Reverse mastering: can the dynamic range of compressed recordings be increased? Dynamic range again

Dynamic audio processing on PC

In the previous article, I talked about software tools for eliminating noise and sound distortions, including listing those "sound cleaning" operations that need to be done with recording a song, starting with correcting errors in the microphone installation and ending with mastering, so that a group of songs, recorded on a disc, from an aesthetic point of view, was a single whole. This topic so serious that it should be devoted to the next few articles.

I'll start, like last time, with the main thesis: the sound recorded by an amateur in a home computer studio, although, of course, does not compare in quality with the results of the work of professional studios, but it can be close to them.

I write, and with the edge of my ear I listen to what the TV is mumbling there. Here's a movie that was featured in the announcement as a "super project." Tsar Peter dying, fighting for the throne. Passions are raging ... Through other channels, the investigator Turetsky is looking for stolen rare tomes, experts have shaken off the antiquity and are again conducting their investigation, because, it turns out, still "someone here and there in our country sometimes does not want to live honestly" ... Such different stories , but they have something in common. It's common - sound. Bad sound. Terrible sound recorded by professionals in professional studios. Especially in the "superproject": when the groans of the dying tsar and the cries of those close to him die down for a moment, background sounds come through clearly, you can even hear how the tape drives of the cameras work.

The following conclusions suggest themselves:

1. It is clear that in our country films have not been re-dubbed in a sound studio for a long time. Probably, there is no money for this. The way the sound is recorded on the set goes into the edited tape.

2. Some professionals do not use computer noise cancellation tools. It's not very clear why. Don't know about them? No time to read special literature? But even the elementary information that is contained in the five pages of my previous article would be enough for a start.

3. Some of those people who record sound for television films do not know how to use dynamics processing equipment.

We'll talk about dynamics processing now. This topic is complex, but if you concentrate, you will definitely understand everything, and the sound in your projects will become professional. Well, not professional, but amateur, but such that everyone will listen to them. For those who have doubts, I propose to evaluate the work of our readers, recorded on the disc, which accompanies the new book "Sonar. Secrets of Mastery". By the way, nothing prevents you from trying your hand. Your composition may well appear in the music collection on the next such disc.

So, dynamic processing. Formally, it consists in changing the dynamic range of audio signals. But to use it for the benefit of sound quality, this phrase is clearly not enough. So let's start from the beginning.

Sound level and dynamic range

The source of sound vibrations radiates energy into the surrounding space. The amount of sound energy passing per second through an area of \u200b\u200b1 m2 located perpendicular to the direction of propagation of sound vibrations is called the intensity (strength) of sound.

When we are in normal conversation, the power flow is approximately 10 μW. The loudest violin sounds can be 60 μW, and the organ sounds can range from 140 to 3200 μW.

A person hears sound in an extremely wide range of sound pressures (intensities). One of the reference values \u200b\u200bfor this range is the standard hearing threshold - the effective value of the sound pressure created by a harmonic sound vibration of 1000 Hz frequency, barely audible by a person with average hearing sensitivity.

The audibility threshold corresponds to the sound intensity Isv0 \u003d 10-12 W / m2 or the sound pressure ps0 \u003d 2CH10-5 Pa.

The upper limit is determined by the values \u200b\u200bof Isv. Max. \u003d 1 W / m2 or pev. Max. \u003d 20 Pa. When a sound of such intensity is perceived, a person experiences pain.

In the range of sound pressures, significantly exceeding the standard hearing threshold, the magnitude of the sensation is proportional not to the amplitude of the sound pressure psc, but to the logarithm of the ratio ps / ps0. Therefore, sound pressure and sound intensity are often estimated as logarithmic units decibels (dB) in relation to the standard hearing threshold.

The range of sound pressure changes from the absolute threshold of hearing to the pain threshold is for different frequencies from 90 dB to 130 dB.

If the ear of a person simultaneously perceives two or more sounds of different loudness, then a louder sound drowns out (absorbs) weak sounds. The so-called masking of sounds occurs, and the ear perceives only one, louder sound. Immediately after exposure of the ear to a loud sound, the hearing sensitivity to faint sounds decreases. This ability is called hearing adaptation.

Thus, the hearing threshold is highly dependent on the listening environment: in silence or against a background of noise (or other disturbing sound). In the latter case, the hearing threshold is increased. This indicates that the interference is masking the useful signal.

The human hearing aid has a certain inertia: the sensation of the appearance of sound, as well as its termination, does not appear immediately.

The audio signal is a random process. Its acoustic or electrical characteristics change continuously over time. Trying to track down random changes in the realizations of this chaos is an exercise that doesn't make much sense. It is possible to curb his majesty the case, to give it features of determinism, using averaged parameters, such as the level of the audio signal.

The audio signal level characterizes the signal at a certain moment and represents the voltage of the audio signal, expressed in decibels, rectified and averaged over a certain previous period of time.

The dynamic range of an audio signal is understood as the ratio of the maximum sound pressure to the minimum or the ratio of the corresponding voltages. In such a definition, there is no information about what pressure and voltage is considered to be the maximum and minimum. This is probably why the dynamic range of the signal determined in this way is called theoretical. Along with this, the dynamic range of the audio signal can be determined experimentally as the difference between the maximum and minimum levels for a sufficiently long period. This value essentially depends on the selected measurement time and the type of level meter.

The dynamic ranges of musical and speech acoustic signals of different types, measured with instruments, are on average:

80 dB for symphony orchestra

45 dB for choir

35 dB for pop music and vocal soloists

25 dB for speakers' speech

When recording, levels need to be adjusted. This is explained by the fact that the original (unprocessed) signals often have a large dynamic range (for example, up to 80 dB for symphonic music), and at home audio programs are listened to in a range of about 40 dB.

There is a drawback to manually adjusting levels. The response time of the sound engineer is about 2 seconds, even if the score of the composition is known to him in advance. This leads to an error in maintaining the maximum levels of music programs up to 4 dB in both directions.

Amplifiers, acoustic systems, and even human ears need to be protected from overloads caused by sudden abrupt changes in the amplitude of the audio signal - to limit the signal in amplitude.

The dynamic range of the signal must be matched with the dynamic ranges of recording, amplification, transmission devices.

To increase the range of FM radio stations, the dynamic range of the audio signal must be compressed. To reduce the noise level in pauses, it is desirable to increase the dynamic range.

