Overview of tuning processing for vocal effects

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

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(1) Finding an ideal tuning effect requires multiple guesses, so it takes time for the instructor.
(2) The better tuning effect is often encountered by chance, which does not help the summary of the tuning rules, and is not 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.

Up to the current effect processing equipment, there are not many technical means for changing the sound of the sound source. Among them, only three basic methods, such as frequency equalization, delay feedback, and limiting distortion, are commonly used. However, different parameters of these effects processing devices The sounds produced by the combination are quite different.

The parameter setting of the effect processor can have many items, especially the delay feedback. The setting of the analog reverb effect parameter can theoretically reach 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 of the effect processing devices only set one or two adjustable parameters, and the adjustable range is also narrow. This simple adjustment of the effect processing device allows people to make tentative adjustments on it without causing too much problems. However, for fine-tuning applications where the effect processing is more sophisticated, such as in a multi-track recording system, more professional effect processing equipment must be used to make more detailed effects processing.

Frequency equalization

Obviously, the more segments of frequency equalization, the higher the level of effect processing. In addition to the graphic equalization, the general tuning equalization unit usually has only three or four frequency bands, which obviously cannot meet the requirements of accurately processing the sound source. In order to be flexible enough to arbitrarily equalize the vocals, we recommend using a four-band frequency equalization with adjustable gain, frequency and width.

Most of the frequency-adjusted tunable parameters 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 device and complicate its adjustment. Therefore, the parametric equalization circuit with adjustable gain, frequency and width is usually only seen on high-end equipment.

In fact, the gain, frequency, and width are all adjustable frequency equalizations, and it is almost impossible to find an ideal tone using a guesswork. Here we must study the physical characteristics of the audio signal, the technical parameters and his correspondence in the human ear.

The spectrum of the human voice source is quite special. In terms of its pronunciation, he has three parts: one is the tone produced by the vocal cord vibration. The pronunciation of this part is the most flexible, and the spectrum changes caused by different pitches and different pronunciations. It is also very large; the second is that the shape of the nasal cavity is relatively stable, so the resonance audio spectrum distribution generated by its resonance does not change much; the third is the frictional sound of the oral airflow between the teeth, which is basically independent of the tone produced by the vocal cord vibration. .

Frequency equalization can roughly separate the three parts of the spectrum. The frequency segment of the nasal tone is adjusted at 500 Hz. The midpoint frequency of the following equalization is generally 80-150 Hz, and the equalization bandwidth is 4 octaves. For example, 100Hz can be set as the midpoint of frequency equalization, and the equilibrium curve should be smoothly transitioned from 100~400Hz. The adjustment range of equalization gain can be +10Db~ -6dB. Here you should be reminded that the monitors that make this adjustment should not use small boxes with low frequency pronunciation to avoid the nasal sounds being unintentionally overweighted.

The spectrum of vocal music varies greatly with the pitch, so the balance curve for adjusting the tone should be very flat. The balanced midpoint frequency can be 1000~3400Hz, and the equalization bandwidth is six octaves. This frequency band controls the brightness of the singing voice, and the upward adjustment can gently increase 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, the vocals that are too bright and bright are mostly near 2500 Hz. Here we can use equalization bandwidth of 1/2 octave, equalization gain of about -4 dB, and find the best frequency around 2500 Hz. Just click.

The spectrum of vocal teeth is distributed above 4 kHz. Since this band also contains some music and audio spectrum, it is recommended that the frequency band of the adjustment tooth tone should be 6~16KHz, the equalization bandwidth is 3 octaves, the equalization midpoint frequency is generally 10~12KHz, and the equalization gain can be adjusted upwards to +10Db. If you want to reduce the loudness of the vocal tones down, you should use an equalization bandwidth of 1/2 octave and an equalization midpoint frequency of 6800 Hz. The equalization gain can be reduced down to -10 Db.

It can be seen from the above analysis that when the frequency equalization processing is performed on the vocal, the frequency band lifting for highlighting a certain sound sensation is tried to use the flat broadband equalization as much as possible. This is to make the spectrum distribution of the three parts of the human voice, the music, and the tooth sound uniform and uniform, so that the pronunciation is natural and smooth. In theory, the loudness should be kept constant when the human voice makes any sound.

In order to deal with the specific effects without destroying the natural feelings of life, 1/5 octave equalization processing can be used, in the following cases:

(1) The sound is narrow and lacks thickness. It can be attenuated by 1/5 octave at 800 Hz. The maximum attenuation can be -3dB.
(2) The squeaking sound of the squeaky squeaky sound, the "嘘" sound lacks a clear feeling, and can be attenuated by 1/5 octave at 2500 Hz, and the maximum attenuation can be -6 Db.

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

Delayed feedback

Delay feedback is the most widely used but the most complicated way to handle effects. Among them, the effects of reverberation, chorus, edging, echo, etc., are basically delayed feedback.

