Building the ultimate NOS DAC using TDA1541A

@ecdesigns

Hi John,
In order not to spam here and since it was originally started by yourself, if you don't mind I will open a new thread - let's say 3.5 mm connectors or something of a kind where we can discuss particularly that kind of interconnection cables and will quote your and other people's opinions about it. Could be interesting for the community - we will see.
Cheers,
Ignat
 
Hi formatcd3,

I am new to this thread and am wondering if you have a kit version of your dac

Sorry for the very late reply, welcome to this thread.

There are no kit versions available.


Could you also share the schematics of your latest dac version.

I added a block diagram of the Mosaic UV (USB + Volume control). We returned to the USB DAC concept after successfully completing discrete ultra low jitter magnetic isolators. Now we can eliminate ground loops and the masterclock can be placed in the DAC again.

Like all other DACs, this one is based on imperfect parts and circuits. So some source interference and jitter will come through, regardless of design. This means that we have some degree of source and interlink dependency. Then it helps to use cleanest possible USB digital audio source and suitable USB interlinks.


These DACs are designed for high performance, and this high performance can only be obtained when all audio components are performing at a similar level (weakest link).


The USB receiver is based on an XMOS processor. The receiver is self powered (clean, isolated linear power supply) so USB bus power supply pollution is not that critical annymore. Because the Mosaic UV does not use the USB bus supply, one can also use a smartphone for streaming.


There are 3 low jitter master clocks running on super clean power supplies, 26 MHz system clock, 22.5782 and 24.576 MHz audio masterclocks. The receiver supports UAC1/2, 192/24 and provides bit-perfect test. It outputs I2S-plus, a modified I2S protocol as I2S is unsuitable for use in this high performance converter.

BCK and WS signals generated by the XMOS still contain jitter so these are synchronously reclocked with the audio clock.

PDATA is not reclocked yet in order to keep the audio clocks as clean as possible.

The I2S-plus signals then enter discrete, ultra low jitter magnetic isolators. Unlike chip isolators, these magnetic isolators are passive and specifically designed to offer extremely low jitter. This is obtained by using selected core materials and specific winding schemes.

These provide -essential- galvanical insulation between both source and DAC. Without this isolation barrier we would couple interference (hum, noise) directly into the audio set by means of a ground loop. When the background isn’t ghostly quiet we will start to loose micro information fast as it will get masked by the ground loop interference.

I2S-plus then enters a I2S-plus to multi-stream decoder. It translates binary to ternary and provides 87.5% I2S-plus interference reduction by means of flash-loading of data into the D/A converter core. It also synchronously reclocks PDATA, now all interface signals will have low jitter.

The multi-stream signals and data burst clock then enter blocker circuits that “pull the plug” and reduce source interference to almost zero after flash-loading of data. This results in very high SN ratio, less masking, more detail.


The actual multibit D/A converter is based on ternary (3-level) logic. It switches between -2V, 0V, and +2V. The ternary logic drives a common rail resistor matrix that could be compared with cascaded Wheatstone bridges. The “Wheatstone bridge” load resistors are all placed in series and form a common rail. The sum of the voltages across these load resistors will produce the output voltage (4Vpp). Because the converter is based on ternary logic, the output is DC-coupled so we don’t need coupling caps. So for clarity, this is not a conventional R2R ladder DAC, it doesn’t even run on binary logic and obviously doesn’t have R2R resistor ratio.

Ternary logic offers a number of advantages over binary logic. Major advantage is that undifided states (where all register outputs are still changing from one logical state to the next) revert to zero volts. This will “mute” glitches, so unlike conventional R2R ladder DACs we have transistions without glitches. The ternary logic also reduces bit errors and output impedance by using “parallel drive” for each bit.


The Mosaic D/A converter outputs enter a relay input selector that also offers 3 AUX inputs. Relay volume control provides 64dB attenuation with 1dB steps. Integrating the volume control in the Mosaic DAC eliminates degrading of one pair of audio interlinks.

The relays and indicator LEDs are controlled by a system controller that communicates with the relay circuit through an isolated RS232 interface (slotted opto sensor). Without an extra isolator at this location we would create a ground loop again.

IR remote control receiver is also provided.

The linear mains power supply provides two isolated supplies (separate transformers). One powers the XMOS and system controller (5V), the other powers the Mosaic D/A converter and input / relay volume control.

