Phasing of acoustic systems. Test fragments (tracks) for checking and tuning audio equipment Phasing acoustic speakers

Polarity when connecting speakers must be taken into account so that they work in phase. In other words - consistent.

Many people probably know that some speakers have polarity indicated on their terminals (“ + " And " - »).

In theory, when connecting speakers, you don’t have to observe the polarity, nothing bad will happen. Especially when it comes to monophonic sound reproduction. Old TVs, for example, only had one speaker.

But with stereophonic, quadraphonic or pseudo-quadraphonic sound reproduction, in-phase activation of the speakers plays a very important role.

For example, in all modern car radios, the connecting wires to which the loudspeakers (speakers) are connected have a color pair - 2 wires per channel.

But on one of the paired wires the insulation has a white or black stripe against the background of the main color of the insulation. This was done so that when connecting acoustics, it is correctly connected to the audio output and maintained in phase.

This is also indicated on the connection diagram, which is on the body of the car CD/MP3 receiver.

When connecting speakers, this feature should be taken into account. It is necessary that the diffusers of each speaker oscillate synchronously. Here's how it can be depicted in the picture.

If the speakers do not work synchronously, then the sound picture may be distorted, since the sound waves will be in antiphase - partially compensating each other. In this case, distortion of the spatial orientation of the sound source, etc., is possible.

And if the speaker diffusers operate in phase, the sound vibrations will add up, thereby creating a complete and accurate sound picture.

If you need to connect several speakers in parallel, then their pluses “+” are connected together and connected to the wire that corresponds to the “+” audio output of the device. We do the same with negative terminals.

The phasing when connecting speakers in series is slightly different. Here we proceed by analogy, as with

So, it is required to correctly phase the frequency bands in the speakers. To begin with, I would like to note that it is advisable to have filters at the junction of the same order - in this case, their phase characteristics are generally “parallel”, providing a fairly stable value of the electrical mismatch. Otherwise, it will change greatly in the band of joint sound and talking about successful mutual phasing throughout the entire band is much more difficult, because there will always be a section where the bands will “interfere” with each other.

Let us assume that the filter is designed qualitatively and that the mutual mismatch is fairly constant. We have two options for phasing: forward and reverse - in one of the options, the vectors will be added rather than subtracted - this is exactly the phasing that will be correct. By the way, this is the main criterion - with correct phasing, the overall sensitivity of the speaker increases.

The technology itself:

We start with the bass/mid transition (let the speakers have 3 bands). Turn off the tweeter (HF) and try the options. When turned on correctly, the mids sound better, but the main criterion is sensitivity, i.e. A “comfortable” volume in a room is achieved with a lower volume control on the amplifier, and this is usually a noticeable value.

Next, we connect the tweeters and phase the second transition in general according to the same scheme. It’s just that for this transition there is another criterion that allows you to immediately tell during normal listening whether the phasing is correct. If the phasing is disrupted, then there is a certain “border” vertically in the sound - a kind of plane. When passing through this “plane” the sound changes depending on whether it is “above” or “below”. Since incorrect phasing is a fairly common mistake, you often see the suffering of the owners of the acoustics, who tilt it back in every possible way, trying to combat this effect. With correct phasing, such a “border” in the sound, as a rule, is absent...

Well, one more very intuitive, but subjectively understandable criterion. Correctly phased speakers sound "as if there was just one big speaker under the false panel" - i.e. the imagination may well complete the “picture” and the image will turn out to be stable. In the event of phasing errors, this image falls apart almost immediately. With correct phasing, the sound is “simple”. If the phasing is incorrect, it can be described as “outlandish”, sometimes “spectacular” in some way, but not “simple”. However, “simplicity” usually costs a lot - a higher quality amplifier also sounds “simpler”...

With correct phasing, the sound is “open”, namely, it creates the impression that the phonogram is sounding outside the speakers - “hanging in the air”. In case of incorrect phasing, the sound, one way or another, is hidden “inside” the box - it may be subjectively liked for its “effectiveness,” but it will come from “from within.”

That’s probably all the criteria + methodology. More precisely, there are no instruments yet... By the way, when using 2nd order filters, the phase at the frequency transition needs to be changed - if all the bands are set in phase (and this is almost always the case), then it is usually necessary to “reverse” the polarity in the middle (MF).

1716

Graphite resistors on wood in the output amplifier on 6p9 pentodes, absolute phasing

Phasing of the output amplifier relative to the speakers

Chute = As far as I remember, there are no signals on the Freedom Disk (FSQ) for absolute phasing, only for relative phasing - on the full signal and in different bands separately.

There are several different discs, the one I have, signed by Svoboda, has an absolute phasing test. For relative phasing, no tests are needed; the acoustics are phased simply and without any tests. Or rather, it must be phased at the manufacturer.

Aovox = Tell me, what is absolute phasing? This is clear between channels. Yes! and respect to Yuri for the correct approach to connecting the output amplifier with 6p9 tubes and a speaker cable without heavy output connectors from “you don’t understand” what. Yuri, what kind of wires are there for the acoustics, similar to Audio Quest? Is this by any chance a mono-core lighting fixture? By the way, he plays decently.

A wire from the past, when I knew nothing about the influence of wires, and therefore bought something from inexpensive professional ones. This is a Japanese Canare, four wires of 1.25 sq. mm each, the total cross-section is 5 sq. mm.

Relative phasing is understandable and elementary, but absolute phasing is not much more complicated. Simply, the acoustics are first phased relative to each other so that either + or - goes out to the same terminals equally for two speakers. Then they, phased in this way, need to be phased relative to the terminals of the output amplifier. Here, too, only two options are possible, this is exactly what it is - absolute phasing. The sound at different absolute phasing is strictly different and it is impossible to confuse them.

Alexander Web = Absolute phasing is done between two speakers, starting with the woofers. When a signal is applied to them, their diffusers must work strictly in one direction and either retract or push out. I do this by applying voltage from a 1.5 Volt battery, alternately on the left and then on the right channel. When the diffuser is pushed out, I assign a plus sign to the corresponding terminal. In many industrial speakers, this phasing is disrupted, which results in low bass quality. Next, the midrange and high-frequency heads are phased, but this procedure is more complicated; you either need to do it by ear, or build the total frequency response of the entire speaker, for the left and right channels separately.

This is precisely relative phasing, but you don’t know about absolute phasing.

Masala-chai = I would add that the effect of incorrect absolute phasing, in addition to the strange bass, is also in the periodically floating sound, it either clings to the acoustics, then suddenly breaks away. With correct phasing, the entire stage “stands” without falling apart during the course of the play, and nothing interferes with it. Absolute phasing - the growth of the wave corresponds to the growth of the output. Antiphase - the growth of the wave at the input corresponds to its decay at the output.

Absolutely accurate, besides - the scene blurs in all directions, like a doughnut. KIZs become unclearly expressed, can separate and “flutter” back and forth, and the bass level drops noticeably. The sound takes on a scattered, uncollected character, distortion increases, the stage flattens, and its 3-dimensional three-dimensional appearance disappears. The musical space is scattered and not dense, the sound takes on an unnatural character, the naturalness of the sound is lost - in general, a complete finish.

Freedom had several disks for testing cars, motorcycles and audio equipment. Moreover, there are also different release times for all of them. As I suspect, the composition of the tests in them is different. But the one I have - it perfectly determines the correctness of absolute phasing based on the quality of reproduction of low-frequency signals. Although, personally, I don’t need any tests, I can hear perfectly well when the phase is incorrect - it’s simply impossible to mix it up.

