### Markus Kasemann (@dudelsound)

I don't really see why using different FFT lengths would be related to undersampling, but maybe I just didn't get it. But the Chinese remainder thing is ingenious!You...

If the carrier is far enough above the noise floor, you could boost the input signal into clipping and just count positive zero crossings with a counter and read/reset...

Hi - I am irritated by the fact that this filter behaves differently for sine and cosine waves - from the point of view of the filter the only difference is at the...

if you can do c, try this:NXP LPCXpresso55S69 - its about 40€+tax and has one Cortex M33 core with float support (plus a second one without), also there is an...

A common problem in audio spectrum analyzers is that due to the log frequency scale you need a very fine resolution for the lower frequencies. Therefore you have...

Sorry - I wasn't clear enough - I do compress the stream but not as in zip or lha, but as in lossy ADPCM - so the 5bit values are still samples representing some...

"You tolerate bit errors but not denary(decimal value) yet the denary value is transmitted as binary."Almost true. I do accept errors in the denary value. I just...

**edit**I first thought, gray code would solve my problem, but it doesn't. While it ensures that consecutive coded numbers differ by only one bit, it does not ensure,...

Can you accept that some values are deemed invalid, or do you need all of them ?** edit **-> after thinking twice: I can accept that some values are invalid....

Hi everyone. I fear that what I am looking for might not exist, but I'll still ask - you never know.I need to transmit quantized measurement values over a noisy...

The rate sounds typical for an audio application, so I assume you are talking about a sigma delta audio converter. These usually sample both channels at the same...

Hi - oversampling is:- inserting zeros in between samples- low-pass filtering of the result.The zero-insertion does not help with your problem, but the low-pass...

Hi - MP3 drops all information above 16kHz - so you just don't have it anymore. Now, you are trying to add something (and using 11-15k duplicated is just one of...

I haven't checked the exact implementation, but I had trouble with this before:You have one coefficient smaller than -2 (-2.0001525) - it might be that this causes...

Hiany 32bit processor with 100MHz should be more than enough. On a STM32 Cortex M4 with 100MHz I implemented several filters, limiter, echo-cancelling and FIR filter...

Since Audacity is open source and the code is quite readable, I also encourage you to download the code and look for yourself. I did this for a different effect...

Ahh - and obviously: Correlation in the time domain is conj(multiplication) in the frequency domain, so bringing both signals to the same length, ffting, multiplying...

depending on the signals you are correlating I have an algrithmic suggestion:Make a coarse search followed by a finer search -> run the entire correlation at...

The input stages of these devices usually have a dynamic range (difference of loudest possible input signal to noise floor) of less than 120dB. That's about 20bits...

yes, mostly. With integers, a large number plus a small number may create a large negative value. For 16bit signed integers:32767 + 1 = -32768

I don't exactly get what you mean, but I'll try anyway.There is no difference between analog and digital clipping (except for numeric overflow which is NOT happening...

I have programmed soft clippers and used them for mastering purposes and would share the source code if I hadn't done that on comany budget :(Pseudo code for positive...

If I understand you correctly, you want to use a digital signal path that does some sort of clipping as some sortz of musical effect - very much like people use...

I doubt there is something very complex going on at the output of your average DAW. I wouldn't really want it to. Within the entire signal chain there is a lot of...

From memory and without confirming the math I am not sure. Things are easier if the frequency offset is a lot bigger than your bandwidth. Then after modulation with...

well - if by 'interleaved' you mean z = [real(z0) imag(z0) real(z1) imag(z1) ...] then you need to deinterleave it and do the same:
zout(n) -> (z(2*n) + i*z(2*n+1))...

on a closer look, the entire code has pointer/value problems. For instance: target = (double*)(*(block->work)); inputDelta = (double*)(*(block->work...

/* init */ *(block->work) = (double *)scicos_malloc(sizeof(double)*3);should probably beblock->work = (double *)scicos_malloc(sizeof(double)*3);

yes, the lines inputDelta = (double*)(*(block->work + 1)); out = (double*)(*(block->work + 2));increment the pointer block->work...

