Bouncing readings in an FFT

Started by d.ka...@iemmag.com in Audio Signal Processing10 years ago

I am trying to pick out a 75hz drive signal from a large open coil which picks up a good magnitude of 60hz. I find that the derived amplitude...

I am trying to pick out a 75hz drive signal from a large open coil which picks up a good magnitude of 60hz. I find that the derived amplitude number of the 75hz bin bounces quite a bit up to about 1-2%. My current scenario is this: I am sampling at 4080hz (I started with 4096 and found that the amplitude peak/centering was at 4080hz) taking 2048 samples. With the samples, I am using Welch'...


Speaker Management System

Started by ramm...@ymail.com in Audio Signal Processing10 years ago 1 reply

Hi.... I'm new to DSP field.I'm asked to develop a speaker management system(to control delay,crossover,level etc) Since i don't have hand on...

Hi.... I'm new to DSP field.I'm asked to develop a speaker management system(to control delay,crossover,level etc) Since i don't have hand on experience in DSP,also feeling difficulty in handling C.Previously i was working in FPGA design category,also i'm thorough with verilog language. Is it possible to develop this system with the help of any FPGA kits. If not please suggest me a programming ...


overlapping in speech analysis

Started by ducn...@yahoo.com in Audio Signal Processing10 years ago 4 replies

Hi, I am a new comer in speech processing. I have a little question - why do we often have to split a long speech signal into shorter blocks with...

Hi, I am a new comer in speech processing. I have a little question - why do we often have to split a long speech signal into shorter blocks with an overlap for example - first block - from 0 to 20 ms (along time axis), second block from 10 ms to 30 ms ... and so on with the overlap between 2 consecutive blocks 10 ms. I just want to know the effect of this work - overlap. And one more question is...


AC3/DTS output through SPDIF

Started by qbq7...@yahoo.com.cn in Audio Signal Processing10 years ago 1 reply

Hi all , Does anyone know how to output ac3/dts es through SPDIF?I mean is there any specification on SPDIF ?

Hi all , Does anyone know how to output ac3/dts es through SPDIF?I mean is there any specification on SPDIF ?


inverting an FIR matrix

Started by Mona in Audio Signal Processing10 years ago

Hi all i've been working on a BSS project where i'm trying to apply ICA methods. if i generated a convolutive mixture of two source signals; and...

Hi all i've been working on a BSS project where i'm trying to apply ICA methods. if i generated a convolutive mixture of two source signals; and i know exactly the FIR filter coefficients. how can i invert the process let me try to explain more i have two source signal S1 and S2 and i am creating two convolutive mixtures as follows Y1(z)= (1 - 0.5 z^-1) S1(z) + S2(z) Y2(z)= S1(z) + (1 -...


audio clock jitter effect

Started by qbq7...@yahoo.com.cn in Audio Signal Processing10 years ago 2 replies

I have a question , every one must know that the jitter will effect the performance of audio ,such as SNR . However , if there is no data input to...

I have a question , every one must know that the jitter will effect the performance of audio ,such as SNR . However , if there is no data input to audio dac (you also can think the data are also 0), would it still effect it? such as noise floor .


Scaling output for cross-synthesis

Started by rela...@hotmail.co.uk in Audio Signal Processing10 years ago

Hi, I'm experimenting with cross-synthesis, and I was wondering how to scale my data after multiplication of the signals. I've been using...

Hi, I'm experimenting with cross-synthesis, and I was wondering how to scale my data after multiplication of the signals. I've been using unscaled fft, but I scale my ifft by 1/N, which seems to work. It seems that I need some other scaling factor for cross-synthesis, or possibly I need to do the scaling in a different place - the result of the ifft takes the combined signal way over ...


dBU and dBV recording

Started by Rohith MP in Audio Signal Processing10 years ago 2 replies

Hello, Is there any relation between dbu and db/dbV. I have IO box with option of Hi-Gain, +4dBu gain and -10dBV gain. Basically these are...

Hello, Is there any relation between dbu and db/dbV. I have IO box with option of Hi-Gain, +4dBu gain and -10dBV gain. Basically these are hardware switches present in the IO box. What does +4dbU recording actually mean? What does -10dBV recording actually mean? While performing recording of a file, I found that @ -10dBV recording, the recorded file used to get clipped, what mig...


dBU and dBV recording

Started by Gene Goff in Audio Signal Processing10 years ago

2. dBU and dBV recording Posted by: "Rohith MP" r...@lntemsys.com mp_rohith Date: Sun Jun 28, 2009 8:59 am ((PDT)) Rohith, 0 dBu =...

2. dBU and dBV recording Posted by: "Rohith MP" r...@lntemsys.com mp_rohith Date: Sun Jun 28, 2009 8:59 am ((PDT)) Rohith, 0 dBu = 20 * log10( V / 0.7745 ) and 0 dBV = 20 * log10( V / 1.0 ). +4 dBu = 1.227 V = professional level coming out of a mixing console for example. -10 dBV = 0.316 V = consumer level coming out of a CD player for example. When you switch your IO box...


what is gated response

Started by elvi...@hotmail.com in Audio Signal Processing10 years ago 2 replies

Hi~ I saw a function in a loudspeaker measurement program called "Gated Response" but there is no explaination. Could someone tell me what is...

Hi~ I saw a function in a loudspeaker measurement program called "Gated Response" but there is no explaination. Could someone tell me what is that? What does "Gate Response" do in audio aquisition? Many many thanks! Alvis


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