I've got a very basic kenwood turntable here. It has a DC motor rated at 12V and a simple power supply.

It came to me non-functional. On investigation the internal fuse was blown. The fuse sits between the transformer and the rectifier (i.e. not on the mains side).

When I replaced the fuse (with the correct value) it initially ran OK. For good measure I decided to replace the smoothing cap as it looked suspect - although I think it was probably glue rather than leaking electrolyte that led me to doubt it... It ran OK for a few minutes before and after capacitor replacement.

I then tried it again and the fuse blew right away.

According to my multimeter the low voltage side of the transformer is providing around 10V AC.

I have checked the diodes (removed from circuit) - they are all OK.

My best guess at this stage is either the motor is bad and drawing too much current, or perhaps the transformer is producing inadequate voltage and similarly causing the motor to draw too much current.

Any thoughts? Inclined to cut my losses at this stage...

This thread is for discussions about the Mezmerize B1 Buffer Pre Amplifier. For more information on this product, please see the product page below.

• Product Page
• Schematic
• Bill of Materials
• Dimensions

Threads on diyAudio that relate to this product (If we have missed one, please post it in this thread and we will add it to the list):

• Build Thread: Mezmerize Build Thread
• Build Thread: http://www.diyaudio.com/forums/anal...otrodded-blue-dcb1-build-132.html#post2700123
• The B-1 Mezmerize is based on this: B1 Buffer Preamp

Enjoying this recent acquisition looking new, through an 85 watt per side Yamaha receiver. I am hearing the most coherent bass than I have heard before from an enclosure of this size.

When I set up a pair of Tannoy E-11 (that I regarded to have the most natural sounding bass within the range bookshelf speakers operate in) for an A/B test, on the Wharfedales male vocals sounded like they were far forward of the rest of the music compared to my reference Tannoys.

Questions - was the Diamond line designed to do this on purpose or is it a side effect derived from something else the Wharfedale engineers were attempting to correct with the crossover?

I feel kind of silly asking, as it just occurred to me that the voicing the different product lines of Wharfedale is more heterogenous than other brands - I don’t think that this was true in the 1970’s, but I think that it has been true since the 1990’s.

Hi does anyone have one. One of mine has failed.

Apparently there are a few variations. Mine has the number 8778 printed on it.

Thank you, David

Hi 👋

I designed and 3-D printed experimental pod speakers. I’ve tried to keep the cost down as much as possible, which is important me. I chose the two following drivers for my tweeter and midbass enclosures:

Pioneer TS-M800Pro

Dayton Audio ND25FA-4


I chose a cheap crossover from Amazon is there any problem to use it or better to buy a different one?




Here is my speaker design concept:



Optional sub pod:


The color is not finalized. You can say it’s ugly too 😆

I have this amp that uses 2 NE5532 and 1 TL072 in the output
the 2 NE5532 are signetics, and the TL072 is a TI

Should I change those with newer op amps ?

The amp is a Hafler Pro5000

see schematics

I’ve had a chance to play with the Sony 2SK79 and am hoping to try out other small signal SITs. Looking at options for a SIT FE in my OPT-coupled SIT amp.

Hello everyone, I'm new to the forum, but I know there are some very skilled technicians in this group.
I'm slowly building my small audio repair lab, and one of my projects is to build a tube audio amplifier.
Among the instruments I’ve painstakingly purchased, I have a beautiful HP-339A distortion analyzer… the only problem is that it doesn’t work!

I’m trying to repair it, but I’ve encountered several difficulties.
The oscillator works correctly; on the oscilloscope, I measure a perfect sine wave with the correct frequency and amplitude from 1 Hz to 100 kHz.
However, the notch filter does not track properly. The LEDs indicating frequency adjustment often stay lit, even when using the internal oscillator directly connected to the distortion analyzer input.

Occasionally, the notch null occurs (usually when the third frequency knob is set to 1x or 100x, almost never at 1k or 10k), but most of the time, it doesn’t.
The problem has worsened over the past few weeks. I’ve checked all the electrolytic capacitors in the instrument, including the tantalum ones, without finding any issues.
I also cleaned the frequency switch contacts with isopropyl alcohol, but it didn’t improve the situation.

Therefore, I’m seeking help from anyone who might have faced a similar problem.
Thank you!

Greetings from Vietnam, I'm Glad to be here to start build some interesting thing like speaker, thanks for accepting!

Hi everyone, this is Roberto. Thanks for the best audio player ever made.

Ian Canada FifoPi Q7II + TransportPi AES + PurePi II complete stack with accessories and Lifepo4 186500 batteries. Will include RPI4 1GB with 16gb sd card for free. Used for less than 100hrs. I am going active with minidsp so it no longer fits in the chain. Location - india. Price- Rs 43000($510)

There were two different goals that forces these ideas and concepts:

1) Only the adjust free power amp concerning bias adjust for idle current
(self biasing - I know only the Quad 405 current dumping amp and some related topologies)
2) Additional no leaving the Class A operation and low idle current at the same time
under all load conditions > 4 ohms (I know only Threshold and Mark-Levinson - Madrigal)

Where I can find an overview of a wide range of commercial brands/ models so as topologies concerning this two categories (only push-pull topologies - not single ended)?

Here patents/examples for this topologies:
Patent US3995228
Madrigal Library: Adaptive Bias
Patent US4160216
(automatic and dynamic/adaptive bias for classA under all conditions without external bias adjustment)
Patent US4077013 (automatic Bias without external bias adjustment)
http://www.diyaudio.com/forums/solid-state/1573-linear-tech-lt1166-chip.html (automatic Bias without external bias adjustment)
Patent US5216379 (automatic and dynamic bias without external bias adjustment)
http://220.232.128.43/audio/quad405/Current Dumping Audio Amplifier - PJ Walker.pdf (current dumping - ClassB without external bias adjustment)

P. S.
In this case also this threads could be of interest:
http://www.diyaudio.com/forums/solid-state/162559-new-class-super-non-switching-need-revival.html
http://www.diyaudio.com/forums/pass-labs/104399-importance-being-biased.html
http://www.diyaudio.com/forums/solid-state/58561-low-biasing-high-biasing-class-ab-amps.html
(complementary push-pull FET output stage with no source degeneration resistors and good working of bias control - i. e. no variation in quiescent current at the same time).

I posed this question, because I'm looking for a simple answer as to why this resistor is needed In series with the speaker when measuring impedance.
My thought would have been that you use your amplifier to send a signal to the speaker, staggering the frequency sweep to one or two seconds at each. frequency and measuring the voltages at each frequency across the speaker. Have an ammeter In series with a speaker. To measure amps. One can then plot a graph of the impedance response.
Please keep in mind that my knowledge is at beginner level, and if you could, think of the simplest individual when constructing your answer. I understand equal. V=IR., but not a whole lot more.

I am hearing a low frequency hum at higher volumes when using my turntable and upon searching as to why that might be I saw a video saying to be sure to have your turntable grounded to your pre amp, receiver, or whatever, the problem is I see no ground lug on the stated turntable however there is a ground lug on my receiver?

Do I have an un-groundable turntable?

Thanks

I wasn't sure where to post this so apologies if it's in the wrong place.
I don't know much about what's available, brands or prices etc..
I'm in the UK and Amazon are selling a trio of JJ ecc83 valves for £50 ($64 ish for our American friends).
Is this about the best I can get for fifty quid and no more!
It's a trio I need btw, and I'll be looking at replacing the rest of the valves in the amp at a later date.
PP EL84 amp btw.

This is Vlad's kit. First image is what he provisioned to me, essentially complete without chassis, wiring between boards, input/output hardware, power inlet/fuse holder.

1541 chip, non-oversampling, 6N2P-EV NOS tubes, nice caps.

Second image is kit housed in larger Galaxy chassis supplied by DIY store.

Fired right up, sounds superior to the low end Project e DAC as expected. A really good value and step up. I run an old high end Sony CDP into a Monarchy anti-jitter box, haven't tried streaming contemporary sources.

