Dynamic range: compressed or standard. Compression in practice. Problems solved by compression

Dynamic compression(Dynamic range compression, DRC) - narrowing (or expanding in the case of an expander) of the dynamic range of the phonogram. Dynamic range, is the difference between the quietest and loudest sound. Sometimes the quietest sound in a soundtrack will be a little louder than the noise level, and sometimes a little quieter than the loudest. Hardware devices and programs that perform dynamic compression are called compressors, distinguishing among them four main groups: compressors themselves, limiters, expanders and gates.

Tube analog compressor DBX 566

Downward and upward compression

Downcompression(Downward compression) reduces the volume of a sound when it begins to exceed a certain threshold, leaving quieter sounds unchanged. An extreme version of downward compression is limiter. Boost compression Upward compression, on the other hand, increases the volume of a sound if it is below a threshold without affecting louder sounds. At the same time, both types of compression narrow the dynamic range of the audio signal.

Downcompression

Boost compression

Expander and Gate

If a compressor reduces dynamic range, an expander increases it. When the signal level rises above the threshold level, the expander increases it further, thereby increasing the difference between loud and soft sounds. Devices like this are often used when recording a drum kit to separate the sounds of one drum from another.

A type of expander that is used not to amplify loud sounds, but to attenuate quiet sounds that do not exceed a threshold level (for example, background noise) is called Noise gate. In such a device, as soon as the sound level becomes less than the threshold, the signal stops passing. Typically a gate is used to suppress noise during pauses. On some models, you can make the sound, when it reaches a threshold level, not stop abruptly, but gradually fade out. In this case, the decay rate is set by the Decay control.

Gate, like other types of compressors, can be frequency dependent(i.e. treat certain frequency bands differently) and can operate in side-chain(see below).

Compressor operating principle

The signal entering the compressor is split into two copies. One copy is sent to an amplifier, in which the degree of amplification is controlled by an external signal, and the second copy generates this signal. It enters a device called a side-chain, where the signal is measured and, based on this data, an envelope is created that describes the change in its volume.
This is how most modern compressors are designed, this is the so-called feed-forward type. In older devices (feedback type), the signal level is measured after the amplifier.

There are various analog variable-gain amplification technologies, each with its own advantages and disadvantages: tube, optical using photoresistors, and transistor. When working with digital audio (in a sound editor or DAW), your own mathematical algorithms can be used or the operation of analog technologies can be emulated.

Main parameters of compressors

Threshold

A compressor reduces the level of an audio signal if its amplitude exceeds a certain threshold value (threshold). It is usually specified in decibels, with a lower threshold (eg -60 dB) meaning that more audio will be processed than a higher threshold (eg -5 dB).

ratio

The amount of level reduction is determined by the ratio parameter: ratio 4:1 means that if the input level is 4 dB above the threshold, the output level will be 1 dB above the threshold.
For example:
Threshold = −10 dB
Input = −6 dB (4 dB above threshold)
Output = −9 dB (1 dB above threshold)

It is important to keep in mind that signal level suppression continues for some time after it falls below the threshold level, and this time is determined by the value of the parameter release.

Compression with a maximum ratio of ∞:1 is called limiting. This means that any signal above the threshold level is attenuated to the threshold level (except for a short period after a sudden increase in input volume). See “Limiter” below for more details.

Examples of different Ratio values

Attack and Release

A compressor provides some control over how quickly it responds to changes in signal dynamics. The Attack parameter determines the time it takes for the compressor to reduce the gain to a level determined by the Ratio parameter. Release determines the time during which the compressor, on the contrary, increases the gain, or returns to normal if the input signal level drops below the threshold value.

Attack and Release phases

These parameters indicate the time (usually in milliseconds) it will take to change the gain by a certain amount of decibels, usually 10 dB. For example, in this case, if Attack is set to 1 ms, it will take 1 ms to reduce the gain by 10 dB, and 2 ms to reduce the gain by 20 dB.

