Tuning of vocal effects

For the treatment of vocal effects, most people use repeated tentative adjustment methods to find the best processing effect of the sound effect. The shortcomings of this tuning method are very obvious:

(1) Finding an ideal tuning effect requires many guesses, so it takes a long time.
(2) The better tuning effect is often encountered by chance, which is not helpful for the summary of the tuning rules, and it will not be easy to reproduce in the future.
(3) The fixed parameters and adjustable parameters of different devices are different, so the experience of using one device is usually not applicable to another device.

Developed to the current effect processing equipment, there are not many technical methods for changing the timbre of the sound source. Among them, the three most common methods are frequency equalization, delay feedback, and limiting distortion. However, the different parameters of these effect processing equipment The sound produced by the combination is quite different.

The parameter setting of the effect processor can have many items, especially the delay feedback, and the setting of this analog reverb effect parameter can theoretically reach as many as dozens of items. Of course, these highly professional parameters are difficult for most people to understand and do not know how to understand. Therefore, most effect processing equipment only set one or two adjustable parameters, and the adjustable range is relatively narrow. This simple adjustment effect processing device allows people to make tentative adjustments on it without causing too many problems. However, for occasions where more precise tuning is required for effect processing, such as in a multi-track recording system, more professional effect processing equipment must be used to make more elaborate effect processing.

Frequency equalization

Obviously, the more frequency-balanced segments, the more detailed the effect processing. In addition to graphic equalization, the equalization unit of general tuning usually only has three or four frequency bands, which obviously cannot meet the requirements of accurately processing the sound source. In order to be flexible enough to perform arbitrary equalization processing on human voices, we recommend using four-band frequency equalization with adjustable gain, frequency and width.

Most of the adjustable parameters for frequency equalization have only one gain, but this does not mean that the other two parameters do not exist, and these two parameters are fixed parameters that are not adjustable. Of course, it is not difficult to set these two parameters to be adjustable, but these will increase the cost of the equipment and make its adjustment complicated. Therefore, the parametric equalization circuit with adjustable gain, frequency point and width can usually only be seen on high-end equipment.

In fact, the gain, frequency point and width are all adjustable frequency balance, it is almost impossible to find an ideal timbre using the method of random guessing. Here we must study the physical characteristics of audio signals, technical parameters and their corresponding relationship in the sense of hearing of the human ear.

The frequency distribution of the human sound source is special. In terms of its pronunciation, he has three parts: one is the musical sound generated by the vibration of the vocal cords. The pronunciation of this part is the most flexible. The spectrum changes caused by different pitches and different pronunciation methods It is also very large; second, the shape of the nasal cavity is relatively stable, so the harmonic frequency spectrum distribution generated by its resonance does not change much; third, the frictional sound of the oral air flow between the teeth, this tooth sound is basically independent of the musical sound generated by the vocal cord vibration .

Frequency equalization can roughly separate these three parts of the spectrum. The frequency band for adjusting the nasal sounds is 500Hz, the midpoint frequency of the following balance is generally 80 ~ 150Hz, and the balance bandwidth is 4 octave. For example, you can set 100Hz as the midpoint of frequency equalization. The equalization curve should be a gentle transition from 100 to 400Hz. The adjustment range of the equalization gain can be + 10Db ~ -6dB. Everyone should be reminded here: the monitor speakers that make this adjustment must not use small boxes with low-frequency sounds to prevent the nasal sounds from being unintentionally excessive.

The frequency spectrum of vocal tones also changes greatly with the tone, so the equalization curve for adjusting the tones should be very gentle, the midpoint frequency of the equalization can be 1000 ~ 3400Hz, and the equalization bandwidth is six octave. This frequency band controls the brightness of the singing voice, and upward adjustment can gently enhance the brightness of the human voice. However, if you want to reduce the brightness of the human voice, the situation will be more complicated. Generally speaking, the vocals with excessively bright sounds have a strong spectrum around 2500Hz. Here we can use equalization bandwidth of 1/2 octave and equalization gain of about -4dB to find a frequency with the best effect around 2500Hz. Just click.

The frequency spectrum of vocal tooth sound is distributed above 4kHz. Since this band also contains part of the music frequency spectrum, it is recommended that the frequency band for adjusting the tooth sound should be 6 ~ 16KHz, the equalization bandwidth is 3 octave, the midpoint frequency of the equalization is generally 10 ~ 12KHz, and the equalization gain can be adjusted upwards to + 10Db ; If you want to reduce the loudness of vocal tooth sound downward, you should use the equalization bandwidth of 1/2 octave, the equalization midpoint frequency of 6800Hz, and the minimum of its equalization gain can be reduced to -10Db.

