a. I also build a 3 way speaker pair that I currently use as my pc speakers. I do want to replace the 3 way speakers in the future with a 3 way speaker with compression drivers, since I do love the dynamics they can deliver.

I manged to buy a pair of new old stock Seas 21TV drivers.
It looks like I need to match this to a high sensitivity tweeter above 3000Hz .
I thought it might work well with my Almarro 205mk2 SEP which is about 4.5 watts.
Any experience or suggestions?
https://www.google.com/url?sa=t&sou...IQFnoECBgQAQ&usg=AOvVaw3vQCZ4eSGGYe7fAXgYFi-j

The MeanWell UHP-500R-48 power supply is passively cooled through coupling with the chassis it is mounted in.

I could use thermal paste, but it can be messy and hard to apply in a thin even layer over a large surface, I am leaning towards using a large thermal pad.

The thermal pad should be ~232mm x 81mm for the power supply,



And I am going to need one for the amplifier board heat sink to couple it to the chassis heat sink, 3e Audio TPA3255 TPA3251 480-1-29A, 120mm x 85mm,



Suggestions on what brand and series thermal pad would be a good choice?

Thank you, David.

How calculate the box to build BLH+FLH (like VOTT type) with woofer BSA PL-10

I want to know if I need to tune the horn at the same freq Back and Front ?

Another question, do you know a good online software to building the enclosure with shematic and wood parts size ?

thanks in advance 😉 !

nicK

About 9-10 years ago I build a 2-way T-line speaker set. They are a quite bad design, but I did put a lot of effort into building a nice looking, stiff and heavy box.
I did some calculations for the bass-mid driver and if I had a bass reflex with the same volume of box, the result should be really quite good.

There are a few things I am wondering about and would like your thoughts on.

1. If I simply close the end of the T-line, and put a vent there instead, will the enclosure behave like a bass reflex? Or will it behave differently. The cross section of the T-line is about 250cm2, which is much much bigger than any vent cross section. The total volume of the box is about 90 litres.

2. Would the placement of the vent matter? It will be most easy for me to add it to the new piece of wood I will close the end of the T-line with, but I could make a hole in another section if that would be a better choice.

3. For the ideal tuning of the box, the first harmonic of my vent will be at around 1kHz, while the crossover is a 4th order butterworth at 1.5kHz. I can't really find what happens if I were to implement that design. I assume it is not great, but I would like to confirm that before I make the bass extension (slightly) worse by tuning the box a little higher with a shorter port. I am already minimizing the diameter of the port while keeping air speed in check at the maximum volume level I intend to actually use.

Thanks for your input.

My desired output power would be 5/7w at 8 ohms and twice that at 4 ohms. the reason i want to lower the voltage is becuse i dont need that much output power and i would have trouble heatsinking said power.

Hi,

I am tinkering with different designs for a 3 way hifi speaker I would like to build in the future. I was considering a B&C DCX354 to use with the FaitalPRO LTH142. I have the following questions:

• If you heard the B&C DCX354 what did you think of the sound?
• What are your thoughts of using a coaxial compression driver compared to two seperated compression drivers for the mids and highs?

I did found a test bench article from AudioExpress. The results show a very high distortion figure of 12% around 4.8 kHz and the frequency response is not really smooth. This makes me wonder if a single 1.4 inch compression driver with a smoother response will be a better option than the DCX354.

Reasons for my interest in this driver:

• One point of sound for the mids and highs
• Smaller build volume by only using one horn
• The mid and high of the coaxial compression drivers are likely tonal matched
• Not too expensive

On Thursday I had a power surge and I’m now thinking it may have caused issues in my main system where I now have buzz on every amp I try.

My original setup was - Gold Note Dac/streamer -> Academy Audio Volume control -> BBA3 front end preamp -> F4 monoblocks -> Magnepan LRS+ (fully balanced system)

I switched my system before I noticed the hum where I put an Aleph JZM right after the Academy Audio into some Zu speakers (fully balanced). I kept having buzzing on the left channel. I swapped the Aleph JZM for an F5 (using rca out from the Academy Audio) and same issue. I then tried an M2x with the same results but louder buzzing in both channels.

I tried removing the Academy Audio and went straight from my Gold Note as a preamp to the Aleph JZM but still have buzzing but in both channels.

I tried XLR and RCA shorting plugs on each amp and all of them are silent. I took the Aleph JZM and F5 into my office system and both of them were perfectly quiet there.

