Tube filter for DACs. DAC with tube output. comments to “A simple tube filter for a DAC or CD player”

To go further in the design of amplifiers, I ran into the problem of a quality source. I really needed a good DAC. I was not fully satisfied with the quality of those that I had at home and that I had listened to before. If this is a classic DAC with operational amplifiers at the output, then this usually leads to problems in reproducing the upper mids and highs. The middle becomes slightly grating, harsh, as if there is sand or metal in the voice, especially at high volumes. With tube DACs, not everything is all right either - often there is no good bass or a flat, inexpressive sound, and besides, for some reason, developers really like to install a cathode follower at the output, which, although it reduces the output impedance, but in my humble opinion the sound to put it mildly, it does not decorate. In general, I came to the conclusion that I had to do it myself.

Why did I choose Ad1955? Its output is designed for an I – U converter with a current of 3 – 5 mA of positive polarity. And here is a wide range of options for connecting to a high anode voltage in such a way that the output current of the DAC chip passes through the lamp.

Yes, of course, I wanted a DAC with a tube output. And given my weakness for cascades with a common grid and transformers, the output was planned on my favorite 6E6P lamp with a transformer output. The choice of this lamp is also due to its low internal resistance in the triode, as well as its high transconductance (30 mA per volt), and in the case of a cascade with a common grid, this gives a lower input resistance - and this is very good for I - U DAC converters, for which the input resistance should tend to zero. It is logical to make the input I - U of the converter on a germanium transistor connected according to a circuit with a common base. This is where the scheme was born. According to my rough estimates, the input impedance of my hybrid cascode is somewhere on the order of 1 Ohm. How did you calculate? We take the formula for calculating the input resistance of a cascade with a common grid Rin = (Ra + Ri)/(u +1). The lamp load is 3.3 KOhm, the 6E6P itself in the triode has about 1500 Ohms. Add and divide by 30 - this is the gain of the lamp. It turns out 160 Ohms. This is the input impedance of a lamp connected according to a circuit with a common grid. Now for the transistor, the lamp is a load Ra. Internal resistance I don’t know the germanium transistor, but we take roughly 50 Ohms, then if its Kus is about 250, then (160 + 50) / 250 = 0.84 Ohms.

If someone finds 6E6P too emphasizing the middle, then it can be replaced with 6ZH9P, 6ZH11P or 6ZH49P. Only in this case, you should pay attention to the fact that the collector of the transistor is connected to terminals 1 or 3 of the lamp socket (and not to terminal 6) - then you can simply plug in the one that seems more melodious to you.

I present the first version of the scheme, although I am sure that it will have to be finalized, because there is no limit to perfection...

In order not to do the digital part myself, I took a DAC board for AD1955 from e-bay and removed the operational amplifiers from it, also unsoldered the 2K resistors from the power supply positive according to the datasheet from the AD1955 outputs, and left 100 pf (capacitors C1 and C2 in the diagram) those that were on the board. I'll give more details a little later.

I tried a transistor stabilizer as a power supply, but it still turned out to be the best-sounding tube doubler on the 6N1P, which was later replaced by the ECC99. The reason for using this rare lamp is simple - to package my DAC I used a case from a Chinese Lite DAC, which died for a long time, thank God I didn’t throw out the case. Both network transformers, the network button and the input/output connectors came in handy. Here is the power supply diagram:

As you can see, the 6E6P filament is powered by DC, but unstabilized.

Now a little about listening. The source was a Denon 1500 CD player and compared it with my DAC, the signal was supplied via an optical digital cable. The amplifier is my cascode for 6E5P - 2A3. Speakers - wideband in OYA from 3AC505. The first impression was very bad, I was very upset and was about to take my creation to the closet in the company with other unsuccessful projects. I found my DAC to be overly harsh on female vocals and trumpets. But then - lo and behold! - it turned out that it was me who mixed up the inputs on the switch in front of the amplifier - something that I was disappointed in - it was just a Denon DAC, but my DAC gives an excellent presentation of the material! And the timbral balance, stage width, and emotional richness will be higher than with Denon. In general, he sings clearly, in detail, transparently, and what especially distinguishes him from my signature Denon is the very soft presentation of vocals and the upper mids and highs in general - no ringing, no excessive harshness at almost any volume, in general - much more natural. Here it is appropriate to talk about the “coloring” of sound. As in colorimetry, when talking about color, it is important to answer the question - what is accepted as the standard for white? If we take transistor sound as this standard, then yes, the lamps give “coloring”. But in my understanding, tube sound is the standard of white. And the operational amplifiers at the output (by the way, always used with deep OOS) give a slightly metallic color and a slightly unnatural upper register, which IMHO is not typical for live performance. Overall, I was very, very pleased with my creation.

