Hi everyone,

I bought this locally to reuse the case. However, the insides seem to have been made with care. I am wondering if I can resurrect this.

From the previous owner, some fuses blew (there are 7). They were not the original DIYers and did not want to fix this. Left channel power rail fuses seem to have blown now. I have not turned it on yet.

The input stage has JE350, JE340, 1 pair per channel.

The output stage has 1 pair of K344, and 1 pair of J100, per channel.

My searches have not turned up anything. I attached a couple of pictures. As far as my experience, I have built a Pass DIY and a Zen Mod amplifier. I have also fixed a Klipsch promedia with a blown output stage.

Before I try tracing schematics, Does any of this sound familiar? What amp could this be? Any links or suggestions ?

Thanks in advance for any help.

Hi,

From searching here and there I found this is a common problem:

the output "protection" relay (what is the official name for this?)
of my trusty PMA-560 (bought in 1990 🙂 ) started tripping occasionally.

It switches off and back on rapidly a few times, and after fiddling
with the A+B speaker selector it seems to go away.

The speakers are on B (mainly because the plastic knobs on the A outputs got too worn).

Maybe related question: I've attached pictures of the output capacitors,
do they look ok? The light brown plastic at the bottom is just to hold them in place I'm guessing.
Is it a good idea to replace them anyway? (since they're 30+ years old 🙂)

Many thanks for any pointers.

SOLD: Toshiba 2SJ201-Y / 2SK1530-Y Matched Quads

SOLD (4) 2SJ201-Y & (4) 2SK1530-Y all (8) pieces 6.35 ID@3Vgs

SOLDQuad Set - (4) pairs - NNNN-PPPP matched $110. + ship

Thks for looking.
Charles

My dad has been trying to get me to do a DIY build for a while, and told me that jeschke's voigt pipes would be a great idea. I said I liked the thought of some striking looking speakers (of course great sound as well), and he figured they'd be great. Plus after looking at the plans I was hooked on the INSANELY simple design

After reading around the internet a bit more I found that the reviews say the speakers are kinda lacking.

Are there any other really simple designs out there that sound decent and are cheap to build? Should I just stick with my plan of building the VPs?

By the way, I'm fifteen, and this'll be my first DIY build. I listen to everything, but these speakers will be mostly focused on '60s and '70s rock. I'd probably be briefly powering them with a sony str-d365 receiver, until purchasing a new amp.

Hi Nelson. I want to build my own version of your H2 generator, since I was not at your Amp Camp😱 I have all the parts in stock, including the J112. In the documentation you said they need to be matched for a pinch voltage of 2.0 to 2.5V for an IDSS of 30-40ma. Is it an error there since the J112 IDDS is about 5ma from the specs sheet, or I'm missing something?

Thanks
SB

Selling my stock of 1G4G directly heated triode, DHT.

A very sweet sounding tube for a small signal stage. Original boxes.

£6.50 a pair plus shipping. Interested in offers for all of them.

Hi all

I'm about to undertake my first DIY tube amp project (power or integrated) and need some recommendations of schematics and designs that would be a good place to start.

I've looked at kits and whilst that is convenient it does seem a lot like lego (and I have a Elekit preamp to cover that). Ideally I'm looking for something that has a fairly well regarded schematic, parts list and if possible an explanation of why certain parts are placed where they are and what function they have. The guitar amp guys seems to have access to lots of these but I can't find much from the audio world.

Obviously I love the Kondo and Luxman stuff but every time I find a schematic I see comments posted that lead me to believe the layouts are incorrect or have errors. I'm just a beginner and yes I'm aware my DIY will sound nothing like a well established company that uses the finest parts, but that's ultimately not the point of this project. The goal is to build something and to learn whilst doing it.

Best

Only 1. New and unused. Type SC
Ship from Asia.

Asking USD400. Item from Asia. Buyer pay ship fee. Thanks

I’m a newbie, still in the initial learning process, mostly watching YouTube. I have heard contrasting recommendations (watts per speaker). Repeating some info from videos and articles, “the amp should be twice the amount of the speak wattage and match the ohms…”

I recently purchased a set of klipsch speakers. Each speaker is 8 ohms and 150 watts.

Can anyone recommend the best Dual Mono Block / pre-amp combination for a dyi project?

With my little knowledge, I was thinking:
• Purchasing the 2 “amp camp amp” as dual mono blocks.
• And a nice matching ss pre-amp.
But the amp camp amp “Power (monoblock)” states 15Watts into 8 Ohms. Would I be wasting my speakers range by purchasing such a low wattage amp.

Any recommendation on DYI kits, two mono block and pre-amp under $1000??

I am thinking to move up from my current Infinity TSS-SUB4000-it's good but I would like to sell that off with the rest of the TSS system it came with. Plus I'd like to go lower
Infinity TSS-4000 Speaker System HT Labs Measures | Sound & Vision
and have a more effortless high volume.

One way to do this is to buy another sub, but I feel more interested to build something. The sub location is in the corner of a long room behind where a French door opens, hence a trapezoidal shape:
- About 7' tall
- 22" along the front wall which has TV and three Wharfedale Diamond 11.4
- 14" along the side wall, to the hinges of the French door
- 8" at the other side of the trapezoid, so the French door can be open some
- About 23" along the "hypotenuse" or long face of the trapezoid.
Yeah, a picture is better. Let's see if this one works


With the woofers facing outwards, it's about 8 cubic feet. If I face the woofers inwards into a tapered slot, somewhat less. The advantage is both acoustic (not firing into the glass panels of the French door) and aesthetic (don't see the woofer cones, I can decorate the side or something).

(1) To port or not to port, that is the question. I'm feeling it will be tricky to calculate and physically put the ports...I guess they would have to fire out the narrow end, which is not so attractive. Or make a long slot port. It would be cool with sealed to pump in like 3 Hz or something and rattle the house ha ha-then again it's nice to have the output bump from porting. What do you think?

(2) What woofers? A wall of 10s? Twin 18s? 12s or 15s? Why? Here I'll specify the total budget for woofers as hundreds, not thousands.
(2b) And this choice affects the internal cabinet construction, since you don't want different channels driving woofers in common enclosures.

(3) Amp also hundreds not thousands. Maybe a Crown or Behringer? Fan is OK only if it does not normally come on.

(4) In the Middle Ages, I used LMS for measuring parameters and LEAP for simulation. What is used these days? (Ideally that is NOT just based on simple electrical filter calculations, i.e. which can take actual data and simulate the true rolloff shape. That's probably more important for ported). I still have a LEAP port key but nothing to run it on.

(5) If I slot load, how big should the slot opening be? I have 3.14159" in mind but that is because I'm craving pecan pie ;-D. I suppose it can be calculate as cone displacement transformed to slot area, then multiply the air velocity at the woofer by the same factor?

I have been thinking about and simulating variations of this tapped pipe for a while now. Today the stars aligned with the moon, my morning tea and my morning mood and access to the car, so I loaded Cypress Hill into my car stereo and went and bought some 22 mm particle board for a cheap and easy to build prototype based on my B&C 12PS100.

I hope this tapped pipe will have it all. Reasonably high efficiency, a great midbass punch and attack, reasonably small footprint, cheap and easy to build and a great extension downwards into the abyss. These desirable properties seldom walk hand in hand, but I hope a healthy dose of positive acoustic feedback will create a lot of synergistic effects enabling the presence of all those desirable properties at once.

The name TP12LL34 comes from "tapped pipe Long L34". "Duality" comes from the fact that the great extension downwards does not stand in opposition to a great midbass. Positive feedback is the pressure that the last long section (L34 in Hornresp lingo) exerts on the cone throughout the midbass.

I got this Class A/B Amp, with mosfets.

2 Channels are ok, 2 are disorted.
Outputs and everything else is ok.
The output on the two channels is reduced, and upper and lower sine are disorted.
I expect the mute circuit, but i dont fully understand, how it works.
Attached a diagram from a two channel amp, but its exactly the same:

Page 2: U25A , on the working channel i have a good input sine. Ground from osciloscope is at rca ground.

on the disorted channels, there is no visible input at U25A...
I expect, that the mute circuit is supressing the signal, but i cannot find, where it is suppressed.

1W or 2W magnetic or not ( US)

For sale from London, UK. Several VT-67 (30 Special) with ceramic base. Also several Type 30 DHTs.

Most are in original boxes. A couple unopened.

Very nice old directly heated tubes - perfect for preamps etc.

Pictures to follow. Prices vary from £9.50 a pair for 30 to £16.50 pair for VT-67. Open to offers for larger quantities. Can ship at cost.

