Tube sound: myth or reality? Speaker (audiophile) N1. Warm tube sound Warm tube sound lurkmore
Somehow I accidentally noticed that 90% of articles on the hub with the tag “warm lamp” talk about anything, but not about lamp technology. At the same time, few publications about tube devices receive many likes and admiring comments.
I no longer remember how and when this strange idea settled in my head - to assemble a tube amplifier. Why is also not entirely clear - I’m not a music lover, I quickly got over home theaters a long time ago, and as a souvenir of that time, I still have the Wharfedale Diamond 8.4 floor-standing speakers, which in recent years have been used exclusively as a decorative flower stand. Be that as it may, the thought settled so deeply in my head that I began a leisurely study of specialized resources, reading forums, searching for tube amplifier circuits “for dummies,” etc. and so on. The lack of any experience with tube technology (the most modern gadget that I remember was a b/w TV in a student dorm in the early 90s of the last century) was frightening and attractive at the same time.
The sluggish search could have continued indefinitely if one day a wonderful resource had not been discovered - http://tubelab.com/. I settled on single-ended amplifier Tube Lab Simple Single End (SSE), ideally suited to my interests, namely: a simple amplifier for beginners with a minimum of components, the absence of any adjustments, at the same time quite universal and, judging by the reviews, proven to be excellent. The board was ordered on the website (shipped anywhere except Russia and Italy), payment via Paypal, short correspondence with the developer, fairly fast delivery of two boards (In addition to SSE, a board was also ordered for the advanced version of Tublab SE - so to speak, “for growth”) . It was decided to order components through e-bay, not quickly, but reliably and inexpensively - delivery times were compensated by convenience (receipt at the post office, leisurely search while sitting at the computer). The process took quite a while for a long time, but I was in no particular hurry (almost 2 years passed from the moment I ordered the boards to the moment of successful activation).
First components received
It makes no sense to describe the process of assembling the amplifier board, detailed instructions with pictures are on the project website. I was especially pleased with the disclaimer
We are not responsible for injury, accidents, acts of random stupidity, burning your house down, exploding parts, and other undesired actions (all of which are possible) resulting from the use of ANY information contained herein.
Translation
We are not responsible for injuries, incidents, acts of random insanity, burned houses, exploded components and other undesirable consequences (all of which are possible) as a result of the use of the information contained on the site
Some recommendations received during the study of materials.
- Never install electrolytes “all the way”; there should be a small gap between them and the board. The fact is that when soldering, the leg heats up and lengthens, and as it cools down it shortens, and, if the fit is tight, it can simply fall off the lining. Considering that in a tube amplifier the heating and cooling process occurs regularly, this point is worth paying attention to.
- The chassis of the output and power transformers should be placed perpendicularly to reduce mutual influence.
- The audio input connectors should be isolated from the chassis to eliminate the possibility of “ground loops” appearing in the signal lines. If the wire is shielded, then the shield should be grounded only on one side.
- Order components with a reserve in order to avoid logistics delays and save on delivery.
- And most importantly, be careful when purchasing components on ebay (more on this a little later).
Finally, the moment came when all the components were received, there was free time and nothing stood in the way of seeing how it all would work. In violation of all safety regulations, all connections were made directly on the table in front of the monitor.
An old LG-P500 was invited to play the role of the signal source, speakers from the music center were invited to play the role of speakers, it took some red electrical tape and a little courage. Sooooo - the switching on took place, nothing exploded, the lamps glowed with a beautiful orange light... and silence, or rather, if you put your ear to the speaker, you could even hear music against the background noise, but it was not at all the “warm tube” sound that I was hoping for hear.
The first thing I decided to check was the voltage at the output of the rectifier, and was immediately unpleasantly surprised, instead of the 375V x √2-27V= 503.33V I expected (voltage at secondary winding multiplied by the root of 2 minus the drop on the lamp), I saw almost 550V at the output of the rectifier and, accordingly, 525V B+ (anode voltage). There was no desire to test electrolytes for endurance (they are designed for 500V), so I had to turn off the power. Having checked the network voltage, I was once again surprised - it turned out to be more than 240V (a further survey of neighbors confirmed that this was the case for everyone). Fortunately, the transformer can be reconnected to this voltage. When turned on a second time, the voltages returned to normal, but the speakers were still silent; further testing revealed the absence of anode voltage on the input triode, which, in my opinion, indicated a malfunction of the only semiconductor device - the IXIS10M45 adjustable current source.