Here are its characteristics

– output voltage at 0 dB – 2 Volts;

– noise level – less than -80 dB, there is simply nothing to measure less;

– total harmonic distortion at maximum level– less than 0.15% – again, I can’t measure it more precisely yet.

– inputs – optical and SPDIF;

– outputs – unbalanced 2 Volts and balanced 10 Volts;

– output resistance – at the unbalanced output – less than 100 Ohms, balanced output – about 2 KOhms;

– the circuit does not contain OOS circuits.

Here is what the device looks like packaged in the case and a photo of the entire set of listening equipment.

The output transformers were wound to order at the Audio Instrument company, for which we bow to Sergei Glazunov. And also - read on the forum http://www.diyaudio.ru/forum/index.php?topic=4180.0. My first attempts (not entirely successful) to make a DAC using only tubes are in another thread on the same forum http://www.diyaudio.ru/forum/index.php?topic=1267.570.

Updated June 6, 2015. I had to adjust the diagram a little. Firstly, at peak volumes, excitation (resonances) was observed and therefore it was necessary to add capacitors C3 and C5 to the lamp grids, as well as C1 and C6 to the anodes. Also, due to voltage drift at the output of AD1955, it was necessary to stabilize the bases of the transistors using a 3.0 volt zener diode D1. Well, nevertheless, I replaced 6E6P with 6Zh49P - out of all those listed earlier, it seemed to me the most balanced in timbre.

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2295

The digital converter has a number of exclusive solutions

• Two lamp shreds (DEM and master);
• Two operating modes: master and slave;
• Separate power supplies for analog and digital parts with ballast lamps instead of chokes and resistors. Parallel stabilizers on P605 germanium transistors are used as a source of stable voltage for the external DAC. The power supply contains 4 high-quality toroidal transformer;
• Single-sided printed circuit boards of increased thickness 2.5 mm with copper foil 100 microns (usually 30-60). The tracks of the digital converter boards are covered with gold and violin varnish;
• The boards are soldered using vintage (30s) solder.

History of the creation of audio DAC abbasaudio 3.0

Finally, the final, “finished” boards of the first of several external DACs, conceived many years ago and having undergone many intermediate modifications, have arrived. These are digital converters of four quality levels, incorporating all the latest research in the field of esotericism, including very unconventional methods of influencing materials (which I will allow myself to remain silent about so as not to confuse the weak in mind). I will even take the liberty to say that these external DACs are the only ones of their kind, since the vast majority of decisions in their design were made not as a result of speculative theories, as is customary among engineers, but after many years of auditory tests, which were also confirmed statistics of reviews of boards for upgrades, on which many nodes of future audio DACs were tested. Without the help of many unbiased listeners who performed hundreds of experiments with my boards, I would have had a much more difficult time. For this, many thanks to everyone who took the time to write about their impressions of the operation of certain devices - clocks, stabilizers, spdif transformers, dem clocks, buffer preamplifiers.

In general, the ideology of building such digital converters was formed under the influence of Anatoly Markovich Likhnitsky. After reading and understanding his conceptual articles, later during live communication and listening to the AML system and several dozen AML+ remasters, many of which for me remain the standard of SD quality.

Gradually, I began to come to the conclusion that in nature there are practically no CD players or audio DACs that would satisfy the requirements for high-quality transmission of well-recorded live music. The decline in recording standards has led to an even more catastrophic decline in reproduction standards. And if the first and second generation players could still provide musical satisfaction to the listener, especially after simple modifications, then the latest models that the industry offers us as household audio are simply contraindicated for use for enjoying music. It is impossible to imagine a more anti-musical and emasculated sound of the digital converters built into them. Moreover, these devices can no longer be upgraded, and even a piece of lamp can play the role of a poultice for the dead. I perfectly understand the contempt for digital that amateurs who have serious analog sources feel. This is completely fair contempt, because what even the high end offers us in the field of digital is something completely obscene and inedible.

Therefore, I made an attempt to go back in time and, using discontinued chips, try to create a digital converter that would be close in quality to good analog sources - tape, vinyl. This difficult work took a lot of free time, but in general terms it is completed and the third in the line of external DAC 3.0 is prepared for release in a small series (no more than 10 copies, half of which have already been ordered). As it turned out, due to the huge number of elements that require quality control, you have to literally purchase small items for such a device separately for each item, which means that even the device is printed circuit board turns into an individual order with a long production, adjustment and adjustment time. Hence the small circulation - it is simply impossible to make more. Since I physically cannot assemble ten external DACs in the form of complete devices with housings, the remaining five copies will be sold as a set of assembled and debugged boards (main and two power supplies).

Description of projects

The above-mentioned digital converter based on PCM58 broke up into several projects, four to be precise.

