Questions tagged [audio]
Audio, or in terms of signal processing, an audio signal is an analog or digital representation of sound, typically as an electrical voltage.
1,109
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140 views
Understanding anti aliasing filter
I'm working on a project for the university about the anti aliasing filter.
I have a small audio file, which has a sampling rate of 22KHz. Now when I downsample it 8 times, the file sounds weird. So ...
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30 views
Using complex number in non-negative matrix factorization (NMF) for signal source separation
In short, I wonder which kind of spectrum can be modeled using complex number in NMF. And could an imaginary part possibly be a vector?
For detail, inspired by audio processing paper that used ...
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1answer
148 views
RF demodulation for dummies
Given a signal that's been FM encoded on a carrier wave, what are the individual steps I need to take to demodulate it, and recover the original signal?
I don't have a background in signal processing,...
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26 views
Incorporate another dimension to non-negative matrix factorization (NMF)?
My first data profile has 3 dimensions; atom pattern (I think it is similar to frequency), time, and signal intensity. I can use supervised NMF for source separation straightforwardly. Because the ...
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2answers
287 views
Extending Goertzel algorithm to 24 kHz, 32 kHz and 48 kHz
I'm learning to implement Goertzel's algorithm to detect DTMF tones from recorded wave files. I got one implemented in python from here. It supports audio sampled at 8 kHz and 16 kHz. I would like to ...
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77 views
How do audiobook players manage to quickly speedup/slowdown playback speed without using more memory?
Many audiobook player apps on Android have a reading speed slider which lets us choose the reading speed e.g. 0.5x 0.75x 1.00x 1.25x 1.50x. This is time-stretching ...
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2answers
53 views
Converting speech audio to telephone audio
I am trying to train a machine learning algorithm on telephony speech audio. However, there isn't really enough data for this anywhere that I can find. My solution is to just use speech audio from ...
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1answer
98 views
How to build a realtime simple audio spectogram? [closed]
I'm trying to build a spectogram view of history time of 4 seconds and I'm stuck at the part where you load/draw FFT vectors into the 2D bitmap. I just don't know how to do that, but i got the FFT in ...
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1answer
46 views
How do I warp a 1 dimensional signal in frequency domain?
I want to implement VTLN (Vocal Tract Length Normalization).
It is equivalent to warping the signal in the frequency domain according to the following graph.
I can't wrap my head around how one ...
2
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1answer
298 views
Compute group delay of an audio file from STFT
I'd like to reproduce results from this paper in which the authors used Group Delay-grams as an input to the Machine Learning model. The Machine Learning side is not the one that I have problems with. ...
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1answer
22 views
Implementing echo on adsp 21262 [closed]
I'm working on an echo implementation on the board adsp 21262. I've added a buffer to have more samples to work with. I end up then with 4 buffers of size 2048 each.
The problem is that I need to ...
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2answers
101 views
How to graph magnitude plots for basic 4-pole filters?
I have developed magnitude plots for basic one-pole and two-pole filters derived from various Physics PDF's and tutorials I have found online and calculated from:
https://www.desmos.com/calculator/...
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1answer
94 views
Deciding an EQ's Center Frequencies
I am currently making a relatively simple EQ of a music app, for which we decide to have 5 bands. Other than that, we want to have a bass boost knob, which, as I view it, is just another Peak EQ ...
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2answers
96 views
Interpolation of audio for new frames
I am trying to upsample a video by existing frame interpolation techniques. In the process, I realize that I also need to interpolate the audio signal for the new frames so that the audio signal is in ...
18
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3answers
3k views
Show others how I hear myself
Sooo .. I've been thinking about this stuff. We all know that we sound different from what we hear of our own voice. It is easy to find out how others hear us by recording oneself and listen to it.
...
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2answers
80 views
How to efficiently control an FIR's magnitude response by altering its phase spectrum
Question:
Extensively searching the space of all possible vectors of length $n$ to satisfy a (non-overdetermined) requirement is possible in principle. Hence, there is a way to calculate a complex ...
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2answers
120 views
Is there an audio renderer?
I wanted to know if there is a sound render like the renderers of 3d modeling software (vray , cycles etc).
I would like to experiment with sound in a controlled environment
if it does not exist ...
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0answers
57 views
FIR Audio Equalizer Crackling Noise
I've implemented a few FIR filters into a song equalizer, but there seems to be a crackling that I'm unsure how to fix.
Heres a video for reference: https://www.youtube.com/watch?v=IlzWRYpv0rk. Noise ...
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1answer
102 views
Unit Step Response of an Integrator
Desinging a very basic 1st order integrator with the continous transfer function:
$H(s) = -1 / (0.001 s)$
[No zeros, one pole at origin]
I was expecting the unit step response to be positive-slope.
...
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3answers
81 views
Does 1 kHz sine tone means $\sin(2(1000)\pi t)$ or $\sin(2(500)\pi t)$?
Does 1 kHz sine tone means $\sin(2(1000)\pi t)$ or $\sin(2(500)\pi t)$?
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1answer
160 views
Extreme quantization of audio
I would like to take an audio sample and reduce it to a very limited set of numbers, with one value for amplitude and another for length.
Example:
I want to find the dimensions of the red boxes, ...
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2answers
79 views
does audio feedback consist of harmonic content?
So sound from musical instruments as well as human voice consists of harmonics. just wondering if feedback from speakers also have harmonics component
I have heard that feedback does not consist of ...
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1answer
98 views
Frequency Modulation to arbitrary Audio signals [closed]
I understand how FM synthesis works, but I was wondering how to FM arbitrary samples or part of samples. I.E any frame of audio samples.
Further more the modulator can itself be an arbitrary frame of ...
