white noise spectrum

Started by jaso...@gmail.com in Audio Signal Processing10 years ago 2 replies

I have use a simple LFSR to generare random number. Set of random number are pass through to FFT(power in db vs freq). However, the freq response...

I have use a simple LFSR to generare random number. Set of random number are pass through to FFT(power in db vs freq). However, the freq response is increasing at low freq. Then only maintain roughly flat for higher freq. Is there any problem with increasing at low freq? If yes may i know the reason? FYI, lot of white noise freq response is almost flat from low to high freq. Link of the whi...


comparaison of 2 audio signals and calculation of filter

Started by roma...@yahoo.fr in Audio Signal Processing10 years ago

Hi everyone, For a linguistics experiment, I have loudspeakers playing sounds in a room and they are directed towards a wall, which acts as a...

Hi everyone, For a linguistics experiment, I have loudspeakers playing sounds in a room and they are directed towards a wall, which acts as a low pass filter when you hear the sounds in the neighbouring room (where I am making recordings with my experimental subjects). The goal is to see the influence of ambient sounds on speech. It would be very nice if i could actually know more about the ...


Re: Re: How do I "flip" the frequencies of a sound file? What I mean is...

Started by Jeff Brower in Audio Signal Processing10 years ago

Possimpable- > Well I have Audacity and Goldwave, and I haven't found such a feature to apply these equations The equation is simple: ...

Possimpable- > Well I have Audacity and Goldwave, and I haven't found such a feature to apply these equations The equation is simple: y[n] = x1[n]*(-1)^n 1) Make a .wav file that contains the sequence 1, -1, 1, -1..., with length equal to your sound .wav file. 2) Multiply them together. > so instead of being getting condescension for my questions I'd rather have a little assistance


How do I "flip" the frequencies of a sound file? What I mean is...

Started by possimpable in Audio Signal Processing10 years ago 1 reply

I want to take a sound file, and geometrically flip every frequency around,= say, A (440Hz). So A# becomes Ab, B becomes G, C becomes Gb... To...

I want to take a sound file, and geometrically flip every frequency around,= say, A (440Hz). So A# becomes Ab, B becomes G, C becomes Gb... To basicall= y flip the frequency scale (f' =3D 440=B2/f). It's a similar concept to voi= ce-inversion-scrambling. The only way I know how to approximate this effect= is by producing a spectrogram (with a logarithmic frequency axis), vertica= lly flippi...


Re: How do I "flip" the frequencies of a sound file? What I mean is...

Started by Jeff Brower in Audio Signal Processing10 years ago

Possimpable- > Okay, but am I supposed to know what program to plug this > equation into? I already know the concept, I'm asking about > ...

Possimpable- > Okay, but am I supposed to know what program to plug this > equation into? I already know the concept, I'm asking about > the execution. You're kidding me, right? That equation is so simple you can use any number of audio editing programs to do it. CoolEdit, Audacity, Goldwave...or technical computing software like MATLAB, LabVIEW, Hypersignal... > I insisted on using


Re: How do I "flip" the frequencies of a sound file? What I mean is...

Started by Jeff Brower in Audio Signal Processing10 years ago

Possimpable- > I want to take a sound file, and geometrically flip every frequency > around, say, A (440Hz). So A# becomes Ab, B becomes G,...

Possimpable- > I want to take a sound file, and geometrically flip every frequency > around, say, A (440Hz). So A# becomes Ab, B becomes G, C becomes > Gb... To basically flip the frequency scale (f' = 440?/f). It's a > similar concept to voice-inversion-scrambling. The only way I know > how to approximate this effect is by producing a spectrogram (with > a logarithmic frequency axis), ver


Difference between original sound and microphone recorded sound

Started by kali...@gmail.com in Audio Signal Processing10 years ago 1 reply

Hi, I was wondering if anyone can formalize the difference between a played audio buffer and the same audio buffer, played from a speaker and...

Hi, I was wondering if anyone can formalize the difference between a played audio buffer and the same audio buffer, played from a speaker and captured by a microphone. For example, if I take the original buffer and the captured buffer and run on both FFT, would the result look the same? what is the expected difference between the two? how can I know it's the same content? Can anyone recom...


Re: Difference between original sound and microphone recorded sound

Started by Ofer Kalisky in Audio Signal Processing10 years ago

I'm aware it's not the same buffer eventually. My question is whether there's a simple test (FFT, or any kind of variant) that would make it...

I'm aware it's not the same buffer eventually. My question is whether there's a simple test (FFT, or any kind of variant) that would make it easy to test a segment and tell whether it's the same original data even if it was recorded from the speaker by the microphone? On Mon, Dec 7, 2009 at 3:37 PM, sada wrote: > Hi , > > Surely there will be difference in the


Audio signal sync

Started by hakpax in Audio Signal Processing10 years ago 13 replies

Hi, I'm developing a real-time audio processing project which consists of 2 DSP separated systems. One system is encoder and other decoder....

Hi, I'm developing a real-time audio processing project which consists of 2 DSP separated systems. One system is encoder and other decoder. between the systems there is a regular audio line. the encoder samples the input audio data (8/16/24 bits, 16Khz sampling rate) separates it into frames and process each frame individually. The processed data is played by the D/A to the audio line and i...


Cross Correlation on a 20kHz tone

Started by "cry...@ymail.com" in Audio Signal Processing10 years ago

Hi there, I am working on a "robot" that determines the source of an audio signal. Right now I am using cross correlation to determine angle...

Hi there, I am working on a "robot" that determines the source of an audio signal. Right now I am using cross correlation to determine angle and it works beautifully if I shout or clap at it. However, I want to use a 20kHz tone because its inaudible and I'd like to be able to ignore other loud sounds in the room. Right now I am working in Labview. It has a very difficult time cross corr...


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