Having decided that the problem arose due to overvoltage and/or a Chinese ebay seller, I ordered a new pair of IXIS10M45 from England, which seemed more reliable and faster. I must say that the next switch-on ended in absolutely the same way as the first and second ones; although the new parts looked completely different, they refused to work in the same way. This is where I began to worry, since both channels behaved completely identically, and there was absolutely no voltage at the 12AT7 anodes. Since in this circuit there was nothing else except the lamp itself, the current regulator and the a priori working small things, suspicion fell on the lamp. An auction on ebay made it possible to buy an ECC81 (European analogue of the American 12AT7) quite inexpensively, and at the same time another batch of IXYS 10M45 (again a Chinese seller, he took it with a reserve just in case). The third batch of 10M45 looked (and sounded) exactly the same as the second; for the purity of the experiment, I immediately replaced the lamp and IXYS, disconnected everything unnecessary (the second stage) and for the fourth time did not find anything on the anode of the first triode.
A complete failure, my mind refused to understand how this could be. On a breadboard, I assembled a simple circuit with an LED and an adjustable current source (I used an untouched one from the third batch), powered it from the laptop power supply - and IT DID NOT WORK!!!
At that moment, the thought of a universal conspiracy began to haunt me; even what was supposed to work did not work... and I again decided to order the problematic microcircuits, only through a trusted seller (Digikey). And once again, difficulties arose even where they should not have existed. The first problem that arose (at Digikey, the minimum delivery cost to my region was $75, even for a $5 order). This problem was solved with the help of an American intermediary, but the second one emerged after placing the order - a letter was sent to my email asking me to confirm that I am not a terrorist, fill out the BIS711 form (for those interested, goo.gl/VAkDYB). I ordered regular radio components to an American address, why do I need to fill out this form I still don’t understand when buying conventional radio components. Having indicated my name, name and home address in all fields, namely: I am the end user, I am the official representative of the end user, I am the buyer, I am the exporter and indicated that I am an individual, I sent the completed form to Digikey, and the very next day I received confirmation of the order and tracking for the parcel.
Next batch appearance was different from all previous ones, which inspired some optimism (picture below)
The test on the breadboard was encouraging; the LED happily changed its brightness depending on the resistance of the control resistor. Five minutes to replace a part on the board...
...the next turn on and MUSIC began to sound from the speakers.
As it turned out during communication on specialized forums, fake radio parts on ebay are becoming a big problem. This is what Diyaudio moderators write
- Fake parts are a real plague by now. No small chance we all get a share of those when fishing for a quick small purchase.
- I never buy semi-conductors or electrolytic capacitors on eBay for this reason.
What will my “author’s column” be about? Yes, about everything related, albeit indirectly, to the world of audio. The word "audiophile", included in the name of the speaker, in the vast Russian Federation received a clearly abusive connotation. So to speak, it immediately classifies the holder of such a “title” as a certain minority. Personally, I prefer to use the term “music lover” in relation to myself, but this does not save me from periodic distortions into audiophilia. So, I hope, we will talk about forum celestials, and about warming up interconnect cables, and about the meaning of audiophilia as a phenomenon - perhaps with the involvement of experts (of course, world-famous, and no less) on these issues.
Let’s start a series of revealing materials with a typically “audiophile” topic, namely, with the term “warm tube sound” that has stuck in our teeth, fortunately, these days are hot. Why the sound - I think there is no question. But not everyone understands why it’s warm, and especially why it’s warm. Also, the considerable price tags for equipment with glowing bulbs contribute to the confusion about the warm and (of course) tube sound.
I do not pretend to fully cover this issue, but I hope that readers will glean at least some interesting information from the material.
Let's first turn to an area directly related to music, namely, its recording. It’s no secret for musicians that thanks to various “tube” gadgets you can get extremely interesting effects when these same lotions are overloaded. But at the final stage they are unlikely to install a lamp - fortunately, recently (about 20 years ago) the recording has been mixed on absolutely soulless computers and along the way it has been run through no less insipid-sounding mixing consoles.