• Two of them are on the TDA-1541, recognized throughout the world as the most analog-sounding DAC, although what prompted me to return to it was not someone’s recognition, but experiments with the DEM clock, which give promising results on the way to bringing digital sources closer to analog.
• As before, an expensive project on the PSM58 is in the works, but with a slightly different configuration - with an ancient Sony SPDIF receiver, with a PLL loop and a tube VCO.
• External mid-range DAC price category on TDA1541A with dem-clock on finger lamps.
• A simple inexpensive digital converter based on PCM56K with linearity adjustment and lamp shred on a miniature EF732 lamp.
• In addition to all this, an I/U converter based on germanium transistors is being tested as an alternative to a good transformer, but more on that later.

I would like to emphasize that even in the simplest external DAC based on PSM56, all the principles underlying expensive designs will be observed. Quality materials and elements selected by ear on the test path, minimalism in power supply, parallel stabilizers, at least one lamp clump on board, a printed circuit board without a mask on the bottom, gold and violin varnish as a coating.

Description of ABBAS AUDIO DAC 3.0

Today we will talk about an external DAC called ABBAS AUDIO DAC 3.0, designed to work with an external tube buffer and preferably a transformer after the DAC chip.

The suitability of absolutely every component, including terminal blocks and sockets, was determined by ear and checked various ways organization of nutrition, I was looking for optimal “bundles” of components that would give best result when combined.

A huge number of tube samples for digital converters were produced, using a variety of tubes, selected by sound. Logic - by year, series and manufacturer.

Audio circuitryDAC

After experimenting with various SPDIF receivers, I finally returned to the crystal chip, having discovered that there are copies on the market made in the USA (I found only two), in South Korea and Taiwan. In the same order, I prefer them in terms of sound.

The external DAC uses only South Korean chips; evaluation and comparison were made by ear, of course. The digital converter can operate in two modes - master and slave. The location of the jumpers is indicated on the board (see photo).

In MASTER mode, a 74LS74 trigger and one jumper are inserted according to the table (the 393rd chip should not be in the socket!!!), in the SLAVE 74LS393 (the 74th chip should not be in the socket!!!)

This somewhat cumbersome switching method guarantees the digital converter a minimum of extra soldering, chips and transitions, which means maximum sound quality. PLISC - and let's leave it to Vega Lab and other kindergartens!

In slave mode, the tube clock is divided and “drives” both the transport and the audio DAC for the TDA1541 DAC, a new frequency grid is created (the effect is wonderful) - word clock and bit clock.

Comparing different variants, I came to the conclusion that a bitclock formed using a lamp gives a big gain in sound, even despite the use of an SPDIF interface, and the connection complexity is small - just one more coaxial cable or twisted pair. The SPDIF receiver itself in this case does not cause a dramatic deterioration in sound, as in traditional audio DACs.

The only drawback of the “slave” mode is that it is necessary to insert the clock into the transport, which, given the difference in clock frequencies, is sometimes a serious problem. It is impossible not to start it - beats will occur between clock frequencies, which can be heard in the form of periodic clicks, increasing in frequency as the signal level increases. For these “hopeless” cases, I provided a switch to the traditional “master” mode, which makes the external DAC a completely independent device, albeit with minor sacrifices in quality. In this case, asynchronous recall is enabled, the solution is somewhat vulgar, but if certain conditions are met, it is extremely effective and sometimes superior to synchronous recall. I don’t have enough knowledge to explain this fact, but I trust my hearing 100%.

Lamps

The clock generator is built on turtles. After many experiments, I returned to where I started - a shred on the EF14. The use of this relatively expensive lamp in an external DAC is completely justified. Among the lamps of the 30s, the EF14 Telefunken has no analogues! High slope, low noise, durability, in addition, the EF14 case is not connected to the cathode, which means it is an effective screen, which cannot be said about the EF12 and EF13

The DEM piece of the digital converter was assembled on the EF13, the “run-up” in the lamps was made not only thanks to the features of the EF14 described above, I still try to avoid repeating the same lamps or components multiple times within the same device. This is a purely intuitive decision. If I lay out all my thoughts on this matter, I will get several more pages of text.

Kenotrons of type EZ11 or EZ12, it is not prohibited to use AZ11 (you will have to install a 4-volt separate incandescent trans). Run-up in kenotrons is also extremely desirable.

Powering up digital chips

TDA1541A, not counting the use of a lamp DEM unit, is turned on in a completely normal way. I don’t see the point in some special modes (differential, parallel) - they don’t justify themselves.

At the PC input, the circuits are filters, but using high-quality resistors and capacitors.

