The Human Voice has been widely characterized, correct?

Started by Ramon F Herrera in comp.dsp7 years ago 39 replies

Of all the possible sounds, since the day the ear was invented, the one that has been most widely studied and *characterized* is the Human...

Of all the possible sounds, since the day the ear was invented, the one that has been most widely studied and *characterized* is the Human Voice. While that statement is a wild/educated guess of mine, I do not believe I am not too far from the truth. There *must* be some software/algorithm somewhere that works like this: % IsThisHumanVoice someaudioclip.wav The output would be Y...


mfcc calculation - help please

Started by jasine in comp.dsp16 years ago 1 reply

Hi everybody, I am new to this group and could find that many of the group members know much about speaker identification. I am a graduate...

Hi everybody, I am new to this group and could find that many of the group members know much about speaker identification. I am a graduate student, now doing my final year project on speaker recognition. The problem is to perform speaker recognition on movie clips. I have no previous experience with speech processing. I could eliminate silence, environment sounds etc. from the audio ...


AC coefficients

Started by glen herrmannsfeldt in comp.dsp13 years ago 1 reply

In the Wikipedia JPEG page describing DCT, the (0,0) coefficient is called the DC coefficient, as usual. The others are then called AC...

In the Wikipedia JPEG page describing DCT, the (0,0) coefficient is called the DC coefficient, as usual. The others are then called AC coefficients. Somehow I don't ever recall them being called AC coefficients before, and it sounds funny to me. Is that really what they are called? -- glen


drums module

Started by Anonymous in comp.dsp17 years ago 1 reply

Hello I'm interested on developing a drums module (I mean something like http://www.harmony-central.com/Synth/Data/Roland/M-BD1-rev.html )...

Hello I'm interested on developing a drums module (I mean something like http://www.harmony-central.com/Synth/Data/Roland/M-BD1-rev.html ) and I'd like to know if someone can point me in the right direction. I'm not sure of the theory behind it. At a first view, it seens simple: You record sounds of real drums in a memory and, when the eletronic drum is hitted, it sends a signal to the dru...


Wavelet ON/OFF

Started by Ishira in comp.dsp18 years ago 2 replies

I know this question is awfully silly. But I just want to make sure that it IS indeed as simple as this and that I havent missed anything. I...

I know this question is awfully silly. But I just want to make sure that it IS indeed as simple as this and that I havent missed anything. I have a file of raw data bits from a signal capture board. I havent seen this file. But I know it's rather huge and that it has 0s and 1s. I need to run this file through a function and determine the ON and OFF periods of the signal. Sounds simple....


LPF order at DAC output

Started by krishna_1105 in comp.dsp13 years ago 3 replies

Says who? The filter characteristics (and hence it's order) should be derived from an understanding of the problem. It sounds like you're...

Says who? The filter characteristics (and hence it's order) should be derived from an understanding of the problem. It sounds like you're applying a cookbook solution that is not only narrow and simplistic, but very possibly wrong in many cases. The filter order does depend on the ADC. If a 12-bit ADC is chosen having an attenuation in stop band of less than 72dB will ensure that the un...


PitchShift using STFT

Started by Himanshu in comp.dsp16 years ago 8 replies

Hi All! Greetings! I was implementing pitch shift using STFT (the algorithm that Mr. Bernsee discusses at his site "dspdimension"). Its...

Hi All! Greetings! I was implementing pitch shift using STFT (the algorithm that Mr. Bernsee discusses at his site "dspdimension"). Its working absolutely fine but as I take the semitone to 12 which yields a pitching factor or 2.0 (one octave up), the output sounds like somewhat of "vibrato" added to it. Its not that clean. If you pitch shift the same file using Audacity at semitone val...


PC Sound card recommendation

Started by Jon Harris in comp.dsp17 years ago 7 replies

I'm looking for a PCI soundcard for Windows 2000 with the following features: dual line level analog input (1/8" stereo jack...

I'm looking for a PCI soundcard for Windows 2000 with the following features: dual line level analog input (1/8" stereo jack preferred) microphone level analog input (1/8" jack preferred) dual line level output (1/8" stereo jack preferred) digital input (S/PDIF optical or coaxial) price


Processor for music Synthesis

Started by hsmdsp in comp.dsp11 years ago 3 replies

Guys/Gals(?), I am working on a synthesizer for an Indian instrument (Tanpura, an instrument with 4 strings,that are plucked in a regular...

Guys/Gals(?), I am working on a synthesizer for an Indian instrument (Tanpura, an instrument with 4 strings,that are plucked in a regular sequence continuously, for most live performances). My algorithm is straight forward and involves STFT based analysis and synthesis. I already have the C program working on PC platforms ( You can ask me (umpmpm@gmail.com) for a demo, it sounds good !!) and ...


Interpolation by cubic splines

Started by Ross Clement (Email address invalid - do not use) in comp.dsp16 years ago

Is there any reason why cubic splines are a poor method for interpolation of audio signals such as speech or musical instrument sounds? Also,...

Is there any reason why cubic splines are a poor method for interpolation of audio signals such as speech or musical instrument sounds? Also, once upon a time I had a copy of a paper written by someone (if I recall correctly) who worked for E-mu systems talking about pitch shifting. Again IIRC, the conclusion was that for a given amount of computing power, polynomial interpolation would al...


