Scheme of switching on the simplest stereo base expander. Stereo base width regulator - audio and sound - schemes - file catalog - radio schemes magazines repair moding. The principle and methods of implementing a wide sound

To begin with, it should be noted what is a stereo base, and why is it needed? For me, this is one of the components of high-quality sound. This is the volume of your soundtrack. Any composition, mixed in the studio by a professional sound engineer, has a sufficient amplitude of the stereo base, which gives it liveliness and freedom to fly. The sound becomes more transparent and velvety. This is when a sound wave does not propagate in a straight line, but, as it were, covering the entire space. And so, we have decided on the concept of a stereo base. But how to achieve such a sound, is such quality possible only when mixing in a studio? No, this can be achieved at home, and in two ways. The first way is software, that is, using plugins (plug-in). The second way is manual, that is, with the help of a panorama and a little trick.

Method one.

Currently, there are many plug-ins and VST tools that can expand the stereo base of the soundtrack in one way or another. There is such a wonderful tool called Virtual Guitarist (Fig. 1), which imitates the playing of a live guitarist, in which this function is built in, and of course it is applicable only to your own sound. With the stereo knob, you can either expand the sound or downmix it to mono. If the guitar part in the mix serves as a melody, then it is desirable to make the sound<<шире>>, for more volume and beauty. If the guitar part keeps the rhythm, then the sound needs to be made clearer, and try not to resort to this function.

fig.1

fig.2

The CuBaseSX3 sequencer has a built-in synthesizer called the Embracer (Figure 2) in which the stereo base can be expanded in parallel with the tone of the sound. Of all the plug-ins with which you can expand the stereo base of the phonogram, you can especially highlight the Izotope Ozone mastering package, which copes with its duties in the best possible way. He does this by diluting the individual frequencies of the sound wave across the panorama. It is also possible to equalize the sound with a four-band equalizer. For a better understanding of what we are talking about, we will analyze everything with examples:

Example 1
This is what the sound wave looks like on the Izotope Ozone diagram when playing a clean, unprocessed sound. As you can see in the figure, the sound propagates strictly in the center, and there will be no volume to the ear. If, however, separate frequencies across the panorama, and not their entire spectrum, then the difference will be noticeable even to an inexperienced listener.

Example 2
This is how the sound wave looks like on the Izotope Ozone diagram when low and medium frequencies are diluted across the panorama by 10% to the right and left channels. In this case, low and medium frequencies in relation to the main sound actually play at different times, but since they are separated by channels, our ears hear them separately, and the brain already assembles them into a single stereo image. The main thing is not to overdo it. If the stereo base of the mix is ​​too broad, it can lose its clarity and dynamics.

Method two.

If you do not have a suitable plug-in, then you can expand the stereo base manually using any audio editor with multi-track recording mode, or you can take any sequencer. The example uses CuBase SX.
First we need a sound that we will process. Import this track into CuBase (File -> Import -> Audio File). We duplicate the original track, after which we should succeed, as in Figure 3.

fig.3

Next, we take the upper track to the left along the panorama, and the lower one to the right. At the same time, we impose a delay of 10 units on the lower track. But you can deviate from this figure, but the main thing is not to overdo it. That's all, we reduce these tracks into one file (File -> Export -> Audio Mixdown), and only in stereo, otherwise there will be no sense from the work done.