Chute = I’m not sure that the correct absolute phasing can be heard so clearly and unconditionally. We once discussed this issue on the AML forum and came to the conclusion that it is not the absolute phase that is heard, which depends on the number of amplification stages, and may be different in different recordings. But rather, the mutual orientation of the conductors of the secondary winding of the output transformer, the speaker coil and speaker wires. A change in the absolute phase of the recording and the source may not be noticeable; it must be caught with instruments.

I have a CD-R with a recording of one of the test discs, given to me by Svoboda when I attended his master classes. Also, this phasing test had no effect on my system. So perhaps you are right and this was a test not for absolute phasing, but for relative one. Then I don't understand why it is needed at all. Typically, speaker wires are colored (as are speaker terminals) and it is not difficult to phase two speakers correctly. In absolute phasing, it is impossible to understand the correct switching without listening. I know one thing - for my system with absolute phasing, this test does not work.

Bertie Wooster = There's something I didn't understand about phasing. If the speakers in the column are initially soldered in phase according to the markings on the speaker terminals + and -, and then connected to the speaker terminals in the same way, there are no questions. It's the same whether it's left or right. Output amplifier: Front at the input - cascade (inversion), another cascade (more inversion) - front at the output. Unless you add another cascade to the channel, it will be the same. If the transformers are wound the same way, it will be the same. And if you solder not randomly, but in the same way, it will be the same. But if the directions are not taken into account (wires, windings, radio elements, etc.) - then there will be differences. But this is no longer quite a phase.

So I never said that I associate this effect specifically with the phase of the signal. There could be many reasons for this. One of them lies on the surface - when rephasing, the direction of the wires changes - this alone can cause a change in sound. But, probably, not the only one, since the effect is very noticeable and it’s impossible to attribute everything to this alone. Here the processes are more complicated, and perhaps you shouldn’t bother with these processes. You just need to take it for granted and use it for good. Well, thank the Lord that he gave us another instrument with which we can radically change the sound.

It is equally possible to connect the wires from the speakers to the output amplifier in two different ways. With different inputs, the sound changes dramatically. For example, let the speaker terminals be red and black, and the output amplifier terminals too - one red, the other black. Wires can connect red to red, black to black - this is one option. Or vice versa - red with black, and black with red. By the way, this changes the connection between the secondary winding of the transformer and the speaker coil. The beginning and end of its windings are connected differently, here is another possible reason for the change in sound. But you never know what else can happen and change. By the way, the phase also changes, this is one of the possible reasons.

Graphite resistors

Abbas advised installing anti-ringing resistors made of graphite rather than constantan in the output amplifier using 6p9 pentodes in the lamp grid. Previously, I had already made such a resistor on mahogany wood with a diameter of 8 mm with a thick layer of graphite, it turned out to be 0.5 kOhm. Now I had a rod with a diameter of 11 mm made of American walnut at hand, and applied a thin layer of graphite to it. I measured it, it turned out to be as much as 7 kOhm - it won’t work. It is necessary to select the diameter of the mandrel, the thickness of the graphite layer and its length in such a way as to achieve low resistance. Well, these ones can be wound with constantan. You need to work with a layer, without it you won’t be able to wind a 0.12 mm wire turn to turn. A graphite resistor for low resistance should be on a mandrel of large diameter, short length and made of a sufficiently thick layer of graphite.

It turns out that such resistors can be made with tens and hundreds of kOhms on mandrels of small diameter, long length and from a thin layer of graphite. You should definitely try them in the tract to see how they sound. As my intuition tells me, they should sound good, better than many factory resistors. All you need to do is try it.

Attempt No. 3. Wood material - maple, diameter 18.5 mm, length 6 mm. While the graphite layer is small, when it dries, I measure it. If there is a lot of resistance, I will add more graphite. It should turn out as much graphite as you need, so we’ll add that much. If necessary, the diameter can be increased. I plan to grind the top layer of graphite with a zero polish, because... it must be ideal, current flows through it first.

For anti-ringing resistors, a resistance of 100 Ohm to 1 kOhm is recommended, depending on the excitation frequency. I think that 400-600 Ohms is an average universal rating. I will make the previously manufactured RCA output amplifier connectors from wood and brass wire “male” with an improved design, taking into account existing operating experience. With those that I currently have on interconnect wires (old design) there are some reliability problems. Replacing them with power ones for low voltages, I think, is also justified, especially if you remove a little plastic from them. Now they are working for me in an experimental output amplifier using 6P9 pentodes. I don’t notice any particular negativity; after all the adjustments and tweaks, it sounds awesome. Perhaps I’ll try anti-ringing resistors in this output amplifier, otherwise the high female voice could be adjusted a little, a little more. It is the graphite in anti-ringing resistors that should help with this.

Dima = To be honest, I’m not in the know - I forgot (and didn’t really know) where to use them, tell me, and I’ll think about it for a couple of days, remember and write down the denominations.

In both lamp grids, especially with high transconductance, the output amplifier is based on pentodes. Moreover, in the second you can immediately set 100-150 Ohms, but with the first you will have to experiment. I think that about 500 Ohms should be enough for the first grid.

Attempt No. 3 failed miserably... I went the wrong way, I had to go in the opposite direction. The resistance turned out to be 20 kOhm. But the attempt was not in vain, now I know that by lengthening this resistor several times and reducing the thickness of the graphite layer by grinding, it is possible to make resistors of more than 100 kOhm. Especially if you increase the diameter of the mandrel - perhaps then you can get to 0.5 mOhm.

Attempt No. 4. Mahogany diameter 7 mm, resistor length 5 mm, drying.

Listening

Since I’m still a child, I still decided to check what my pentode output amplifier is capable of at a maximum power of 2 Watts. I installed the GOA X electronic chisel, turned it on, I almost got carried away with the chair, I was definitely pressed into it. The bass is so scary that it makes the hair on your head stand up. The whole apartment is sounded and probably the neighbors are on the 2nd floors below me, I live on the last 5th. To avoid excesses in the form of fights and showdowns, I didn’t listen for a long time and after five minutes I turned off the output amplifier. In general, I’ll tell you - 2 watts with an acoustic sensitivity of about 102 dB, this is something terrible. And this is from a single-ended single-stage amplifier with just one 6p9 tube. Moreover, the control at the low end from the output amplifier over the speaker is complete. The movement of the diffuser is already visually visible a little, on particularly “heavy” things - under 1 mm.

Test audio tracks (tracks)

for testing and setting up household and car acoustics and audio equipment
Description, instructions for using test audio discs "Audio Doctor FSQ" and "Audio Doctor FSQ 2"

Quite often one hears the opinion that " THIS IS MY AUDIO SYSTEM PLAYING JUST GORGEOUS", however, during test listening, something often comes to the surface that is quite difficult to describe in words.
And it happens that the owner of the complex becomes fixated on the fact that his equipment is working “SOMEHOW WRONG” and alterations, corrections and many different, sometimes unnecessary, movements begin.
The first time I encountered test signals was after purchasing the “PRIBOY E-024S” equalizer, which had a built-in generator with fourteen fixed frequencies. That’s when I realized that in addition to sound, overtones are also important. When tested with a test signal of 31 Hz, it turned out that it was at this frequency that window glass began to resonate, and at a frequency of 63 Hz, the plates stacked under the bed began to jingle.
The conclusion suggested itself - test signals or fragments of audio recordings are necessary not only for setting up and testing ultra-high-quality audio equipment and acoustics, but it is far from superfluous to check the audio path of a household complex.
In 2005, the magazine "AUTOZVUK" released an audio disc with test audio fragments and a description of how to use this disc when adjusting the acoustics and the complex as a whole. Although the disc is already quite old, it has not lost its relevance to this day. We offer you recommendations from this test disc from the manufacturer and an online test of your computer audio system, as well as a copy of the disc recoded into WAV format, with virtually no loss of quality from the original CDA format.