The first "Miss World Pageant" winner of 1951. From the Wall St Journal:

One evening in 1951, over dinner at their family’s home in Stockholm, Kiki Håkansson’s older brother—who had once told her she had “the face of an angel and the body of a pig—proposed entering her in a coming beauty pageant.

Håkansson said she wasn’t interested. Her brother entered her anyway.

Next thing Håkansson knew—and in retrospect, much of her life would unfold in this same, less-than-fully-autonomous way—she was at the Lyceum Ballroom in London, competing in the first-ever Miss World pageant.

Miss World is now the longest-running major international beauty competition in the world, having preceded Miss Universe by one year. It remains a glitzy annual event and has expanded its focus to include contestants’ humanitarian work. But that first summer, when it was a part of a much larger event, the point of the competition was simple: Its official title was “The Girl Bikini Contest of the Festival of Great Britain.” (Because contestants from various countries entered, the press nicknamed the contest “Miss World,” and the next year the pageant officially adopted that title.)

Hi,

as a followup to my othe thread (New diy open baffle project: 3-way with AMT) starting my open baffle project I am a bit further now in decision making:

Tweeter: AMT23D6.1-R (88dB, 8Omhs) is already ordered
Mid woofer: Audiotechnology C-Quenze 18 H 52 06 13 SDKA – last perhaps in a version with the J-magnet instead 90dB

I am now looking for a matching subwoofer, xo is planned around 200 Hz. I am thinking of one of the following:

• Acoustic Elegance Dipole12 or Dipole15
• Supravox 285 GMF
• SB Audience Bianco 15OB350 Open Baffle (or the 12OB150)

I would prefer the 12" drivers as I awant to build a rather slim baffle but with side wings (U-frame). I tried to chosse drivers with higher sensitivity to compensate for the baffle loss for low frequencies. Are there reasosn one of them would not match well with the above mid/high drivers?

Another question:
Is there a way to simulate U-frame baffles? I wand to make them triangular shaped, is there a way to simulate this as well? Can I use Basta! for this?
Is the response the same when I "flip" the wings to the front as Basta! can only simulate flat baffles as far as I know....

I am searching for a strobe light for my Micro Seiki DDX-1000. According to the parts list part designation is "NL-77". I was checking the archive of this forum and it looks like that this part is NLA. Does anybody have an idea how to fix this or even found a new source for this neon light? The light is internally pulsed so a direct hook up of the light to line voltage and -frequency is not possible here.

Hello everyone and thanks for accepting me into the diyAudio community!

I work as a Broadcast Technologist / Electronics Technician at a local college here in Calgary. Before coming to Canada, I was a Telecom Technician, so my career is a diverse one 😉 but basically revolves around electrons! It's same with my hobbies: any DIY electronics, ham-radio, computers, programming, house-fixing, woodworking, ...

diyAudio popped up many times when I was searching inter-webs for a solution to a problem. Finally, decided to join you for my own benefit and also to try and help out other members if it's something that I have some sort of expertise. At the moment i don't have many projects going on. Latest one is tinkering with an old Sony STR-V2 receiver whose amplifier doesn't work any more. Instead of fixing it, idea is to use a class D module as an output amplifier and build an Arduino controlled preamp and audio switch. Prototype of the preamp is in the testing phase, it works. It uses NJW1194 - digital volume and tone control with 4-input switch. Project is going slow a bit, since we are in a house moving mode, which will be over by spring, i hope. Until then, I might not be a very active.

Cheers to everyone and thanks again!
Goran

I'm a new member from Europe. Hi-Fi audio is my love.

Hello,

Once again with a DAX problem. I now have a DAX decade balanced that works great on all inputs but it came with the ADX analog to digital converter that allows to use 3 different analog inputs with the DAX and its built in preamp. The DAX sold with this option have an additional board and a differently programmed Microprocessor. My issue is that the DAX can command the ADX module, correctly switching between the 3 inputs but it doesn't lock on the digital signal fed back from the ADX through BNC coax. Any other dac I tried, equipped with a BNC input does lock on the ADX without a problem and it sounds good. The actual BNC input on the DAX works fine with other sources.

At one point the DAX managed to lock on one of the 3 inputs but displayed Ncode on the screen and there was no sound.

Happy Thanksgiving everyone! We have a lot to be thankful for! Safe travels to all that are moving about today. Have a good weekend as well.

Hi folks,

I'm a casual builder of a lot of things, whenever i need something, ranging from sketching to programming, a little music, grafics, things in and around the house, shelves, modification of furniture, piping, recumbent bikes, bike trailer, car trailer mod for the recumbents, pc, audio, mod of speakers, and finally my very own omnis (for the bedroom) which i'll introduce and share with you later - and are the reason i registered here...

i was long "in love" with my Duevel Planets, which perform really great with any kind of music, all 3 amps i've tested them with and in any room and environment i tested them - true omnis in every meaning. They are in the music room (named after the piano and the keyboard, not the omnis), and i have a 6.1 set in the living room consisting of Jamo Speakers I purchased bit by bit during my studies and a Denon digital surround amp, 20yo meanwhile - time flies, and a multi room streaming system consisting of two Raspis connecting to the amps via bluetooth. Im a bit of an audiophile but not that very much that i would spend a lot of money on that, i like 3 dozen styles of music, and a handfull i really dislike and as written above i have a lot of other hobbies and work to do...

regards, Dan

Hi there,

Someone needed with knowledge in speaker measurements.

I am using this bible by Kimmo Saunisto for measurements speaker simulation: https://kimmosaunisto.net/Software/VituixCAD/VituixCAD_Measurement_REW.pdf
Most of it is clear and made for newbies, like me, but some parts of it still are confusing, one of it is related to nearfield measurements. In page 9 of it there is one sentence, which is very straightforward, but still leaves more questions than answers:

"Locate woofer cone close to floor or wall to make half space conditions at LF."

What does it mean? I measure farfield measurements of speaker with REW at home, with "good enough" conditions - driver at ~1.2m height from floor, and distance is around the same or more from nearest walls and big objects. Usually it is 1 meter distance from source to mic, as in most instructions. Gating is ~4-4.5ms. Then I move microphone stand close to the driver, at ~0.005m distance and measure it. The same for the BR vent. Then I do importing into Vituix and start sim.
EDIT: Yes, I do the merger and everything else as in this instruction

The question is - do I have to move speaker close to the wall when doing nearfield measurements??? That instruction says that, but I haven't found it mentioned nowhere except there. If I simply put it close to the wall at 90 degree angle, I am getting results of big SPL boost at ~150-500 Hz and drop below 150 Hz. When I import it into Vituix and do exactly the same with both nearfield measurement options the crossover needs to be adjusted a little. That is for 2-way. For 3-way or 4-way speakers the data for biggest woofers will get skewed even more. And yes, with near wall measurements the simulation looks uglier. I don't care much about the "beauty" but more about accuracy.

Can someone with experience comment about this?
Thank you.

Hello. I have a pair of Mission 753's that were donated to me by my son. I have enjoyed them for the last 2 years as a significant improvement to my listening despite having a cheap amplifier. My amp is an old Cambridge 30watt thing with a bi-wire feature which how iy is connected to the speakers.
Over the last month one speaker had begun to distort, then it increasingly started to fade in and out volume wise and recently it has began to play up and then drop out for extended periods. I have swapped it over at the amp output and the fault remains and the good speaker stays good when swapped.
Help to work out what is going on and how remedy this would be appreciated.

Thanks.

Hi all, please forgive what to a lot of you are very newbie questions. I have searched quite a bit for answers, but it's kind of hard to digest and apply everything in practice...

I'm building a subwoofer for music / HT use in my living room. It's going to replace an XTZ 10.17 (non-edge). For several reasons, I'm limited to a relatively "thin" and small box. I've bought a used RSS315HO-44 in very good condition that I'm going to use for the project. I understand this is a driver that should work well in small enclosures. Haven't bought an amplifier for it yet.

