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5

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 ...


5

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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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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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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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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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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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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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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