I have no spcific knowledge about using C blocks in xcos, but it seems to me the problem must lie with using static variables within your block to save system states.A...

HiI'm from Germany in the Stuttgart Area. My company is still up and running. Production has been changed from one larger shift to two shorter shifts to increase...

It has been some time, so maybe this is not entirely correct, but to my knowledge it is like this:I your rf bandpass signal is centered around fc, and you multiply...

HiI cannot help you improve your problem, but I can tell you that using up-conversion or downconversion, your BER should not change at all (at least not significantly)....

k is the discrete time index or symbol index. k is the current symbol, k-1 the last one and so on.phi and I and Q are related by the formula:I + j*Q = a * exp(j*phi)so...

When you are filtering the input signal with your raised cosine that was also used when generating this sequence, this is equal to a corellation of the signal with...

"The differnece between DQPSK and QPSK should not be vissible to me at this point" - That depends on the quality of your DQPSK signal and on the exactness of your...

Hithe widely spread rumor that Q is I phase-shifted by 90° is a dangerous one. It is true if you look at a continous harmonic oscillation - this can be described...

a bit error rate of 0.5 is equivalent to guessing, so I think you have some sort of general bug in your implementation (it does not work at all) and not an issue...

Hiwe do that a lot and we usually excite the room with a log sine sweep followed by a pause and a pink noise signal of some seconds and record that with a microphone...

I don't know much about numpy, but from the comments I figure this is about an IIR filter, namely a biquad. Since there are only 5 coefficients, they must be normalized...

Yes, while synthesizing reverb has been studied a lot, dereverberation is a hot topic and to my knowledge there is no best-practice on this yet. Most papers on the...

Hi, it has been a while since I used these algorithms, so I may be wrong, but my understanding is this:You estimate the symbol duration T and sample the incoming...

I also paid 0$ for my 5 years of university in electronic engineering (Diplom - German predecessor to the masters degree). And I think that today I would be paying...

A simple brick wall limiter would be implemented like this:if (filtered_signal[k] > desired_signal_max) filtered_signal[k] = desired_signal_max;if (filtered_signal[k]...

HiWhile I agree with the others that your resolution is very low and I don't know exactly what you try to achieve, but your desired signal looks pretty much like...

It might work, but only if you have at least statistical information about the source signal - if it is a mastered pop music signal I wouldn't waste any effort since...

I don't really get the problem, I think... An allpass is a filter that alters the phase only and lets the amplitude of all frequencies of interest unaltered. If...

For an oversampling as big as yours, I would simply calculate the linear interpolation of the two neighbouring input samples. If your output sample is taken at time...

Hi.The generated files are heavily clipped versions of the real files. If you amplify the real file by factor 1000 and then clip saturate them at +-1.0, they will...

Spectral Subtraction produces artifacts over the entire frame length of your current STFT (short term fourier transform). To reduce these artifacts, start with overlapping...

Hias some above have stated the linearity of the DFT tells us that averaging the measurement results in the time domain or in the frequency domain is equivalent....

You can always do direct conversion of the RF signal (sample RF signal at very high rate and do demodulation in digital domain) if you have the power and the money.A...

That sound like frequency shift keying to me - so google "FSK detector" or "FM demodulation" or "superhet" for hints.

I know I don't answer your question, but to me you seem to have a error in your assumption: You are talking about 'high' reflections - indicating they are relevant...

I think I would use the terms "smearing" or "smoothing" instead of "filtering" in the frequency domain - if you multiply the time domain signal with a window function,...

Hi everyoneI am trying to dig into the topic of dereverberating an audio signal - e.g. a microphone signal picking up direct sound and room reflections of a speaker...

You can try not to filter the hum from the signal, but to cancel it. To do so, you have to create an synthesized 50Hz-with-harmonics signal, like a clipped 50Hz...

Two noise vectors OF INFINITE LENGTH are mutually uncorrelated - taking 100 samples of two uncorrelated noise processes will not produce uncorrelated vectors..More...

I don't know about that specific Matlab function, but I woulod assume 'sps' to stand for samples-per-second and thus 228000...