On many compressors the Attack and Release parameters can be adjusted, but on some they are pre-set and cannot be adjusted. Sometimes they are designated as “automatic” or “program dependent”, i.e. change depending on the input signal.

Knee

Another compressor parameter: hard/soft knee. It determines whether the start of compression will be abrupt (hard) or gradual (soft). Soft knee reduces the noticeability of the transition from the dry signal to the compressed signal, especially at high Ratio values ​​and sudden increases in volume.

Hard Knee and Soft Knee compression

Peak and RMS

The compressor can respond to peak (short-term maximum) values ​​or to the average level of the input signal. The use of peak values ​​can lead to sharp fluctuations in the degree of compression, and even distortion. Therefore, compressors apply an average function (usually RMS) to the input signal when comparing it to a threshold value. This gives a more comfortable compression, closer to human perception of loudness.

RMS is a parameter that reflects the average volume of a soundtrack. From a mathematical point of view, RMS (Root Mean Square) is the root mean square value of the amplitude of a certain number of samples:

Stereo linking

A compressor in stereo linking mode applies the same gain to both stereo channels. This avoids stereo shifts that may result from individual processing of the left and right channels. This shift occurs if, for example, a loud element is panned off-center.

Makeup gain

Since the compressor reduces the overall signal level, it usually adds a fixed output gain option to achieve the optimal level.

Look-ahead

The look-ahead function is designed to solve problems associated with both too high and too low Attack and Release values. An attack time that is too long does not allow us to effectively intercept transients, and an attack time that is too short may not be comfortable for the listener. When using the look-ahead function, the main signal is delayed relative to the control signal, this allows you to start compression in advance, even before the signal reaches the threshold value.
The only drawback of this method is the time delay of the signal, which in some cases is undesirable.

Using Dynamic Compression

Compression is used everywhere, not only in musical soundtracks, but also wherever it is necessary to increase the overall volume without increasing peak levels, where inexpensive sound reproducing equipment or a limited transmission channel is used (public address and communication systems, amateur radio, etc.) .

Compression is used when playing background music (in shops, restaurants, etc.), where any noticeable changes in volume are undesirable.

But the most important area of ​​application of dynamic compression is music production and broadcasting. Compression is used to give the sound "thickness" and "drive", to better combine instruments with each other, and especially when processing vocals.

Vocals in rock and pop music are often compressed to make them stand out from the accompaniment and add clarity. A special type of compressor tuned only to certain frequencies - a de-esser - is used to suppress sibilant phonemes.

In instrumental parts, compression is also used for effects that are not directly related to volume, for example, quickly decaying drum sounds can be made longer lasting.

Electronic dance music (EDM) often uses side-chaining (see below) - for example, the bass line may be driven by a kick drum or similar to prevent bass and drums from clashing and create a dynamic pulsation.

Compression is widely used in broadcast (radio, television, internet broadcasting) to increase perceived loudness while reducing the dynamic range of the source audio (usually CD). Most countries have legal restrictions on the maximum instantaneous volume that can be broadcast. Typically these limitations are implemented by permanent hardware compressors in the air chain. Additionally, increasing perceived loudness improves the "quality" of the sound from the perspective of most listeners.

See also Loudness war.

Consistently increasing the volume of the same song remastered for CD from 1983 to 2000.

Side-chaining

Another commonly encountered compressor switch is the “side chain”. In this mode, sound compression occurs not depending on its own level, but depending on the level of the signal entering the connector, which is usually called the side chain.

There are several uses for this. For example, the vocalist has a lisp and all the “s” stand out from the overall picture. You pass his voice through a compressor, and feed the same sound into the side chain connector, but passed through an equalizer. With an equalizer, you cut out all frequencies except those used by the vocalist when pronouncing the letter “s.” Typically around 5 kHz, but can range from 3 kHz to 8 kHz. If you then put the compressor in side chain mode, the voice will be compressed at those moments when the letter “s” is pronounced. This resulted in a device known as a de-esser. This way of working is called “frequency dependent”.

