knowledge share on my vocoding

emerald tablet

New member
a technique for all you vocoders out there
vocoded voices sound often static and to robotic ... use the following technique to make it more human

1: sing the part you want to hear vocoded in Goldwave (record to wave) or use any other wave editing program

2: use wave goodbuy (a free wav to midi tool) and make a simple monophonic midi-1 file (or use another wave 2midi like program such as inteliscore, Digital Ear or Audio Works-Sound2Midi AKoff Music Composer etc.. etc

4: open your vsti host and select the sound you want to use as carier.

5. use that sound on your midi file and render the result to a wave file

6: select that wave file as carier in anolog-x vocoder

7: select the wavefile of step 1 as voice

you can also place the two audiotracks on 2 channels in your vstihost .. pan wave one to the left and wave two to the right.
put them on the same effect channel and open a real time vst-e vocoder such as : Evoc20 / Fusion vocoder / vocodomatic / orange vocoder / steinberg vocoder / delaydots spectral morpher / vocoder

8. render the stuf.


cheers

remco
 
Do you ever use hardware vocoders? My ms2000 has nice vocoder capability. But, yeah, you normally sound like a robot or the devil. But the real time physical controls are worth it.
 
yes i do

i have vocoding possibilities in my kurzweil k2500s
and i have a harmonix golden throat clone

not my intention to sound like good old satan :)
 
So, I pm'd you about the k2500 from another post... I'd be interested in learning some key pointers on all that if you've got the links or whatever. Like the onscreen feature you spoke of...

Check your private messages in the user op feature up top.
 
Hey there man

it is a nifty little program that is called Kurzsee
i will look up where i got it from when i get home tonight (at the office right now0
It works via your standard midi cable
do you have a k2500 rack version or key version ?

on the the vocoding part on your kurzweil
you need to have KDFS on your kurzweil (which is not standard)
and the latest kurzweil firmware
to my knowledge the k2500s with latest firmware is exactly the same as the k2600

i also use a program called krz2wave
and wave2krz .. i guess i don`t have to explain what they do :)


Remco
 
http://www.kurzweilmusicsystems.com/html/alternate_tunings.html

for detuning your kurzweil

for the vocoder in the k2500s check out the folowing

This feature requires v v4.21b or later of the operating system.

Vocoding is a special feature which allows you to speak into a microphone and use that signal to control another audio signal. Typically you would use the output from a synthesizer, although in fact you can use any sound source

The first part of this document describes setting up the K2500 for vocoding. You can either use the K2500 itself to generate the synthesis sound to feed into the vocoder, or you can use an external source, such as another keyboard or rack. You can even use audio output from a CD or tape.

The second part of this document describes the theory of vocoding and how the special vocoding programs work.




SETTING UP AND USING THE VOCODER FEATURE:


I. HARDWARE SET UP.


A. Using the K2500 for both vocoding and slave source signal:

1: You must have an insert cable (Y cord) with a 1/4" STEREO (Tip/Ring/Sleeve) jack on one end and 2 mono jacks on the other end. The right side mono jack should be 1/4". The left side should be a female 1/4" or female XLR. (You will be plugging a Mic into the left side, so if the insert cable has a 1/4" jack, you will need an adapter from XLR to 1/4".)

2: Plug the stereo side of the insert cable into the 1/4" stereo Sample Input.

3: Connect the LEFT mono jack of the insert cable to a Microphone. It must be the LEFT input only!!!!

4: Connect the RIGHT mono jack of the Insert cable to the B RIGHT separate output on the K2500.


B. Using the K2500 for vocoding with an external sound source for the slave signal:

Method One

1: You must have an insert cable (Y cord) with a 1/4" STEREO (Tip/Ring/Sleeve) jack on one end and 2 mono jacks on the other end. The right side mono jack should be a 1/4" plug. The left side should be a female 1/4" or female XLR. (You will be plugging a Mic into the left side, so if the insert cable has a 1/4" jack, you will need an adapter from XLR to 1/4".)