Yes I am actively deciding on which bins to modify in real time. My question is general because I can think of several things I could use this for, but consider...

I have used QMFs for a low-latency audio compression I designed a few years ago - I haven't considered them for my current task for some reason. Thanks for the hint.In...

Hi everyone.For various tasks in my day-to-day job I run into the same problem. I would like to do the following:- Slice a one dimensional signal x into chunks of...

I think windows aligns itself to the audio interface clock... And no, you don't have to use a timer at all. Everytime you receive N samples from the radio, you count...

Who defines the rate at which samples are 'picked up' (the 192kHz) - is this defined by the audio interface? Who defines the input rate (the high rate)? Is it the...

I think googling these key words might get you going:Timing RecoveryEarly-late gateMueller and MullerGardner algorithmRegards (that's not a key word :)

some micro-controllers offer hard-wired instructions to count the number of bits in a variable. Sometimes these instructions can be accessed within c by using compiler...

Yes, new samples are added to the tail.If you only want to monitor the signal (no changes made), this method will work just like that.If you want to alter the signal,...

well, you could collect only 128 samples and fill the rest with zeros - your fft will have a resolution of less than 1Hz then, but also the spectrum of your signal...

Your specification of the problem isn't exactly clear to me, but I'll give an answer to what I think you mean.Q1: Whatever you want to do with your signal afterwards...

Sorry, my week-end got in the way :) Thanks again for the hints. I will look into the sub-band stuff...

Thanks for the hints!HOW would you divide the spectrum into subbands? I tried MDCT, but the results were poorer than with the NLMS adpative filter. I also tried...

Well, maybe it is more lean than nice :) Basically it is just an adaptive filter. Length is limited by processing power which isn't much, but in simulations the...

HiI am designing the echo-cancelling/removal algorithm of a speakerphone device and am trying to get ideas on how to improve the performance.I know, physics are...

Quote:As I've explained above, there is only other frequencies, no noise at
all. And, by sure, that frequencies are sampled according to nyquist
rate, avoiding...

I do not use Matlab much so I may be wrong, but my first guess would be: Does Matlab mesaure THD+N instead of THD? That could make a huge difference depending on...

The absolute value of your correlation will always be 1, but the phase will change. Maybe your software just shows you the absolute value?

In a two-pole filter you can create conjugate complex pole-pairs - in a single-pole filter you can only create real poles - unless you have complex coefficients,...

Hithe echo-canceller adapts slowly over time. When fed with a broadband signal it will converge to the room impulse response with correct delays. If you feed the...

Firstly, I assume CW is something with "continuous" in it? Abbreviations can obscure a problem, so it is usually helpful to give the complete notion once.If you...

Another thing: Depending on what you mean by "reconstructing the sine wave" the solution might very well be simpler. if you just need to know the frequency of the...

Looks like a PSK (phase-shift-keying) modulated carrier to me - that leads me to support bholzmayer's suggestion to use a PLL to reconstruct the carrier...

Not sure I'm helping, but if I get your numbers right, you have a cosine burst at 0.75 of your original Nyquist frequency (3/8 of your original sample rate - three...

Generally speaking, you can say that IIR filters only need special attention if you want to design filters with high Q and/or f << Fs - then the precision...

I read the answers to the posts and cannot resist to add this reply even though I think the methods below will solve your specific problem well enough.I am a little...

Hi everybodyI have an interesting problem to solve. I am working on a product that runs on two AA-batteries - could be alkaline or NiMH (or Lithium or NiCd, but...

Sounds like a software optimization problem to me. Did you try to increase the optimization level of the compiler - does that minimize your 45msec some?

Most important aspects have been mentioned before, but I wanted to give another viewpoint on lossless compression: The redundancy that can be removed from a wave-file...

Firstly, you will run into trouble if you want good results with a channel spacing this narrow and a data rates that high. Your single channel carrier will significantly...

HiGMSK is a special case of GFSK and can thus be demodulated as any FM based modulation can be:A very simple implementation:Roughly filter your RF signal to exclude...

