Acoustic term 2)

Psychological acoustics

 This paper studies the relationship between subjective hearing and physical quantity of sound. It focuses on the relationship between sound stimulation and response. People's correct feeling and comprehension of sound are very important to the Evaluation of sound.

   Phase

When the phase difference between two sound signals is equal to O, two States are specified in the audio system: one is that when two (or more) Speakers input the same signal, the vibration direction is consistent, when the speaker is in the same voice, the sound is superimposed, the sound is correctly positioned, and the sound is powerful. When two (or more) microphones pick up the same sound, the phase difference between the output signals is equal to O.

Signal-to-Noise Ratio

The abbreviation of the signal-to-noise ratio refers to the ratio of the average signal power to the average noise power. The higher the signal-to-noise ratio, the smaller the background noise of the system, and the weaker the subsection of the sound signal is not easily overwhelmed by the noise, the dynamic range of the device is also increased.

Phase distortion

The abbreviation of frequency and phase distortion is an important aspect of the linear distortion of the audio system. Because the phase shifting of audio signals of different frequencies is different when they pass through the circuits of resistance and impedance, as well as the time sequence in which the speaker emits sound at different frequencies to the listener is different, the phase (I .e. time) Relationship between each frequency component of the sound source is changed, the output sound signal waveform is no longer the same as the original sound waveform. Phase distortion will produce a certain impact on the sound that reappears (changing the phase relationship between the fundamental wave and the harmonic wave) and the Image Positioning (chaotic between the front and back of the sound and the left and right of the sound, in addition, it causes issues such as bass blur and tweeting level deterioration. In the stereo sound system, the effect of phase distortion on the restored audio/video positioning is particularly serious. It is a distortion that cannot be ignored, so we should minimize the phase distortion in the audio system.

On the reverb time

The time required for the sound pressure level to fade to a level that the ears cannot hear after the sound source stops.

Harmonic distortion

A type of nonlinear distortion. The waveform distortion of the new harmonic component is generated after the signal is replayed, and the distortion is expressed by the harmonic component in the output signal and the ratio of the total output sound signal. Research shows that the odd harmonic has the greatest damage to the sound tone. For example, the third harmonic changes the sound to a sharp point and the Fifth Harmonic generates a golden sensation, seven or more odd harmonic waves produce extremely sharp and harsh sounds, while even harmonic waves are different. For example, if the second harmonic wave is higher than the fundamental frequency, it does not sound as non-harmonious, on the contrary, the sound color can be enriched. The modern vibrator uses this feature to artificially add a Secondary Harmonic to the sound, thus improving the reproduction of the sound. However, any serious harmonic distortion will make the sound split, burst, hair, and blow, and the harmonic distortion of the audio equipment should be minimized.

Hearing Threshold

The minimum sound pressure that can cause hearing, that is, the minimum sound that the human ears can hear.

Ripple

It is also called the top switch. Because the audio signal is too strong or the dynamic range is too large, the top of the peak signal is scaled out of the linear area. The phenomenon of Ripple cutting leads to signal ripple distortion. Ripple distortion not only damages the sound quality, but also may burn equipment. If the high-frequency harmonic generated, it will burn the speaker audio head, And the DC component can also burn the bass unit. The method to avoid this is to adjust the signal level appropriately to ensure that the Ripper lights (peak value display) in each audio system cannot be on when the maximum sound signal is displayed.

Speaker sensitivity

Generally, when the speaker inputs a power signal of 1 Watt, the sound pressure level obtained at one meter on the axis is used as an indicator. The larger the sound pressure level, the higher the speaker sensitivity, the maximum sound pressure level of the speaker can be calculated based on the speaker's sensitivity and rated power.Latency feedback Rate

 The attenuation of Multiple echoes over time reflects the sound absorption coefficient of the room interface. In the latency effect, it is used to control the number of echo requests. When the feedback rate is continuously adjustable between 0% and 99%, the feedback rate is 0%, which is the latency effect; when the feedback rate is 99%, it is an endless echo.