2: Plug the TRS plug into the Stereo Analog Input of the Sampler

3: Connect the LEFT mono jack of the insert cable to a Microphone. It must be the LEFT input only!!!!

4. Connect the RIGHT mono jack of the Insert cable to your external sound source (most likely, an external synthesizer).

Method Two

1: Connect a Mic into LEFT SIDE low impedance input (XLR) of the sampler

2. Connect your external sound source to the RIGHT SIDE low impedance input (XLR) of the sampler.


C. Final Audio Output:

You must have audio cables connected from either the Mix Outs or A Separate Outs to your mixer or amp.

D. MIDI

If you have a K2500 keyboard, and your external slave is a rack (or it is a keyboard but you want to use the K2500's keyboard to control the slave), then you must of course have a MIDI cable going from the MIDI Out of the K2500 to the MIDI In of the slave.


II. SETTING UP THE K2500

1: Go to the sample page

2: Set Input to Analog

3: Set Source (Src) to External (Ext)

4: Set Mode to LiveIn

5: Verify that mic signal is on the left side only. Adjust the Gain parameter as needed, to get a good signal level.

6: Verify that your sound source (either the K2500 or external source) is on the right side only.

7. Go the Effects Mode page and make sure that FX Mode is set to Auto and FX Chan is set to Current.


III. ENABLING THE VOCODER MODE

1: Load the file VOCODER.K25 into any bank.

2: Press the Master Button.

3: Press Mast2 soft button.

4: Turn the Vocoder On..

5: Exit all the way out of Master mode.

IMPORTANT! Enabling the Vocoder mode loads special micro code into the second Hobbes chip (the two Hobbes chips are the chips that do VAST.) This replaces the code that is used for the SHAPE2 and AMP MOD OSC functions in the F3 block of an algorithm. Therefore any programs you have which use SHAPE2 and AMP MOD OSC in the F3 block will no longer play correctly! Turning the Vocoder parameter Off will restore those DSP functions and disable vocoding.


IV. USING THE VOCODER

1: Go to Setup Mode and select one of the Setups in the bank you just loaded the file into. If you are using an external sound source for your slave, choose the Setup Vocoder-ExtSlave. If you are using the K2500 as the source for the slave, then you can choose either Vocoder-22 Band or Vocoder-20 Band. The 22 band Vocoder will allow you to play up to 4 voices of polyphony on the slave program; the 20 band Vocoder will allow you to play up to 8 voices of polyphony on the slave program.

2. Play a note or chord on your keyboard and speak into the microphone. You should be able to hear what you are speaking, but the sound will be a string sound (assuming you are using the K2500 as the slave source), pitched to the note or chord you are playing.

3: If you have a K2500 keyboard, try moving Sliders A, B, & C and listen for changes in the sound. If you have a K2500 rack, you can send MIDI controller numbers 6, 12, & 13 from your keyboard. Since the Setups contain Entry values for these 3 Sliders, you may have to move the slider across the full range of travel before it will begin to take effect.


V. EFFECTS/OUTPUT ISSUES FOR THE VOCODER

The Studio assigned to the Vocoder Setups is configured in the following manner: If you are using the K2500 for the slave source signal, the Slave program (in zone 3) has its output assigned to KDFX-B, which is being routed to the FXBus2, with No Effect. Then on the OUTPUT page, Output B is set to FXBus2, thereby sending the signal from the slave program to the B outputs and from B Right into the right side of the sample input. The Slave program has its output panned hard right within the program, so if you decide to try using a different slave program, you will probably want to edit the program itself to pan its output hard right, so you get 100% of the signal. You don't need to worry about setting the output pair within the program, because the Out parameter on the CH/PRG page of the Setup Editor, is set to KDFX-B in zone 3, thereby overriding any settings from within the program.

Of course, all of the previous paragraph is not important if you are using an external slave.