I also tried plugging in a different source to rule out the Gold Note and had the same noise.

The only way I am able to get a completely quiet system is to go

Gold Note Dac/streamer -> BBA3 front end preamp -> F4 monoblocks -> Zu Audio Unions (all balanced)
Adding the Academy Audio preamp back in caused buzzing and noise on just the left channel.

I unplugged my whole system hoping maybe it needs a reset. Going to try that after a few hours of rest.

Any other recommendations to check after a power surge?

I'm planning to build this minimal circuit. The source is an I-out DAC. One opamp creates a virtual ground–strong enough for 32ohm HPs to sit on, the other a DAC filter+driver for the headphones.

The main candidate has been the OPA1622 so far, for its large output current. Then I discovered the OPA1688, weaker but maybe enough for me and easy to solder. Also good with capacitive loads. Simulating both I found something puzzling...

Crucial here is that DC offset is extremely low I guess, since headphones are DC coupled. On paper the OPA1622 has 0.1mV VOS, the OPA1688 slightly more 0.25mV. But when I simulate both opamps in LTSpice, the 1622 shows quite surprisingly high(er) offset!

OPA1688: 0.22mV
OPA1622: 64mV

This is simply comparing the voltage difference between the opamp's output and VGND.

Are there any probably explanations to this? I read somewhere @johnc124 explaining that the OPA1622 model is quite detailed, not a 'macromodel'. So maybe there is a problem with the OPA1688 model not being accurate enough? Or this particular application creates troubles with the OPA1622 for having bipolar inputs, compared to the CMOS inputs of the 1688?

Just trying to double check which opamp is more suitable for this application and all comments are very much welcome!

For a project I'm working on, I'm having a hard time deciding wich way to go. I have several options and I'm sure they can all lead to good results.
Base parameters are a 12" (mid)bass handing over to a waveguide/compression driver around 1000Hz. Fully active + DSP.
If there will be subs, they need to be integrated in to the main speakers. There's not enough space for separate (placed elsewhere) subs. Playing only music, no HT.

Drivers are not set in stone, my question is more about 'design philosophy'

I could do a FaitalPRO 12PR320 in a sealed, 50L box and below that a 12" sealed sub taking over somewhere around 80-120Hz
or
a single FaitalPRO 12RS430 in a 90L vented box running from 35 to 1000Hz.

Both provide enough SPL, both give me ~100L boxes. The 2 way is simpler and cheaper but uses a 90gr Mms cone all the way down. The 3 way uses a lighter 50gr Mms cone and hands the lowest frequencies over to a dedicated driver.

Would the more agile driver relieved from the lowest notes results in a nicer midrange and will the sub provide a better bottom end?
Or will it be a PITA getting good integration of the subs and will the 2-way not only be cheaper and easier to build but also a more coherent sound?
What would you do and why?

I have retired and wanted to spend time listening to a loved audio system from earlier years. I am now 75. My background is an Electrical Engineer from many moons ago and have practiced medicine for decades. I have an ADCOM GFA 5500 bought in the 90's with one channel not working and some distortion in the other. I would love to have advice how to go about making the diagnosis for what looks like the need for electro-surgery or get psychotherapy to accept the equipment's mortality. Thanks for listening!

Hoping someone can confirm for me.

These old woofers have loose terminals. I’m going to glue them back in place, but wanted to confirm this silver ring should be on the terminal itself rather than attached to the cone.

It seems like it’s embedded in the glue, must have moved down with gravity while things were hot.

I am wondering has someone built a DIY or a commercial home speaker with two or more lines of the same type drivers side by side? I would love to see some examples. I know it is rare, since it requires at least double the amount of drivers than normal single line.

I am not interested in the typical line array PA speakers, which of course often have two of the same type woofer/mids in the same enclosure. Those have the tweeter in the middle and arrays are assembled from separate enclosure modules. In other words, I am only interested in a single enclosure, multiple line speakers for home use.

The inspiration is the seas a26 kit (smooth 10" crossing to a single capped tweet).

I think the b102 can be run wide open with a notch.

I figure sensitivity matches with a eminence b102 and a morel cat378 (with 5.6uF).

https://www.madisoundspeakerstore.com/horn-tweeters/morel-cat378-morel-horn-tweeter/
http://www.eminence.com/pdf/Legend_B102.pdf


If notch 2,800hz for the woofer (10uF + .33mH), both can be 6db down around 2,000 and have matching sensitivities (around 92db).

