Search
Soft start protection
When a high current absorbing audio device is powered on, the current flowing through it can reach 4-10 times its average current value, which is an impact on the power grid and the device itself. In severe cases, it can damage the device. At this point, the soft start circuit can suppress the influx of current at the moment of equipment switching, allowing it to smoothly achieve normal protection of the equipment and not cause fluctuations in the power grid. This function is usually realized by the Negative temperature characteristic of thermistor (NTC)
DC protection
When the output stage of the power amplifier is damaged or the static bias is offset, it is possible to transmit a DC signal. For speakers, their working mode only generates impedance for AC signals, and for DC signals, they do not generate any impedance (equal to zero impedance). At this point, the current is infinite, so the coil of the speaker is equivalent to a heating wire that will quickly burn out under DC signals. Therefore, accurate and fast DC protection circuits are very important. The DC protection activation value of the power amplifier is usually set at 2V. When it is greater than or equal to this value, the power amplifier will cut off the output to protect the speaker. Of course, there are also power amplifiers that will cut off the output by burning the built-in DC fuse
Short circuit protection
When a short circuit occurs at the output end of the power amplifier due to certain reasons, the current output by the power amplifier will cycle through its own circuit and become infinite. This situation is very dangerous, so it is necessary to have accurate and fast short-circuit protection circuits to protect the power amplifier equipment. Each power amplifier must have a short-circuit protection function, which is vital and cannot be ignored
Usually, when a short circuit occurs in a power amplifier, it first controls the input signal to reduce its amplitude or even zero. If the situation does not improve (the current flowing inside the power amplifier still exceeds the safety value), it will suppress the output current, making the current flowing inside the power amplifier always lower than the safety value of the output stage transistor
Overcurrent protection
When the load of the power amplifier is too low but has not reached the short-circuit state, the short-circuit protection will not operate, but the output current will be very large and exceed the safe use value of the power amplifier. At this time, the overcurrent protection circuit will intervene and work. The usual approach is to control the input voltage and output current to ensure that the power amplifier always operates within a safe range
Overheat protection
A well-designed power amplifier will not experience overheating protection under normal use, and will only operate when the external usage environment is harsh or internal faults occur. The hottest part of the entire power amplifier is the C-pole (collector) of the output stage transistor, so the temperature sensor for overheat protection is generally installed in the closest place to the C-pole of the transistor or the hottest place on the heat sink. The threshold for overheating protection is generally 95 ℃, and there are also 105 ℃. The extreme withstand temperature of the transistor is 105 ℃.
In recording acoustics, loudness, loudness level, sound intensity, sound intensity level, sound pressure, sound pressure level, decibel, square, level, gain, pitch, and sound score are always headache inducing
Several concepts, let's briefly explain their meanings and differences, and organize their order.
Decibel
The decibel is the most commonly used unit in sound level measurement, abbreviated as dB. The lowercase d represents the decibel in English, while the uppercase B represents the bell in English
The use of lowercase d and uppercase B mainly indicates that the relationship between decibels and bell is 1:10, which means that 1 decibel is equal to one tenth of a bell.
It should be noted that 0dB does not represent a completely silent state, but rather represents the threshold point of the human ear, which is the lowest sound pressure level that can be detected by people with normal hearing.
Doubling the power represents a 3dB increase in gain (such as in a mix where one track of sound is 100dB, copying this track and playing it together will result in a total volume of 103dB, and
Non 100+100=200dB), while doubling the voltage represents a gain increase of 6dB.
level
A time variable, such as power or field, that is calculated as a mean or weighted value in a specific way within a specific time interval. Its units can be measured relative to the reference value
Represented in logarithmic form, such as' decibels'.
In recording, simply understood, level is a way of reflecting the current volume of sound in an electrical expression. For example, "Increase the level value of this track by 3dB"
It can be understood as' turning up the volume along this way by 3 decibels'.
gain
The degree of increase in current, voltage, or power of components, circuits, equipment, or systems is usually specified in decibels (dB). Here, it can be simply understood as
A state of increase.
pitch
Refers to the characteristic of hearing that distinguishes between high and low musical sounds. It is determined by the frequency of sound wave vibration, with higher frequencies indicating higher pitch; Low means low sound.
Cent
To improve the accuracy of measuring the height of sound, each "half tone" interval (such as C~# C or B~C) is defined as 100 minutes in metrology to facilitate the calculation of its error rate.
That is, 1cent is one percent of the Minor second interval.
Acoustic energy
The total amount or overall energy exhibited by sound in motion is usually expressed as sound energy.
sound intensity
The average sound energy per unit area passing through the direction perpendicular to the propagation of sound waves per unit time is called sound intensity. The sound intensity is expressed in I, in watts per square meter.
Sound intensity level
Research in Psychophysics shows that people's perception of sound intensity is not proportional to sound intensity, but proportional to its logarithm. This is precisely how people use sound intensity levels
To indicate the reason for sound intensity. The decibel value obtained by logarithmic operation of sound intensity that corresponds to human auditory perception, expressed in dBSPL. (SPL is the sound intensity level Sound
Intensity Level
sound pressure
The difference between the pressure and static pressure in the medium in the presence of sound waves.
