Re: OT Re: trying to generate a wave file of 440 Hz...

Started by Bryan in comp.dsp8 years ago

I'm not sure exactly what you're asking, but if I interpret it correctly in that you're wondering why intonation is the way it is, and why middle...

I'm not sure exactly what you're asking, but if I interpret it correctly in that you're wondering why intonation is the way it is, and why middle C and A(440) tend to be anchor notes scales are built from, I recommend the following book(s): http://www.musimathics.com/ Very cool if you like math and music. The first volume covers what I just mentioned as well as the physics behind sounds and ...


multi-band equalizer

Started by WYChen in comp.dsp15 years ago 6 replies

For CD signals (sampling rate 44100Hz), Butterworth method is applied to design 10 bandpass filters. Center frequencies of these 10 bands are:...

For CD signals (sampling rate 44100Hz), Butterworth method is applied to design 10 bandpass filters. Center frequencies of these 10 bands are: 31, 63, 125, 250, 500, 1K, 2K, 4K, 8K, 16KHz, respectively. As weighting values (0.5, 0.5, 0.5, 0.8, 1.1, 1.3, 1.6, 1.6, 1.6, 1.6) are set to these corresponding 10 badns, the "Full Treble" style of music is achieved. It seems that the function of "Ful...


Interesting Audio Processing

Started by d99n in comp.dsp11 years ago 23 replies

Hi, I've always been interested in DSP methods to decompose a piece of music into discrete instruments. I came across this new tool that's...

Hi, I've always been interested in DSP methods to decompose a piece of music into discrete instruments. I came across this new tool that's pretty fascinating: http://www.celemony.com/cms/index.php?id=dna&L=0 How do you think something like this works? Considering an acoustic guitar chord and all its intermixed, varying and slightly non-integer harmonics, how would one go about separatin...


transient

Started by Anonymous in comp.dsp15 years ago 20 replies

Hi, My mother-tongue isn't English. What does it mean "transient" in music signal processing world? I have found physics meaning: transient -...

Hi, My mother-tongue isn't English. What does it mean "transient" in music signal processing world? I have found physics meaning: transient - (physics) a short-lived oscillation in a system caused by a sudden change of voltage or current or load Best Regards, Andrzej


Some Notes on Strategies for Automated Music Transcription.

Started by Mark in comp.dsp7 years ago

I'm looking through the net for papers, references, and other writeups on the topic and am not finding a whole lot -- at least not a lot that's...

I'm looking through the net for papers, references, and other writeups on the topic and am not finding a whole lot -- at least not a lot that's particularly insightful. The key feature of the filter is that it should have a reproducing result: the ability to play back what it transcribes. What I'm looking for is some way to organize the analysis around a large digital sample archive (fre...


wireless audio device

Started by Programmer in comp.dsp14 years ago 7 replies

This isn't directly related to DSP as of now, but at the same I'm hoping that someone in here may be able to help (after all it's music...

This isn't directly related to DSP as of now, but at the same I'm hoping that someone in here may be able to help (after all it's music related!) I've been gearing up to take on a new hobby project, building a wireless MIDI controller. I've done a fair bit of research and I am pretty straight on what parts I need for the task and so on. The one roadblock I've hit is making the setu...


phase distorsion (music dsp)

Started by Robert Frunzke in comp.dsp15 years ago 8 replies

Hello, there is the following function (the base for a phase distorsion algorithm): f(x) = sin( 2*pi * PD(phase) ) phase is 0..1 How...

Hello, there is the following function (the base for a phase distorsion algorithm): f(x) = sin( 2*pi * PD(phase) ) phase is 0..1 How can I choose PD() so that fx(x) results in a triangle wave (approximation)? PD() should be small (in x86 code size) and preferably without branches and without any 'memory'. Thanks in advance, Robert


Wideband DOA estimation

Started by sherrykhan78 in comp.dsp13 years ago 1 reply

Hi, I am working on DOA estimation using super resolution methods for wideband signals, I am in need matlab code for wideband MUSIC and ESPRIT...

Hi, I am working on DOA estimation using super resolution methods for wideband signals, I am in need matlab code for wideband MUSIC and ESPRIT coherrent and incoherrent. Your help regarding these will benifit me a lot. Thanks in advance Khan


Question about FM

Started by sj in comp.dsp15 years ago 1 reply

I am trying to implement FM algorithms for computer music applications. The language I'm using (Nyquist) has a primitive FM oscillator called...

I am trying to implement FM algorithms for computer music applications. The language I'm using (Nyquist) has a primitive FM oscillator called fmosc. I want to use modulators in series which is a very common technique in some Yamaha synths. In pseudo code it is the composition fmosc(fmosc(osc())) where all oscillators output sine waves. When I have tried this there are strong non-harmonic co...


looking for broadband engineering info.

Started by Bowen T in comp.dsp14 years ago 2 replies

Hi, I am a programmer and DSP/audio engineer looking to get my foot in the door with the integrated services industry, particulary digital...

Hi, I am a programmer and DSP/audio engineer looking to get my foot in the door with the integrated services industry, particulary digital cable. I'm actually back after about a year and a half of music school and military, over which time I did no engineering whatsoever. I'm trying to fill this gap by learning my stuff as good as possible. 2 questions: 1. What is a more appropria...


Acoustic beamformer data acquisition question

Started by Tom in comp.dsp15 years ago 2 replies

I was looking for eight microphone inputs and something not too expensive. I wondered (even though the application is for music) if this would...