Project it was gonna be used for didn’t end up happening so my loss is your gain.

$55 PayPal friends and family and it’s yours shipped or cover the PayPal fees. Save a few bucks on the cost of a new one and avoid paying shipping.

Send a PM if you’re interested, will mail right away!

Hi!
I have been through a number of threads here about motor controllers for turntables. But several of these threads are quite old and new products have come on the market. What are your`e thoughts about the following solutions:

Solution 1: Pi Pico generates a sine curve for each of the motors windings. The sine curves are amplified with an affordable class D amplifier that drives the motor. One channel for each winding. If so: Do ​​I need a transformer between amplifier and motor? And what specifications must this transformer have? The motor on td 166 is 16v ac.

Solution 2:
Use a PWM generator to synthesize a sine and switch the signal to and from the two windings? I do not quite see how to do this. This may require that the PWM signal is generated with varying duty period throughout a cycle. Is this😕 possible? In addition, should a low-pass filter be installed? If so, where in the chain should it be placed, and how is such a filter calculated?

Which of these solutions will be easiest to succeed with? Are there other similar solutions that may be possible for someone with limited experience?
As you can see, I'm totally new to this, so sorry for the stupid questions.

Cheers!

[WIKI=https://www.apogeeweb.net/electron/what-is-a-tantalum-capacitor.html
]DVP12SA211T[/WIKI]
Has 8 digital inputs including:
3 high speed inputs 100KHZ
5 10KHZ average speed inputs
Has 4 digital outputs including:
2 high speed output 100KHZ
2 10KHZ average speed output
16KSTEP programming capacity
Upgradable input / output
DVP-SA2 series has two models, both models have 8 inputs. The main difference between these two models is in the type and number of outputs, so that the DVP12SA211R model has a relay output, but the DVP12SA211T model has a transistor output.

The 2nd generation slim DVP-SA2 series PLC offers greater programming capacity, optimal execution, high-speed output of 100 kHz and counting functions. It can also be upgraded with left and right DVP-S series development modules.

The second generation of advanced slim type PLC

Advanced PLC with the ability to support left high speed modules

32 bit CPU

Programming capacity: 16K STEP

10K word data storage

Execution speed:

LD: 0.35 .s

MOV: 3.4 .s

It has one RS232 port and two RS485 ports

Supports standard MODBUS ASCII / RTU protocol and plc communication functions

No battery required, uninterrupted timer operation 15 days after power failure

Supports DVP-s series right and left modules

Hi, is there anybody out there that has any experience with the Dali 15a mk2 Loudspeaker - I think they were produced around the mid 90s and were the standmount version in Dali's top line, alongside the 18 and the flagship 40. Thanks

Ever wanted to build a CMOY using modern parts? 😀

This project uses johnc124's (TI's) new OPA1688 dual channel headphone driver IC that can source/sink up to 75mA per channel at extremely low THD+N levels, while consuming just 1.6mA idle current per side. FET input and just 0.25mV (250uV!) DC output offset. I've just received the 4th version of the 4-layer board back from fabrication. I'll list those at-cost in the vendor forum in a few days if anyone is interesting in building one up.

The project is intended to be a "mini NwAvGuy O2 headamp" as much as it is a CMOY. To that end it uses two 9V batteries with a *real* ground. No virtual grounds here, no rail splitter chips. And just as with the O2 Headamp to make that work a power management circuit has to be used to cut off the batteries if one battery "disappears" or if both batteries simply get discharged. Surprisingly easy for that to happen with broken wires, intermittant battery snaps, a shorted cell or in the case of lithium "9V" rechargeable cells the on-cell protection circuit doing a low battery cutoff. If one battery suddenly disconnects a large amount of DC would appear at the amplifier output without a protection circuit like this.

I've updated NwAvGuy's O2 Headamp's Power Management circuit with some really cool new parts, the CT128 optical-mosfet solid state relay from Coto Technology (Mouser #816-CT128). The part has an "on" resistance of just 0.05 ohms! That is AC-wired. Here they are DC-wired which further reduces the resistance to a ridiculously-low 0.0125 ohms. Not a typo, 0.025 ohms per SSR mosfet. AC-wired they are in series, but DC-wired they are in parallel. By way of comparison the mosfets used in the NwAvGuy O2 Headamp have "on" resistances of around 0.5 ohms each, 40 times higher.

Even better, since these SSR's are opto-mosfet parts the internal LEDs are strung in series with a current source and the power switch (on the pot), guaranteeing that the positive and negative power rails go up/down at exactly the same time. In the O2 the mosfets were powered separately by comparators leading to time lags between on/off of the pair, resulting in one power rail up while the other was down for an instant (= a DC output transient in the O2 Headamp causing on/off thumps). In this amplifier the SSR LED string grounds by a single comparator, which in turn cuts off if the total battery voltage drops below 14Vdc (7V per battery), exactly like the O2's PM circuit.

And speaking of comparators... this CMOY uses the brand new TI TPS3701DDCT. Look it up, it is an amazing part. Internal 400mV reference, hysteresis, and just 8uA of idle current meaning it would take 30,000 hours to run down a lithium 9V cell. That is part in the photos on the small adaptor board near the battery wires. Originally I had the part on the back of the board, but given the size I found it best to put it on a DIP6 adaptor board. I have the adaptor boards and connection pins at cost, but I'm also having a run of professionally assembled adapter boards done too. With the DIP6 adaptor every part on this CMOY is through-hole for easy assembly, except the OPA1688 chip itself which is a SOIC-8 SMD.

This Super CMOY has other good stuff from the O2 Headamp, including the input RF filter and blocking capacitors to keep any source DC out of the headphones. Preserves that nice 0.25mV or less DC output offset. With the 3.3uF film caps specified the resulting low-end corner frequency is just 0.9Hz (the low end of the frequency response), below the audible range. The amp has battery polarity protection diodes and diode reverse rail clamps.

And of course it includes a headphone relay - same circuit I've used on the O2 Booster Board - to insure zero turn-on or turn-off thumps. 🙂 Works great. In the first couple of fabricated version of this amp I didn't have the relay, just the OPA1688 output directly. The chip does produce some small on/off thumps by itself without the relay circuit. One of the revisions used two more of the opto-mos SSRs on an adaptor board for the muting relay, which worked perfectly, but in the end I didn't have the board space.

The board mounts upside-down in an Altoids tin, sitting on the 3.5mm jack cases and screwed onto the front of the mint tin with the 3.5mm jack nuts and the pot nut. Adydula here on the forum built up the previous V3.0 and did his usual fantastic job of centering and cutting the holes! I'm using a paper punch - goes right through the thin mint tin metal - then enlarging it from there.

The project materials will all be put here over the next few days:

https://drive.google.com/folderview?id=0B67cJELZW-i8b0VRa29ScjFac2M&usp=sharing

A bunch of spreadsheets that calculate the amount of voltage and current a headphone amplifier needs to supply for various headphones and Sound Pressure Levels can be found here:

https://drive.google.com/drive/folders/0B67cJELZW-i8eFVrX2taU25LUWs?usp=sharing

Read the "readme" in there for an explanation of the sheet. Worst case, with lithum batteries that are nearly dead (6.4Vdc cutoff) you will get around 4Vrms maximum swing from the Super CMOY. With throw-away or NiMH batteries add a volt to that, around 5Vrms. A single-chip Super CMOY can supply 75mA per channel, while a dual can supply twice that at 150mA per channel. Compare those numbers with the rms voltage swing and rms current your specific headphones need in the sheets to hit 90dB SPL and 110 dB SPL loudness levels.

Click on the arrows in the lower left corner to enlarge any of these photos.

Non complementary power MOSFETs generally disregarded by diyaudio community for their limited use as output devices. What if we are looking for good candidates for circlo or quasi like topologies that uses same type devices on both sides? In such cases, number of candidates becomes big but when we become more picky about specs, situation becomes different.

I think experienced members here can name some good but not well known N-MOSFETs regarding such conditions below:

Low (preferably very low) Input Capacitance
High Transconductance
Pd (Watts, higher is better for same or lower capacitance)
Specified Pd can be verified as DC SOA in datasheet
Low Thermal Resistance (especially for TO220 ones)

I came across some devices produced by Alpha & Omega Semiconductor which seems spot on most specs.

eg: AOT7N60, AOT8N50, AOT9N50, AOT9N40

Can you name some other ones?

Hello,

I had a few questions that I was hoping to get answered.

