Unclipping

Started by Anonymous in comp.dsp6 months ago 7 replies

I have lately become interested in the processing of audio signals recorded at too high a level, and so have clipping. First, I have a signal...

I have lately become interested in the processing of audio signals recorded at too high a level, and so have clipping. First, I have a signal that clipped at five samples (but only in one channel). The easy fix is to convert to a mono signal with the unclipped channel, but I might try to interpolate new values for the clipped samples. But I have another one that has about 17000 clipp...


Theory for convolution

Started by Anonymous in comp.dsp6 months ago 11 replies

Here's a mind-challenge with no solution as far as I know. Maybe someone he= re will have an idea.=20 Let's say I have a continuous...

Here's a mind-challenge with no solution as far as I know. Maybe someone he= re will have an idea.=20 Let's say I have a continuous discrete-time signal that is 5 seconds long w= ith the Dirac impulses occurring once per second starting at 0. Now lets sa= y that I multiply this signal by a rectangular pulse that starts at t=3D1/2= and ends at t=3D4.5, so the first and last sample are 0. Ther...


Sample Rate of a Digital Audio Delivery Medium (i.e., CD)

Started by Randy Yates in comp.dsp6 months ago 70 replies

Perhaps I'm forgetting something, but why would the sample rate of the digital audio delivery medium such as CD have _anything_ to do with...

Perhaps I'm forgetting something, but why would the sample rate of the digital audio delivery medium such as CD have _anything_ to do with the ease of sampling the original signal? It is certainly true that it is better to oversample the original analog input so that the antialiasing filter requirements can be greatly relaxed. However, once we're in the digital domain we can run some very ...


CORDIC

Started by rickman in comp.dsp6 months ago 14 replies

I don't think I've ever really dug into the CORDIC algorithm enough to appreciate the logic involved. I do recall coming to the realization...

I don't think I've ever really dug into the CORDIC algorithm enough to appreciate the logic involved. I do recall coming to the realization that the shift and add/subtract algorithm is not overly different from a multiplication, so I had trouble understanding why touted as a way to avoid the use of a multiplier. Looking at it harder, it would appear that the level of complexity is th...


So DC motors are obsolete!

Started by Anonymous in comp.dsp6 months ago 10 replies

Talked to a motor manufacturer in Germany. They don't even do DC motors any more, no call for them at all. Just ac and brushless DC which is...

Talked to a motor manufacturer in Germany. They don't even do DC motors any more, no call for them at all. Just ac and brushless DC which is really ac under a different name. Apparently with a good ac controller you can get an induction motor to have the same torque as a DC motor and you get max torque at zero speed - all down to flux vectoring. So do we throw out our brushed motors


Linear equalizers and similarities

Started by Peter Mairhofer in comp.dsp6 months ago

Hello, The usual setup: Suppose that I can model a channel as FIR filter h such that the received signal is y=h*x+w (*: convolution; x:...

Hello, The usual setup: Suppose that I can model a channel as FIR filter h such that the received signal is y=h*x+w (*: convolution; x: transmitted signal; w: measurement noise). The goal is to find an FIR equalizer g such that xhat=g*y is close to x. Writing x/y as vectors, the relation can be written as y=Hx+w. The equalizer xhat=Gzf y with Gzf = (H^T H)^-1 H^T is called the Z...


Create FM stereo from Matlab.

Started by disney_snoopy in comp.dsp6 months ago 34 replies

Hi all, I would appreciate if anyone here could help me. I was thinking to create a FM stereo with the RDS and L+R and L-R channel in...

Hi all, I would appreciate if anyone here could help me. I was thinking to create a FM stereo with the RDS and L+R and L-R channel in Matlab. But i doesnt know how to do on that... Can anyone here help me plz? Thankz


Physical continuation of analog filter (physical resampling)

Started by Peter Mairhofer in comp.dsp6 months ago 1 reply

Hi, Suppose I have frequency domain measurements from a channel between 1 and 10 GHz - for example like (c) in this...

Hi, Suppose I have frequency domain measurements from a channel between 1 and 10 GHz - for example like (c) in this figure: https://www.researchgate.net/profile/Sergey_Rylov/publication/2983309/figure/fig1/AS:39471058680 6272@1471117745152/Fig-1-Backplane-channel-characteristics-a-Backplaneline-card-application-b.pp m What is the best way to upsample the impulse response to a higher rate?


CPFSK receiver architecture

Started by gct in comp.dsp6 months ago 12 replies

I'm evaluating receiver structures for the ETSI DMR spec, but not having much luck finding any work in this area. They use continuous...

I'm evaluating receiver structures for the ETSI DMR spec, but not having much luck finding any work in this area. They use continuous phase modulation, so I anticipate I'll have to use a viterbi demodulator to get it done. Unfortunately they choose h=.27, so I have at least 200 states, times whatever number of symbol cycles their pulse shape is over (not clear on this, three maybe?). So I'm loo...


Michrophone amp with D/A for continuous speech recognition

Started by Richard Owlett in comp.dsp6 months ago 19 replies

My theory dates from 3 years towards a BSEE in the 60's. Real world experience focused on instrumentation ( 1 mV large signal, 10Hz very wide...

My theory dates from 3 years towards a BSEE in the 60's. Real world experience focused on instrumentation ( 1 mV large signal, 10Hz very wide band ;) [Ever try to convince a fresh chemistry post doc that demanding a system gain-bandwidth product allowing measuring less than 1 electron per second wasn't reasonable?] My voice recognition background is limited to reading end user oriented ...


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