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ahmad
open ...
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readsf~
<<modified signals>>
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writesf~ open ...I want to store a file a.wav into a1.wav, but when i write it using above logic. It only creates a file "a1.wav" of 44 bytes, where as the actual size of file is around 700KB and when i listen that file, no sound in it.
so any idea, how to save a file.
please find the attched file for details.thanks.
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ahmad
input channels = 0, output channels = 0
input channels = 0, output channels = 0
audio I/O error history:
seconds ago error type
2.66 unknown
2.66 unknown
2.66 unknownI dont, know why I am getting this error. when ever I test Media->Test Audio and MDI, i got this error. any body knows how to correct it?
Last day I tried a number of PD versions, from 38 to 40, but now this error is in all version. I also tried install/uninstall loop, but no benefit.
Either it is PD mad , or this error would help me in madness.
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ahmad
I want to load cyclone library that i copied in pd folder. when i use -lib cyclone. it is not loading? any idea , how to load external libraies in pd.
thanks.
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ahmad
what is the way to convert a signal into numbers.
adc~
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FFT~
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atanAfter FFT~, I want to do (atan) of ratio of imaginary and real parts to calculate the phase. but imaginary and real parts are in signal from. Is there any way that i convert this signal into a number and then pass to atan (as atan demands a number input not a signal input).
or is there any variant of atan, who take signal input.
The formula to calculate the phase is
PHASE=atan(imaginary/real).
or is there any pd object that calculates the phase directly
thanks.
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ahmad
I want to do fft~ of a file named a.wav.
so when i read a file and attach it to fft~ object. it says fft~ inlet expected signal not a file.I undersatnd the error , that fft need signal inlet , not the file.
so how I can do the fft~ of a wave file. I want to do the fft of a wav file (stored at disk) and then want to store the imaginary part in an array.
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ahmad
I have two array , Imaginary1 and Imaginary2 , each of length 512. I want to multiply each item of Imaginary1 to Imaginary2 and store it in third array Imaginary3.
So how I can do this?I have an idea of tabwrite and tabread, but how to do manipulation at index level.
tabread Imaginary1 tabread Imaginary2
thanks.
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ahmad
I am using two simple objects.
adc~ and dac~. I want to control the output at dac~ in a left or right speaker. When i click left, voice should come only from left speaker and when i click right , voice should come from right. (just like the test sample of PD).The test sample of PD use the toggle object but when i use toggle in my program for adc~ and dac~. It dont work.
Thanks.
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ahmad
I want to calculate the pitch (frequency) of a sound. so how I can calculate it using PD. Suppose there is a "dog.wav" file and I want to calculate the pith of dog sound .
thanks. -
ahmad
I want to convert person A voice into person B's voice. so can I do this using PD.
and if yes, what objects of pd would be used.eg consider I want to change the pitch of voice into another pitch (male voice into female voice) , what objects of pd I should use.
As I am a new to pd, so I would highly appreciate the detailed answer.
Thanks.
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ahmad
http://www.busim.ee.boun.edu.tr/~speech/research.html
these guys are actively working in speech processing domain. Few years back I also studied some of their papers to extract information. Have a look over their research work and even make a request for guidance , may be some one guide you as well.
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ahmad
Well... though I do not know the exact solution of your problems as it is long time that I am away from DSP world. Anyhow few years back, I was too at your position and really curse the PD tool a lot with my friends ..... that what the hell this PD is ....:) there is no main function as we are often familiar with C/JAVA etc... it is a different world but believe me after a very short period of time, I found it very interesting and simple to program in PD. so just gear up yourself, it is new world for the orthodox programmers but very easy world. I must say.
Now come to the problem one by one. suppose we want to calculate the amplitude of voice. Do the following steps. ( some senior/junior DSP programmer can correct me, if I am wrong)
step 1: Use the adc~ object to take input
step 2: perform the fft~ to convert the input into a complex number
step 3: calculate the magnitude by using the formula sqrt (real*real + imaginary* imaginary). you have to use the object +, * and sqrt for this purpose.step 4: show the output in some display object.
step 5: try to learn write this info by using write~ or some other kind of object into file.
step 6: use the dac~ object in the end to listen your voice... -
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ahmad
I solved it by deleting the PD registry values, now it works fine.
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ahmad
Thanks. I found the solution. I used the "Pd-0.38.4-extended", and it contains the cyclone library.
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ahmad
I find some thing like "hans pd extended download" posted by hardoff, I downloaded that patch of pd, but it is for MAC OS. it contains a number of other libraries such as GEM,.... etc, but intrestingly (for my hard luck) it does not contain cyclone. and also it is not at windows plateform.
Revised Problem: I want to load cyclone library to use atan~, sin~ etc objects, in WINDOWS.
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ahmad
I finally did, how to load libraries in pd, but now there is another problem started
tried D:\\pd\\bin\\cyclone.dll and succeeded
this is cyclone 0.1, 54th alpha build
tried hammer.dll and succeeded
hammer.dll\\hammer.dll: couldn't load (problem)
tried sickle.dll and succeeded
sickle.dll\\sickle.dll: couldn't load (problem)
error (miXed): version mismatch
use a more recent Pd release (or recompile the cyclone).I am using, pd version 039-2, where as this cyclone library requires 037-038 or higher. that i quite have.
how I can recompile the cyclone? there is a makefile? but dont know how to compile at windows XP.
PROBLEM: I want to load cyclone library to use atan~, sin~ etc objects.
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ahmad
I find File->startup the right place to write -lib cyclone. but when i run pd next time, it says can not load library. so still waitning at the same STATION.
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ahmad
In Max/MSP, there are objects cartopol~ (cartesian to polar) and poltocar~
(polar to cartesian). The first object take imaginary and real parts of
FFT and give its polar values (amplitude and phase), whereas the second
object takes amplitude and phase , and convert it back to real and
imaginary. for details see
"http://www.cycling74.com/story/2006/11/2/113327/823".I searched these objects in pd, there is one object
named cpole~, but this object does not clearly states that it produces
amplitude and phase, as MAx/MSP object says.
and also pd does not have inverse of (polar to cartesian).so any idea, of these two MAX/MSP objects in pure data.
>>There is a signal domain [atan~] unit. Can't remember what library it's in though, >>have a search for it.
thanks. I try to search it. -
ahmad
<<i think you might rather record to disk the imaginary portion of the transformed sound using [writesf~] and then open it into an array using [soundfiler].>>
Actually the recorded voice is done by some other program. (may be window recorder, so I dont have a choice to store only transformed portion. what i have, I have a recorded wave sound)
<<You shouldn't discard the real portion though, if you want to ifft~ at any point. the fft~ produces a complex signal, that is to say that the real and imaginary portions are entwined.>>
Yes you are right. I would not discard it in the actual program.
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ahmad
I opened file by using read, and set the "source" as an array.
read source
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soundfiler
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atombang
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tabplay
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fft~
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tabwrite imgbut I could not get the imaginary part of fft.
pls see the attched file for details.
Objective: My objective is to load a sound file and do fft and store the imaginary part in 512 array.
thanks.