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Forte Model 3 - One channel running hot

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Hello Forum -

I have enjoyed my Forte Model 3 running it into a pair of Magnepan 1.6 with great results. The amp was left one night and since then I noticed the left channel running quite hot (hot to the touch) compared to the right. I have not used it since with the exception to get you theses images. I have not messed with the bias but it looks like that adjustment has slipped (? as visually inspected by the amateur). I have not been able to find what test point I should use to compare the two sides to decide if a bias adjustment is the next step. Note, the IR images is after 5 seconds on time. Any help would be greatly appreciated.


Okto DAC8 - a high-performance, 8-channel D/A module with Sabre ES9028PRO

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Hello everyone,

It's our pleasure to present to you OKTO DAC8, an 8-channel D/A module. Based on numerous prototypes with different I/V stages and many, many hours of PCB design and testing with R&S UPV audio analyzer, DAC8 is made to excel as a part of multi-way or multi-channel reproduction chain.

All the features, measurements and documentation can be found on our website: Okto Research.

Key features:
  • Up to 384kHz I2S, DSD256, DSD128 over PCM or 96kHz SPDIF input
  • Measured -124dB (0.000063%) THD / -117dB (0.00013%) THD+N analog output performance
  • 8 analog stages with fully differential OPA1632 op amps, analog outputs accessible on onboard XLR connectors and 0.1" pin headers
  • Access to ES9028PRO's 32-bit internal volume control with a potentiometer thanks to onboard MCU, direct access to control registers via I2C
  • Headphone amplifier capable of delivering 100 mW into 32 ohm load with -116dB (0.00016%) THD
  • Highly optimized 4-layer PCB design with attention to RFI immunity (cannot stress out enough how small loop area and short traces are important for clean output spectrum without high-order spuriae!)
Typical applications:
  • As a part of hardware-based digital crossover together with MiniSHARC, FreeDSP or any other DSP module with digital (I2S or DSD) output
  • As a part of software-based digital crossover like 64-bit DSP engine in JRiver and any USB>I2S module like USBStreamer or DIYinhk Multichannel
  • As a standalone 8-channel DAC for multichannel playback paired with any of the aforementioned USB>I2S module
  • As a stereo preamplifier and/or headphone amplifier future-proofed by 6 extra channels
Available for 489 EUR. OKTO DAC8 only comes fully assembled and tested due to high number of difficult-to-solder parts.

Franta Blazek,
Okto Research

Attached Images
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File Type: png OktoDAC8_bal_thd_1kHz_1.png (27.5 KB)

FS: assembles ACA amp

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Hi,

I have the ACA I made a couple of years ago. I used it in a small audio setup, but hardly listen to it anymore. Decided to let it go. This is one of the first generations of the Amp but with the bias mod. Also it has a proper Zen9 like psu powered by CRC and Antek transformer in the front. Our all mighty ZM suggested me this configuration and that improved the sound for sure.
Heatsinks are barely warm in this case. Now it has an aluminum black face plate, but if you like I can make you a solid wood one if it fits your interior better :-). As usually all mi soldering is nice and clean (kinda have OCD for that one :-).

Price is $280 shipped which barely covers parts cost.

I can send more pictures upon request. Thanks for looking.
Attachment 701562Attachment 701563

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File Type: jpg IMG_0254.JPG (85.7 KB)

36 volt amp.

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I have a been using a Sure 2050 4x100 on 24volts (SLA) to drive my mobile PA but now have a big block of 36volt lithium to use and I am now searching for an amp that can get the most out of that voltage.

The TDA7498E will do 2x160watts @ 4ohms on 36volts, anything else I should consider for driving four or 10" PA drivers.

Multiple PCM2704 sound cards

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Hi,

For a CBT array I was thinking of using 8 of PCM2704 to produce 8 stereo channels, Raspberry as software DSP for delay and Eq.

If multiple PCM2704s are used over USB bus for multi channel sound output, will the output from all of them track each other closely enough for such an application where constant inter output delay (whatever is configured by software) is to be maintained at all times?

Thanks and Regards,
WA

Mini-amp for Output Tube Distortion

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We have had a few threads where the OP's were interested in output tube distortion, one asked it to go to a pedal. So how about a pair of 6AK6's driven by a 12AX7? Using a 12V laptop supply which supplies the heaters and a step up module for the high voltage. The speaker can just be a resistor and a pot used to send the signal to the next device. Should have no preamp distortion. Or maybe should go Paraphase? Hmmm...



Click the image to open in full size.

X-Formers out of Phase?