There is practically no ceramic in a digital converter; if we want to get analog sound from a digital signal, we should forget about ceramics. One ceramic capacitor is located near the comparator, and only because of urgent need. The use of windows that I don’t like in two places is again associated with problems of interference emitted by some nodes. That is why the section with the comparator has a polygon on the bottom and top and vias - this is an extremely “noisy” unit that produces a lot of garbage both along the power bus and on the ground. It needs to be properly powered and “chained”, preventing interference from scattering around.

The main thing in our business is to take the right start! I don’t have to worry about building a product line from cheap consumer goods to the very high-end. Therefore, I can afford to immediately choose the digital-to-analog converter chip I like and build a design around it. So, the “mystical DAC” was taken as the basis "as they call it on the Internet. I will not make it from a small microcircuit big secret, but let's still keep the intrigue first.

Build a good DAC I’ve been planning for my beloved since the last century, but somehow I didn’t get around to it and more priority tasks took over. And here, to my delight, a customer appeared, on the one hand able to appreciate good sound, on the other hand, willing to put up with a certain level of “homemade” in the finished device. Naturally, I will make every effort to ensure that my clients are satisfied with their choice. What my “pre-production” products lose in comparison with serial devices of popular brands is:

1. part of the editing is done with cobwebs on mole rats, and not on print, which has a positive effect on the sound quality, but, alas, will not be available in production samples;
2. I don’t skimp on little things like a surge protector or shunt capacitors, which, by the way, has been caught by recognized authorities more than once;
3. My “brand” is not yet very widely known in narrow circles :)

Let's start, pay attention...

Where to begin? That's right, it's best to start with a ready-made device, even a simple one, but containing key components. In China for US $ 50 I purchased a generally good kit for self-assembly of a DAC. As I already said, the Chinese economic genius is not distinguished by any special technical talents, so everything in that set was at a minimum, exactly according to the datasheets. Except that the creators of the set built, as it seemed to them, very high quality: they stuck " KRENOK" with garlands. But the kits came with very appropriate R-core transformers.

At this stage, the task was not to somehow specifically control the digital receiver or DAC, so the hard-wired minimalist S/PDIF->I2S->DAC chain suited me quite well.

I didn't consciously try to find a DAC with a USB input. The reason is simple: the computer generates a lot of noise and there is no desire to let all this garbage into the audio device. Of course, there are methods, but I still haven’t come across a single DAC with proper isolation USB input(devices for 1K green and higher, as well as products of Russian audio “left-handers” do not count).

I consider it necessary to note that despite all my quibbles about the circuit design, etc., the quality of the printed circuit board is simply excellent!

Taking control of the situation into our own hands

In the documentation for the DAC, in one place it is written that the analog power leg must be bypassed with an electrolyte of 10 μF and ceramics of 0.1 μF. In the diagram, leg 18 is bypassed exactly like this.

A little further in the same document it is said that it is advisable to bypass the input on pin 17 with an electrolyte of 10 μF and ceramics of 0.1 μF. The developer acted in full compliance, dutiful comrade, just great!

Another place in the documentation says that 17 leg Can run it straight to analog power. This is what we see in the diagram :)

What’s funny is that not only in the circuit, but also on the printed circuit board, everything is laid out like this: with two electrolytes and two 0.1 µF capacitors, with a short one right between the 17th and 18th legs of the chip (the path to the capacitors from the 17th leg goes under the chip body) :

Everything came just this dirty from the factory. How I washed it is a different story :)

For those who are especially curious: the pitch of the legs of the microcircuit body is 0.65mm.

I once came across a gorgeous picture from my friend Vadich-Borisych on VKontakte: " resistance is futile". Here, it inspired me, it is as useless here as the duplicated shunt capacitors in the diagram above, I redrawn the “circuit” especially for you:

I needed to control what was happening on the 17th leg. I had to cut him alive. It’s good that they haven’t put a jumper under the chip yet - the prospect of unsoldering one leg of the SSOP case is somehow not encouraging.

Mediocrity goes overboard

What digital-to-analog converter is complete without operational amplifiers?

That's right, only high quality DAC. So I simply did not solder the modest filter on the NE5532. Maybe it was worth it to have something to listen to for comparison and make sure how unconvincingly deep loop-backed op-amps play... But I already have a CD player from a venerable manufacturer, which very diligently plays the very mediocre sound of op-amps, although hidden behind the sonorous name HDAM and soldered into small screens. And there are plenty of other similar “samples”.

Study, study, and... think!

Perhaps on all, without exception, DACs from manufacturers from the “heavenly Empire” I see the same locomotives from “KRENOK” (the photo on the right is not mine, caught on the Internet). By fanning out serial voltage stabilizers, the developers are obviously trying to achieve better power supply isolation and reduce the penetration of interference from the digital part to the analog part. Unfortunately, the masses lack what I call “current thinking” in circuit design. In fact, everything is simple and... a little sad.