To fully check the audio system with test signals, you will need the text included with the disc:

PART I. TECHNICAL TRAILS
Turn on and warm up the audio equipment, put in a disc, arm yourself with the remote control and sit comfortably in your usual place. If you want to increase the accuracy (and therefore reliability) of measurements, try to get a sound level meter, it will significantly simplify the work.
So, first, let's check the headroom for the undistorted volume level, this is one of the most important indicators.
In a well-tuned sound path, even with the volume control fully turned up, no wheezing or other distortion should be heard from the speakers. However, the maximum undistorted volume level of equipment of different classes is different - one can bring down the ceiling, another can only block a loud conversation.
What is the optimum for home conditions?
For reference:
A loud but calm conversation between two or three interlocutors standing next to each other usually reaches 75 - 80 dB.
The average volume level in the stalls of a medium- and large-capacity auditorium from a jazz quartet playing does not exceed 80 - 85 dB, a symphony orchestra (not higher than Forte) 85 - 90 dB, and at a rock concert it can reach a pain threshold of 120 dB. Learn more about volume and audio sources.
Theoretically, you can achieve 120 dB at home, fortunately modern technology allows this. But let's look at things realistically: if you live in an ordinary panel house, where the walls and ceilings rarely have sound insulation of more than 40 - 45 dB, then even completely peaceful neighbors will be forced to call the police.
Therefore, it is generally accepted that the average volume level when listening to music in an apartment is 85 dB. And if your equipment can develop an undistorted volume 10 dB more, i.e. 95 dB, then this is quite enough. If this is not enough for you, you will have to fork out not only for more powerful equipment, but also for additional soundproofing of the room.
If the room requires additional sound insulation of 10 - 12 dB, then it will be provided by a carpet with a thickness of at least 1.5 cm over the entire floor area (if there is parquet or bitumen-based parquet boards) plus pasting at least 75% of the walls with additional absorbing materials (Daekwell , Cotex and similar). In addition, there will be a need to cover the entire ceiling area with gypsum plasterboards at least 1 cm thick.
The undistorted volume level is determined by the first track of the test disc. It plays a musical fragment where the vocal and bass parts are additionally compressed. Gradually increase the volume level from zero until the beginning of overload, when non-linear distortions begin to be clearly heard on the bass and vocals, perceived by the ear as wheezing. This is the limit of the sound path in terms of undistorted volume. Remember this position of the regulator.
To accurately determine the value, you need to use a sound level meter. A small-sized digital device (FWE 33-2055 or similar in functions and overall dimensions) with a weighing filter “C” is very convenient here. The measurement procedure is simple: the sound level meter is installed on a tripod in the place where you usually stand when listening. Without changing the position of the volume control, play track number 15 with a pink noise signal. The device will show the exact value of the undistorted volume level, by which you can judge whether you will disturb your neighbors too much.
The next step is to set the standard volume level. All of the following tracks on the test disc should be played at the same volume level. If you are used to listening to music at a very specific position of the volume control, set it to this mark. If you prefer the 85 dB level mentioned above, use the sound level meter again. Turning on track No. 15, use the amplifier control to set the device scale to 85 dB (don’t forget to turn on “C” on the weighting filter).
If you don’t have a sound level meter, invite two or three friends and ask them, without forcing their voices, to discuss some problem. Focusing on the volume of the conversation and periodically playing track No. 1, try to set the amplifier control to the same volume. The accuracy of this operation depends on your patience.
Remember the position of the volume control and do not change it until the end of listening to the test disc!
On test tracks No. 2 - 4 The phasing of the audio path between the channels is checked.
For reference:
With correct phasing on a monophonic signal, the diffusers (forward and backward) of the left and right channel speakers move synchronously. In this case, the sound image reproduced by the stereo system will be perceived exactly from the middle between the right and left speakers. If the phasing is disrupted and one diffuser lags behind or advances the other, then the sound image in the center blurs, becomes unclear, or even shifts to one side.
On a stereophonic soundtrack, incorrect phasing leads to distortion of the sound perspective. For example, some of the musicians in a symphony orchestra may end up in completely different places. Or a rock vocalist, soloing in the center of the sound stage, will suddenly appear in the corner or even at the back of the stage.
On the AudioDoctor FSQ disc, phasing is determined separately for medium, low and high frequencies. On track No. 2, the voice of the announcer is recorded with the words: “Mid frequencies. Phase". These words should be heard from the center of the sound stage. Then the announcer says: “Mid frequencies. Antiphase." In this case, the announcer's text should be played at a lower volume level and (or) defocused for the listener and (or) shifted to one side or another from the center. If the speaker's voice during the last words sounds louder and focused in the center, then in the mid-frequency region the audio system is acoustically out of phase.
The phasing is checked in the same way in the HF band along track No. 3 and in the LF band - along No. 4. For greater simplicity, or if the home audio system is single-sided, the phasing check is carried out immediately across the entire frequency band using the pink noise signal recorded on track No. 16. The phase signal must be located exactly in the middle of the sound stage.

TREATMENT. If you find that the audio path is out of phase throughout the entire band, change the polarity of the speaker wires on one of the speakers. If antiphase is detected only in one of the bands, the situation is worse. Then pick up a soldering iron or take the speaker (usually this is an acoustic defect) to the workshop.

The presence of interference, rattling, extraneous sounds and noise in the sound path and listening room is checked using test audio fragments No. 5 and No. 6. It is clear that any of the above does not decorate the sound, being superimposed on it in the most inappropriate places. We had to listen to a “duet” of a vocalist with a poorly screwed amplifier cover periodically “singing along” with him. Its contribution to the sound was insignificant, so the owner of the system attributed the change in the performer's well-known voice to poor quality recording and almost threw out the CD. On a real music signal, especially a polyphonic one, when many instruments sound simultaneously, it is difficult to accurately track interfering sounds.
Therefore, for testing on the test disk, a tone (sine wave) signal is used, the frequency of which smoothly changes from the lowest to the highest frequencies (colloquially “sweep tone”). Separately, first for the left and then for the right channels. And here sometimes such “dirt” comes out that you are amazed. There is the rattling of loose glass in a window, bookshelf or sideboard, and much more.