Thus, tubes are used when recording music to introduce some “pleasant” distortion. Let's remember this fact.
However, recorded music must be played back somehow. Let's leave issues of digital-to-analog conversion outside the scope of this material, although it should be noted that some DACs (for example, MHDT Havana) use tubes at the output. Let's look at amplifiers with glowing bulbs. For example, Woo Audio WES for 5 thousand US dollars.
There is no doubt this is a great amplifier. The huge number of positive reviews leaves no doubt about this, and I won’t play “tear the covers off”, I’ll just recommend looking at the picture. In my opinion, it's a beautiful thing. By the way, the connectors on it are designed for Stax and Sennheiser Orpheus headphones, which cost comparable and big money. You can also purchase various upgrades for this engineering marvel, for example, the right capacitors ($1,280) and 50-year lamps (4 pieces for $520, one-year warranty).
Why is all this needed, and most importantly, how are such wonderful lamps found that should sound good and at the same time cost as much as a good mp3 player? In the article about the GAME card amplifier, I wrote that the music lover’s movement essentially has no end - you can always change something in your system, and thanks to the wide selection of components in the low, mid, high, and for what in general there is so much money” segments – the range is very wide.
I hope the attentive reader follows my thoughts. So, the fact that lamps introduce distortion, and this is used in the recording process, is an obvious fact. Also, software and hardware solutions based on transistors that are comparable in cost do not provide such color.
Now let's combine all of the above. If the recording may already have warmth and tube-like quality, is it worth further distorting the sound? At first glance, definitely not, because it is widely known that the so-called. “Audiophiles” do not strive for stronger distortion, but, on the contrary, for higher fidelity (hence the well-known and vulgarized abbreviation hi-fi by marketers). But let’s not lie – there can’t be so many on the market various devices, each providing the best sound quality. Yes, various discussions are possible here about what exactly is considered the “best” quality, and what, on the contrary, is a stronger embellishment. The emotional aspect is often put at the forefront, namely, so that a lovingly selected combination of audio components brings joy and positive vibrations when listening to your favorite music. If you don’t go deep into replacing interconnect wires, modifying headphones and acoustics, correct phasing of the plug, power conditioners, then the easiest way is to change the amplifier or its components.
Yes, you can also change your headphones (or speakers). The question is - why do this if the sound as a whole is satisfactory, but the current sound style is somewhat boring? So to speak, you can paint your life with new colors. Of course, the inquisitive reader will say, you can use VST plug-ins that programmatically “decorate” the sound even in a gray-brown-crimson color. But what should people do who do not want to use a computer as the main source and buy licensed discs the old fashioned way, listening to them on expensive audio players (for example, Accuphase DP-510 for 6 thousand US dollars)? And let’s not forget that a lamp is an analog element, which is difficult to emulate in software and not very promising.
But even if you are not yet “ripe” before purchasing a high-quality sound source, you can always connect the amplifier to a laptop or built-in sound card. You don’t need any special headphones; inexpensive “Fishers” and some “Sennheisers” will do.
So all that remains is to choose a tube amplifier and correct set lamps for it, fortunately some models allow you to replace them without much difficulty. Let's say, the inexpensive (only $200) Laconic HA-06 is quite suitable for experiments:
By the way, this is a Russian development that is also being promoted on the Western market, seemingly with some success, and this is encouraging.
Yes, all these amplifiers produce distortion in the sound. Let's take a break. There is such a wonderful science called “psychoacoustics”, the tasks of which, among other things, are to create more and more advanced algorithms for masking artifacts in compressed recordings. Simply put, the difference between MP3-like recordings and CDs is gradually disappearing. Yes, now records that were once described as being of CD quality are no longer perceived so well. And modern codecs, say, AAC, no longer allow more and more more respondents distinguish CDs from MP3s in blind tests. And here the psychological factor comes to the rescue; I have repeatedly read and observed from my own experience that long-term listening to MP3 is more tiring than listening to a CD. Although, I repeat, with a fleeting comparison the difference may not be noticeable.