After all, fashion, which dictates its conditions in all spheres of human activity, including in sound recording, requires a rich, dense sound of modern music, which is achieved by a sharp narrowing of its dynamic range.

Sound wave (volume envelope) of a fragment of S. Rachmaninov's opera "Aleko",

and contemporary dance music.

In classical music, nuances are important, dance music should be "potent".

This dictates the need to use devices for automatic signal level processing.

Dynamics processing devices

Devices for automatic processing of signal levels can be classified according to a number of criteria, the most important of which are: response time and function.

According to the criterion of response inertia, there are non-inertial (instantaneous) and inertial (with a variable transmission coefficient) auto-level regulators:

When the signal level at the input of the inertia-free autoregulator exceeds the nominal value, a trapezoidal signal is obtained at the output instead of a sinusoidal signal. Although the inertia-free auto-adjusters are simple, their use results in severe distortion.

Inertial is an autolevel control in which the gain is automatically changed depending on the input signal level. These auto-level controls only distort the waveform for a small amount of time. By choosing the optimal response time, such distortions can be made hardly audible.

Depending on the functions performed, inertial automatic level controllers are divided into:

Quasi-maximum level limiters

Auto level stabilizers

Dynamic range compressors

Dynamic range expanders

Compander Squelchs

Threshold Squelchs (Gates)

Devices with complex dynamic range conversion

The main characteristic of the dynamic processing device is the amplitude characteristic - the dependence of the output signal level on the input signal level.

The level limiter (limiter) is an autoregulator, in which the gain changes so that when the input signal exceeds the nominal level, the signal levels at its output remain almost constant, close to the nominal value. With input signals not exceeding the nominal value, the level limiter works like a conventional line amplifier. The limiter should react to level changes instantly.

Amplitude characteristic of the limiter

Auto level stabilizer is designed to stabilize signal levels. This is sometimes necessary to equalize the sound volume of individual fragments of a phonogram. The principle of operation of the autostabilizer is similar to that of the limiter. The difference is that the nominal output voltage of the auto stabilizer is approximately 5 dB less than the nominal output level of the limiter.

A compressor is a device whose gain increases as the input level decreases. The action of the compressor leads to an increase in the average power and, therefore, in the loudness of the sound of the processed signal, as well as to a compression of its dynamic range.

Amplitude characteristic of the compressor

The expander has the opposite amplitude characteristic to the compressor. It is used when it is necessary to restore the dynamic range converted by the compressor.

Amplitude characteristic of the expander

A compander is a system consisting of a series-connected compressor and an expander. It is used to reduce noise in the recording or transmission paths. sound signals.

The threshold squelch (gate) is an auto-regulator in which the transmission coefficient changes so that when the input signal levels are less than the threshold, the output signal amplitude is close to zero. With input signals that exceed the threshold, the squelch acts like a conventional linear amplifier.

Auto-adjusters for complex dynamic range conversion have multiple control channels. For example, a combination of a limiter, an auto-stabilizer, an expander and a threshold noise suppressor allows you to stabilize the sound volume of various fragments of a composition, withstand maximum signal levels and suppress noise during pauses.

Dynamic processing device structure

The inertial level controller has a main channel and a control channel. If the signal is fed to the control channel from the input of the main channel, we are dealing with direct regulation, and if from the output - with reverse.

The main channel in a direct control circuit includes audio amplifiers, a delay line and an adjustable element. The latter, under the influence of the control voltage, is able to change its transmission coefficient. The main channel in the reverse control circuit contains all of the above elements, except for the delay line.

The fundamentally important elements of the control channel are the detector and the integrating (smoothing) circuit. Until the voltage at the input of the circuit does not exceed the threshold (reference), the control channel does not generate a control signal, and the transmission coefficient of the controlled element does not change. When the threshold is exceeded, the detector generates impulse voltageproportional to the difference between the current signal value and the reference voltage. The integrating circuit averages the differential voltage and generates a control voltage proportional to the signal level at the input of the control channel.

The delay line in the main channel of the direct control circuit allows the control channel to work with some anticipation. It will detect a spike in the signal level before the signal reaches the adjustable element. Therefore, there is a fundamental possibility of eliminating unwanted transients. Level drops can be handled almost perfectly. However, the phase response of the analog delay line is not linear. The difference in phase shifts for different spectral components of the signal leads to distortion of the shape of the wideband signal when passing the delay line. Digital delay lines do not have this disadvantage, but to use them, the signal must first be digitized. AT virtual devices signal processing is processed in digital form, and there are no problems with the algorithmic implementation of functional elements.

When we are in normal conversation, the power flow is approximately 10 μW. The loudest violin sounds can be 60 μW, and the organ sounds can range from 140 to 3200 μW.

The audibility threshold corresponds to the sound intensity Isv0 \u003d 10-12 W / m2 or the sound pressure ps0 \u003d 2CH10-5 Pa.

The upper limit is determined by the values \u200b\u200bof Isv. Max. \u003d 1 W / m2 or pev. Max. \u003d 20 Pa. When a sound of such intensity is perceived, a person experiences pain.

In the area of \u200b\u200bsound pressures that are significantly higher than the standard hearing threshold, the magnitude of the sensation is proportional not to the amplitude of the sound pressure psc, but to the logarithm of the ratio ps / ps0. Therefore, sound pressure and sound intensity are often evaluated in logarithmic units of decibels (dB) relative to the standard hearing threshold.

The range of sound pressure changes from the absolute threshold of hearing to the pain threshold is for different frequencies from 90 dB to 130 dB.

The human hearing aid has a certain inertia: the sensation of the appearance of sound, as well as its termination, does not appear immediately.

The dynamic ranges of musical and speech acoustic signals of different types, measured with instruments, are on average:

80 dB for symphony orchestra

45 dB for choir

35 dB for pop music and vocal soloists

25 dB for speakers' speech

Amplifiers, acoustic systems, and even human ears need to be protected from overloads caused by sudden abrupt changes in the amplitude of the audio signal - to limit the signal in amplitude.

The dynamic range of the signal must be matched with the dynamic ranges of recording, amplification, transmission devices.

To increase the range of FM radio stations, the dynamic range of the audio signal must be compressed. To reduce the noise level in pauses, it is desirable to increase the dynamic range.