Speaker Frequency Response

The output characteristics of the speaker change with frequency, which is mainly determined by the inertial system components of the Speaker and the resonant frequency. For example, when acoustic radiation is performed, the acoustic impedance is reduced, and the sensitivity of low frequency segments is decreased. the inertia of the vibration system reduces the sensitivity of high frequency segments. Through reasonable design of the sound box structure, excellent speaker units and speaker materials are selected. It can improve the frequency response characteristics of the speaker and compensate for the frequency defects of the speaker.

Phase shifting effect

A special sound effect in the effect device. There is a phase difference between the direct sound emitted by the sound source and the delayed reflection sound during the communication of the room. When the two sounds come together, it will produce a phenomenon called the comb filter effect in acoustics, that is, the mutual enhancement on some points forms the peak point, and the mutual offset forms the valley point on other points. The effect of the phasing is to take advantage of this phenomenon. It has the function of adjusting parameters of the amount of time delay between the direct sound (I .e. the unprocessed sound signal) and the reflected sound. it can control the peak and valley positions of the comb filter effect, so that the odd harmonic enhancement, even harmonic reduction, or odd harmonic reduction and even harmonic enhancement can be realized, in order to improve the sound tone and filter out excess harmonic components produced by certain distortion. The difference in the bee Valley amplitude of the Hang filter is determined by the mixing ratio of the delayed signal and the direct signal. The mixing ratio of the two is, with the largest difference and the most obvious effect, at this time, the peak amplitude is 6 decibels higher than the direct signal before the hybrid, and the Valley amplitude is O. A comb filter usually uses a short delay, which is between 1 to 20 milliseconds.

Relative reverb time

The time required for the sound pressure level to fade to a level that the ears cannot hear after the sound source stops.

Speaker Distortion

The speaker output sound signal is distorted from the original audio signal. the harmonic wave is generated because the amplitude of the speaker vibration system does not change linearly with the input level, and the transient characteristics of the speaker vibration system cannot keep up with the changes of the electrical signal, this distortion is inherent in the speaker.

Music forks

The metal, which is similar to the English letter U, has a handle at the lower end, and uses a hammer to hit the upper end, that is, a certain frequency of sound. The two arm is long and thin, and the pronunciation frequency is low. The two sides are short and thick, and the pronunciation frequency is high. Because it contains very few phoneme components and the sound is close to the pure sound, it is often used as a standard for determining the tone. It can also be used as an experiment for producing standing waves by sound interference.

Ears Effect

People rely on the volume difference, time difference, and chromatic aberration between ears to determine the sound azimuth effect. Due to the orientation and distance of the ears, the two ears hear different sounds, I feel that the sound comes from a large volume, an earlier arrival, and a better tone.

Transient Characteristics

Also known as adaptability, it refers to the ability to respond quickly and clearly to pulse signals. There are many quenching signals in music, such as piano and percussion, and their rising edges are steep, if the sound equipment cannot keep up with the signal changes in time, it cannot truly reflect the original characteristics of the sound. The starting and ending sections of the sound signal must have an appropriate response speed, too slow, it is difficult to follow the abrupt change signal, the sound sounds muddy, of course, too fast or excessive changes exaggerated will bring a abrupt sense, it does not sound comfortable.

Diffraction

It is also known as diffraction. When a sound wave is transmitted, if it is blocked by an object close to the sound wave wavelength or equal to the wavelength, it will bypass this object and continue traveling. When the barrier is small (compared with the wavelength), it can still clearly hear the sound behind it; but when the barrier is large, it will form a significant reduction in the volume of the phonograph.Speaker impedance curve

 It describes the characteristic curve of the speaker's impedance changing with frequency. The impedance reaches the maximum value at the harmonic peak frequency, and the impedance reaches the minimum value at the reverse resonant frequency (Valley, generally, this value is used as the speaker's rated impedance. When the frequency is higher than the frequency corresponding to the reverse harmonic vibration peak, the sensitivity of the speaker's line ring increases, and the impedance curve continues to rise, the impedance curve can be used as a reference for speaker design and impedance matching.