Another use of this function is called "ducker". For example, at a radio station, the music goes through a compressor, and the DJ's words come through a side chain. When the DJ starts chatting, the music volume automatically decreases. This effect can also be successfully used in recording, for example, to reduce the volume of keyboard parts while singing.

Brick wall limiting

The compressor and the limiter work approximately the same way; we can say that the limiter is a compressor with a high Ratio (from 10:1) and, usually, a low Attack time.

There is the concept of Brick wall limiting - limiting with a very high Ratio (20:1 and above) and a very fast attack. Ideally, it does not allow the signal to exceed the threshold level at all. The result will be unpleasant to the ear, but this will prevent damage to the sound-reproducing equipment or exceeding the channel capacity. Many manufacturers integrate limiters into their devices for this very purpose.

Clipper vs. Limiter, soft and hard clipping

The sound level is the same throughout the entire composition, with several pauses.

Narrowing dynamic range

Narrowing of the dynamic range, or more simply put compression, is necessary for various purposes, the most common of which are:

1) Achieving a single volume level throughout the entire composition (or instrument part).

2) Achieving a uniform volume level for songs throughout the album/radio broadcast.

2) Increased intelligibility, mainly when compressing a certain part (vocals, bass drum).

How does dynamic range narrowing occur?

The compressor analyzes the sound level at the input by comparing it to a user-specified Threshold value.

If the signal level is below the value Threshold– then the compressor continues to analyze the sound without changing it. If the sound level exceeds the Threshold value, then the compressor begins its action. Since the role of the compressor is to narrow the dynamic range, it is logical to assume that it limits the largest and smallest amplitude values ​​(signal level). At the first stage, the largest values ​​are limited, which are reduced with a certain force, which is called ratio(Attitude). Let's look at an example:

Green curves display the sound level; the greater the amplitude of their oscillations from the X axis, the greater the signal level.

The yellow line is the threshold (Threshold) for the compressor to operate. By making the Threshold value higher, the user moves it away from the X axis. By making the Threshold value lower, the user brings it closer to the Y axis. It is clear that the lower the threshold value, the more often the compressor will operate and vice versa, the higher it is, the less often. If the Ratio value is very high, then after the Threshold signal level is reached, all subsequent signals will be suppressed by the compressor until silence. If the Ratio value is very small, then nothing will happen. The choice of Threshold and Ratio values ​​will be discussed later. Now we should ask ourselves the following question: What is the point of suppressing all subsequent sound? Indeed, there is no point in this, we only need to get rid of the amplitude values ​​(peaks) that exceed the Threshold value (marked in red on the graph). It is to solve this problem that there is a parameter Release(Attenuation), which sets the duration of the compression.

The example shows that the first and second exceedances of the Threshold threshold last less than the third exceedance of the Threshold threshold. So, if the Release parameter is set to the first two peaks, then when processing the third, an unprocessed part may remain (since exceeding the Threshold threshold lasts longer). If the Release parameter is set to the third peak, then when processing the first and second peaks, an undesirable decrease in the signal level is formed behind them.

The same goes for the Ratio parameter. If the Ratio parameter is adjusted to the first two peaks, then the third one will not be sufficiently suppressed. If the Ratio parameter is configured to process the third peak, then the processing of the first two peaks will be too excessive.

These problems can be solved in two ways:

1) Setting the attack parameter (Attack) - a partial solution.

2) Dynamic compression - a complete solution.