And you can even slide the tweet back if time alignment is your cup of tea.

Whatcha think ?

Hi,

I’m starting this build because I’m a newbie and I would like to ask some simple questions on how to build speaker cabinets.

Today, together with my friend, we did some trials on a junk piece of wood.

Is this correct, or the 45 degree should start from the inner part of the speaker?

Thanks

I am looking for a DIY SE kit to use with Klipsch Speakers. Will enjoy browsing what others are doing with their projects, Speakers SS amps ect.

I want to convert my wired DIY subwoofer to wireless subwoofer. Now it is connected to my main system with 8 feet long audio cable, I want to convert it to wireless. Can any one suggest a bluetooth transmitter and receiver pair for this? Transmitter board will be placed inside my main system and the receiver inside the sub. They must be per-paired or something like that so every time I power them on, they will get connected automatically.

SUB: Pyle PLPW8D, IRS2092S 500W Mono amp.


Thanks in advance.

I have a pair of 10” MB Quart subwoofers. I’m looking for an amplifier to driving them. There are two amplifiers available here; Nakamichi PA-302 and Soundstream Reference 405.

The Nak is a 2-ch amplifier and the SS is a 5-ch amplifier. If the Nak is used, the satellite speakers will be driven by the Pioneer head unit. But if the SS is used, all speakers will be driven by it.

IMO, the Nak should perform better for subwoofer driving because it has no duty to drive the satellite speakers, while the SS would have to share power or input current to 4 satellite speaker channels at the same time.

However, according to the specifications, the Nak is only listed 80Wrms x 2ch @ 4 Ohms, whereas the SS is listed 100Wrms x 1ch @ 4 Ohms (mono subwoofer channel).

Consequently, which amplifier is more suitable (or better) for powering a pair of subwoofers?

Hi there, thanks for letting me in ;-)

I am from Denmark, but live in sunny south of Spain. This is the 10th country where I live, which meant a lot of international moving, which again meant getting rid of my Large and Heavy B&W speakers and ditto amps etc. But now, settling down, I want to experience that high end sound again. So cue Troels Gravesen Illuminator 5 project. But.... the WAF (Wife Acceptance Factor) was really nil on his build, so I decided for a total overhaul.

I liked several things about this project, they are time aligned, the quality of the drivers involved, the fact that Wilson Audio Alexx is exactly the same. I will be amping the bottom with a Hypex FusionAmp with active crossover and having a passive crossover for the mids and tweeter with external amp.

And now, after 3 weeks of spending every spare hour (read: 8 hours per day ) designing and planning and drawing, the finalized design is done. The enclosures correspond to all volumes in the original project, but obviously the upper part looks (and will behave) very different from the original design, so.... crossover modifications etc. But that's more than enough for a later thread. Suffice to say that this is a complex design, and while paper is grateful, it also has to be possible to actually assemble the darn thing, so a lot of planning went into that.

For now, I'd like to ask here if anyone has experience with casting of Cabinets such as these mid-tone and tweeters. What resin or material did you use ? Molding or casting in 2 shell parts or in one go ? I am thinking along the lines of reinforced epoxy with chopped glass fiber, which should give a lot higher density and very strong modulus. But this decision is not set in concrete (wait, perhaps I should try that ?! 🧐) and I for sure would like some suggestions before beginning - which is sometime next weekend.

I enclosed a few screenshots of how the design looks like, and for sure I'd also love to hear all sorts of comments and advice on what problems you see in this, your solutions and basically whatever you can think of.

Just bumped into a news item on Audioxpress about a potential new material for loudspeakers magnets.

It's called Niron and it sounds like it's less expensive, more sustainable as well as less constrained than Neodymium.

https://audioxpress.com/article/next-generation-rare-earth-free-magnets-are-coming

Of all the news lately around loudspeaker development, I can only hope that his will actually prove what it claims and eventually will catch on as well.
Hoping that speaker prices will go down as well and this new technology isn't being used to overprice products instead.

I'm in the final phase of securing a DAC I've wanted for some time that uses 4 PCM1702 20-dip chips per channel.

So, I'm a mechanical guy, and all this digital stuff is giving me a headache.

I have found that the manufacturer used the PCM1702, whereas some companies used the PCM1702U, others the PCM1702J, and I could type for a while, but what is the difference between the chips.