The Vocoder programs themselves are assigned to KDFX-A, which is being routed to FXBus1, with #3 Natural Room. Then on the OUTPUT page, Output A is set to Mix. So the final output of the vocoder programs is run through the effect and then comes out either A Outs or the Mix Outs (depending on where you have plugged in the cables).

If you choose to change the effects, you may find it easier to edit the our Vocoder Studio, and try changing the effects assigned to FXBus1, FXBus2, and AuxFX. But if you want to change to a different studio, you will need to make sure the following parameters are set correctly - on the FXBUS page, for FXBus2, set the Level parameters for both Aux and Mix to Off, and on the OUTPUT page, set Output B to FXBus2.



HOW VOCODING WORKS IN THE K2500


A vocoder is a device which analyzes the time-varying spectrum of one signal (the "Master") and imposes that spectrum as a filter on a second signal (the "Slave".) The method we use is an emulation of the traditional analog technique involving banks of bandpass filters and envelope followers.

The Master signal will be what you send from the microphone, and the Slave signal will be what you send from an external synthesizer or other sound source, or a program from the K2500.

The Master signal is sent to a number of bandpass filters in parallel. The center frequencies are spaced so as to cover the frequency range we are most interested in. The lowest frequency filter is a low pass rather than a bandpass, so as to group all low frequency components together. Likewise, the highest filter is a high pass. The outputs of all these bandpass filters go into individual envelope followers, which detect the level of signal present in each band. The output of the envelope follower is then used as a control for the Slave signal.

The Slave signal is also sent to the same number of bandpass filters. These will generally have the same center frequencies as the master bandpasses. The output signals from the slave bandpasses are multiplied, one by one, by the outputs of the envelope followers (from the Master signal). The resultant products are all summed together for the final output.

Since each band requires two layers (one for Master and one for Slave), the largest number of bands you can have for vocoding is 24. (24*2=48, which is your maximum polyphony.) The programs in the Vocoder-ExtSlave Setup use 24 bands. If you want to use the K2500 to generate your slave signal, then you have to use either the 22 or 20 Band Vocoder Setups, which have fewer bands, and therefore leave 4 or 8 voices of polyphony available for the Slave signal program.

Since 48 (or 44 or 40) layers are used, and a drum program has a maximum of 32 layers, we use two 24 (or 22 or 20) layer programs, on different MIDI channels, that are combined in a setup.

Each of the Setups has 3 zones. In the 22 and 20 Band Vocoder Setups, the first two zones are used for the vocoding programs and the third zone plays the internal program that is used for the slave signal. In the Vocoder-ExtSlave Setup, the third zone is set to transmit out via MIDI only, on channel 1. (This allows you to play your external sound source, if it is a rack, but won't play a K2500 internal program.)

Layers are grouped in pairs, with the master signal going to the first layer, and the slave to the second. All odd numbered layers are Master and all even numbered layers are Slave. If you look at the algorithms in the vocoding programs, you will see that the first two DSP blocks (after PITCH) of each layer are a bandpass filter (or low pass or hi pass filters for the first and last bands). The first layer then has a DSP called MASTER, while the second layer has a DSP called SLAVE. These Stages are then followed by an AMP stage. These DSP blocks perform the function of an envelope follower and gain multiplication.

The signal flows from the odd numbered (Master) layer to its associated even numbered (Slave) layer (for example, from layer 1 to 2), which is something that does not happen in other algorithms. The low pass frequencies controlled by the third time-slot for each layer set the response speed of the envelope follower. They are normally set to the same frequency. The Master layer controls the frequency of one pole of low pass filtering, and the Slave layer controls two more poles.

The AMP page on the Master layer does nothing. There is no output from this layer, so any settings on the OUTPUT page don't matter. The Slave layer's AMP page does do an actual amplitude control. The output pages for Slave layers are active, and can be used to choose the output group and set the step panning.

All of the Master layers use the LiveIn Left keymap and all of the Slave layers use the LiveIn Right keymap. That is why you must plug the microphone into the left side of the sample input and the slave source into the right side.

