Basic concepts in acoustics. Maximum volume and pitch Sound limit

Youtube

Psychoacoustics, a field of science bordering between physics and psychology, studies data on a person’s auditory sensation when a physical stimulus—sound—is applied to the ear. A large amount of data has been accumulated on human reactions to auditory stimuli. Without this data, it is difficult to obtain a correct understanding of the operation of audio transmission systems. Let's consider the most important features of human perception of sound.
A person feels changes in sound pressure occurring at a frequency of 20-20,000 Hz. Sounds with frequencies below 40 Hz are relatively rare in music and do not exist in spoken language. Very high frequencies Ah, the musical perception disappears and a certain vague sound sensation arises, depending on the individuality of the listener, his age. With age, a person's hearing sensitivity decreases, primarily in the upper frequencies of the sound range.
But it would be wrong to conclude on this basis that the transmission of a wide frequency band by a sound-reproducing installation is unimportant for older people. Experiments have shown that people, even if they can barely perceive signals above 12 kHz, very easily recognize the lack of high frequencies in a musical transmission.

Frequency characteristics of auditory sensations

The range of sounds audible to humans in the range of 20-20,000 Hz is limited in intensity by thresholds: below - audibility and above - pain.
The hearing threshold is estimated by the minimum pressure, or more precisely, the minimum increment of pressure relative to the boundary is sensitive to frequencies of 1000-5000 Hz - here the hearing threshold is the lowest (sound pressure about 2-10 Pa). Towards the lower and higher audio frequencies hearing sensitivity drops sharply.
The pain threshold determines the upper limit of the perception of sound energy and corresponds approximately to a sound intensity of 10 W/m or 130 dB (for a reference signal with a frequency of 1000 Hz).
As sound pressure increases, the intensity of the sound also increases, and the auditory sensation increases in leaps, called the intensity discrimination threshold. The number of these jumps at medium frequencies is approximately 250, at low and high frequencies it decreases and on average over the frequency range is about 150.

Since the range of intensity changes is 130 dB, the elementary jump in sensations on average over the amplitude range is 0.8 dB, which corresponds to a change in sound intensity by 1.2 times. At low levels hearing these jumps reach 2-3 dB, at high levels they decrease to 0.5 dB (1.1 times). An increase in the power of the amplification path by less than 1.44 times is practically not detected by the human ear. With a lower sound pressure developed by the loudspeaker, even doubling the power of the output stage may not produce a noticeable result.

Subjective sound characteristics

The quality of sound transmission is assessed based on auditory perception. Therefore, it is correct to determine technical requirements to the sound transmission path or its individual links is possible only by studying the patterns connecting the subjectively perceived sensation of sound and the objective characteristics of sound are height, volume and timbre.
The concept of pitch implies a subjective assessment of the perception of sound across the frequency range. Sound is usually characterized not by frequency, but by pitch.
A tone is a signal of a certain pitch that has a discrete spectrum (musical sounds, vowel sounds of speech). A signal that has a wide continuous spectrum, all frequency components of which have the same average power, is called white noise.

A gradual increase in the frequency of sound vibrations from 20 to 20,000 Hz is perceived as a gradual change in tone from the lowest (bass) to the highest.
The degree of accuracy with which a person determines the pitch of a sound by ear depends on the acuity, musicality and training of his ear. It should be noted that the pitch of a sound depends to some extent on the intensity of the sound (at high levels, sounds of greater intensity appear lower than weaker ones.
The human ear can clearly distinguish two tones that are close in pitch. For example, in the frequency range of approximately 2000 Hz, a person can distinguish between two tones that differ from each other in frequency by 3-6 Hz.
The subjective scale of sound perception in frequency is close to the logarithmic law. Therefore, doubling the vibration frequency (regardless of the initial frequency) is always perceived as the same change in pitch. The height interval corresponding to a 2-fold change in frequency is called an octave. The range of frequencies perceived by humans is 20-20,000 Hz, which covers approximately ten octaves.
An octave is a fairly large interval of change in pitch; a person distinguishes significantly smaller intervals. Thus, in ten octaves perceived by the ear, more than a thousand gradations of pitch can be distinguished. Music uses smaller intervals called semitones, which correspond to a change in frequency of approximately 1.054 times.
An octave is divided into half octaves and a third of an octave. For the latter, the following range of frequencies is standardized: 1; 1.25; 1.6; 2; 2.5; 3; 3.15; 4; 5; 6.3:8; 10, which are the boundaries of one-third octaves. If these frequencies are placed at equal distances along the frequency axis, you get a logarithmic scale. Based on this, all frequency characteristics of sound transmission devices are plotted on a logarithmic scale.
The loudness of the transmission depends not only on the intensity of the sound, but also on the spectral composition, the conditions of perception and the duration of exposure. So, two sounding tones, middle and low frequency, having the same intensity (or the same sound pressure), are not perceived by a person as equally loud. Therefore, the concept of loudness level in backgrounds was introduced to designate sounds of the same loudness. Behind volume level sound in the backgrounds take the sound pressure level in decibels of the same volume of a pure tone with a frequency of 1000 Hz, i.e. for a frequency of 1000 Hz the volume levels in backgrounds and decibels are the same. At other frequencies, sounds may appear louder or quieter at the same sound pressure.
The experience of sound engineers in recording and editing musical works shows that in order to better detect sound defects that may arise during work, the volume level during control listening should be maintained high, approximately corresponding to the volume level in the hall.
With prolonged exposure to intense sound, hearing sensitivity gradually decreases, and the more, the higher the sound volume. The detected decrease in sensitivity is associated with the reaction of hearing to overload, i.e. with its natural adaptation. After some break in listening, hearing sensitivity is restored. It should be added to this that the hearing aid, when perceiving high-level signals, introduces its own, so-called subjective, distortions (which indicates the nonlinearity of hearing). Thus, at a signal level of 100 dB, the first and second subjective harmonics reach levels of 85 and 70 dB.
A significant level of volume and the duration of its exposure cause irreversible phenomena in the auditory organ. It has been noted that young people's hearing thresholds have increased sharply in recent years. The reason for this was a passion for pop music, characterized by high sound volume levels.
The volume level is measured using an electroacoustic device - a sound level meter. The sound being measured is first converted into electrical vibrations by the microphone. After amplification by a special voltage amplifier, these oscillations are measured with a pointer instrument adjusted in decibels. In order for the device readings to correspond as accurately as possible to the subjective perception of loudness, the device is equipped with special filters that change its sensitivity to the perception of sound different frequencies in accordance with the characteristics of hearing sensitivity.
An important characteristic of sound is timbre. The ability of hearing to distinguish it allows you to perceive signals with a wide variety of shades. The sound of each of the instruments and voices, thanks to their characteristic shades, becomes multicolored and well recognizable.
Timbre, being a subjective reflection of the complexity of the perceived sound, has no quantitative assessment and is characterized by qualitative terms (beautiful, soft, juicy, etc.). When transmitting a signal along an electroacoustic path, the resulting distortions primarily affect the timbre of the reproduced sound. The condition for the correct transmission of the timbre of musical sounds is the undistorted transmission of the signal spectrum. The signal spectrum is the collection of sinusoidal components of a complex sound.
The simplest spectrum is the so-called pure tone; it contains only one frequency. The sound of a musical instrument is more interesting: its spectrum consists of the frequency of the fundamental tone and several “impurity” frequencies called overtones (higher tones). Overtones are a multiple of the frequency of the fundamental tone and are usually smaller in amplitude.
The timbre of the sound depends on the distribution of intensity over overtones. The sounds of different musical instruments vary in timbre.
More complex is the spectrum of combinations of musical sounds called a chord. In such a spectrum there are several fundamental frequencies along with corresponding overtones
Differences in timbre are mainly due to the low-mid frequency components of the signal, therefore, a large variety of timbres is associated with signals lying in the lower part of the frequency range. Signals belonging to its upper part, as they increase, increasingly lose their timbre coloring, which is due to the gradual departure of their harmonic components beyond the limits of audible frequencies. This can be explained by the fact that up to 20 or more harmonics are actively involved in the formation of the timbre of low sounds, medium 8 - 10, high 2 - 3, since the rest are either weak or fall outside the range of audible frequencies. Therefore, high sounds, as a rule, are poorer in timbre.
Almost all natural sound sources, including sources of musical sounds, have a specific dependence of timbre on volume level. Hearing is also adapted to this dependence - it is natural for it to determine the intensity of a source by the color of the sound. Louder sounds are usually more harsh.