It should be noted the main instruments that need this, and those that absolutely do not need stereo phase expansion.
Guitar.
The guitar is the instrument that is most often subjected to such processing. It does not depend on the type of guitar, be it acoustic or electric guitar, or the style of performance. In rock music, guitar rhythm parts are the basis of rigidity and dynamics, and should cover almost the entire sound area, if it is a lyrical arpeggio part, then the width of the stereo phase emphasizes the beauty and elegance of the melody.
piano.
Expanding the stereo base of a piano does not hurt, but you can do without it. It all depends on the place of the instrument in the mix, and on the style of the music being played.
Violin.
It is desirable that the violin would sound<<широко>>, but here everything again depends on the violin part and the style of the musical composition.
Vocal.
Backing vocals are subject to a mandatory expansion of the stereo base, otherwise they will sound like the main voice, expanding them, we leave room for solo vocals. But you can hear this arrangement on solo vocals as well.
Bass.
It is strictly forbidden to expand the stereo base of the bass guitar, as it is the foundation of the composition, and must be strictly in the center, this applies to live styles. Everything is possible in electronic music, for example, the expansion of the stereo base of the bass is often found in the Trance style, since the drum rhythm section is the basis there.
Drums.
Again, in live music, they should be strictly in the center, you should only breed individual components along the panorama.

This module is as simple as the MP Harmonic Exciter.

Each range has its own stereo expansion control. Level 0 means that no extension is applied in this frequency band. Positive values ​​imply expansion, while negative values ​​represent a "negative expansion" or summation of the channels, bringing them closer to the center.

When working with the stereo channel expander, keep an eye on the controllers on the right. The horizontal bar is the phase correlation meter (or phase meter), and the "radar type" box below it is the vectorscope screen. Both are used to provide information about the channel separation (or "width") of your mix.

Phase Meter

Indicates the degree of similarity or "correlation" (correspondence) between the left and right channels.

When the sound in the left and right channels is similar, the meter "draws" to the right of this strip. When the left and right channels are the same, then the correlation value will be +1, our meter will be moved all the way to the right.

When the left and right channels do not correlate or are very different, the meter walks in the left corner. If the left and right channels are out of phase, in such a situation the correlation is -1, the meter will be completely left. Over time, the phase of the signals in the channels changes, but the bar remains colored to give us a picture of the history of the correlation change. Bright white coloration indicates that the meter has spent more time in that area. Thus, by listening to the mix, you will quickly discover the extremes of the phase correlation as well as the most common areas.

Note that you can "zero" the meter by clicking on the bar.

Most records have a pattern of correlation values ​​between 0 and +1. Brief outliers to the left are not necessarily a significant problem, but may represent a possible mono-incompatibility. You can quickly check if the phases are mono compatible by clicking on the Show Channel Ops additional options. This menu allows you to get the mix in mono, invert the polarity of the left or right channels and swap them.

As stereo broadening is applied, the phase correlation will tend to drift towards the left as the left and right channels become "wider" or less similar.

By default, the phase meter is placed at the end of the signal chain, so you "see what you hear". A useful side effect of this is that if you mute either band, the phase meter only shows stereo correlation for the band(s) you are listening to.

vector scope

Vectorscope also offers a visual image of the signal.

Typically, a stereo recording drawn in Vectorscope is usually taller than wide (as shown in the screenshot above). Vertical shapes mean the left and right channels are similar (approaching a mono signal that is a vertical line). The horizontal shapes of the pattern mean that the two channels are very different, the mix will sound wider, but this can mean mono compatibility problems.

Options in vectorscope:

You can click on the phase meter to update the picture.

If you want to hide this display, you can turn it off in the Options Screen menu.

MP Stereo Delay

We have kept the most interesting part of the stereo display module for the latest version of the program. Ozone offers a stereo delay control that allows you to adjust the delay between the left and right channels. At first glance, such a delay may not bring a qualitative improvement to the mix, but the use of a multi-band delay can produce some very interesting stereo effects.

Of course, you know that in order to make sound come from the right, you must increase the level of the right channel. This is true in the sense that the sound that sounds on the right side is louder in the right ear. But there is another factor as well. The sound coming from the right reaches the right ear first. There is a short delay before it reaches the left ear. If you delay the channel by a few milliseconds, you can effectively move the mix sections around the stereo field.

To experiment with stereo imaging using delays, try moving the delay control to the left or right. By default they are grouped so that they all move together in sync. You should be able to hear how the mix responds and if you're making a live acoustic recording with a pair of stereo mics, this can be an effective tool to adjust the stereo image without shifting the channel-by-channel balance. Thus, delay can be very useful for mastering live recordings. Use it and you will get more experience every time.