Echo Cancellation - non acoustic application

Started by martini in comp.dsp16 years ago 8 replies

all: I dont know if this will work or not, but from what I read it sounds like acoustic echo cancellation is fundamentally the same problem I...

all: I dont know if this will work or not, but from what I read it sounds like acoustic echo cancellation is fundamentally the same problem I have. I am recording pitch measurments with a single sensor located at the center of gravity of a vehicle. I would like to extract the road profile from this measurement through a suspension model for the front and rear suspension. For the time b...


Biquad cascade

Started by baeksan in comp.dsp13 years ago 2 replies

I was wondering what the most computationally efficient way of implementing a biquad cascade would be? I've implemented an alsa application...

I was wondering what the most computationally efficient way of implementing a biquad cascade would be? I've implemented an alsa application to generate the filter coefficients and process some audio. This works great on my linux machine, but when I put the same code on a less powerful embedded system, it sounds horrible and takes almost all the cpu. The linux machine can handle around 20-30 ...


Non-harmonic data in speech

Started by Raeldor in comp.dsp11 years ago 2 replies

Are there any good techniques to either remove or detect the amount of non-harmonic data (white noise?) in speech? I really want to remove or...

Are there any good techniques to either remove or detect the amount of non-harmonic data (white noise?) in speech? I really want to remove or detect the 's', 'z' etc. sounds in the sample. Is this possible? Thanks Rael


Breakthrough in noise reduction?

Started by HardySpicer in comp.dsp14 years ago 3 replies

http://www.embedded-computing.com/news/db/?10289 Does anybody know of this chip - sounds too good to be true. We'll all be out of a...

http://www.embedded-computing.com/news/db/?10289 Does anybody know of this chip - sounds too good to be true. We'll all be out of a job! Hardy MOBILE WORLD CONGRESS - BARCELONA, Spain - February 11, 2008 - Audience today announced the industrys first voice processor based on the intelligence of the human hearing system and began sampling these new chips to mobile handset manufacturers. ...


Volume

Started by Anonymous in comp.dsp15 years ago 5 replies

Hello All, I'm currently using the Goertzal algorithm to find a couple frequencies in some audio samples. However, how am I supposed to deal...

Hello All, I'm currently using the Goertzal algorithm to find a couple frequencies in some audio samples. However, how am I supposed to deal with volume differences? On occasion, the tones are quieter and thus don't seem to be picked up by the decoder. Same thing with loud tones: they don't necessarily get picked up. Should I boost/lower volume on the fly (sounds real hard!) or tweak th...


calculation of transfer function

Started by stereo in comp.dsp15 years ago 9 replies

Hi everyone, this questions sounds possibly too simple, but nevertheless I don't get it: taken a measurement, say sweep measurement of a room,...

Hi everyone, this questions sounds possibly too simple, but nevertheless I don't get it: taken a measurement, say sweep measurement of a room, I want to get the acoustic room transfer function. I have the stimulus signal X(k) and the measured response Y(k), k being the discrete frequency. All formulae I found say: the transfer function is calulated by the cross spectrum X(k)*Y(k) divided b...


measuring the quality of mixer

Started by srikk in comp.dsp15 years ago 1 reply

Hi, I have implemented the Audio Mixer, which will mix three Audio Signal. I have done implementation using MATLAB. I am getting the Mixed...

Hi, I have implemented the Audio Mixer, which will mix three Audio Signal. I have done implementation using MATLAB. I am getting the Mixed Signal. I performed listening test on it. It sounds good. I want some quality criteria to be defined to validate Mixed Output signal. Your Input will help me. Thanks in Advance, Regards, Sri Kant


NEEDed: well formed stereo .wav file

Started by Richard Owlett in comp.dsp13 years ago 9 replies

I need a "well formed" test file with following characteristics: 1. *STRICT* conformance to "wav" file format { In another field, I've...

I need a "well formed" test file with following characteristics: 1. *STRICT* conformance to "wav" file format { In another field, I've seen 20 mA current loop working with RS-232C "compliant" receiver choke CHOKE *CHOKE* ;o 2. Exactly two channels 3. Would prefer very different 'sounds' on each channel 4. short - 30 seconds or so would be fine I've written a little rou...


FFT real input and output

Started by dcl in comp.dsp18 years ago 4 replies

Hello, I am trying to find the answer to the following: If I generate a test file to test a FFT program is the following true: Do I have to...

Hello, I am trying to find the answer to the following: If I generate a test file to test a FFT program is the following true: Do I have to generate samples that only represent cosines or if I only generated sines would that be a real signal? I know it sounds like a stupid question. But I have read in DSP books that a signal from an A/D converter is considered Real. So I was wondering do...


Sample rate conversion help

Started by KWhat4 in comp.dsp14 years ago 2 replies

I am very new to dsp in general and I am attempting to convert 16khz PCM stream (byte array) to 8khz stream (byte array) in Java. I...

I am very new to dsp in general and I am attempting to convert 16khz PCM stream (byte array) to 8khz stream (byte array) in Java. I have figured out that I can copy over every other frame from the 16khz stream to the 8khz one but I get what I think is called dithering? My question are the following: How do I clean up the audio so it sounds close to normal? Is there a better way todo th...