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Yes, it will work to some extent, but you're basically just making a directional mic out of two mics. If you make two recordings, it effectively allows you to adjust the polar pattern after the fact, though. Use one directional mic facing the target, and one in the same location but facing the opposite direction (with its null end facing the insect), record ...


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First of all: I'm 100% with Hilmar. I'm really not an acoustic expert, but I remember helping students build a minimal audio data transmission system, and the take aways are really these; you can, if you limit volume to "nice and far below max volume of the system" practically ignore 1., 2., and 6., and if you filter harmonics well enough at the ...


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I mean it's just acoustic waves through air No it's not. Not unless you are operating in an anechoic chamber (which is unlikely). The total transfer function of your channel can be quite complicated. You have Driver and codecs (if any) D/A & Amplifier. You have to make sure that you get decent signal to noise ratio and not overdriving or clipping ...


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I TOO WANTS ANSWER I have a CDLA earphone and a Phone that support hi-res has 24bit audio LDAC etc. etc. Lossless Audio Checker has given “Clean” ranking to this file, and the “spek” software waveform/pics/details is shown in the figure. The First file name is ; ACE OF BASE THE SIGN.flac, 192000 Hz, 5585 kbps, size is 22 mb ACE OF BASE THE SIGN 2.flac ...


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My question here is: How fast is such an histogram expected to decrease? When I look at an audio software which shows the histogram graphically, I see a slow and smooth decay, but I'm thinking that may be to please the eye rather than a real loudness level. Say "give the brain a chance to register what's going on" than "please the eye", ...


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Sound is a compression wave in a fluid media. In other words: sound is air molecules wiggling back and forth. No air = no sound. Sound is generated if something vibrates and jostles the air molecules. For example: if you hit a drum head, the head gets displaced and wiggles back and forth for a bit. This in turn wiggles the air around it. The moving air ...


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A big question. Plz have a look at this first https://en.wikipedia.org/wiki/Sound_recording_and_reproduction. Sound is highly related to vibration. Sound is generated by vibration of a sound source, and you can hear a sound is because of the vibration of eardrum. Sound Recording Sound pressure is the local pressure deviation from the ambient (average or ...


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If you want to do any overlap and arbitrarily large vectors while being fast it is a bit cumbersome. If you only do 50% overlap, then it is straight forward to write an unbuffer function


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It would be enough to divide by three, but dividing by 4 might be faster as it can be just be shifted by two. And yes, if only one 16-bit sine channel is playing, it would make the resulting output sum to have 14 bits precision per channel. The signals could be generated with amplitude 10922 so three of them can just be summed together.


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This JAES paper gives a close form of the Fourier transform of the synchronized swept-sine (SSS) signal which has the same form as the ESS $$ x(t) = \sin \big\{2\pi f_1L \big[\exp(t/L) -1 \big]\big\} $$ where $$ L = \frac{1}{f_1} \mathrm{round}\left[\frac{\hat{T}f_1}{\ln(f_2/f_1)}\right] $$ and $\hat{T}$ is the approximate time length of $x(t)$. The authors ...


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Although I cannot directly provide a solution to your problem I think I can point you towards the "most well established" approach. To the best of my knowledge, non-uniform sampling (both in time and space if this is of interest to you in some generic way) is related to uniform sampling via interpolation. I believe that the "simplest" and ...


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An asynchronous resampler should do the trick. Basically positioning a continuous time windowed sinc at the desired (uniform) output time instants, sampling it by a neighbourhood of input time instants, choosing sinc width (inverse of bandwidth) either as a function of the largest input inter sample spacing, or locally as a function of sample density. Unless ...


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I think the way to go here is oversampling, lowpass filtering and, optional, consecutive downsampling. This process will yield equidistant audiodata. 1. Oversampling Choose a target oversampling period $T_{os}$ and correspondig oversampling frequency $f_{os}=1/T_{os}$ that is well below $T_{\text{min}}$, the shortest time distance occuring in your data. The ...


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This is actually an excellent place to start with ML/DNN tools. Noise Reduction, Speech Processing and Recognition are driving a lot of the innovation in sound in this space. Recurrent Neural Networks and LSTM models are good at identifying patterns - which can be useful in this context. https://jmvalin.ca/demo/rnnoise/ If you’ve got an NVIDIA GPU you could ...


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Assuming that you know the set of fundamental frequencies, you should be able to calculate the THD for either signal (original or encoded) as $$\frac{\sqrt{v_2^2 + v_3^2 + \ldots + v_n^2}}{v_1}$$ where $v_1$ is the signal amplitude at the fundamental frequency and $v_2, \ldots, v_n$ are the signal amplitudes at the harmonics of interest for that fundamental ...


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First of: "AAC" is not a single codec; there's AAC-LC, HE-AAC, and probably a lot more. Since "audio quality" is a perceptive quality, there's no objectively "best" answer. It's, however, very likely that in all direct human comparisons, MP3 at 128 kb/s would fare worst. Also "audio quality" means something very ...


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after dividing the short values by participant count so it never gets clipped That's not a great idea, you are loosing a lot of SNR this way. I recommend running the processing in floating point are 32-bit fixed point and only reducing it to 16-bit when you create the output of the algorithm I tried reading webRTC AGC modules but it's too complex for me to ...


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