Sound wave (volume envelope) of a fragment of S. Rachmaninov's opera "Aleko",

and contemporary dance music.

In classical music, nuances are important, dance music should be "potent".

This dictates the need to use devices for automatic signal level processing.

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Introduction

One of the five senses available to humans is hearing. With it, we hear the world around us.

Most of us have sounds that we remember from childhood. For some, it is the voices of relatives and friends, or the creak of wooden floorboards in grandmother's house, or, perhaps, this is the sound of the train wheels on the railway that was nearby. Each will have their own.

How do you feel when you hear or remember sounds familiar from childhood? Joy, nostalgia, sadness, warmth? Sound can convey emotions, mood, induce action, or, conversely, calm and relax.

In addition, sound is used in various spheres of human life - in medicine, in the processing of materials, in the exploration of the depths of the sea and many, many others.

At the same time, from the point of view of physics, this is just a natural phenomenon - vibrations of an elastic medium, which means, like any natural phenomenon, sound has characteristics, some of which can be measured, others can only be heard.

Choosing musical equipment, reading reviews and descriptions, we often come across a large number of these very characteristics and terms that authors use without appropriate clarifications and explanations. And if some of them are clear and obvious to everyone, then others for an unprepared person do not carry any meaning. Therefore, we decided to tell you in simple language about these incomprehensible and difficult, at first glance, words.

If you remember your acquaintance with portable sound, it began quite a long time ago, and it was just such a cassette player, presented to me by my parents for the New Year.

He sometimes chewed the film, and then he had to unravel it with paper clips and a strong word. He ate batteries with an appetite that would have been the envy of Robin Bobin Barabek (who ate forty people), and therefore my, at that time, very meager savings of an ordinary schoolboy. But all the inconveniences paled in comparison with the main plus - the player gave an indescribable feeling of freedom and joy! So I "got sick" with the sound that you can take with you.

However, I would be wrong if I say that since that time I have always been inseparable from music. There were periods when there was no time for music, when the priority was completely different. However, all this time I tried to be aware of what is happening in the world of portable audio, and, so to speak, keep my finger on the pulse.

When smartphones appeared, it turned out that these multimedia combines can not only make calls and process huge amounts of data, but, which was much more important for me, store and play a huge amount of music.

The first time I got hooked on the "telephone" sound was when I listened to how one of the music smartphones sounds, in which the most advanced sound processing components were used at that time (before that, I confess, I did not take a smartphone seriously as a device for listening to music ). I really wanted this phone for myself, but could not afford it. With this, I began to follow lineup this company, which has established itself in my eyes as a manufacturer quality sound, however, it turned out that our paths with her constantly diverged. Since that time, I have owned various musical equipment, but I never stop looking for a truly musical smartphone for myself that could rightfully bear such a name.

Specifications

Among all the characteristics of sound, a professional on the move can overwhelm you with a dozen definitions and parameters, which, in his opinion, you must, well, you must definitely pay attention and, God forbid, some parameter will not be taken into account - it's a problem ...

I will say right away that I am not a supporter of this approach. After all, we usually choose equipment not for the "international audiophile competition", but still for ourselves, for our souls.

We are all different, and we all value something of our own in sound. Someone likes the sound "bassier", someone, on the contrary, is clean and transparent, for someone certain parameters will be important, and for someone completely different. Are all parameters equally important and what are they? Let's figure it out.

Have you ever come across the fact that some headphones play on your phone in such a way that you have to make it quieter, while others, on the contrary, force you to turn the volume up to full and still not enough?

In portable technology, resistance plays an important role in this. Often, it is by the value of this parameter that you can understand whether you will have enough volume.

Resistance

Measured in Ohms (Ohms).

Georg Simon Ohm - German physicist, deduced and confirmed experimentally a law expressing the relationship between the current in a circuit, voltage and resistance (known as ohm's law).

This parameter is also called impedance.

The value is almost always indicated on the box or in the instructions for the equipment.

There is an opinion that high-impedance headphones play quietly, and low-impedance headphones play loudly, and for high-impedance headphones you need a more powerful sound source, and low-impedance headphones will have enough of a smartphone. You can also often hear the expression - not every player will be able to "rock" these headphones.

Remember, low impedance headphones will sound louder on the same source. Despite the fact that from the point of view of physics this is not entirely true and there are nuances, in fact, this is the easiest way to describe the value of this parameter.

For portable equipment (portable players, smartphones), headphones with an impedance of 32 ohms and below are most often produced, but it should be borne in mind that for different types different impedances will be considered low. So, for full-size headphones impedance up to 100 Ohm is considered low impedance, above 100 Ohm - high impedance. For headphones of the in-ear type ("plugs" or earbuds), a resistance value of up to 32 ohms is considered low impedance, above 32 ohms - high impedance. Therefore, when choosing headphones, pay attention not only to the resistance value itself, but also to the type of headphones.

Important: The higher the impedance of the headphones, the clearer the sound will be and the longer the player or smartphone will work in playback mode. High impedance headphones draw less current, which in turn means less signal distortion.

Frequency response (amplitude-frequency characteristic)

Often in the discussion of a device, be it headphones, speakers or a car subwoofer, you can hear the characteristic - “shakes / does not shake”. You can find out whether the device will, for example, "swing" or is more suitable for vocal lovers, without listening to it.

To do this, it is enough to find its frequency response in the description of the device.

The graph allows you to understand how the device reproduces other frequencies. In this case, the fewer the differences, the more accurately the equipment can transmit the original sound, which means that the closer the sound will be to the original.

If in the first third there are no pronounced "humps", then the headphones are not very "bass", but if on the contrary, they will "swing", the same applies to other parts of the frequency response.

Thus, looking at the frequency response, we can understand what the equipment has a timbre / tonal balance. On the one hand, you might think that a straight line would be the ideal balance, but is it?

Let's try to figure it out in more detail. It just so happened that a person for communication uses mainly medium frequencies (MF) and, accordingly, is best able to distinguish this particular frequency band. If you make a device with a "perfect" balance in a straight line, I'm afraid that listening to music on such equipment will not be very pleasant for you, as most likely the high and low frequencies will not sound as good as the mids. The way out is to look for your balance, taking into account the physiological characteristics of hearing and the purpose of the equipment. For the voice one balance, for classical music - another, for dance - the third.