I don't listen crazy loud, but I like proper low extension, and I feel the XTZ lacks just a little bit in that area. It works OK enough for my needs, and I could probably have lived with it for the sound alone, but it doesn't really fit in the living room so I want to get rid of it. My living room has one open side, meaning the total room is quite large.

I was originally thinking to build a sealed box. According to WinISD, in a sealed box of ca 25 liters (I could go up to 41 liters), the driver at Xmax will give me around 97.5 dB @20hz and 104 dB @30Hz (700W) (using the SPL tab in WinISD). I believe this is around 1 dB better than the XTZ at 20Hz but around 5 dB worse at 30Hz. The -3dB point is in the mid 40s. If I make the box vented, 32 liters net volume (when vents, bracing, driver etc are subtracted), tuned for 24 Hz, vent pipe dia 9.74 cm, vent length 114 cm, I get 102 dB @ 20Hz and 110 dB @ 30 Hz, with a very flat response from 100 to 27 Hz. Port velocity is a bit high at 28 m/s max, but I believe I'll never reach that level with music, and for HT it might be OK. Driver excursion seems to be OK provided I high pass it @17 Hz. I'm thinking to stuff the box with polyfill.

Ok, so here are my questions:

1. Based on the WinISD simulation, the vented version seems like the obvious choice. It gives a lot flatter response in general, and a lot more dB below 45 hz. But still, I see a lot of people recommend sealed boxes and one of the arguments is that room gain gives the sealed box quite a bit more output in-room than the simulation indicates. What do you guys think? The sealed version is of course a bit easier to build, especially for a first timer.

2. I understand I could do EQ or even Linkwitz transform for the sealed box (provided I could get a proper DSP in the chain), but I would still be limited by the xmax of the driver, right? So I can't really get more output @ 20 and 30Hz than what WinISD gives me above?

3. Since I'm very inexperienced, dB numbers are a bit hard to relate to. I don't really know if 97.5 dB @ 20Hz and 105 @ 30 Hz might actually be quite sufficient for me. But as I said, I feel the XTZ lacks just a little bit (not much) of power for my needs in that large room. What are your experiences, what dB numbers would you think I should aim at (using the SPL tab in WinISD)

4. If I do vented, do you see a problem with the long vent pipe? I'm thinking to use flared ports (inlet and outlet). I know it's generally recommended to not go above 20 m/s but I believe my music listening will not exceed that, and for HT it's maybe not that important?

5. If I do vented, I'm thinking to use thin PVC pipe (HVAC air pipe) with 1.3 mm thick walls for the vent, and buy "aeroports" for the flared ends. Do you see any problems with this, or do you have better solutions to recommend?

6. The driver can be connected in 2 or 8 ohm configuration. It's easiest and cheapest to get an amp that can do 700 watts @ 8 ohm (bridged) as far as I can see. Does this give any disadvantages over using 700 watts @ 2 ohm (paying a bit more for an amp that is 2 ohm stable)?

7. Any general input or recommendations for the build?

Greetings from Western Australia: I have been messing with speakers, amplifiers, etc since the early 70's. I am currently setting up a system incorporating electrostatic panels from ER Audio, harking back to my early experiences hearing Quad Electrostatics belonging to the father's of girls I was dating.

A few years ago, I made a 814 push pull amplifier like this:



step-up tranformer as input stage, driven by a small amplifier.
It was nice but today I would like to make a more "conventional" amp with a tube driver.

As I already have a pair of LL1660 10mA, I wonder if a 5687 (200V/10mA) can drive the 814's into A2 region or definitively not ?

From the datasheet, control grid at +30V eats 10mA of current, will it works ?

It may be better with a direct coupled CF or even better with solid state but as I have these 1660's, I am curious..

I bought these tubes recently for some tube rolling from HotRox

Fantastic little tubes, but unfortunately due to high frequency damage to my ears, I have to sell them on.

Been used for approx 180 hours (150 hours burn in and then some listening)

These are a matched pair

Price - £95 GBP

Location - Glasgow (UK) - Can be picked up

eBay link is below for safe purchase

https://www.ebay.co.uk/itm/267080761574

I've been planning for stereo upgrading in my car. I'm going to build 3-way system. This is my very FIRST TIME using multiple amplifiers. I'm not sure if my car's electrical could support abundant amplifiers used. My car is 1983 380SEL.

Attached pics are diagrams of bi-amping vs. tri-amping that I could figure of:

• System A is tri-amping utilizing 3 amplifiers. The Soundstream amp is actually 5-ch configuration but it could be switched to 3-ch by bridging front and rear channels together.
• System B is bi-amping utilizing 2 amplifiers. The Soundstream amp now will perform 5-ch configuration so as to reduce an amplifier used--the tiny Sony amplifier.

As mentioned earlier I've never experienced with multiple amplifiers system before, I need advice from experienced people to suggest which system you recommend. And are there any modifications on my car's electrical system required to support the enormous current consumption for these systems?

Here is a diagram of power connection of tri-amping system with wire size labelled. I need assistance to review the diagram if it's acceptable and has proper wire sizes and fuse used.

PS. All components will be located at the rear trunk of a sedan except head unit, fuse, and battery.
PS2. The ga. unit may refer to AWG standard, I'm not sure. I just called it ga. following the user manual of Soundstream.

Hello everyone,
We are a small engineering company «ChipDip.» has released three complete uHAT projects for the Raspberry Pi.
These are expansion boards for high quality sound.
The driver and overlay for Raspberry Pi OS is called ChipDip DAC
To build a streamer, you can use any Raspberry Pi. The microcomputer is installed on a 40-pin audio card connector. As software, use any player or OS image that is recorded on a microSD card for Raspberry Pi. Operating systems automatically detect the ChipDip DAC audio card and configure sound output to it. In some players, you may need to select ChipDip DAC as the audio output device.
All files just posted on PCBWAY.

We really want you to make any of the projects and share your comments.
I will answer any questions and support you in assembly and programming.

The first project is DSP Machines 1.
DAC, audio processor, amplifier for all Raspberry Pi. ADAU1701 + TPA3255.
PCBway.

• Compatible with all Raspberry Pi
• Compatible with SigmaStudio and ChipStudio software for audio processor rebuild
• Official support for RaspberryPi OS / Volumio3 / rAudio1 / Album Player / OpenELEC / Manjaro OS / MoOde and more
• DSP ADAU1701, TPA3255 amplifier chip for best sound quality.
• Output Power 2х120 W
• Audio output connectors: two pairs of screw terminals (left and right)
• Built-in EEPROM for automatic OS configuration (with write protection).
• Automatic frequency switching according to I2S input signals.
• Sample rate and bit depth range from 44.1kHz/16bit to 192kHz/32bit

ChipStudio
SigmaStudio & ChipStudio projects DSP Machine1

Few photos.





The second project is DSP Machines 2.
DAC, audio processor, for all Raspberry Pi. PCM5242.
PCBway.

• Compatible with all 40-pins Raspberry Pi
• Compatible with PurePathStudio and ChipStudio for audio processor rebuild
• Official support for RaspberryPi OS / Volumio3 / rAudio1 / Album Player / OpenELEC / Manjaro OS / MoOde and more.
• High quality PCM5242 DAC for the best sound quality.
• Audio output connectors: 2 x XLR (left and right)
• Built-in EEPROM for automatic OS configuration (with write protection).
• Automatic frequency switching according to I2S input signals.
• Sample rate and bit depth range from 44.1kHz/16bit to 192kHz/32bit

ChipStudio
PurePathStudio & ChipStudio projects DSP Machine2

Few photos.




The third project is DSP Machines 3.
DAC, audio processor, for all Raspberry Pi. ADAU1452 audio processor and AD1934 DAC.
PCBway.