TREATMENT
1. The method of dealing with rattling is clear and does not require special investigation.
2. If on a sweep tone there are overtones of the heads in the speakers or, which is really bad, self-excitation of the amplifier - it smells like a visit to a repair shop. The diagnosis of self-excitation is a spontaneously occurring “hustle” in the sound in the high frequencies, noise that can be heard especially well in pauses between tracks on a disc.
3. Sometimes at high frequencies (above 8 - 10 kHz) a series of soft squeaks, increasing in frequency, are heard. This is not self-excitation of the amplifier, but the effect of the appearance of a standing wave in the system: amplifier output stage - cables - acoustic load. The audio signal returns from the acoustics to the output stage of the amplifier and is combined with a slightly higher frequency sweep tone, which is where the beats arise. On a real sound signal, such a selection of components makes the sound at high frequencies frankly dirty. The fight against this phenomenon is quite simple - changing the length or brand of the speaker cable. Sometimes bad contact in the terminal connections is involved.
4. The most unpleasant symptom is a low-frequency buzz at the very beginning of the sweep tone. In the worst case scenario, it can become so loud that it blocks your ears. Here you will have to “treat” not only the equipment, but also the room. We will learn about what exactly needs to be done from the next track (No. 7), designed to evaluate the low-frequency section of the audio path. Two parameters are determined - the lowest frequency of the operating range and the unevenness of the frequency response up to 150 Hz. The assessment mechanism here is based on one feature of human hearing - good memory and preference for the perception of low-frequency sounds.
And this is psychoacoustics.

For reference:
Try to conduct an experiment at your leisure (if you have two audio frequency generators): feed a signal with a frequency of 5 - 7 kHz to the amplifier. After this, from the second generator with the same level - one frequency, approximately 50 - 80 Hz. You will be surprised: the low-frequency tone will be clearly audible, and the mid-range tone will either disappear completely or be barely noticeable.
This is called the masking effect, proving our ear's preference for bass. The phonogram, first for the left and then for the right channels, contains a recording of a number of fixed sound frequencies in the low-frequency range. First, the announcer will announce that the frequency is 60 Hz. Let's call it “support”. Concentrate and remember its volume level. The announcer will then announce 20Hz, 25Hz, 30Hz and so on. In the vast majority of cases, the frequency of 20 and even 25 Hz will be quieter than the reference one, and then the volume will begin to increase. The first pure low-frequency tone (without distortion and turbulent sip), which coincides in volume with the reference tone, determines the lowest operating frequency of the audio path. Remember it and continue listening. Ideally, the volume of the remaining tones up to 150 Hz should be the same, but in practice dips and spikes in levels are clearly audible. This is the unevenness of the low-frequency section of your system.
Judging by the reviews from users of the test disk, this phonogram is so effective that some amateurs even use it to adjust the bass reflexes in speakers. We would like to note that here our ear surpasses even very cool spectrum analyzers in the accuracy of perception.
But let's return to the possible low-frequency hum that we mentioned above. If it makes your ears clog, select on track No. 7 the frequency at which the maximum hum is observed. It is from this that you will have to “treat” your room. This, as already mentioned, is the result of the interaction of acoustic systems and the room, a standing low-frequency wave. Acousticists call them mods. In any room there are at least three of them (length, width and depth). But if they are close to each other in frequency, which happens with multiple room sizes (1:1, 1:2), then it is extremely difficult to fight them.

TREATMENT in such cases the matter is not easy. Most often, processing is carried out over the entire audio frequency band (it’s easier this way) using sound-absorbing coatings - floor and wall rugs, upholstered furniture. This is usually done taking into account the overall design of the room. You just need to remember that synthetic carpets in winter, when the air in the apartment is dry, can accumulate a significant static charge that can damage the displays on the equipment. The solution is portable humidifiers. Good results are also obtained when laying the ceiling, and sometimes walls, with slabs of dry gypsum plaster (DSG). If it is possible to change the geometry of the room, a very effective way can be to lay an additional wall (0.5 bricks) next to one of the walls, extending 3 - 5 degrees from the existing plane.
Very effective, although also quite expensive, is the use of stretch (“French”) ceilings made of dense fabric. A slanted suspended ceiling in the form of a monolithic sound-absorbing structure also significantly reduces modal resonances. But if the mode frequencies are known exactly (using track No. 7), then it is best to place sound-absorbing sheets near the ceiling and on the walls, the geometric dimensions of which are a multiple of the mode wavelength. For example, to eliminate the 63 Hz mode, two sheets of 1.25 x 1.25 (1/4 wavelength) perforated plywood suspended from frames near the ceiling reduce resonance by 8 to 10 dB.

The unevenness of the frequency response in the region of mid-sound frequencies is most noticeable to our ear, especially sharp surges and dips following each other (pros call them “fence”). To evaluate this parameter by ear, without a spectrum analyzer, use track No. 8. The soundtrack on it is a high-quality stereo recording of the applause of a large number of spectators in the hall. Clapping your hands in a fairly echoey room is equivalent to a diffuse field - noise - evenly distributed across the spectrum.
However, against the background of this monotonous noise, the human ear manages to distinguish the very beginning of the pops (bursts). On an audio path with a linear frequency response, you really hear applause, but if it is uneven (“fence”), it becomes similar to the noise of pouring rain. And the greater the unevenness, the more natural the downpour seems, and individual pops that stand out from the general sound background are in this case perceived as annoying drops knocking heavily on the windowsill.

TREATMENT
The main source of “rain” is acoustics. Manufacturers usually draw horizontal frequency response lines that are pleasing to the eye on packaging boxes, but the reality, especially for speakers with woofers on polypropylene cones, can be simply terrifying.
In addition, multi-band speakers are characterized by “raininess”, and most often the maximum unevenness appears at the junctions between adjacent frequency bands, especially with not very high-quality crossovers. Here there is an incorrect choice of crossover frequencies, and the joint radiation of heads spaced apart from each other at the cutoff frequencies (due to the insufficient cutoff steepness of the filters). The magnetization of the inductor cores also makes a significant contribution.
The treatment method is the most radical - it is better to get rid of such speakers.

Track No. 9 determines the linearity of the stereo image across the width of the sound stage. This is also necessary to check the correct placement of the columns. The phonogram contains seven drum beats, smoothly moving from left to right across the entire width of the stereo image. The impacts are precisely localized in direction, and their movement in space is linear, i.e. the angles between the impacts are the same. The first impact is heard on the first sound plane from the very left edge of the soundstage; the second is a little closer to the middle and a little deeper; the third beat is placed a little further into the sound stage and closer to its middle. The fourth blow should be perceived by the listener exactly from the middle of the sound stage, in depth, on the second or third sound plane. The fifth and sixth beats are similar to the third and second, respectively, but on the right side of the middle of the stage. The seventh strike is in the foreground on the very right side of the scene.

TREATMENT
1. The first blow merges with the second, and the sixth with the seventh - move the speakers apart, they are too close.
2. There is no movement in depth - move the speakers forward.
3. The angles of movement of the impacts are asymmetrical - pay attention to the furniture located next to the speakers or its different sound-absorbing properties. A soft sofa on one side and a polished wardrobe on the other is a sure provocation of such an illness.
4. If the impacts themselves are not very clear in orientation in space (not focused), then there may be two reasons:
- Insufficient resolution of the audio path, most often due to poor quality of the source. This is where the difference between expensive and cheap CD players is especially clearly visible. No less important here are the parameters of the amplifier, in particular its phase-frequency response. Cables, and even more so interconnects, have a very noticeable effect on the music resolution. Very often, the wrong direction of the cable makes the sound faded and smeared. You may not believe it, but when you hear the effect live, you will understand that you were wrong. If, of course, the class of the audio path allows you to hear the difference. And not the least role is played by the speakers, and to a greater extent their design.
- Acoustic treatment of the listening room (not to be confused with sound insulation, which we discussed above). In an insufficiently muffled, boomy room, there are always a lot of reflections in the mid and mid-bass frequencies, blurring the localization of the sound stage, although the sound itself usually has a pleasantly bright, rich character. At the same time, a muted room is always better from a localization point of view, but the sound loses its cheerfulness and becomes dry. It is clear that in such a matter a reasonable compromise is needed, which track No. 9 will help to achieve.
A specific example: listen to a moving drum in a room with varnished parquet flooring, and then again with a carpet occupying 40 - 50% of the floor area. Localization will improve noticeably. And then unfold the carpet and cover 100% of the floor area. Localization will be a little better, but the sound will be drier. The same experiments can be done with wall and ceiling coverings, using the acoustic materials and draperies mentioned above. But don’t get carried away with sound absorption and don’t forget about sound dispersion. It has to be both, in compromise. Good studios always have a large set of sound absorbers in the form of suspended curved or asymmetrical structures that improve the diffuseness of the sound field.