Some are about DRC in general and some are more about DRC Designer

1) I am more trying to get people opinion on this one. But is DRC Designer a good way to get ones feet wet and get 95% of the way there? Or should I really just cut to the chase and learn it through the way outlined by gmad?

I have a 7.2 setup but am mainly looking to improve my stereo music listening. So I run my PC --> AVR (which looks after bass management and crossover at 100hz) --> L&R HTM-12s (they have a woofer and a compression driver) and to a behringer amp that has its own PEQ of dual 18” subs. So would that be simple enough for DRC Designer? Or should I deep dive into the method outlined by gmad? More so looking for opinions on this one.

2) My understanding is that even if the initial recording is done in REW with the mic calibration in REW that it also needs to be loaded into DRC correct?
If that is the case, I tried to edit my mic calibration text file to follow the required formatting but this is the error that I receive through DRC Designer:

“Reading mic compensation definition file: E:\John\Audio - HiFi\REW\UMIK-1\7060719_90deg - DRC.txt
Mic compensation direct inversion.
Allocating mic compensation filter arrays.
Mic compensation FIR Filter computation...
FIR Filter computation failed.”

Any ideas what I am doing wrong? Or do I need to even worry about this if I do my initial recording in REW with its mic calibration activated?

3) Should I do my baseline / raw measurements with or without my pre-existing parametric EQ in place? I spent a bit of time making a combination of nearfield gated measurements and then MMM in room measurements to create a quite good sounding PEQ file that I have been using and really enjoying. I have since become interested in DRC and thus I am wondering if it would work best if I ‘layered’ DRC ontop of the existing EQ or if I should start from the ground up and just measure the raw speaker/sub response with no EQ. Really the question is, if one wants to add PEQ should they do it pre or post DRC?

4) Is there any easy way to make correction files for surround channels and also for the LFE/Sub channel? I am mostly going to use if for stereo listening but I know the upgrade bug will bite at some point. So could I just record them in REW and then export and label them as if they are left or right channels and then when I get the correction files then rename / assign them back to their actual channel? Do people do multichannel EQ with DRC and what’s the best approach for that?

5) Can I use Equalizer APO as a convolver? As it looks like DRC Designer puts output convolver WAV files in “ConvolverFilters” file, so can I use these and import them into into EAPO? Is that adequate? Sorry it is just unclear how Convolver VST fits into the system flow and if it is needed if one uses EAPO?

Thanks for any input and replies! Looking forward to learning more about DRC and trying it out!

Hi fellow fullrangers.

This is a small story of my experiments with a couple pairs of nice drivers from Mark Audio, namely their high-end Alpair12P and bottom-end CHN50, both of which I bought last year. I built a pair of small rear ported nearfield PC bookshelves with the CHN50s, and made a pair of onken-type speakers with the A12P, having 6 vertical rectangular vents in front of each box.

The CHN50's sound obviously doesn't compete with the paper-coned A12P's when it comes to overall quality and SPL, but these smaller alloy-coned drivers have something special going on in their frequency response. Simply put, I can play Any type of music and can sit back, relax and enjoy the music without ever getting the feeling of turning it down due to any unwelcome jump in SPL, starting from about 55Hz upto as high as I can hear. In other words, I absolutely adore these tiny drivers now for more than how they look! The frequency response is very near perfect for my ears. Here they are: (currently powered from a barebones 15Watt TPA3110D2 module)



Enter the A12P! The drivers sound truly beautiful hands down... but... (like that classy high-end girlfriend you can't get your eyes off because she's so this here and she's so that there but then she doesn't know when she switches from speaking to screaming when talking casually to someone standing one feet in front of her) the drivers have occasionally been a pain in my ears but no way I'm gonna let her go! It's complicated, and gets even more complicated when you realize that the published response graph in the datasheet has 20dB increments in the Y axis! Yes I missed that at first glance, second glance and third glance as well until someone told me to look close. You can see a couple pictures of the speakers in the fullrange gallery thread >Here<.

So, for the last two months I have been playing with line level passive filters to control that dreadful midrange shout, and have finally reached a design that is simple, sounds good and sits comfortably between the preamp and the power amp. I may sound a bit hypocritical when I say that I don't want to spend on large inductors and capacitors after spending so much time and money into building these speakers, but it's just an honest bias against putting stuff between the amp and speakers, nothing more.

Now, Going over many different variations of the same basic design and listening to the results I noticed that while many of the iterations succeeded in taming the shout, sometimes they sounded really bad and a few times sounded really good. It's not about parts quality; on inspection, it became clear that for implementing the new response curve without sacrificing the music's soul, maintaining a relatively flat midrange phase is essential! It's not at zero degrees in the filter, but is more or less flat over the range of 100Hz to 1.5KHz.

Two things I should mention. Firstly it is my guess that the A12P's peak/dip response may be affected by the amp that is driving them, in that the shout might not be too much of an issue with amps presenting a low damping factor e.g. tube amps, but all my amps are solid state and of high damping types. The drivers seemed to scream slightly lower with a resistance of 5ohms in series between them and the amp. Secondly, overall response with the filter added goes down by about 3dB. So for the same midrange SPL the volume knob may need to be cranked up more.

So here is the filter (texts explained below):

=

The contraption:



A bit of explanation:

As the filter is supposed to sit at the preamp's output, it is driven from a low impedance source. For different power amp input impedance the same filter response can be acheived by following the formula mentioned in the above picture.

"R AMP" is the power amp's input impedance in Ohms. Say it is 50Kohm (50000 ohms). Following the formulas in the picture of the schematic the values of the components can be calculated.

R1 = 50000 / 10 = 10000 = 10kohm
R2,R7,R8 = 50000 = 50Kohm
R3 = 50000 / 1.47 = 34013 = ~33000 = 33kohm
R4, R5, R6 = 50000 / 4.6 = 10869 = ~11000 = 11Kohm
C1 = 47000 / 50000 = 0.94 = ~1nF
C2 = 220000/50000 = 4.4 = ~4.7nF
C3,C4 = 1000000/50000 = 20nF

Using these values it will give a response very close to the green curve shown in the picture below.



Notice that the red curve is the result of the combination of component values calculated for 10Kohm impedance and it is very close to the green one. My application is the same. For driving the filter into 10Kohm input impedance of the amplifier I chose the following values.

R1 = 1Kohm
R2,R7,R8 = 10Kohm
R3 = 6.8Kohm
R4,R5,R6 = 2.2kohm
C1 = 4.7nF
C2 = 22nF
C3,C4 = 100nF

Here is phase response:



The design was not derived from calculations, rather the calculations have been derived from noticing changes in frequency response behavior with differing loads to the filter i.e. different input impedance of the power amplifier. The filter is sensitive to the load that is presented to its output. My main power amp has an input impedance of 10Kohms. So while experimenting I have been choosing values that give a preferred/equalised response only for that particular amplifier. If the design is built for an amp with a particular input impedance, and then it is driven into another power amp with a different input impedance, the response will change. But if your combination of preamp+amp is expected to be constant then that problem goes away. Besides, it is so simple and cheap that you can make one filter for each amp, with all the combos giving the same end-result.

Now, the A12P drivers have some fire in the trebles. So in case the mids sound a bit muted, the level can easily be equalised without too much affecting the overall response dips that act against the drivers' peaks. It can be done by changing the value of R6. Increasing its value will decrease the mid response and decreasing its value will increase mid response. Closing J1 will short the resistor, resulting into the green response curve. The red curve is the default response and the blue curve is with double the calculated R6 value. Changing R6 value will also change the overall response curve shape slightly and the main dip near 3KHz shifts a bit towards 4KHz. Shown in this picture.



If lows are to be kept same but you only want to change the mid+high level then R8's value can be played with. In this case the dip's level varies a bit, but doesn't shift much from 3KHz. Here is the response with changing R8 value.



Notice that all the above modifications also change the dip depth along with overall levels slightly, i.e. precision isn't great. But at the end it has brought the A12Ps to where I can sit with them all day every day without a hint of the shout that plagued the sessions before, for a cost of about 100 rupees (1.5dollars). They been playing with the new filter for more than a week now. The resultant frequency response the filter is giving in my room comes pretty close to that of the CHN50 drivers - in that the shout is gone and overall SPL balance from about 38Hz right up to the highs has been established. The most prominent of the peaks in the driver's response sit around 3KHz and the filter adds a -6dB dip there. There is another noticeable peak at around 800Hz but it's not as loud, so the attenuation is about -3dB. It's just a simple RC filter but as the effect has been well worth the effort for me, I thought it's worth writing a few words as well.

