Audio knowledge Summary

CoolEdit basic recording tutorial (ultra-complete)
I have been paying attention to the original sound for many days. It's really a good place. Everyone is eager to show their talents, but it makes me a little lost.

I hope there are too few technical posts here. No one will teach you how to record songs. Four years ago, I went crazy.

I have collected a lot of information on the Internet, and I have summarized some information. I 've been using my computer for many years.

Help me. I will share some content with you today.

From cooledit to k8 (I am mainly engaged with these two, and there are also a lot of auxiliary software later), I think it is more professional

Cooledit. Theoretically, the effect of the recording studio is cooledit's hardware.

It means that cooledit can achieve the effect of the recording studio. Of course, it is not easy to achieve this. The following describes in detail.

Click.
(Most of the content below comes from others' painstaking efforts. I just collect and sort out the content. If there is any infringement, please inform me)
From the very beginning, the necessary theory should be a bit more! Let's talk about the audio preliminary.

Audio preliminary (1)

Audio: audio in English. You may see the audio output or input port on the video recorder or vcd backplane. In this way, we

Audio can be easily interpreted. As long as it is an audible sound, it can be transmitted as an audio signal. Related music

Because the physical attribute of frequency is too professional, please refer to other materials.

Now we know that computers can also Process audio, which is different from video recorders or vcd because they do not have large-capacity storage devices,

Therefore, you can only transmit audio signals, but cannot save and process them. The computer has a hard disk and can store large volumes of information.

They can record any external sound to a computer, save it in wav format, and then use various audio processing software

Processing, and recording again, then you will hear the effect is different from the original. Like sound forge, cool

Software such as edit and acid are powerful tools in this area.

In fact, computer-based audio processing is inseparable from the rapid development of computer hardware. A few years ago, I used computers to Process audio.

It's a fantasy. A one-minute wave file requires 4-5 MB of hard disk space. Based on the 80386 and several hundred megabytes of hard disks

Audio may be processed. However, the rapid development of computer hardware and the emergence of large-capacity hard disks have changed everything. Some specialties

Audio Software has been released in the industry. In the past, it was impossible to do the work at home. Now you can do it at home.

For example, you have the accompaniment of a favorite song (not karaoke, everyone knows the sound quality of karaoke), tape

Or cd. You really want to sing it, but it cannot be sung. Do you want to downgrade it? (Sorry! Not karaoke !) No.

, You just need to record it to the computer (for example, recording it with sound forge, and try again later), and drop it through the software.

Call. Can you do this at home a few years ago? Do you know how much it will cost to use other machines?

? After downgrading, I want to record a voice for myself. Have you ever used a recorder to record it for yourself? The effect is unbearable, right !?

Now you can directly record your voice to your computer, add the effect, and then output it. A little noisy? No, Noise Reduction

The software removes it, and your home is simply a small studio! This is the case.

Audio preliminary (II)
The following describes the basic attributes of audio.

Everyone acknowledges that it is a digital age, and many people make unremitting efforts to pursue excellent sound quality. With the advent of the digital age

Everyone acknowledges that digital audio is superior to analog audio. What is a simulated signal? In fact, any sound we can hear

Transmission through audio channels or microphones is a series of analog signals. We can hear the analog signal. While digital signals

It is to use a bunch of digital marks to record the sound, rather than using physical means to save the signal. (Recording with ordinary tape is

Physical means) digital signals that we cannot hear.

In this way, we can briefly compare the differences between the recording production in the simulation age and the digital age: the Simulation Age refers to the original

Number is recorded physically on the tape (of course completed in the recording studio), then processed, cut, modified, and finally recorded

To the carrier that the audience, such as tape and lp, can appreciate. This series of processes are all simulated, and each step will suffer some losses.

The signal is naturally far behind the audience, let alone hi-fi. The digital age is the first step to get the original letter

Digital audio data is recorded, and then processed using hardware or software. This process is superior to the simulation method.

Because it has almost no loss. For machines, it's just a matter of processing numbers. Of course, it's possible to lose the code.

But it will not happen if the operation is reasonable. Finally, this pile of digital signals are transmitted to digital recording devices, such as cd, causing loss.

The consumption is naturally much smaller!

If you pay attention to the cdns around you, you will see many cdns, such as: add, aad, and ddd. Three letter generations

The method used in the recording, editing, and finished table is analog or digital.

. Of course, a represents a simulation, and d represents a number. Aad indicates that the recording and editing are simulated, and the number of recording is used.