Musical sound sources

A number of factors characterizing the primary sound sources have a great influence on the sound quality of electroacoustic systems.
The acoustic parameters of musical sources depend on the composition of the performers (orchestra, ensemble, group, soloist and type of music: symphonic, folk, pop, etc.).

The origin and formation of sound on each musical instrument has its own specifics associated with the acoustic characteristics of sound production in a particular musical instrument.
An important element of musical sound is attack. This is a specific transition process during which stable characteristics sound: volume, timbre, pitch. Any musical sound goes through three stages - beginning, middle and end, and both the initial and final stages have a certain duration. The initial stage is called an attack. It lasts differently: for plucked instruments, percussion and some wind instruments it lasts 0-20 ms, for the bassoon it lasts 20-60 ms. An attack is not just an increase in the volume of a sound from zero to some steady value; it can be accompanied by the same change in the pitch of the sound and its timbre. Moreover, the attack characteristics of the instrument are not the same in different parts of its range with different playing styles: the violin is the most perfect instrument in terms of the wealth of possible expressive methods of attack.
One of the characteristics of any musical instrument is its frequency range. In addition to the fundamental frequencies, each instrument is characterized by additional high-quality components - overtones (or, as is customary in electroacoustics, higher harmonics), which determine its specific timbre.
It is known that sound energy is unevenly distributed across the entire spectrum of sound frequencies emitted by a source.
Most instruments are characterized by amplification of fundamental frequencies, as well as individual overtones, in certain (one or more) relatively narrow frequency bands (formants), different for each instrument. Resonant frequencies (in hertz) of the formant region are: for trumpet 100-200, horn 200-400, trombone 300-900, trumpet 800-1750, saxophone 350-900, oboe 800-1500, bassoon 300-900, clarinet 250-600 .
Another characteristic property of musical instruments is the strength of their sound, which is determined by the greater or lesser amplitude (span) of their sounding body or air column (a greater amplitude corresponds to a stronger sound and vice versa). The peak acoustic power values (in watts) are: for large orchestra 70, bass drum 25, timpani 20, snare drum 12, trombone 6, piano 0.4, trumpet and saxophone 0.3, trumpet 0.2, double bass 0.( 6, small flute 0.08, clarinet, horn and triangle 0.05.
The ratio of the sound power extracted from an instrument when played “fortissimo” to the power of sound when played “pianissimo” is usually called the dynamic range of the sound of musical instruments.
The dynamic range of a musical sound source depends on the type of performing group and the nature of the performance.
Let's consider dynamic range separate sound sources. The dynamic range of individual musical instruments and ensembles (orchestras and choirs of various compositions), as well as voices, is understood as the ratio of the maximum sound pressure created by a given source to the minimum, expressed in decibels.
In practice, when determining the dynamic range of a sound source, one usually operates only on sound pressure levels, calculating or measuring their corresponding difference. For example, if the maximum sound level of an orchestra is 90 and the minimum is 50 dB, then the dynamic range is said to be 90 - 50 = 40 dB. In this case, 90 and 50 dB are sound pressure levels relative to zero acoustic level.
Dynamic range for this source sound is a variable quantity. It depends on the nature of the work being performed and on the acoustic conditions of the room in which the performance takes place. Reverberation expands the dynamic range, which typically reaches its maximum in rooms with large volumes and minimal sound absorption. Almost all instruments and human voices have an uneven dynamic range across sound registers. For example, the volume level of the lowest sound on a forte for a vocalist is equal to the level of the highest sound on a piano.

The dynamic range of one or another music program is expressed in the same way as for individual sound sources, but the maximum sound pressure is observed with a dynamic ff (fortissimo) tone, and the minimum with pp (pianissimo).

The highest volume, indicated in the notes fff (forte, fortissimo), corresponds to an acoustic sound pressure level of approximately 110 dB, and the lowest volume, indicated in the notes ppr (piano-pianissimo), approximately 40 dB.
It should be noted that the dynamic nuances of performance in music are relative and their relationship with the corresponding sound pressure levels is to some extent conditional. The dynamic range of a particular musical program depends on the nature of the composition. Thus, the dynamic range of classical works by Haydn, Mozart, Vivaldi rarely exceeds 30-35 dB. The dynamic range of pop music usually does not exceed 40 dB, while that of dance and jazz music is only about 20 dB. Most works for orchestra of Russian folk instruments also have a small dynamic range (25-30 dB). This is also true for a brass band. However, the maximum sound level of a brass band in a room can reach a fairly high level (up to 110 dB).

Masking effect

The subjective assessment of loudness depends on the conditions in which the sound is perceived by the listener. In real conditions, an acoustic signal does not exist in absolute silence. At the same time, extraneous noise affects the hearing, complicating sound perception, masking to a certain extent the main signal. The effect of masking a pure sine wave by extraneous noise is measured by the value indicating. by how many decibels the threshold of audibility of the masked signal increases above the threshold of its perception in silence.
Experiments to determine the degree of masking of one sound signal by another show that a tone of any frequency is masked by lower tones much more effectively than by higher ones. For example, if two tuning forks (1200 and 440 Hz) emit sounds with the same intensity, then we stop hearing the first tone, it is masked by the second (by extinguishing the vibration of the second tuning fork, we will hear the first again).
If two complex sound signals consisting of certain sound frequency spectra exist simultaneously, then a mutual masking effect occurs. Moreover, if the main energy of both signals lies in the same region of the audio frequency range, then the masking effect will be the strongest. Thus, when transmitting an orchestral piece, due to masking by the accompaniment, the soloist’s part may become poorly intelligible and inaudible.
Achieving clarity or, as they say, “transparency” of sound in the sound transmission of orchestras or pop ensembles becomes very difficult if an instrument or individual groups of orchestra instruments play in one or similar registers at the same time.
The director, when recording an orchestra, must take into account the features of camouflage. At rehearsals, with the help of the conductor, he establishes a balance between the strength of the sound of the instruments of one group, as well as between the groups of the entire orchestra. The clarity of the main melodic lines and individual musical parts is achieved in these cases by the close placement of microphones to the performers, the deliberate highlighting by the sound engineer of the most important this place works of instruments and other special sound engineering techniques.
The phenomenon of masking is opposed by the psychophysiological ability of the hearing organs to single out from the general mass of sounds one or more that carry the most important information. For example, when an orchestra is playing, the conductor notices the slightest inaccuracies in the performance of a part on any instrument.
Masking can significantly affect the quality of signal transmission. A clear perception of the received sound is possible if its intensity significantly exceeds the level of interference components located in the same band as the received sound. With uniform interference, the signal excess should be 10-15 dB. This feature of auditory perception is practical use, for example, when assessing the electroacoustic characteristics of media. So, if the signal-to-noise ratio of an analog record is 60 dB, then the dynamic range of the recorded program can be no more than 45-48 dB.