___________

EDIT:

As pointed out by XRK in the next post, the line level filter's response may have significant dependence on the preceding and following stages i.e. the preamp and the power amp's output and input impedance respectively. I admit I could have been more thorough in writing the original texts and sharing the graphs; so following the formulas I prepared a couple charts.

This chart is a bit tall. It shows the response of the filter with different power amplifier input impedance values starting from 5Kohms to 100Kohms. As can be seen, with appropriate values chosen, the frequency response is virtually unchanged from the lowest amp input impedance to highest.



The chart below shows effect of preamplifier output impedance. Obviously the response of the filter will change as the source impedance changes. But it's also very easy to solve this problem. As preamp's output impedance is changed from a low value to a high value, all we need to do is to decrease R1's resistance accordingly. In this picture one graph set is take with preamp's output impedance of 100ohm, and the other with 1Kohm. With R1 shorted in case of 1Kohm preamp output impedance the only thing that changes is total gain by about a dB. but the curve shape and positions of the dips/notches stays more or less constant.


So it can be seen that the filter is quite flexible when it comes to suiting different scenarios consisting of different sets of preamp+amplifier impedance values. The good thing about all this is that, the preamp's output impedance and the amplifier's input impedance are not reactive components unlike loudspeaker impedance, and are fairly constant values over their specified frequency response bandwidth. So when modding the filter for a set of pre+amp combo, we can be assured that the resultant response will be close to that with driving the filter from and to purely resistive source and load.

___________

Thanks for reading and I hope this info helps anyone who wants to experiment with the 12Ps the same way. Feel free to comment if you have suggestions and/or questions. I'm no expert in designing filters and would love to learn if the filter can be modified further in ways that makes it more effective without getting too complex.

p.s. you can see all the pictures in this google album - Alpair Filter

shaan

SOLD!

I am selling my RAAL 70-20XR-AM Pair Brand New in Box. $850 shipped in the USA.

What are a few good brands of car audio speakers, not subwoofer. door and rear for a chevy c1500 truck. I prefer quality sound over high power.

I have a variety of amps and a couple of active crossovers. I want to do some amp rolling.



One amp will be doing tweeter and squawker, another doing the woofer.

The crossover between the tweeter and squawker will be passive.

There will be a sub getting signal from the woofer amp's outputs.


Should I pony up for Forte threes or fours or would I be just as well served with OG Fortes?

Odd behaviour...I make a post and then, within moments, wish to edit it.... Normally all good...but

Then your site goes down for 'Maintenance" and when it comes up again the time limit on editing has passed.... no fault here, simply that your "maintenance" does not queue edits...???

Rather counter-intuitive and unable to be allowed for when posting... Just a big guessing game or 'When will the Earthquake strike ?" scenario 🙁

For the problem of the inductor has a "hiss, hiss, hiss" or "ji ji ji" sound in the normal operation of the class D circuit. this is has big effect of the sound quality . here is my sharing of analyse and solution .

Cause Analysis:
1. The winding of the inductor enameled wire is loose. (The coil is impregnated need dip in varnish to avoid the coil loose )
2. PWM dimmable output power supply, inductor noise. (PWM produces audio resonance and EMI interference signals during dimming. The ripple current increases with power changes. (Use magnetic powder core inductors/one-piece inductors instead, varnish strengthens the inductor coil and increases the output filter capacitor capacity. Can be reduced noise.)
3. The inductor exceeds the maximum current and the margin is insufficient, resulting in the decrease of the inductance electrical characteristics and the deterioration of the magnetostriction. (we need choose a large-size inductor to ensure that the peak current of the inductor in the circuit is 1.3 times greater than the inductor's rated current.)
4. The circuit ripple current is too large, the electrical characteristics of the inductor are reduced, and the irregular vibration of the inductor coil is aggravated. (Varnish is impregnated to strengthen the inductance coil. Increase the inductance in the BUCK circuit, and increase the filter capacitor in the boost circuit. Reduce the electrolytic ESR or use ceramic capacitors.)
5. The switching frequency is too low, within the audible range of human ears (20-20K). (Where Varnish reinforces the coil, or use a magnetic powder core with low hysteresis expansion, or apply an integrated inductor. Adjust the power switching frequency on the circuit. Avoid the audible range of the human ear.)

There is an axample of the sound of the power amplifier channel is poor. customer's project tested the inductance of 3 defective products, and the inductance of the 3 channel inductance was 22uh, 2.8uh, and 4.7uh. customer usesCODACA digital power inductor CSD1013B-100M, which is applied to the output of the digital power amplifier.

Cause Analysis:
Because the customer's inductance is tested on the PCB board, it is possible that the chip, other components and connections will affect the accuracy of the test. Therefore, the two inductors are removed separately and tested separately to be accurate.
Defective product 1: 4.7uh measured value 9.7uh (10uh is normal)
Defective product 2: 2.8uh measured value 0.84uh, (10uh becomes smaller)

Open the measured 0.84uh inductor and observe the inner coil of the inductor through a microscope. It is found that the varnish water between the coils is black. It is suspected that the flat enameled wire is damaged during the IQC or the production process. In the case of the customer's high current test, the layer is short-circuited. , Resulting in smaller inductance.

Solution :using varnish to strengthen the coil and test frequency is increased to 100K/1V to prevent the defective .

For learn CSD1013 design click https://codaca.com/en/product/230.html

I have been using a pair of RCA 6L6GB's with the screen tied directly to the plate. I have recently seen some schematics where there is a 100 ohm resistor in between. One of these models I saw was using a switch to change between UL mode and Triode mode.

So my question is who has used a low value resistor between the plate and screen for triode mode and why?

I do not see anything about this resistor in the datasheets, RCA states G2 connected to plate for triode mode which is what I have done.

-bird

hello, it's been a long time i have post here...

during my covid force vacation i started soldering again and bought a phono pre.

was wandering if there is any substitute for the three 12AY7/6072 tube in this design?

6N4P

6J32P (6Ж32П 6ZH32P EF86 6267)

thanks

Martin



h t t p s : / / Phono stage RIAA PCB stereo ANK LP3 track version one piece ! | eBay

Dear all,

I'm moving into a new house in the countryside with more available space and I would build and install two fullrange speakers in the livingroom for music and tv as well.

I would dedicate to them a PP with EL84 that gives around 20 Wrms per channel, or a KT88 SE with around 30 Wrms per channel.

The kitchen&livingroom is an L (imagine a square 7x8 meters with a missing quarter).
I'd need to install the speakers close to the wall and it is ok to have an height of 1.8-2 meters if performances can benefit (footprint would be more critical due to WAF).

Cabinets will be built together with a friend, and I would prefer not exceed 4-500 euros for speakers.

Can you suggest me something and guide me to some readings to better understand how to set/arrange the speakers in the room and fine tune the system?

Thank you in advance,

Roberto

Hi there. My Ninka's have blown up but I can't bear to throw them out so I want to replace the drivers. I have sourced some Vifa P17WJ-00-08 and will get some Hiquphon OW-1 with a rebuild of the crossover. (This is my first foray into DIY Audio). My first challenge is tracing the FRD and ZMA. I have tried tracing programs but I can't seem to get the impedance curve to behave. It is wildly off. Does anyone have FRD and ZMA files for the speakers above? I am getting quite frustrated with my tracing efforts.

Thanks in Advance.
Mike

Note: 15-july-2018.
With Chris’ retirement we no longer have access to the shop, so barring some new, different arrangements we are suspending flat-pk manufacture.

I do have some stock of small flat-paks.

dave

07-august-16 we are more or less in continuous production, Price is still $290/pr USD for FH3 flat-paks.
The larger Frugel-Horn XL flat-paks are also slowly starting to flow, Price is $390/pr USD. 1st shipments of Frugel-Horn Lite are underway. There is still often a queue.

We have reached 100 flat-paks with very few issues. None have been damaged in shipping or lost.

All have shipped to North America (with a couple pair off to the other side of the Pacific compliments of the US Military). I will ship overseas but the pain of cost of shipping will be high.

Flat-paks include precut cabinet panels of 15mm maple veneered quality plywood, with an 18mm baffle (rebated and dadoed, all angles cut for easy assembly). drivers (optional, 10% discount if bought with flat-pak), terminals (5-way posts, 2" hole, good for sonics, tight for fat fingers, damping, wire (24g copper (from CAT5 cable)) for a pair of speakers. You will need to provide glue, screwdriver, soldering iron, drill & need to finish them. Clamps are useful & recommended, but a slow methodical assembly can usually be done with tape (masking usually -- anything that leaves a residue should be avoided)

Standard rebates for Fostex FE126 (FE127 & FF125wk) Mark Audio CHR/CHP/EL70, Alpair 7, or blank. One went out for FE108eS so that can be ordered. Others. We'll try (really helps to have a driver on hand to do it).