Most of these records convert the recorded music into a CD file without any modification. Add is a modified

Cheng, many classical music masters play or direct the recordings in the simulation age. Now we hear the cd is a bucket after modification.

Many such records are marked with "add. The ddd recording must be a modern recording product. Naturally, the CD must be d

The end, and the tape can be regarded as aaa, although it does not seem like this.

Therefore, digital audio is a way for us to store sound signals and transmit sound signals. It features that the signal is not easy.

Loss. The analog signal is the last thing we can hear. However, the modification of analog signals is a disaster, causing losses.

It's too big. Is there a rare grain? If it were not, it would be stunned. And digital audio won't be copied for 100 times.

There is a loss. Do you believe you can copy a wave file?

Audio preliminary (III)
The most important step in digital recording is to convert analog signals to digital ones. For the computer, analog sound signals are recorded.

To be a wave file, this work can also be done with the recorder that comes with windows, but its function is very limited and cannot be full

We use other professional audio software, such as sound forge. The recorded file is

The wave file describes two main indicators: Sampling Accuracy and bit rate. This is a digital sound

There are two very important concepts in frequency production. Let's take a look at them.

What is sampling accuracy? Because wave is a digital signal, it uses a bunch of numbers to describe the original analog signal, so it wants

After analyzing the original analog signal, we know that all the sounds have their waveforms. The digital signal is the original analog signal.

In the signal waveform, "Fetch points" is performed at intervals, and each point is given a numerical value. This is "sampling ",

Then we can connect all the "points" to describe the analog signal. Obviously, the more points we get in a certain period of time

The more accurate the waveform is, the more precise this scale is called "Sampling Accuracy ". The most common sampling accuracy is

44.1 khz/s. It means 44100 sampling times per second. The reason why this value is used is that after repeated experiments

We found that this sampling accuracy is the most suitable, and lower than this value will have a significant loss, and higher than this value of the person's ears already

It is difficult to distinguish and increases the space occupied by digital audio. We will also use

The sampling accuracy of 48 k or even 96 k. In fact, the difference between the 96 k sampling accuracy and the 44.1k sampling accuracy is definitely not like 44.1k and

The difference of 22 k is so big that the sampling standard of cd we use is 44.1 k. Currently, 44.1k is the most common

Standard, some people think that 96 k will be the trend of the recording industry in the future. Improving sampling accuracy is a good thing, but sometimes I want

Can we really hear the difference between music with 96 k Sampling Accuracy and music with 44.1k sampling accuracy? Regular

Can the audios at home name tell us the difference between them?

Bit Rate is a common term. digital recording generally uses 16 bits, 20 bits, and 24 bits to make music.

Is it "bit "? We know that the physical elements that affect the sound are amplitude. As a digital recording

It also needs to be able to accurately represent the sound of music, so there must be a precise description of the waveform amplitude, "bit" is this

A 16-bit unit refers to dividing the waveform amplitude into 216 or 65536 levels.

When it is divided into a certain level, it can be represented by numbers. The higher the bit rate, the more detailed the sampling accuracy is.

Reflects the minor changes in music. 20 bits can generate 1048576 levels, representing the dynamic music of the symphony.

There is no problem. The term "dynamic" was mentioned just now. It actually refers to the most loud and lightest piece of music.

In comparison, we often say "dynamic range". The unit is db, and the dynamic range is compared with what we use during recording.

The baud rate is closely combined. If we use a very low bit rate, we only have a few levels.

It can be used to describe the strength of the audio. Of course, we cannot hear a large comparison between the strength and weakness. Dynamic Range and Bit Rate

The link is; each time the bit rate is increased by 1 bit, the dynamic range is increased by 6 dB. So if we use a 1-bit recording, then we

The dynamic range is only 6 dB, and such music cannot be heard. The dynamic range is 96 dB in 16 bits. This can satisfy

General Requirements. The dynamic range is 120db for the 20th bits, and the strong symphony can be used freely

The strength of music is more than enough. Fever-level Sound Recorder still uses 24 bits, but like sampling accuracy, it does not compare

20 bits have obvious changes. In theory, 24 bits can achieve a dynamic range of 144 db, but it is actually hard to achieve.

Because any device will inevitably produce noise, it is difficult to achieve the expected effect at least 24 bits at present.
We will continue to go back to cooledit and do not talk about how to install it. Now it is very easy to install cooledit.

It's so complicated to use. The following must be used properly when your computer has microphones and headphones.

How to record.

Cooledit recording method!