Look at some LM317 from the output side. You will probably find a 10 µF electrolyte and a few other small containers. Now let's estimate the time constant in this circuit: just look at the datasheet and make sure that the output resistance of the "crank" is very small, which is what the developers of the integrated stabilizer sought. To be honest, I’m too lazy to count now, but interference with frequencies from, say, 100 kHz and below the roll “sees” right at its output, that is, the control electrode and, as it was designed, transmits these pulsations “upstream on command,” diligently trying to maintain the voltage on its way out.

Current fluctuations reach the output of a higher voltage stabilizer. Following the same logic, fairly high-frequency current changes still flow almost unhindered throughout the entire chain of stabilizers. And they whistle and make noise to everyone around.

I see the only rational grain in the use of two linear stabilizers in a row is that small precision stabilizers usually do not tolerate high input voltages, and kits for self-assembly of DACs often fall into the hands of soldering riggers, who often do not even bother to look into documents for the used components. And the kits should still work...

The spread of sufficiently high-frequency interference can be easily prevented by adding... ordinary resistors to the circuit. Simple RC filters by entrance linear stabilizers will provide excellent decoupling of RF ripples in both directions, sharply reducing the “distance” in the circuit where surge currents reach (including the “ground” wire!)

So the power supply has undergone major changes on the board. Alas, it was not without a couple of cut tracks and hanging installation.

Sometimes a small resistor is much more effective than a large capacitor:

We respect the heritage of our ancestors

Instead of a stupid bridge, we put super-fast diodes in the rectifier, which significantly reduces the current “shocks” when the diodes are turned off. This technique is quite popular and quite meaningful, so we will use it too:

By the way, it is precisely the lack of understanding of how to decouple linear stabilizers at HF ​​that leads meticulous developers to start installing a separate transformer for each block of the circuit. Another very popular, but also costly solution to the problem of series stabilizers: the use of current source-parallel stabilizer combinations. IN in this case Everything is fine with the decoupling, but the power has to be dissipated with a considerable margin.

Let's not demand too much from the "whale"

A separate article is needed to describe a series of experiments with various stabilizers. Here I’ll just note that, to the credit of the developers from the Middle Kingdom, the LDO stabilizer they chose, lm1117, may best option from commercially produced and relatively affordable integrated stabilizers. All sorts of 78XU, LM317 and others like them simply rest due to the incongruously high output impedance (measured at 100 KHz). Alas, the precision LP2951 went into the same basket. The TL431 behaves a little better in a shunt stabilizer circuit, but it has its own story: TL431 can be very different, depending on who made them. 1117 wins by a landslide. Alas, it also turns out to be the noisiest stabilizer. It rumbles and squeaks, both with and without load.

I had to assemble the stabilizer myself, using discrete components. From just two modest transistors, following the HotFET ideology, we managed to “squeeze out” everything that in an integrated design requires dozens of transistors and still falls short. Of course, to ensure the work of the “sweet couple”, several more active components were required... but that’s again a completely different story.

An interesting result of macro photography: I didn’t notice with the naked eye that the board was not completely washed off from the flux.

Polymers rule the roost

The latest modification aimed at achieving the most accurate sound transmission was the “smoothing” of the power supply.

In critical places, the usual (albeit good ChemiCon) aluminum electrolytes from the kit were replaced with solid-state aluminum Sanyo OS-CON. Since I collected two identical sets in parallel, it was possible to arrange “A/B” testing. The difference is barely audible, but it is there! Without a signal with conventional electrolytes, at (very) high gain, there was a certain “noise space” in the headphones. Polymer electrolytes take us into the absolute.

Sanyo OS-CON - purple barrels without a notch on the lid.

If you don't want to think with your head, work with your hands

On almost all boards and DAC kits using the CS8416 digital receiver, the Chinese put a toggle switch so that the user can choose between an optical and copper S/PDIF input (the photo on the right is a typical example caught on the Internet). So: there is no need for a switch there, the receiver chip can easily listen to two inputs without any outside help, be it a crude toggle switch or a smart microcontroller.

I’m sharing with you a trick I spotted on a demo board from Cristal Semiconductor themselves. It is enough to connect, for example, copper S/PDIF to RXN, and the output of the optical TOSLINK receiver to RXP0.

I hope there is no need to explain how this works? 😉

Even in the reference design, the companies screwed up and forgot the shunt capacitor in the TORX power supply :)

Economy or illiteracy?

It can be very useful to read the documentation of manufacturers, especially those that make the very microcircuits that audiophiles then swear by. I’m revealing the most secret secret: reference design board, evaluation board and similar “probes” from manufacturers usually contain examples literate the use of those same microcircuits. Moreover, it is not at all necessary to buy all these boards, and the price tags for such “samples” can be very different: 50, 400, and can exceed a thousand greenbacks. But, my dear developers, the documentation for all these boards is publicly available! Okay, good to teach.