Here is Chris doing a mock assembly: Frugel-Horn Mk3 Flat-Pak Mock Assembly Instructions





I've borrowed Colin's (aka Toppsy) pictures of his lovely build as examples. From the flat-pak you can build something broadly similar, we have round terminal cups that we like bettr than Colin's and the optional outriggers we have are a little different.



Maximum shipping in Canada or USA for the flat-pak alone (no drivers) will be ~$130 to Florida. Package is 98x52x18 cm and 25.5 kg.

Postage quotes

We are encouraging local manufacturers to start providing these so that shipping does not skew the value. Speakers & flat-paks have/are made in Europe. Unfortunately the Australian flat-pak maker is no longer in production.

eMail me if interested.

Here are relevant threads on the Frugel-Horn Mk3:
FH3 Picture Gallery
Frugel-Horn Mk3 discussion thread
Frugel-Horn Mk3 beta Builds thread
Frugal Horn Mk3 build for CHP70 and FE126En

dave

Okay so I want to cheat on enclosure design!

I wanted to know how to calculate the new parameters for my driver the Eminence Delta 12LFA and then how to calculate the components to be used in the amplifier design. I would like to use an LM3886 amp if that's possible. I have a 42l enclosure kicking around if I could use that it would be great.

Thanks
Boscoe

While looking for completely different things, I found this old magazine by chance from November 1992
Radio Electronics (November 1992) : Free Download, Borrow, and Streaming : Internet Archive
and an excellent diy power amp project at hard to beat price with LM1875 and AD711 - go to the attached PDF file.

Are there more advanced versions in this topology with operational amplifier IC's from last generation like TI's OPA1611 or OPA1656 - go to
Burr-Brown OPA1611/1612 are here
together with traditional power amp ICs like LM3886 or TDA7293/TDA7294 ?

I am mainly interested in finished amplifier devices (both vintage for easy upgrade and newer versions) but also in diy projects.

Best thanks for an advice and threads here on diyaudio.

Check in this case also this URLs:
ESP SIM (Sound Impairment Monitor)
https://www.nanovolt.ch/resources/ic_opamps/pdf/opamp_distortion.pdf
Choosing of best sounding OP AMPs for the lowest possible THD+N -really the best Way?
Low-distortion Audio-range Oscillator
(page 960, post #9596)

McIntosh MX112 Front Glass , New old stock ,never been used. excellent condition , includes plastic insert's for panelock holes.
$165 , email me arg.allan@shaw.ca Txs Rich

Since some of you are talking about 74HC74s and their replacements with 74VHC74s I thought I'd try this question. The reference to halves in the following referes to the fact that there are 2 seperately usable logic gates on each 74HC74.

The DAC board on my Arcam 6 uses three of these chips, two halves per channel as part of an "integrator" circuit (taking the output from an SM5864AP) and another half to cut the clock frequency from 22MHz to 11MHz for the PCF2705 laser controller chip.

I've just bought a Superclock 1.2 which I've fitted to my Arcam 6. The clock frequency I got was 45MHz (the original was a 22MHz) which I fed through a 74HC74 (there was a spare half unused on the board) and then to the the PCF2705.

The unit still works after this mod but on a scope I can see that the output wave from the Superclock deteriorates the second time through the 74HC74 i.e. it enters at 45MHz happily goes to a buffer amp 74HCU04 and returns to the other half of the 74HC74 happily at 22MHz and then when it comes out again its almost a sine wave at 11MHz.

Is this because the two sides of the chip are interfering with each other because of the speed mismatch? Or is the 50MHz rating a sum of the speeds on each half.
Can anyone think of another reason for this? I'm a bit suprised the chip follows the Superclock's wave shape (a steep rise, small dip and then up again and then negative in a clock cycle) anyway as the spec sheet claims the logic voltages are generated internally rather than just tracking the input. i.e. once I'm over the 1 state the chips should stay at 1 until it drops to 0.

Any insights welcomed.
Bill

Hello.
Integrated amplifier NAD 3140.
Power amplifier works fine.
Pre amp fuse F1 blown.
With 60w bulb on fuse F1 pre Amplifier works but the sound if very low.

SM
NAD 3140 SERVICE MANUAL Pdf Download | ManualsLib

I'm considering buying a new integrated amp whose input impedance is only 1.2kOhms. My DAC's output impedance is 2.3kOhms-5.8kOhms. Is using this DAC with this amp going to result in weak bass?

Hi guys,

I'm building NAC-152 clone which I bought from aliexpress, same same this kit Assembly 2N5087 / 2N5089 Mono Preamplifier Board Based on NAC-152 Preamp | eBay

There are many 2n5087 and 2n5089 on time align filter area and gain area. Should I match some of these transistors to improve the sound quality? If I should, please show me the locations on circuit, many thanks.

Fiter circuit


Gain circuit


Excuse me for my bad English, thank you guys for helping 🙂

I've been looking at build a decent DAC, but I am still a newbie to DIY electronics. I found this on eBay (based on an AD1955):

Finished DIR9001 AD1955 (MK3A) DAC &Transformer in Case | eBay

This is in my price range, and given my capability, a decent start to DIY. Anyone have any opinions on the setup?

I am designing a good DIY preamplifier project to go with the class AB amplifier I made ( GitHub - profdc9/PowerAmpAudio: Power Amplifier based on Michael Chua's C300 amplifier ). It is at


GitHub - profdc9/PrettyGoodPreamplifier: A Pretty Good Preamplifier for Stereo and Subwoofer Use


I would like it to be constructed with relatively simple tools and readily available parts, with substitutes available. My github has many educational projects on it and I am working on audio projects that I hope will be both educational, simple to build, and useful as real equipment.



It is a stereo preamp with a subwoofer output. It has four unbalanced inputs selected by relays. It supports being remotely triggered to be turned on and remotely triggering other devices (the amplifier for example).


It has volume and a simple Baxandall tone control (treble/bass) and a relay to bypass the tone control. It has an adjustable low pass filter and gain for a subwoofer output. It also has balance control.


There are both unbalanced and balanced outputs for the stereo and subwoofer channels.


I was debating whether to use the THAT1646, but since there is a similar chip the DRV134, I think it is safe to include. Otherwise I might use a op-amp based balanced driver since substitution is easier for op-amps than balanced driver ICs.



If there's any interest or comments for such a project, let me know. The current PCB design and schematics are below.

Years ago I put a lot of stuff away (projects to be built) for retirement. The bad thing about being older and retired is I can't remember that "Safe Place" that I put the manual to a long forgotten Aikido project. Its not really that I need another Aikido since I have several its just that well I bought this board probably 15 or so years ago and I have probably given it time to age like a fine wine. Its probably time to put it together now. Anyone have a manual to share on it?

I'll post a picture.

One of the first boards available from Glassware if memory is still functional. Octal board with no power supply.

As the Title says, I'm looking for the Hornresp Parameters for the Altec 288 compression driver.

Specifically, I have a 288-8G & 288-16H, but any 288 Alnico parameters will do for now.

I'm looking to simulate a LEcleche, Tractrix, radial (like a Fostex wood horn), wood multicell, like a 805b, and a paper horn like the inlowsound.

My listening distance is only 10 feet, and I'm not too concerned about directivity, but more concerned about low extension under 500hz, despite only having a horn depth of around 20" including driver. I have a single Altec 805b multicell horn, that fits nicely on top my Onkens. I currently have 802-and 511b sitting on top. I would love to source another multicell, but haven't had any luck.

I'm asking if there's any issues operating two of Maida-LT3080 power supplies like this:



So I have both referencing to ground but the current flow circulates through both isolated secondaries (my concern here is getting similar secondaries).

The ~350V secondary side output seems to give the desired ~320V output at about 250mA. I suspect that only a single choke on each is required instead of the two I have setup in LTspice. LTSpice doesn't actually register any Vripple just the normal initial stabilisation and smooth operating.

The chokes were modelled after Torroidy's 9H 29ohm toroidal choke. The transformer is simply a ~1 ohm resistance and made to fit with the secondary voltage output. It seems a dual secondary (0-350/360V each) that cantebury windings do may suit this.

The questions are two fold - any issues? Plus a sanity check would be good.

The first is an old project.
Simulation is on improved designs with reduced noises.