Do you think this could happen with “cold transistor” sound? Why not? But a simple argument comes to the aid of “digital” apologists - they say, the lamp “smears the sound.” You can’t argue against this, the transistor is “faster”. However, by choosing the right amplifier, you can “touch up” the sound without losing much in dynamics. Expensive models are almost completely free from the disease of “stringy” sound. Let's say the author of this article had the opportunity to compare a tube and transistor amplifier from STAX. Yes, the “tube” plays a little less “fast”, but in combination with the “fast” STAX headphones you hardly notice this. Therefore, I don’t have any prejudices regarding the “warm tube sound”, everyone selects the sound they want to get and does it the way they want to do it.
Of course, we should not forget about the possible placebo effect. Believing that there are changes, especially if they are noticeable only over a long period of time, is not so difficult. But sometimes it’s more pleasant to see a carrot in front of your nose and trail behind it than to be kicked by a boot and move in the right direction. So dreams of the “right” sound probably help many people somehow escape from life’s troubles and immerse themselves in an illusory world. But don’t all one hundred percent of the world’s population do this?
I would also like to add that in my materials I do not focus the reader’s attention on subtle nuances, as well as on details that I am not sure about. What is important, in my opinion, are those moments that are heard almost immediately. Some details emerge during prolonged listening and form the final assessment, but I won’t risk listening to, say, the results of the warm-up - the gap between listening is too long, it’s easy to make a mistake in the assessment.
So don’t worry, I won’t overload my articles with subtle matters. As for the “warm tube sound”, this is far from the monster it is portrayed as on various sites. But I wouldn’t recommend buying some hand-assembled “lamp” from a shaggy year of manufacture and drawing conclusions from it about all products with lamps inside. Ruin the whole impression for yourself. These are toys for those who know why they should play.
Finally, I note that for the first time at the end of the article I add the address of my Email and twitter - write what you would like to read in future issues. I hope that communication in this “live” format will be productive.
Ilya Tarakanov (
This article is an informal continuation of the previous one: “Analog vs Digital: the fight that never happened.” Since the above article had a wide response (and still does ;), I decided to develop this topic further. I will not give any graphs, as in the previous article - as it turned out, for the general reader this is in some cases even harmful. I’ll talk about the topic of digital sound in a more casual manner, trying to touch on topics that were not well covered in the previous essay.
Next I'll be in large quantities use the word "audiophile". I note that this word is used mainly as a diagnosis - this article will be no exception. A person who appreciates high quality sound and understands it, it is usually called a music lover. But audiophilia is an addiction to supposedly “high-quality” sound, based on myths, legends and, as a rule, a lack of personal experience and knowledge.
What is "quality sound"?
The funniest thing in the entire history of debates about various sound reproduction technologies is that there is simply no exact definition of “high-quality sound”.
Let's start with the fact that the same sound can be of high quality for one individual and completely low-quality for another. For example, someone loves bass more and suffers from its lack. But someone, on the contrary, likes “strong” highs - and if they are “soft”, then discomfort arises when listening. What's even more interesting is that these preferences for certain ranges can change over time, even within the same person. This all happens because the human ear is a rather subjective instrument for perceiving sound. The ear can “adjust” to the sound, thereby cleverly deceiving its owner (here one immediately recalls red gold cables made using the latest nanotechnology).
The “hearing tests” that audiophiles rave about are essentially subject to wild errors and generally cannot be seriously considered as reliable evidence of the “badness” or “goodness” of sound. It is impossible to enter the same water twice - it is equally impossible to hear the same sound, even from the same speaker.
Further, any sound reproducing system will a priori distort the original sound. The sound was distorted during recording, then during processing, and then in the amplification paths and speaker system. There is no way it can be 100% consistent with the original one for the simple reason that ideal recording/playback technology does not exist (and it is unlikely that it will ever appear). Moreover: after recording, the sound is deliberately distorted to obtain one effect or another. The number of treatments through which sound passes in modern recording studios amounts to dozens. As a result, everything turns out beautifully - just like in the picture of a Hollywood movie, which is 99% far from reality. But nevertheless, everything sounds quite good (unless, of course, the sound engineer was a layman). Therefore, you should take it into your head: the sound in the final track is purified and refined. Moreover, refined, not with the aim of worsening it, but vice versa.
As a rule, the required sound reproduction system is selected very simply: by sound. You turn on the system and hear a sound that you either like or don't like. Looking for “transparency”, “warmth”, “volume” is pure audiophilia, which in this case does not lead to anything good. You either like the sound of the system or you don’t - it’s simple. And interestingly, as the cost of the system increases, the sound usually improves. Is this strange or not? It seems to me - not very much.