• Compatible with all Raspberry Pi
• Compatible with SigmaStudio and ChipStudio software for audio processor rebuild
• Official support for RaspberryPi OS / Volumio3 / rAudio1 / Album Player / OpenELEC / Manjaro OS / MoOde and more.
• High quality ADAU1452 audio processor and AD1934 DAC.
• Audio output connectors: 2 x XLR (left and right), SPDIF, Coaxial
• Built-in EEPROM for automatic OS configuration (with write protection).
• Automatic frequency switching according to I2S input signals.
• Sample rate and bit depth range from 44.1kHz/16bit to 192kHz/32bit
• Color indication of sampling rates

ChipStudio
SigmaStudio & ChipStudio projects DSP Machine3

Few photos.

Hello everyone

Have to know resistor code and value of R22

On right channel i got a RN60/78ohm
on the left channel it was no one

In the attached file some one have written down R22 is a RN65 / 75ohm is this true ??

if some one know the original value and code of R22 plz let me know!

Can some one who got one BGW 750C that dont have bean tutched maby take it apart and look ?

Seeking SBA SB34SWPL76-4 subwoofer ideally in Canada

Patrick died late last week, after a long illness (here: Patrick-other-concerns)

Patrick was entertaining and provocative company, a talented man and a fine designer and builder of valve amplifiers.

He was a powerful cyclist with a fine nose for a good bottle of shiraz and an ear for a decent LP. And generous with information, including medical data that he shared with other cancer survivors.

caveat...I'm an older audiophile and Linux user. I claim no coding skills and won't be supporting this how-to and any querys. However, it does work for me today and, as has been said before on this forum, 'even a blind squirrel sometimes finds a nut" 🙂

ADC Analog to Digital Converter AKM5720 I2S 24Bit / 96kHz - Audiophonics
Lusya lossless digital audio I2S ADC decoder Support 24bit 96K I2S Signal output A1 003|Digital-to-Analog Converter| - AliExpress

This walk through should set-up the above mentioned ADCs for recording.
Connection to the ADC is as for an i2s dac but changing the default Pi output GPIO Data pin to the Pi Data input GPIO pin...look up the GPIO pins for those.

The setup script will auto-detect your Pi model. I used a Pi2B with no other peripherals attached.
I am assuming you have a little Linux knowledge and are running a Linux PC.

Download Raspberry Pi O/S lite
Operating system images – Raspberry Pi

Burn to a micro-sd card using Balena Etcher
balenaEtcher - Flash OS images to SD cards & USB drives

With the micro-sd card still in your computer write an empty file titled ssh to the boot partition of the card. Option #3...
SSH (Secure Shell) - Raspberry Pi Documentation

Eject the card and insert it into your Pi (ethernet wired) then boot.

ssh to your Pi from another PC and follow the prompt to change the password.

Run..
and in the Advanced tab select to Expand the filesystem. Exit raspi-config but don't reboot yet.
Run...
and uncomment this line to read..
Save and exit.

Now reboot and log back in.
Do both..
Then reboot and log back in and..
then..
and finally...

When that's all done reboot, log back in and ..
and change this line..
to read..

save and then rerun the installer script..

reboot log in and run ..
output should be...

then run ..
output should be..

If that is all good try recording with..
which should give (with no input to the adc card yet) the following output and a basic vu meter showing 00% for both channels. (ctrl + c to stop)

Connect a source to the card and adjust the #1 dip-switch for gain then rerun the recording command with some content playing.

With much thanks to forum member Phofman posts 7 & 9 here..
SBC with I2S codec drivers for playing/recording

and the information from this site..
Raspberry Pi Wiring & Test | Adafruit I2S MEMS Microphone Breakout | Adafruit Learning System

If IIUC the set-up above uses the scripts for an i2s device (microphone) modified to work with the ADC by changing it from slave to master mode ( CBS_CFS to CBM_CFM)

You will most probably need a Pi4 with 4GB ram (or better) (or set up a ram-drive for lesser Pi. I did this to allow one side of an LP to be recorded then moved the files to my PC ...rinse and repeat.)
To send the audio elsewhere in real time you may need to research piping to external devices or over the network.

Hello,

I would like to share what I am working on and possibly get some help on area's I am not very familiar with.

Introduction:

A few years back I was doubting to change out my Yamaha FZ1 Fazer motorcycle for some other type, however I failed to find a suitable replacement. Then I stumbled upon a really cheap turbocharger on a local marketplace and since I always wanted to do something with a turbo I decided to convert my FZ1.

I bought the turbo and started on the project, I crafted spagetti exhaust headers from stainless steel to fit the turbo just below my radiator and close to the engine block. Then I crafted an aluminum plenum for the inlet side, changed out the OEM ECU and OEM dash to fit an EFIgnition 46 and KOSO dash. Fitted an extra fuel pump, oil return pump, water cooled- inter cooler, wiring loom, etc, etc, etc..

Wonder well thing started on the first attempt and did some ECU parameter tuning (ignition + fuel) to get it running atmospheric. I found that the turbo kicks in really early (0.7 bar @ 5000 rpm) to I was concerned about engine knock which will destroy aluminum pistons and cylinder head in seconds. So I decided to do research on measuring engine knock. One thing to reduce the chance of engine knock it to lower the compression ration of the engine. So I stripper the engine, added a 1.5mm aluminum plate to the base of the cylinder block and rebuild the engine. The other thing the reduce the engine knock it to run a very rich mixture, normally an engine runs on about 13-16:1 air/fuel ratio, on turbo engines 10-13:1 is more common. The there is ignition advance, which can cause the mixture to burn to early and push back on the rising piston (not good!) So, the only way to see where you are going is to measure stuff. I have a wideband- and a narrow band lamda sensor to measure the air/fual ratio, I have an exhaust temperature sensor, now I just need to add the knock sensor to complete things and hopefully keep my engine in one piece during ECU tuning.

Engine knock (or detonation) is uncontrolled burning of the air/fuel mixture, it's more of an explosion then anything else (read) This is caused by lean mixtures and/or high gas temperatures within the cylinder. If knock occurs this can be measured with a knock sensor which is a piezo microphone bolted to the engine block. Each engine layout has a specific knock frequency (calculator) , for me this will be around 7 KHZ. Now I once bought this nice arduino engine knock shield , but this knock shield has a band pass filter that need to be configured for the correct frequency to work... since I do not know this frequency exactly , relying on this shield to work is doubtful for me, so I want to do FFT on the signal coming from the knock sensor, visualize it and make it audible (headphones)

For this I bought a Stereo ADC Board WM8782 I2S 24bit 192kHz from Audiophonics, and connect this to a Raspberry Pi4 over I2S.

Following the excellent post of DRONE7 I have the I2S ADC converter up and running now with a knock sensor (KS4-P) connected:



The WM8782 is configured to do 44.1KHz sampling @ 16bit/128fps , on higher sampling frequency's or other bit resolutions the audible sound was horrible (a lot of noise) I managed to improve the sound quality a bit using the twisted pair layout (as described in the WM8782 evaluation board document from Cirrus) , so on the current settings I can make out a clear 'tap' on the knock sensor, I have noticed the sound quality improving when I clamp the sensor , I assume the voltage of the piezo rises during clamping which improves the signal/noise ratio. Further improvement would be to add an electromagnetic shielding case around the WM8782, since I am going to use it on a motorcycle which has an electromagnetic unfriendly environment (ignition sparking).

Next items to develop/investigate... Raspberry Pi software to read the audio from I2S (which is/was now done with arecord) and process it.

Kind regards,

Dirk

Hello everyone,


I have a high-end Swiss Physics Model 5 preamplifier, which is both beautiful and great-sounding. Despite this, it has a few flaws. Since it’s my favorite preamp, I’d like to learn how to repair it and possibly improve it, as it doesn’t really have a resale market.


First off, if anyone finds the schematics, I’ll buy you a pizza.


Some modules are resin-potted specifically to prevent them from being “copied,” which brings me to my first question. Given that the modules are fairly repetitive, wouldn’t it, in theory, be possible to remove the resin with a dental drill to figure out their composition—assuming someone had nothing better to do in life? I’m just starting out, so I’m curious.


