when i run this program, it gives me this error (no such array), what is wrong? . This pd file is the implementation of VoCoder.

2nd Question:
What is diference between (adc~ and inlet~) and (dac~ and outlet~). Both are used for audio input and output and are same ( at least to me.)

http://www.pdpatchrepo.info/hurleur/timbre.pd

The file you gave is only part of a bigger patch. You need the other bit that has a [tabwrite~] in it.

[inlet~] and [outlet~] are used to send signals in and out of a subpatch or abstraction.

They do not send and receive signals to the soundcard,

Use [adc~] and [dac~] to do that.

I have written this piece of code from the book "The Theory and Technique of Electronic Music ". The book discuss it under the article "Timbre stamp (``vocoder")". that I am assuming would change the timbre of voice.

I have not a detailed knowledge of DSP, as I am not a DSP student.

I am just using the approach of "bulit-in component" of software engineering. So can you add some thing in this code, so that it works, and change the timbre.

I am thankful to you for your already assistance in pitch shifting module.?

I have to implement two things.
1- pitch shifting
2- timbre shifting

Thanks.

Patience young padawan, your training is not complete

A vocoder based on FFT and resynthesis can be used to change timbre, but that is complicated. If you do not understand the method you will not learn much. It will not help you if I just change it to work. Neither will you learn much if somebody just completes your assignment for you It is better to take a simpler approach and break it down.

Millers book is for advanced students and you need to know DSP.

A solution to part 2 does not have to be a vocoder. There are many ways to change timbre.

A simpler solution to part 2 can be an ordinary filter or a modulator that adds new harmonics. For example try researching "ring modulation".

Here is a very *very* basic filter bank used with FFT. There are 16 frequencies we find. See them appear in the table when a sound is played. See how they change in time.The rest of the patch takes each frequency and applies it to the level of a fixed filter. A pulse wave is passed through the changing filter.Hear how it sounds a bit like the voice after processing? It's timbre follows the voice. You need many more than 16 filters to make a proper vocoder but this demonstrates the idea.

Listen to the pulse wave on its own, then through the filter bank. The filter bank is just like an an "EQ" that sets the level of frequencies that are already in the sound being processed.

Use that as a basis for experimenting and try different inputs to the FFT and filter bank. Don't hand in the example I gave you, your master probably reads this board too

We can discuss this example if you have more questions.

have fun

http://www.pdpatchrepo.info/hurleur/voc16.tar.gz

Hey andy, any chance of posting this in an (evil) windows zip file

Thx,

Domien

certainly, here ya go

(ps it's not much use for music - just demonstration, it is a terrible vocoder. The filters would need to be steeper and accurately tuned to the detected bands A voc made with individual filters would be very expensive too)

http://www.pdpatchrepo.info/hurleur/voc.zip

Yes U are right but I am also not wrong:) Actually neither I am a signal processing student nor I have any ambition in this field. My actual task is some thing different.

I have to evaluate (Only EVALUATION) of voice morphing technology. There are some other avaialble voice morpher tools (such as AVS, Vodi, and many others etc) but those tools do not serve my purpose as they are hard to integrate with my other application.

So if I would not make myself dirty with DSP, it would not hurt my master:) I and my master goal and intentions are different. Even my master is not from Engineering field . He is a socialgist . at least her last degree is in socialiogy.

I am also not an from electrical engineering field and I also know, that may be at higher level I would undersatnd the meaning of

Filters, FFT, pitch, timbre, formant, pitch shifting, Vocoder, phase alignment etc.

but I would never understand this domain in depth (not due to my incompetency rather due to my different domain)

so as you said

"It will not help you if I just change it to work. Neither will you learn much if somebody just completes your assignment for you ".

So it would works for me, as my final goal is some thing different. (that goal is experientaion/evaluation, Human Psycholgy etc )

So Guru, I assure you, that if you do this work for me. You would not be penalized by Plagrism act:). rather I would fully document your name in my final thesis.:)

I give you some idea, what I want to do

A says: let's go to cienma (centre)
B says: no busy in course work (left of A)
C says: me too (right of A)
B says: hmm... we would go on TUESDAY. (morph into C,s voice)

(morphed B,s voice into C's voice and send at right channel of A ).

Now in this all conversation, A brain is tune up that what ever is coming from left is B's voice and on right it is C's voice.

so Here i would challange the human brain in voice recognition process? and would try to identify that how much location matters in voice recognition?

so you can guess that I have to do nothing with DSP:) Sorry I try to convience you - though I know It is really hard (impossible or even sin) to convience TEACHERS.

Aha! This is a very interesting subject. Please forgive my presumptions.

The full FFT vocoder patch is attached. To use it you must experiment with loading different files and cross synthesising them. It is from the help files and should work as is.

Some problems will present themselves very quickly...

1. To cross synthesise two voices you must ensure that two speakers make exactly the same
   utterances which are phonetically aligned. This is hard as I can tell you from experience of recording many voice artists. Even the same person will not speak a phrase the same way twice.

2. The result is not a "timbral morph" between the two speakers. The human voice is very complex. Most likely the experiment will be invalidated by distracting artifacts.

Here's some suggestions.