Process of creating a self-singing song (five steps in total)

1. Recording the original sound (if you use vcd as the companion sound, you can check the post first)

Recording is the basis for all post-production and processing. If there is a problem in this process, it cannot be remedied by post-production. Therefore,

If there is a major problem with the original recording, record it.
1. Open ce and enter the multi-track interface. Right-click the blank area of audio track 1 and insert the mp3 accompaniment file for the song you want to record. The wav file can also be used.

(Figure 1 ).

2. Choose to record your voice on Track 2 and press the "r" button. (Figure 2)

3. Press the red recording key in the lower left to start singing and recording with the accompaniment music. (Figure 3)

4. After the recording is completed, you can click the broadcasting key in the lower-left corner to test whether serious errors occur and whether to record the recordings (figure 4)
5. Double-click audio track 2 to go To the waveform editing page (figure 5) and save the original voice file you recorded as mp3pro (figure 6

Figure 7). In the previous introduction, we saved everyone in wav format. In fact, mp3 is also absolutely acceptable, and it can save a lot of space.

.

Note: When recording, you must turn off the speaker, use headphones to listen to the accompaniment, and follow the accompaniment for singing and

Before recording, you must adjust your total volume and microphone volume. This is important! Mike's volume is better off

If the volume exceeds the total volume, it is better to be slightly smaller, because if the microphone volume is too large, the recorded waveform will become a square wave.

The sound of a waveform is distorted, and such a waveform is useless. No matter how high your level is, it is impossible to handle it.

Satisfactory results.
In addition, if your microphone always enters the sound of accompaniment music from the headset, we recommend that you use an ordinary big microphone

You need to add a large-to-small connector to use it directly on your computer. You will find that the recorded effect is much cleaner.
Ii. Noise Reduction

Noise reduction is a crucial step. Doing well is conducive to further beautifying your voice. Otherwise, sound distortion will occur.

Completely destroys the original sound. This step alone is enough to provide a dedicated explanation. You can understand this.
1. Click the horizontal zoom-in button in the lower left of the waveform (the two with the plus sign are horizontal zoom-in and vertical zoom-In respectively) to zoom in the waveform,

To find a suitable waveform for noise sampling (fig. 8 ).

2. Click the left mouse button and drag until the highlighted area completely overwrites the selected waveform (figure 9 ).

3. Right-click the highlighted area and select "copy as new" to extract the waveform (Figure 10 ).

4. Enable "effect-noise elimination-noise reduction" to prepare for noise sampling (Figure 10 ).
5. Perform noise sampling.
The parameters in the noise reduction device can be changed by default values, which may cause great distortion of voice after noise reduction (

Figure 11)

6. Save the sample result (Figure 12)

7. Turn off the noise reduction device and the waveform (no need to save ).

8. Return to the voice file on the waveform editing interface, open the noise reduction device, and load the previously saved noise samples for noise reduction.

Before determining the noise reduction, You can preview the noise reduction effect (if the distortion is too large, it indicates that the noise reduction sampling is not suitable ).

, Need to re-sample or adjust parameters, one thing to note, no matter which method of noise reduction will have a certain damage to the original sound)

(Figure 13 Figure 14 figure 15)

Iii. tweeter incentive handling

1. Click "effect -- directx -- bbe sonic maximizer" to open the bbe tweeter (figure 16 ).

2. After loading various effects from the preset drop-down menu (or manually adjusting the three knob), click "preview" in the lower right of the actuator"

Repeat the audio repeatedly until the desired effect is reached, and then click "OK" to motivate the sound. (Figure 17, figure 18)

(Note) The purpose of this process is to adjust the tweeting and bass of the recorded voice to make the sound clearer, brighter, or heavier.

. The incentive function is to generate harmonic waves, modify and beautify the sound, and produce a pleasant auditory effect, which can enhance the sound.

The audio frequency is dynamic, which increases the definition, brightness, volume, warmth, and warmth to make the sound more tense.

Iv. pressure limit handling

1. Click "effect -- directx -- waves -- c4" to open the wave filter (Fig. 19 ).

2. Load various effects from the preset drop-down menu (you can also adjust them manually if you have enough knowledge about digital audio)

Click "preview" in the lower-right corner to review the audio repeatedly. After the desired effect is adjusted, click "OK" to apply pressure limit to the original sound.

(Figure 20 ).

(Note) the purpose of the pressure limit is to adjust the overall balance of your recorded voice, without being too large or small,

High and low.
 