The +2V and -2V are provided by a discrete quatrode tracking regulator that keeps both voltages perfectly matched for very low DC-offset. The +0V7 and -0V7 balanced digital power supply for the multi-stream decoder is derived from diode shunts. This voltage is kept as low as possible in order to minimise ground-bounce.
 

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Congratulations, dear John, and thank you very much for this new invention. It looks luscious and I hope I can have n° 001. :)

As always, you are so generous to share detailed explanation of the working principles, but also maybe a little reckless, save that you already applied for the patent. I bet not only the usual suspects, but also main players would jump into the idea.

Best wishes of success,
M.
 
While we wait, a couple of questions:

1) Does the Mosaic UV need special drivers to work?

2) How the volume level is set and shown and does it set at a level after powering-off?

One of the inconvenients of some electronic volume controls is that they begin at -80db after power-on, so one has to climb a lot and use the remote control batteries. A sort of "memory" level would be welcome. :)

Best wishes,
M.
 
Hi maxlorenz,

1) Does the Mosaic UV need special drivers to work?

The Mosaic UV contains a XMOS USB receiver that will work without drivers on sources that support UAC1 (max. 96/24) or UAC2 (max. 192/24). It is possible to use XMOS drivers on Windows systems, bit-perfect playback can be checked with the Mosaic UV bit-perfect test feature and a bit-perfect audio test file.


2) How the volume level is set and shown and does it set at a level after powering-off?

The volume setting is indicated with 8 LEDs, each LED has 8 different brightness levels. This way we can indicate 64 volume levels. One can set the volume through the included IR remote control or through the device volume setting (not the system volume setting) on the USB audio source. The system volume control setting is always set at maximum for bit-perfect playback. The device related volume setting on the USB audio source operates the relay volume control in the Mosaic UV through the XMOS.

Default volume settings for each input can be programmed with the IR remote control. These settings will be remembered, even when power is disconnected. After power up volume is set to minimum so one has to press the volume recall button once in order to set the programmed volume level.

When powered up, the Mosaic UV also remembers the volume setting for each of the 4 inputs.


I attached a photograph of the new audio set. The website will be updated soon.

On the left and the right are the new Equilibrium ZF (Zero global feedback) monoblocks. These are based on 48A / 0.75KW source followers running in class A with sliding bias in order to obtain low output impedance without global feedback. The output power of these test versions equals 50W rms, the production version will offer 100W rms.

The large heat sinks are required because the Equilibrium ZF runs in class A and has to dissipate 50 watts. The mains transformers are mounted at the front, as far away from the amplifier circuits as possible. That’s why the housing has this shape.

I tested global feedback circuits exhaustively and while technically correct (THD measurements) there are sound smearing effects and related loss of clarity. Global feedback also leads to a rather analytical uninvolving sound despite excellent THD specs. Finally, zero global feedback also reduces hum and noise to inaudible levels.

The box on the left is the Mosaic UV, it comes as 24 bit unit only. The yellow LEDs (bottom left) indicate the sample rate (44.1 … 192 KHz). The 8 orange LEDs (bottom right) indicate the volume setting. The 4 LEDs above the volume setting LEDs indicate the selected input.

The 6 LEDs above the sample rate indicator LEDs indicate USB receiver status, there is a green LED for bit-perfect test. The larger circular window in the center is the IR remote control receiver.

On the rear there are RCA and 3.5mm jack sockets, large USB-B socket, UAC jumper, mains entry and mains power switch. There are both fixed and variable outputs.

The heat sinks on this device are used to match the design and to make the housing even more rigid (mechanical resonances).


The box on the right is the SSD music server, it is based on the RaspBerry Pie 4, one or two SSD drives and a built-in clean linear power supply. Here the heat sinks are required for cooling the linear voltage regulators.

This is the modern replacement for the SD-transport. SSD capacity keeps increasing, Samsung already offers 4TB SSD's right now.

Because there are no perfect parts we will always end up with some USB source and interlink dependency as it is impossible to fully block source interference and jitter using imperfect parts. Therefore we developed this cleaner digital audio source. It is operated by a pad or smartphone (Lumin app) and communicates through an external, low power WIFI USB dongle.

There is an index button that initiates indexing after a USB stick or SD-card with USB converter is inserted or the data on the SSD has been changed. There is also a power-down button. When the unit has powered down, all front panel LEDs go off, now the power can be removed safely.

Using external wiring (LAN) and external mains powered peripherals will create ground loops resulting in significantly increased source interference levels. The SSD music server can stream through a network but this is not recommended.