Audio range

The distance between two sounds. The unit of calculating the phoneme is called the degree. The two questions contain several levels, which are called the degree.

Pain Threshold

The sound pressure when people are uncomfortable with the sound. different frequencies of sound have different frequencies of pain points, for example, 50Hz of sound pain points at around 10 Pa, the pain threshold of 1000Hz sound is about 200 Pa, and the pain threshold curve of various frequencies is drawn into a curve called "pain threshold curve ".

Audio domain

It refers to the range between the lowest sound and the highest tone that a certain instrument or voice can send.

Audio Zone

The whole phoneme of an instrument or human voice can be divided into several parts based on its sound height and tone characteristics. Each part is called a music area. The voice field is called the "audio area". Most audio areas can be divided into three audio areas.

Audio

The audio frequency ranges from 20Hz to 20Hz.

Auditory location

The human ears can determine the direction and distance of the sound source. The accuracy of the determination of the sound source distance is poor, but the determination of the sound source direction is quite accurate. Auditory location is caused by the dual-in-ear effect. When the sound source reaches the two ears, the Volume Difference and time difference are generated. When the frequency is increased to 1400Hz, the intensity difference plays a major role, when the time difference is less than 1400Hz, the time difference plays a major role in the identification of the direction of the sound source by G ears, and the horizontal direction is better than the vertical direction. When the sound source is in the front, that is, when the horizontal azimuth is o degrees, a normal auditory person, in a quiet and echo-free environment, it can be used to identify the changes in the horizontal orientation between 1 and 3 degrees and the changes in the sound level between the left and right ears (from 0 to 60 degrees, the ears have a good ability to identify the location, and over 60 degrees will rapidly deteriorate. In the vertical direction, the positioning capability of human ears is relatively poor, but the vertical positioning capability can be greatly improved by swinging the head.

Concealed Effect

When listening to a sound, the more often it is masked, the more concealed it is; the higher the sound pressure level of the masking sound, the larger the masking volume. The low-frequency sound is easy to conceal the high-frequency sound, and the high-frequency sound is difficult to mask the low-frequency sound. In the process of music, people do not feel the noise, but when the music stops or stops, people can feel the background noise from the speaker. This effect is the masking effect.

Audio Band Division

In terms of sound quality evaluation and audio system adjustment, the audio range is usually divided into several frequencies. The Rise and Fall of the voice signals in different frequencies are different for the listener, according to different requirements, the audio frequency band can be divided into 3, 4, and 7 segments, audio can be divided into up to seven frequencies, including the subwoofer, mediaintone, mid-tone, high-pitched, and high-pitched. The frequency range of the subwoofer is 20 to 40Hz, which is responsible for the heavy volume of sound. The amount of this frequency determines the heavy volume of sound. When appropriate, the sound is strong.
  The sound of thunder, bass drum, bass, and organ can be controlled. Excessive improvement may confuse the sound. The frequency range of the bass is 40 to 150Hz, which is responsible for the sound width A. the guitar, drum and other bass instruments are located in this frequency band. Excessive lifting will make the sound soft and sound long, when appropriate, the bass should be relaxed, and when the voice is insufficient, the sound should be thin and plump. The frequency range of the sub-woofer is 150 to 500Hz. It is responsible for the intensity of the voice. The voice is in this frequency band. When this frequency band is insufficient, the singing sound will be drowned by the music, and the sound will be soft, if it is too strong, the Bass will be stiff. When it is appropriate, the Bass will be strong and tough. The central audio frequency ranges from 500 to 2 k Hz. It is responsible for the brightness of the sound, including the low harmonic and extensive sound of most instruments. When the intensity is too strong, it will generate sounds similar to those heard on the phone, however, the characteristic sounds of percussion such as Xiaojun drum are clear and bright when the range is appropriate, and the sound is choppy when the intensity is insufficient. The frequency pattern of a high-pitched voice is 2 k Hz. It is responsible for the transparency of the voice and is the most sensitive part of human speech, the characteristic sound of the stringy instrument (such as the friction sound between the string bow and the string, and the sound of the finger-hitting string) is located in this frequency band. When the intensity is too strong, the speech recognition will be masked, when there is not enough, the sound penetrating power drops. The frequency range of the tweeter is 5 kb to 10 KB Hz. being responsible for the legality of the voice, affecting the sound's sense of distance, intimacy, and color. If the sound is too strong, the sound of the wood pipe (such as the short flute and long flute) and the violin will be highlighted, and the voice of the language will be obvious. The frequency range is 10 KB to 20 KB Hz, which is responsible for the fine voice. When appropriate, the sound of the triangle iron and the vertical sound is clear and lifelike, and the rhythm of the sand hammer is clear and recognizable, when the voice is insufficient, the details cannot be heard.Pitch