V. reverb Processing

1. Click "effect -- directx -- ultrafunk fx -- reverb r3" to open the reverb effector (Figure 21 ).

 

2. After loading various effects from the preset drop-down menu (you can also manually adjust them), click "preview" in the lower right corner to review the results.

Until the desired reverb effect is adjusted, click "OK" to process the original sound. The common effect is shown in Figure 22.

Figure 23 ).

 

After the reverb processing, you can make your voice look less dry, smooth and heavy.

At this point, the processing of voice is all over.

6. Hybrid reduction and Synthesis

1. Click Edit-merge to zoom in to file-all waveforms to merge accompaniment and processed voice and voice (figure 24)

).

2. Click "file -- save as" to save the merged file in mp3pro format (figure 25 )!!

Your masterpiece is complete! Some tips can improve the effect of the above steps.
 
Cool edit pressure limit processing

The pressure limit is a volume adjustment button. When your voice is too loud, you can turn it off a little and give it to you when your voice is too small.

A little higher means that your volume is always on the average line.

Go to the Boeing editing page and choose directx -- waves -- c4.

 

The vertical direction is the volume value, which is in the normal range within the range of plus and minus 6 dB (within the purple range). If the range is exceeded

. Then you can see that there are four regions horizontally separated, which is the essence of c4-segmentation compression. In fact,

After c4, I have seldom balanced people's voices. static balancing is always suitable here and not suitable.

Yes, it is very troublesome to adjust it. The c4 balance and pressure limit are dynamic and closely linked. below

I will focus on the benefits of Dynamic Balancing and combination of pressure limit.

Assume that your work contains four instruments: Bass, guitar, drum, and baseline plus your voice. In terms of frequency band, bass

And the base drum in the low-frequency section, the high-position guitar and high-tone band strings in the high-frequency band, and then with his central audio area, Bass panic

The string's central audio area and your voice are all in the central audio area. This is just our division. In fact, when every instrument is mixed together

The signal is not clearly divided, and a group of voices are often overlapped in the areas of the frequency band.

It doesn't sound so obvious. It's all fun together, so I won't be able to hear Beth for a while, and then I'm done again. C4

The function is to divide the effect into different frequency bands and clarify the sound of each frequency band through dynamic pressure limits and balancing. For example

Said Beth, through processing, it will honestly occupy the low frequency, the sound exceeding the low frequency will be squashed out, so which is

The voice in the frequency band is in which band, so everyone is safe and everyone's voice is clearly identifiable. So I usually try to shrink at the end.

Use c4 for processing (it should be sorted.

See the following figure.

 

This is also a predefined value of c4 called pop vocal. Do you have to check whether it is different from the above standard voice processing? (Image

As I mentioned earlier, the pop voice often adds a lot of reverb sound, so in order to be clear, it is generally necessary to increase the high frequency by a little,

Because in the voice, the highest sound direction is the strongest, and the lowest sound is the worst .) Look at the Purple area! Upgrade 1 in 3 K

Until 16 k, this is the main part of the voice of the average person (also the clearest and most nice part) if your voice is male

You can adjust the gray card in the 4 K area manually, and then adjust the balance accordingly,

Generally, there is no fixed preset value. Because of the different style of the Song and category, the adjusted value should also be different.

From the mastering point of view, the whole music will be full only when the sound is evenly distributed to various frequencies.

The job is to allocate the sound of where to go, so don't let him run around.

In addition, the parameters of the reverberation are ever-changing. The parameters are satisfactory to me. View

 
 
Reverb parameter!

Reverb hall 1 simulates the reverb effect of a large concert hall
Parameter value range description
Rev. time2.8s0.3-30.0s reverb time
High ratio0.80.1-1.0 high-frequency attenuation rate
Diffusion60-10 reverb Diffusion
Ini. dly40.0ms0.1-200.0ms latency between direct and early reflection sound
Cut-off frequency of lpf7.0khz1khz-16 khz and thru low-pass filter
Cut-off frequency of hpfthruthru, 32hz-8khz high-pass filter
Reverb hall 2 simulates the variant of the large concert hall.
Parameter value range description
Rev. time3.2s0.3-30.0s reverb time
High ratio0.70.1-1.0 high-frequency attenuation rate
Diffusion80-10 reverb Diffusion
Delay Time between ini. dly38ms0.1-200.0ms direct and early reflection sound
Lpf6.3khz1.0khz-16.0 khz, cut-off frequency of thru low-pass filter
Cut-off frequency of hpfthruthru, 32hz-8khz high-pass filter