This situation is not the only creative use of delay. Click on “Group all band delays” and ungroup the delays. Now you can use the delay between the left and right channels for each band arbitrarily. By moving the delay slider to the right, you will delay the right channel, to the left, respectively, the left channel.

If you expect to hear "echo", you won't hear it. This echo here is very short delay periods from 0 to 30 msec. In this time range, you will not be able to distinguish between two distinct signals delayed in time by as much as you can hear the echo.

So what can you do about MP delay? Try moving the position of the bass using Delay in Band 1. You can move the perceived position of the bass without setting the stereo position. Or try shifting Range 3 to the right and Range 4 to the left for a treble extension effect. Experiment.

02/06/2015 at 12:00

There are many methods for expanding the stereo image in a mix, all of which combine to produce excellent results. This includes panning, reverb, various effects, playing with phase, delay between right and left channels, Mid/Side processing, and even saturation and compression can give us width. There is a lot of controversy about the use of these methods, since the output can be porridge or a lot of antiphase, which can make our mix unreadable in certain sources. Of course, when recording live instruments, there is no such problem, as a rule, the sound is captured by several microphones, the sound source itself and its environment (room) are removed, with this recording method it is not necessary to apply any effects, it is enough to set the desired levels, clean and tighten for readability of the original signal. But when working with already recorded fragments (samples), the picture is completely different. When using samples mixed with the room, with tails, we do not have the opportunity to control the stereo image of our mix to the fullest, which gives a very bad result.

In the book " The Beatmaker's Bible» I was attracted by the method that sound engineers used before, it consists in dividing the channel into 2, divorced all the way to the left and right, oppositely duplicated with phase inversion. This method is called " Left minus right” and was used on analog mixers.

The huge advantage of this method is the preservation of mono compatibility, that is, when listening in mono, the sound is not distorted, but gives a good result. You can repeat it in your virtual mixer, but be aware that the levels should change strictly in parallel and in direct proportion to how much you changed the levels of the left channel, change the level of the right channel by the same amount. (see above)

IN FL Studio there is a plugin that mimics this circuit - Fruity Stereo Shaper. « Fruity Stereo Shaper. Stereo effect with the ability to separately control the volume of the left, right channels and their inverted copies, as well as their delay and phase shift. Appeared in FL Studio 9 » - Wikipedia. The scheme of this plugin is exactly the same as " Left minus right”, in addition, it has several convenient controls and 2 modes of operation - PRE And POST, which imply a place in the processing chain, corresponding to the name of the mode.

I use this plug-in on the snare drum, as it is very difficult to achieve its liveliness and depth, and this is very important, it is also important for the reason that it is worth leaving a small margin for vocals in the middle in the mix, otherwise they will conflict. I put it after the equalizer, in mode PRE, as it actually processes the “dry” sound, I unscrew the knob PHASE on 90-100% to the left, followed by shifting the channel in the mixer to the left by 5-7% by turning the stereo knob to 100% . It looks illogical, but when summing these offsets, it is not audible at all, however, the snare drum sounds much wider, while without antiphase. Also, I send the snare drums to a send channel with room reverb, which gives even more depth. This plugin can also be used on the group channel of all elements of the rhythm section, after all the processing, I personally have it: saturation, filtering, compression. After all the processing, I put this plugin in the mode POST, handle PHASE all the way to the left or right, by ear, and I get an even wider sound, but already of the entire rhythm section. It is important to remember that, the larger the phase shift, the less it is audible and the less its effect on the stereo field. Try, experiment, there are no strict rules in modern production.

Stereo Width Control

STEREO WIDTH CONTROL - RUMBLING FILTER

According to studies, the elimination of rumble (anti-phase low-frequency components) together with the expansion of the stereo base (amplification of the difference signal at midrange and high frequencies) psychoacoustically improves the sound, all other things being equal, more than reducing the non-linear distortions of the UMZCH from 0.1 to 0.01%.
Therefore, the introduction of an appropriate device into the amplification path is very justified and expedient.