Parameter Astill (Attack) is intended to set the time after which the compressor will start operating after exceeding the Threshold threshold. If the parameter is close to zero (equal to zero in the case of parallel compression, see the corresponding article) - then the compressor will begin to suppress the signal immediately, and will work for the amount of time specified by the Release parameter. If the attack speed is high, then the compressor will begin its action after a certain period of time has passed (this is necessary to give clarity). In our case, we can adjust the parameters of the threshold (Threshold), attenuation (Release) and compression level (Ratio) to process the first two peaks, and set the Attack value close to zero. Then the compressor will suppress the first two peaks, and when processing the third, it will suppress it until the threshold is exceeded (Threshold). However, this does not guarantee high-quality sound processing and is close to limiting (a rough cut of all amplitude values, in this case the compressor is called a limiter).

Let's look at the result of sound processing with a compressor:

The peaks disappeared, I note that the processing settings were quite gentle and we suppressed only the most prominent amplitude values. In practice, the dynamic range narrows much more and this trend only progresses. In the minds of many composers, they make the music louder, but in practice they completely deprive it of dynamics for those listeners who may be listening to it at home and not on the radio.

We just have to consider the last compression parameter, this Gain(Gain). Gain is designed to increase the amplitude of the entire composition and, in fact, is equivalent to another sound editor tool - normalize. Let's look at the final result:

In our case, compression was justified and improved the quality of the sound, since the prominent peak is more likely an accident than a deliberate result. In addition, it is clear that the music is rhythmic, therefore it has a narrow dynamic range. In cases where high amplitude values ​​are intentional, compression may be a mistake.

Dynamic compression

The difference between dynamic compression and non-dynamic compression is that with the former, the level of signal suppression (Ratio) depends on the level of the input signal. Dynamic compressors are found in all modern programs; the Ratio and Threshold parameters are controlled using a window (each parameter has its own axis):

There is no single standard for displaying a graph; somewhere along the Y axis the level of the incoming signal is displayed, somewhere on the contrary, the signal level after compression. Somewhere the point (0,0) is in the upper right corner, somewhere in the lower left. In any case, when you move the mouse cursor over this field, the values ​​of the numbers that correspond to the Ratio and Threshold parameters change. Those. You set the compression level for each Threshold value, allowing for very flexible compression settings.

Side Chain

A side chain compressor analyzes the signal of one channel, and when the sound level exceeds a threshold (threshold), it applies compression to another channel. Side chain has its advantages of working with instruments that are located in the same frequency region (the bass-kick combination is actively used), but sometimes instruments located in different frequency regions are also used, which leads to an interesting side-chain effect.

Part Two – Compression Stages

There are three stages of compression:

1) The first stage is compression of individual sounds (singleshoots).

The timbre of any instrument has the following characteristics: Attack, Hold, Decay, Delay, Sustain, Release.

The stage of compression of individual sounds is divided into two parts:

1.1) Compression of individual sounds of rhythmic instruments

Often the components of a beat require separate compression to give them clarity. Many people process the bass drum separately from other rhythmic instruments, both at the stage of compression of individual sounds and at the stage of compression of individual parts. This is due to the fact that it is located in the low-frequency region, where in addition to it, only bass is usually present. The clarity of a bass drum means the presence of a characteristic click (the bass drum has a very short attack and hold time). If there is no click, then you need to process it with a compressor, setting the threshold to zero and the attack time from 10 to 50 ms. The roll-off (Realese) of the compressor must end before the next kick drum hit. The last problem can be solved using the formula: 60,000 / BPM, where BPM is the tempo of the composition. So, for example) 60,000/137=437.96 (time in milliseconds until a new downbeat of a 4-dimensional composition).

All of the above applies to other rhythmic instruments with a short attack time - they should have an accentuated click that should not be suppressed by the compressor at any stage of the compression levels.

1.2) Compressionindividual soundsharmonic instruments

Unlike rhythmic instruments, parts of harmonic instruments are rarely composed of individual sounds. However, this does not mean that they should not be processed at the sound compression level. If you use a sample with a recorded part, then this is the second level of compression. Only synthesized harmonic instruments apply to this compression level. These can be samplers, synthesizers using various methods of sound synthesis (physical modeling, FM, additive, subtractive, etc.). As you probably already guessed, we are talking about programming the synthesizer settings. Yes! This is also compression! Almost all synthesizers have a programmable envelope parameter (ADSR), which means envelope. Using the envelope, you set the time of Attack, Decay, Sustain, and Release. And if you tell me that this is not compression of each individual sound - you are my enemy for life!