Temporal characteristics of auditory perception

Hearing aid, like any other oscillatory system, is inertial. When the sound disappears, the auditory sensation does not disappear immediately, but gradually, decreasing to zero. The time during which the noise level decreases by 8-10 backgrounds is called the hearing time constant. This constant depends on a number of circumstances, as well as on the parameters of the perceived sound. If two short sound pulses arrive to the listener, identical in frequency composition and level, but one of them is delayed, then they will be perceived together with a delay not exceeding 50 ms. At large delay intervals, both impulses are perceived separately, and an echo occurs.
This feature of hearing is taken into account when designing some signal processing devices, for example, electronic delay lines, reverberates, etc.
It should be noted that, due to the special property of hearing, the sensation of the volume of a short-term sound pulse depends not only on its level, but also on the duration of the pulse’s impact on the ear. Thus, a short-term sound, lasting only 10-12 ms, is perceived by the ear quieter than a sound of the same level, but affecting the hearing for, for example, 150-400 ms. Therefore, when listening to a broadcast, loudness is the result of averaging the energy of the sound wave over a certain interval. In addition, human hearing has inertia, in particular, when perceiving nonlinear distortions, it does not feel them if the duration of the sound pulse is less than 10-20 ms. That is why in level indicators of sound recording household radio-electronic equipment, the instantaneous signal values are averaged over a period selected in accordance with the temporal characteristics of the hearing organs.

Spatial representation of sound

One of the important human abilities is the ability to determine the direction of a sound source. This ability is called the binaural effect and is explained by the fact that a person has two ears. Experimental data shows where the sound comes from: one for high-frequency tones, one for low-frequency tones.

The sound travels a shorter distance to the ear facing the source than to the other ear. As a result, the pressure of sound waves in the ear canals varies in phase and amplitude. The amplitude differences are significant only at high frequencies, when the sound wavelength becomes comparable to the size of the head. When the difference in amplitude exceeds a threshold value of 1 dB, the sound source appears to be on the side where the amplitude is greater. The angle of deviation of the sound source from the center line (line of symmetry) is approximately proportional to the logarithm of the amplitude ratio.
To determine the direction of a sound source with frequencies below 1500-2000 Hz, phase differences are significant. It seems to a person that the sound comes from the side from which the wave, which is ahead in phase, reaches the ear. The angle of deviation of sound from the midline is proportional to the difference in the time of arrival of sound waves to both ears. A trained person can notice a phase difference with a time difference of 100 ms.
The ability to determine the direction of sound in the vertical plane is much less developed (about 10 times). This physiological feature is associated with the orientation of the hearing organs in the horizontal plane.
A specific feature of spatial perception of sound by a person is manifested in the fact that the hearing organs are able to sense the total, integral localization created with the help of artificial means of influence. For example, in a room, two speakers are installed along the front at a distance of 2-3 m from each other. The listener is located at the same distance from the axis of the connecting system, strictly in the center. In a room, two sounds of equal phase, frequency and intensity are emitted through the speakers. As a result of the identity of the sounds passing into the organ of hearing, a person cannot separate them; his sensations give ideas about a single, apparent (virtual) sound source, which is located strictly in the center on the axis of symmetry.
If we now reduce the volume of one speaker, the apparent source will move towards the louder speaker. The illusion of a sound source moving can be obtained not only by changing the signal level, but also by artificially delaying one sound relative to another; in this case, the apparent source will shift towards the speaker emitting the signal in advance.
To illustrate integral localization, we give an example. The distance between the speakers is 2 m, the distance from the front line to the listener is 2 m; in order for the source to move 40 cm to the left or right, it is necessary to submit two signals with a difference in intensity level of 5 dB or with a time delay of 0.3 ms. With a level difference of 10 dB or a time delay of 0.6 ms, the source will “move” 70 cm from the center.
Thus, if you change the sound pressure created by the speaker, the illusion of moving the sound source arises. This phenomenon is called summary localization. To create summary localization, a two-channel stereophonic sound transmission system is used.
Two microphones are installed in the primary room, each of which works on its own channel. The secondary has two loudspeakers. The microphones are located at a certain distance from each other along a line parallel to the placement of the sound emitter. When moving the sound emitter, different sound pressure will act on the microphone and the time of arrival of the sound wave will be different due to the unequal distance between the sound emitter and the microphones. This difference creates the effect of total localization in the secondary room, as a result of which the apparent source is localized in a certain point in space located between two speakers.
It should be said about the binaural sound transmission system. With this system, called an artificial head system, two separate microphones are placed in the primary room, spaced at a distance from each other equal to the distance between a person's ears. Each of the microphones has an independent sound transmission channel, the output of which in the secondary room includes telephones for the left and right ears. If the sound transmission channels are identical, such a system accurately conveys the binaural effect created near the ears of the “artificial head” in the primary room. Having headphones and having to use them for a long time is a disadvantage.
The organ of hearing determines the distance to the sound source using a number of indirect signs and with some errors. Depending on whether the distance to signal source, its subjective assessment changes under the influence of various factors. It was found that if the determined distances are small (up to 3 m), then their subjective assessment is almost linearly related to the change in the volume of the sound source moving along the depth. An additional factor for a complex signal is its timbre, which becomes increasingly “heavier” as the source approaches the listener. This is due to the increasing amplification of low overtones compared to high overtones, caused by the resulting increase in volume level.
For average distances of 3-10 m, moving the source away from the listener will be accompanied by a proportional decrease in volume, and this change will apply equally to the fundamental frequency and harmonic components. As a result, there is a relative strengthening of the high-frequency part of the spectrum and the timbre becomes brighter.
As the distance increases, energy losses in the air will increase in proportion to the square of the frequency. Increased loss of high register overtones will result in decreased timbral brightness. Thus, the subjective assessment of distances is associated with changes in its volume and timbre.
In a closed room, the signals of the first reflections, delayed relative to the direct reflection by 20-40 ms, are perceived by the hearing organ as coming from different directions. At the same time, their increasing delay creates the impression of a significant distance from the points from which these reflections occur. Thus, by the delay time one can judge the relative distance of secondary sources or, what is the same, the size of the room.

Some features of the subjective perception of stereophonic broadcasts.

A stereophonic sound transmission system has a number of significant features compared to a conventional monophonic one.
The quality that distinguishes stereophonic sound, volume, i.e. natural acoustic perspective can be assessed using some additional indicators that do not make sense with a monophonic sound transmission technique. Such additional indicators include: hearing angle, i.e. the angle at which the listener perceives the stereophonic sound picture; stereo resolution, i.e. subjectively determined localization of individual elements of the sound image at certain points in space within the audibility angle; acoustic atmosphere, i.e. the effect of giving the listener a feeling of presence in the primary room where the transmitted sound event occurs.

On the role of room acoustics

Colorful sound is achieved not only with the help of sound reproduction equipment. Even with fairly good equipment, the sound quality may be poor if the listening room does not have certain properties. It is known that in a closed room a nasal sound phenomenon called reverberation occurs. By affecting the organs of hearing, reverberation (depending on its duration) can improve or worsen sound quality.

A person in a room perceives not only direct sound waves created directly by the sound source, but also waves reflected by the ceiling and walls of the room. Reflected waves are heard for some time after the sound source has stopped.
It is sometimes believed that reflected signals only play a negative role, interfering with the perception of the main signal. However, this idea is incorrect. A certain part of the energy of the initial reflected echo signals, reaching the human ears with short delays, amplifies the main signal and enriches its sound. In contrast, later reflected echoes. whose delay time exceeds a certain critical value, form a sound background that makes it difficult to perceive the main signal.
The listening room should not have big time reverberation. Living rooms, as a rule, have little reverberation due to their limited size and the presence of sound-absorbing surfaces, upholstered furniture, carpets, curtains, etc.
Obstacles of different nature and properties are characterized by a sound absorption coefficient, which is the ratio of the absorbed energy to the total energy of the incident sound wave.