Of course, people with high sound requirements choose a system in more detail. For example, for this case I have several tracks with me - a couple of listens - and everything becomes clear. No amplifier has an ideal frequency response - which means you need to choose the one that sounds most pleasant (ultimately, it all comes down to how acoustic system reproduces well certain frequencies necessary for an individual to listen comfortably). Moreover, an amplifier with an ideal frequency response in a subjective test will most likely lose to an amplifier that reproduces certain frequencies with greater gain (or, conversely, suppresses them) - as they say, who cares what.
Today, the world of audio is dominated by digital technology. This should not surprise a specialist in this field: digital is an excellent way to save and reproduce sound. In a way that is much more advanced than the methods that existed before it. However, as is the case with all relatively new technologies (although the figure is no longer “new”), digital technologies still receive little deserved criticism. “Critics” are mainly divided into two camps: people who are savvy in theory - and, accordingly, those who are not savvy and have no experience at all. The former (apparently due to pathological conservatism and personal preferences) invent myths that can influence the latter. The latter are happy to spread these myths and argue until they drop at conferences, without understanding the essence of the subject as such. With all this, without changing in any way the fact that everything around is digitized and will no longer be converted back to analogue.
In general, I am not an ardent defender digital technologies sound recording/playback (on this moment, this is not required). I had to hear both analog and digital. Naturally, they sound different. But who said that analog sounds better? This is completely unprovable. The main advantage of digital is its replicability and eternity, enormous post-processing capabilities. And the digital sound, forgive me audiophiles, is no worse than the analogue. More precisely, it sounds better.
In the previous article, I did not cover some of the “cherished” topics with which typical audiophiles try to “destroy” digital. The topics are, in general, hackneyed and made up. I will try to analyze them here in more detail and hope that I will be able to arrange the most understandable educational program.
Dynamic range
"Dynamic range!!!" - the first cry with which an audiophile rushes into the embrasure of controversy. Absolutely all the audiophiles with whom I spoke on the topic of sound mentioned these two words. And absolutely all of them did not really know the true meaning of these words and the real picture of the matter.
Roughly speaking, dynamic range is the difference between the quietest and loudest sound. IN general case, the larger it is, the better: after all, this means that the system can record both very loud and extremely quiet sounds with equal quality. The dynamic range calculated for a CD "by mathematics" is about 96 dB. The dynamic range of the best analog media (without noise reduction) is 50-60 dB. In total, it seems like there is a gain of 30-40 dB for the figure (which is extremely large), but it’s not that simple. The fact is that below the range of 50-55 dB, the nonlinear distortion coefficient of a CD increases. That is, the dynamic range of analogue is limited by noise in which sound is lost. And the figure (in its CD version) has acceptable distortions. It turns out that the dynamic range in both cases is approximately the same (and the figure does not lose, even at this stage of the argument). However, there are several nuances.
First point. What is better: when the sound is completely hidden in the noise, or when it is still heard through the noise? However, let better sound will be, than it will not be.
Second point. Sound at -50 dB is almost inaudible. Non-believers can try to normalize any sound file to -50 dB in some editor and listen (of course, you don’t need to turn the volume up to maximum - let it remain at the normal level). That is, somewhere there, beyond the -50 dB range, distortion occurs in the CD. It’s just that there’s no way to hear them - that’s the catch, no one simply records music at this level - in this volume range you can only hear the aftersound at the end of the track. Well, with analog media there is just noise, that’s all.
Third point. Audio science has long known about nonlinear distortions at low signal levels in CDs (quantization noise). And for a long time there has been a technology that allows these distortions to be masked (dither). This technology is used in the process of creating AudioCD. In fact, dither is invisible (due to the fact that it affects low levels that are not audible anyway). But you can do a fun experiment: dither on an 8-bit file! In this case, distortion will practically disappear (though due to increased noise), despite the low bit resolution. Thus, distortions in a virtually inaudible range of levels can also be effectively masked!
