Audio Processing - why Subband? why Octave banks? and why Uniform banks?

Started by Yan.L in comp.dsp14 years ago 1 reply

Hi, all I believe this is a very basic and old question. I know there are a number of advantages of subband processing. Here I'd like to ask...

Hi, all I believe this is a very basic and old question. I know there are a number of advantages of subband processing. Here I'd like to ask for your opinions of why subband is so widely used in audio signal processing applications, not only in audio compression, but on many other audio related topics, e.g BSS, surround effects... Researchers also talked about octave-band filter bank...


implement biquad filter in octave or matlab

Started by katwalatapan in comp.dsp9 years ago 1 reply

Hello, I'm very new to the implementation of the biquad filter. It would really help if someone could post links where I can learn how to...

Hello, I'm very new to the implementation of the biquad filter. It would really help if someone could post links where I can learn how to implement a biquad filter as well as cascaded biquad filter in octave or matlab. I'd appreciate if someone could point me as to how do I decide the filter co-efficient and test them with respect to the desired frequency response. Thank you.


some suggestions on my octave-band spectrogram analysis in fixed point DSP implementation

Started by Simon in comp.dsp13 years ago 6 replies

i am designing one 1/3 octave band spectrogram analysis I use multirate filter bank to realize that spectrogram. It goes this way 1) first,...

i am designing one 1/3 octave band spectrogram analysis I use multirate filter bank to realize that spectrogram. It goes this way 1) first, from the biggest frequency value, I use three IIR bandpass filter, then calculate the std value. 2) Then decimate the input by 2 through(one 30 order FIR filter, then resampling the input), then use the same three coffeicents to calculate the succ...


Pre-emphasis in the frequency domain...

Started by Mohamed in comp.dsp14 years ago 3 replies

It is common practice in speech research to apply a pre-emphasis filter before FFT in order to boost high frequency by 6dB/octave. I was told...

It is common practice in speech research to apply a pre-emphasis filter before FFT in order to boost high frequency by 6dB/octave. I was told that multiplying the spectrum points by its frequency would yield the same 6dB/octave effect on the spectrum. The only mention of such method I could locate was in (Oppenheim 1970): "If S(w) represents the spectral section to be displayed, the high ...


how to realize octave band analysis through FFT?

Started by Simon in comp.dsp12 years ago 8 replies

I have realized the octave band analysis through multi-rate filter bank design. But yesterday, somebody told me he realized it through FFT. For...

I have realized the octave band analysis through multi-rate filter bank design. But yesterday, somebody told me he realized it through FFT. For some reason, he won't tell me how to realize it. Then I just wonder with these two questions. 1. Due to the low resolution in the low frequency, we need really too big size of FFT to improve its resolution, while it's imposible to be implemente...


Generation of Pink Noise using 3dB/octave filters on digital white noise

Started by bademiya in comp.dsp13 years ago 23 replies

Hi guys. I want to know how is it possible to generate digital white noise in a ADSP-BF561 board? I have tried searching for algorithms on...

Hi guys. I want to know how is it possible to generate digital white noise in a ADSP-BF561 board? I have tried searching for algorithms on digital white noise generation but i have been getting a lot of variations on the algorithms and the source codes. My intention is to generate digital white noise and use 3dB/octave filters to make the output into a digital pink noise. This output will th...


1/N octave analysis

Started by naebad in comp.dsp12 years ago 1 reply

I have read a little about this but exactly when would you use such an approach? Is it when you roughly know the frequency you are looking...

I have read a little about this but exactly when would you use such an approach? Is it when you roughly know the frequency you are looking for and need more accuracy? I understand it is a filter with bandwidth 2^(1/N) for a normalised freq centred on unity. What is wrong with ordinary FFTs? Naebad


plot 1/3 octvave frequency

Started by zohasaba in comp.dsp7 years ago 3 replies

Hi everybody I'm trying to compute plot 1/3 octave frequency of wave file . As anyone any idea about the right algorithm? Thanks a lot

Hi everybody I'm trying to compute plot 1/3 octave frequency of wave file . As anyone any idea about the right algorithm? Thanks a lot


My convolution using FFT now works.

Started by Les Cargill in comp.dsp7 years ago 10 replies

Yay! It may not seem that you helped, but you guys did. Octave helped, too. Nice to have a known-good implementation to compare...

Yay! It may not seem that you helped, but you guys did. Octave helped, too. Nice to have a known-good implementation to compare to. Multiplication, as it turns out, is multiplication. -- Les Cargill


Third Octave Equalizer

Started by Dirk Bruere at NeoPax in comp.dsp12 years ago 4 replies

Does anyone do one of these that will run standalone on a PC? I need one that will fit itself between the output data stream and whatever is...

Does anyone do one of these that will run standalone on a PC? I need one that will fit itself between the output data stream and whatever is creating the audio stream, whether WMP, RealPlayer, WinAmp etc There doesn't appear to be anything that can do this. Is it even possible? I'm even willing to pay a fair amount of money for such a product. -- Dirk http://www.onetribe.me.uk - Th...


FM or PM?