So, what the Chinese did not read, or what they saved on: modest shunt ceramic capacitors of 1000 pF in parallel to 10 μF and 0.1 μF. It would seem - why, because with such capacitors we bypass frequencies from tens of megahertz and higher. The audio range is considered to be up to 20 kHz, well, up to hundreds of kHz. But no one has canceled the digital part in the digital-to-analog converter. So it is precisely the interference at tens of megahertz that freely walks through inexpensive home-built DACs, causing all PLLs to tremble in fear and thereby creating ideal conditions to cause the dreaded JITTER to occur.

Another popular way to save on matches

The vast majority of manufacturers of both digital audio sources and digital-to-analog converters save 30...50 cents on each device. We, the users, pay for this. Read details.

What's high-end without lamps?

I am amused by the hordes of tube-DAC and tube-headphone-amplifier "s in the price range from one and a half hundred to hundreds of dollars that have flooded the market recently. People seem to like how a light bulb hisses and distorts at 15...24 volts anode. However, analysis of all the problems of such DACs and pseudo-tube amplifiers for headphones is a topic for a separate article, but not just one.

(photo on the right is an example, I don’t have such a lamp-tac)

Rich topic. I just skimmed the surface here and didn’t touch on the analog part at all. And how interesting it can be to properly plant the “ground” or organize a simple and, at the same time, convenient control apparatus. And what are attenuators worth - after all, you can choose them with different resistances, build them according to different topologies, and include them in different parts tract. Coordinating sources with load is a very, very interesting question, you know!... But for today it’s time for me to wrap up.

BOM, or Bill of Materials

Of course, the matter is not limited to fifty dollars. The ceramic capacitors from the kit were replaced with film. Schottky diodes, high-quality electrolytes, and a lot more had to be added, not to mention the housing. And, of course, my HotFET amplifier: only 2 (two) amplification stages from the DAC output to the headphones or amplifier output. Neither more nor less, but in the amplifier itself I counted 32 transistors in the stereo version. Yes, all transistors are JFETs and depletion MOSFETs. No way I can’t fit into the green fifty kopecks even in terms of components :) And note, this is without any audiophile esotericism. Well, yes, I also have my own opinion on this matter. After all, there are people who believe that by installing the “right” components, any circuit can be made to sound. If you, dear reader, are from their ranks, teach me, I will listen, argue, listen and tell everyone about my experiences right on this site.

So where is the promised freebie???

Friends, this article is just thoughts, notes in the margins, it was written hot on the heels of remaking a Chinese DAC. I myself would never get involved in such an adventure again: although it turned out well, it was too expensive in terms of time and effort. And I don’t recommend it to anyone. When I dealt with that set, the poison simply oozed out, which was reflected in the article :) I apologize for the slightly arrogant style of presentation, and if I did not live up to your expectations and did not offer the distribution of almost free high-end DACs to the population 😉

If you were interested, please let me know. There is still a lot of material in the bins, but the strength, motivation to publish and formalize all this comes mainly from reviews and comments from my readers.

3181

Exclusive PCM58 DAC with EF11, EF13 Telefunken “turtle” tubes in the master oscillator

The Telefunken turtle lamp is soldered directly into the PCM58 DAC board, its service life is 10-15 years

Selecting a Digital Filter

So, having chosen the final version of the digital-to-analog converter on the Burr-Brown PCM58 DAC chip in front of me in full height The problem of integration into a digital filter circuit arose. I want to say that I don’t like digital-to-analog converters that use delta/sigma and similar algorithms for the unnatural effects that occur at their output. I tested a lot of digital filters and never came to a clear conclusion whether they are needed as part of a high-end DAC or not. Some fragments of music and entire compositions without a digital filter sound much more prominent, lively and richer than with it. And some without a digital filter are generally impossible to listen to, such a duality is incomprehensible... Here a lot depends on what the non-over-sampling DAC is working with, but the result is ambiguous in any case.

Even in my first digital-to-analog converters, I made toggle switches that allowed me to either connect a digital filter to the output of the DAC chip, or work directly. Five years of constant clicking have convinced me that audiophiles need to be given the opportunity to choose their own DAC operation: with or without a digital filter. In this regard, in the circuit of an experimental top-class digital-to-analog converter based on PCM58 Burr-Brown chips, I provided a connector with six modules that change within a few seconds. You can install either a shift register in the connector my development(see link), or a digital filter from the list below:

• CXD1144 in X4 mode;
• CXD1244;
• SM5842;
• SM5813 (DF1700);
• PMD100 in X8 mode.