Note: Using track #9, you can also estimate the crosstalk attenuation between channels. As you know, a snare drum has tensioned springs at the bottom, which are clearly audible. If, when moving the drum to the right channel, the after-sound of springs is heard in the left channel even after the fifth or sixth hit, the sound path cannot be considered high-quality. Most often, the amplifier or source is to blame for this, but sometimes the situation can be improved by replacing the interconnect cables.

PART II. MUSICAL MATERIAL
In this part you should be especially careful, since for each of the phonograms you will need to evaluate at least two or three parameters. Study the description of the tracks in advance, then diagnosing the path will be noticeably easier.
And treatment methods will depend on the specific type of components you have, financial capabilities and personal musical preferences.

Track No. 10 determines the microdynamics and depth of the created sound stage. The phonogram is a small musical fragment with two instruments - a double bass and a drum set. The recording is of exceptionally high quality. It was produced in a large music studio using two X-Y condenser microphones, 24-bit/96 kHz. The analog signal was digitized immediately after the microphones and transmitted to the console in digital form.
The drummer and his drum kit are located in the middle of a not very wide sound stage, in its very depths (on the third or fourth sound plane). The double bass player is also located far away, slightly to the left of the drum kit. At the beginning of the fragment, both musicians play very quietly. Nevertheless, their instruments are clearly audible, and the music is perceived clearly, with exceptionally high detail. The sound of the double bass is bright and full-bodied. Even at such a low volume level, you can clearly hear the movement of the musician’s bow along the strings and the light tapping of his fingers on the fingerboard. When playing pizzicato, the double bass sounds clear and distinct, without disturbing boominess or blurriness. The drum hits are full-bodied and elastic. The drummer’s “run” through them literally amazes with its clarity and clarity. The cymbals sound very authentic, both when the musician plays very quietly at the very beginning of the fragment, and at the end when he plays loudly.

Sound rating
1. Unacceptable sound stage depth is considered if the musicians are visually located on a horizontal line between the speakers (that is, in the foreground).
2. With unsatisfactory microdynamics at the very beginning of the phonogram, quiet hits on the drums and cymbals are not audible at all, and bowing on the double bass is difficult to distinguish. Microdynamics can be considered satisfactory if the drums, cymbals and double bass are audible, but in the sound of the double bass you can’t hear the musician’s fingers tapping on the fingerboard and (or) when the double bass player plays with a bow, you clearly don’t hear the “resting” movement of the bow along the strings. And the microdynamics will be good if the double bass player’s fingers are heard clearly and clearly. The sound path has excellent microdynamics and exceptionally high quality if a very quiet rustle is heard (time 1’09”), when the drummer accidentally touches the cymbal with his elbow and immediately clamps it with his hand. You can be proud of this sound path.

Track No. 11 of the test disc determines the naturalness of the sound transmission of the musical attack, as well as the position and focusing of the sound stage in width (in the horizontal plane) and height (in the vertical plane).
The soundtrack shows a fragment of a drum solo. Clearly expressed localization of cymbals in direction and depth allows the listener to correctly and accurately assess the spatial arrangement of all “components” of the drum kit. It was recorded “close-up”, i.e. located close to the listener across the entire width of the sound stage. The sound is bright, full-bodied and beautiful. At the very beginning of the soundtrack, attention should be paid to the musician’s playing. The drums sound bright, with emphasized elasticity and “meatiness”, very dynamic and attractive to the ear. The second part of the soundtrack focuses on the cymbals and hi-hat, the articulatory clarity of their sound and the accuracy of their position in stereo space. The hi-hat is located slightly to the right of the middle of the stage, slightly above the snare drum. When the “interruption” on the cymbals begins, the “second” cymbal is visually perceived by the listener to the right, higher and slightly closer to the hi-hat, the “third” - a little to the left.
Next, the musician’s playing moves to the left, and the next, “fourth” cymbal sounds much to the left and is already noticeably higher than the hi-hat. Then another cymbal is hit, which is heard even further to the left, higher and closer to the listener. Behind it you can hear the “sixth”, perceived slightly higher and deeper than the previous one, and to top it off, the seventh and eighth beats sound almost simultaneously, moved even further away from the listener in depth and located slightly lower than the previous ones. The naturalness of the musical attack is assessed by the first part of the phonogram, the focusing of the cymbals in space - by the second.

Sound rating
1. An unacceptable transmission of attack is considered if the sound of the drums is dull, there is no elasticity and “meatiness”; unacceptable - if the sound of the drums is quite dynamic, but has an element of “cardboard” in the beat.
Treatment: if the midbass and bass lack elasticity and clarity, place the speakers with spikes on marble slabs 3 - 5 cm thick. In nine cases out of ten, the sound will improve.
2. It is considered unacceptable or unacceptable if the sound stage is narrower than the space between the speakers (the extreme right and left plates are shifted to the center), and also clearly below or above the listener’s eye line.
3. It is unacceptable or less acceptable if the cymbals and hi-hat are at the same height (in the vertical plane) or the difference is insignificant (the last hits of the left cymbals are only slightly higher than the hi-hat position).

Track No. 12 evaluates the timbre and musical sound balance. The phonogram is a fragment of a jazz piece with male vocals; the quality of the recording can serve as an example of musical balance of sound. Saxophone, piano, electric guitar, bass guitar and drum kit are located across the entire width of the sound stage, on the first and second sound planes, as if arranged in a line next to the listener. The instruments are spatially clearly oriented in front of the listener, musically balanced among themselves and perceived with the same volume.
There is a piano on the left, a guitar and bass on the right. In the middle of the sound stage, slightly behind the main instruments, there is a drum kit. It is recorded wide, the drums, cymbals and hi-hat are sort of placed along the front plane. In the center, in front of the drum kit, a saxophone is heard a little closer to the listener. While playing, the musician sometimes migrates from the middle a little to the right, and the movement of the saxophone is felt on the recording. Male vocals are heard exactly from the center of the stereo image. At the very beginning of the play, the singer from the back of the stage approaches the microphone - his voice moves from the background to the foreground and “remains” there until the end of the play. Timbrally, the vocals sound soft and full-bodied, with a good content of low components.
It is clear, precise and legible, but in no way harsh. The piano is perceived as full, dynamic, with a bright attack and in several places is accentuated in level. The bass is dense, thick, and has a very pleasant timbre. In the overall sound picture, it is located between the foreground and background and does not protrude forward. The guitar, whose main role in this piece is accompaniment, is also visually located between the first and second sound plane.
Timbral balance (natural sound of instruments) and musical balance (balance between instruments and vocalist in level) are assessed separately by the listener.