As is always the case with Live Mode, a note message is required in order for an incoming signal to be processed through VAST. Therefore, the two layers in the Setup assigned to the vocoding programs have Pswitch2 set to generate a C4 with a velocity of 127, as soon as the Setup is selected. That note remains on until you select a different Setup. The Setups are edited so that none of the notes on an 88 note keyboard are assigned to either of the two vocoding programs

REAL TIME CONTROL OF THE VOCODING PROGRAMS

The most important control parameter is the envelope follower speed, set by the third time-slot low pass parameters. These are set to C6 on all the layers for the initial level. Slider A (controller #6) lowers the cutoff up to 8 octaves (-9600 cents). Therefore, the higher you raise the slider, the slower the envelope follower speed. C6, as a filter cutoff, has a time constant on the order of one millisecond. This is generally rather too fast. For best results, this should be lowered about 4 octaves to C2 (half the range of the DATA slider), to a time constant of 16 milliseconds. Too slow and the vocoder will not respond to quick transients, like consonants, and too fast will result in a jittery sort of sound, as the envelopes follow every little fluctuation. At the fastest possible setting, the envelopes follow the master audio signal itself, and an extremely harsh intermodulation is heard between master and slave. The Vocoder Setups have an Entry Value of 64 for this slider, so when the setup is selected it is the equivalent of having the slider halfway up.

Slider B (controller #12) is used to control the width of the band pass filters (for all bands except the lowest and highest). The Vocoder Setups have an Entry Value of 10 for this slider, the equivalent of having the slider at the first dot above the bottom.

Slider C (controller #13) transposes the center frequencies of all the slave bandpasses upward together. It gives you the same result as pitch shifting the master signal up. Vocal formants will be munchkinized as you bring the slider up. The Vocoder Setups have an Entry Value of 0 for this slider, the equivalent of having the slider at the bottom.

SOME ADDITIONAL NOTES & ADVANCED PROGRAMMING SUGGESTIONS

1. The classic application of a vocoder is to make instrumental sounds talk/sing. The slave signal has to have a lot of high frequency content, or the consonants will not be heard clearly. However, there is no rule set in stone that you must speak words into the microphone. Using the vocoder just as a timbral control can be just as interesting. Using one's voice to control a lead line, where one is doing the articulation and filter control by talking, singing along (tunelessly, if that's all you can do; it doesn't really matter too much), or just making various vocal sounds can give very expressive results. You can get some of the same types of results you would by using a breath controller. It's a little like having a 24 band graphic equalizer, but instead of controlling it with your hands, you use... your mouth!

Furthermore, you don't even have to use a microphone as the master. You can send a signal from anything else that has varied timbral content and get interesting results. For example you could use a drum loop or some other recorded sounds which change timbres regularly as your master.

2. The analog sample inputs on the K2500 are line level, not mic level. This means you have to boost the Gain on the sample page to get a good signal. But this also increases the general noise level of the input signal. If you have a mic pre-amp, or plug the mic into a mixing board before sending the signal to the Kurzweil, you can lower the Gain parameter and start with a much cleaner signal. This is highly recommended.

In addition, you will find you get better results if you run the pre-amped mic signal into a compressor before sending it to the Kurzweil. This can also be done for the slave source signal. Using compressors will give you a much more even dynamic result, making it easier to play and control your sound. This is because the dynamic range of the master and slave signals is added together. For example, lets say both the master and slaves signals have a dynamic range of 20dB. The resulting signal will have a dynamic range of 40dB, giving you a very wide range between the softest and loudest signals you can produce.

3. One way to improve intelligibility is to mix in a little of the master signal into the final audio output. This can be done in a couple of ways. If you run the mic into a mixer, you can split the signal, sending it both to the Kurzweil as well as to your final mix.