Hi.I can not make it with proteus. Which pin programs can do it ? Thanks

I picked up this sub really cheap to have a go at fixing it but I have no experience with digital amps and am not sure where to start. The former owner said it was intermittent, i.e., it would work one day and not the next.

I've been unable to get it to fail completely, but it cuts out at very low volume. I've eliminated the speaker driver itself and it happens on either line level input, so it's not likely the typical loose solder joint at the mechanical interface of RCA to board.

Would a cold solder joint in the early amp stages be a likely culprit?

Hello everyone,

Picked up a Harmon Kardon F-3 / FA-3000X / XAM Mark IV (EJ Korvette's) yesterday. According to the previous owner it worked - but needed a lot of work. It is certainly filthy.

A visual inspection revealed a blown 32uf electrolytic and a poorly soldered pair of 120 ohm 5W resistors in series that had broken free from the bias potentiometer. Wired to ground.

Questions to all the more knowledgeable experts here:

1. The schematic does not have either the 240ohm 5W resistor or the 32uf electrolytic. In the schematic it is wired straight to the pre-amp heaters and then to ground via a capacitor. Any ideas why resistor and capacitor would have been added?

2. The receiver would have originally had 7408 power amplifier tubes. They now have 6V6GT tubes. I understand they are direct replacements. Is it possible that the resistor/capacitor modifaction was done in order to accept the 6v6 tubes?







Thank you,

ikonw8

I don't have your home address, and no reply from above e-mail addy

I also don't have any info about your nick here ....... so ................

either PM me here , oe reply to >zenmodiyaudio@gmail.com<

Waiting a few years for a project that never comes, I put for sale a pair of Fostex ft96h in perfect condition. They were tested long time ago maybe a couple of hours and never installed.

One of them has a small damage in the face plate that I can’t remember how it happened but it’s only a cosmetic issue that can’t be seen at listening distance and hardly to be appreciated in pictures.

Overall very very good condition.

I’m asking 400€ + PP fees + Shipping.

I'm currently in the process of creating a few PCBs using Sprint Layout and a few questions have come to mind. I'd like to get a few responses and points of view for PCB layout specifically for solid state design (which is why I posted here).

There are a few good nuggets of information scattered in various projects, particularly with respect to grounding and bypassing. I have been finding it difficult to get a high level overview of the subject with good generalizations.

Some of my most basic questions for those who are good at producing quality layouts:

1) What components to place first, where and why?

2) General workflow direction, from input to output, or from output to input?

3) What traces are laid out first, power and grounds or signal?

4) Choosing PCB size and shape, what are the primary factors in selecting PCB dimensions.

5) What components to place on the amplifier module and what should be on the PSU, such as fusing. If one places fuses in the PSU outputs is it still considered 'good form' to put a set on the amplifier PCB?

I welcome any advice and opinions pertaining to the above or otherwise that will help me down the right path when doing a design. Thanks.

I spent my quarantine the past year designing a particular preamp and its now my obsession to tweak it the best quality possible.

My concern now is the input grounding. Ive been using a star ground method to summate the input ground, output ground and the feedback resistor ground to a single point as close to the regulator output as possible and thats been hum free so far and sounding good.

But thinking about it wouldnt it be a better practice to summate the input ground and the output ground to the feedback resistor node (of the amplifying circuit) then trace that with one wire to the psu ground? Because after all the feedback ground is where the correction occurs.

Please enlighten me

Thanks

There are two grounds: Analog GND and digital GND (D-GND)
The kind of creation I don't understand and I have never seen by others.

At first look it shows like a symmetrical supply voltage about +/-5VDC.
But there is only one voltage: +5Vdig.

Because no additional secundary winding for independent creating of the +5Vdig is in use, this way was choiced from Anders Thule.
The neg regulator drives a bidirectional diode (normally anti paralleling 1N4148) and creates 0V7DC refer to GND (A-GND). But the actually GND wire for the neg regulator goes here to +5Vdig !!
What about this idea from Anders Thule?

Who can give me a commonly circuit describtion ?

Im looking at using either of these 2 tweeters. I'd like to crossover at 2khz to a 7 or 8 inch woofer. One is the Peerless DX25BG60 and the other is the SS D2608. Are there any major differences between them? I have a high end system that I want to get the very best sound out of them. I like a relaxed sound so I dont need every last drop of detail if it means it will sound etched.

Hello,

As you probably know, around 2015, ASC started selling IsoThermal Tube Traps. They are reported to perform a lot better, especially with the low end.

Here's a section of their site where they explain how it works in quite a bit of detail.

Quote:

"How do we achieve converting our capacitors into isothermal vessels? We evenly distribute fibers, a single micron in width, through the air cavity of a TubeTrap, spaced a few dozen microns apart.




Such an exposure of surface area provides very rapid, very aggressive thermal action upon the gas molecules, which return to room temperature extremely quickly. This added speed allows the operating frequency of any diameter IsoThermal TubeTrap to be lower than a non-isothermal variety."

My question is - what do they mean by 'we evenly distribute fibres... throughout the air cavity'?

My assumption is they use 1nm fabric, the kind that is used in various filters, and lined the inside of the TubeTrap with it.

Or is there something more complicated going on?

Hi. I'm looking at building a matching MC/MM phono stage for the RedRoo amplifier. I keep on coming back to using an input transformer and a D3A input tube with ccs anode load, per Stuart Yaniger's design in the articles section on this forum.

Trouble is the D3A is getting expensive and goodness knows how long they will be available for.

Has anyone tried paralleling 6DJ8/6922 in this position to reduce noise, followed by a cathode follower to drive the RIAA network, or any other solution? I know John Broskie uses the 6DJ8 CCDA circuit in his RIAA pre-amp, and I like his designs for their simplicity, but not sure about the tube noise.

Any ideas or comments greatly appreciated!

Hi to all 🙂
Since there are extremely experienced people on this forum and I know nothing about Mosfet matching, I'd really appreciate a hint 🙂
I'm building a Crescendo Millennium Edition (dual mono version) from the Elektor Magazine article and I'm stuck on the choice of the two pairs of 2SK1530/2SJ201 Mosfets. Apart from the fact that they are not easy to find any more (and I live in Europe) I don't understand what is mosfet-matching about. There's no reference about matching on the Elektor article 🙁
So far I have found the following sources to buy them:

1. This Forum
User "NicMac" is selling these Mosfets on the forum from this topic:

http://www.diyaudio.com/forums/swap...s-matched-2sk1530-2sj201-f5-f5-turbo-f5x.html

And this is the list of his available Mosfets:

https://docs.google.com/spreadsheet...BBqLqnYq03e5_z798RoW0/edit?pref=2&pli=1#gid=0

Looking at the list, I really don't know which one I should choose... 🙁


2. Tech DIY Company Store
http://www.tech-diy.com/Store/LatFets.htm

They sell a pair of these mosfets for 12$ and a matched pair for 40$, a significant difference in the price!!

3. AmpsLab
2SK1530 2SJ201 Toshiba Mosfets - Buy Online @ AmpsLab

40$ for two pairs. No specification about matching.



Many thanks in advance 🙂

I got tired of going round the back my 8600S and having to tread carefully because of the wires at the back so decided to use some old parts lying around.
I dont hear any audio degradation but was surprised that I could hear the 1st input's audio when I switched to input 2 but I also had the volume cranked up on my Zu speakers, but in any event I dont play more than 1 source at a time and happy with the "el cheapo" outcome 🙂

Hello!

I plan to build a high end three way speake. I plan using the following drivers:

- Looking for a Tang Band W8-1808 woofer? - SoundImports
- Looking for a SB Acoustics TW29RN-B-8 tweeter? - SoundImports
- Dayton Audio - DS315-8 12" Designer Series Woofer Speaker 8 Ohm

And I plan using the following amplifier

- Looking for a Hypex FA123 plate amplifier? - SoundImports

Do you have any feedback or comments to my choices?

Preample
I was looking for a speaker build with point source mids and highs, so I looked for 12" coax with 12" woofer untill I learned that coaxial compression drivers exists. Having very positive experiences with horns and compression drivers, I'm now fixated on a BMS 4594HE, I just need a suitable horn and a 10-12" woofer.

