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6 votes

Algorithm to "serialize" impulse responses

You convolve the two impulse responses. $$h_{cascade}[n] = h_1[n]*h_2[n]$$
Hilmar's user avatar
  • 46.4k
6 votes

How can the quality of a resampling algorithm be measured?

which resampling algorithm should I look at? Polyphase FIR is the way to go. It's actually pretty east to implement from scratch. Much easier than anything FFT related. And you get a lot of knobs to ...
Hilmar's user avatar
  • 46.4k
4 votes
Accepted

Why do we have negative sign dB in spectrum in audacity?

The audio amplitude of 100% is full scale and that will be 0 dB in comparison to the 100% full scale. That's the amplitude of 1.0 in Audacity. As your audio amplitude is not full scale 100% but less, ...
Justme's user avatar
  • 2,321
4 votes

How to reduce clicking in an audio signal with seek jumps?

Clicking noise on seek is a result of discontinuity in the audio signal. One way of reducing this noise is applying a fade-in effect for few frames post seek.
SakSath's user avatar
  • 665
4 votes

How are the results of a fourier-transform interpolated to cover the entire frequency range?

This is coming from an audio developer (me): Most FFT analyzers in audio plugins are actually very simple. From the 1024-4096 frequency bins that are usually output by the FFT a 2D Path is created. ...
Tobias Hienzsch's user avatar
4 votes

What measures are used to identify a faulty speaker?

Faultiness in the sense of mechanical damage is commonly referred to as "Rub and Buzz" distortion. It can be caused by loose particles, rubbing of the voice coil on something else, voice ...
Max's user avatar
  • 2,383
4 votes
Accepted

Create a minimum phase filter from an elliptic digital filter

The impulse response hmp is computed by convolving the impulse response of the elliptic filter with itself. We know that the elliptic filter is marginally minimum-...
Matt L.'s user avatar
  • 90.7k
3 votes

What kind of distortion effect is this?

If a sine wave was going in, because of the asymmetric waveshape coming out, it can't be a straight memoryless waveshaper. The half wave coming out looks the impulse response of an overdamped or ...
robert bristow-johnson's user avatar
3 votes

Algorithm to "serialize" impulse responses

The answer Like Hilmar describe in their answer all you have to do is convolve the two impulse responses to get $$ h_{tot} \left[ n \right] = h_{1} \left[ n \right] * h_{2} \left[ n \right] \tag{1} \...
ZaellixA's user avatar
  • 1,314
3 votes
Accepted

low shelving filter frequency response diagnosis

Your filter coefficients and your difference equation are wrong. You assumed that the output signal $y[n]$ is fed back to the input, but what is actually fed back is the signal before the multiplier $...
Matt L.'s user avatar
  • 90.7k
3 votes
Accepted

Adding mono sources to Ambisonic tracks

The best approach to add a mono source to an ambisonic project really depends on how you want the mono source to behave spatially so you will have to elaborate more on that end. If you want the mono ...
Ahsan Yousaf's user avatar
  • 1,583
3 votes

Why do we choose 44.1 kHz as recording sampling rate?

The detailed historical reason for the choice of 44.1 khz vs basic Nyquist 40K, etc is given in The Art of Digital Audio by John Watkinson. The excerpt is shown at: http://www1.cs.columbia.edu/~hgs/...
JDiddy's user avatar
  • 31
3 votes
Accepted

What does the frequency of oscillating microphone feedback depend on?

Let's look at a simple block diagram The microphone receives the input sound but also reproduced sound from the loudspeaker. There are two transfer functions in place. One, $A(\omega)$ from the ...
Hilmar's user avatar
  • 46.4k
3 votes

Laplace transform of this simple parallel RLC circuit? (For audio speaker simulation ...)

That is an annoyingly difficult problem. Physical modelling of the transfer function of an electrodynamic loudspeaker in an enclosure is well understood and there is an industry-wide accepted "...
Hilmar's user avatar
  • 46.4k
3 votes

Difference between Level, Signal Strength, Amplitude and Volume?

I'm afraid there is no easy "one size fits all" answer to this. While most of these terms do have well defined scientific definitions, they are often used as "slang" where the ...
Hilmar's user avatar
  • 46.4k
3 votes
Accepted

What are the theoretical maximum values of the complex bins in an unscaled FFT of size N?