A second way is to include it in the vocoder program. You can do this by editing one of the programs in the 22 or 20 band vocoder setups. You would want to add a layer to the program (it doesn't matter which one of the two programs you edit). set the Keymap for the layer to LiveIn L and choose algorithm 1 with the DSP function set to NONE. You could then control the amount of the signal by editing the Adjust parameter on the F4 AMP page (or even assign a control source to vary the amount).

You could then try various algorithms and DSP functions to further modify the signal. Running the signal through a high pass DSP to emphasize vocal articulations is one obvious example. Just make sure that you don't use the SHAPE 2 or AMP MOD OSC DSP functions. In that case, the master signal won't be output.

4. If you are using the K2500 as the source for your slave signal, try editing the Slave Vocoder Program. A simple thing to try is to choose a different keymap. The AMPENV in this program has been set to User, with a lengthy Decay, so you can even choose decaying sounds such as guitar, and get interesting results. And of course, you can choose other programs as the slave.

And of course, you should try making some of your own programs to use as a source. Just edit the Setup and change the program in zone 3 to your new program.

5. Some other things to try:.

¥ Use an LFO to modulate the center frequencies of the slave bandpasses, or the master bandpasses.

¥ Try panning alternate bands of the slave layers to L and R to create a "fake stereo" program.

¥ Try different center frequencies from the ones used in the preset programs.

¥ Currently the center frequencies of the slave layers match the master layers. Try scrambling the slave frequencies relative to the master frequencies.

¥ If you are using the K2500 for the slave source and need more polyphony, you can delete some of the layers in the vocoding programs. Make sure to delete matching sets of master and slave layers. You will probably want to readjust the frequencies and widths of the remaining layers accordingly.

6. Even more obscure and advanced applications:

¥ Instead of using a microphone or other external source for your master, you could use the K2500 to generate BOTH the master and slave signals. There are two ways you could set this up. You can either edit the setup to add another program on a 4th zone, or you could edit the slave source program to add more layers. Then split the keyboard so that one side plays the master zone/layers and the other side plays the slave zone/layers. On the OUTPUT page, make sure all the master layers are assigned to B and panned hard left and the slave layers assigned to B and panned hard right. You will then have to alter the wiring setup described at the beginning of this document so that the B Left jack is going to the left side of the stereo sample input.

¥ If you edit width of the Master layers so that they are extremely narrow, and set the frequencies to a specific scale pattern, then if you sing into the microphone, you will only hear sound as you sing the specific pitches in that scale.

¥ If you edit the width of the Slave layers so that they are extremely narrow, then you will get a very pure tonal sound, hearing only very specific pitches depending on the harmonic content of the Master.

¥ Another possibility for using very narrow width Master layers: Edit the Slave layers so that instead of using a series of bandpass filters, each slave layer uses different DSP functions in the F1 and F2 slots (remember that the F3 slot still needs to be set to LPCLIP in order for the vocoding function to work - you can change algorithms as long as the algorithm allows LPCLIP to be selected for the F3 slot). Now, if you sing various pitches, the slave signal will be played through the various corresponding VAST algorithms.

¥ It is actually possible to use samples in RAM (or ROM) instead of the Live Mode In for either the Master or Slave signals (or even both of them). Just change the Keymap parameter on the KEYMAP Page. (Remember that you need to edit the Keymap parameter on all Master and/or Slave layers.) In this case, the keymap would be playing a single held sample, so you will want to use a looped sample. Loops with changing harmonic content will work best. The note used in the setups is C4, so you would want the sample root at C4 to hear it back without transposition. You will need to edit the layers, save the programs, and reselect the Setup before you will hear the change. If both the Master and Slave layers call up samples in the unit, then as soon as you select the Setup, you will hear sound without even touching the keyboard! You might want to assign a slider to the F4 AMP page on the slave layers to control the amount of output. If the Master and Slave layers are loops of slightly different lengths, then you will hear a continually changing sound that could appear to go in indefinitely without changing.