The graph above shows the balance of these headphones. Low and high frequencies are more pronounced, in contrast to the middle ones, which are less, which is typical for most products. However, the presence of a "hump" at low frequencies does not necessarily mean the quality of these lowest frequencies, since they can be, albeit in large quantities, but of poor quality - booming, humming.

The final result will be influenced by many parameters, ranging from how well the geometry of the case was calculated, and ending with what materials the structural elements are made of, and you can often find out only by listening to headphones.

In order to roughly imagine before listening to how high-quality our sound will be, after the frequency response, you should pay attention to such a parameter as the harmonic distortion.

Harmonic distortion

In fact, this is the main parameter that determines the sound quality. The only question is what quality is for you. For example, the well-known Beats by Dr. Dre at 1kHz has a harmonic distortion of almost 1.5% (above 1.0% is considered a rather mediocre result). At the same time, oddly enough, these headphones are popular with consumers.

It is desirable to know this parameter for each specific group of frequencies, because the allowable values \u200b\u200bdiffer for different frequencies. For example, for low frequencies, 10% can be considered a permissible value, but for high frequencies, no more than that same 1%.

Not all manufacturers like to indicate this parameter on their products, because, unlike the same volume, it is rather difficult to comply with. Therefore, if the device you choose has a similar graph and you see a value of no more than 0.5% in it, you should take a closer look at this device - this is a very good indicator.

We already know how to choose headphones / speakers that will play louder on your device. But how do you know how loud they will play?

For this, there is a parameter that you most likely have heard about more than once. He is very fond of using nightclubs in their advertising materials to show how loud the party will be. This parameter is measured in decibels.

Sensitivity (volume, noise level)

The decibel (dB), a unit of measure for the intensity of sound, is named after Alexander Graham Bell.

Alexander Graham Bell is a scientist, inventor and businessman of Scottish descent, one of the founders of telephony, the founder of Bell Labs (formerly Bell Telephone Company), which determined the further development of the telecommunications industry in the United States.

This parameter is inextricably linked with resistance. The level of 95-100 dB is considered to be sufficient (in fact, this is a lot).

For example, the volume record was set by Kiss on July 15, 2009 at a concert in Ottawa. The sound volume was 136 dB. By this parameter, the Kiss group surpassed a number of famous competitors, including such groups as The Who, Metallica and Manowar.

At the same time, the unofficial record belongs to the American team The Swans. According to unconfirmed reports, at several concerts of this group, the sound reached a volume of 140 dB.

If you want to repeat or surpass this record, remember that loud sound can be regarded as a violation of public order - for Moscow, for example, the standards provide for a sound level equivalent to 30 dBA at night, during the day - 40 dBA, maximum - 45 dBA at night, 55 dBA during the day ...

And if the volume is more or less clear, then the next parameter is not so easy to understand and track as the previous ones. It's about dynamic range.

Dynamic range

Basically, it is the difference between the loudest and quietest sounds without clipping (clipping).

Anyone who has ever been to a modern cinema has experienced what a wide dynamic range is. This is the very parameter thanks to which you hear, for example, the sound of a shot in all its glory, and the rustle of the boots of the sniper creeping along the roof, who made this shot.

More range on your equipment means more sounds that your device can transmit without loss.

At the same time, it turns out that it is not enough to convey the widest possible dynamic range, you need to manage to do it so that each frequency is not just audible, but audible with high quality. Responsible for this is one of those parameters that almost everyone can easily evaluate when listening to a high-quality recording on the equipment of interest. It's about detailing.

Detailing

This is the ability of the equipment to divide sound into frequencies - low, medium, high (LF, MF, HF).

It is this parameter that determines how distinctly individual instruments will be heard, how detailed the music will be, whether it will not turn into just a jumble of sounds.

However, even with the best detail, different hardware can produce very different listening experiences.

It depends on the skill of the equipment localize sound sources.

In reviews of musical technology, this parameter is often divided into two components - stereo panorama and depth.

Stereo panorama

Reviews usually describe this parameter as wide or narrow. Let's see what it is.

From the name it is clear that we are talking about the width of something, but what?

Imagine that you are sitting (standing) at a concert of your favorite band or artist. And before you on the stage, the instruments are arranged in a certain order. Some are closer to the center, others are further away.

Have you presented? Let them start playing.

Now close your eyes and try to distinguish where this or that instrument is. I think you can easily do it.

And if the instruments are put in front of you in one line one after another?

Let's bring the situation to the point of absurdity and move the instruments close to each other. And ... let's put the trumpeter on the piano.

Do you think you will like this sound? Will it be possible to make out where which instrument is?

The last two options can most often be heard in low-quality equipment, the manufacturer of which does not care what sound his product produces (as practice shows, the price is not an indicator at all).

High-quality headphones, speakers, music systems should be able to build the correct stereo panorama in your head. Thanks to this, when listening to music through good equipment, you can hear where each instrument is located.

However, even with the ability of the equipment to create an excellent stereo panorama, such sound will still feel unnatural, flat due to the fact that in life we \u200b\u200bperceive sound not only in the horizontal plane. Therefore, no less important is such a parameter as the depth of sound.

Sound depth

Let's go back to our fictional concert. We will move the pianist and violinist a little further into our stage, and put the guitarist and saxophonist a little ahead. The vocalist will take his rightful place in front of all the instruments.

Did you hear that on your musical equipment?

Congratulations, your device is capable of creating a surround sound effect by synthesizing a panorama of imaginary sound sources. And if it's simpler, then your equipment has good sound localization.

If we are not talking about headphones, then this issue is solved quite simply - several emitters are used, placed around, allowing you to separate the sound sources. If we are talking about your headphones and you can hear it in them, congratulations for the second time, you have very good headphones for this parameter.

Your equipment has a wide dynamic range, excellent balance and good localization of sound, but is it ready for sudden changes in sound and rapid rise and fall of impulses?

How is her attack?

Attack

From the name, in theory, it is clear that this is something swift and inevitable, like the blow of the Katyusha battery.

But seriously, here's what Wikipedia tells us about it: Sound attack - the initial impulse of sound production, necessary for the formation of sounds when playing any musical instrument or when singing vocal parts; some nuanced characteristics of various methods of sound production, performing strokes, articulation and phrasing.