The Beef
I've been scrolling through dozens and dozens of specsheets and learned that there are masses of good twelve inchers out there, so I haven't really looked at 10" drivers yet. What I don't know which ones would fit the sound quality of this CD and my existing sealed 14" subs (which is a secondary concern). For horn I was looking at RCF HF94, because of wide dispersion and loading down to 500 Hz and it's quite pretty. Still open to suggestions though. Needs to be pretty enough for other than blind tests ;-)

So technically I'm going for

• coax 1.4" cd, xo @ 6.3 kHz, this is rated down to 300 Hz
• horn with wide dispersion, because I'm using them at couch and at a desk between the speakers, and good loading down to woofer range
• 10-12" woofer, xo from horn to woofer @ 500-900 Hz, depending on selected horn and woofer. With HF94 I'm looking at around 650 Hz.
• xo to sealed subs @ 60-100 Hz, depending on selected woofer and fine-tuning
• fully active with dsp-plates.

For the woofer, I really have no idea which to choose. Some requirements:

• sealed, since subs are also sealed
• doesn't need to go below 80 Hz, since will be crossed to subs!
• preferably (much) less than 30 litre cabinet
• I've been looking at >=8 Ohm speakers, because fully active.
• Sound quality is what I'm looking for here, not just "best value". If there is a reasonably priced driver with performance where any improvement wouldn't really matter anymore, that's my choice.

Being relatively new to DIY I really don't know which parameters to look for in a driver for this application. All forums and articles are full of tips about how to extend bass response, which does not apply here in the same sense. Dedicated mid range drivers seem to have completely different EBP than sealed subwoofer drivers, but this is something in between. I'm just looking for top quality bass-to-upperbass woofer. Also the proper enclosure size is unknown to me for this frequency range, where I'm not looking for maximal extension. Simulator suggests that almost any box size that can house the driver goes for >100 Hz, is this actually correct?

Some pointers of candidates:
- Faital PRO 12PR320?? Top of the list right now. Also other models look good.
- Beyma, P80ND/V2?? 12MC700ND??
- Eighteen sound?? Many models.
- BMS 12N810 looks fine for upper frequencies, but doesn't go very low. There are many other models and this manufacturer seems to be quite respected, although their specsheets are not always so good. Just honest maybe? BMS is the only one that I remember posting their distortion graphs.
- Precision Devices, like PD.124NR1??
- Acoustic Elegance TD12M or TD12S, a bit too expensive with shipping to Europe
- I haven't yet even looked at B&C, Ciare, Eminence, Fane, JBL, RCF, SB Audience etc or any hifi-like manufacturers
- I have got diy speakers with "starting at" prices, but I would like to try something higher now and use them for many years.

Preliminary Cadding
Woofer sims. 2 quite different drivers:

Faital PRO 12PR320 in sealed 22,9 litre box.

BMS 12N810 in sealed 18.6 litre box.
(cyan in excursion if cone force)

What I learn from these sims is almost nothing. They both have unlimited performance for home use and can be EQed to any FR target. The BMS is slightly better around 120 Hz, not sure what causes the hump for the Faital. Probably a sim parameter thing.

Cabinet drafts in attachments.

I built a battery powered amplifier that uses a Sure ME-DV42331 voltmeter display. I burned out 3 of these displays before I finally realized my mistake (I think). The display is inline with the charge circuit. That means both the battery and the charger are feeding into it. The display is meant to check the voltage of the battery and is switched, working outside of amplifier circuit.

The battery is 24V. This means it tops at 29V. Whatever voltage the smart charger is pumping in probably puts me over the 30V max of the Sure display. Oddly, the display works fine all the way through the end of the charge. But the next time I check the amplifier charge after charging, magic smoke--it's happened 3 times like that. So I am assuming the issue is over voltage.

Upon figuring out the issue, I've been disconnecting the display during charging. This is not ideal. Assuming the display is a voltage limit issue, I would like to reconfigure that. However, I have neither the knowledge nor SMD skills to inspire faith. So I am wondering if someone would be interested in doing this for a fee? Option to mod a new display or repair/mod my 3 mistakes. I assume this is just a matter of replacing caps with ones that can handle 48V or something. Thanks

Hello everybody!
I just bought a Technics Rs1500 reel to reel recorder. It comes with a Nab equalization. I would like to modify it to have a switchable IEC equalization that is used on some master tapes.
I have been looking at the schematics(I have attached them). It appears to me that the equalization is obtained with the feedback circuit circled in red plus the one circled in green which changes based on tape speed (the transistor Tr127 is connected to the speed switch).
The circuit circled in red has (if I have done my calculations correctly-I got the formulas here:Andy's Active RIAA Phono Equalization Design Page) two time constants:the first around 3180 μs which is the bass lift and the second around 25μs which is not a time constant of the NAB equalization but I suspect it is then corrected by the second circuit (circled in green).
Now my questions are:
1)Is my reasoning correct? (I'm new in the electronic world)
2) If I wanted to implement the Iec equalization for 15IPS (Two time constants:∞ and 35 μs) what should I do? I think I could increment the value of the of the capacitor 162 or resistor 182 until I get a bass roll off under 20 Hz (∞ is not possible, I've read that people place it somewhere under 20 Hz) and then decrease the value of the capacitor 169 or trim pot vr105 or resistor 181 till I get the correct frequency response ( high frequency roll off at 4500Hz) taking into account the circuit circled in green (I would probably need a calibration tape to do this correctly).
Am I at least partially right? Or completely wrong?
Thank you very much
Benjamin

Who knows about this?

I have a used pair of JBL 2451J drivers on my floor and a new pair of factory aquaplas coated diaphragms ready to install.

There is a small card in the diaphragm container that says the diaphragm is not self aligning and that the warranty is void if you install it by yourself. I have seen this same disclaimer card in all the other factory JBL diaphragm boxes I have looked into (6 or 8 or so)

The JBL 2451 specification sheet says “The JBL manufacturing process permits the use of rim centered diaphragms for instant interchangeability and ease of field service.”
Does not sound like anything special is needed to me.
I do not care about the warranty. I have installed several factory diaphragms in JBL compression drivers; they all look to have no possible adjustment, they all look to be rim centered by design.
Nothing odd shows up on the APx Rub and Buzz sweep.

So what is it? Why the printed caution in the diaphragm box.
Is it just a caution to protect the employment of JBL techs and to protect the warranty from some unfortunate person slipping with a screwdriver?

Thanks DT

Hi all. I'm starting to put an Aleph 5 together and have ran into some questions I figured I'd run by the folks over here.


If you care to know the reasons why, you can read on after the questions, but suffice it to say that my build is going to be 3 boxes: 1 P/S sitting on the ground and 2 mono blocks getting their DC feed from the P/S. Obviously the amps will not have IEC ground as that'll only be on the P/S. The questions:



1. Should I still do an isolated star ground in the amps? Or is it okay to liberally use their chassis for ground?
2. Should I duplicate everything from the P/S after the coil in each amp? Or should I go ahead and finish the P/S outside the amps and just supply the final voltage to each amp?
3. Am I okay running my DC power from the P/S over to one amp and then the other?


Thanks!


The reasons:

So, after reading posts for a few weeks on DIY Aleph 2 I finally decided that I couldn't stomach a 600W heater for 8-9 hours a day, 5 days a week, so I decided to take the plunge and settle for a 300W heater instead. The decision wasn't an easy one, since my NHT 3.3 speakers are only 87db SPL and would have appreciated the Aleph 2, but I decided that I'd instead do passive bi-amp, keeping my ATI AT-1502 for the subs and the double A5 for everything else. That would bring my overall power requirements much lower, since the 3.3 splits the woofer at 100Hz and 60W for everything above that should be fine..



One of the things I wanted to do, since aesthetics don't mean much to me (although I love the stock look of the A2 and A5), was to separate the P/S from the amp, per one of Nelson's preferences. So after reading much I settled on a 30-0-30 1000va transformer (hoping it'll be +/- 37v) and running +/-V over to the amp. Then I thought purchasing a massive chassis with nothing in it seemed like a silly idea, so I decided to do 2 very small chassis. Each comes with 2 heat sinks of 0.36 C/W each, so my total after 4 heat sinks would be 0.09 C/W which I think should be able to handle 300W without any issues. The chassis run 10" wide and would sit side-by-side.


I'm still trying to figure out how to do volume matching between not only these two amps, but also between this set and the ATI that is going to drive the subs, because I'm using a home-made passive preamp with just one volume control and no balance, so I may have to do an op-amp with a volume control when I build the two A5s to feed the existing amp.

I bought this of eBay a few years ago as I figured it was priced not much more than the cost of the parts, and I had a 2A3 itch. It has worked reliably, it came with JJ tubes, I got some Sovteks for day to day.
It was never very convincing though. I recently acquired some Sonus Faber Electas. The 2a3 will, says with gritted teeth work nicely with the Electas, bit they don't. So I finally measured the amp today, it is significantly rolled off at the top and bottom, and manages a whopping 1.5 watts with the Sovteks, and 2 watts with the JJs.
I have been plotting their reinvention for some time, and have just got some ISO FC20s, and am open to ideas for a circuit, but I feel maybe a three stage with SRPP driving the grid, or maybe fixed bias and a cathode follower for the grid.