I think he wants to say that he might want to use the seed-creation-method above and then just directly use the seed as the random number instead of using it as...

"For
example, let's say I was choosing a random number from 1 - 10 using my
algorithm. Are you saying that the distribution of those random numbers
would be more...

I wouldn't use audio as a source in the first place - why would I - I just said I wouldn't go through all of the hassle described above just to get a number that...

That seams to be an unreasonable effort to me - if I were to use audio as a source for random numbers, I'd take one audio sample (as in one sample: T = 1/44100),...

There is something similar in linear predictive coding (LPC or ADPCM). here you try to estimate the next sample by a linear combination of past samples. Sometimes...

AFAIK, it is used heavily in the open source 'Opus' codecs (ogg vorbis) that you can find at xiph.org and it's resources. They developed and use something they call...

a shift in the frequency domain is performed by - well - shifting - not multiplying by some complex value.A frequency shift in the time domain, however, is performed...

HiI find Mark's summary very useful - I've done quite some stuff in fixed point and found that the biggest danger is in getting confused by all the effects that...

What makes these calculation hard is that in FSK and its derivatives, the modulating signal is in the argument of a cosine - hence it is a highly non-linear function....

Again - that's all clear - it is just a common mistake to think that since we multiplied two 16 bit values, we just take the upper 16 bit of the 32 bit result and...

I didn't read everything below, but a frequent mistake immediately comes to mind: If you multiply two signed 16bit integers, the result is only 31 bits long, not...

That depends if your "noise" is crosstalk (correlated) or noise. If it is crosstalk, you could try to remove it using a short adaptive filter:ChB' = ChB - AdaptFIl...

Convolution is all about phase-cancellation. What the convolution actually does is take your original signal and create N copies of it (for an N sample impulse response)....

Another trick might be to run a simple low-pass filter at high speed in the 3MHz domain - say a 1st order feed-back filter or a single biquad. This will of course...

Q: Does the power at the receiver matter at the RF or the baseband stage?A: The receiver adds noise to the receive signal - so more signal power means more signal-to-noise-ratio...

As spetcavich said it all depends on what you want to predict. Prediction is only possible if you have some valid assumptions about the nature of the process you...

I agree with dgshaw6.If you have complete control over every dry signal that's handed over to person Y you might make Y's life harder by introducing a very sharp...

Sorry for the hasty formulas below, I was too lazy to make real formulas.Yes I agree to that. And could I retrieve the impulse response from a dry and a wet signal?...

I agree with the others that assumptions 2 and 3 are wrong. I just wanted to say that convolution is just another word for delaying and dropping the amplitude. The...

Looks pretty cool, but is restricted to non-commercial use (as is the tool suggested by asn). Does anyone know of a really free (as in free for all uses) filter...

OK - so if I get you correctly, you want to do some sort of echo-cancelling - you transmit a signal and want to receive only the reflections, not the direct path.The...

I've implemented a few adaptive FIRs myself and what you observe doesn't strike me as too weird. I agree with Tim: By upsampling and interpolating you effectivly...

OK - I think you might be confusing yourself by mixing up a few things that are all correct in some sense, but in different senses :)Firstly: Your filter does not...

In your case it seems to me that absolute phase is not that important, so I'd say you have a few given fix-points of your filter transfer function given by gain...

I don't quite understand what you are trying to do and what that result of yours is supposed to show... If you want to show the deviation of the shifted AM/AM curves...

You can! But only if you are synthesizing the TX signal - in other words if you create those signals in matlab like you did above, you can of course change the synthesis...

I may have gotten the problem wrong, but as I see it, this is not about pre-emphasis - which in the standard usage of the term would mean frequency and phase alteration...

You may well be right - but the last little 'machine' I did this on came with a coprocessor which was extremely useful to me, but unfortunately only programmable...

If you are into writing more efficient code for very special cases (which is something I really enjoy) - you will probably find this site very nice:http://bits.stephan-brumme.com/Regarding...

I use Scilab (http://www.scilab.org/) extensively. I think the syntax is a tiny little bit different from Matlab, but it has been too long since I last used Matlab...

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