Reverb room 1 simulates a great deal of ECHO in the cement wall room * adds a sense of presence to the drum tone
Parameter value range description
Rev. time1.4s0.3-30s reverb time
High ratio0.80.1-1.0 high-frequency attenuation rate
Diffusion70-10 reverb Diffusion
Delay Time between ini. dly5.0ms0.1-MS direct and early reflection sound
Cut-off frequency of lpfthru1khz-16khz and thru low-pass filter
Cut-off frequency of hpfthruthru, 32hz-8khz high-pass filter

Reverb room 2 room1 Variant
Parameter value range description
Rev. time1.8s0.3-30s reverb time
High ratio0.60.1-1.0 high-frequency attenuation rate
Diffusion60-10 reverb Diffusion
Delay Time between ini. dly17ms0.1-MS direct sound and early reflection sound
Cut-off frequency of lpf9khz1khz-16khz and thru low-pass filter
Cut-off frequency of hpf80hzthru and 32hz-8khz high-pass filter

Reverb stage is similar to program 1, but it is brighter and has a sense of presence.
Parameter value range description
Rev. time3.4s0.3-30s reverb time
High ratio0.90.1-1.0 high-frequency attenuation rate
Diffusion80-10 reverb Diffusion
Delay Time between ini. dly45ms0.1-MS direct sound and early reflection sound
Cut-off frequency of lpfthru1khz-16khz and thru low-pass filter
Cut-off frequency of hpf70hzthru and 32hz-8khz high-pass filter

Reverb plate is a steel disc-type reverb system with wide adaptability, especially human voice, drum and percussion.
Parameter value range description
Rev. time2.4s0.3-30s reverb time
High ratio0.70.1-1.0 high-frequency attenuation rate
Diffusion80-10 reverb Diffusion
Delay Time between ini. dly16ms0.1-MS direct and early reflection sound
Cut-off frequency of lpf8khz1khz-16khz and thru low-pass filter
Cut-off frequency of hpfthruthru, 32hz-8khz high-pass filter

Rev ambience 1 simulates the reverberation around the instrument, used for voice, chorus, and percussion
Parameter value range description
Rev. time1.2s0.3-30s reverb time
High ratio10.1-1.0 high-frequency attenuation rate
Diffusion80-10 reverb Diffusion
Delay Time between ini. dly19ms0.1-MS direct and early reflection sound
Cut-off frequency of lpf9khz1khz-16khz and thru low-pass filter
Cut-off frequency of hpf45hzthru and 32hz-8khz high-pass filter

Rev ambience 2 variants of program 7
Parameter value range description
Rev. time0.8s0.3-30s reverb time
High ratio0.60.1-1.0 high-frequency attenuation rate
Diffusion80-10 reverb Diffusion
Delay Time between ini. dly0.1ms0.1-MS direct and early reflection sound
Cut-off frequency of lpfthru1khz-16khz and thru low-pass filter
Cut-off frequency of hpf56hzthru and 32hz-8khz high-pass filter

Rev live room 1 simulates the reverb effect of the on-site room, which is stronger than the recerm room.
Parameter value range description
Rev. time2.4s0.3-30s reverb time
High ratio0.80.1-1.0 high-frequency attenuation rate
Diffusion70-10 reverb Diffusion
Delay Time between ini. dly0.1ms0.1-MS direct and early reflection sound
Cut-off frequency of lpf7khz1khz-16khz and thru low-pass filter
Cut-off frequency of hpfthruthru, 32hz-8khz high-pass filter

Rev live room 2 variant of program 9
Parameter value range description
Rev. time2.2s0.3-30.0s reverb time
High ratio0.50.1-1.0 high-frequency attenuation rate
Diffusion60-10 reverb Diffusion
The delay time between ini. dly12.0ms0.1-200.0ms and early reflection.
Cut-off frequency of lpf4.0khz1.0khz-16108khz.thru low-pass filter
Cut-off frequency of hpfthruthru, 32hz-8khz high-pass filter

Reverb vocal used for voice and chorus reverb
Parameter value range description
Rev. time1.9s0.3-30.0s reverb time
High ratio0.50.1-1.0 high-frequency attenuation rate
Diffusion60-10 reverb Diffusion
Delay Time between ini. dly16ms0.1-200.0ms direct and early reflection sound
Cut-off frequency of lpf12khz1.0khz-16108khz.thru low-pass filter
End frequency of hpf100hzthru and 32hz-8khz high-pass filter