1, reverberation

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 program for simulating the reverb effect is also complicated and varied. The common parameters are as follows:

Reverberation time: A digital reverb that can realistically simulate natural reverberation has a complicated program. Although there are many technical parameters that can be adjusted, the adjustment of these technical parameters will not be more than the original effect. Natural, especially reverberation time.

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

Diffusion: This parameter can adjust the growth rate of the reverberation sound array density. The adjustable range is 0~10. When the value is higher, the reverberation effect is richer and warmer. When the value is lower, the reverberation effect is better. Empty and secluded.

Pre-delay: The establishment of the natural reverberation array is delayed for a period of time, and the pre-delay is set for the analog sub-effect.

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

Frequency Modulation: This is a technical parameter because the acoustic density of the electronic reverberation is sparse than the natural reverberation. In order to make the reverberation sound smoother and more consistent, the delay time of the reverberant sound array needs to be modulated. This technology can effectively eliminate the cracking sound of the delayed sound array, and can increase the soft feeling of the reverberation sound.

Treatment depth: refers to the depth of adjustment of the above FM circuit.

Reverb type: The natural reverberation array of different rooms is also different, and this difference is not one or two parameters. In a digital reverb, different natural reverbs require different procedures. The options are generally S-Hall, L-Hall, Room, Random, Reverse, Plate, Sprirg, and the like. Among them, reverberation in small halls and halls is a natural reverberation effect; steel plate and spring reverberation can simulate the effect of early mechanical reverberation.

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

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

The balance between early reflection sound and reverberation sound: the early reflection sound of reverberation is closely related to its processing effect characteristics, while the sound of reverberation sound array is not so varied, so the generation of these two parts of digital reverberator is separate, this parameter It is used to adjust the loudness balance between the early reflected sound and the reverberant sound array.

Delay time of early reflection and reverberation: delay time control between early reflection and reverberation. This time is longer, the front part of the reverberation effect is clearer; this time is shorter, the early reflection sound and the reverberation sound will overlap, and the front part of the reverberation effect is more turbid.

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

2, delay

The delay is to delay the sound source for a period of time before you want to play the effect. Depending on the delay time, chorus, chrome, echo, etc. can be generated separately.

When the delay time is between 3 and 35 ms, the human ear does not feel the existence of a lag sound, and when it is superimposed with the original sound source, it will produce a "comb filter" effect due to its phase interference, which is the edging effect. If the delay time is above 50ms, the delay sound is clearly identifiable, and the processing effect at this time is the echo. Echo processing is generally used to produce a simple reverb effect.
Adjustable parameters for delay, chorus, chrome, echo, etc. are similar, specifically the following:

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

Since the effects of delays are complex and variable, if it is not an effect processing expert, it is recommended to use the preset parameters provided by the device, because the processing results given by these preset parameters are generally better.

Acoustic excitation

By performing a shallow clipping process on the source signal, the sound produces a "saturated" sound effect that allows the sound to have an louder effect without increasing its actual loudness.

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

Since the limiting distortion is mainly caused by the generation of additional high-order harmonic components, the newly designed exciter, in order to make its processing effect softer, simulates the limiting distortion by setting the high-order carrier component in the sound source. To create a less hoarse sound effect.

In addition, the original signal is processed by a high-pass filter for enhancing the 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 incentive processing.

The excitation process is similar to the overload distortion of the audio equipment, so excessive excitation of the sound source can produce an unpleasant and noisy feeling. Because the fidelity of early audio equipment is not high, people have become accustomed to the slightly noisy sound, but for the high-fidelity sound with clean sound, they are not used to it, and their pronunciation is too weak. Among the human voice sources, except for a small number of specially trained people, most of the speeches lack strength, so the incentive processing here is very necessary.

There are several situations in which the vocal incentives are handled:

(1) For the excitation processing of vocal music, the spectrum distribution is 2500 Hz as the midpoint. The effect of this kind of incentive is more natural and comfortable, and the effect on increasing the sense of sound source is also obvious.
(2) For the excitation processing of the human voice nasal sound, the spectrum distribution is 500 Hz as the midpoint. This kind of incentive can effectively increase the sense of stiffness of the human voice.
(3) Excitation of the vocal 800Hz nearby can increase the sensation of the sound source. Of course, the use of this processing method should be very cautious, and it is best to use it only for rock music.
(4) It is not advisable to use the excitation process for the spectrum in the range of 3500-6800 Hz, because it is easy to make the sound source unpleasant and noisy.
(5) Excitation processing should generally be avoided for vocal sounds, as the distortion of this band is easily noticeable. Of course, if a digital actuator with a softer excitation effect is used, it is also possible to perform a slight excitation process on the tooth tone for aggravating the sense of sound of the tooth. The spectrum it processes should be above 7200 Hz.