The SSD music server is completely quiet as there are no moving parts. Even a HDD or small cooling fan produce background noise that will mask details. In order to hear most bits we need a listing room that is free of ambient noise.

This set gets the last bits of detail from a recording, so we need high resolution speakers in order to preserve this information as good as possible (weakest link).


So I developed suitable high resolution speakers that received the name finesse. Bass is based on a sonic resonator, one midwoofer is driven by the power amp, the other serves as “acoustical shock absorber” that minimises coloration and keeps the woofer under control, even when playing extreme bass. The housing dimensions are based on the golden ratio (approx. 0.6 : 1 : 1.6) in order to minimise issues with multiple standing waves of similar frequency.

Midrange and trebles are reproduced by an open baffle. I used 4 dipole fabric dome midrange units (80cm2 radiating surface area) for high resolution and low distortion. Trebles are reproduced by a dipole AMT (Air motion transformer) with 12.5 cm2 radiating surface area.

I use first order filtering on all units. I tested first order …. 6th order crossovers and observed following. With higher order filters the sound gets analytical and uninvolving and sound looses coherency.

The midwoofers have usable frequency range of 32 Hz … 9000Hz. The fabric dome midrange units cover 300Hz … 12 KHz and the AMT covers 1500Hz … 38,000Hz.

The speaker cabinet is made from MDF and HDF panels.

The speaker internal wiring is glued into routed slots in the wood so it cannot resonate. The crossover filters are integrated in the housing.

The finesse consists of two units that are mounted together.
 

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Hi maxlorenz,

The second woofer of your speakers is a "passive radiator" then?

Here is some information about passive radiators:

How Passive Radiators Work -All About Passive Radiator Speakers


The sonic resonator uses two drivers with motor. One is driven by an amplifier. The other generates an ac voltage that is loaded by a (frequency dependent) impedance.

So during correction, an ac current flows through the second woofer voice coil while it is not connected to the power amp. One can observe that the actively driven woofer cone excursion is much greater compared to the second woofer cone excursion. So the net result is dominated by the first woofer.

A radiator that moves in opposite phase, hence "shock absorber" as you described, to privilege coloration control instead of bass amplitude.

The second woofer membrane can move anywhere between in-phase and out of phase compared to the primary woofer. We observed this by comparing the electrical signal generated by the second woofer with the electrical signal applied to the primary woofer. The correction is highly complex and varies dynamically with music content.


We demonstrated the new set for the first time at a Dutch audio club, here is a youtube vid.

https://www.youtube.com/watch?v=VrxlYUps8bg&feature=youtu.be

Audio components:

Finesse prototype speakers, 3-way, first order filtering @ 1 KHz and 10 KHz, sonic resonator, open baffle.
Litz wire speaker interlinks (isolated copper strands)
Equilibrium ZF (zero global feedback) monoblocks, 50W rms.
Mosaic UV (Isilated USB interface, second generation ternary logic Mosaic D/A converter, built-in passive relay volume control.
Low cost 3.5mm jack interlinks from HQ (between Mosaic UV and Equilibrium monoblocks).
SSD music player (universal plug & play), clean, stand-alone digital audio source.

Set is controlled by an iPhone running Lumin app.
Ipad is used to display the artwork.

Audio tracks for demonstration, 44.1/16 “low res”

The Mosaic supports up to 192/24 files.



They documented this demonstration.

This is what was said about the sound:

“Na de theoretische wordt gestart met muziek. Gordon heeft een set van 5 nummers geselecteerd. En wat gebeurt ? Bijna totale stilte gedurende 5 nummers. Deze set weet ons te raken. Het geluid laat zich (in mijn eigen woorden) omschrijven als gecontroleerd, een hele brede stage, exacte plaatsing, en snel, vooral heel snel. Het is net alsof de muziek wordt losgelaten en allerlei barrières die er blijkbaar ongemerkt toch zijn, zijn verdwenen.”

Translation:

“After the theory it’s time for music. Gordon selected a set of 5 audio tracks. And what happens? Almost total silence in the audience during all 5 audio tracks. This set managed to touch us. The sound can be described (in my own words) as controlled, very large sound stage, exact focus, and fast, especially fast. It’s as if the music is released and all barriers that still appear to exist are gone.”
 