2) Second stage – Compression of individual parts.

By compression of individual parts I mean narrowing the dynamic range of a number of combined individual sounds. This stage also includes recordings of parts, including vocals, which require compression processing to give it clarity and intelligibility. When processing parts by compression, you need to take into account that when adding individual sounds, unwanted peaks may appear, which you need to get rid of at this stage, since if this is not done now, the picture may worsen at the stage of mixing the entire composition. At the stage of compression of individual parts, it is necessary to take into account the compression of the stage of processing individual sounds. If you have achieved clarity of the bass drum, then incorrect re-processing at the second stage can ruin everything. It is not necessary to process all parts with a compressor, just as it is not necessary to process all individual sounds. I advise you to install, just in case, an amplitude analyzer to determine the presence of unwanted side effects of combining individual sounds. In addition to compression, at this stage it is necessary to ensure that the parts are, if possible, in different frequency ranges so that quantization can be performed. It is also useful to remember that sound has such a characteristic as masking (psychoacoustics):

1) A quieter sound is masked by a louder one coming in front of it.

2) A quieter sound at a low frequency is masked by a louder sound at a high frequency.

So, for example, if you have a synthesizer part, then often the notes begin to play before the previous notes finish sounding. Sometimes this is necessary (creating harmony, playing style, polyphony), but sometimes it is not at all - you can cut off their end (Delay - Release) if it is audible in solo mode, but not audible in playback mode of all parts. The same applies to effects, such as reverb - it should not last until the sound source starts again. By cutting and removing unnecessary signal, you make the sound cleaner, and this can also be considered as compression - because you are removing unnecessary waves.

3) The third stage – Compression of the composition.

When compressing an entire composition, you need to take into account the fact that all parts are a combination of many individual sounds. Therefore, when combining them and subsequent compression, we need to make sure that the final compression does not spoil what we achieved in the first two stages. You also need to separate compositions in which a wide or narrow range is important. when compressing compositions with a wide dynamic range, it is enough to install a compressor that will crush short-term peaks that were formed as a result of adding parts together. When compressing a composition in which a narrow dynamic range is important, everything is much more complicated. Here compressors have recently been called maximizers. Maximizer is a plugin that combines a compressor, limiter, graphic equalizer, enhyzer and other sound transformation tools. At the same time, he must have sound analysis tools. Maximizing, the final processing with a compressor, is largely necessary to combat mistakes made at previous stages. Errors - not so much in compression (however, if you do at the last stage what you could have done at the first stage, this is already a mistake), but in the initial selection of good samples and instruments that would not interfere with each other (we are talking about frequency ranges) . This is precisely why the frequency response is corrected. It often happens that with strong compression on the master, it is necessary to change the compression and mixing parameters at earlier stages, since with a strong narrowing of the dynamic range, quiet sounds that were previously masked come out, and the sound of individual components of the composition changes.

In these parts, I deliberately did not talk about specific compression parameters. I considered it necessary to write about the fact that when compression it is necessary to pay attention to all sounds and all parts at all stages of creating a composition. This is the only way in the end you will get a harmonious result not only from the point of view of music theory, but also from the point of view of sound engineering.

The following table provides practical advice for processing individual batches. However, in compression, numbers and presets can only suggest the desired area in which to search. The ideal compression settings depend on each individual case. The Gain and Threshold parameters assume a normal sound level (logical use of the entire range).

Part Three - Compression Parameters

Brief information:

Threshold – determines the sound level of the incoming signal, upon reaching which the compressor starts working.

Attack – determines the time after which the compressor will start working.

Level (ratio) – determines the degree of reduction in amplitude values ​​(relative to the original amplitude value).