From the above analysis, it can be seen that when frequency equalization processing is performed on the human voice, the frequency band enhancement for highlighting a certain sound sense, try to use a broadband equalization with a gentle curve. This is to make the frequency distribution of the three parts of vocal nasal, musical and dental sounds even and coherent, so that their pronunciation is natural and smooth. Theoretically, the loudness of the human voice should be kept constant at any sound.

In order to process specific effects on the basis of not destroying the natural sense of life, you can use 1/5 octave equalization processing, specifically in the following situations:

(1) The sound is narrow and lacks thickness. It can use 1/5 octave attenuation at 800Hz. The maximum attenuation can be -3dB.
(2) The tongue-shaped tooth sound has a screaming sensation, and the "hush" sound lacks clarity. It can be attenuated by 1/5 octave at 2500Hz. The maximum attenuation can be -6Db.

For the equalization processing of the sound source, it is best to use an equalizer that can display the equalization curve. The equalization gain adjustment button on the general digital mixer equalizer is marked with "G", the equalization frequency adjustment button is marked with "F", and the equalization bandwidth adjustment button is marked with "F" or "Q".

Delayed feedback

Delay feedback is the most widely used in effect processing, but it is also the most complicated method. Among them, the basic processing methods of reverb, chorus, flanger, echo and other effects are delay feedback.

1. Reverb

The reverb effect is mainly used to increase the sense of fusion of the sound source. The delayed sound array of natural sound sources is very dense and complex, so the procedures for simulating the reverberation effect are also complicated and variable. Common parameters are as follows:

Reverberation time: Digital reverberators that can realistically simulate natural reverberation have a complex set of procedures. Although there are many technical parameters adjustable, the adjustment of these technical parameters will not be more than the original effect. Naturally, especially the reverberation time.

High frequency roll-off: This parameter is used to simulate the effect of air absorption of high frequencies during natural reverberation to produce a more natural reverberation effect. The general adjustable range of high frequency mixing is 0.1 ~ 1.0. When this value is higher, the reverb effect is also closer to natural reverb; when this value is lower, the reverb effect is clearer.

Diffusion: This parameter can adjust the growth rate of the density of the reverb sound array. The adjustable range is 0 ~ 10. When the value is higher, the reverb effect is richer and warmer; when the value is lower, the reverb effect is more Open and cold.

Pre-delay: The establishment of the natural reverb sound array will be delayed for a period of time, and the pre-delay is set for simulating secondary effects.

Sound array density: This parameter can adjust the density of the sound array. When the value is higher, the reverb effect is warmer, but there is obvious sound staining; when the value is lower, the reverb effect is deeper, and the cut sound staining is also more weak.

Frequency modulation: This is a technical parameter because the sound array density of electronic reverberation is sparse than natural reverberation. In order to make the reverberation sound smoother and coherent, the delay time of the reverberation array needs to be modulated. This technology can effectively eliminate the split sound of the delayed sound array, and can increase the softness of the reverb sound.

Modulation depth: refers to the modulation depth of the above frequency modulation circuit.

Reverb type: The natural reverb sound arrays in different rooms are also quite different, and this difference cannot be expressed by one or two parameters. In digital reverb, different natural reverbs require different procedures. The options are generally small hall (S-Hall), hall (L-Hall), room (Room), random (Random), anti-reverberation (Reverse), steel plate (Plate), spring (Sprirg) and so on. Among them, the reverberation in the small hall and hall room is a natural reverberation effect; the steel plate and spring reverberation can simulate the processing effect of early mechanical reverberation.

Room size: This is set to match the natural reverb effect and is easy to understand.

Room activity: Activity is the reverberation intensity of a room. It is related to the sound absorption characteristics of the room wall. This parameter is used to adjust this characteristic.

The balance of early reflection and reverberation: the early reflection of reverberation is closely related to the characteristics of its processing effect, and the sound of the reverberation array is not so varied, so the two parts of the digital reverberator are generated separately. This parameter It is used to adjust the loudness balance between the early reflected sound and the reverb sound array.

Delay time of early reflected sound and reverb sound: that is, the delay time control between early reflected sound and reverb sound array. If this time is longer, the front section of the reverberation effect will be clearer; if this time is shorter, the early reflections and reverberation will overlap, and the front section of the reverberation effect will be muddy.

In addition to the above adjustable parameters, the reverberation effect has some other auxiliary parameters, such as low-pass filtering, high-pass filtering, and loudness balance control of direct / reverb sound.

2. Delay

Delay is the effect processing after delaying the sound source for a period of time, and then wanting to play it. Depending on the delay time, it can produce chorus, flange, echo and other effects.