We demonstrated the new set for the first time at a Dutch audio club, here is a youtube vid.

https://www.youtube.com/watch?v=VrxlYUps8bg&feature=youtu.be

Audio components:

Finesse prototype speakers, 3-way, first order filtering @ 1 KHz and 10 KHz, sonic resonator, open baffle.
Litz wire speaker interlinks (isolated copper strands)
Equilibrium ZF (zero global feedback) monoblocks, 50W rms.
Mosaic UV (Isilated USB interface, second generation ternary logic Mosaic D/A converter, built-in passive relay volume control.
Low cost 3.5mm jack interlinks from HQ (between Mosaic UV and Equilibrium monoblocks).
SSD music player (universal plug & play), clean, stand-alone digital audio source.

Set is controlled by an iPhone running Lumin app.
Ipad is used to display the artwork.

Audio tracks for demonstration, 44.1/16 “low res”

The Mosaic supports up to 192/24 files.
John, I'm quite impressed by the video-recording of the session... Usually recorded sound in these cases is blurred, but the atmosphere is preserved quite well... The experience in the room must have been overwhelming .

BTW, who'se the artist/song?
 
Dear all,

Time ago -EC- shared his idea of using power attenuators between amps and speakers, using power resistors. I tried it then without much success, probably due to system-dependency of the idea.

Recently, I wanted to improve the sound of the "patio" system, which uses digital level attenuation, and which I hate, for lack of dynamics and detail, so I bought cheap power level autoformers (Sonance brand) for volume control level (I don't have specs nor manual) and I installed them on that system. The idea is to increase digital level output without reaching clipping state on the power amp and then using the autoformers to reduce the SPL to desired volume, and without bothering the neighbors. I can say that I am pleased with the result as I can listen more dynamics and detail even at lower levels, resulting in a more involving musical experience. I can imagine inductive attenuation is better than resistive attenuation also at power levels.
What do you think?

Sorry for this off topic.
Cheers,
M.

PS: dear John, how is going our Mosaic UV ?
 
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Hi maxlorenz,

Time ago -EC- shared his idea of using power attenuators between amps and speakers, using power resistors. I tried it then without much success, probably due to system-dependency of the idea.

The problem is increased output impedance, reduced damping factor and related loss of control over the speaker. But it works with speakers that have good self damping properties.

Recently, I wanted to improve the sound of the "patio" system, which uses digital level attenuation, and which I hate, for lack of dynamics and detail, so I bought cheap power level autoformers (Sonance brand) for volume control level (I don't have specs nor manual) and I installed them on that system. The idea is to increase digital level output without reaching clipping state on the power amp and then using the autoformers to reduce the SPL to desired volume, and without bothering the neighbors. I can say that I am pleased with the result as I can listen more dynamics and detail even at lower levels, resulting in a more involving musical experience. I can imagine inductive attenuation is better than resistive attenuation also at power levels.
What do you think?

By maximising the signal that enters the power amp you get improved SN ratio and more detail. The autoformers probably offer low enough output impedance to keep the speaker under control.

My personal experience is that manipulations in the digital domain will degrade perceived sound. Possible causes could be a damaged dither pattern and or rounding errors caused by calculations with finite resolution.


Recently we finally managed to integrate the volume control in the Mosaic D/A converter core by using some smart tricks.

The end result is a greatly improved, virtually loss-free volume control (2Vrms output) with constant 750 Ohms output impedance regardless of volume setting.

Signal path: Ternary logic cells -> resistor matrix -> power amp -> speakers.



PS: dear John, how is going our Mosaic UV ?

Another redesign as a result of new developments (volume control, novel USB receiver with native multi-stream interface), more on this later.
 
The autoformers probably offer low enough output impedance to keep the speaker under control.

That's what I thought. Apart that they are good to replace resistive attenuation inside passive crossovers. :cool:

My personal experience is that manipulations in the digital domain will degrade perceived sound. Possible causes could be a damaged dither pattern and or rounding errors caused by calculations with finite resolution.

Recently we finally managed to integrate the volume control in the Mosaic D/A converter core by using some smart tricks.

The end result is a greatly improved, virtually loss-free volume control (2Vrms output) with constant 750 Ohms output impedance regardless of volume setting.

Signal path: Ternary logic cells -> resistor matrix -> power amp -> speakers.

Good.

Another redesign as a result of new developments (volume control, novel USB receiver with native multi-stream interface), more on this later.

He, he. Well, we our fans, are used to it. :D
No problem from my side, safe that it will not arrive for my 50th birthday :(

A question: will "direct out" be still offered?
As I've mentioned before, I use a TVC (to which "variable out" will be compared) which offers me galvanically insulated connection.

Thank you and have fun,
M.
 
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