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Hi all,

I'm in the process of re-doing a mid sixties Japanese amp and found some curious wiring. If the photos are readable, note the OP transformers have the red and green wires from the same B+/Plate connections. The wires however go to different connections. One green and one red to filter cap and the others to the tube plates? Out of phase? Things look factory and untouched until now.

Thanks

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File Type: jpg Sockets.jpg (102.7 KB)

Paul carmody Tarkus mods

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Id like to build the tarkus but WAF is a big issue with the 10inch woofer. So instead, i was wondering if i could use 2 or 3 more 830657 in parallel for woofer duty so that i can keep the enclosure narrow

Tarkus - undefinition

Woodside Sc27 volume pot

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Hi
My preamp has been rebulid by former owner.
Balance removed, and stepped attenuator for volume mounted instead of the original.
Anyone have schematics of the original volume/balance circuit. The diagrams on the web has not this part in details.
Br Morten

Tube Amp Preheat broke

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Howdy.

My amp used to have a preheat function but it recently died, should I be concerned, 300B aren't cheap?
The amp was automatically switch on by my streamer and a 12v relay.

Before:
  1. Turn on.
  2. Red light on.
  3. Wait ~20 seconds.
  4. Hear relay click.
  5. Green light on.
  6. Audio pass through.

Now:
  1. Turn on.
  2. Orange (Red and Green) light on.
  3. Audio pass through.

Schematic attached.

The TC4011BP seems to be the controller for it (I'm a schematic novice).
I'm assuming it warms up the tubes w/ 1 circuit and then switches the high voltage circuit on.

But now the high voltage circuit is on all the time so it doesn't matter if music is playing or not.

What would you do?

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File Type: jpg 300C.jpg (754.8 KB)

Thoughts on these Tubes

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Bought a 60's tube amplifier and it came with the following tubes.

What would be keepers and what would you replace?

It sounds fantastic as is but just want to know what members thoughts are?

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Another question about coupling capacitors

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I'd like to put this out to the forum:

Is there any advantage in bypassing film coupling caps with smaller values like .1 and .01 uF? Or is that just for electrolytics?

For example, if I have a 2.2 uF CBB film capacitor in the signal path between preamp stages, I would add both a .1 and a .01 film capacitor in parallel.

And with PSU filter electrolytics, is there any advantage to bypassing with both .1 and .01?

There's a lot of differing information on the internet, and I'd like to clear this up so I know that I haven't been doing it wrong all these years.

Thanks in advance.

modify TDA7057Q amp lepai lp-a2

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Hi there I am being weird and modifying a much more obscure amplifier that actually cost me a grand total of £7! It was the cheapest amplifier I could find at the time straight from china on ebay and uses the TDA7057Q pdf, TDA7057Q description, TDA7057Q datasheets, TDA7057Q view ::: ALLDATASHEET :::! Thing is I actually really love the sound from this little amplifier I used it for many years. After buying "upgrades" and usually being disappointed I decided maybe I should just modify this little badass for more clarity.

I took some photos of it's internals:
Prime Photos

I don't really know anything about this stuff but I did modify the Lepai 2020 and it sounded great! Only time I retired my LP-A2.

I assume I need to get a higher rated cap for the power input, maybe 10000uf?
If it can fit in the box.

And replace the other capacitors with similar values but high quality film caps?

I assume making these changes will improve the sound. I guess I could look into disconnecting the EQ from the circuit to?

Either that or I buy a couple TDA7057Q's and just build something from scratch, whatever I do I will do it as cheaply as possible.

Any advice will be much appreciated! :) Would buying a better transformer and making an external power supply help?

Dom.

Sanity check 400VA 4x18VAC Toroid

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My confidence is waning, so wanted to ask the forum.

I have ordered two PSU boards from Tea-bag, and the plan is to feed the boards from a single 400VA toroid from toroidy.pl, custom-wound with 4x18V secondaries.

No centertaps, just 0-18V x 4.

The SissySIT(Class A) uses a symmetric supply +/-25VDC at a Iq=~2Amps and i'm building a stereo pair.

I feel this should work, but wanted a second opinion before i order this custom tranny.

If i understand Nelson's writings here he seems to recommend that for a Class A amp i should go with 3 times the draw, meaning 600VA. That is one big mother of a toroid...

Think i'll go with a single 600VA toroid with quad 18V secondaries.

