voice sampling

Started by sukh_ece in Audio Signal Processing11 years ago 1 reply

i m new to this subject... i m little confused over voice sampling... it is easy for me if voice were composed of single frequency... but voice is...

i m new to this subject... i m little confused over voice sampling... it is easy for me if voice were composed of single frequency... but voice is made up of a band of frequencies... how one sample at a paticular instant sould look like....because one sample would mean simultaneous amplitudes of many frequencies... so how a code is assigned to a sample that is made up of many amplitudes... thanks...


Test Signal to get Static Compression Curve?

Started by mark...@gmail.com in Audio Signal Processing11 years ago 1 reply

What is the ideal test signal to use to get a Static Compression Curve for a dynamics processor (compressor)? Can you assume that the processor...

What is the ideal test signal to use to get a Static Compression Curve for a dynamics processor (compressor)? Can you assume that the processor operates the same across 20Hz-20kHz? In that case, then a simple 1 kHz sin waveform that is gradually increased in db amplitude should work as long as its slow enough to not trigger the attack and release sections of the compressor? The compresso...


Res: Test Signal to get Static Compression Curve?

Started by Christian Herrera in Audio Signal Processing11 years ago 2 replies

Hi Mark, I really didn't understand what you called "Static Compression Curve", but anyway... I would try a burst signal, sine wave....

Hi Mark, I really didn't understand what you called "Static Compression Curve", but anyway... I would try a burst signal, sine wave. Then, compute the envelope and take the parameters. Regards, C.Herrera ________________________________ ________________________________ De: "m...@gmail.com" Para: a...@yahoogroups.com Enviadas: Ter=E7a-feira, 14 de Abril de 2009 13:17:


WAV Data

Started by kash...@gmail.com in Audio Signal Processing11 years ago

Hi I am new to this so any help would be highly appreciated. I would like to know how to calculate SNR for a wave file. The following are...

Hi I am new to this so any help would be highly appreciated. I would like to know how to calculate SNR for a wave file. The following are the wav properties: Version : libsndfile-1.0.19 ======================================== File : 1.wav Length : 108336780 RIFF : 108336772 WAVE bext : 602 fmt : 16 Format : 0x1 => WAVE_FORMAT_PCM Channels : 1 Sample Rate


Non relaxed systems linear?

Started by gpal...@yahoo.com in Audio Signal Processing11 years ago 2 replies

Hello all, Can non relaxed systems be linear? Why is the condition of linearity(superposition) defined only for relaxed systems? System...

Hello all, Can non relaxed systems be linear? Why is the condition of linearity(superposition) defined only for relaxed systems? System like y(n) = Ax(n) + B is linear in its equation but does not satisfy the linearity test of superposition. What can be said about such systems? Thanks Pallavi


University for the course Audio Signal processing, Audio Engineering-programming

Started by jr_vignesh in Audio Signal Processing11 years ago 2 replies

Hello all, I am Vignesh doing B.E. in India. I wish to apply for the course:Audio Signal Processing I could not exactly find the university...

Hello all, I am Vignesh doing B.E. in India. I wish to apply for the course:Audio Signal Processing I could not exactly find the university for the above mentioned course. All the courses that i searched were concerned with Music and were not fully engaged with Acoustics and Digital Signal Processing. I wanted to apply for course that is related only to DSP and acoustics. Do anyone kno...


4rth order digital Butterworth filter coefficient calculation

Started by etha...@peavey.com in Audio Signal Processing11 years ago 1 reply

Hello, I'm trying to calculate coefficients for a 4rth order Butterworth filter, to be implemented as two cascaded biquads. It seems easy...

Hello, I'm trying to calculate coefficients for a 4rth order Butterworth filter, to be implemented as two cascaded biquads. It seems easy enough to come up with the poles in the s domain and split the transfer function into two pairs of complex conjugate poles, so that I have two 2nd-order sections that I can BLT into biquads. Say I want the complete fourth order filt...


Relation between IMDCT and IFFT

Started by peter jebaraj in Audio Signal Processing11 years ago 2 replies

Hi all, Could you please explain the relation between IMDCT and IFFT? Thanks in Advance, Jebaraj S.Peter.

Hi all, Could you please explain the relation between IMDCT and IFFT? Thanks in Advance, Jebaraj S.Peter.


Chamberlin state variable filter transfer function

Started by kt...@am3d.com in Audio Signal Processing11 years ago 4 replies

Hi, I am comparing coefficient quantization errors for second order IIR filter structures (for fixed point implementation). I have found the...

Hi, I am comparing coefficient quantization errors for second order IIR filter structures (for fixed point implementation). I have found the Chamberlin state variable structure to perform well at low frequencies and have seen in the article "The modified Chamberlin and Zölzer filter structures" that the quantized pole distribution is dense in the low frequency area of the unit circle in the z-pl...


Choice a DSP's Model for Audio applications

Started by "ahe...@ymail.com" in Audio Signal Processing11 years ago 1 reply

Hello everybody! I'm new in this Community! I'm trying to find DSP's models= for Audio Digital Applications in the professional way. I've found...

Hello everybody! I'm new in this Community! I'm trying to find DSP's models= for Audio Digital Applications in the professional way. I've found the TMS= 320C672x, but i can=B4t find the starter kit's for this DSP's. Now, I need to know if exist Starter Kit's for TMS320C672x or another model= s for similar applications.=20 Thanks! _____________________________________


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