To increase the sound-absorbing properties of the carpet (and reduce noise in the living room), it is advisable to hang the carpet not close to the wall, but with a gap of 30-50 mm).

Psychoacoustics is a science that studies the psychological and physiological characteristics of human perception of sound.

Prerequisites

In many applications in acoustics and audio signal processing, it is necessary to know what people hear. The sound produced by air pressure waves can be accurately measured with modern equipment. However, understanding how these waves are received and reflected in our brain is not such a simple task. Sound is a continuous analog signal which (assuming air molecules are infinitesimal) can theoretically carry an infinite amount of information (there could be an infinite number of frequencies containing amplitude and phase information).

Understanding the processes of perception will allow scientists and engineers to focus on the capabilities of hearing and ignore the less important capabilities of other systems. It is also important to note that the question “what a person hears” is not only a question about the physiological capabilities of the ear, but in many ways also a question of psychology, clarity of perception.

Limits of sound perception

The human ear nominally hears sounds in the range of 20 to 20,000 Hz. The upper limit tends to decrease with age. Most adults cannot hear above 16 kHz. The ear itself does not respond to frequencies below 20 Hz, but they can be felt through the senses of touch.

The frequency resolution of the sound in the middle of the range is about 2 Hz. That is, a change in frequency of more than 2 Hz is felt. However, it is possible to hear an even smaller difference. For example, if both tones arrive simultaneously, as a result of the addition of two oscillations, a modulation of the signal amplitude occurs with a frequency equal to the difference in the original frequencies. This effect is also known as runout.

The range of loudness of perceived sounds is enormous. Our eardrum in the ear is only sensitive to changes in pressure. The volume of sound is usually measured in decibels (dB). The lower threshold of audibility is defined as 0 dB, and the definition of the upper limit of audibility refers rather to the question at what volume the ear will begin to destroy. This limit depends on how much we hear the sound. The ear can tolerate short-term increases in volume up to 120 dB without consequences, but long-term perception of sounds louder than 80 dB can cause hearing loss.

More careful studies of the lower limit of hearing have shown that the minimum threshold at which sound remains audible depends on frequency. This graph is called the absolute hearing threshold. On average, it has a region of greatest sensitivity in the range from 1 kHz to 5 kHz, although sensitivity decreases above 2 kHz with age.

The absolute hearing threshold curve is a special case of the more general curves of equal loudness. Equal loudness curves are lines along which a person perceives different frequencies of sound to be equally loud. The curves were first obtained by H Fletcher and W A Munson, and published in the work "Loudness, its definition, measurement and calculation" in J. Acoust. Soc Am.5, 82-108 (1933). Later, more accurate measurements were performed by Robinson and Dutson (D W Robinson and R S Dadson “A re-determination of the equal-loudness relations for pure tones” in Br. J. Appl. Phys. 7, 166-181, 1956). The resulting curves differ significantly, but this is not an error, but different conditions carrying out measurements. Fletcher and Manson used headphones as the source of sound waves, and Robinson and Dutson used a front-facing speaker in an anechoic room.

Robinson and Dutson's measurements formed the basis of the ISO 226 standard in 1986. In 2003, the ISO 226 standard was updated to include data collected from 12 international studios.

What do we hear

Human hearing is in many ways similar to a spectral analyzer, that is, the ear recognizes the spectral composition of sound waves without analyzing the phase of the wave. In reality, phase information is recognized and is very important for the directional perception of sound, but this function is performed by the parts of the brain responsible for processing sound. The difference between the phases of sound waves arriving at the right and left ear allows one to determine the direction to the sound source, and information about the phase difference is of paramount importance, in contrast to changes in the volume of the sound perceived by different ears. The filtering effect of head transfer functions also plays an important role in this.

Masking effect

In certain cases, one sound may be hidden by another sound. For example, a conversation at a bus stop may be completely impossible if a noisy bus is approaching. This effect is called masking. They say that faint sound masked if it becomes indistinguishable in the presence of a louder sound.

There are several types of camouflage:

According to the time of arrival of the masking and masked sound:

simultaneous (monaural) masking
temporary (non-simultaneous) masking

By type of masking and masked sounds:

pure tone pure tone of different frequencies
pure tone noise
speaking in clear tones
speech monotonous noise
speech with impulse sounds, etc.

Simultaneous camouflage

Any two sounds, when heard at the same time, influence the perception of the relative loudness between them. A louder sound reduces the perception of a weaker one, until its audibility disappears. The closer the frequency of the masked sound is to the frequency of the masking one, the more strongly it will be hidden. The masking effect is not the same when the masked sound is shifted lower or higher in frequency relative to the masking one. Lower-frequency sounds mask higher-frequency sounds more.

Temporary disguise

This phenomenon is similar to frequency masking, but here the masking occurs over time. When the masking sound stops, the person being masked continues to be inaudible for some time. Under normal conditions, the effect of temporary masking lasts much less. The masking time depends on the frequency and amplitude of the signal and can reach 100 ms.

In the case when the masking tone appears in time earlier than the masked one, the effect is called post-masking. When the masking tone appears later than the masked one (this is also possible), the effect is called pre-masking.

Post-stimulus fatigue

Often, after exposure to loud, high-intensity sounds, a person’s hearing sensitivity sharply decreases. Restoring normal thresholds can take up to 16 hours. This process is called "temporary shift in auditory threshold" or "post-stimulus fatigue." The threshold shift begins to appear at sound pressure levels above 75 dB and increases accordingly as the signal level increases. Moreover, the greatest influence on the shift of the sensitivity threshold is exerted by the high-frequency components of the signal.

Phantoms

Sometimes a person can hear sounds in the low-frequency region, although in reality there were no sounds of such a frequency. This happens because the vibrations of the basilar membrane in the ear are not linear and vibrations can occur in it with a difference frequency between two higher frequencies.

This effect is used in some commercial audio systems to expand the range of low frequencies that can be reproduced when such frequencies cannot be adequately reproduced directly.

Psychoacoustics in software

Psychoacoustic hearing models allow high-quality compression of signals with loss of information (when the reconstructed signal does not match the original), due to the fact that they allow us to accurately describe what can be safely removed from the original signal - that is, without significant deterioration in sound quality. At first glance it may seem that this is unlikely to ensure strong compression signal, but programs using psychoacoustic models make it possible to reduce the volume of music files by 10-12 times less than uncompressed ones with a very slight difference in quality.

These types of compression include all modern audio compression formats:

Ogg Vorbis
Musicam (used for digital audio broadcasting in some countries)
ATRAC used in MiniDisc format

In everyday life, we describe sound by, among other things, its volume and pitch. But from the point of view of physics, a sound wave is a periodic vibration of the molecules of the medium, propagating in space. Like any wave, sound is characterized by its amplitude, frequency, wavelength, etc. Amplitude shows how strongly a vibrating medium deviates from its “quiet” state; It is she who is responsible for the volume of sound. Frequency tells how many times per second the oscillation occurs, and the higher the frequency, the more alt we hear.

Typical values of volume and frequency of sound, which are found, for example, in technical standards and characteristics of audio devices, are adapted to the human ear; they are in the range of volume and frequency that is comfortable for humans. Thus, a sound with a volume above 130 dB (decibel) causes pain, and a person will not hear a sound wave with a frequency of 30 kHz at all. However, in addition to these “human” limitations, there are also purely physical limits on the volume and frequency of the sound wave.

Task

Estimate the maximum volume and maximum frequency of a sound wave that can propagate in air and water under normal conditions. Describe in general terms what will happen if you try to emit sound above these limits.