And the last, fourth point: all these “terrible limitations” of dynamic range apply only to CDs. Studios have long been recording and processing with a bit resolution of at least 18 bits (usually 24 bits). 24-bit offers a dynamic range of over 140 dB, leaving all analog technologies far behind. Now it’s difficult to say which format will steadily replace AudioCD, but we can say for sure that it will not be 16-bit resolution. However, so far the majority are satisfied with even AudioCD - based on the above, I don’t see anything strange in this.
Thus, fairy tales about the limited dynamic range of digital images are nothing more than fairy tales. Which, firstly, are tied to a specific AudioCD format, and secondly, even AudioCD’s range is fine.
There's been some controversy in the comments about the CD's DD, so I'll give further clarification here. The fact is that digital technologies are so advanced that the practical (namely achievable) dynamic range of a CD (16 bits) is about 120 dB! Using dither and noise shaping, having at your disposal an original 24-bit audio file, you can make a 16-bit file where signal levels of -100 dB and lower will be heard. The price to pay for this will be noise, which will make a recording at this level not only of poor quality, but simply unsuitable for listening. But the fact is the fact: the dynamic range of a CD with the use of tricks is simply enormous. Another thing is that no one really needs him like that. Firstly, sounds at levels less than -50 dB are practically never found in recordings (except perhaps for the “fading” of tracks or relatively rare classical works), because this is a very “quiet” zone. Well, secondly, the noise that appears from dither-shaping is also not a gift. All the inexperienced reader needs to know: the dynamic range of a CD exceeds any analogue sound carrier released in the “pre-digital” era.
Jitter
Jitter - instability of the sampling frequency. Can occur both during recording and playback. Audiophiles have long been accustomed to scaring others with the dreaded word “jitter.” In fact, it's simple. Jitter occurs in low-quality ADCs/DACs - that is, in cheap, household and non-professional ones. And in expensive ones - professional and high-quality ones - there is no jitter. That's all, actually.
Most often, jitter is found in cheap sound cards for computers. The sound card must reproduce sound at completely different sampling rates (typically from 8 to 48 kHz). Naturally, no one will insert into it a dozen stable generators for different frequencies. They will make one generator, and all the necessary frequencies will be obtained using a frequency synthesizer, which will skip some of the pulses and thus generate an unstable sampling frequency (producing jitter).
Calling jitter “one of the problems” of digital audio is the same as calling an MK-60 cassette tape a problem with analogue audio. If you know what I mean. ;)
Recording level
We often hear that due to “problems with digital overload,” sound engineers “lower the recording level” to a margin of as much as 12-16 dB. Which, naturally, leads to an increase in quantization errors, corresponding to signal distortion, as well as a decrease in the dynamic range. Just a couple of nuances are enough to destroy this myth.
Firstly, nowadays no one writes in 16 bits (and it is for this bit resolution that underestimating the recording level will be a problem). That is, the problem may have existed in the 90s with people trying to record something on sound card class SB16.
Secondly, even when I recorded in 16 bits, I never left such a huge margin and did not lower the level to such a minuscule level. Simply because there is no need to do this: you need to adjust the recording level to -3 -4 dB and record for your own pleasure. In addition, when I was working with 16 bits, I was doing track-by-track recording: each instrument had its own track (this is the usual scheme). With this scheme, even at 16 bits everything turns out delicious: each instrument is recorded with a large dynamic range (since “no one bothers” the instrument). In the final mix, the instruments were mixed and the actual dynamic range was greater than could be achieved with a "all in a bunch" recording.
This Kotelnikov of yours is just a theory
We often read discussions that the practical application of Kotelnikov’s theorem, on the basis of which the recording and playback of digital sound is made, faces obvious problems - which supposedly does digital audio"doesn't stand up to any criticism." Problems do exist: both recording and playback of figures encounter pitfalls. The only question is that these pebbles are the size of a speck of dust, given the finite resolution of the human ear, which is simply not able to calculate these pebbles. And the description of the “problems”, as always, is based on the bare AudioCD format - as if others do not exist. The matter, as a rule, is further complicated by the fact that audiophiles present their damning “hearing tests” obtained from the JingHuang mp3 player into the Genius speakers.