When the delay time is between 3 ~ 35ms, the human ear can't feel the existence of lag sound, and after superimposed with the original sound source, it will produce a "comb filter" effect due to its phase interference. This is the flange effect. If the delay time is more than 50ms, the delay sound is clearly discernible, and the processing effect at this time is the echo. Echo processing is generally used to produce simple reverb effects.

The adjustable parameters of delay, chorus, flanger, echo and other effects are almost the same, specifically the following items:

* Delay time (Dly), which is the delay time adjustment of the main delay circuit.
* Feedback gain (FB Gain), that is, gain control of delay feedback.
* Feedback high frequency ratio (Hi RaTIo), that is, high frequency attenuation control on the feedback loop.
* Modulation frequency (Freq) refers to the frequency modulation period of the main delay.
* Depth of modulation (Depth) refers to the modulation depth of the above frequency modulation circuit.
* High frequency gain (HF) refers to high frequency equalization control.
* Pre-delay (Ini Dly) refers to the adjustment of the pre-delay time of the main delay circuit.
* Equalization frequency (EQ F), where frequency equalization is used for tone adjustment, this is the midpoint frequency selection for equalization.

Since the effects produced by the delay are more complicated and changeable, if you are not an expert in effect processing, it is recommended to use the preset parameters provided by the device, because the processing effects given by these preset parameters are generally better.

Acoustic excitation

By performing shallow amplitude limiting processing on the sound source signal, the sound will produce a "saturated" sound effect so that its pronunciation will have an effect of increasing the loudness without increasing its actual loudness.

Some digital effects are also equipped with a nonlinear saturation effect, which is to process the amplitude of the signal, simulating the nonlinearity caused by the saturation of the large battery signal on the triode, thus producing a "hard" sound effect.

Because the limiting distortion is mainly caused by extra high-order harmonic components, the newly designed exciter, in order to make its processing effect softer, is to simulate the limiting distortion by placing high-order carrier components in the sound source. , To create a less "hoar" sound stimulation effect.

In addition, the original signal is processed by a high-pass filter for enhancing higher harmonics, and then superimposed on the delayed original signal to create a clear sound effect. Obviously, this kind of processing can produce less noisy excitation processing.

Excitation processing is similar to overload distortion of audio equipment, so excessive excitation of the sound source will produce an unpleasant noisy feeling. Since the fidelity of early audio equipment was not high, people were used to the slightly noisy sound, but for the high-fidelity sound with clean sound, they were not used to it and felt that their pronunciation was too soft. Among the human voice sources, except for a small number of specially trained people, most of the speeches lack stiffness, so the stimulation process here is very necessary.

There are several situations for the stimulation of vocals:

(1) For the excitation processing of vocal sounds, its frequency spectrum distribution takes 2500 Hz as the midpoint. The effect of this kind of excitation is more natural and comfortable, and it has a more obvious effect on increasing the prominent sense of the sound source.
(2) For the excitation processing of vocal nasal sounds, the frequency spectrum distribution is centered at 500 Hz. Such stimulation can effectively increase the stiffness of the human voice.
(3) Excitation of the vocals around 800Hz can increase the hustle and bustle of the sound source. Of course, the use of this processing method should be very careful, and it is best used only for rock music.
(4) Excitation processing should not be used for the spectrum in the range of 3500-6800 Hz for human voices, because it easily causes unpleasant noisy sounds from the sound source.
(5) Tooth sounds of human voices should generally avoid using excitation processing, because the distortion in this frequency band is easily noticeable. Of course, if you use a digital exciter with a softer excitation effect, you can also do a slight excitation process on the tooth sound to increase the sense of clarity of the tooth sound. The frequency spectrum it handles should be above 7200 Hz.

The incentive process of singing pronunciation is usually conservative. In actual tuning, the sound effect of the excitation process may gradually weaken with long-term listening, so when adjusting the excitation effect, the time should not exceed 10 minutes.

It is best to use a digital effects processor for the excitation processing of human sound sources. It usually has the following adjustment parameters:

1. Input gain (Gmn), used to adjust the input level, pay attention not to overload the device here.
2. Tuning frequency (Tuning), select an appropriate frequency according to the frequency band to be processed.
3. Drive level (Drive), used to adjust the depth of excitation. When the driving level is large, the effect is noisy; when the driving level is small, the effect is gentle.
4. Mixing ratio (Mix), that is, the loudness ratio of the original signal and the effect signal.

Overall planning for effect processing

For the fine processing of human voice sources, you need to use an all-digital mixer, at least three digital effects and a digital exciter, and the connection method is shown in the figure.