Which is quite enough to select the sound character of a DAC to suit almost any taste. There is a separate article about comparing the sound and features of using different digital filter microcircuits. First of all, I can say that from the presented list, I like the digital filter chip CXD1144 the most, but this particular chip is very scarce, it is almost impossible to get it from suppliers and it will not be installed in a serial DAC based on PCM58 Burr-Brown.

Shift register

As for shift registers, just like with digital filters, I tried a lot variety of options. On the Internet, information on shift registers is distributed by some incompetents or saboteurs who write about the “ten-story” schemes necessary for their implementation. In fact, in order to connect DACs with a resolution of 18, 20, 24 bit to a signal processor via the i2s bus and the Sony data transfer protocol, you need only 3 logical chips. In this case, there is no need to embed anything into the data bus. this leads to severe sound degradation.

I'm not talking about the fashion - installing complex PLIS format converters that simultaneously serve as a shift register. I tried this PLIS converter once as an experiment and became convinced that it was completely unsuitable for obtaining high-quality sound. The shift register is designed to delay the DAC load update signal by 2, 4, 8 bit clocks, respectively (for 18, 20 and 24 bit). The shift register itself must be assembled using high-quality vintage elements, tested for musicality and have a well-organized linear power supply. For the serial version of my DAC, I provided a shift register based on Signetix logic chips from the 80s, powered by a “parallel” voltage regulator on a vintage Telefunken transistor.

S/Pdif receiver

I'll tell you about the S/Pdif input receiver. The choice of the Yamaha YM3623 microcircuit was more spontaneous than based on any calculation. According to all Internet publications, this antediluvian microcircuit has huge jitter, which is unacceptable from the point of view of an engineering approach to the design of a high-class DAC. However, everything is not so simple here. It is this synchronous S/Pdif receiver with reclock that sounds much cooler than much newer and more sophisticated ones. Which raises a legitimate question: does the much worse sound of newer devices depend on the internal sophistication? Maybe that’s the point: the Yamaha YM3623 input S/Pdif receiver is made inside in such a way that it couldn’t be simpler: minimum logic, minimum formats, current consumption less than 10 mA. Especially in comparison with the chip from Crystal cs8412 and the now fashionable DIR microcircuits.

All this mass of logic inside DIR and Crystal requires high-quality power supply and generates noise along the internal buses, which naturally creeps into the output of the microcircuit. After all, logically good sound“The simpler the structure of the microcircuit, the more environmentally friendly, cleaner and more natural the environment inside it.”

These fabrications were confirmed in comparisons of the sound of a PCM58 DAC prototype with the ability to hot-plug S/Pdif receivers different manufacturers and years of manufacture. As a result, I settled on the Yamaha YM3623, although it is criticized by all and sundry. Remember, the most expensive external digital-to-analog converters of the 80-90s, which were equipped with this particular microcircuit! The Yamaha YM3623 was also used in numerous professional audio processing equipment. For DAC upper class I chose this chip as a base one and supplemented it with an external receiver with AM26LS32 type hysteresis (in a ceramic housing) and an S/Pdif input transformer.

Tube Master Oscillator

Well, the main feature of my digital-to-analog converter is the built-in tube master oscillator based on Telefunken EF13 “turtles” and kenotron power supply based on an E311 lamp. The choice of these particular lamps for the serial DAC on the PMC58 is due to the fact that from the mass of vintage they are the ones that are easily obtained and equipped with metal body acting as a screen. Their sound is more expressive than that of finger-type triodes, and their service life is so long that in the gentle modes of a DAC tube clock they can work for decades.

In my digital-to-analog converter on PCM58, I provided jumpers that allow you to select operating modes of the clock generator:

• Synchronous Recall. The clock is fed to the digital filter, to the resynchronization triggers and to the transport (in the transport, the signal may have to be divided by 2 or 3. For this option, I have production program there is a universal divider, described in the article about tube master oscillators);
• Asynchronous Recall. The digital filter receives the clock frequency from the S/Pdif stream, and the DAC itself is connected to the transport only with an S/Pdif cable. Thus, the master oscillator (clock) is involved only in the resynchronization node. The asynchronous relock option is slightly worse in sound than the synchronous one, but it allows you to connect the DAC to various CD players and transports, which is important for audiophiles who have not yet decided on a CD drive.

In both versions, the tube master oscillator operates constantly. All supply voltages are supplied to it from a high-quality external power source.

Supply system

Great attention has been paid to the quality of the DAC's power supply. It does not contain any standard parametric (serial with feedback) voltage stabilizer. This digital-to-analog converter has the best-sounding (IMHO) parallel shunt-type regulators. Most of them were collected from the simplest scheme on two parts High Quality and sound tested: vintage zener diode: Telefunken, Mullard, Motorola and vintage ballast resistors: NCF, Allen Bradley, Siemens.