If you try to translate this into understandable language, then this is the rate of rise of the amplitude of the sound before reaching a given value. And if it's even clearer - if your equipment has a bad attack, then bright compositions with guitars, live drums and fast sound drops will sound cottony and dull, which means goodbye to good hard rock and others like that ...

Among other things, in articles you can often find such a term as sibilance.

Sibilants

Literally - whistling sounds. Consonants, when pronounced, the flow of air rapidly passes between the teeth.

Remember this fellow from the Disney cartoon about Robin Hood?

There are very, very many sibilants in his speech. And if your equipment also whistles and hiss, then alas, this is not a very good sound.

Remark: by the way, Robin Hood himself from this cartoon is suspiciously similar to the Fox from the recently released Disney cartoon “Zootopia”. Disney, you repeat yourself :)

Sand

Another subjective parameter that cannot be measured. And you can only hear.

In essence, it is close to sibilants, it is expressed in the fact that at high volume, with overload, the high frequencies begin to disintegrate into parts and the effect of crumbling sand appears, and sometimes high-frequency rattling. The sound becomes somehow rough and at the same time loose. The sooner this happens, the worse, and vice versa.

Try at home, from a height of a few centimeters, to slowly pour a handful of granulated sugar onto the metal lid of the pan. Have you heard? Here, this is it.

Look for a sound that is free of sand.

frequency range

One of the last direct parameters of sound that I would like to consider is the frequency range.

Measured in hertz (Hz).

Heinrich Rudolf Hertz, main achievement - experimental confirmation of the electromagnetic theory of light by James Maxwell. Hertz proved the existence of electromagnetic waves. Since 1933, the name of Hertz has been the name for the unit of measurement of frequency, which is included in the international metric system of SI units.

This is the parameter that you with a 99% probability will find in the description of almost any musical technique. Why did I leave it for later?

To begin with, a person hears sounds that are in a certain frequency range, namely from 20 Hz to 20,000 Hz. Anything above this value is ultrasound. Anything below is infrasound. They are inaccessible to human hearing, but are available to our smaller brothers. This is familiar to us from school courses in physics and biology.

In fact, for most people, the real audible range is much more modest, and for women the audible range is shifted upward relative to the male, so men are better at distinguishing low frequencies, and women are better at distinguishing high frequencies.

Why, then, do manufacturers indicate on their products a range that is beyond our perception? Maybe it's just marketing?

Yes and no. A person not only hears, but also feels, feels the sound.

Have you ever stood near a large speaker or subwoofer playing? Remember your feelings. The sound is not only heard, it is also felt by the whole body, it has pressure, strength. Therefore, the larger the range indicated on your equipment, the better.

However, you should not attach too much importance to this indicator - you rarely find equipment whose frequency range is narrower than the boundaries of human perception.

additional characteristics

All of the above characteristics are directly related to the quality of the reproduced sound. However, the final result, and therefore the pleasure of viewing / listening, is also influenced by the quality of the original file and which sound source you use.

Formats

This information is on everyone's lips, and most already know about it, but just in case, we will remind you.

In total, there are three main groups of audio file formats:

uncompressed audio formats such as WAV, AIFF
lossless compressed audio formats (APE, FLAC)
lossy compressed audio formats (MP3, Ogg)

We recommend reading more about this by referring to Wikipedia.

We note for ourselves that it makes sense to use APE, FLAC formats if you have professional or semi-professional equipment. In other cases, the capabilities of the MP3 format, compressed from a high-quality source with a bit rate of 256 kbit / s, are usually sufficient (the higher the bit rate, the less losses during audio compression). However, this is more a matter of taste, hearing and individual preferences.

A source

The quality of the sound source is equally important.

Since we were originally talking about music on smartphones, let's consider this particular option.

Not so long ago, the sound was analog. Remember bobbins, cassettes? This is analog audio.

And in your headphones, you hear analog sound that has gone through two stages of conversion. It was first converted from analog to digital, and then converted back to analog before being fed to the headphone / speaker. And from what quality this transformation was, in the end the result will depend - the sound quality.

In a smartphone, a DAC, a digital-to-analog converter, is responsible for this process.

The better the DAC, the better the sound you will hear. And vice versa. If the DAC in the device is mediocre, then whatever your speakers or headphones are, you can forget about the high sound quality.

All smartphones can be divided into two main categories:

Smartphones with dedicated DAC
Smartphones with built-in DAC

At the moment, a large number of manufacturers are engaged in the production of DACs for smartphones. What to choose, you can decide by using the search and reading the description of this or that device. However, do not forget that among smartphones with a built-in DAC, and among smartphones with a dedicated DAC, there are samples with very good sound and not very much, because optimization plays an important role operating system, the firmware version and the application through which you listen to music. In addition, there are kernel software audio mods that can improve the final sound quality. And if engineers and programmers in a company do one thing and do it competently, then the result turns out to be worthy of attention.

It is important to know that when directly comparing two devices, one of which is equipped with a high-quality built-in DAC and the other with a good dedicated DAC, the latter will invariably win.

Conclusion

Sound is an inexhaustible topic.

I hope that thanks to this material, a lot in music reviews and texts has become clearer and easier for you, and previously unfamiliar terminology has acquired additional meaning and meaning, because everything is easy when you know.

Both parts of our educational program about sound were written with the support of Meizu. Instead of the usual praising of the devices, we decided to make useful and interesting articles for you and pay attention to the importance of the playback source in obtaining high-quality sound.

Why does Meizu need it? The other day, the pre-order of the new musical flagship Meizu Pro 6 Plus began, so it is important for the company that regular user knew about the nuances of high-quality sound and the key role of the playback source. By the way, having issued a paid pre-order before the end of the year, you will receive a Meizu HD50 headset as a gift to your smartphone.

And we have prepared a music quiz for you with detailed comments on each question, we recommend you try your hand:

HI-FI AUDIO.RU / Alexander / edited

When choosing music discs (CDs), recording dynamic range (DR) is of great if not decisive importance to the listener. Precisely because of the dynamic range of recording on a CD deliberately narrowed (compressed) by the sound engineer, claims to the sound can arise.

Compression over the sound range is used more and more often not only at the stage of final preparation of the disc. Any DR compression is detrimental to the listening experience. If you have a persistent feeling of mush and confusion while listening to a CD, a "dirty" sound is a sign that the disc is most likely compressed in its dynamic range mercilessly.

What is dynamic range and why should it be compressed at all?