Not All Balanced Inputs Are Created Equally

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Interesting study and simulations of a power amplifier with 'pseudo' balanced line input.
Quote:

On closer examination the balanced input is implemented in identical fashion to a typical opamp difference amplifier with some modifications due to the power output stage and the requirements of stability. In essence by well matching the resistors at the input and feedback stage one should be able to achieve a fairly good common mode rejection at low frequencies although how well this performs at higher frequencies will be entirely limited by the component matching, loop gain and frequency response of the output stage which probably won’t be anywhere near as good as implementing it with a differential amplifier using a wideband opamp that feeds directly into a proceeding power amplifier stage.
Attachment 701698
http://www.analog-precision.com/Down...ed_equally.pdf


Dan.

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FS: F6 Boards with two pairs of R125 Semisouths

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I have the following critical components for making PassDIY F6 clone amplifier for sale.

1. Semisouth R125's 2 pairs.
2. F6 Amp boards pair.
3. Rectifier board.
4. PSU Board.
5. Jensen daughter boards pair.
6. Daughter board mounting hardware.

I bought it along with a friend from the Tea-bag's group buy but it's a shame that I won't be able to take up the build anytime soon. Hence thought of offloading it to someone who can make use of these gems.

Asking for $220 + shipping from India.

16x Digital interpolation filter - drive PCM56, PCM58, AD1865 and so on up to 768 kHz

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Hello

In the last month I've been working on a custom digital interpolation FIR filter implemented on an FPGA (Spartan-6 XC6SLX9). This filter does pretty much the same thing as well known DF1706, SM5847, PMD100, and so on. However, its ability to reconstruct and attenuate a signal is way beyond those :)

The filter contains 8192 coefficients and it interpolates the data by a factor of 16 times (e.g. 44.1 kHz to 705.6 kHz and 48 kHz to 768 kHz). It has several FIFOs built in since its core is running asynchrounsly (at 225 MHz) while the MCLK is used only to clock data out of FIFO and to create LE (latch enable) signal for the DAC.
In fact, this filter does always interpolate to 705.6 kHz and 768 kHz (an integer factor of the input) and accepts data up to 768 kHz / 32 bits. In order to do that it works like a sample rate converter, so it interpolates and decimates at the same time. However, it should be noted that for 44.1 kHz and 48 kHz it does not decimate at all, but for anything higher than 44.1 kHz and 48 kHz it still interpolates data 16 times, so e.g. having 96 kHz input means that the data is interpolated up to 1536 kHz, but decimated back to 768 kHz. The same goes to 768 kHz input which is interpolated to 12.288 MHz (in a mathematical sense of course) and decimated back to 768 kHz.

It should be noted that along with huge amount of coefficients (8192) this filter incorporates multiply-accumulate units of 32x35 bits wide. It means that the input data word is fully accepted up to 32 bits (without any truncation for that matter) and coefficients are quantized on 35 bits resulting in unmatchable accuracy of the math it does to calculate the output sample ;)

In the title of this thread I did mention you can drive PCM56, PCM58, PCM63, AD1862, AD1865 and so on up to 768 kHz. How is that possible? The filter contains a self reconfiguratable oscillator which sets its frequency depending on the output length (16, 18, 20 or 24 bits). It means that the output bit clock (CLK) is running fully asynchronously from the LE (latch enable) signal which is generated by dividing the provided MCLK signal, so data is latched in almost all DACs without any extra jitter introduced by the oscillator itself. In fact, the filter contains 3 sets of FIFOs (6 FIFOs for both channels) - one in its input (I2S) before going to the core, another one on the core output and the last one for the final oscillator to clock data into the DAC. Besides all of that this technique introduces a quiet zone after the latch signal going down, so it should give DAC some time to settle down with its output before clocking in another sample ;)

Depending on the word length the following frequencies are created on the CLK output:

16 bits - 14 MHz
18 bits - 15.5 MHz
20 bits - 17 MHz
24 bits - 20 MHz

It means that certain DACs such as PCM56 and AD1865 will be running at the edge with 768 kHz stream, but they will work just fine according to my tests. The LE (latch enable) signal is always 705.6 kHz or 768 kHz depending on the input (either multiply of 44.1 kHz or 48 kHz).

Below are bunch of measurements using a PCM58:

18 bits:

Click the image to open in full size.

0 dBFS @ 20 kHz:

Click the image to open in full size.

-60 dBFS @ 1 kHz:

Click the image to open in full size.

Jitter test @ 48 kHz with LSB toggled @ 250 Hz:

Click the image to open in full size.

Filter attenuation - white noise @ 48 kHz:

Click the image to open in full size.