First of all, on the mixer, use the channel equalization control unit to adjust the timbre of the human voice, so that its sound feel can be improved. Here are a few common examples.

(1) The frequency band around 8OOHz can cause a certain amount of boredom, so it can be attenuated by a maximum of 15dB in this frequency band, and the frequency bandwidth is 1/5 octave, which is used to improve the overall impression of human voice pronunciation;
(2) The frequency band around 68O0Hz can make the human voice produce a squeal and harsh feeling, and can be attenuated by a maximum of 10dB in this frequency band, and the frequency band width is â…• octave to reduce the squeal of the tooth sound;
(3) For those who feel too bright and have ear sticks, the maximum attenuation is 8dB at 3400Hz, and the frequency bandwidth is 1/3 octave;
(4) For those with excessive nasal sounds, it can be properly attenuated in the frequency band below 500Hz, and the attenuation bandwidth is 3 octave;
(5) Due to the influence of human ear sensitivity, the ultra-high frequency band of dental sound needs to be increased by 6dB at 12KHz (band width is 2 octave) so that its loudness can be balanced with the musical sound of human voice.

The above equalization process is more suitable for live amplification. If it is multi-track recording or program forwarding, the gain adjustment should be halved.

After the balance is adjusted, adjust the exciter again. First adjust the drive level and mixing level of the exciter to the maximum state, and the frequency tuning is placed at 2500Hz. At this time, if the pronunciation is already noisy or the sound is too hard, you can lower the drive level. You should pay attention to this adjustment. What changes is the hardness of the sound source. If the drive level is adjusted to a higher position, and only the mixing level is lowered, the sound of the high-hardness sound will remain unchanged, but it will be slightly masked by the original sound without the excitation treatment. This phenomenon is more obvious when the depth of excitation is very strong. The former sound is the original sound, and the latter sound can produce two layers of sound, which has the effect of increasing the level of human voice.

Generally, one exciter can only handle one frequency band, and the connection of many single-function exciters requires that they cannot be connected in parallel, only in series. If you need to add excitation to multiple frequency bands of the sound source, it is recommended that in the connection of the equipment shown in the figure, the reverb should use multiple effects with excitation processing (such as YAMAHA SPX990). 800Hz and 7200Hz frequency bands, the excitation function on the reverberator is used to process the 2500Hz frequency band.

Once again, everyone is reminded that the adjustment time of the incentive process should not be too long, so as to prevent the human ear from fatigue, it is impossible to accurately identify whether the level of incentive is appropriate.

The last step is to adjust the reverb effect. The reverb effect here contains two aspects, one is basic retouching and the other is strong staining.

The basic retouch processing is mainly to increase the harmony of the sound source, but it can not make people hear the reverberation of the room. The strong coloring effect of the reverberation process here is mainly used to generate the reverberation rendering of the sound source. The processing methods are as follows:

(1) Generate a sense of space. Use the hall or room reverb effect. The obvious natural reverberation effect of analogue reverberation is a simple and effective way of reverberation processing. The frequency band around 3500 Hz on this effect channel is slightly improved to produce a high-brightness sound with good penetration. Of course, there is also a shortcoming, that is, the treatment effect is relatively turbid, sometimes with a "sullen can" sound.
(2) Generate echo. The delay feedback processing with long delay time can simulate the valley echo effect; the processing delay time is generally in sync with the rhythm of the singing song. In order to make its effect more distant, the frequency band below 1600Hz and above 3800Hz can be moderately attenuated. Simulating the valley echo effect, there are ready-made programs available on many digital effects processors.
(3) Generate a blended sound background. The reverberation effect of the reverberation of aftertones is very effective for the beautification of human voice sources, and almost all vocal singing must use reverberation. On the premise that it does not cause the pronunciation to become muddy, or cause a "sullen jar" sound, we think that the stronger the reverberation effect, the better, but in fact, when the reverberation effect is still very weak, the pronunciation has become muddy and causes obvious The "sullen jar" sound.

In order to create a harmonious sound background without causing the pronunciation to become muddy, or to cause a "sullen can" sound. The following effect processing method is recommended, namely the delay-reverb series processing method. The delay time of this kind of processing is generally 200-600ms, the feedback gain is 40% -60%, the reverberation uses the hall reverberation effect, and the reverberation time is 2-8s. The reverberation effect after series processing requires smoothness and coherence. If the processed sound head is exposed, the following adjustments can be made, one is to shorten the delay time, the other is to increase the reverberation loudness, and the third is to increase the reverberation time.

The strong dyeing effect of the reverberation treatment should generally be carried out under the premise of basic retouching, so that the strong dyeing treatment can be weaker.

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