new to this - Changing sample rate to mix tracks on CD's

Started by AL in comp.dsp17 years ago 19 replies

Hi, I'm a university student studying Electronics Engineering and Bournemouth University in England. For my final year project I need to...

Hi, I'm a university student studying Electronics Engineering and Bournemouth University in England. For my final year project I need to understand how CD mixers work. I know they alter the sample rate to change the speed (pitch) of the track being output thus allowing the DJ to mix the tracks (getting the tempo's to match). How is this done? Are there specific DSP processors to alter...


funding fundamental frequency(pitch)

Started by cyberaishu in comp.dsp14 years ago 26 replies

HI, We re working on a project dealing with south Indian music signals. We re right now stuck with finding out the fundamental frequency of...

HI, We re working on a project dealing with south Indian music signals. We re right now stuck with finding out the fundamental frequency of the signal. Wat we ve done so far: 1)Segmented the signal and store it in an array 2)Find out DFT of the segments and store. 3)Find the cross correlation between adjacent elements from the DFT values and store. 4) Take the maximum value from the cr...


Optimum Window Duration

Started by Huzefa in comp.dsp16 years ago

Hi all we are doing a voice verification project in MATLAB. I have done a pitch detection algorithm which is working ok ok depending on...

Hi all we are doing a voice verification project in MATLAB. I have done a pitch detection algorithm which is working ok ok depending on the length of the input as well the length of hamming window. Now for male input, the results are quite good if the window duration is 120-150 samples (sampling rate is 8 KHz) whereas for female voice i have to reduce window length to almost half if the voi...


Something like cross correlation for the time domain

Started by in comp.dsp8 years ago 2 replies

From time to time I have to do cross correlations of signals with different sampling rates. The problem is that I do not know the ratio of...

From time to time I have to do cross correlations of signals with different sampling rates. The problem is that I do not know the ratio of the sampling rate, e.g. because longer recordings are taken with different crystal oscillators for the sampling rate. The idea is to pass two recordings to an algorithm and get the pitch and the delay as result. Strictly speaking the result is two d...


Calculating the fundamental frequency of a human voice

Started by adityaram_k in comp.dsp14 years ago 2 replies

Dear all, I am working on a speaker identification system.I have a hurdle of calculating the fundamental frequency of the input voice...

Dear all, I am working on a speaker identification system.I have a hurdle of calculating the fundamental frequency of the input voice signal.The input for my application is a UDP packet from the telephone line and we go ahead calculating mfcc and lpcc for the voice print,then compare the signals using these variables. Can anyone help me with a CODE in C# to detect PITCH(fundamental...


live concert vocal autotuners?

Started by Anonymous in comp.dsp18 years ago 3 replies

There's an article about "autotuners" on www.slashdot.com today. Does anyone know what types of DSP algorithms are used by these...

There's an article about "autotuners" on www.slashdot.com today. Does anyone know what types of DSP algorithms are used by these autotuners that some pop vocalists use at live concerts to do singing pitch correction in real-time? Isn't processing latency a problem? Thanks. -- Ron Nicholson rhn AT nicholson DOT com http://www.nicholson.com/rhn/ #include


How do I handle harmonics for autocorrelation peaks? Need an algorithm I think.

Started by noodle22 in comp.dsp13 years ago 12 replies

Hi, I am trying to determine the pitch of an audio signal using autocorrelation. My autocorrelation peaks come out quite clearly and after a...

Hi, I am trying to determine the pitch of an audio signal using autocorrelation. My autocorrelation peaks come out quite clearly and after a bit of processing, I have a signal where one of the following is true a) all the peaks are close to the same height as their neighbors b) Every second peak is significantly taller than the previous peak c) Every third peak is significantly taller tha...


Fast (real-time) time stretch code

Started by R in comp.dsp13 years ago 5 replies

I am looking for code for slowing down music without altering pitch. If anyone is interested, here are some programs that are designed...

I am looking for code for slowing down music without altering pitch. If anyone is interested, here are some programs that are designed for doing this for music transcription. A couple examples here with free trials: Amazing Slow Downer: http://www.ronimusic.com/ Transcribe!: http://www.seventhstring.com/ Slow Gold: http://www.worldwidewoodshed.com/products.htm I'm looking to do some...


Implementing a graphic equalizer on speech

Started by J. Mike in comp.dsp17 years ago 4 replies

I'm a bad with signal processing but an experience c++ programmer. I have implemented a code on a real-time man's speech that makes a pitch...

I'm a bad with signal processing but an experience c++ programmer. I have implemented a code on a real-time man's speech that makes a pitch change. The algorithm works fine in the time domain and implements overlapping windows(50% overlapping). Now I want to add a graphic equalizer and use the same blocks for that. I was told that implementing an fft procedure on the blocks with a variable f...


autocorrelation code for full signal

Started by Anonymous in comp.dsp14 years ago 1 reply

Hi: i got the source code of autocorrelation (AC) for a frame (for example: 600 samples out of 40000 samples) of a signal. Can anyone show...