Only two consumers are connected through a powerful parallel voltage regulator using a vintage Motorolla germanium PNP transistor. This is the PCM58 DAC power bus with a voltage of -12 V and a digital filter or shift register assembly. Some microcircuits consume a current of more than 50 mA, which a simple parametric stabilizer using a ballast resistor and a zener diode cannot produce.

I describe the differences in the sound of parallel and series voltage stabilizers in almost every article, and the parallel stabilizer always turns out to be better. Although it consumes significantly more current than series and, accordingly, requires a power transformer of greater power.

Electrolytic capacitors in the wiring of the DAC microcircuits are also very audible. In my DAC I have vintage 25 uF 35 V jars from Hydra, which outperform 90% of expensive electrolytes and sound simply excellent. In less critical places, where minimum dimensions are required, nichicons are installed, soldered from first-generation CD players of a vertical design. Unfortunately, I was unable to find modern electrolytes of similar dimensions with the same transparent sound. Therefore, I use proven vintage (naturally not dried out by time). In several places of the DAC there are ELNA Cerafine electrolytes and a lonely Black Gate of the NX series (thoughtlessly “plugging” Black Gate wherever possible harms the sound and wallet much more than their complete absence).

There are no ceramic capacitors or CMD elements in the wiring of PCM58 chips. In those places where it is necessary to suppress interference, Siemens and Philips film capacitors are installed; their number, type and ratings in each device are selected by ear. There is not a single DAC with soldered parts “in the image and likeness” of the pilot sample. Each digital-to-analog converter is purely individual (almost) and is configured individually, and I don’t work by eye...

By the way, I noticed that increasing the value of electrolytic capacitors above a certain value, as a rule, makes the sound heavier. It’s probably not in vain that in the most musical and “soulful” vintage CD players, the electrolyte rating in the DAC chip circuit does not exceed 20-50 uF.

The power supply of the digital-to-analog converter contains full-wave (FWW) rectifiers using vintage 1N5060 diodes. These are the diodes that were installed in the first generation Philips CD players, which are still the standard of digital sound. Trying to replace these diodes modern devices Schottky, Ultrafast, etc. leads to complete degradation and killing of sound... So, even in low-power rectifiers - only vintage and nothing else... Windings power transformers made with vintage wire with a midpoint. The DPPV circuit migrated to the DAC from tube amplifiers, and everyone knows that it plays better than a bridge one.

Trimming of microcircuitsPCM58

The signal to the PCM58 microcircuits is supplied from D flip-flops from Fairchild Semiconductor or 74LS74 Signetics, and the DAC update signal is replayed in them. In my opinion, updating the remaining data is harmful and pointless.

At the output of the digital-to-analog converter, I installed transformers with k.tr. 1/10 on vintage Telefunken permalloy. I once wound them for a preamplifier as MM/MC transformers. In a serial DAC, I will most likely install transformers with two coils based on permalloy from UTC industrial transformers, because by ear they appear air-transparent, and by instruments they are extremely broadband. The second pair of experimental post-dac transformers does not fit on the board, so in the photographs they stand next to it.

The need to use a ballast resistor in the positive power bus of a DAC on PCM58 chips prompted me to the solution that I used in a hybrid amplifier - to use a lamp filament as a ballast resistor. In that amplifier I loaded a powerful field-effect transistor with a quiescent current of 3 amperes per filament of a GM-70 lamp. The device played very expressively and was as simple as a board, but in terms of heat generation and dimensions it was “monstrous” and unsuitable for series.

In the experimental DAC, this role was taken on by a finger lamp installed in the power supply. It uses only filament, and for a digital-to-analog converter its performance does not play any role, the main thing is that the filament is intact. The nature of the sound can be selected by plugging in a variety of lamps that match the filament voltage and current.

And one significant nuance: it was possible to carry out a very simple and effective adjustment of the linearity of the 4 most significant bits of the PCM58 chip. This unit contains German carbon tuning resistors from the 70s. Each channel is adjusted individually and only by ear. Trimmer resistors for military purposes are characterized by increased reliability.

I somehow caught my eye on a DAC circuit based on PCM2704. And I really wanted to repeat it. Simplicity and good feedback. Then, when I began to gradually gain knowledge, it turned out that this chip was not the only one, and there were a dime a dozen sold amateur DACs. After reading some forums I found out. There is an opinion that the PCM2702E chip, although it has less functionality, but, according to reviews from writers, gives a more pleasant sound. So I decided to check these statements. Having rummaged around on the Internet, I found out that the PCM2702E is still considered a good DAC, although it has long crossed the age limit of 10 years. Moreover, there are many different circuits for implementing this converter with a filter and an amplifier, both on silicon and on tubes. Well, since lamps are now of greater interest to me, I opted for two schemes from Laconic Lab.

But first, about the implementation of the DAC module on PCM2702E.