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

For the incentive processing of human sound sources, it is best to use a digital effects processor. It usually has the following adjustment parameters:

1. Input gain (Gmn), used to adjust the input level, be careful not to overload the device here.
2. Tuning frequency (Tuning), select a suitable frequency according to the frequency band to be processed.
3. Drive level (Drive), used to adjust the depth of the excitation. When the drive level is large, the effect is noisy; when the drive level is small, the effect is mild.
4. Mix ratio (Mix), which is the loudness ratio of the original signal to the effect signal.

Overall planning of effect processing

For the fine processing of the human sound source, it is necessary to use one full digital mixer, at least three digital effects and one digital actuator, and the connection method is as shown in the drawing.

First, on the mixer, use the channel equalization control unit to adjust the tone of the vocals to improve the sound. Several common examples are given here.

(1) The frequency band around 8OOHz can cause some kind of annoyance, so it can be attenuated by up to 15dB in this frequency band, and the bandwidth is 1/5 octave, which is used to improve the total impression of vocal pronunciation;
(2) The frequency band around 68O0Hz can make the human voice produce a screaming and harsh feeling. It can be attenuated by up to 10dB in this frequency band, and the frequency band is l/5 octave, which is used to reduce the squeaky sensation of the tooth sound;
(3) For those who are over-sounding and have the feeling of frying ear stick, the maximum attenuation can be 8dB at 3400Hz, and the bandwidth is 1/3 octave;
(4) For those with too much nasal sound, it can be properly attenuated in the frequency band below 500 Hz, and the attenuation bandwidth is 3 times octave;
(5) The ultra-high frequency band of the tooth sound is affected by the sensitivity of the human ear, and it needs to be improved by 6dB at 12KHz (the frequency band is 2 octave), and its loudness can be balanced with the vocal music.

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

After the balance is adjusted, adjust the exciter. Firstly, the driving level and mixing level of the exciter are adjusted to the maximum state, and the frequency tuning is placed at 2500 Hz. If the pronunciation is already noisy or the tone is too hard, the driving level can be lowered, and it should be noted that this adjustment has What is changing is the hardness of the source. If the drive level is adjusted to a higher position and the mix level is only lowered, the high-humidity sound remains the same, but it is slightly masked by the unstimulated original sound. This phenomenon is obvious when the depth of excitation is very strong. The former one gives a sense of sound to the original sound, and the latter produces two layers of sound, which has the effect of increasing the vocal layering.

Generally, one exciter can only process one frequency band, and many single-function exciter connections are required to be in parallel and can only be connected in series. If you want to add excitation to multiple frequency bands of the sound source, it is recommended that in the device connection shown in the figure, the reverberator should use multiple effects devices with excitation processing (such as YAMAHA SPX990), then you can use the actuator to process 500Hz, In the 800 Hz and 7200 Hz bands, the 2500 Hz band is processed using the excitation function on the reverberator.

Once again, everyone is reminded that the adjustment time of incentive treatment should not be too long, so as to avoid the possibility of accurately identifying the degree of incentives after human fatigue.

The final step is to adjust the reverb effect. The reverb effect here has two aspects, one is the basic retouching and the other is strong dyeing.

The basic retouching of the reverberation process is mainly to increase the fusion of the sound source, but it does not make people feel the room reverberation. The strong dyeing effect of the reverberation process here is mainly used to generate reverberation and rendering performance for the sound source, and the processing method has the following three situations:

(1) Generate a sense of space. Use the hall or room reverb effect. The natural reverberation effect of the simulated residual sound is a simple and effective way of reverberation processing. The frequency band around 3500 Hz on the effect channel is slightly improved, and a high-brightness sound with good penetration can be generated. Of course, there is also a disadvantage that the effect of the treatment is rather turbid, sometimes with a "squeaky can" sound.
(2) Generate an echo. Long delay time delay feedback processing can simulate the valley echo effect; the processing delay time is generally in line with the rhythm of the singing song. In order to make the effect more distant, it can be moderately attenuated in the frequency band below 1600 Hz and above 3800 Hz. Simulating valley echo effects, there are ready-made programs available on many digital effects processors.
(3) Generate a blended sound background. The reverberation effect of the lingering sound is very effective for the beautification of the human voice source, and almost all vocal performances use reverberation. Under the premise of not causing the pronunciation to become sloppy, or causing the sound of "squeaky can", we think that the reverberation effect is as strong as possible, but in practice, when the reverberation effect is still very weak, the pronunciation has become ambiguous and causes obvious The "squeaky can" sound.

In order to produce a sound background without causing the pronunciation to become sloppy, or causing a "squeaky" sound. The following effect processing methods are recommended, namely, delay-reverberation serial processing. The delay time for this type of processing is generally 200-600ms, the feedback gain is 40%-60%, the reverb uses the hall reverb effect, and the reverberation time is 2-8s. The reverberation effect after serial processing requires smooth and consistent. 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 loudness of the reverberation, and the third is to increase the reverberation time.

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

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