Dynamic range is the range between the quietest and loudest sounds in a track. Naturally, the larger it is, the more subtly and accurately the musical material will be presented, where everything will be heard in three-dimensional space - from air turbulence from a conductor's baton to a cannon shot. Based on the foregoing, there is no need to compress the dynamic range; its compression can be perceived as a distortion of sound.

In many difficultly composed and masterfully performed musical works, the dynamic range is very large and there are places where musicians play extremely quietly, and there are places where expression grows and the music rumbles. When listening, in such compositions, the volume of the amplifier is set high enough and it becomes perfectly audible, both the quietest sounds and, as they grow, very loud.

In portable devices (smartphones, tablets) there are low-power amplifiers, which, it is doubtful, can play all this in the full range with an acceptable volume. Therefore, they began to use compression - the quietest sounds in volume are pulled up to the loudest (it turns out that they actually start to scream in a whisper), the dynamic range is narrowed, but the volume in general increases by 30%, which is a plus for mobile devices that are listened to in an aggressive listening environment (noisy street, metro). Thus, "music for mobile phones" in all cases is a compromise between quality and convenience. Manufacturers are willing to sacrifice sound quality for amateurs mobile musicbut ultimately spoil the music for everyone.

On the example of the album of the group ZZ Top - the disfigurement of the sound by later releases. In the remaster of 2008, the original contours are no longer even guessed. Click on the picture for dynamic display.

Music lovers faced the difficult task of selecting CDs for their collection that were not mutilated by compression of the dynamic range, which is now becoming an increasingly insoluble problem.

To determine the DR of any piece of music, it is enough to install the Dynamic Range Meter plugin which measures the dynamic range in the foobar2000 player. More precisely, it measures a certain crest factor - the difference between the peak levels and RMS (root mean square value of the sound level in an album or audio track). If the crest factor DR of a phonogram is 14, this is an excellent indicator, and above 15 is close to fiction, but it should be understood that this indicator will be different for the genres in which the music is performed.

So for rock music in general, a good result starts with DR 10. For example, the band's album Nazareth "Sound Elixir" on CD has DR \u003d 10 and at the same time sounds great thanks to the use of electronic instruments. For heavy music, this may be quite and sufficient, if the musicians have not used strong sound differences. However, a wider dynamic range is required to reproduce acoustic instruments - guitar, saxophone, etc. In such cases, the difference in the range from 13 to 15 will please.

In general, most solid CDs show DR from 11 to 14. At the same time, there are discs with a dynamic range of 15 (for example, a group TV "Fatherland of Illusions") and even 18. Discs with high DR are listened to with great pleasure - their sound is open, natural, devoid of digital dryness and weight.

Minimum DR table according to musical style.

So, if the sound of the disc is dirty, but tolerant, then most likely it is a disc compressed in terms of the dynamic range with a value of no more than 8. Many early concerts of the group go on with this value. Nazarethand others - this is depressing, since such interesting and rich in instruments music is worthy best quality... It is truly bewildering when a priori audiophile performers release recordings of their concerts with strong compression. For example disk Sade "Soldier of Love"released in 2010 (!), has a DR of dynamic range equal to only 10. At the same time, the compositions are filled with beautiful female vocals and acoustic instruments. The range compression is clearly audible here and is very disappointing. It becomes unclear for whom then such CDs are released according to the principle - if for audiophiles this quality is not very suitable for listening, and the music is clearly not commercial in nature.

It is doubtful that today anyone will listen to music on the street from a portable CD player, when in a mobile environment, instead of uncompressed CD formats, music files have long been used, in most cases these are not audiophile formats (mp3, AAC), which also have a destructive nature and limitation also in the frequency range. Then a reasonable question arises: why spoil the CD over DR and write discs without compression? After all, there is no common sense to distort a CD recording for a higher volume, however, the marketing machine of the war for loudness is launched on full power and no return is expected. Unfortunately, statistics show that the manufacturer is increasing the compression of sound material every year, which of course negatively affects the sound quality on Hi-Fi equipment.

Indeed, an uncompressed disc on a cheap portable player or smartphone will sound "ineffective" due to external noises that will mask the quietest sounds, and the compressed sound will seem better due to the fact that the volume of quiet sounds is hyperextended and located above the external noise. It's like the sound engineer is puzzled with the goal of making a disc that will sound great with a jackhammer in motion. It may seem fine in such situations, but can you seriously talk about the sound quality if you use deep compression?

In any case, low-quality and low-grade playback and for high-quality playback on good Hi-Fi / Hi-End devices, compressed recordings are not suitable.

Most audiophiles do not care about the volume of the disc, it can be set to any on the amplifier, the clarity and detail of the sound is important, and many other parameters.

With the advent of modern high-end amplifiers, music has opened up a new dimension that adds another delightful dimension to it - the possibility of greater engagement through audiophile portrayal of musical events. In this dimension, not only the melody is perceived, but also every sound that sings and delights in a good path, clings to the strings of the soul.

That is why most modern discs after purchase you want to immediately throw out, for example, an album Madonna "Handy Candy"... The sound on them is terribly dirty, mushy, depressing to the ear. The reason is easily determined by checking the DR of the dynamic range. On the disc, it is equal to the depressing value of 5. Good-sounding discs can be considered records with a range of at least 10 and higher. The CD range from DR 8 and below does not provide the best listening experience.

Many will offer as a panacea listening to vinyl discs, where compression is unlikely, but compression is unlikely on all original CDs of old issues (DR up to 18 occurs), and modern vinyl can be compressed as well. This is the first argument, and the second comes from the fact that when measured, the DR value of the dynamic range of modern vinyl discs is not very high. The DR value is 12-14 for different vinyl discs. But serious suspicions remained that the lower boundary was not determined by the most quiet sound, and the rumble and noise of the vinyl record itself due to the mechanical nature of data reading, and then, probably, the real DR has an even worse value. At the same time, it is not uncommon to find recordings on CDs with DR of the dynamic range equal to 15, and, in addition, channel separation and many other indicators are significantly better on the disc.

From the above, we can conclude that the compression intensity of the DR sound range has a great influence on the sound quality of a CD. As a response to this situation, special "audiophile" discs without compression began to appear on the market, for example, the compilation Audiophile World.

For the curious: websitewww.dr.loudness-war.info contains a catalog of DR measured values \u200b\u200bfor a large number of audio CDs.