1. Don't "morph" the voices, simply crossfade/mix them.

2. For repeatable results (essential to an experiment) a real-time solution is probably no good. Real time processing is very sensitive to initial conditions. I would prepare all the material beforehand and carefully screen it to make sure each set of subjects hears exactly the same signals.

3. If you want a hybrid voice (somewhere between A and then vocoding is not the best way. There are many tools that would be better than Puredata which are open source and possible to integrate into a wider system.

Csound has a LPC suite. Linear predictive coding is particularly well suited to speech.

ii) Tapestrea is a wonderful tool that uses wavelet analysis. It also has a graphical front end that makes alignment of phonemes easier.

iii) Praat (Boersma and Weenink - Amsterdam Institute of Phonetic Sciences) is a great voice synthesis system based on articulatory tract models, where you can morph speaker models. You may find that a purely synthetic method yields data more suitable for this experiment.

> It is really hard (impossible or even sin) to convience TEACHERS.

Did you mean even cos, sin is an odd function, write it out 50 times!

http://www.pdpatchrepo.info/hurleur/3.phase.vocoder.pd

Also here's the sheep-goat you asked for....

(ruminants not included)

http://www.pdpatchrepo.info/hurleur/06.sheepgoat2.pd

1. To cross synthesise two voices you must ensure that two speakers make exactly the same utterances which are phonetically aligned. This is hard as I can tell you from experience of recording many voice artists. Even the same person will not speak a phrase the same way twice.

<< This is not possible in my experiment, as I am supposed to morph the actual conversation, so it is upto subjects what ever they want to speak. There is some work done by (Oytun Türk , Levent M.Arslan) who conducted experiment in passive enviornment (not at real time).

2. The result is not a "timbral morph" between the two speakers. The human voice is very complex. Most likely the experiment will be invalidated by distracting artifacts.

Here's some suggestions.

1. Don't "morph" the voices, simply crossfade/mix them.

<< yes I also want to do this, crossfading and mix, as I just want to create illusion, so that listner start thinking whether it is B's voice or A's voice>>

2. For repeatable results (essential to an experiment) a real-time solution is probably no good. Real time processing is very sensitive to initial conditions. I would prepare all the material beforehand and carefully screen it to make sure each set of subjects hears exactly the same signals.

<<Yes, I agree, but it is demand of experiment, I can not control the environment, but to create some good illusion (or to distract listner, I may add noise in the siganl at real time, it would challange the listner brain in identification, so I may use such kind of tricks for the sucess factor)

3. If you want a hybrid voice (somewhere between A and then vocoding is not the best way. There are many tools that would be better than Puredata which are open source and possible to integrate into a wider system.

<<Actually now I have a little familarity with pure data, so it is more good for me to stick with it for a while (due to short time), yes if i continue my phD in this domain, I would explore other tools as well. for the time being it is a kind of pilot study>>

Question :
Is VoCoder alone is sufficient for morph/mix/crossfade among two voices?
or I should also add the pitch shifting module with VoCoder to get some more qualitative results.

Question 2:
I already tried this VoCoder example, but could not change it according to my requirements. In my requirements, I have a target voice (the target voice is phonetically rich) and now source speaker is speaking (what ever he want to speak) and source voice is changing into target voice. (illusion/crossfade/mix)

The (changetimbre1.pd ) file that I attached first, give you an idea of what kind of operational interface I am looking for.

Question 3:
what should be the ideal length of target wave file?

Before the start of experiment, I would collect the voice sample of all participants.

I am highly oblige for your earlier help and looking for more (greedy). Meanwhile I would once again study this vocoder example to change it according to my requiement. ( though i doubt I may change it).

Thanks.

Hi Daisy,

I followed your reference to Türk and Arslan, I found many interesting claims but no published material. Critically I cannot find a published algorithm nor any experimental data. Can you give a reference to the algorithm as a complete DSP block diagram, pseudo code or source code please. Further, I found a reference to a patent on this claimed approach which as a scientist I have moral objections to. If this work is not in the open and free to build upon it has no utility. One abstract mentions training. This implies a neural, expert system or Markov approach. This aligns with my proposal given in the other thread to use a dictionary of transformations on the intermediate data.
As you said you are not a DSP programmer so I believe this is probably beyond the scope of your research. Have you tried contacting the authors to ask for their code?

I must admit I do not entirely understand your proposed experiment. It seems flawed in two repects. It has arbitary constraints and too many variables for what I understand are the experimental objectives.

To answer your questions as best I can:

> Is VoCoder alone is sufficient for morph/mix/crossfade among two voices?
> or I should also add the pitch shifting module with VoCoder to get some more qualitative results.

A vocoder, even if combined with pitch shifting is insufficient to create a morphology between two people speaking unpredictable sentences in real time.

> Question 2:
> I already tried this VoCoder example, but could not change it according to my requirements. In
> my requirements, I have a target voice (the target voice is phonetically rich) and now source
> speaker is speaking (what ever he want to speak) and source voice is changing into target
> voice. (illusion/crossfade/mix)

I'm sorry, I can't find the question.

> Question 3:
> what should be the ideal length of target wave file?

5 seconds

Time permitting I may be able to help you and your supervisor design the appropriate framework for this experiment. Please contact me by private email (look in the profile), as we are now off-topic for this forum.

best regards,

Andy