¥ Continuing with the previous suggestion, you could set the Slave layers to different keymaps, each layer assigned to a different sample loop. Edit the DSP functions on the slave layers so that F1 and F2 are set to NONE, or some other DSP function. Set the Master layers to very narrow widths. Now, as your Master signal changes frequencies you will hear different sample loops fading in and out.



Center frequencies (or cutoff for first and last bands) in the 24 Band Vocoder

Part 1
Layer Note Hz Width (shown as % of an octave. 1/12=.083=1 semitone)
1 B2 123 N/A
3 E3 165 .320 oct
5 G#3 208 .320 oct
7 C4 262 .320 oct
9 E4 330 .320 oct
11 G#4 415 .320 oct
13 C5 523 .240 oct
15 D#5 622 .240 oct
17 F#5 740 .240 oct
19 A5 880 .200 oct
21 B5 988 .170 oct
23 C#6 1109 .170 oct

Part 2
Layer Note Hz Width
1 D#6 1245 .170 oct
3 F6 1397 .170 oct
5 G6 1568 .170 oct
7 A6 1760 .170 oct
9 B6 1976 .170 oct
11 C#7 2217 .170 oct
13 D#7 2489 .200 oct
15 F#7 2960 .240 oct
17 A7 3520 .240 oct
19 C8 4186 .240 oct
21 E8 5274 .320 oct
23 A8 7040 N/A

Center frequencies (or cutoff for first and last bands) in the 22 Band Vocoder

Part 1
Layer Note Hz Width
1 B2 123 N/A
3 F3 175 .500 oct
5 B3 247 .420 oct
7 D#4 311 .320 oct
9 G4 392 .320 oct
11 B4 494 .320 oct
13 D#5 622 .320 oct
15 G5 784 .240 oct
17 A#5 932 .200 oct
19 C#6 1109 .170 oct
21 D#6 1245 .170 oct

Part 2
Layer Note Hz Width
1 F6 1397 .170 oct
3 G6 1568 .170 oct
5 A6 1760 .170 oct
7 B6 1976 .170 oct
9 C#7 2217 .170 oct
11 D#7 2489 .200 oct
13 F#7 2960 .240 oct
15 A7 3520 .240 oct
17 C8 4186 .240 oct
19 E8 5274 .320 oct
21 A8 7040 N/A

Center frequencies (or cutoff for first and last bands) in the 20 Band Vocoder

Part 1
Layer Note Hz Width
1 B2 123 N/A
3 F#3 185 .500 oct
5 C4 262 .420 oct
7 F4 349 .320 oct
9 A#4 466 .320 oct
11 D#5 622 .320 oct
13 G5 784 .240 oct
15 A#5 988 .200 oct
17 C#6 1109 .170 oct
19 D#6 1245 .170 oct

Part 2
Layer Note Hz Width
1 F6 1397 .170 oct
3 G6 1568 .170 oct
5 A6 1760 .170 oct
7 B6 1976 .170 oct
9 C#7 2217 .170 oct
11 D#7 2489 .240 oct
13 G7 3136 .320 oct
15 B7 3951 .320 oct
17 D#8 4978 .400 oct
19 A8 7040 N/A
 
Forgot the vocoder file you need

Make sure the latest firmware is loaded !

If not this thingy can completely confuse your device

... theres always the possibility to load a fresh os and all the roms and settings ... but i think its smarter to be carefull
 

Attachments

  • vocoder.zip
    1.6 KB · Views: 13
Hey, Emerald Tablet!

Welcome to the board.

You seem to know a lot about vocoders, and I have a (pretty dumb) question:

Are Vocoders the thingies that some guitarist put in their mouths and sound like the guitar was "talking"? (i.e. David Gilmour in "keep talking" or Ritchie Sambora in "Living on a Prayer")

Thanks for your time!
 
nope. Those are talk boxes. The original one from the 60's was made by Heil - speaker driver into surgical tubing. They make re-issues of them now for about $150-ish.
 
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