Second question: call it a thesis project, call it something else, but this preamp keeps breaking down. I don’t replace it because it sounds better than anything else. Plus, it’s gorgeous.


One issue—or maybe not an issue, but still...


After the preamp, my signal goes into professional mixing/mastering monitors, specifically the Dynaudio BM 15A. They’re active speakers without a volume knob. The problem is that I end up using the preamp at only 2 or 3% of its volume range.


For normal home listening, I’m basically moving the volume knob by just half a millimeter.


So, I was wondering: could I add a trimmer?
Something to lower the base signal level.


Since this is a high-end system, anything I add could affect the sound, so the more transparent the solution, the better.


And one more thing: since the preamp is so powerful, it has a noticeable background noise. True, lowering the signal strength should also reduce this, but generally speaking, if I’m dealing with slight hums or background noise, where should I focus to make it quieter?

Devore made video about cost of high end:
Login to view embedded media

Some numbers from the video:
-Devore O/baby

Fixed cost: 1500 usd
Retail cost: 5700 usd
Price to dealer: ??

Speculation:
He says he can make 300% normal margin by selling trough online without dealers,
also say that 30% discount from 5700 is still more than normal margin:

-5700 usd about 300% normal margin = 1900 usd
-30& discount still more than normal margin trough dealer = 3990 usd

1900 < ? > 3990
33% < ? > 70%

So around 40-50% margin probably?



-Any thoughts? Is this honest?
-Is this fair margin what you have to have to survive as small scale high end manufacturer?

-I think this is self serving and bit dishonest video since he used as an example the lower margin product in his line..
You will not have much more fixed cost for Devore 0/96 (probably 2000-2500 usd) that is retail $15,900..

JVC "EXU-901a" (KRELL KPS-25sc Phase Tech CT-1 CT1) better than VAM1254, CDpro2LF ?

JVC cd mechanism "EXU-901a" (KRELL KPS-25sc Phase Tech CT-1 (CT1) - better than VAM1254 (CD-pro2M + CDpro2LF) ?

Who know more about the JVC cd transport mechanism "EXU901a" (EXU901 EXU-901).
I don't find datasheets, information about optical pick up (laser unit) and order sources for spare parts so as the complete JVC device.
I know about only two cd player brands, where this mechanism is inside:
1) Krell KPS25SC
2) Phase Tech CT-1
to find by follow URLs:
http://www.audiogon.com/cgi-bin/cls.pl?dgtlplay&1253308225
http://www.phase-tech.com/html_japan/productspage_ct-1.html
www.phase-tech.com/pdf/CT-1_catalog1.pdf

Who know more about this transport unit? Thank you for your advices

I'm shopping for motor run caps for tube amp power supplies. I have a smal collection of the classic aluminium can caps like Bosch MP etc.
That style of caps is no longer made so it seems. New ones have a white plastic case. My question is: are these as good as the old style ones?
I've been looking at datasheets, they say 'metalized polypropylene' and what used to be oil is now polyurethane resin.

https://www.tme.com/html/EN/capacitor-polypropylene-450v-sr-passives/ramka_11117_EN_pelny.html

Thanks!

(edited)

USB transport Reflex allows you to play files with a maximum sampling frequency of 192 kHz and a maximum resolution of 32 bits in PCM, TDM and SPDIF formats. The module supports multi-channel sound, the maximum number of channels is 8 in the 7.1 configuration (Surround Sound). The device is defined as a standard USB audio class 1 sound card, works with computers, tablets, smartphones running Windows, Linux, Android, macOS.
Assembled on the STM32F446RCTx microcontroller.



Possible audio channel configurations


Selecting the audio channel configuration
The audio channel configuration is selected by setting jumpers on contacts 6, 7, 8


SPDIF format
When the SPDIF format is activated, the data is output to the DATA pin of the first I2S port.

Selecting the TDM format
Selecting the TDM format is done by setting the jumper to pin 9. If the jumper is not set, the output format is I2S. If the TDM format is selected, port 1 is active. In the TDM format, the LR signal is a pulse with a duration of one BCLK clock cycle by default. It is possible for LR to be a rectangular signal with a pulse duration of 50%, as for I2S. To do this, connect the DATA pin of the 2nd I2S port to GND.

Sync modes
The following synchronization modes are possible:

• master with internal synchronization
• master with external synchronization
• slave

In the "master with internal synchronization" mode, the quartz resonator installed on the board is used as a source for generating I2S/TDM signals. In the "master with external synchronization" mode, the clock signal with a frequency of 256*Fs is supplied to the "EXT CLK" contact of the module. In the "slave" mode, all clock signals are generated by the external master device.


Changing output states when changing frequency and bit rate
When changing frequency and bit rate in the OS settings, the module changes the states of the outputs: 22 and 23 - for bit rate; 25, 26 and 27 - for frequency. This can be used for external synchronization to select a clock source or for indication.

State of outputs depending on bit rate

Output state depending on frequency

Software update
The module software is updated using the "DfuSe" program, which is available for download on the ST website. To update the software on the module, you need to install a jumper on the "B" contact and connect the "DATA" contact of port 2 to the "GND" contact.

Pin assignments

What you can do.

If it's just plugging into USB (without setting jumpers), then it's a stereo USB/I2S transport with one I2S port. Connect the DAC and enjoy the music.
But Reflex can do more.
You can connect many DSPs and DACs to it. You can load Sigma Studio or Pure Path Studio projects into it. You can connect Buttons, Regulators, LEDs associated with Sigma Studio or Pure Path Studio projects to the GPIO pins. Everything will work!
Follow the headlines ChipDipDAC did it, so can you. I will show examples of how to do it.

KiCad files and firmware are attached so you can do it yourself.

I'm a new member from Europe. Hi-Fi audio has always been my passion. And for the last 10 years, it has also been my professional career. At this moment, I work for a small loudspeaker factory, which leads to the opportunity to do some DIY stuff 🙂

Hello
I am looking for help or suggestions to try to resolve a fault.
The amplifier is a kicker kx 700.5 and the fault symptom is lack of power in the 5 audio outputs.
Check the muting stage, it seems that is not the problem.

Hi guys.

A customer sent me a Gemini RS-410 monitor. After turning on, it makes an awfully loud noise that goes away after a while and then works all right. If it is turned on after a long time, the noise takes longer to disappear, when turning power off and then on again, it is not there. It's there with volume and level controls all the way down. Appears to be temperature dependant, or e-cap (charge) related, or something in this direction. Could also be a cold solder joint.

These monitors are cramped with components and glue and other stuff that makes it almost impossible to work on.

Also, I couldn't find a service manual anywhere.

Does anyone have the same issue with these powered speakers? How did you solve it?

I read that a MOSFET can simulate a Triode to produce even harmonics, said to give a more pleasing listening experience. As a Source Follower a massive 3rd harmonic is produced, but as a Grounded Source the harmonic is the 2nd. Your comments, please.

New Old Stock chips.
CS8412-CP and SM5842APT, one each.
Never been inserted into any sockets before.
That slight outward angle of the legs that every new chips have is the tell-tale sign.
Stored in anti-static tube all these years.
Both chips + shipping registered & tracking = US$68 or SG$88
Paypal FF only please.
50 packages sent to Europe, US, Asia. All received by members without a glitch.
I am not a scammer.

Brand new, unused Cerafine 220uf + 220uf @350V , with clamps provided.
No kinks, no dents, no scratches. Perfect virgins for your tube projects.
Two pieces of caps + shipping registered with tracking = Singapore dollars 330
They will be packed very carefully to avoid blemishes, i can assure you.
Paypal FF only please.
Check my track record of sales here.
All are happy buyers. No Lemons.

Hello greetings from a windy cold flatland.Today i opened the bought DATS v3 casing and looked at the board.My Dats is from 2023 (See picture).It also has the 1000 ohm calibraton resister built in.It has only 1 LMV op amp.



For the reference cal value i took a precision resistor of 100Ω first. Later i used a 4.952Ω closer to minimum impedance speakers. REW shows fairly accurate down to 0.2% 0.4%. Dats shows approx the same values.. REW with dats is somewhat more noisy but averaging should help.Higher sweeplevels -4db help too.