Tricky. When choosing quantization for any signal $x[t]$ your SNR (signal to noise ratio) for the quantization noise is determined by the crest factor of the signal which is the peak divided by the ...
Hilmar's user avatar
  • 46.4k
3 votes

Create a minimum phase filter from an elliptic digital filter

A filter is minimum phase when all it's poles and zeros are inside the unit circle. The easiest way to check is to inspect the zeros and poles of the filter and look at the magnitudes ...
Hilmar's user avatar
  • 46.4k
3 votes

Audio is sampled at 48 kHz, or 48000 samples per second. Does that mean 48000 bits (or 6000 bytes) of storage?

For raw, uncompressed audio, the number of bits per second is the product of three values: $$ numBitsPerSecond = f_{sampleRate}*numBitsPerSample*numStreams $$ where $f_{sampleRate}=48000$ in your ...
vml's user avatar
  • 111
3 votes
Accepted

Will open, edit, store from/to lossy audio format significantly degrade quality?

This really depends a lot on what exactly the processing is. Cutting out unwanted parts of the signal will have very little impact on the parts that you keep. Ogg Vorbis is block based and the largest ...
Hilmar's user avatar
  • 46.4k
2 votes
Accepted

BPM Detection using Matlab xcorr function

$$\texttt{BPM} = \frac{60 \cdot f_s}{\texttt{lag}} $$ Assuming a sampling frequency $f_s=44100$, we’d expect the highest correlation peak to be at $\texttt{lag}\,29400$, which is more or less in ...
Jdip's user avatar
  • 6,555
2 votes

What books describe convolutional neural networks for DSP?

To my knowledge there might not be good books on it already. I'll assume you have enough background in classical DSP and CNNs ( and NNs in general): go read the papers that things like WaveRNN, ...
Marcus Müller's user avatar
2 votes
Accepted

How to merge audio better?

Your denoising algorithm seems to be nonlinear in what it is leaving behind. The leftover noise is not phase aligned with the original noise or the noise of the next chunk. So, by overlapping and ...
Max's user avatar
  • 2,383
2 votes
Accepted

How to identify the middle of the first trough in a waveform?

Compute a moving average or a moving median $p_a[n]$ of the signal power, with a long enough window that it doesn’t get affected by spurious dips in power. Find the point at which $p_a[n]$ dips below ...
Jdip's user avatar
  • 6,555
2 votes
Accepted

Mixer and control systems

If we define "frequency domain" as the Fourier Transform: multiplication in one domain is equivalent to convolution in the other domain. So multiplication in time is the same as convolution ...
Hilmar's user avatar
  • 46.4k
2 votes

Why will the output be imprinted with delay effect or echo after convolving two direct dry signals' FFT values (convolution filtering)

The correct way of doing this is overlap add. Neither version of your code looks remotely like a correct overlap add implementation, so chances both are wrong. The proper way to determine this is to ...
Hilmar's user avatar
  • 46.4k
2 votes

Why will the output be imprinted with delay effect or echo after convolving two direct dry signals' FFT values (convolution filtering)

The difference in the behavior of the two methods, particularly the delay or echo effect in the first method, is primarily due to the way the kernel is being used in both methods. In your first method,...
Ahsan Yousaf's user avatar
  • 1,583
2 votes
Accepted

Audio EQ Cookbook Stability

Do I need to be concerned about stability? Yes, but not for the reasons you think (we'll get to this). Or are the coefficients that come out guaranteed to keep all poles inside the unit circle? Yes,...
Hilmar's user avatar
  • 46.4k
2 votes
Accepted

Laplace transform of this simple parallel RLC circuit? (For audio speaker simulation ...)

Do what r b-j suggests: get a Laplace domain equivalent first, and then transform from $s$ to $z$. The equivalent circuit will be something like: \begin{align} H(s) &= R_1 + sL_1 + \frac{1}{\frac{...
Peter K.'s user avatar
  • 25.9k
2 votes

Difference between Level, Signal Strength, Amplitude and Volume?

In audio terminology, these terms are often used interchangeably. They all influence each other (I’d say the term volume is easier to differentiate because it is a subjective quantity whereas the ...
Jdip's user avatar
  • 6,555
2 votes

Is there a Mcu/ DSP IC which can help me convert wav to MP3 (44.1 Khz sampling) in real time and store it into an embedded SD card?

yeah, any 32 bit microcontroller that you can buy these days should be able to do that, given enough RAM. The more capable ESP32 should be more than capable enough: They're dual-core machines that ...
Marcus Müller's user avatar

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