Sound pressure level
Although sound intensity can theoretically represent the amplitude of sound waves at a certain point and can be measured to obtain its value, it is not commonly used in daily work
To elaborate on the magnitude of sound amplitude. Due to
In the entire audio circuit, there is a negative feedback structure. The deeper the negative feedback, the better the fidelity. If negative feedback is reflected in the feedback loop, it will cause flying marks. That is, howling. This is a headache for all operators. The best way now is to use other devices for frequency correction. The most effective device is a frequency shifter. It is to move the required frequency and amplify it. This way, there will be no negative feedback reflected in the sound reproduction. Usually, an equalizer is used. It can also be easily and effectively controlled. The equalizer will change the composition of the sound, causing it to vary strangely. Because using an equalizer for correction can change the composition of the sound, there is a possibility of feedback squealing in the microphone's pickup sound system.
The harm of microphone whistling is significant, mainly manifested in the following aspects:
1. When self-excited, the power amplifier will produce a large power output, which may exceed the capacity of the sound reinforcement equipment, and burn out the power amplifier and sound producing equipment.
When the feedback coefficient approaches 1, due to the comb filtering effect, the superposition between the delayed sound field and the direct sound field will make the amplified sound field narrower in terms of sound sense than the original sound field.
2. The delayed feedback of the speaker sound field will cause a series of delayed echoes throughout the entire system, and this echo will exacerbate the comb filtering effect, resulting in significant distortion of the reverberation tail - just sound distortion.
3. When howling, the output sound pressure is very high, which seriously affects the atmosphere of various activities.
To eliminate feedback howling, one must start with the necessary conditions for generating feedback howling. As long as one of the conditions can be broken, the goal can be achieved.
1、 Adjusting distance method
One of the most effective ways to avoid whistling and increase the amplification volume is to pick up the microphone as close as possible to the sound source, while using a non directional microphone. To be clear, directional microphones (especially sharp directional microphones) have minimal pickup attenuation from distant sound sources, and adjusting the distance has little effect on increasing the amplification volume and preventing whistling. Whether the sound reinforcement system is prone to howling is not directly related to the sensitivity of the microphone. However, highly sensitive microphones are highly directional and prone to squealing. Shortening the distance between the sound equipment and the audience can actually improve the loudness of the amplification. The total gain of the system can be appropriately reduced. If accompanied by a near field speaker with wide directionality, the microphone can be kept slightly away from howling. For the direct feedback sound field of speakers, the farther the microphone is from the speaker, the better, and the closer the speaker is to the audience, the better. The microphone should be placed on the back of the speaker's radiation direction. If the microphone is likely to be carried around, the speaker should be placed in a place where the microphone cannot be very close.
2、 Frequency equalization method (broadband notch method)
Due to the frequency curve of microphone pickup and sound production equipment not being ideal flat straight lines (especially for some poor quality playback equipment), as well as the acoustic resonance effect of the hall sound field, the frequency response fluctuates greatly. A frequency equalizer can be used to compensate for the sound amplification curve, adjusting the frequency response of the system to an approximate straight line, so that the gain of each frequency band is basically consistent, and improving the sound transmission gain of the system. Equalizers with 21 or more segments should be used, and parameter equalizers should be configured in areas with higher requirements. Feedback suppressors can be used when higher requirements are required. Actually, the sound reinforcement system is experiencing a backlash
3、 Feedback suppressor method (narrowband notch method)
In high demand situations, such as some live singing places, audio feedback automatic suppression devices are commonly used. This device can automatically track the frequency of feedback points, adjust the Q value bandwidth, automatically eliminate audio feedback while maximizing the protection of sound quality. The principle is to suppress whistling by trapping waves. For example, Sabine's FBX series feedback suppressor is a 9-band narrowband automatic voltage limiting device controlled by a microcomputer, which can effectively distinguish feedback self-excited signals from music signals,
When the system is self-excited, it can react quickly, and set a very narrow digital filter at the feedback frequency point. Its notch depth will also be automatically set. The filter bandwidth is only 1/3 Octave band. Such a narrow notch frequency band will hardly affect the loudness and timbre
4、 Reverse offset method
Anti phase cancellation to prevent self-excitation is common in high-frequency amplification circuits. Two microphones of the same specification can be used in the audio amplification circuit to pick up direct sound and reflected sound, and the reflected sound signal can be phased out before entering the power amplifier through an inverting circuit, effectively preventing self excitation of whistling.
5、 Phase modulation method
The self-excited howling of the amplifying system has a positive feedback loop. If the microphone signal is phased, it will disrupt the phase condition of self-excited howling, thereby preventing the system from self-excited howling. There is data indicating that the stability is best when the phase deviation value is 140 °; Moreover, the higher the modulation frequency, the better the stability of the system. In order to ensure that the processed sound quality does not experience significant distortion, the maximum allowable value of its phase modulation frequency is 4Hz.