I was looking for eight microphone inputs and something not too expensive. I wondered (even though the application is for music) if this would fit the bill http://www.aardvarkaudio.com/aasd-v1/products/q10-main.html It comes with software - maybe I could create 8 .wav files with this kit and process off-line.I don't need real-time processing at present. Anybody used it for anything or got...


Low-pass filter algorithm

Started by Cristiano in comp.dsp15 years ago 5 replies

I'm trying to write (in C++) a low-pass filter for audio frequencies. I used this: H[i]= sin(2*PI*FC * (i-M/2)) / (i-M/2) * * 0.54 -...

I'm trying to write (in C++) a low-pass filter for audio frequencies. I used this: H[i]= sin(2*PI*FC * (i-M/2)) / (i-M/2) * * 0.54 - 0.46*cos(2*PI*i/M) where M= 100 points and FC is the cutoff frequency (0 to 0.5), but I guess I don't know how to use it because the filtered music is really bad. I have the real and the imaginary part of the transformed input signal. Could anybody te...


vorbis encoder/decoder for TI C54x Platform?

Started by Randy Yates in comp.dsp14 years ago 8 replies

Does anyone know if such animals exist? -- % Randy Yates % "Bird, on the wing, %% Fuquay-Varina, NC % goes...

Does anyone know if such animals exist? -- % Randy Yates % "Bird, on the wing, %% Fuquay-Varina, NC % goes floating by %%% 919-577-9882 % but there's a teardrop in his eye..." %%%% % 'One Summer Dream', *Face The Music*, ELO http://home.earthlink.net/~yatescr


Decimating

Started by Luis Fernando in comp.dsp15 years ago 10 replies

Hello I'm trying to implement a distortion effect (following http://music.calarts.edu/%7Eglmrboy/musicdsp/FAQs/guitar_distortion_FAQ.html )...

Hello I'm trying to implement a distortion effect (following http://music.calarts.edu/%7Eglmrboy/musicdsp/FAQs/guitar_distortion_FAQ.html ) with 48000 Hz sampling and minimum "useful" frequency = 20 Hz. My problem is with aliasing. To minimize its effect, I want to oversample, apply the distortion and then decimate it. The idea seens simple and straighforward, but what's the buffer size...


An interesting lecture.

Started by Jerry Avins in comp.dsp7 years ago 3 replies

I just returned from a Princeton Section ACM/IEEE-CS meeting at which the speaker was Joo Won Park. He is an assistant professor of music at...

I just returned from a Princeton Section ACM/IEEE-CS meeting at which the speaker was Joo Won Park. He is an assistant professor of music at Community College of Philadelphia and an electronic musician and composer. Those who read Korean might like his website at http://www.joowon.net/. It somehow reminded me of an old New Yorker cartoon whose caption was, "Yes, but it it art?" I enjoyed...


MUSIC Aglorithm (Source Localization)

Started by rudykeram in comp.dsp5 years ago 5 replies

Hi, I am trying to understand the MUISC (Multiple Signal Classification) algorithm. I am new in this topic. So, I am sorry if my questions seem...

Hi, I am trying to understand the MUISC (Multiple Signal Classification) algorithm. I am new in this topic. So, I am sorry if my questions seem so elementry. First of all, I would like to make sure that Source Localization is different than beamforming. Even though in Source Localization, we are dealing with phase arrays, but the goal is not to form (amplify) the beam in one direction and suppr...


Looking for Stefan Sprenger

Started by Frank in comp.dsp15 years ago 3 replies

Hi there, I've been reading some interesting articles on the music-dsp list written by a Stefan Sprenger and I need to contact him about an...

Hi there, I've been reading some interesting articles on the music-dsp list written by a Stefan Sprenger and I need to contact him about an idea I'm having. I've heard that he is a frequent comp.dsp'er so I'm wondering if anyone here knows him, and where to find him. The email address I found so far (sms@prosoniq.com) doesn't seem work. Any help would be appreciated. Thanks --Frank ...


Wow! 320 GMACs!!!

Started by Randy Yates in comp.dsp8 years ago

http://www.linuxfordevices.com/images/stories/ti_c66x_chart.jpg -- Randy Yates % "So now it's getting late, Digital...

http://www.linuxfordevices.com/images/stories/ti_c66x_chart.jpg -- Randy Yates % "So now it's getting late, Digital Signal Labs % and those who hesitate mailto://yates@ieee.org % got no one..." http://www.digitalsignallabs.com % 'Waterfall', *Face The Music*, ELO


A discrete Fourier transform application

Started by Rick Lyons in comp.dsp10 years ago 9 replies

Hi Guys, did you see the discrete Fourier transform application at: http://blog.wired.com/music/2008/10/how-a-professor.html See...

Hi Guys, did you see the discrete Fourier transform application at: http://blog.wired.com/music/2008/10/how-a-professor.html See Ya', [-Rick-]


Re: I am back

Started by Stewart Pinkerton in comp.dsp13 years ago

On 1 Oct 2005 11:51:53 -0700, "Radium" wrote: > Richard Crowley wrote: > > MPEG *IS* "PCM". > > Thats like saying MP3 is the same as...

On 1 Oct 2005 11:51:53 -0700, "Radium" wrote: > Richard Crowley wrote: > > MPEG *IS* "PCM". > > Thats like saying MP3 is the same as WAV. Which is wrong. No offense. WAV is a container format, doesn't tell you anything about the contents. Perhaps we should refer to LPCM on the one hand, and compressed digital on the other. -- Stewart Pinkerton | Music is Art -