What is dynamic range?

Dynamic range can be defined as the distance between the level of the quietest and loudest possible signal. For example, if the instructions for the processor indicate that the maximum input signal level before distortion is +24 dB, and the noise threshold at the output is -92 dB, then the total dynamic range of the processor is 24 + 92 \u003d 116 dB.

The dynamic range of the orchestra is on average between -50 dB and +10 dB. Which adds up to 60 dB. While 60dB dynamic range might seem like little to you, with simple calculations it turns out that +10dB is 1000 times louder than -50dB!

The dynamic range in rock music is much smaller, typically -10 dB to +10 dB, or 20 dB in total. Therefore, mixing different signals in rock music into a single mix is \u200b\u200bpretty boring.

Why do we need compression?

Let's say you are mixing a rock recording with an average dynamic range of 20 dB. And you want to add uncompressed vocals to the mix. The average vocal dynamic range is approximately 40 dB. What does this mean for the mix? Too quiet vocal parts will simply not be heard, and too loud ones will stick out from the overall picture. In this situation, the compressor is necessary to reduce (compress) the dynamic range of the vocal within 10 dB.

In this case, the vocals will be at about +5 dB. The range is 0 dB to +10 dB. Quiet phrases will now be above the lowest signal level in the mix, and loud phrases will not stick out. It turns out that the vocals take their place in the mix.

The same principle works for any instrument in the mix. Each instrument has its own place in the mix, and a good compressor helps the engineer mix them correctly.

Is a compressor needed for everything?

Usually, in response to this question, you hear: "Of course not! Overcompressed tracks sound terrible." This statement is true only in one case - if you clearly hear how the compressor works on the recording.A high-quality expensive compressor, being correctly tuned, sounds imperceptible! Overcompressed sound is a consequence of errors in the processing of specific instruments, unless of course this is done deliberately in order to get a special effect ...

Why do you think all expensive mixing consoles have their own compressor on each channel? The answer is simple - most instruments need compression, however subtle. This helps them to be heard in the mix.

Why do we need noise gates?

Let's take an example with vocals. Let's say you set it to a 20 dB range. Problems start when the compressor boosts the quietest signals in a vocal track. All sorts of unwanted noises in the background, pieces of phonogram that got into the microphone from the headphones, etc. You can try to simply turn the volume down during the pauses, but this usually ends in complete failure. A much better way is to use a noise gate. We can set the threshold of the noise gate, for example, to -10 dB, which corresponds to the lower limit of the dynamic range of the vocal in our case. Thus, the gate will automatically nullify all unwanted signals between phrases.

If you've ever tried mixing a live recording, you know how many problems arise with a drum kit, namely with the iron that gets into the microphones mounted on toms. As soon as you add highs on the EQ to make the toms brighter, they begin to climb up the cymbal. And this is especially well heard through the HF loudspeakers in monitors. If we use gates on mics recording toms, so that the hardware no longer sounds through them in pauses, we will very much clean up the overall mix and make it much more legible.

Dynamic processing types

Dynamics processing is the process of changing the dynamic range of a signal to enhance the capabilities of the equipment through which the signal is recorded or played back. In other words, we get the ability to record or play the recorded signal without distortion and / or noise, thereby simplifying our mixing task.

Compressor and Limiter

Punchy, well audible, with good presentation - these are all descriptions of sound signals obtained by processing them with compressors and limiters.

Compression and limiting are forms of controlling the dynamic range (loudness) of a signal. Audio signals have a fairly large spread in volume levels. A peak signal can overload the recording circuit, which in turn will distort the signal.

A compressor / limiter is a kind of amplifier in which the volume level depends on the level of the audio signal passing through it. By selecting a certain compressor / limiter value, the signal will automatically attenuate above the preset level or threshold level.

In essence, compression is the process of attenuating the input signal in a given proportion. Used to narrow the dynamic range of a voice or musical instrument, allowing distortion-free recording. Also used when creating a mix, reducing the frequency difference of each track.

A vocalist, for example, is constantly moving in front of the microphone and the output signal fluctuates up and down, which sounds strange. In this case, the compressor will solve the problem by lowering the volume of individual phrases so that the result is smooth vocals.

The amount of signal attenuation depends on the ratio of the compression to the threshold level. A ratio of 2: 1 or less is considered weak compression, where the output signal that exceeds the threshold level is halved. Ratios above 10: 1 can be called strong limiting.

The lower the threshold level, the more part of the signal is compressed (at a certain input signal level). It is important to know when to stop, since too strong compression kills the dynamics of the recording (while some sound engineers kill it on purpose as an effect)!

Limiting is a type of signal processing in which loudness bursts (amplitude jumps) are suppressed.

A compressor / limiter is used for many sound processing tasks, for example:

The kick drum sound can get lost among electric guitars. And no matter how loud the track sounds, the kick drum sounds "dirty". Compression will straighten the kick drum sound against the background of the guitars.

The range of voice on the recording is wide enough. Loudness peaks can bulge out of the overall sound. There can be many such peaks, and they are all different, so it is almost impossible to align them through the mixer. The compressor / limiter automatically controls the volume without distorting the subtleties of the vocals.

The guitar solo is muffled by the rhythm. Don't crank the fader all the way, compression will put the lead guitar in its place in the mix.

Bass guitar is difficult to record. A smooth sound with good attack is achieved through proper compression. And no need to cut the low end of the mix - the compressor / limiter allows the bass to manifest at any frequency

Expander

There are two main types of expansion: dynamic and downward. Expansion expands the dynamic range of the signal when it is above the threshold. Dynamic expansion is essentially the opposite of compression; dynamic expansion is applied on TV and radio to de-compress just before the audio signal is transmitted. Compression followed by expansion is called companding. Currently, downward expansion is most commonly used. Unlike compression, which lowers the signal above the threshold, expansion lowers the signal below the expansion threshold. The degree of decline is determined by the expansion ratio. For example, a 2: 1 ratio cuts the signal in half (this means that if the signal is below the threshold by 5dB, the expander will lower it to 10dB). Expansion is often used to reduce noise, it is a very powerful and simple noise gate. The main difference between an expander and a noise gate is that the expansion depends on how far the signal goes “below the threshold,” whereas when the noise gate is operating, this does not matter.