 In linguistics, the posts of voice are low. The speed of sound wave vibration determines the length, elasticity, and thickness of human vocal cords. Call the audio in the music.

Refraction

Sound waves change the propagation direction on the contact surface of two substances (or materials with different density and media) due to changes in sound speed, and then enter the second kind of material phenomenon, for example, when the sound enters the wall from the air, the direction changes.

Homophonic

It refers to the sub-audio in which the frequency and pitch frequency are an integer multiple... Homophonic sounds are called the second or third homophonic sounds when the frequency is twice or three times the baseline.

Early reflection

It is also referred to as the recent reflection sound. The sound that arrives within 50 milliseconds after the direct sound is reflected once or twice. In the sound field, an appropriate early reflection sound can make the sound thickened, aggravated, or even strengthen the direct sound. However, when the sound is too strong, it will damage the Audio-Visual Positioning. It should be designed through acoustics, make proper use of and control the early reflection sound on the interface.

Direct voice

It is the main component of a sound source (that is, a speaker) that directly sends a sound to the speaker. In the audio system, unprocessed sound signals are also called direct sound. During the transmission process, the direct sound is not affected by the indoor reflection interface. The distance from the sound source is doubled. The sound pressure of the direct sound is reduced by 6 decibels, and the sound quality is pure, but it sounds dry, modern sound field design requires the full use of direct sound from the speaker, reasonable control of the reflection sound, speaker hanging is the best way to get direct sound. The conditions for getting direct sound from the speaker in the listening area are as follows: (1) All speakers can be seen in the listening area; (2) the listening area is located in the Cross-radiation area of all speakers.

Latency

The time difference between the frontend and backend of the same sound. In the room, the distance between the sound source and the reflected surface divided by the sound speed can be used to calculate the short-term (less than 50mm) Wind delay time returned after the sound is sent as the early sound reflection effect, for a long period of time, it is the echo of trembling and echo effect. Some delimiters refer to the pre-delay time before the early reflection sound and the entry time before the reverb sound as the delay time, regardless of whether the initial delay or mixed noise delay. The delay time of the caster is adjusted to a short time (less than 50 milliseconds), and the sound is similar to the reverb sound. In the range of 50 milliseconds to 0.2 seconds, the vibrato effect with different frequencies can be created; the echo interval is greater than O.2 seconds.

Valid Value

It is also called the root-mean-square value. The actual volume and intensity of the sound signal are very close to the human's auditory intensity. Therefore, it is generally necessary to determine whether the sound signal is suitable Based on the display of the valid value.

Far Field

Sound Field with a wavelength greater than twice, and the long wavelength of the sound wave (that is, when the frequency is 20Hz) is 17 meters P. Therefore, for the whole audio range, the sound field with a wavelength greater than 34 meters is a far field, when the size of a room reaches the far field, it is a large room. In the far field, there is no interference between sounds. Each time the distance is doubled, the sound pressure level degrades by 6 decibels.