I'll post a comparison of the solo Dats vs REW-with dats later with higher resolution.In Rew's the impedance peaks (Most clearly the left one) shows 1ohm higher then dats is measuring.
Not shure if this is a large value or not.... I'm just starting with this measurements and threw away most cause i need to know if i'm doing it right. I didnt take any free air measurements yet to find T.S parameters.
The purpose is to get familiar with this stuf so i measure the speakers in box now and later on measure the seperate components of the crossover filters and speakers.This is a stereo system with 2.5 way floor standing speakers .nahh .with some 20+ year old elko's witch i want to replace/upgrade. I wil only take comparison measurements using dats vs rew-with-dats.Measuring with same room temperature and speaker placement. so no impedance RIG.. I'll try to get the levels about the same.
I have a modified behringer DEQ2496 ultracurve (parametic eq) where the "grainy sounding " output stage is bypassed.Signal goes out direct via highpass filter and coupling caps.All is done with online help.Im not much of a techy but know some tiny basic stuff.My L-shaped room is far from ideal and plan on doing some accoustic treatment in the future.I listen in a fairly small triangle (speakers and me) to reduce room interfearance and let early reflections not take over.Thats it for now.Time for bed.

I have a pair of vintage 1.5” dome midranges from ADS L780/2 speakers. Now I plan to install them in my car. I’ve been wondering whether these drivers can be used without low-pass filter (high frequency cut). Because I believe car environment is different from home environment and the passive crossovers in these complete speaker units may not work well in a car due to the different mounting location. The tweeters and midranges in factory cabinets are located next to each other. But, when they are installed in the car, with reference to my car’s locations, the tweeters are placed on the dashboard facing up towards the windscreen, while the midranges are located at the front bottom corner of the door, close to the kick panel. So, there’s a distance between two drivers.

I observed my another car, 1990 Volvo 760 Turbo Intercooler, it has full audio system from the factory. The dash speakers are tweeters with passive high-pass filter on them. And the front door speakers are coaxial full-range speakers without passive filters.

So, I just wonder if I could build my audio system similar to the Volvo’s, tweeters on the dash with second-order high-pass filter and 1.5” dome midranges on the front doors with only high-pass, without low-pass.

What I’m afraid is the durability of the midranges as they are now responsible for reproducing high frequency. I’m not sure if it will cause problems such as distortion, over temperature, etc.?

Note: the 1.5” dome midranges are equipped with woven soft dome with sticky coating and they utilize ferrofluid for cooling.

It has been obvious to myself and a lot of other people that we are targets of "focused advertising." Big corporations are gathering data on those of us who search and shop on the web. This is supposed to be anonymous and not detailed to the last degree. It has become obvious over the last few years that far more detail is being gathered, and it is attached to one user or more likely one IP address. A relative works for Ebay as an "information specialist" and she provided me with some info as to how deep they dig into their user base, and to whom they divulge (sell) that info to. It's a bit deeper than you realize. I have purchased a few small items on Ebay in the last few years so there isn't much to find on me there, but one of their partners is Amazon where there is some considerable history. One of the most intrusive details showed up right after I bought a cheap Motorola cell phone on Amazon. Every time I drove past the Mercedes Benz dealership near Pittsburgh I would get a pop up advert for Mercedes cars on that phone. Obviously Big Brother doesn't know what kinds of cars I drive.......

After zero new car purchases since 2008, I bought my wife a new vehicle about 2 months ago. I guess Big Brother got wind of what we were looking at since the new object in our driveway looks EXACTLY like the one pictured in these pop-ups (two on every page) that started appearing on my Start page daily about a week ago. They even got the color right! We paid under $40K out the door. We also looked at about a dozen other vehicles in the same mid sized SUV market and took lowest priced car of the bunch. I still get pop ups for Toyota Highlander vs Hyundai Santa Fe comparisons. We looked at the Highlander, but it was far beyond our budget (about 50K).

Many years ago, I had a Honda Element. I subscribed to the Element Users group forums to learn all about how to fix my Element, but the anomalous behavior I had turned out to be somewhat normal, and the Element was eventually traded in almost 10 years ago. I still get links to the latest posts on that forum via email, and occasionally I read some posts. LAST NIGHT I read a post about changing the starter. THIS MORNING, I get adverts trying to sell me a starter. How can the details of what I'm reading in a forum propagate this quickly, and can it happen here?

I sell 8 kt77 JJ, matches, I don't have the boxes, they are new from Tubestore, 300, or 410 in Canada+ shipping

Cables have been a lifelong hobby until recently after spending 30 years in nuclear and aerospace, when I converted the hobby into a commercial business and fulltime activity. 18 months on and it's going better than expected!

The business model focuses on supplying well-built, durable cables for audiophiles and there is an opportunity to first explore if there can be a synergy and improvement within their own systems when using them. To that end, I offer an 'audition/tryout' service with no obligation to purchase. You only pay a deposit and shipping costs. It's helped a lot of people explore different cabling possibilities and accounts for about 20% of sales. It's the most sensible way of justifying the investment in cables I would suggest.
Purchase prices reflect the time it takes to build a set of cables and include a modest profit margin, making them very competitively priced when compared to others.

CMRR (Common Mode Rejection Ratio) as implemented in cables via a braided geometry is my 'thing' at the moment. It's real, it's very effective at mitigating EMI/RFI and it's measurable. If you're looking to achieve as dark a noisefloor as possible then CMRR-shielded cables can be a good choice.
I don't make any outrageous or unprovable claims that my cables will magically transform your system but 'neutrality' and 'transparency' are the key objectives when it comes to the presentation, almost as though the cables are an invisible component in the signal chain.
My own estimation is that cables will bring a maximum of 10-15% improvement over stock 'bootlace' cables. Anything more than that is psychoacoustics and/or placebo. However, this small margin is critical for some to explore and exploit.

10% discount for diyaudio members from my Etsy platform using the code 'DIYAUDIO'.
I operate 4 selling platforms with Marktplaats.nl and Etsy being the primary outlets.
I thought about a website but I doubt I would be able to keep up with demand. I currently have a 10-14 day build queue with about 45% sales coming from referrals and repeat orders, so that's just about manageable for the moment!

Etsy
Marktplaats.nl

A few examples:

'The Braided One'
Uses 3 braided, isolated conductors per cable to implement CMRR and achieve a dark noisefloor


'The Indigo.Y'
Still utilising 3 isolated, parallel conductor wires is essentially a copy of Chord's discontinued 'Indigo Plus' cable.
Chord's cable had a pricetag with 5 figures, mine has just 3 figures! 😉


'The Core Function[SIGNAL]'
The objective was to develop a cable which sounded epic, but without an epic price-tag. Starting at €30/pair it's probably the best bang-for-buck cable anywhere. Got a very favourable review from Alpha-Audio, here.


'The Crystal Clear Phono'
If you're a TT user then you know all about the need to create head-room in the signal path and having the lowest capacitance cables can facilitate this.
36.2pF/m was about the lowest I could achieve using braided SPC. It's doing well amongst vinyl-junkies!



'The Crystal Clear XLR'
Not the easiest cable to build due to the wire gauge, but the transparency it facilitates is worth the effort.
Strangely popular with Accuphase amplifier owners for some reason.


'The Silkworm' and 'The Mamba' braided powercables.
All the benefits of CMRR via a braided geometry and substantially cheaper than the equivalent AQ braided power cables.



Headphone cables constitute about 30% of the overall build queue at any given time.
There are a few models in the range and again, a braided geometry features heavily.

'The Crystal Cranium v2.0' 16-core braided, hybrid SPC/copper


'The Coffee Cranium' 4-core, braided copper

https://www.youtube.com/@ChicagoHornLoudspeakerGuys

Discussions on vintage commercial and DIY horn loaded speaker systems. How to achieve the best sound quality out of them.