Finally, after adjusting various devices, it is strictly prohibited for others to move around, including DJs who are unfamiliar with the performance of the equipment and only know how to turn it on, off, and adjust the volume. Therefore, Aircross Audio has also taken certain measures on this, that is, each Effects unit can be locked. Professionals can press the lock key after debugging.
When using the Mixing console, no matter it is simple, complex, high-grade and low-grade, if you want to give full play to its performance, you must master the skills and experience of using the Mixing console. Taboos to use the Mixing console mainly include: Do not press buttons and adjust various knobs randomly, do not adjust the channel gain knob randomly, do not use or use parametric tone controller randomly, do not pay attention to the audio image adjustment button, do not pay attention to the auxiliary output knob, do not pay attention to the cooperation between the shunt volume fader and the main output volume fader, do not turn off unused volume faders in time, and do not turn on the phantom power switch randomly, Below is a detailed introduction from Gedi Audio:
1. Avoid randomly pressing buttons and adjusting various knobs
Although the Mixing console is simple and complex, the number of channels (channels) varies. In cases where we are not familiar with the audio console or do not understand the on-site system connection and usage requirements, it is strictly prohibited to press buttons and adjust various knobs indiscriminately. As the core equipment of the audio system, different systems have different connections. When the system is debugged, the relevant knobs are already in working state. If we press and adjust them carelessly, their working state will be destroyed. The effect will be destroyed in the light, and the system equipment may be damaged in the heavy.
2. Avoid randomly adjusting the channel gain knob
Generally, each input channel of the Mixing console has a gain selection button (with non attenuation and attenuation of 20 dB) and a gain adjustment knob. The function of these two buttons is to control the size of the audio signal entering that channel. If the signal is too large, it will inevitably cause the input amplifier to enter a saturation state, causing serious distortion of the audio signal; If the signal is too small, it either makes the sound very light or reduces the signal-to-noise ratio, which affects the sound reinforcement effect. Do not randomly adjust the channel gain knob. The correct approach should be to select the gain selection button based on the type and size of the input signal. This button has two options. If pressed, the attenuation will be 20-30dB (depending on the Mixing console); If not pressed, there will be no attenuation. When we input a microphone type signal (usually a few millivolts in size), there is no need to press the attenuation button; When we input a line level signal (such as DVD, card holder, etc.), we must press the attenuation button. The second step is to adjust the gain adjustment knob again to ensure it is in the correct position. When the PEAK indicator light is very or constantly on, turn the knob counterclockwise by 5-10 ° (a scale mark)
3. Avoid indiscriminate use of parametric tone controllers
Generally, the Mixing console is equipped with 3 or 4 segment parametric tone controller, which is composed of 6 or 8 knobs. These knobs are divided into two categories: one is gain control, labeled with -10 to+10, and the other is frequency knobs, which can be increased or decreased according to on-site needs. The adjustment of these knobs requires a certain level of tuning experience. In cases where you are unfamiliar or lack tuning experience, remember not to adjust these knobs indiscriminately, especially the gain knob, which is generally at "0". But we must also oppose the use of parametric tone controllers in any case
4. Avoid not paying attention to the sound and image adjustment buttons
In each input channel of the Mixing console, the audio and image adjustment button PAN is set and marked with L or
The speakers that provide ground return sound reinforcement for bands and choirs on stage are generally referred to as stage monitoring speakers or stage return speakers
Why is optimizing the gain before feedback before graphic equalization the most commonly misunderstood part of sound reinforcement systems? The return speaker can be placed anywhere on the stage, and the following tips can help you overcome misunderstandings and gain more feedback before returning
Even if your microphone hasn't screamed yet, you may still hear a deep "hollow" sound. Although the acoustic environment of the building itself can also cause this problem, it is highly likely that the microphone has picked up too many signals and fed them into the system signal chain, and then you will hear a very harsh feedback howl
Before we start discussing system optimization, there are some knowledge points that you must master. To understand the placement of stage monitoring speakers, you must have an understanding of the type of microphone you are using
According to the polar pattern of microphones, they are generally divided into omnidirectional microphones and single directional microphones. Omnidirectional microphones pick up sound evenly from all directions, so they are generally rarely used for live performances. A single directional microphone only picks up sound from a specific direction
When we point the microphone towards the stage monitoring speaker, there is a high possibility of feedback howling, so pointing the microphone away from the stage monitoring speaker seems to be the best choice. But the prerequisite is that you are using a cardioid microphone. The heart pointing is named after its polar range, which is very similar to a heart
Next, let's take a look at the polar pattern of a supercardioid microphone. You will find that pointing the microphone away from the monitoring speaker is not necessarily the best method, as the sensitivity of a supercardioid microphone to sound from the front and rear is almost the same. You will find that the best placement method for supercardioid microphones is to slightly deviate from the axial direction of the monitoring speaker. Using two stage monitoring speakers, or adjusting the angle of the microphone to face the performer as closely as possible, will have a better effect.