Release – defines the time after which the compressor will stop working.

Gain – determines the level of increase in the incoming signal after processing by a compressor.

Compression table:

The information was taken from various sources referenced by popular resources on the Internet. The difference in compression parameters is explained by different sound preferences and working with different materials.

, Media players

Records, especially older ones that were recorded and produced before 1982, were much less likely to be mixed to make the recording louder. They reproduce natural music with a natural dynamic range that is preserved on the record and lost in most standard digital or high-definition formats.

There are exceptions to this, of course—listen to Steven Wilson's recent album from MA Recordings or Reference Recordings and you'll hear just how good digital audio can be. But this is rare; most modern sound recordings are loud and compressed.

Music compression has come under a lot of criticism lately, but I'm willing to bet that almost all of your favorite recordings are compressed. Some of them are less, some are more, but still compressed. Dynamic range compression is a scapegoat for bad music sound, but highly compressed music isn't a new trend: listen to Motown albums from the '60s. The same can be said about the classic works of Led Zeppelin or the younger albums of Wilco and Radiohead. Dynamic range compression reduces the natural relationship between the loudest and softest sounds in a recording, so a whisper can be as loud as a scream. It's quite difficult to find pop music from the last 50 years that hasn't been compressed.

I recently had a nice chat with Tape Op magazine founder and editor Larry Crane about the good, the bad, and the ugly aspects of compression. Larry Crane has worked with bands and artists such as Stefan Marcus, Cat Power, Sleater-Kinney, Jenny Lewis, M. Ward, The Go-Betweens, Jason Little, Eliot Smith, Quasi and Richmond Fontaine. He also runs the recording studio Jackpot! in Portland, Oregon, which was home to The Breeders, The Decemberists, Eddie Vedder, Pavement, R.E.M., She & Him and many, many others.

As an example of surprisingly unnatural-sounding but still great songs, I cite Spoon's 2014 album They Want My Soul. Crane laughs and says he listens to it in the car because it sounds great there. Which brings us to another answer to the question of why music is compressed: because compression and additional “clarity” make it easier to hear in noisy places.

Larry Crane at work. Photo by Jason Quigley

When people say they like the sound of an audio recording, I think they like the music, as if sound and music were inseparable terms. But for myself, I differentiate these concepts. From an audiophile's perspective, the sound may be rough and raw, but that won't matter to most listeners.

Many are quick to accuse mastering engineers of overusing compression, but compression is applied directly during recording, during mixing, and only then during mastering. Unless you were personally present at each of these stages, you will not be able to say what the instruments and vocal parts sounded like at the very beginning of the process.

Crane was on a roll: “If a musician wants to intentionally sound crazy and distorted like the Guided by Voices records, then there’s nothing wrong with that—the desire always outweighs the sound quality.” The performer's voice is almost always compressed, and the same thing happens with bass, drums, guitars and synthesizers. With compression, the volume of vocals remains at the desired level throughout the song or is slightly raised from the background of other sounds.

Properly done compression can make drums sound more lively or intentionally strange. To make music sound great, you need to be able to use the necessary tools. This is why it takes years to figure out how to use compression without overdoing it. If the mix engineer compresses the guitar part too much, the mastering engineer will no longer be able to fully restore the missing frequencies.

If musicians wanted you to listen to music that had not gone through the stages of mixing and mastering, they would release it onto store shelves straight from the studio. Crane says the people who create, edit, mix and master recorded music aren't there to get in the way of musicians - they've been helping artists since the beginning, for more than a hundred years.

These people are part of the creation process that results in amazing works of art. Crane adds, "You don't want a version of 'Dark Side of the Moon' that hasn't been mixed and mastered." Pink Floyd released the song the way they wanted to hear it.

People who are passionate about home audio exhibit an interesting paradox. They are ready to shovel the listening room, build speakers with exotic drivers, but shyly retreat in front of the canned music, like a wolf in front of a red flag. But actually, why can’t you stand up for the flag and try to cook something more edible from canned food?