Hi everyone,

As I think about building some bass towers next year I am reminiscing about some of the most exceptional speakers I ever heard, the Snell A/III. Among the attributes I was impressed by was the bass performance. Just outstanding feeling that the speakers were pushing all the air in the room at once. I believe it's also called coupling to the room. As I recall, these speakers had a smallish 4" mid and tweeter in a hemispherical baffle in a separate enclosure which sat on top of an over large bass cabinet. There was quite a bit of distance between the mid and the floor slot.

Another attribute which I learned later was that the woofer was mass loaded. Sadly a reason so many have now failed.

On the negative side, these speakers were very sensitive to the amps used and as I recalled seemed to really not perform well at all with tubes.

My question for any who remember, or want to opine, is what made those speakers perform so well in the bass range?? Was it the floor firing woofer? The mass? something else?

I'm thinking about this in large part because I would love to recreate that experience, but not necessarily the design choices. Maybe knowing how Snell achieved the performance will help me put some of that into my next build.

Thank you,

Erik

Hey, I am very new to repairing electronics with capacitors as well as using a multimeter so I need so help/guidance fixing this issue

recently I was handed a family members old NAD C320BEE that they had in their loft for storage for a few years but when powered on it went into power safety mode due to a problem on the protection circuit somewhere before it went into storage I was told it worked fine I found lots of people online talking about how the capasitors are known to be pretty bad in this model and can cause this issue the only problem being I don't have much experience sorting this so I thought it might be a good opportunity to learn I found the online schematics for the board although I don't really know how to read them and tried checking some power levels around the board the power from the power supply mini board seemed correct but the +18 and -18 rail seemed to be incorrect from what I could figure out so I tried to check the capacitance of a lot of the capasitors on the board and I'm not too sure if I've found the issue or not the following capacitors stood out more than the rest but lots were slightly out of spec so I'm not sure how to take it if they need replacing or not:

C431 0.75mf instead of 10uf
C430 13.36mf instead of 10uf
C12 111.4uf instead of 47uf
C326 and c325 0.46mf instead of 47uf
C421 580mf instead of 4.7uf
C49 104900uf instead of 47uf
C47 860uf instead of 47uf
C413 and c414 350uf instead of 220uf
C48 1280mf instead of 47uf
C410 170uf instead of 47uf
C342 250uf instead of 47uf
C425 1.8mf instead of 2.2mf
C436 is 0.25mf instead of 100uf
C324 is 10.74mf instead of 47uf
C323 is 450uf instead of 47uf
C44 is 610uf instead of 220uf
C43 is 1.43mf instead of 220uf
C26 is OL on max settings should be 220uf
C25 is 18.4mf instead of 220uf
C15 is 13.6mf instead of 220uf
C16 is 14.26mf instead of 220uf

as I said I'm not experienced at all with this type of stuff so would be really appreciated for any guidance that you could give I'm not sure how big of an error of margin I should give c410 doesn't seem too far off but its still 3x over what it should be so should it be replaced or is it an issue further back down the line that is making this capacitor seem broken I'm not sure and don't really know where I should go

TL;DR
I need help fixing a NAD C320BEE but don't have much experience with electronics and don't know what to do

I tried to attach PDF files to a post which according to your help site should work but it does not at all, no issues with jpg

Hi,

I have two sets of

Matched JFETs Toshiba 2SK170/2SJ74 Quad 8-11mA x 1 $45.00

I offer one set for 35 Euros plus shipping from Germany, or both for 65 plus shipping.

Hi, I am selling some stuff that I never got to use!

For convenience and transparency, I am listing exactly what I have and the original DIYaudio price.
I am located in Germany, Berlin.

Here, I have two sets for the Burning Amp.

Burning Amplifier Complementary Bias and Output Set for the BA-2 & BA-3 x 2 $29.00 each
Burning Amplifier Gain Stage No Feedback for BA-3 x 2 $19.00 each
F-4/BA-3 Transistor Kit F-4/BA-3 Transistor Kit x 2 $59.00 each
Total 2 Kits 107 USD each without taxes originally.

I am offering one set for 80 Euros or both for 150 Euros plus shipping from Germany.
If you take both, I will also include some random transistors and caps that I marked as belonging to the BA, as you can see on the picture.

Hi everyone,
I'm part of a non-profit org in L.A. and we have a pair of Mackie Thump 15 powered speakers. Usually we use them in our hall, but sometimes for outdoor events. When in our hall we're getting a buzz. Sometimes loud, sometimes not, sometimes it goes away. But it's behavior isn't connected to any particular action. In other words, we can switch mixers, turn the fluorescent lights off, plug them into the same circuit or separate circuits, but still can't figure it out. The AC wiring to our electrical outlets are in a conduit that runs along the wall, and again - fluorescent lights. I know those might be the problem. But here's something I'm curious about after reading snippets of info by googling. Is it actually that the mic is picking up vibration from the AC current or from the ballasts in the lights? Some answers from the google searches said that the equipment was very sensitive and wiring and also fluorescent lights could create vibration that the equipment picks up. Does that mean that the mic is picking it up? Could I test it by standing with the mic outside the hall with the mic to see if it goes away? Or maybe try just unplugging the mic?
And, if that's the case, whether its the mic or electrical noise through the AC source, is it possible that I could build a little box around all of the conduit with some type of shielding inside? In other words I could just use some 1 x 2 and dry wall to box the conduit in, but is there some kind of shielding material I could stuff in there that would help mitigate the buzzing? It would be a lot of work but I would do it if it was a sure thing. Already I'm thinking of switching out the fluorescent lights for LEDs or incandescent. Big undertakings but if I were confident that either or both jobs would solve the buzzing, I'd do it.
Any words of wisdom would be really appreciated.
Thanks,
Scott

Hello all!

Thank you to the administrators for approving my account.

I live in sunny Singapore and recently got into DIY audio. I've recapped a Technics RS1500 tape deck, built a Bottlehead Moreplay tube preamp and most recently completed GR Research NX-Studios and Double Troubles.

My current hyper-fixation is Voigt pipes and I would appreciate very much if anyone can link me to learning material 🙂

Cheers!
Timothy

Hi all - RA here. Looking to start my journey with DIY amplifiers.
It started by hearing a Blue Hawaii SE on some electrostats that got me addicted to the sound. I now have some Stax SR-009s (driven by a SRM-D50).
I'm looking to start small, and maybe in a few years time build my own Blue Hawaii SE or DIY T2.

An OPT is a big investment. Question is how it would work from 350Hz and up for my horns, with an MOFO buffer for my favourite 6B4G?

This topology will output 4W at ca 1,6V input(clipping at 6W 5%), while the 4P1L has 0,7Vrms sensitivity.

It seems quite simple using a depletion mode MOSFET and a low budget Hammond choke, when we don´t have to deal with the lowest octaves.

Off course this is just brainstorming from a cheapskate, will anyway try it with an amp for 6.5kHz and up with stuff from the parts bin.

Hi

Some months ago, I finally got my hands on an Audio Research Mkiii and I want to learn as much as possible about it. Its a fully balanced linestage, uses 8x 6922 tubes. Since it may need new tubes, I have some questions.

1. When buying tubes, how should they be matched, in pairs or quads?
2. What is the purpose of RV2 and RV3 in the schematic
3. Which current production tubes? I'm currently looking at Gold Lion 6922's of Tubedepot, or perhaps some Phillips ECG 6922's off an ebay seller. Due to the high number of tubes, I'm not really interested in NOS tubes like NOS Amperex 7308 (even if they're the best sounding), my upper budget would be the Gold Lion 6922's , they're currently about $75 each.

Attaching circuit diagram if it helps answering the tube matching question, hopefully its clear enough.
Looking at the schematic, looks like this is for a single channel?
Any ideas on what Q1-Q5 do? They're color coded with paint by ARC.

Thanks in advance!

Hi, OJG.

