New answers tagged audio-processing
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50 uS is about the time it takes sound to travel 6mm. So if your application is doing TDOA or stereo source positioning, you will introduce 6mm additional error or calibration requirements. If it’s sound from a fixed speaker to a listener (or singer to a mic), then it won’t make much difference unless you clamp the persons head position to less than 6mm of ...
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Assuming that this is an analog audio in/analog audio out application, in addition to the other great responses, you would see comb filtering if the input were summed with the output. All things being equal, which at such low latency seems impractical, a signal delayed by 20us would not be perceptible. However, the sum of the input and output, even if the ...
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Latency is inevitable in signal processing—you receive a sample, and some time is taken to perform computations and return a modified value. Because it's usually important to be predictable and synchronous, a full sample delay is usually the minimum. But note that when using the processor of a general purpose computer, which is running an operating system ...
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One sample latency is pretty much unavoidable so this is fairly normal. Please note that you need to look at the total system level latency which includes DA and AD conversion. Many audio applications use Sigma Delta converters which have much higher latency than one sample, so your processing latency wouldn't make much of a difference.
The impact of latency ...
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Low latency is often required in applications where the DSP is part of a feedback loop. Excessive delay in a feedback loop can limit the maximum feedback gain that can be achieved without oscillation. A classic example is closed-loop acoustic noise cancellation.
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This is an FDTD based, wave equation solution. It is the best free solution I have found. None of the other answers has helped so far.
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Firstly, here is how you can fix what you are tryin to do:
noisepsd{k} = psdvectors{k};
noisepsd{k}(signalpeaklocs{k}) = noisepsd{k}(signalpeaklocs{k}) - signalpeaks{k};
Secondly, here is why I think it is not the greatest idea: You are finding the peaks and subtracting them from the spectrum. This is actually equivalent to setting the points where you have ...
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Perhaps you can use a game development platform like Unity or Unreal Engine to build the room and to simulate the acoustics using a plug-in package like Microsoft's Project Acoustics.
For Project Acoustics, the computation will take place in Azure Batch and Azure Storage, which require a subscription. So this is not a free solution.
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As this post over at the apple stackexchange says, .aax files from Audible are indeed encrypted m4b files, which means that if I would decode an .aax file without converting it, I would get a .m4b file.
This is possible with the following command:
ffmpeg.exe -activation_bytes XXXXXX -i input.aax -c copy output.m4b
The -c copy ensures that the data will not ...
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Coupled rooms are hard siince you have two weakly coupled mode spectra to deal with. Also the details of the coupling (size, boundary conditions, geometry) become very important. If there is no direct line of sight, you need a really good diffratcion model to get the high frequencies correctly.
The method that you cite seems to be only for rectangualr rooms ...
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