Hi: i got the source code of autocorrelation (AC) for a frame (for example: 600 samples out of 40000 samples) of a signal. Can anyone show me the AC code for the whole 40000 samples which means continuosly analyse every 600 samples up to the 40000th samples, so that the result would be stored in an array. i need this for pitch tracking. or can anyone give me a better idea on doing the...


Higher upsampling with minimum phase downsampling produces more aliasing

Started by jungledmnc in comp.dsp7 years ago 20 replies

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by...

Hi, I'm programming a sound generator, based on wavetables. I have 8192 point wavetable. I create several band-limited "subwavetables" by taking DFT, zeroing high octave(s) and IDFT. For generating particular pitch a choose a wavetable, which has all harmonics until 20k. Sound good so far, way better than just upsampling the original non-band-limited wavetable. The harmonics that exceed 22k ...


low frequency bandpass filters with high sampling rate

Started by ssnyder in comp.dsp15 years ago 5 replies

I am trying to design a set of bandpass filters for an audio spectrum analyzer. Since human perception of pitch is spaced...

I am trying to design a set of bandpass filters for an audio spectrum analyzer. Since human perception of pitch is spaced logarithmically according to frequency, my low pitches have quite low and narrow bandpass frequencies. I am working in matlab using designing elliptical filters. I am having issues where my lowest filters, passband 8hz to 62hz and 64hz to 247hz are not working. My sampling ...


New low spectral noise window function

Started by Bill Cox in comp.dsp11 years ago 18 replies

You can read about the OLA-Hann window function at: http://vinux-project.org/ola-hann. Features of this technique include non-tapering at...

You can read about the OLA-Hann window function at: http://vinux-project.org/ola-hann. Features of this technique include non-tapering at the edges, more like a rectangle window, zero spectral noise from sine waves that fall exactly in the middle of FTT buckets, or exactly in between buckets, and far lower spectral leakage overall. It is especially handy for pitch-synchronous speech ana...


Fundamental DSP/speech processing patent for sale

Started by Dude Whocares in comp.dsp10 years ago 23 replies

US Patent 7,124,075 =93Methods and apparatus for pitch determination=94 will be auctioned as Lot 147 at the upcoming ICAP Patent Brokerage Live...

US Patent 7,124,075 =93Methods and apparatus for pitch determination=94 will be auctioned as Lot 147 at the upcoming ICAP Patent Brokerage Live IP Action on November 17, 2011 at The Ritz Carlton, San Francisco. The patent addresses a core problem of signal processing in general, and speech signal processing in particular: period (fundamental frequency) determination of a (quasi)-periodic sig...


DMFX-1: Open Source HW/SW Audio Digital Multi Effects

Started by rezzonics in comp.dsp6 years ago 4 replies

Hello, I am working on an Open Source HW/SW project, DMFX-1, Audio Digital Multi Effects, originally conceived as a guitar pedal that will...

Hello, I am working on an Open Source HW/SW project, DMFX-1, Audio Digital Multi Effects, originally conceived as a guitar pedal that will include multitude of effects: Distorsion, Overdrive, Fuzz, Compressor, Equalizer, Phaser, Chorus, Delay, Reverberation, Flanger, Tremolo, Pitch Shift, Wah-wah ... HW is based on two TI TMS320C5535 DSP, one is used for digital audio and the other one as user ...


Update on Think DSP, and an explanation of the missing fundamental effect

Started by AllenDowney in comp.dsp7 years ago 38 replies

Last week I posted blog article on Chapter 4 of Think DSP, which is...

Last week I posted blog article on Chapter 4 of Think DSP, which is about noise: http://thinkdsp.blogspot.com/2015/01/think-dsp-chapter-4-noise.html The article includes this notebook, where I present the missing fundamental effect, where we hear a low pitch even if the signal contains no power at that frequency: http://nbviewer.ipython.org/github/AllenDowney/ThinkDSP/blob/master/code/sax...


Audio: Time and phase shift with BF533

Started by Anonymous in comp.dsp8 years ago 1 reply

Hi there! I am using a BF533 EZ Kit lite board and I want to implement an audio vocoder (phase shift and time dilation). The matlab codes for the...

Hi there! I am using a BF533 EZ Kit lite board and I want to implement an audio vocoder (phase shift and time dilation). The matlab codes for the same have been well explained and detailed by many, for eg: http://www.ee.columbia.edu/ln/rosa/matlab/pvoc/ and.. http://www.mathworks.com/help/dsp/examples/pitch-shifting-and-time-dilation-using-a-phase-vocode r.html Are there any example file