I found some original prices on the net that seem to indicate that the K, U-K, and the U were more expensive - does this equate to better SNR? Is it an indicator of better QA techniques and mean that these chips are better matched?

Thanks in advance.

Hi All,
I am a stockist based in Singapore. ( in South East Asia )
Mostly obsolete and vintage Audio IC from the 70's , 80's and 90's .
Also very popular now are the Sony CX20185 , CXA1417S , Curtis CEM3372 , Technics OD503AQ , AP etc.

My listing on Ebay is getting a lot of attention at the moment on this Analog Devices item.
AD1865R - 3 pcs at USD45 , Free postage by Singapore Post with tracking # provided.
Payment : Paypal

Can offer very competitive price on Group buy based on quantity with free postage also.
Best price on the market , new and Original 100% guaranteed , just check on the feedbacks and the quantities that I have sold.
If interested just let me know , thanking all in anticipation for your time and efforts.

I knew that this topic would eventually come up:

I have inherited an Audiolab 6000CDT in as new condition. I have not yet put it in my system. If any of you have this transport, how do you like it? Does it take CD-R discs?

Inherited one of these in like new condition.
If anyone has one, what do you think about it? Usefulness as a streamer?

Fully working, top notch components with two matched J grade Japan made chips, set up for external IV stage. With PCBs for the Tube IV stage by Grunf and Michelag.

Superb DAC, very highly recommended. The Tube PCB kit comes with tube holder. High quality DAC board Enig plated.

Looking for £130 for DAC and tube IV PCBs. Plus shipping £10. Will send anywhere in the observable universe...

Hello,

Is there a reliable way of getting a trigger signal from a Mac (Mac Mini) when it enters (trigger off) or exits (trigger on) standby mode? The trigger should ultimately switch on my Poweramp (12V nominal required).

With "trigger signal" I mean any reliable signal source that can be converted into 12 V. At the moment I use a 12V line from my PC power supply, that switches on when the PC is leaving standby. This works flawlessly. My keyboard is allowed to wake the PC via USB.

The problem with Mac is apparently (depending on the source) that their USB ports remain powered when entering standby. USB ports would actuall be easiest, since I could just use a DC/DC converter to get 12V.

One solution, with drawbacks, is to set standby to hibernation mode. However, restarting will take long and must be done with the the power switch. This is also some unreliable, sind ever generation of Macs might behave differently.

Another solution I found uses a microcontroller that pings a server on the Mac. When the ping is answered, the trigger output is set on. However, his relys on WIFI and even though it is a neat solution, the complexity is large and a little bit over my head.

Are there any other ways to extract a "I am on now"-signal from a Mac without opening the internal power supply?

Thanks
Florian

Does anybody know if Narrow Band Technology transformers by Plitron are still available somewhere? Or is there any equivalent produced by Noratel or someone else? I tried contacting Noratel, but they don't seem to be interested in answering my emails.... I want to use one as an isolating transformer for my audio stack, need ~2,5kVA one, 230:230V.

Hello everyone,

I was repairing a Class D Type 4 amplifier with a simple output section failure, which had a dead FET on the low side. After removing that FET, I noticed it’s running very hot at idle with a 4-ohm load connected.

I replaced all the FETs, but the issue remains.

The gate drive is not perfect but not terrible either. However, the PWM output from the optocoupler on the low side looks pretty bad.

Additionally, the rails have some nasty noises.

Any help would be greatly appreciated. 🙏

*Sorry, I misplaced the LS and HS

My system is 2.1 channel comprising 5.25” coaxial speakers at front doors and a subwoofer in the rear trunk. I love to hear treble, so I decided to install additional tweeters on the dashboard. The additional tweeters need high-pass filters to protect them. I simply add single 3.9uF capacitors to them, yes, it’s first-order filter. I tried listening to them with and without polarity reversal. The first-order gave clearly different result, so I could make decision easily. Yet, I have 0.3mH inductors unused, I decided to add them. Now, I have converted the filters to be second-order. Nevertheless, it’s not easy to decide like the first-order anymore. I found little difference between reversing tweeter’s polarity and without. One gave boosting treble, the other gave cutting it. Since this is car audio installation, I‘m not convenient to setup RTA equipment in the car. I’d like to ask whether there’s a way to determine tweeter’s polarity “in car audio” without using measurement equipment.