Clue

Recall that loudness, measured in decibels, is a logarithmic scale that shows how many times the pressure in a sound wave (P) is stronger than some fixed threshold pressure P 0 . The formula for converting pressure into volume is as follows: volume in decibels = 20 lg(P/P 0), where lg is the decimal logarithm. It is customary to take P0 = 20 μPa as the threshold pressure in acoustics (in water, a different threshold value is accepted: P0 = 1 μPa). For example, a sound with a pressure P = 0.2 Pa exceeds P 0 ten thousand times, which corresponds to a volume of 20 lg(10000) = 80 dB. Thus, the loudness limit arises from the maximum possible pressure that a sound wave can create.

To solve the problem, you need to try to imagine a sound wave with a very high pressure or a very high frequency and try to understand what physical limitations arise.

Solution

Let's find first volume limit. In calm air (without sound), molecules fly chaotically, but on average the density of the air remains constant. When sound propagates, in addition to rapid chaotic movement, molecules also experience a smooth back-and-forth displacement with a certain period. Because of this, alternating areas of condensation and rarefaction of air arise, that is, areas of high and low pressure. It is this deviation of pressure from the norm that is acoustic pressure (pressure in a sound wave).

In the region of rarefaction, the pressure drops to P atm - P. It is clear that in a gas it must remain positive: zero pressure means that in this region at a given moment in time there are no particles at all, and it cannot be less than this. Therefore, the maximum acoustic pressure P that a sound wave can create while remaining sound is exactly equal to atmospheric pressure. P = P atm = 100 kPa. It corresponds to a theoretical volume limit equal to 20 lg (5 10 9), which gives approximately 195 dB.

The situation changes slightly if we are talking about the propagation of sound not in a gas, but in a liquid. There the pressure can become negative - this simply means that they are trying to stretch and tear the continuous medium, but due to intermolecular forces it can withstand such stretching. However, in terms of the order of magnitude, this negative pressure is small, on the order of one atmosphere. Taking into account a different value for P 0 this gives a theoretical limit for loudness in water of about 225 dB.

Now we get sound frequency limit. (In fact, this is only one of the possible limits on frequency; we will mention others in the afterword.)

One of the key properties of sound (unlike many other, more complex waves) is that its speed is practically independent of frequency. But the wave speed relates the frequency ν (that is, the time at th periodicity) with wavelength λ (spatial periodicity): c = ν·λ. Therefore, the higher the frequency, the shorter the sound wavelength.

The frequency of the wave is limited by the discreteness of the substance. The length of a sound wave cannot be less than the typical distance between molecules: after all, a sound wave is a condensation-discharge of particles and cannot exist without them. Moreover, the wavelength must be at least two or three of these distances: after all, it must include both areas of condensation and a region of rarefaction. For air under normal conditions, the average distance between molecules is approximately 100 nm, the speed of sound is 300 m/s, so the maximum frequency is about 2 GHz. In water, the discreteness scale is smaller, approximately 0.3 nm, and the speed of sound is 1500 m/s. This gives a frequency limit of about a thousand times higher, on the order of several terahertz.

Let's now discuss what happens if we try to emit sound that exceeds the found limits. A solid plate immersed in a medium, which a motor moves back and forth, is suitable as a sound wave emitter. It is technically possible to create an emitter with such a large amplitude that at maximum it creates a pressure much higher than atmospheric pressure - for this it is enough to move the plate quickly and with a large amplitude. However, then in the vacuum phase (when the plate moves back) there will simply be a vacuum. Thus, instead of a very loud sound, such a plate will be “cut A"breathe air" into thin and dense layers and throw them forward. They will not be able to propagate through the medium - when they collide with still air, they will sharply heat it up, generate shock waves, and collapse themselves.

One can imagine another situation, when an acoustic emitter oscillates with a frequency exceeding the found limit of sound frequency. Such an emitter will push the molecules of the medium, but so often that it will not give them a chance to form a synchronous vibration. As a result, the plate will simply randomly transfer energy to the approaching molecules, that is, it will simply heat the medium.

Afterword

Our consideration was, of course, very simple and did not take into account the many processes occurring in matter that also limit the propagation of sound. For example, viscosity causes a sound wave to attenuate, and the rate of this attenuation increases rapidly with frequency. The higher the frequency, the faster the gas moves back and forth, which means the faster the energy is converted into heat due to viscosity. Therefore, in a too viscous medium, high-frequency ultrasound simply will not have time to fly any macroscopic distance.

Another effect also plays a role in the attenuation of sound. From thermodynamics it follows that with rapid compression the gas heats up, and with rapid expansion it cools. This also happens in a sound wave. But if the gas has high thermal conductivity, then with each oscillation, heat will flow from the hot zone to the cold zone, thus weakening the thermal contrast, and ultimately the amplitude of the sound wave.

It is also worth emphasizing that all the restrictions found apply to liquids and gases under normal conditions; they will change if conditions change significantly. For example, the maximum theoretical volume obviously depends on pressure. Therefore, in the atmosphere of giant planets, where the pressure is significantly higher than atmospheric pressure, an even louder sound is possible; conversely, in a very rarefied atmosphere all sounds are inevitably quiet.

Finally, let us mention one more interesting property of very high frequency ultrasound when it propagates in water. It turns out that when the frequency of sound significantly exceeds 10 GHz, its speed in water approximately doubles and is approximately comparable to the speed of sound in ice. This means that some fast processes interactions of water molecules begin to play a significant role when oscillating with a period of less than 100 picoseconds. Relatively speaking, water acquires some additional elasticity at such time intervals, which accelerates the propagation of sound waves. The microscopic reasons for this so-called " fast sound", however, were understood

You can buy the most expensive system in the world, but if you place it in a small cubic room, the cost will no longer matter. Definition right place for your speakers is the single most important factor in getting good sound in your room. Very precise speaker placement can open up a new sonic dimension for you. Any speakers do not exist on their own. They are an inevitable compromise with the listening room. There are no just good speakers - there are suitable ones. With a lot of desire and a little luck, your room can become your happiest place. We will assume that all the furniture and furnishings in the room existed before the acquisition of speakers or equipment that should be integrated into your room without disturbing the existing dynamics in it. The goal of a good listening room is to minimize coloration, which is strongest in the bass region between 20 and 200 Hz. At higher frequencies the room also has an effect, but resonances are much less problematic since it is much easier to achieve absorption of high frequency resonances. Any room will resonate at many frequencies.

The accuracy and height of the resonant peak depend on the absorption properties of the room. A room with a lot of upholstered furniture, carpets on the floor and drapes will be acoustically relatively “dead”. Peaks and dips in the frequency response in such rooms have an unevenness of 5-10 db. A room with bare walls and floor will be very lively, with peaks and valleys varying by 10-20 dB or more. General rule This is: in an acoustically good and correct room, you can place the speakers fairly close to reflective surfaces with minimal negative consequences. In acoustically poor rooms, the main strategy is to place the speakers as far away from the boundaries of the room and the listener as possible.

If we feel a series of deep dips or peaks in frequency, then this is the result of reflections. Reducing the level of reflections flattens the actual frequency response curve. The most important thing is to minimize early reflections (less than 20ms) as much as possible. Reducing them improves sound quality and stereo image. How can we improve the acoustics of a room so that this curve is flattened? This can be done by using absorbent materials to cover hard surfaces near the speaker. The best, most useful listening environment is a complete combination of the principles of “live” and “dead” room acoustics. I personally prefer a slightly dead room as opposed to a live room. How can this be determined without special instruments? Clap your hands. Does it seem to you that the sound decays naturally, or does it fade away too long (live), or, on the contrary, fades away too quickly (dead)? The best solution is to provide the room with a reasonable balance of dispersion and absorption. A room with bare walls will have a strong echo, which will impair clarity. Wall art, bookshelves, drapery, and floor coverings will provide sound absorption and dispel harmful reflections. Uncovered windows, bare floors and walls are not desirable.

