# Tag Info

3

The f[i] * i together determine the phase and frequency at every i so sine wave is only rendered correctly if there is no speed change, so the algorithm must be changed to allow for continuous time i and any change in frequency must not cause a phase jump. So try a phase accumulator approach, where phase for each i is incremented as required by the frequency....

3

One important of them is the dynamic range (in sense of "loudness war"). By which in this context is not meant the SNR but the (naive said) "variance" of the "occuring energy". Loudness War is neither dynamic range nor SNR nor variance of occuring engery (although it's related). It's mostly about peak limiting. For better or ...

0

I don't like the other answers. You can just zero pad the DFT to obtain higher resolutions. The primary reason is that the spectrum is tainted by pitch harmonics, and thus the accuracy of just using a DFT will be limited by the pitch (higher pitchs will not be very accurate). What you really want are the peaks of the spectral envelope, and the peaks of the ...

4

I'm almost tempted to answer with a counter question: what is frequency? It's a question I like to ask students that have already taken signals and systems courses and it's surprising how often they don't know what to answer. While the whole answer involves some more discussion, what we can definitely say is that frequency is associated to a rate of change ...

0

Feedback without a gain factor (or a very high one). Incidentally, this is the principle behind Schroeder reverberators.

0

When you spoke the first few words, the microphone recorded it and the audio was played through the desktop's speakers. Since the audio output of the desktop would be louder than your voice, the microphone picked up that audio signal. This recorded audio would be played on the desktop again, and it would get recorded by the microphone. This forms a closed ...

1

I learned decades ago in electrical engineering classes that the Frenchman, Joseph Fourier, realized that any periodic waveform can be represented by an infinite sum of sines and/or cosines. The frequencies of the sines and cosines are integer multiples (including one) of the frequency of the original periodic waveform. That frequency is called the ...

3

A reasonable, analytically derived cubic interpolation method will give a straight line if the inputs are on a straight line. With $a = 0$, $b = 1$, $c = 2$, $d = 3$ you get from cubic_bourke $A = d - c - a + b = 3 - 2 - 0 + 1 = 2$ as the coefficient for $x^3$ whereas for a straight line you would have $A = 0$ (and $B = 0$ as the coefficient for $x^2$). So ...

0

I'm going to chalk this up as an optimization with a trade-off. I wrote a program to plot the results of the two interpolation methods, the one I inlined above (Bourke) and the other one that I originally derived (Breeuwsma). The interpolating functions used: # https://www.paulinternet.nl/?page=bicubic def cubic_breeuwsma(x, a, b, c, d): return b + 0.5 * ...

0

One of Microsoft's many audio APIs (MME, AKA "waveOut") has a bug in Windows 7. https://www2.iis.fraunhofer.de/AAC/ie9.html I am using the library PortAudio, so my solution is to compile it without MME support, forcing it to use DirectSound or WASAPI instead.

3

Could be a few things You may be borderline clipping. Your sine wave has an amplitude of 1, which is just at the edge of clipping (depending on how its rendered). Try it with an amplitude of 0.5 Your hardware is sloppy. For example, cheap laptop sound cards often cut corners in the anti aliasing filters and or clipping management Your operating system is ...

0

That's is most probably because the filtering post sampling is not adequate. After sampling you would repetitions of the frequency spectrum of the sinusoid, which need to be filtered out. Or there are non linearities being produced in the acquisition. The reason you only hear it with certain sampling rates is the filtering (bandwidth) in Audacity amplifies ...

1

Hello NatanBackwards and welcome to DSP SE. Your results seem valid to me but I would like to mention a couple of things here. First of all, you have to keep in mind that due to the way "log-sine sweeps" work the part before the impulse response is comprised of the distortion components. Judging from the way you have handled the measurements I ...

5

If you calculate the error, 44102/44100 is only about 45 parts per million. That is well within operating tolerance of many crystal oscillators used in consumer equipment to generate audio sampling rates and USB communication clocks. You can be quite pleased it is not even more off. Another problem is that if you simultaneously use different devices for ...

1

Here is an excellent reference on the topic. It's a survey on acoustic feedback control. http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.231.9808&rep=rep1&type=pdf

0

WOW long gap to answer this question, maybe if I have a Einstein brain I can understand the image, what is not my case lol. I haved touched in this algorithm in a long time ago, and I remember that was a surprised for me that almost no one around the Web talks about the Algorithm from Morita Naotaka, I tryed seach his original paper but I cant found online, ...

Top 50 recent answers are included