I offer to your attention a scheme of such a regulator, developed back in 1989 by M. Starostenko, Miass, Chelyabinsk region. ("RADIO", 1989 No. 11, pp. 52-54). The regulator was assembled by me and a number of my friends, and has proven itself only on the good side.

Main technical characteristics:

Rated sensitivity. . . . . . . . . . . . . . . . . . . . . . . . . . 0.5V
input impedance. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . not less than 70 kOhm
Voltage transfer coefficient. . . . . . . . . . . . . . . . . . . 1
Maximum stereo expansion. . . . . . . . . . . . . . . . . . . . up to 2 times
Expansion rumble suppression level
stereo base 1 time (2 times), dB, at frequency, Hz
8 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26 (20)
20 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18 (12)
50 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10 (4)
Nonlinear distortion. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . no more than 0.05%
Signal to noise ratio. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . not less than 80 dB
overload capacity. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20 dB
Consumption current. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . no more than 8 mA

The device operates on the principle of sequential sum-difference conversion of a stereo signal. On the dual resistor R1, the adder of the signals of the left and right channels is assembled, and on the op amps DA1 and DA2, the difference conversion unit. C1 and C2, together with R1.1 and R1.2, perform the functions of a rumble filter. In the extreme left (according to the diagram) position of the R1 sliders, we have the maximum expansion of the stereo base (and minimal rumble suppression), in the right position we have the “MONO” mode with maximum rumble suppression.
Variable resistor R1 -- with a linear relationship. A properly assembled product does not need adjustment.

A few tips from VEPR:

If the OU K544UD1A is replaced by K544UD2B, then the non-linear distortion will decrease to 0.02-0.03%, and if on LF351 (LF147) - then to 0.02%. The experience of using this regulator has shown that the MONO mode is not needed. Yes
and the roar in those days came mainly from the "turntables" (EPA) - where are they now. Therefore, you can take R1 \u003d 33 kOhm, and turn on a constant resistor of 33 kOhm between its right (according to the diagram) and R2 conclusions. Many simply put on input fixed resistors 68 kOhm - saving controls and space
on the control panel.

Danich Alexander, aka VEPR.
Nizhyn, Chernihiv region

In the article by Yu. Kuznetsov, M. Morozov and A. Shityakov under this title (“Radio”, 1985. No. 1, p. 27h; 28), a description was given of a device that, despite its relative simplicity, could perform two functions at once :

• reduction in the level of rumble;
• improved stereo separation.

At the same time, as shown by the analysis of circuit solutions, a similar device can be built using a much smaller number of passive and active elements. A schematic diagram of such a regulator, developed by the author, is shown in fig. 1.

Main technical characteristics:

• Rated input voltage, V.....0.5;
• Input resistance, kOhm, not less.... 70;
• Voltage transfer coefficient .... 1;
• Maximum expansion of the stereo base, times.... 2;
• The level of suppression of low-frequency anti-phase components of the rumble in the "Stereo" mode (maximally extended stereo base), dB, at frequencies, Hz: 8 . . .26(20) 20 . .18(12); 50......10(4);
• Harmonic coefficient at rated input voltage, %, no more. 0.05;
• Signal to noise ratio. dB, not less... 80;
• Overload capacity, dB, not less. 20;
• Consumed current, mA, no more. 7.

The device operates on the principle of sequential sum-difference conversion of a stereo signal. It consists of a node for summing the signals of the left and right channels on a dual variable resistor R1 and a node for the difference signal conversion on two op-amps DA1 and DA2. In the first one, the width of the stereo base is adjusted from the nominal value to zero, the second one expands the stereo base by a fixed value.

Serial connection of these nodes made it possible to obtain a device that regulates the width of the stereo base from zero to the maximum value determined by the difference transformation node.

The functions of the rumble filter in the described design are performed by transition capacitors C1 and C2 together with the stereo width controls R1.1 and R1.2.