I'm wanting to add balanced outputs to some of my pre-amps. I've read the Jensen document and am planning to use the 'simple alternative', which I'm attaching for convenience. This just involves adding two resistors and a capacitor to balance the impedence.

Since this question comes up often enough, I'm thinking it would be worth it to make up a simple PCB that would contain the required components. The other two attachments show how this would go in the context of the Pass B1 Korg and then just the parts that would be on the PCB. Of course the actual board would be designed so that any values could be used. Can someone verify that I've got this right before I fire up KiCad to build the board?

One slightly unrelated question about KiCad, in case anyone reading this happens to know: How do you add mounting holes?

I stumbled across a random video on Youtube, then spent most of the morning watching every video in their catalog and following some of the links.

This band is mostly from the Ukraine, Russia or eastern Europe, English is not their native language, and they have never seen any of the "Chicago" incarnations play live. Somehow they have learned to play old CTA stuff better than the current "Chicago" does.

(I've Been) Searchin' So Long - Leonid & Friends (Chicago cover) - YouTube

25 or 6 to 4 – Chicago (Leonid & Friends сover) - YouTube

Questions 67 & 68 – Chicago (Leonid & Friends cover) - YouTube

Leonid & Friends | Official Band Website

In a class AB, How much output power can I get from 2 pair of these into 4 ohm? My simulations show about 4A and 18w per device dissipation for 125W output. Simulations was for TIP122/127. I don't have 142T/147T models.

https://www.mouser.com/datasheet/2/308/1/TIP142T_D-2320223.pdf

https://flic.kr/p/2m9pr6g by https://www.flickr.com/photos/64593884@N08/

I currently have a Topping E30 DAC connected to a Schiit SYS (10k volume pot box) connected to a Hafler DH-101 preamp (using for tone controls since I’m hearing impaired and my idea of “neutral sounding” may be different than yours) and that is connected to a Hafler DH-120 amp (22k input impedance) or sometimes another amp with a 47k input impedance. The output level of the DAC is so high that the volume level of the preamp is either all the way off or too loud, hence the Schiit SYS pot box inline. Interconnect cables are very short, the longest being 3’ from the Hafler preamp to the power amp.

Having wired the preamp myself and looking over the schematic, the signal from the DAC/pot box goes directly through the DH-101’s own internal 50k volume potentiometer before it goes through the active high level gain stage. The DAC does have a remote and digital volume control (It can be put in pure DAC mode or preamp mode.) Should I just use the preamp mode instead? I bought the Schiit SYS since my other DAC (Schiit Modi 3+) is fixed line level out only. The Modi 3+ has an output impediance of 75 ohms. Since the Modi 3+ and SYS are wired in what is basiaclly series, it’s 5.075k ohms maximum (Schiit’s website says the SYS’s output impediance is 5k ohms maximum.) correct? I’m not sure what the output impedance of the Topping E30 is, but it would be xx ohms + 5k (max).

I ordered a Schiit Loki Mini+ a few days ago and my plan is to stop using the DH-101 and just use whatever DAC (Modi 3+ or E30), the SYS, and Loki Mini+. Is there a correct way to connect these 3, or does it not matter? I was thinking DAC > SYS > Loki Mini+ > power amplifier. Or do I want the DAC connected to the Loki Mini+ and then the Loki Mini+ connected to the SYS?

I have asked this question before and been told that it is ill-advised but I suspect the answers are led by the lack of facilities to calculate.

Logically, two 6.5" woofers in a 20l cabinet will perform better than one. By the same logic two 6.5" subwoofers in that same cabinet will perform better than one. Subsequently, one subwoofer and one woofer will offer better bass performance than two woofers. Perhaps lower the bottom end by 10hz?

I ask the question because many floor-standing speakers appear to be a waste of volume.

The simplest solution would be to wire the drivers in parallel. However, a rumble level high-pass filter on the woofer could be used, but this brings the passive radiator effect into the equation.

What factors govern the frequency response of a driver? e.g. If I feed 25Hz into the sub driver it oscillates wildly, The same signal into the woofer is largely ignored.

Thoughts.

After months of seeking the AK4137EQ all over the possible places, I finally managed to buy 5 pieces still sealed in reel. I do have to pay an overpriced total for them though, not a surprised at all. These all come from a shop in Japan. I only share 4 of them, want to keep one for my DSD-PCM build.

Price is $48/ea plus fee and $15 shipping by Registered Airmail. Unfortunately I'm not able to ship to destinations that have very strict custom like India or Russia.

Hey People!


As I like diy and music, I was hooked from the first second. They're not really complete yet, but right now I'm hearing Nick Cave with my first diy speakers and it' great. It's a 2.1 - 7-channel fully active 2.5-way + SW system with chipamps built on a pretty low budget, but it sounds just wow!


Sure I'm thinking of what could come next.
One of the many ideas could be the next step getting deeper in "real" diy audio. I have no experience, therefore I ask you.


What do you think about a FAST, with Dayton PS220-8 ( 404 Not Found ) and built in Pass Class A mono blocks (and a 12 or 15" active SW).


Hell yeah or let it be?

Hi,
I am guessing it's MLTL or slot-loaded vented subwoofer, which one is it or ....something else. Thanks,

Greetings,
trying to understand i/O voltage/Impedance matching:
to explain:
1) purchased Kenwood Basic M1 Amplifier, and Basic C1 Preamp; very satisfied
M1 Amp: Input sensitivity: 1 V 47KΩ..
C1 Preamp: "Output Voltage & Impedance: 1 V/less than 600Ω, maximum output: 5V Load Impedance: 47kΩ"... everything wunderbar

2) Went shopping for additional Basic M1 (for my workshop), purchased via Ebay, UK
M1 Amp purchase also included Basic C2 Preamp (oh goody! lots of extra features!):
C2 Preamp: "Tape Rec: 150mV, 220Ω", PRE OUT: 1,000mV, 100Ω"
Assumed was included in Kenwood Basic series, was compatible...
However, at approx. 1/3 - 1/4 volume... distortion extreme (mis-match... really stupid move by Kenwood??)

To the Point:
Audient iD14 as DAC/master volume control:
(one audio source: my PC/TOSLINK)
"Maximum output level - +12dBu (0dBFS digital maximum)"
"Output Impedance <100Ω"

Audient iD14 has Balanced TRS outputs - intend to use unbalanced TS-RCA adapters

Is the iD14/M1 Basic combo a mismatch problem? (haven't tried it yet... would really hate to brick something)
solved with HiZ/Loz adapter? or DI box (before the Amp)
thank you for a useful recommendation (if this works, C2 will become a doorstop or boat anchor)

Had trouble finding the detailed spec sheets, here and index, click on the part number,
then click on the .pdf link:
Tube Guitar Amplifier - Output Transformers (1750 Series) - Hammond Mfg.

That is for 1750 and 1760 part numbers.

I finally got around to a modification to my DAC1600 and was wondering if anyone else has tried the same.

I do not use the balanced outputs, since the rest of my system doesn’t support it. I had decided early on that I could pull the unused complimentary PCM63s and stack them onto the existing chips. I was hoping to see how much better/smoother the DAC could be.

Before I pulled them I reviewed the schematics and realized that the complimentary ‘balanced’ half of the circuit was identical to the ‘unbalanced’ circuit with the exception of running the signal through a hex inverter to make the signal a mirror image.

I decided in the end to try and remove the inverters so both chips pass the same signal and sum them after the IV stage at the XLR jack with a custom cable. Tests proved that this worked and is easily reversed back to stock if I ever wanted to.

The difference is subtle, but noticeable if you look for the differences. I have especially noticed that some background acoustic guitar picking sound much more realistic than the stock unbalanced unit the guitars went from sounding ‘hard’ and like steel strings to much more detailed ‘gut’ strings with better attack and decay.

Ultimately I have just created a jumper plug out of an old XLR connector, which sums the two signals and can be passed through the standard unbalanced jack with my normal interconnects. The unbalanced jack shares a trace with half of the XLR so making the jumper plug just connects the two signals which pass it to the unbalanced jack. Pulling the jumper reverts the unit back to a stock unbalanced output so A/B testing is also not too difficult.

Has anyone ever done a similar modification to their DAC? I’ve searched the web but didn’t find anyone who summed the signals after the opamp/buffer section.