Standing Wave

The ups and downs of sound produced by superposition interference of the opposite two columns of sound. Sound in the media interface (such as walls), the incident wave generates reflection, the reflection wave overlays with the human radiation wave, and the sound of the two sound sources will form a standing wave, the standing wave is the main cause of the phenomenon of sound dyeing (also called voice dyeing) during spatial transmission.

 Subjective Evaluation

The method of evaluating the voice based on the listening results of human ears is an important aspect of the sound quality evaluation. It can make a qualitative evaluation of the sound quality, which is simple and easy to use, however, the evaluation results are subjective and have high requirements on the evaluator's hearing level.

   Latency feedback Rate

The attenuation of Multiple echoes over time reflects the sound absorption coefficient of the room interface. In the latency effect, it is used to control the number of echo requests. When the feedback rate is continuously adjustable between 0% and 99%, the feedback rate is 0%, which is the latency effect; when the feedback rate is 99%, it is an endless echo.

Turning frequency

It is also called the cutoff frequency. The signal passing through the full level is the dividing frequency between the signal passing through the attenuation or cutoff signal. signals above this frequency can pass through the whole electric flat, signals of low frequency and this frequency cannot pass at all (in fact, the signal is quickly reduced ). For example, the frequency marked next to the low-cut or high-pass filter function key is the turning frequency, meaning that the sound below this frequency no longer exists, and the sound above this frequency passes normally, the turning frequency of some devices is continuously adjustable.Diffraction

 It is also known as diffraction. When a sound wave is transmitted, if it is blocked by an object close to the sound wave wavelength or equal to the wavelength, it will bypass this object and continue traveling. When the barrier is small (compared with the wavelength), it can still clearly hear the sound behind it; but when the barrier is large, it will form a significant reduction in the volume of the phonograph.

Noise Control

A device that uses the extensioner principle to reduce background noise. When the input signal is smaller than a certain degree (threshold), there is no output of the Nov. If the input signal is greater than a certain value, it is output normally, it can eliminate background noise during intermittent sound. In addition to reducing background noise, it can also be used to increase sound separation and process drum sound.

Refraction

Sound waves change the propagation direction on the contact surface of two substances (or materials with different density and media) due to changes in sound speed, and then enter the second kind of material phenomenon, for example, when the sound enters the wall from the air, the direction changes.

Total Noise Level

The background noise level emitted by the speaker when no sound signal is input. The total noise level of the system is related to audio engineering quality, sound system design, audio system debugging, and audio equipment.

Noise Waste Control

It is also known as equal sound control, which is a control method used to compensate the human ears for being sensitive to the middle voice and being slow to the bass and tweeter. It does not work when the amplifier turns on the high volume, when the amplifier volume is off for an hour, the control circuit in the audible area can increase the same head and bass of the signal to obtain the compensation for the audible frequency. As the voice of a person feels better at both the bass and the treble when the volume is large, and the voice and the high voice experience are poor at the volume hour, people will feel that the high voice and the low voice are appropriate when the volume is large, when the volume is small, the high-pitched bass is obviously insufficient. The audible control is a volume controller with compensation. It can compensate for the differences in auditory characteristics between ears at different volumes, regardless of whether the volume is on or off, the voice of the human ears only changes the sound and the tone remains unchanged.

Active Frequency Division

It is also known as electronic, voltage, or front-level. The divider is located before the power amplifier. After dividing the audio signal, it is allocated to each power amplifier according to different frequencies. Each power amplifier sends the audio power signals of different frequencies to each speaker, because the current is small, it is possible to implement an electronic active filter with low power. The advantage is that it is easy to adjust, The electroacoustic indicator is high, the signal loss is small, and the sound quality is good. However, because each method uses an independent power amplifier, the cost is high and the circuit structure is complex, it is applicable to professional sound reinforcement systems.