Sound rating
1. It is considered unacceptable or hardly acceptable from the point of view of timbre balance if any of the instruments sounds unnatural and (or) if the vocal timbre has a harsh or unpleasant sounding character.
2. It is considered unacceptable or hardly acceptable from the point of view of musical balance if the vocals or any of the musical instruments clearly go out of their sound plane, i.e. clearly stands out in terms of volume (pushed forward) or falls out of the “general rank” in terms of volume (pushed back).

Track No. 13 evaluates the linearity of the sound path in terms of volume level, its macrodynamics and ability to transmit a polyphonic sound image. The phonogram contains a high-quality recording of a symphony orchestra, performed in the Great Hall of the Moscow Conservatory. The recording was initially digital (the audio signal was digitized directly after the microphones) in 24-bit/96 kHz format and, after mastering, was reduced to the standard CD format of 16-bit/44 kHz. According to the sound engineer, the listener should be somewhere in the middle of the hall and feel the overall sound of the orchestra with maximum airiness and volume. Therefore, musicians are perceived as distant from the listener. The fragment consists of four main parts, differing from each other in volume level and dynamics. And the first part, sounding very quietly (pianopianissimo), and the second (piano), and the loud third (forte), and the fourth, final (forte fortissimo), should be perceived equally naturally. The pizzicato of the group of string instruments in the first movement, despite the low volume, should be crisp and clear, the listener can freely and clearly distinguish the “pinch” of the musicians’ fingers. The solo brass instruments in the second part of the phonogram are light, distinct and well localized in their location in the orchestra.
The third, loudest part of this phonogram is not at all easy for the sound path. The orchestra sounds very powerful here. A group of cellos and double basses enters, adding grandeur to the orchestra's sound. To the ear, it seems that the overall picture seems to unfold in front of the listener and visually rises slightly upward. The perception of a large number of string and wind instruments should be polyphonic - remain pure and natural, where not only string and wind groups, but also individual instruments in them, can be clearly and clearly heard. A sound path with good dynamics conveys this part of the soundtrack easily, musically and dynamically. It should not seem “cloudy” and should not merge into a common “cloud” filled with tools.

Sound rating
1. It is considered unacceptable or less acceptable if the string pizzicato in the first part is completely unintelligible or sounds too quiet, sluggish and indistinct in comparison with the next, louder part.
2. It is considered unacceptable or less acceptable if in the third movement (after the entry of a group of cellos and double basses) there is no noticeable jump in volume (forte) and then, in the final, another jump (forte fortissimo), in other words, the sound clearly lacks lightness and dynamics , energy.
3. It can be considered unacceptable or hardly acceptable when in the third and fourth parts of the phonogram non-linear distortions are clearly audible or there are no distortions, but the orchestra clearly does not reach forte fortissimo in volume.
4. It is unacceptable if the orchestra already in the third movement begins to sound like a general “mess”, to merge, individual groups of musical instruments are poorly distinguishable or these groups are completely indistinguishable.

Track 14. An additional track for assessing the sound quality of the lowest sound frequencies. It is needed for sound paths capable of reproducing the lowest bass and operating in acoustically treated rooms. The phonogram contains a nine-second excerpt of the sound of a symphony orchestra, which includes a large (Turkish) drum with a very low register. It can only be heard if you have a high-quality subwoofer that naturally reproduces frequencies of 20 - 25 Hz. For ease of evaluation, the passage is repeated three times in a row, and the Turkish drum enters the track at the 3rd, 17th and 32nd seconds.

Sound rating I
If you feel that from the indicated seconds a clearly discernible low bass punch is added to the sound of the orchestra every measure and the room is not buzzing, you can be heartily congratulated.
For the ON-LINE test of acoustics and audio system, the contents of the "Audio Doctor FSQ" disc are presented, recoded into MP3 with maximum quality.

Acoustics and audio system test online

In 2006, the magazine "AVTOZVUK" released the second version of the disc with test fragments. The second version contained the same 16 tracks for testing and tuning audio equipment, to which additional tracks were added for fine-tuning. Initially, the test fragments were in CDA format, which was the most optimal at the time of the disc's release. However, the gradual extinction of this format forced the conversion of these audio fragments to WAV. Thus, the quality was not affected, but the ability to copy test records to USB flash drives significantly expanded the options for using these tests.
The archive, packed with minimal compression, contains the contents of the "Audio Doctor FSQ 2" disc, which makes the archive more universal, since it contains both versions of the disc.

Well, for a better understanding of what it is, here is a copy of the accompanying text from the “Audio Doctor FSQ 2” test disc, which allows you to more accurately configure the speaker systems and amplifier complex:

FROM THE COMPILERS OF THE TEST DISC "AUDIO DOCTOR-2", DESIGNED FOR CHECKING AND ADJUSTING SPEAKER SYSTEMS AND AUDIO EQUIPMENT:

This disc received its name in the most logical way. The point is not only that this is a new edition of the most popular “test and tuning” CD in our country. The new edition of the disc with test signals for checking acoustics consists of two completely independent parts. Yes, and their compilers are different. In the first part it is Dmitry Svoboda, in the second it is Andrey Elyutin.

PART ONE. DIAGNOSIS AND TREATMENT

Judging by numerous reviews, we underestimated the level of competence of our readers. The first "AudioDoctor" was made on the basis of the judge's disc "Car Audio FSQ" with some simplifications. Thus, we increased the pauses between tracks to leave more time for analyzing what was heard. To check the phasing of the audio path at low, medium and high frequencies, we recorded tracks that were difficult to understand, repeating them twice.

In the second edition of AudioDoctor, it was decided to abandon this and return to the form that was included in the original FSQ test disc. Therefore, “AudioDoctor-2” is designed for fairly experienced listeners, which, apparently, are the majority of our readers. The selection of test tracks itself has not changed, since the original disc has been “polished” over six years to become a generally recognized reliable and proven tool for operational acoustic examination. However, for greater convenience and versatility when setting up the audio path, we decided to add several technical tracks. The content of the first 14 tracks has not undergone any changes, and here we refer the reader to the booklet for the first “Audio Doctor” published on the magazine’s website.