The speakers should be located in an acoustically dead zone, occupying approximately 1/3 of the room's space. Then there is a very live zone of the room, in which there should be objects that dissipate, but do not absorb sound. The closer the absorbing surface (carpet) is to the speaker, the better. Different types of carpets and the lining (backing) of the carpet most affect the upper mids and high frequencies. The thicker and larger the rug or rug, the more it will “absorb” these frequencies. Carpets and curtains reduce reverberation in the room and, as a result, the transfer of sound energy to the walls. Carpeting has little effect on the low frequencies, but the mid frequencies can overwhelm. I prefer a wall-to-wall rug that isn't thick. This is reasonable, if only because the majority of speaker manufacturers conduct critical listening sessions of their products in rooms with completely muted floors.

Many experts believe that the base of the carpet/covering should be made of natural fibers, and not rubber or foam rubber, because... they absorb frequencies selectively—some frequencies are significantly attenuated, while others are not attenuated at all. The most important thing is to minimize early reflections. Reducing them improves sound quality and stereo image. All recording studio designers try to reduce early reflections as much as possible. How to properly place speakers in a room? You should have 2 main goals: flat frequency response and good 3D imaging. Even though you have good speakers, room influence is a very important factor. In many cases, it is more important to pay attention to the acoustics of the room than to spend 2 times more money for new speakers.

Symmetry

The environment behind and to the sides of the speakers should be symmetrical. The environment immediately next to the listener is less important. Regarding the symmetry of the front and rear walls, there are many supporters of various measures. Most (but not all) agree that the wall behind the listener should be highly reflective.

Professionals believe that the entire area around the speakers should be dimmed to reduce reflections as much as possible. Another point: it is advisable to dampen the side walls only immediately in front of the speaker to minimize close reflections of the side wall. To best reproduce a three-dimensional sound image, the room must have good symmetry between and around the speakers. This means that if the speakers are not placed symmetrically, the early reflections from the back wall of the first speaker will be different from those of the second speaker, and critical parts of the stereo signal will be damaged. It is imperative that the distance from you to both speakers be as identical as possible. IN good systems a deviation of several cm will be clearly audible. It is generally believed that the speaker and listener should form an equilateral triangle, but this is not an absolute rule. Some manufacturers give their recommendations for the placement of their speakers. Remember that any recommendation is just a start, a beginning for an experiment; by experimenting properly, you will achieve the desired results.

Directed sound from a speaker is primarily responsible for imaging, while reflected sound is most responsible for changing the speaker's tonal balance - in terms of sound density, or attrition, etc. Any reflective surface - wall, floor, furniture - creates reflections. Based on this, the speakers should be located. The most important thing is to reduce natural reflections as much as possible. Early reflections reach the listener almost simultaneously with the direct sound, degrading the signal. For example, speakers with wide front panels - planars, etc., are less critical of nearby side walls and surfaces, but are very critical of proximity to the rear wall. In general, the further away from reflective surfaces and the further away from the back walls, the greater the depth of the soundstage and the more “air” there will be.

Listening position

The listener should sit exactly in the middle between the speakers, the distance to the listener is slightly greater than the distance between the speakers. If you don't follow this rule, you will never hear a good sound picture. In a room with proportional dimensions, the best listening position is 30-90 cm from the back wall. If you are sitting directly against a wall, you should dim the space on the wall directly behind your head a little. Your brain won't be able to process these reflections, but trust me, they can make a big difference in the sound in this case.

Remember one thing - having your head close to the back wall has two positive effects. Firstly, close to the walls the sound pressure is highest, and the speed of sound waves is the lowest. Positioning in the maximum pressure zone gives better perception of deep bass. Second, the reflected sound waves are shorter than the circumference of the head, so the brain cannot measure the time delay between the ears. When the brain cannot detect reflections, it ignores them.

This is a simple example of how the brain ignores unwanted or irrelevant information and confirmation of the Haas effect - if information from the speaker comes first, then any distortions and reflections (even unpleasant ones) will come later and at a much lower volume - and our brain will ignore them.

Often the listener sits too far from the speaker. The farther you sit, the more free space room affects the sound, especially the mid and high frequencies, but close is also bad - the sound does not have time to form into a picture. Great importance has height AC. It's best when the tweeter is located just above the ear (but not always) - experiment with whether you sit higher or lower. Camber - this method achieves focusing the sound image (imaging) and adjusting the tonal balance, as well as optimizing the mid and high frequencies by adjusting their directionality. The easiest way to do this is with two people. First, aim the speakers so that they face a point slightly behind the listener's head - maintaining the same distance from the ear to the tweeter of each speaker. Play music with vocals or violin. One person should watch the trick. The other should rotate the AC around the inner front spike. The listener must discover which speaker placement is best. When this is done, install the second speaker identically to the first. Some speakers work better turned inward, others differently, but it is best not to turn them much inward or not to touch them at all. Follow the manufacturer's recommendations.

The most important thing is to correctly fill the central images without sacrificing the width of the soundstage. The tilt of the speakers is also an important factor - forward, backward, inward, etc. – also affects the sound. Many manufacturers tilt the front panels of their speakers negatively to achieve proper imaging and sound coherence from the speakers.

Listening height

In two-way speakers, your ears should be on the conventional line between the tweeter and the woofer, in 3-way speakers, on the line between the tweeter and midrange speaker. Keep in mind that the best location for creating a spacious soundstage may not be the ideal location for bass. We must find a compromise in which these characteristics are maximum in our opinion. Depending on personal taste, you can sometimes sacrifice one for the other. Decoupling from the floor is the most important point when installing speakers. Only after solving this issue will you be able to hear your speakers as they really are. Speakers are most susceptible to resonances, and therefore most require rigid fixation. The most important thing that a rigid speaker installation gives is clear focus, clarity, detail, cohesion, and well-articulated bass. The sound will become denser and clearer, especially at high volumes. The more expensive your system, the greater the speaker installation requirements. Placing the speakers too low reduces the dynamic range. Improving the acoustic performance of your room can completely change your opinion about the quality of your system. What room characteristics affect the sound. All sound within the confines of your room will depend on a combination of three acoustic characteristics: reflection, dispersion, absorption. A good listening room will have a proportionate amount of these characteristics. The smaller the distance between the walls where the speakers and the listener are located, the more sonorous the sound. longer distance between these walls, the deeper the bass. Reflections: All or most of the sound energy consists of reflections that occur in the room according to the rule: the angle of incidence is equal to the angle of reflection. Hard flat and smooth surfaces - bare walls, glass, bare hard surfaces of furniture - reflect sound energy.

Diffusion

All or most of the sound waves reflected back into the room are already there in a disordered state - a randomly scattered sound mass. Solid, non-flat, rough, ribbed surfaces, cylindrical and round objects scatter sound. Absorption As opposed to reflections, most sound energy is absorbed. Soft porous surfaces: carpets, floor coverings, upholstered furniture, thick fabric draperies, etc. - absorb.

The quality of the bass in your room largely depends on the room itself. Because the wavelength of bass frequencies is so long, most furnishings, wall and floor designs do very little to change the bass frequencies in a room/speakers combination. Therefore, optimizing low frequencies is a matter of choosing a listening room with optimal dimensions (ratios) and placing speakers in this room. Low frequency energy travels spherically in all directions equally. When a low-frequency sound wave hits an obstacle (a wall), the bass energy is - for the most part - reflected back into the room, bouncing off every obstacle - floor, walls, ceiling. The woofer should be unequally spaced from the three closest side planes of the room. All this is significant, because the reflective plane closest to the speakers enhances some bass frequencies.

