When it comes to speaker parameters, power is the most mentioned by us. Although in the past few years, everyone has not talked about the peak power of PMPO, nor has it been seen that the power value of hundreds of kilowatts is often presented in the speakers, but can the current power of speakers and multimedia speakers be really credible? We can easily see that two speakers with the same power have completely different performances at medium or high volume, and sound distortion is even more frequent. Is this set of data related to power output a "digital game"?
The main function of a power amplifier in a speaker is to amplify signals and provide sufficient power to the load (speaker). The impact of power amplifiers on sound quality mainly depends on whether the input signal can be amplified and transmitted without distortion, providing sufficient power to the load. The signal amplified and transmitted by a power amplifier is different from a simple harmonic signal and is a complex signal with instantaneous changes.
If we look at this signal from its waveform, the original signal has many spikes, their energy is not large, but the peaks are very sharp and high. These peaks have a small contribution to loudness, but a significant impact on sound quality. If clipping occurs, the amplified sound sounds dry and hard. This has a certain relationship with the details in what we commonly refer to as objective listening. If during power amplification, we only pay attention to the transmission of energy (corresponding to loudness) and do not pay attention to the changes in the waveform during the transmission process (resulting in distortion), then we may hear a loud but unpleasant sound.
For an active speaker, the power amplifier is partially located inside the speaker, and its job is also to drive the speaker and bring sufficient output power to the speaker. However, the nominal wording of speaker power we see is not very standardized. Ordinary speaker manufacturers label the power of "speakers" as the "output power (RMS)" of the power amplifier (a part of the power amplification circuit of an active speaker), while RMS (root mean square) refers to the root mean square. Currently, in multimedia speaker labeling, most of them are labeled as root mean square power.
The root mean square power is different from the uniform power and rated power. The detailed algorithm is to take the mean square of the power values at each point in the sample and then square it. We will not delve into the calculation of the root mean square for now. What we need to discuss next is the relationship between the "root mean square" power and rated power, as well as speaker power.
As mentioned earlier, the signal amplified by a power amplifier is a complex signal. Based on the investigation results of various instruments and program signals from different genres in acoustic engineering, the ratio of the maximum root mean square power (i.e. the peak to peak power of the program signal on the load) to the uniform root mean square power (i.e. the uniform power of the program signal on the load) of a large part of the program signal is 3-10, with a maximum of 12.7.
If the rated power of the power amplifier corresponds to the uniform root mean square power of the program signal, then the maximum output power of the power amplifier should be 3-10 times that to ensure that the output signal does not exhibit clipping. This is why we choose a power amplifier with a much higher power than the uniform root mean square power of the amplified program signal, which is also commonly referred to as power storage.
From the current low-end products, the maximum output power of power amplifiers should not be able to store power 10 times the root mean square power of the signal, and power storage is definitely different when designing. This is one of the reasons why we encounter distortion issues at different or higher volume levels during normal testing. On the other hand, when labeling the power of multimedia speakers, there is little clarification on whether this power is rated power, maximum output power, output RMS power, or even speaker power, which is a very chaotic parameter indicator.
Additionally, if we pay attention to the nameplates on some speakers, which also have a value related to power, what is the relationship between this value and the output of the power amplifier? In the Software design description, we can see the following statements: "In order to ensure the safety of the loudspeaker system to which the power amplifier is connected, the rated output power of the power amplifier is requested to be equivalent to the nominal power of the loudspeaker system to which it is connected", "In order to ensure sufficient power storage, the power amplifier with 1.2~2 times of the loudspeaker power is usually selected", etc. This formulation is incorrect in practice, as the power of a power amplifier is not the same concept as the power of a speaker.
The output power of a power amplifier generally refers to the sinusoidal output power under certain distortion limitations. We usually see that manufacturers indicate that the regular Total harmonic distortion is 0.1% after the power. When the output signal of the power amplifier on the rated load reaches this distortion, the output voltage is called the maximum output voltage. Using this voltage to calculate the output power of the power amplifier is the nominal output power of the power amplifier, which can also be understood as the maximum output power of the power amplifier.
The nominal power of the speaker, which is often provided by manufacturers, is the powder noise power, which refers to the power that is fed within the rated frequency range of the speaker, with a regular simulated program signal, and operates continuously for 100 hours without generating thermal or mechanical damage. Obviously, these two powers were regulated and tested from completely different perspectives, and the two are incomparable. If the manufacturer can provide the sinusoidal power of the speaker (referring to the power fed back when using a sinusoidal signal as the test signal), then the two are comparable.
However, manufacturers generally do not provide this data. So, can there be a certain correspondence between the powder noise power and sinusoidal power of speakers? The answer is - no! The powder noise power and sine power of speakers are completely different for different configurations, materials, and specifications of speakers, and the latter is also related to frequency. Therefore, it is not advisable to compare the power of power amplifiers with the nominal power of speakers in the design of speakers and amplifiers to characterize their power storage.
Obviously, comparing the power of the speaker with the power of the amplifier in numbers is meaningless. From the previous text, everyone can also understand that the current topic of whether similar power is sufficient and whether power storage is sufficient can only be based on objective listening experience. It is meaningless to see the authenticity of the labels on the speaker. Due to different labeling methods and specifications, there is naturally no comparability.
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.
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
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