In the right (according to the diagram) position of the sliders of the resistor R1, the non-inverting inputs of both op-amps are combined, and, therefore, the signals at their outputs are equal in magnitude to half the sum of the signals of both channels (“Mono” mode).

In the left position of the sliders of this resistor, the signals at the inputs of the op-amp DA1 and DA2 are equal to the signals of the left and right channels, respectively, coming to the input of the entire device, and the difference conversion node expands the stereo base to the maximum value. In the intermediate position of the sliders of the resistor R1, the voltages at the non-inverting inputs of the op-amp are determined by the expressions: Ul1 \u003d Ul0 (1 - a / 2) + Ul0 * a / 2, Up1 \u003d Ul0 * a / 2 + Up0 (1 - a / 2) at R2> =R1, and the output voltages of the entire device:

• Ul2 \u003d Ul1 (1 + R / R4) - Up1 * R / R4;
• Ul2 \u003d - Ul1R / R4 + Up1 (1 + R / R4);

where R=R3=R5; Ul0, Ul1, Ul2 (Up0, Up1, Up2) - the voltages of the left (right) channel, acting respectively at the input of the regulator, the non-inverting inputs of the op-amp DA1 and DA2 and at the output of the device, and - the relative displacement of the sliders of the variable resistor R1 (a \u003d 0. ..1). In particular, when choosing the ratio R / R4 = 0.5 (which corresponds to the maximum expansion of the stereo base twice) in the middle position of the sliders of the resistor R1 (a = 0.5), a stereo signal Ul2 = Ul0, Up2 = Up0 is formed at the outputs of the op-amp and stereo is nominal.

When deriving relationships for the voltages acting at the controller outputs, it was assumed that the output resistances of the signal sources of the left and right channels are significantly less than the resistance of the resistor R1. If this condition is not met, the effect of the output resistances of the signal sources Rg on the device operation can be taken into account by formal substitution to determine Ul2, Up2 variable a to a new variable a1 = (a+RgR1) / (1+Rg/R1).

In practice, the influence of the output impedances of the signal sources is reduced to a decrease in the adjustment range of the stereo base in the direction of its expansion, and this influence can be practically eliminated by a corresponding increase in the ratio R/R4.

Rice. 1. Schematic diagram of the stereo width controller.

Rice. 2. The printed circuit board of the stereo width regulator.

In the low frequency region (f<= 1/2пR1C1 = 1/2пR1C2 = ~ 160 Гц) зависимость напряжения гармонического сигнала, лежащих в области частот значительно ниже 160 Гц, напряжения на входах ОУ приближаются к полусумме сигналов левого и правого каналов.

For anti-phase components of the roar (Ul0=-Un0), the transfer coefficient of the device is described by the following expression: K (jw) = Ul2 / Ul0 = Up2 / Up0 = 2jwR1C(1-a) / (1 + jwR1C), where C=C1=C2; w is the circular frequency of the rumble.

Resistor R2 sets the operating mode of the op-amp for direct current and, together with capacitors C1 and C2, determines the lower cutoff frequency of the frequency response of the device. With the values ​​\u200b\u200bof the elements indicated on the diagram; at the same time, its technical characteristics, such as harmonic coefficient, signal-to-noise ratio, current consumption, are determined only by the applied op-amp.

Dual variable resistor R1 - any, with an adjustment characteristic of group A. The resistance of the resistor R1 may be different than indicated in the diagram, nominal. The capacitance of capacitors C1 and C2 in this case must be changed in inverse proportion to the magnitude of the change in the resistance of the resistor R1.

Drawing of the printed circuit board of the device, designed for the installation of MLT resistors (R2 - R5). SPZ-23b (R1), KM capacitors (C1, C2)) is shown in fig. 2.

A properly assembled device does not need to be adjusted.

It should be noted that due to the low output impedance, the regulator is well matched with other functional units of the stereo path.

M. Starostenko. Miass, Chelyabinsk region Radio 1989, 11.