-------------------------------------------------------------------------------

Chorus reverb
Parameter value range description
Mod. freq0.8hz0.1-20Hz Modulation Speed
Mod. depth40 % 0-100% modulation depth
Mod. dly1.3ms0-24ms delay time before initiation of modulation
Rev. time2.4s0.3-30s reverb time
High ratio0.70.1-1 high-frequency attenuation rate
Diffusion70-10 reverb Diffusion
Delay Time between ini. dly30ms0.1-139ms and early reflection sound
Lpf6.3khz1khz -- 16 khz, the cutoff frequency of the thru low-pass filter
Hpfthruthru, 32 hz -- cut-off frequency of 8 kHz high-pass filter
Rev. depth24 % 0-100% reverb depth

Flange reverb
Parameter value range description
Mod. freq1.4hz0.1-20Hz Modulation Speed
Mod. depth22 % 0-100% modulation depth
Fb gain + 45%-99 -- + 99% after processing, the signal returns erratic gain
Mod. Delay Time Before dly13ms0-15.5ms start Modulation
Rev. time2.4s0.3-30s reverb time
Diffusion80-10 reverb Diffusion
Delay Time between ini. dly26ms0.1-MS direct sound and early reflection sound
Lpf4.5khz1khz -- 16 khz, the cutoff frequency of the thru low-pass filter
Hpf45hzthru, 32 hz -- cut-off frequency of 8 kHz high-pass filter
Rev. depth30 % 0-100% reverb depth

Delay l-c-r independent delay in left, middle, and right Audio Channels
Parameter value range description
Dly l250ms0.1-661ms left channel delay time
Dly r500ms0.1-661ms right channel delay time
Dly c125ms0.1-661ms central channel delay time
Level c700-100 central channel latency volume
Fb. dly500ms0.1-661ms latency before feedback starts
Fb. gain + 40%-99 -- + 99% signal after processing returns the gain of latency
High ratio0.80.1-1 frequency modulation attenuation rate

Monodly-chorus single-channel delay followed by stereo chorus
Parameter value range description
Delay400ms0.1-618ms Delay Time
Fb. gain + 32%-99 -- + 99% signal after processing returns the gain of latency
High ratio0.80.1-1 frequency modulation attenuation rate
Mod. freq0.4hz0.1-20Hz chorus Modulation Speed
Mod. depth10 % 0-100% chorus modulation depth
Mod. dly0.1ms0-24 Ms latency before the chorus starts Modulation

Chrous-dly l c r stereo chorus followed by left, center, right audio channel independent latency
Parameter value range description
Mod. freq0.8hz0.1-20Hz Modulation Speed
Mod. depth24 % 0-100% modulation depth
Mod. dly5.9ms0-24ms delay time before initiation of modulation
Dly l26.4ms0.1-618ms left channel delay time
Dly r33.2ms0.1-618ms right channel delay time
Dly c13.1ms0.1-618ms central channel delay time
Level c600-100 central channel latency volume
Fb. dly40.5ms0.1-618ms latency before feedback starts
Fb. gain-48%-99 -- + 99% signal after processing returns the gain of latency
High ratio0.10.1-1 frequency modulation attenuation rate

Delay-chrous two-level latency followed by stereo chorus
Parameter value range description
Dly 1250ms0. 1-618ms1 Delay Time
Dly 2500ms0. 1-618ms2 Delay Time
Fb. dly500ms0.1-618ms latency before feedback starts
Fb. gain + 33%-99 -- + 99% signal after processing returns the gain of latency
High ratio0.70.1-1 frequency modulation attenuation rate
Mod. freq1.2hz0.1-20Hz Modulation Speed
Mod. depth25 % 0-100% modulation depth
Mod. Delay Time Before dly10ms0-24ms start Modulation

Karaoke echo 1 karaoke Effect
Parameter value range description
Dly l220ms0.1-332ms left channel delay time
Fb. gain l + 40%-99 -- + 99% gain returned by signals processed by left channels
Dly r223ms0.1-332ms right channel delay time
Fb. gain r + 40%-99 -- + 99% gain returned by signal after right channel processing
High ratio0.40.1-1 frequency modulation attenuation rate

Karaoke echo 2 karaoke Effect
Parameter value range description
Dly l220ms0.1-332ms left channel delay time
Fb. gain l + 44%-99 -- + 99% gain returned by signals processed by left channels
Dly r180ms0.1-332ms right channel delay time
Fb. gain r-55 %-99 -- + 99% right channel signal return gain after processing
High ratio0.20.1-1 frequency modulation attenuation rate