Started by Anonymous in comp.dsp13 years ago 1 reply

Take a normal analogue FM radio system (like the ones in a car). Now FM is defined as cos(Wct + k integral( f (t) dt ) ) and PM...

Take a normal analogue FM radio system (like the ones in a car). Now FM is defined as cos(Wct + k integral( f (t) dt ) ) and PM as cos(Wct + kf(t) ) Where f(t) is the signal. where k is a constant depending on the depth of modulation etc. Now it is known that broadcast 'FM' uses pre-emphasis at the transmitter (a 6dB/octave slope with break-frequencies of say 1kHz to 15kHz ...


Pitch recognition of a musical note on a smart phone

Started by mzincali in comp.dsp9 years ago 6 replies

Hi. With limited resources such as slower CPUs, code size and RAM, how best to detect the pitch of a musical note, similar to what a tuner...

Hi. With limited resources such as slower CPUs, code size and RAM, how best to detect the pitch of a musical note, similar to what a tuner would do? Should I use: - Kiss FFT - FFTW - autocorrelation - zero crossing analysis - octave-spaced filters other? Thank you in advance. -mz


Assumptions to be made while simulating for a wireless channel

Started by Kumar Appaiah in comp.dsp12 years ago 2 replies

Dear comp.dsp, I am going to try to learn a bit more about the wireless channel (specifically using OFDM) by first creating a set of symbols...

Dear comp.dsp, I am going to try to learn a bit more about the wireless channel (specifically using OFDM) by first creating a set of symbols and perform a transmit and receive. It is a simple begining I want to make, and I will probably use GNU Octave and, later, when I do know the basics, shift to IT++. My doubts are: 1. Let us say I am going to use just BPSK over OFDM. Then, I can l...


Demodulation with Hilberttransformation

Started by The Grue in comp.dsp6 years ago 10 replies

Hello, Using " highpass | rectify | lowpass | downsample" for ages to demodulate signald, I've just found that demodulation using...

Hello, Using " highpass | rectify | lowpass | downsample" for ages to demodulate signald, I've just found that demodulation using Hilberttransformation seems to be very easy (speaking matlab/octave): function y=demodulate( x ) y=sqrt(x.^2 + real(hilbert(x)).^2); endfunction (I implemented "hilbet" myself, using fft) That's the "how", easy ;) But I do not understand the "why". I u...


Audio synthesis problem

Started by Brent in comp.dsp12 years ago 2 replies

Hi, I would like to generate an audio clip with a varying frequency. A few attempts using either matlab/octave or Python code have made...

Hi, I would like to generate an audio clip with a varying frequency. A few attempts using either matlab/octave or Python code have made "interesting" sounds but not what I'm looking for yet. The sound I want will have an initial frequency F0 which will shift (interpolate) to a frequency F1 midway through the clip, and remain at F1 until the end. For example, in a two-second clip the fr...


PitchShift using STFT

Started by Himanshu in comp.dsp14 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...


DFT/FFT in log frequency scale...?

Started by DigitalSignal in comp.dsp11 years ago 1 reply

I wonder if there is a convenient fast transform available to compute the frequency spectrum with bins distributed in a logarithmic or...

I wonder if there is a convenient fast transform available to compute the frequency spectrum with bins distributed in a logarithmic or near- logarithmic frequency scale. Here is the application background. In sound/acoustic/vibration signal processing, it is common to use a so called fractional octave analysis. Usually the cctave resolution is 1/1, 1/3, 1/6, 1/12, 1/24 or 1/48. a 1/3 octav...


Re: Low Pass Digital Filter Stability Question

Started by Jerry Avins in comp.dsp8 years ago

Octave and Matlab have filter-design capability. A stand-alone application with useful bells, whistles, and horns is ScopeFIR,...

Octave and Matlab have filter-design capability. A stand-alone application with useful bells, whistles, and horns is ScopeFIR, http://www.iowegian.com/scopeiir. There's a free trial version that's only a little crippled. There are several applets available on the web. http://ptolemy.berkeley.edu/java/Filter.html is one. Jerry


GNU Octave for older ver. Windows (98SE, ME)

Started by Dave in comp.dsp11 years ago 1 reply

All, The sourceforge seems to indicate that the latest version is for all versions of Windows, but when the installer is run it aborts...

All, The sourceforge seems to indicate that the latest version is for all versions of Windows, but when the installer is run it aborts because the Windows OS in ME. Where can I find a version that is known to run under these versions of Windows? NOTE: I don't need a lot of grief about the OS - it is what I have to use because there are no funds for new computers and Linux is not an ...


DSP Hardware

Started by lm317t in comp.dsp11 years ago 5 replies

I have been playing with DSP audio for years on general purpose PC's using octave/matlab and implementing the algorithms in C and verilog. I...

I have been playing with DSP audio for years on general purpose PC's using octave/matlab and implementing the algorithms in C and verilog. I have noticed that hardware DSP is starting to be embedded in various MCUs like AVR32, Blackfin, etc. Basically I'd like to play with some real hardware but I don't know what eval board to look into. How does the DSP performance of the AVR32 on the N...