If the reflective planes are equidistant from the speakers, some bass frequencies will be greatly enhanced. Those. If your speakers are placed the same distance from the back wall, the side wall and the wall of the closet or chest of drawers, then you will get a triple boost of certain groups of bass frequencies, which will lead to a very audible hum at those frequencies. If the doors are in the corners of the room, the bass may simply “leak” through them. For serious listening, you need to close the doors. This is not the case for the mid and high frequencies, where the energy is directed in a more concentrated and controlled manner, in a cone-like, horn-like fashion. Low-frequency reflections and resonances can be quite easily adjusted by manipulating the placement of speakers, varying the distance from the speaker to the nearest wall.

The more all three of these parameters (distances) differ from each other, the less “unison” will be, and accordingly, the less unwanted resonances will be. Standing waves are low-frequency reflections (resonances) between two parallel walls, the main enemies of good sound. They color the sound in your room, emphasizing certain musical notes and creating a rough and unnatural distribution of acoustic energy within the room. The propagation of standing waves is a property of the physical characteristics of the room and has nothing to do with the equipment. In rectangular rooms, standing waves occur in all three directions simultaneously, exerting a very complex distribution of pressure within the room. Standing waves cause noticeable coloration above approximately 300 Hz. However, isolated or random standing waves can be noticeable below this frequency. Standing waves are essentially fragments of some frequencies huddled together in some places in the room. Evenly distributed colors are almost unproblematic compared to standing waves. Understanding what standing waves are and how they work will be helpful to better optimization your room and your speakers.

The determination of the axial standing wave constant between two parallel walls can be easily calculated by the following equation: (1) Fo = 1130 / 2L or (2)Fo=565/L (where the constant 1130 – speed of light in feet per second, L – distance between walls in feet example: calculating fundamental standing waves in three cardinal directions for a room of size 4.8 (w) * 7.8 (d) * 2.4 (h) between short walls Fo w = 565/16 = 35 Hz between long walls Fo l = 565/26 = 22 Hz between floor and ceiling Fo h = 565/ 8 = 70 Hz .

Please note that in this example the height of the wall is 2 times less than the length of the short wall Foh = 2Fow = 70 Hz . This room would have significant coloration at 70 Hz, 140 Hz, 210 Hz and further multiples of 70. The worst possible tonal distribution occurs when the room dimensions are equal in all three directions, i.e. when the room is a perfect cube. In such a room, the harmonics of all resonant frequencies will be equal, and the low frequency resonances will be extremely rough and colored. The best possible tonal distribution will be in a room whose dimensions are not related by a single integer (multiple) number. L24*W24*H8 -bad example - all sections are multiples of 8 L26*W15*H8 – good example. The smoothest bass extension will be achieved if the frequencies of the reflected energy are distributed evenly and do not clump together.

Identifying bass in a room. The number 550 is half the speed of sound per second above sea level. Dividing this number by some bass frequency, say 20 Hz, we get the smallest distance between the walls at which this frequency will be supported by the room. If we divide this number by the bass frequency of 20 hertz, we get 27.5 feet - this is the minimum distance between the walls of your room in order to support this frequency. If the distance between the opposite walls where the listener is located and the speaker is 12.8 feet, then 550/12.8 = 43 Hz is fine for a mid-sized UK speaker, but a shame for an Infinity Bass Tower speaker.

Let's say you want bass below 35Hz - 550/35= 15.7 feet is the minimum distance between walls to support 35Hz. But that number—15.7—almost double the height of a standard room—is bad news. The room will have the same standing waves in two directions. But don't worry, it's unlikely that these dimensions will be a strict multiple of two. The sound stage and sound image depend on the location of the speakers, their orientation and the acoustics of the room. Optimizing speaker placement is a difficult task. Since speaker placement is equally important for soundstage and good bass reproduction, you have to find a compromise between these characteristics - it is much better to sacrifice a little bass reduction to get good staging/imaging. Stage depth is best when the speakers are located at some distance from the front wall - this will reduce the effect of early reflections, improve the focus of images, and allow the speakers to “breathe”. In systems highest resolution, precisely positioned in the acoustic space, the sound stage can extend far beyond the listening room: the rear of the stage does not rest against the back wall, but naturally extends inward. Stage width The final width will be affected by the distance between the speakers and the wheel alignment of the speakers. But remember that on most recordings this sonic characteristic is poorly recorded.

Determining the distance between speakers

Play a recording with a good focus on the central image - for example vocals. Place the speakers approximately 1.8 - 2 meters apart, and so that they are aimed at a point slightly behind your head. Listen to see if the sound is focused enough. Move the speakers further – 30 centimeters and listen again, etc. When the center begins to thin out and blur and become scattered, know that you cannot move the speakers further apart. You now know how widely you can place the speakers without losing the soundstage and density of the central image (focus). Focus is largely, but not entirely, related to the transmission of high-frequency speakers. Our ear uses them to outline an object. Experiment with wheel alignment.

HF propagates very directionally. A happy side effect of narrow directivity is that it reduces stray reflections from nearby surfaces, minimizing the echo of reflected frequencies that affect the sound image.

Balance adjustment

If the balance of the system is adjusted so that the sound is uneven across the entire front and is poorly focused, then the reason may be that one speaker is closer to you than the other. For example, if a lead vocal that should be centered comes to you from the right, the right speaker should be moved back or the left speaker pushed forward. Usually, even a 2-3 cm difference in the distance to you is already clearly audible.

AC movements

All lateral movements of the speakers affect the midbass more, and moving forward and backward affects the depth of the bass more.

The density of the sound image is one of the unusual and musically very beautiful characteristics - the ability to concentrate not only high-frequency energy, but also the wealth of musical energy concentrated in the midrange and upper bass. Due to the wide dispersion characteristics of these frequencies, the density of the image in this part does not depend on whether the edges of the speakers are sharp or rounded. A narrow body with highly rounded edges reduces reflections from the front panel, but there are problems with standing waves inside the box. The narrow body promotes good playback MF, because The narrower the body, the more omnidirectional the sound becomes. If a speaker with a wide polar pattern (narrow body) is placed in a loud room, then the timbre of its sound will be greatly distorted. The narrow body and small speakers result in a lack of physicality and imagery. Such speakers should be placed away from reflective surfaces. A happy side effect of narrow HF directivity is that spurious reflections from nearby surfaces are reduced, minimizing primary reflections that affect the sound image.

Wide front panels and shallow cabinets are the key to the most correct directional characteristics and balance of the low-frequency range in a real listening room.

By Peter Qvortrup

If the speakers have a narrow directivity (wide body), and the acoustics of the room are dull, you will hear the actual sound of the speakers.

Bryston research on acoustic design and speaker placement

The resonant characteristics of a room depend on its configuration (proportions) and design. A square room with bare walls would have the worst possible acoustics for an audio system. In square rooms, standing waves appear in three directions at once, weakening and changing some frequencies and strengthening others, enhancing resonant peaks in a very narrow range. These peaks change the sound greatly. Bare walls have problems with early reflections (High Q) - they prevent the sound from opening up, making it ringing, narrowing the dynamic range and greatly affecting the tonal balance. In a concert hall, we have three main effects that influence what information our brain receives about the acoustic qualities of that environment:

The first direct sound wave coming to us from instruments.
The second sound wave is reflected from nearby walls.
Reflected energy, which is random overtones from all objects inside and has no direction.

Direct sound tells the brain where the sound is coming from. Early reflections, if they reach us within 10-20 ml/seconds, will distort the sound image, tonality, etc. Late reflections (ambience), on the contrary, will add a feeling of spaciousness, spatiality, airiness of the environment. In a good concert hall, direct sound reaches the listener at 20-30 ml/sec. earlier than the primary reflections. And secondary reflections come later by as much as 100 ml/sec. Obviously, in our listening room we should strive to obtain similar results.