Noise suppression

Noise reduction is the process of removing unwanted noise from a recording by limiting the signal below a specified threshold. As described above, the operation of the noise gate does not depend on the signal level below the threshold. The device's output is open as long as the signal is above the threshold.

The duration of the output opening is determined by the attack speed. The duration of the device operation when the signal is below the threshold is called the hold time. The closing speed of the output is determined by the return time. The level of unwanted signal rejection in the closed position is determined by the range.

Brief glossary of terms

It is scientifically proven that if you want to quickly learn a subject, you must first understand the basic concepts. The same principle applies to sound recording and further work with sound. Most of the instructions and textbooks assume the presence of basic knowledge, without which it is difficult to read them. Hopefully the next section will help you clean up your mind and get you down to the basics.

Compressors

Attack.

The attack determines the speed at which the compressor acts on the input signal. A long attack (clockwise control until it stops) initially allows the signal (so called initial transient) to pass unprocessed through the compressor, while a short attack (counterclockwise until it stops) immediately processes the signal according to the compression ratio and the set threshold level.

Auto.

The compressor operates in automatic attack and return mode. In this case, the regulators do not affect the process, but the programmed parameter values \u200b\u200bare used.

Compressor Sidechain.

The side channel input interrupts the signal with which the compressor determines the required compression level. With the side channel disabled, the input signal goes directly to the main compressor circuit. When it is turned on, no signal is sent to the main circuit. Now you can process the control signal with an equalizer, for example, by applying de-essing (voice frequency equalization). After processing, the control signal is fed back to the compressor via the channel output. Typical side channel applications are using a compressor to mute background music during the presenter's performance or lowering the volume of the rhythm guitar against the background of vocals. The voice is now easily distinguishable. In this case, the vocal track goes to the side channel, while the background music goes to the main compressor circuit. The compressor now lowers the background music (a process called ducking) when the vocalist starts to sing or speak.

Hard and soft compression (Hard / Soft Knee)

With hard compression, the signal is attenuated as quickly as possible when the threshold is exceeded. When softer, the signal is attenuated more smoothly after it has exceeded the set threshold, which provides a more natural sound for music.

Limiters.

A limiter is a compressor that does not allow the signal to rise above the threshold level. For example, if the threshold is set to 0 dB, the “Ratio” parameter is turned fully clockwise, the compressor will start in limiter mode at 0 dB and the output signal will never exceed this value.

Makeup Gain

With compression, the compression of the signal usually affects the overall volume level. The gain control allows you to restore the level lost during compression.

Ratio.

Ratio is the relationship between the output and input signals, this parameter sets the compression slope. For example, by setting a ratio of 2: 1, any signal above the threshold will be compressed at a ratio of 2: 1. For every decibel at the input to the compressor, there is 0.5 dB at the output, thus creating a compression that doubles the signal. As the ratio increases, the compressor gradually enters the limiter mode.

Release time.

Return time is the time that elapses between the time the input signal falls below the threshold and the moment the compression level returns to zero (the compressor stops attenuating the signal). The short return produces an uneven, chopped sound, especially on bass. A long return “squeezes” the sound too much, flattening it. Any value of the return time can be used - select by ear.

Threshold.

The compression threshold (compression threshold) determines the value above which the signal attenuation begins. Normally, turning the threshold control to the left will increase the signal that is being compressed (at a ratio greater than 1: 1).

Expanders

Downward Expansion.

Downward expansion is most commonly used in professional recording. The signal is attenuated below the threshold. This is the standard way to suppress noise.

Ratio.

The expansion ratio determines the amount of attenuation of the signal when it falls below the threshold. For example, with an expansion ratio of 2: 1, every decibel below the threshold value is attenuated in half. At a ratio of 4: 1 and higher, the expander works almost like a noise gate, only without the possibility of adjusting the attack, delay and return times.

Noise Gate

Attack.

The "attack time" parameter sets the amount at which the gate opens. Fast attack is suitable for percussion instruments, while vocals and bass require a smooth opening. Attacking them too fast will result in a noticeable “click” when mixing. A click when opening is inherent in any gate, but when correct setting you can't hear it.

Hold time (Hold).

Hold time is a fixed period of time during which the gate is open when the signal is below the threshold. The value of this parameter plays a role when gating, for example, a snare drum - after hitting it, a certain time passes, after which the gate closes abruptly.

Range.

Gate Range - The amount of attenuation of the signal when the gate is closed. Thus, when this parameter is set to 0 dB, no signal attenuation occurs at all. A value of -60 dB means that when the gate is closed, the signal will be attenuated (gated) by 60 dB, etc.

Release time.

The gate return time determines the speed at which the gate goes from open to fully closed. The return time is usually adjusted to preserve the natural decay of the instrument or vocals. A high return speed removes noise, but can cause drum stuttering, which is eliminated by a slow return speed. Adjust this parameter carefully for the most natural effect.

Threshold level.

Gate Threshold sets the value at which the gate opens. The principle is simple - any signal above the threshold is passed intact, and the signal below is attenuated by an amount depending on the range setting. If you turn the knob all the way to the left, the gate will be disabled (i.e. always open), and any signal passes without attenuation.

Below are the compression presets used in PreSonus BlueMax. These presets are standard settings, a kind of starting point for working with sound.

Vocals

Warm vocals. These are parameters for light compression with a low ratio and wider range, mainly for lyric songs performed live. The vocals are "in place".

Screaming.Parameters for loud vocals. A pretty harsh compression for vocalists who don't care about mic distance. The voice protrudes strongly from the mix, creating a sense of presence.

Left / right (stereo) overheads. The "ratio" and "threshold" parameters are low here, which gives a wide range that even cymbals will fit. Deep lows, overall sound is lively with low reverberation. More punchy sound, less room effect.

Acoustic guitar. The preset emphasizes the attack of the acoustic guitar and provides an even sound that will allow the guitar to remain audible.

Keyboard instruments

Piano. A special preset for equalizing the entire range of the piano - from the lowest sound to the fifth octave. The parties of both hands are clearly audible.

Orchestra. The settings are suitable for both strings and other orchestral synthesizer “sets”. The overall dynamic range has been reduced for easy addition to the mix.

Circuit.The settings expand the range of the main mix.

Threshold	Ratio	Attack	Release
-13.4 dBu	1.2:1	0.002 ms	182 ms