From time to time, plaintive questions arise on the forum: “Recommend well-recorded albums.” This is understandable. Special audiophile publications, although they delight the ear for the first minute, no one listens to them until the end, the repertoire is too dull. As for the rest of the music library, the problem seems obvious. You can save, or you can not save and pour a ton of money into components. Still, few people like to listen to their favorite music at high volumes and the capabilities of the amplifier have nothing to do with it.

Today, even in Hi-Res albums, the peaks of the soundtrack are cut off and the volume is driven into clipping. It is believed that the majority listens to music on all kinds of junk, and therefore it is necessary to “step on the gas”, to make a kind of loud compensation.

Of course, this is not done on purpose to upset audiophiles. Few people remember them at all. Perhaps they thought of giving them the master files from which the main circulation is copied - CDs, MP3s, etc. Of course, the master has long been flattened by a compressor; no one will deliberately prepare special versions for HD Tracks. Unless a certain procedure is performed for vinyl media, which for this reason sounds more humane. But for the digital route it all ends the same way - with a big fat compressor.

So, currently, 100% of published phonograms, minus classical music, are subject to compression during mastering. Some perform this procedure more or less skillfully, while others perform it completely stupidly. As a result, we have pilgrims on the forums with a line of DR plugins in their bosoms, painful comparisons of editions, a flight to vinyl, where we also need to mine first presses.

The most frostbitten at the sight of all these outrages literally turned into audio Satanists. No joke, they are reading the sound engineer's holy scripture backwards! Modern audio editing programs have some tools for restoring a clipped sound wave.

Initially, this functionality was intended for studios. When mixing, there are situations when clipping gets into the recording, and for a number of reasons it is no longer possible to redo the session, and here the arsenal of an audio editor comes to the rescue - a declipper, a decompressor, etc.

And now ordinary listeners, whose ears are bleeding after the next new product, are increasingly reaching out to such software more and more boldly. Some people prefer iZotope, some prefer Adobe Audition, others divide operations between several programs. The point of restoring the previous dynamics is to programmatically correct the clipped signal peaks, which, resting at 0 dB, resemble a gear.

Yes, there is no talk of a 100% revival of the source code, since interpolation processes take place using rather speculative algorithms. But still, some of the processing results seemed interesting to me and worthy of study.

For example, Lana Del Rey’s album “Lust For Life”, consistently swearing, ugh, swearing! The original song “When the World Was at War We Kept Dancing” was like this.

And after a series of declippers and decompressors it became like this. The DR coefficient has changed from 5 to 9. You can download and listen to the sample before and after processing.

I can’t say that the method is universal and suitable for all ruined albums, but in this case I chose to keep this version in the collection, processed by a root tracker activist, instead of the official 24-bit edition.

Even if artificially extracting peaks from the sound stuff does not return the true dynamics of a musical performance, your DAC will still thank you. After all, it was so difficult for him to work without errors at extreme levels, where there is a high probability of so-called inter-sample peaks (ISP) occurring. And now only rare flashes of the signal will jump to 0 dB. In addition, a silent soundtrack when compressed into FLAC or another lossless codec will now be smaller in size. More “air” in the signal saves hard drive space.

Try to revive your most hated albums that were killed in the “loudness war.” To reserve dynamics, you first need to lower the track level by -6 dB, and then run the declipper. Those who don't trust computers can simply plug a studio expander between the CD player and the amplifier. This device essentially does the same thing - it restores and stretches the peaks of a dynamically compressed audio signal as best it can. Similar devices from the 80-90s are not very expensive, and it will be very interesting to try them as an experiment.

Once upon a time, equalizers were a taken-for-granted component of an audio system, and no one was afraid of them. Today there is no need to level out the high-frequency roll of a magnetic tape, but something needs to be done about the ugly dynamics, brothers.