It should be noted that pop and rock music is usually recorded in an acoustically dead studio environment in the “near field”, which tends to prevent primary reflections and High Q sonority. (this is probably why studio monitors often sound loud and harsh in rooms, because in studios they are heard in the near field and in a very muffled environment, where this ringing and harshness does not manifest itself, but all the details of the recording are heard clearly).

So, if your room acoustics are close to concert hall, rock music will sound great. How to achieve similar results in a regular room of 12*18*9 feet (almost a standard Russian room, I must say, V.M.)? You should place your speakers so that the direct sound reaches your ears first, using absorbents (absorbers) where the first reflections from the side walls occur. But there should be more space behind you to create a larger sound field. Sit in a chair. Have someone move the mirror along the side wall. When you see the reflection of the speaker in the mirror, this is the first point from which early reflections will follow. Sound is reflected like light - the angle of incidence.... This is where the absorber should be placed. Sit 20-30 cm away from the back wall. Do not place any absorbent materials behind your head. There can only be sound-diffusing materials, distributing random, non-directional sound energy that adds a spacious feel to the room because this random energy (late reflections) arrives much later than direct sound. Place absorbent materials in the corners of the room.

Other measures are soft chairs, flowers, statues, etc. They will also scatter or absorb secondary reflections. Obviously these items won't be as effective as specialty items, but they are a step in the right direction. The main goal you need to remember is that the early reflections and the lack of late random reflections are used by the brain to detect the fact that you are in a small room. Therefore, by reducing the effect of early reflections, reducing the effect of standing waves and sonority, you will increasingly feel as if you are in the hall with the performers.

This information is based on scientific research and observation, as well as the experience of some of the most successful dealers. Solutions presented here. are aimed at limiting your room's sound interference. We will help you place your speakers through the use of psychoacoustics and physics. This method can produce excellent results through experimentation, without the need for special room treatments. How do we arrange sound events in space? Our brain determines the time delay when sound occurs between our two ears. If there is no delay, then the sound is coming from a point directly in front of us. If the sound wave reaches the right ear first, then the sound is on the right, etc. This spatial information—sound transients—is instantly detected by the brain. By determining the delay between the right and left ears, our brain determines with extraordinary accuracy how much to the right or to the left, or how much closer or further, the sound source is from us. It is by the delay of sound between our ears that the brain determines the most important sound characteristic - tonality. This has recently been proven in scientific studies. And is believed to be a critical part of our historical survival. In other words, we first identify the source of the sound - for example, a potential danger - and then try to identify what was the source of the sound.

The first step to getting a good stereo soundstage is to eliminate early reflections from major transients as much as possible. Or, practically, you have to ensure that the sound from the speakers reaches your ears before any reflections from that sound. According to a psychoacoustic phenomenon called the Haas effect. the brain will give priority to the first sound wave not distorted by reflections.

Determining the best speaker location based on room size

Audio Physic called this method room mapping. The principle of this technique is based on the wave phenomenon (phenomenon). Accurately measure the room and draw its plan. Divide the room into equal parts. There are two ways – even and odd number of zones. When dividing the room plan into an even number of zones. By placing the speakers and/or your chair not even at the intersection point, but at one of the separated parts, you will get a natural bass boost from interaction with the room. At the intersection points, the bass frequencies will be enhanced. The bass and midbass tuning method assumes a similar principle - reducing rather than enhancing low frequencies. This happens if the room is divided into an odd number of zones. To do this, you move the speakers to odd-numbered parts of the room layout. It is important to remember that a room can be divided into many more parts than 3 or 4. In even sections, the bass is strengthened, in odd sections it is weakened. Another example (Bryston) is that if you place speakers with excellent frequency response in the corners of the room, you get about -6 db of bass boost. This rise is clearly an anomaly, but the same thing happens elsewhere in the room, only to a lesser extent. We carried out research and found that the increase or decrease occurs at certain nodes (points) of the room. At odd nodes the excitation has minimum value and vice versa. For example, your room is 14*18 feet (ft = 0.3 m). Take any size - length or width - and divide into an odd number of parts, say 18 divided by 3,5,7... you will get values = 6, 3.6, 2.57 - three possible positions (positions) when placed against a long wall. Divide 14 into three parts – we get values = 4.67, 2.8, 2. – possible locations against a short wall. Now place the speaker at a point that is the fifth in length and seventh in width of the room. The fifth value of the length = 3.6 feet, the seventh value of the width = 2 feet. The speakers should be placed at the intersection point, where the excitation of low frequencies will be minimal. Remember to test all options for optimal results. An important detail - the intersection point should not pass through the front or back panel AC, and through the woofer magnet. If this rule is followed, you will feel a clear result. Experimentation is the key to success. In the process, you will discover many things that are not working correctly and you will be able to minimize these shortcomings. Most importantly, standing waves and early reflections must be minimized as much as possible.

It's pretty hard to expect that correct setting sound will be found immediately after the start of the concert. It usually takes more or less time to get the entire system to sound exactly as the engineer wants. In addition, the sound engineer is obliged to take into account the gradual change in the state of listeners and performers that occurs during the concert, so that even after receiving the ideal setup, it cannot be considered final. Therefore, it is usually necessary to make constant adjustments to the sound of all systems of the complex until the sound starts working, and then carefully monitor that the delicate and dynamic balance of this sound does not fall apart.

The sound at the concert will work until the sound engineer stops supporting it.

Recording of a concert performance

It’s a good idea to record all concerts with your participation on magnetic tape. Listening to these recordings, one can discover many typical mistakes, which are repeated every concert. After analyzing these errors, you can try to improve or change the individual form of sound mixing. You can trace all the moments that escape attention during direct work over the sound. However, when assessing the quality of mixing from a phonogram, you need to be able to accurately take into account the influence of recording and listening conditions, as well as the influence of the recording process itself, for example, the narrower dynamic range of magnetic recording compared to a concert. If you record from the main outputs of a mixing console, there will be an overabundance of vocals in it, since the sound of vocals at a concert turns out to be softer than in the recording.

There's no doubt that the performers will also want to hear a recording of the performance, so prepare to be horrified by listening to your raw soundtrack, which will be a far cry from the quality of the special CD live recordings. Therefore, if you want to get a more or less complete recording of a concert, try to provide the necessary recording conditions so that the resulting recording of the concert can be at least remix again.

In most cases, a full-fledged stereo recording of a concert is an unjustified luxury that takes a lot of time and effort, but a monophonic recording that conveys the sound atmosphere of a concert well can be obtained if one of the channels of a two-channel tape recorder is connected to one of the outputs of the mixing console, and the second to a microphone located in the hall, close to the sound engineer’s workplace. Such a recording allows you to evaluate the signal of the sound reproduction system, the sound in the hall, and also obtain, with appropriate mixing of the signals of both channels, a relatively acceptable version of the concert recording. Of course, with this recording method, the balance of the concert sound will be thrown off, so in order to maintain it, you need to use the sum signal of both channels of the mixing console for recording and choose the right microphone position. If you want to get full stereo balance of the concert, you will have to use a four-channel tape recorder. Spend a little time before the concert selecting the sound characteristics of the recorded signals and determining the position of the microphones and you will get very good material for a stereo demo recording of a concert.

Sound mixing for independent artists

The sound of concerts with independent artists is mixed a little differently than the sound of bands, even if the independent artist is performing as a regular group of performers.

Related information:

B) In the following sentences, underline the predicate verb, determine its tense form and voice. Translate the sentences into Russian.

Basic concepts in acoustics. Maximum volume and pitch Sound limit

Prerequisites

Limits of sound perception

What do we hear

Masking effect

According to the time of arrival of the masking and masked sound:

By type of masking and masked sounds:

Simultaneous camouflage

Temporary disguise

Post-stimulus fatigue

Phantoms

Psychoacoustics in software

Task

Clue

Solution

Afterword

Popular articles

Latest articles

Sections

Pages

Special projects

Contacts