From the description of AudioDoctor FSQ 1
However, I would like to make some comments on these tracks. We have already mentioned a nuance on track 10 (timer 1:07), where the drummer accidentally slightly touched the cymbal with his elbow, but then immediately pressed it with his hand to prevent it from ringing. She managed to ring, very briefly and so quietly that it can only be heard on a sound path with a very high musical resolution. That's why we left this sound on the soundtrack to use it as a test sound.
Judging by the letters from readers, this fragment solves its problems more than successfully. In Krasnoyarsk, for example, two audiophiles, with the help of AudioDoctor, in front of amazed sellers, staged a “review competition” of audio equipment displayed on the counter. Of the wide range of equipment offered there, only a few samples had truly high detail. By the way, not the most expensive products... Another “highlight” of this phonogram (timer 1:47) is the barely audible sound of the drummer jumping up. The recording then was long and difficult, and after this successful take the musician, having completed the last hits on the cymbals, jumped up, full of musical ecstasy. This place turned out to be at the very end of the soundtrack and could easily be cut out by the mixer during mixing. But we decided to leave this sound as a test sound. And if at normal volume levels you can hear both of these sounds, you can be justifiably proud of your equipment. What else is “hidden” in the paths you already know? The well-known phonogram No. 9 consists of a drum “moving” from left to right and serves to determine the linearity of the width of the sound stage. But this same recording is very convenient to use for assessing the transition attenuation between channels of the audio path, a very important parameter, but undeservedly ignored. If, when moving the drum to the right, after-sounds are heard in the places from which it just sounded, be wary - this may be a sign of insufficient cross-fade. If the echo of the seventh beat (the rightmost) is heard not only in the place of the fourth (central), but also the very first (left), then the transition attenuation is clearly insufficient, the stereo image will be blurry and unnatural. The main source of this audio path defect is the power amplifier. And more about this track. The snare drum that sounds in it is structurally equipped with metal springs stretched outside the lower head (usually there are four to six of them). They, of course, contribute to the sound, which is why the frequency range of this instrument extends all the way to the highest frequencies. Experienced FSQ experts can quickly and accurately evaluate the frequency response of the audio path based on this drum sound. Practice, maybe you will succeed... Regarding the assessment of transient attenuation and plus the amplifier's tendency to self-excitation, there is one more piece of advice. If extraneous sounds are heard on the sweep tone being played in one channel (tracks 5 and 6), in another channel (where there is no signal), then you have a problem, you can expect a muddy, modulated nature of the sound and distortion of the sound perspective. This is a consequence of poor crosstalk attenuation between channels and self-excitation of amplifiers. The first is the scourge of hastily made amplifiers, where the audio signal penetrates into the adjacent channel through the general power circuits or due to illiterate layout of printed circuit boards. And self-excitation is already an unscrupulous circuit design. In the worst case, spontaneous self-excitation (usually at HF) can even burn out the tweeter.

Now about the technical tracks.

Track 15 is a recording of uncorrelated pink noise in both channels. Noise is a random process, and in this phonogram the processes in the channels occur independently of each other. To the ear, such noise is perceived as a sound cloud of impressive size hanging in the air. The first edition of "AudioDoctor" also had this track, but with in-phase pink noise. The correlated pink noise is now recorded on track 16 in phase and out of phase. In this case, the signal in the stereo channels is the same, in-phase noise should be focused between the speakers, and anti-phase noise should be smeared in space, trying to “stick” to the acoustics of one or another channel, whichever happens to be closer.
On tracks 17 and 18 pink noise was recorded, filtered by a high-pass filter with a cutoff frequency of 500 Hz, separately for the left and right channels. Practice has shown: it happens that in stereo mode, phasing separately into HF, MF and LF (tracks 2 - 4) gives encouraging results, and listening reveals some unnatural sound. This may be due to the fact that in the left or right channel the LF and HF emitters are out of phase with each other. For more accurate localization of low-frequency risers when placing left and right speaker systems in home spaces, use tracks 19 and 20. They can also be used when setting up stereo subwoofers in car audio (this happens). The tracks are completely similar to track 7, but are recorded separately for the left and right channels.

PART TWO. HEARING TEST

The Scripture also says: “Physician, heal yourself.” For those who wish (we won’t force anyone), we have supplemented the tools for setting up sound systems with a set of special audio tracks to assess the noticeability of certain distortions in the sound signal or the listener’s ability to notice them. The specially processed signals for this part of the disc were largely developed by American Arnold Krueger as part of his research into blind comparative listening techniques. The tracks are very short, in order to completely eliminate the factor of short-term auditory memory, but there are a lot of them, so for ease of work, we recorded a CD-text on this disc, and the contents of each track are displayed on the display (if your equipment has this) in abbreviated form - conditionally. When describing tracks, we will show their designation on the display during playback. The degree of noticeability of distortions artificially introduced into phonograms depends on their magnitude and nature and varies from “very easy” to “almost impossible.” This is meant under the condition of an impeccably working tract and a very talented listener. If you wish, you can learn a lot about both by experimenting with different paths (and/or different listeners). Collective exercises with test fragments become especially exciting, when one of the listeners makes up the “blind” part of the audience; the compilers tested this both on themselves and on those who came to hand. In summary, materials for “hearing tests” are divided into three groups: frequency distortions of various mechanisms, nonlinear distortions and noise.

FREQUENCY DISTORTION

Tracks 22-26 contain a phonogram consisting of two series of castanet strikes. In each pair, the first fraction is the original recording, and the second is made through a low-pass filter with a very high slope. You are asked to evaluate your ability to notice the bandwidth limitation of the audio path at the top.

Track 22 cutoff frequency 5 kHz LP 5 kHz very easy
Track 23 cutoff frequency 9 kHz LP 9 kHz easy
Track 24 cutoff frequency 12 kHz LP 12 kHz more difficult
Track 25 cutoff frequency 15 kHz LP 15 kHz difficult
Track 26 cutoff frequency 18 kHz LP 18 kHz very difficult

Track 27, like 21 before it, is used to pause between sections and display information about the upcoming section on the display. Tracks 28-31 contain two sets of snare hits. In each such pair, the first series is the original, reference, the second is recorded through a high-pass filter with a different cutoff frequency. The frequencies are frankly low (some are indecently low), but truly experienced listeners catch this; to be happy, they need a path with a wide low-frequency band.

Track 28 cutoff frequency 50 Hz HP 50 Hz easy
Track 29 cutoff frequency 32 Hz HP 32 Hz more difficult
Track 30 cutoff frequency 20 Hz HP 20 Hz hard
Track 31 cutoff frequency 10 Hz HP 10 Hz almost impossible

Track 32 displays "TILT DOWN". This is what it is: a chord is struck on the brass. On each track, it is first played in its original form, and then - after passing through a path with a frequency response that has a uniform rise to the lower frequencies and the same uniform decline to the upper ones. Like a slanted straight line, the slope you notice will indicate the sensitivity of your hearing to a skew in the overall tonal balance.

Track 33 +5 dB at 20 Hz, -5 dB at 20 kHz Down 10 dB easy
Track 34 +2 dB at 20 Hz, -2 dB at 20 kHz Down 4 dB harder
Track 35 +1 dB at 20 Hz, -1 dB at 20 kHz Down 2 dB hard
Track 36 +0.5 dB at 20 Hz, -0.5 dB at 20 kHz Down 1 dB almost impossible

On the following tracks, the slope of the frequency response is opposite, with a rise to the upper frequencies (TILT UP). Please note: with the same amount of frequency distortion, the degree of noticeability will be different.

Track 38 -5 dB at 20 Hz, +5 dB at 20 kHz Up 10 dB very easy
Track 39 -2 dB at 20 Hz, +2 dB at 20 kHz Up 4 dB easy
Track 40 -1 dB at 20 Hz, +1 dB at 20 kHz Up 2 dB harder
Track 41 -0.5 dB at 20 Hz, +0.5 dB at 20 kHz Up 1 dB difficult

Tracks 43-46 illustrate the noticeability of deep dips in the frequency response. A frequency band centered at 4 kHz is cut out from the original phonogram (using a digital notch with -100 dB attenuation). The width of the cut strip varies, as does the degree of visibility of such vandalism.

Track 43 bandwidth 1/2 octave -1/2 oct. very easy
Track 44 bandwidth 1/3 octave -1/3 oct. easily
Track 45 bandwidth 1/6 octave -1/6 oct. more difficult
Track 46 bandwidth 1/12 octave -1/12 oct. difficult

On tracks 48-51, a burst of constant height (+6 dB) with different widths was created on the frequency response.