Thank you for sharing! I successfully replicated your ThatMicPre project and made a few minor modifications:

1. Slightly reduced the width of the mainboard to fit aluminum cases available on China's Taobao.
2. Adjusted the panel layout height and the positions and sizes of the mounting holes to match the aluminum cases on Taobao.

For those interested, you can check out my fork.

I also replaced the NC3MBH1 with the NCJ6FA-H, which is compatible with both 6.35mm TRS and XLR connectors. However, I haven’t fabricated or tested the new mainboard. Anyone interested can fabricate it directly; I believe the design should work without issues.

During the process, I used That1512 and SSM2019 chips purchased from Taobao, but neither produced an output. Finally, I bought a second-hand AD620 from Taobao for just $0.6, with free shipping, and it worked as expected.

I hope my experience can help others avoid unnecessary trouble.

Hello,

I bought these aluminium cast horns for few money.
I searched for a long time but couldn't find the brand and the model.

they are 51 cm width , 23 cm high , and 35 cm long,
the diameter of the throat is 20.5mm,
the frequency seems to be around 800hz

The angle is 90°,

weight is 6.6kg

the thread is 1' 3/8.

They are near altec 811b or rcf 4823....

Could someone please identify them?

Hi,
I need a well designed high quality class AB power amplifier using A1941 and C5198 at the Output stage.
I'm reading Douglas Self's Book and i noticed that designing a good amp needs a lot of calculations and lots of knowledge
can you help me to do this? or any designed circuit diagram?

Hello folks:
I was looking at my tube collection and checking out the the discussion about the 38he7 being able to be split into an 15Watt pentode and a diode. If the heater is wired to pin 10 and 12 the diode is left blank. The pentode goes from 10W dissipation to perhaps 15W or so. So I decided to check out the 33gy7a and found a similar thing. This time it is pin7 and pin12 to lite the pentode and I think the pentode is a 6ge5 which is a 6dq6b @ 18W. This tube may be a bit of a sleeper. More research must be done to confirm.
Interesting possibility when tube still only costs a buck or two.

Hello everyone. My name is Sina I'm from Iran. I love Electronics specially Audio Circuits; and i want to communicate

I'm looking to piece together a Linkwitz Orion or Phoenix clone and found the GR M-165NQ driver designed for dipole operation. It's a bit smaller than the SEAS driver in the reference design, so I'll probably use two in MTM arrangement.

Anyone have experience using this driver in dipole applications?

Hello Guys!


I'm restoring a luxman CL34 preamp and i d like your advice for the parts, here's the beast:



It is working properly but could benefit from a little refresh and improvements 🙂


I d like to start by the power supply




Bridge rectifier :





It is 1N4007 diode so i will swamp em for UF4007, i think it's an easy choice; less noise & faster


To continue with diodes, here's a bunch of WZ-310 above the second quad of 1N4007. Seems to dissipate a lot of heat !





I was thinking of BZX79C30T50A for the replacement and to mount them 5mm above the pcb to help with the heat. Any better choice or tips here?


Let's continue with the caps





The big caps are 47uf+47uf 450V (35x50mm) here i could swap them for Mundorf Mlytic HV 50uf+50uf 500V


The others are 33, 47, 100 & 220uf from 16 to 350v, i guess i could use any good 105° power supply caps from panasonic or nichicon




The preamp board :




Most of the time i would not change that kind of film caps but the 15k ohms 2w ressitor on the right of the 1uf gray cap started to toast it so... let's swamp 'em all!


Here's the point is to find a good quality replacement for the 0.47uf 250v and 1uf 400v caps with quality and size in mind. PIO, Wax,... are too big.


I could change the resistor by a 15k 3w and take it away a little bit from the caps. Any suggestion here too?


Thanks for your help


Edit: bigger pictures

Hello,
What I am trying to accomplish is to be able to look at a phono preamp (either solid state or vacuum tube) and understand how the RIAA EQ was accomplished, either with negative feedback or passive design. To be able to understand how the combinations of resistance and capacitance were chosen to form the necessary curve. My goal is to understand enough to be able to change the RIAA playback EQ to NAB/IEC tape playback curve in a preamp. I would appreciate any suggestions of texts that explain how RIAA EQ is calculated.
Thank you!

Having made some electrostatic panels for headphones in the past I thought well lets play, when I saw a pair of Audiostatic ES300R for 100 Euro claiming they needed new panels just 70km from my flat in Hamburg Germany.

When I got home I stupidly plugged them both in and the second one went BANG and set of the circuit breaker in my flat, so I left that speaker to later. The other played but clicked, around about once a minute. After a little while I realized the clicks where coming from the moving coil speaker on the other channel. Removing the electrostatic speaker connections from the amp removed the sound. So some how the Electrostatic speaker is making sound in the moving coil speakers through the Quad 306.

I guess this could be due to the mains voltage direct coupled and arcing (hopefully not in the output trafos), so upsetting the Quad 306 Amp? Any suggestions or tests I can do are very welcome.

I found some circuits here in DIY audio that are approximately correct for the Audiostatic ES300R (Resistor and capacitor values differ in the driver circuit but the voltage doubler is identical). Thanks to bolserst for the doubler circuit and maudio for the Audiostatic ES300 driver circuit that I stole from other threads in DIY audio and attached here.

The speaker that went BANG had a loose connection on the mains wire, and has obviously done it before, and taken out quiet a few of the PCB tracks near the mains connection. This still was not a killer but, sadly on further investigating one of the 2 stator driving transformers is open circuit on its secondary.

So although the Audiostatic ES300R pannel and stators looks rather well made I am not so impressed with the electro mechanical side. No detachable mains lead, no issolation transformer for the high voltages, no fuses, no screw terminals on the PCB, not even PCB pins, just bare mains wires soldered into the PCB.

As for the sound from the one working speaker the sensitivity is much lower than a Quad ESL63. Subjectively it has a similar quality to the highs and midrange, posibly even better, but I am guessing here but 300Hz and lower seem non existent. The capcitors in the Audiostatic ES300R mention west germany so I suspect they would need to be replaced.

So I probably will give up renewing the speakers, unless some one can sell me a driver transformer cheap, and instead I will sell the transformers cheap.

Happy DIY'ing

Owen

Hi ! i read often about phase in speakers and the importance that a speaker is phase coeherent, expecially for a good rendering of the virtual soundstage that i love immensely
I have at hand a Dayton Audio DATS v3 Is it possible to check in some way if a speaker is phase coeherent with this device ?
could you please link a phase graph of a speaker that is indeed phase coeherent and i can keep as reference please ?
this phase issue has always been difficult for me to understand
Thank you very much
Have a nice day before it is too late

Hello from Australia.

I am stupid and unknowledgable in regards to audio but very keen to get started building my own speakers.

Wish me luck, and I hope to become acquainted with you all!

- Cell

Hello, I am Thomas from Bavaria in Germany. I like to repair all sorts of broken things and am interested in DIY audio equipment. Not being very proficient, I still managed to make simple things like a mic preamp or a stereo mixer. I would like to make my own valve headphone amp one day! I am here for inspirations and of course help, which I appreciate very much in my projects.

Good morning all,

I joined the forum after finding the 'Lusya/audiophonics ADC for Pi' post from DRONE7 , the post helped me a lot in getting my AudioPhonics WM8782 I2S ADC going on a Raspberry P4 board but there were some modifications needed to the post due to changes in Raspberry Pi releases/kernels. So here I am adding the changes for other tinkerers to find.

I am not into human audible audio, I will be using my setup for DSP of knock sensor on a turbofied motorcycle engine which will allow me (hopefully) to keep my engine in one piece during ECU parameter tweeking.

Hi everybody,

I've been occasionally reading through the forum over the years, I am into Audio DIY since a couple of years, mostly synth diy spiced up with the one or the other outboard gear project like compressors or EQs. I've a small cosy home studio with mostly vintage and DIY instruments and it's always slowly growing 🙂
Recently one of my Event Opal speakers developed a problem and I hope to find some help / information in one of the active threads, so I finally pushed the button and registered 🙂

Cheers,
Andreas