Phase shifting effect

A special sound effect in the effect device. There is a phase difference between the direct sound emitted by the sound source and the delayed reflection sound during the communication of the room. When the two sounds come together, it will produce a phenomenon called the comb filter effect in acoustics, that is, the mutual enhancement on some points forms the peak point, and the mutual offset forms the valley point on other points. The effect of the phasing is to take advantage of this phenomenon. It has the function of adjusting parameters of the amount of time delay between the direct sound (I .e. the unprocessed sound signal) and the reflected sound. it can control the peak and valley positions of the comb filter effect, so that the odd harmonic enhancement, even harmonic reduction, or odd harmonic reduction and even harmonic enhancement can be realized, in order to improve the sound tone and filter out excess harmonic components produced by certain distortion. The difference in the bee Valley amplitude of the Hang filter is determined by the mixing ratio of the delayed signal and the direct signal. The mixing ratio of the two is, with the largest difference and the most obvious effect, at this time, the peak amplitude is 6 decibels higher than the direct signal before the hybrid, and the Valley amplitude is O. A comb filter usually uses a short delay, which is between 1 to 20 milliseconds.

Maximum sound pressure level

In the Sound Amplifier System, the maximum steady-state sound pressure level that the speaker can emit is higher than the maximum sound pressure level, indicating that the power reserve of the system is large, and the sound sounds powerful, dynamic, and powerful. The main factors that determine the maximum sound pressure level of the Sound Amplifier System are power amplifier, total speaker power, and sound field size.

Longitudinal Wave

The propagation direction is the same as that of the vibration, also known as the dense wave. The sound wave is a longitudinal wave, which transfers the change of air pressure caused by the vibration to the mined area, and the air pressure is high (positive pressure, air in places with low air pressure (negative pressure) is sparse.

Damping Factor

One of the indicators that reflect the transient characteristics of audio equipment, the calculation method is: speaker impedance/amplifier internal resistance * wire impedance. When the speaker sends a sound, the reciprocating vibration of the basin will lead to low-frequency resonance. As long as the internal resistance of the power amplifier and the impedance of the sound box is very small, it is possible to short-circuit the induction EMR generated by the sound Band During the resonance of the speaker, to suppress resonance, so that the sound is clear and clear. The damping factor is too small, causing turbidity when the sound is dragged. If the sound is too large, the sound is hard and dry and tasteless. Generally, it is more suitable between 10 and 30.

Column Wave

Sound waves with a coaxial cylinder are generally produced by a linear sound source (such as a sound column) or a sound passing through a longer slit. The attenuation in transmission is smaller than that of a ball wave, doubling the distance, sound Pressure-level attenuation of 3 decibels, making the speaker send a column wave is an important means to increase the sound transmission distance of the sound system.

Pre-Delay

It is also called the initial delay, which refers to the time interval between early reflection and direct sound. The pre-delay time of rooms of different shapes and sizes is different, but it is mainly related to the room size, the average free route of the room can be used for calculation. When the Preset Delay of the timer is large, the system can achieve large space and Hall effects. At the same time, the system can avoid the sound dyeing caused by direct interference of the reflection sound, however, it is not recommended that the audio space be adjusted too long. Generally, the audio space should be adjusted to the actual space size of the room, and the sound should be clear and sound-visual.

Free Sound Field

Open spaces, such as open spaces, open spaces without any buildings, and open-air performances. The room with excellent Sound Absorption Performance (sound absorption coefficient close to 1) also belongs to the free sound field, such as the room and some acoustic laboratories, these rooms are generally used for the measurement and acoustic experiments of electro-acoustic devices (such as microphones, speakers, and speakers. In a free sound field, the sound is not reflected on the interface, which is equivalent to an infinite volume of space. There is no sound interference caused by reflection, so the sound is pure, but it sounds dry, the reverb time is almost equal to zero. Each time the distance is doubled, the sound pressure level degrades by 6 decibels.

 Yinyin

The frequency is a pure audio modulated by Sine. It is often used in the measurement of Hall Acoustic Characteristic indicators such as the reverb time, it can reduce the standing wave interference caused by sound interference and make the measurement results more accurate.