Track 48 bandwidth 1/2 octave +1/2 oct. very easy
Track 49 bandwidth 1/3 octave +1/3 oct. easily
Track 50 bandwidth 1/6 octave +1/6 oct. more difficult
Track 51 bandwidth 1/12 octave +1/12 oct. difficult

Tracks 53-56 are devoted to the visibility of a constant width dip in the frequency response. A dip around the same frequency of 4 kHz is created using a parametric equalizer with a quality factor of Q = 0.5, this means a bandwidth of approximately two octaves, and the depth of the dip varies.

Track 53 - 3 dB at 4 kHz -3 dB easy
Track 54 - 1 dB at 4 kHz -1 dB harder
Track 55 - 0.6 dB at 4 kHz -0.6 dB difficult
Track 56 - 0.4 dB at 4 kHz -0.4 dB almost impossible

NONLINEAR DISTORTION

Track 58 records a 1 kHz tone with minimal harmonic distortion. On subsequent ones - with artificially introduced distortions in the form of a typically occurring mixture of harmonics. We're not giving an estimate of difficulty here, but you'll be surprised at how early distortion can begin to be heard on a pure tone.

Track 59 - 0.3% distortion at 1 kHz THD 0.3%
Track 60 - distortion 1.0% at 1 kHz THD 1%
Track 61 - distortion 10% at 1 kHz THD 10%

The following tracks focus on the salience of harmonics in a music signal. A second harmonic is introduced into the short piano fragment, distorting the symmetry of the signal. Please note: it is not that noticeable, even with very high content.

Track 63 - original Piano REF recording
Track 64 - 2nd harmonic, 0.1% 2nd 0.1%
Track 65 - 2nd harmonic, 1% 2nd 1%
Track 66 - 2nd harmonic, 10% 2nd 10%

There is a similar series for the third harmonic, it is much more noticeable.

Track 68 - original Piano REF recording
Track 69 - 3rd harmonic, 0.1% 3d 0.1%
Track 70 - 3rd harmonic, 1% 3d 1%
Track 71 - 3rd harmonic, 10% 3d 10%

A piece of phonogram to which noise is mixed with a predetermined level relative to the signal level.

Track 73 noise level -80 dB -80 dB
Track 74 noise level -70 dB -70 dB
Track 75 noise level -60 dB -60 dB
Track 76 noise level -50 dB -50 dB
Track 77 noise level -40 dB -40 dB
Track 78 noise level -30 dB -30 dB

The graphs show what type and magnitude of frequency distortions were introduced into the phonograms on tracks 33-56. The name of each graph will appear on the display at the beginning of the section (if CD-text is displayed), and the designations of the curves will appear while they are playing.

Descriptions of the tracks of part 2 of the disc are given in the following format:
Track number / content / CD-text / degree of difficulty

From the author of the test disc:
YOU HAVE PROBABLY NOTICED THAT people who are completely satisfied with the sound of their equipment are not very common. Something is always not as we would like, because of which we involuntarily have to listen to the advice of friends and sympathizers. But sound perception is a subjective and highly individual matter, so following such advice is pointless. The result can be paradoxical - by changing the components to others, usually more expensive ones, you will not get the desired peace of mind. Therefore, it is best to deal with your problems yourself, and with proper formulation of the problem, you can achieve success.
As you know, the main thing in the treatment of any disease is the correct diagnosis. The selection of the necessary medications, potions, etc. will come later. If you really want to enjoy music at home, the diagnosis of the sound image in general is as important as in medicine. By the expression “in general” we mean the general perception of music, which depends both on the selection of audio equipment, cables and discs of appropriate quality, and on the acoustic properties of the room itself.
It's no secret that the most beautiful audio equipment at home may not sound. At the same time, in a good, acoustically verified room, the process of selecting components and their correct configuration is greatly simplified.
The "Audio Doctor FSQ" test disk will facilitate diagnostics and correct configuration of the home path. The FSQ technique is designed specifically for objective and subjective acoustic testing. We will describe all the tracks in detail and tell you what you should hear. And, of course, we will carry out a course of treatment if the sound is not all right.

The subjective-statistical method “Fast Sound Quality” (FSQ) was developed at the Acoustic Center of the Department of Broadcasting and Electroacoustics of MTUSI to conduct professional subjective-statistical examinations (testing) to assess the sound quality of the audio path. It allows you to obtain high reliability of results with little investment of expert time. The method includes an optimal selection of objective and subjective parameters that determine sound quality, a test disc with specially selected and recorded phonograms, and a methodological development of listening.
In 2001, the method was adapted to assess sound quality (QQ) in a car interior. An original expert (judicial) protocol was developed and a test disk “Car Audio FSQ” was published. MTUSI has begun training qualified experts capable of conducting sound auditions (judging) in cars.
In 2002, the method was presented in detail at the 21st international conference of the AES (International Society of Acoustical Engineers), and the following year a Car Audio section was organized at the AES.
In 2003, the FSQ method began to be used to evaluate the short circuit of multimedia audio systems and studio professional near-field monitors with its own “Multimedia FSQ” test disk and an expert (judging) protocol.

FOR REFERENCE

THE FSQ METHOD IS DESIGNED FOR PROFESSIONAL EXPERTS, but its accessibility allows it to be used by experienced listeners as well. The main thing is to carefully study the phonograms included in the disc and the approach to evaluating their sound. Don't be discouraged if you don't hear all the sound information the first time - at first it's really not that easy. The most important thing is to fully concentrate on the musical material, do not hesitate to repeat a fragment that is not entirely clear to you several times.
Now about what and how we will listen.
Which is clear, your audio system and exactly in the place in the room where you usually are. We emphasize this specifically, since the sound field in the room is non-uniform, there may be places where hum and cross-reflections occur.
But we will tell you in detail how to listen.
Since a physical quantity that unambiguously describes sound quality does not exist in nature, experts use different kinds of terms. From the simplest and non-specific “better”, “worse” to the more precise “clear”, “blurry”. More correctly, these words are called subjective criteria. There are more than 100 of them and many of them are vague or duplicate each other, which greatly complicates acoustic examinations and sometimes even neutralizes the results. Attempts to unify terminology have been carried out all over the world for decades, but so far they have not been successful.
The FSQ method clearly defines the main and secondary subjective criteria for assessing sound quality. The main ones that will appear on the “Audio Doctor FSQ” test disc include:
Headroom for undistorted volume level.
Correct phasing of the stereophonic audio path.
Uneven amplitude-frequency response (AFC).
Microdynamics of the sound path.
Macrodynamics of the sound path.
Natural timbre balance.
Natural musical balance.
Ability to reproduce linear low frequencies.
Presence of noise and interference.
Linearity of the stereo image across the width of the sound stage.
The width and height of the sound stage, its position (orientation) in the horizontal and vertical planes.
Sound stage depth (separation).
Natural transmission of musical attack.
Linearity of the stereo image at different volume levels.
The ability of the audio path to transmit polyphony.

A solid list, but everything is not as complicated as it seems. The phonograms on which these assessments will be made are very accessible to perception. Don't be embarrassed by the fact that you'll have to stop and listen to parts of Audio Doctor FSQ a lot at first. Practice shows that after four or five listenings, most difficulties disappear.