Filter attenuation - white noise @ 44.1 kHz:

Click the image to open in full size.

All measurements were performed using PCM58.

The filter has I2S input with signals of MCLK, BCLK, LRCK and DATA. However, MCLK can be fed by the same source as BCLK signal (if no MCLK is available) assuming that BCLK has a rate of 32x, 64x, 96x, 128x Fs or similar since the filter has to determine its frequency to know how to divide it in order to create LE (latch enable) signal. Any exotic values of BCLK rate will not work as MCLK, so keep that in mind. The jitter and synchronization of FIFOs depends purely on the MCLK signal, so in the long term it needs to be synchronous with LRCK (in almost all cases it is, since BCLK and LRCK should be derived from a divided MCLK clock).

Following frequencies are supported as MCLK:

49.152 MHz
45.1584 MHz
36.864 MHz *
33.8688 MHz *
24.576 MHz
22.5792 MHz
18.432 MHz *
16.9344 MHz *
12.288 MHz
11.2896 MHz
9.2160 MHz *
8.4672 MHz *
6.144 MHz
5.6448 MHz
4.608 MHz *
4.2336 MHz *
3.072 MHz
2.8224 MHz
1.536 MHz
1.4112 MHz

Those with asterisk are usually used in CD-Players and it should be possible to use the filter within a CD-Player once input is attenuated using 170 Ohm or so resistor per line (in order not to damage the FPGA and its I/O pins due to 5V logic levels). The filter can be powered by an external supply (5V or higher) or by providing a direct 3.3V power supply (it is up the user).

Following outputs are provided by the filter:

CLK - clock for data
LE - latch enable signal
SD_L - serial data for the left channel (and its inversion with the line above it, so you can create a differential DAC for XLR outputs)
SD_R - serial data for the right channel (and its inversion with the line above it, so you can create a differential DAC for XLR outputs)
3.3V - main power supply
GND - ground

It should be noted that the filter does have a TDPF (triangular probability density function) dithering algorithm as well. It can be turned on or off by a jumper depending on your preferences.

The price for a fully assembled and ready to use board will be around 125 EUR, however, it's yet to be determined. That's for a second run since the first batch is used for prototyping only and the new one does include a few minor changes. Photos of an assembled prototype board are included below:

Click the image to open in full size.

Click the image to open in full size.

It can be directly powered by an external USB to I2S converter, but it draws up to about 300 mA of extra current, so keep that in mind:

Click the image to open in full size.

It is a 4-layer PCB. Dimensions are 50 mm x 47 mm.

This thread is for a technical discussion only and to check whether there is an interest in such project. Anyway, if there are any questions feel free to ask :)

Help with distortion

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I’m still having trouble.

Hertz Mille Pro 6x9 front door. Hertz 6.5 rear doors, hertz 3 inch component center channel, Hertz Mille Legend A pillar tweeters, Alpine Type R Thin 12 inch dual subs.

75 watt RMS to the speakers and 1200 watts to the sub.

The sub sounds perfect.

The speakers start to develop distortion over 4000hz at high volume. The gains are set with an oscilloscope and level matched with a SMD CC1. The entire system is EQ adjusted with an RTA.

I’m scratching my head why there is some distortion entering the mix. Is it the speakers? They are high end Hertz. Should I have done Focal? The sound is very clear and tonality is very bright. Is it my 8 channel JL Audio XD800 amp? Should I return it for two JL hd600 4 channel amps? I have the wire capacity to run either.

I just don’t want to keep changing things and not see an improvement.

Thanks

need some help- redplate and cooked pcb

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hi,

i have forever had an issue with fuses popping, but i let a friend borrow it, after many hours of testing without any other issues.

he gave it back to me with a failed channel and he said he didnt notice anything was wrong until it quit working, which was after the capacitor desoldered itself from the board and fell onto the cabinet.

i flipped the amp over to see a 560 ohm resistor had cooked the pcb, and the large cap next to it had bulged and desoldered itself. i replaced the cap(1500@63 volt) and the resistor measured poorly so i replaced it too.

i flipped the amp power on, i ended up with the same channel going red plate.

after turning it off, the capacitor was very hot, along with the resistor, which was not as hot.

can someone please help me troubleshoot this?

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FS: PEAK DCA55 + SOT23 test jig

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DCA55 Still in perfect working order, with a few light marks. Also included is a SOT23 test jig for use with breadboard etc.

Asking for £30 including postage UK. Overseas please enquire for shipping rates.

Attached Images
File Type: jpg _DSC0004.JPG (410.3 KB)
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