When recording audio, the main problem that arises is the limitation of the input spectrum of the signal. If this is not done, then frequencies above the limit (22.05 kHz for AudioCD) will “creep” down during digitization, creating low-frequency distortion (aliasing). Signal filtering is a non-trivial process and, in general, it will still not be possible to completely filter out the entire HF spectrum above the desired frequency without significant distortion of useful frequencies. However, the problem is easily solved by using higher sampling rates (oversampling) - both during recording and during processing. For example, 88.2 kHz instead of the traditional 44.1 kHz (at studios, hardly anyone in their right mind still writes at 44.1). At a sampling frequency of 88.2 kHz, the cutoff frequency of the input signal is 44.1 kHz, which allows the design of filters low frequencies more “relaxed”, given that the desired range is ultimately the frequency range up to ~20 kHz.
When playing digital audio, problems arise with interpolation: it is necessary to restore the original signal as accurately as possible. Again, the problem is often solved by software increasing the sampling frequency (upsampling). Here, audiophiles will happily cry out that software interpolation requires great billions of operations and that no computer is capable of this. Ideally, yes, but in reality, you can apply very simplified formulas that are sufficient to restore a signal with a quality that analog media has never dreamed of. An example with graphs for this matter is given in the previous article, which shows how accurately the signal is restored even for the AudioCD format (which is traditionally kicked with audiophile feet). Let me also clarify that I did not steal those graphs from somewhere on the Internet, but built them myself - using my own program for modeling digital signal processing systems sDCAD. Fortunately, a supercomputer with billions of operations was not required for this.
Flat sound
Audiophiles often hear the term “flat sound” in relation to digital. The term can vary: “plastic”, “artificial”, “non-living” and the like. How exactly does analog sound differ from digital sound?
Firstly, analogue is characterized by softness (blockage) in the reproduction of high frequencies. The softness arises from the banal shortcomings of analog technologies. In the case of vinyl, this is the inertia of the needle. In the case of magnetic tapes, gradual demagnetization (occurs immediately after recording). In short, the analogue sounds soft and delicate (however, the softness is due to the “chewing” of the high frequencies).
The numbers are a different matter: what you wrote down is what you received. If the sound reproduction path is of high quality, we hear what was recorded and nothing is lost. Some digital tracks sound very hard because they were recorded that way - there is nothing surprising here; not every sound engineer likes softness. Especially considering the trends in modern music, where it is common to distort everything possible, including the vocalist’s voice. But the fact is that a digital sound can also reproduce soft sound - you just need to record it appropriately.
Listeners of the “old school” are accustomed to hearing juicy “hooting” bass from vinyl or tape, which appears due to the natural blockage of high frequencies and the accompanying emphasis of low frequencies against this background. With the advent of digital technologies, sound engineers were able to efficiently operate with the entire spectrum, as a result of which recordings became more saturated in high-frequency terms. And they really sound more attractive than the old ones - if you put aside prejudices. However, to get a powerful “hooting” bass, it’s enough to do a simple operation: turn up the lows. If, of course, your music center is equipped with an equalizer at all...
In general, the advent of digital sound recording technologies has changed the sound itself that we hear from tracks. Is there anything surprising about this? Don't think. Is digital sound bad? No, digital sound is good. When used correctly, just like with everything else.
It would also be right to end all rants by mentioning the fact: in disputes about sound technology, it is common to forget about the music itself. We still listen to, for example, early Beatles records and enjoy it. Despite the fact that these recordings were made on washboards, it is hardly possible for an unprepared person to fully imagine the progress in the field of sound technology that has occurred since those times. Each musician has his own view on conveying ideas, and take my word for it, the last thing we think about is warm tube sound and spherical vinyl in a vacuum. The last thing a musician thinks about is that someone will listen to his recording with golden wires and speakers, around which a shaman with a tambourine previously danced, having previously sharpened the pickup needle in a pyramid. The musician thinks about conveying his idea to the listener. Knowing full well that in 90% of cases his music will be listened to on very budget equipment, which often does not withstand any criticism.
And then, for thirty years now the world has been under the rule of synthesized sound. Sound that does not come from live instruments, but from a variety of electronic devices. And the concept of “flat sound” regarding an electronic instrument cannot exist at all. Who said that the synthesizer sound we hear on a recording should sound something different?
It seems that I have covered all the topics not covered in the previous article. Have questions? Welcome to the comments.
Musical addition to the article: “Vinyl” (not recommended for people without a sense of humor and audiophiles)