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1answer
259 views
audio signal reconstruction from MFCC
Transforming an audio signal to Mel Frequency Cepstral Coefficients is broadly used in tasks involving learning on audio. I was wondering, is this transform invertible with some good approximability ...
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0answers
61 views
Artefacts from playing ultrasound through PA system
I encode data in audio by energising 5 frequencies between 17.5kHz and 19.5kHz:
I play the ...
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1answer
2k views
Write a 440 Hz Sine Wave to .wav-File using Python and SciPy [closed]
This is my first post on DSP.SE! I'm a student in Computer Sciences and am just getting started with Signals and Systems, thus getting in touch with discrete signals and analyzing them.
I wanted to ...
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1answer
100 views
WAV FILE FORMAT
I'working with esp32 and for some project I need to know something about wav files. I saw into many wav files with HxD and play them with Audacity. Still I do not understand why Audacity play silence ...
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2answers
277 views
Estimating effect of filter on headroom
I am working with beamforming algorithms for speaker arrays. One of the problems we face is that we need to estimate the maximum amplitude of our test signals after we applied FIR/IIR filters to them. ...
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1answer
19 views
Is there a software tool or web service to detect problems in speech recordings?
I am looking for a software/library or a web service/API that can detect problems in short samples of recordings of speech.
For example echo, clipping, pop, background noise...
Do you know of any ...
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1answer
233 views
how to create a harmonic mask from fundamental frequency?
Using an algorithm that gives the fundamental frequency across time, example https://www.mathworks.com/help/audio/ref/pitch.html (or some other one?)
how would I then use that, to create a harmonic ...
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2answers
40 views
Find original Hz of converted audio-file?
I have received an audio file which is 44.1kHz wav. My ears suspect this audio clip was originally lower than 44.1, and encoded to 44.1 later, much like re-encoding a 720p video to a 1080p video. Is ...
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1answer
46 views
Find identical audio signal
I have a long audio file (in wav) and a short audio file (in wav) that has been copied from the long file. Now I need to find the correct location where the short signal has been copied from. I tried ...
2
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2answers
491 views
Measuring microphone Signal to Noise ratio (SNR) with A-weighting
Foreword: I have already read Applying A-weighting and user @endolith's A-weighting code
I would like to measure the SNR of a microphone with the same method applied by the main manufacturers, that ...
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2answers
93 views
Modeling the human perception of noise
Add together some sinusoids with irrational or "nearly" irrational frequency ratios, and you get something pseudoperiodic. If there aren't too many, you can hear that it has only pitched components, ...
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1answer
109 views
Find filter coefficients to model a device using its measurement
I am trying to realize a digital filter that has the same freq. response of an existing speaker. I have fed an audio sine sweep to the speaker and measured the speaker output, both at 48kHz. Then I ...
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2answers
312 views
What is the proper way to implement a real time spectrogram?
I am trying to implement a real time spectrogram and I am not sure that I am implementing it the right way. First I decided to use Matlab, I was taking a short recording of lets say 0.1 seconds and ...
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1answer
103 views
Extracting feature from audio track using DWT
Our teacher gave us an assignment: Write a program to transform an audio track to time - frequency domain, and create a vector include at least 4 features extracted from transformed output signal.
I ...
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1answer
60 views
4-phase FIR LPF in MATLAB
I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on.
I'm ...
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1answer
93 views
Frequency detection in audio and what parameter or set of parameters help/limit frequency detection
I'm trying to learn audio and voice/speech processing techniques out of interest. I intend to learn, experiment with algorithms and design, implementation of such systems, frameworks using programming ...
2
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1answer
123 views
Separate two sinusoids with very close frequencies (5 Hz difference) in a 44.1 Khz audio signal, with good time-domain resolution [closed]
Here is an audio signal which is a mix of (WAV file available here):
a 2000 Hz sinusoid, beginning at 1.00 sec, without fade-in/fade-out
a 2005 Hz sinusoid, beginning at 1.031 sec, with a slow fade-...
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2answers
99 views
Recommended resources for noise reduction
I know the question What Resources Are Recommended for an Introduction to Signal Processing (DSP)? and I have already read some general DSP books, such as Rick Lyons's "Understanding DSP," and also ...
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0answers
130 views
Frequency translation with FFT
I'm new to DSP and I would like to make an SDR receiver. For this I want to shift bins in the frequency domain in order to bring to 0Hz any portion of the base band spectrum (I know that there are ...
0
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1answer
207 views
Convert 96 Khz to 48 Khz audio: is this simple downsampling method ok?
I have an audio signal x sampled at 96 Khz. I have already lowpassed it with a cutoff somewhere between 20 Khz and 22 Khz, so there should be nothing left of ...
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2answers
150 views
Any way to reverse engineer professional audio algorithms or those in e.g. rack units?
Any way to reverse engineer professional audio algorithms or those in e.g. rack units?
In order to study how they're implemented?
Not suggesting to steal ideas, but as education.
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3answers
72 views
harmonic algorithm for extending compressed bandwidth of audio file
I just found this site and feel at home :)
I've been playin around with python and some compression ideas and was wondering if there's anything like what I am trying to figure out...
I've noticed ...
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1answer
98 views
What is the name of this very simple spectral subtraction technique?
Let $S = X + N$ be the sum of two audio signals $X$ and $N$ which are both stationnary (let's think X is a constant volume 440 Hz sinusoid and N is constant volume noise).
If the sum S has a -20 dB ...
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2answers
57 views
How can I calculate what a recording would sound like if it was played through my speakers and mic?
I am trying to isolate the sound of someone speaking while music plays back in a noisy environment.
I have multiple speakers and a microphone in my system. I am able to
play back just music and ...
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1answer
98 views
What does it take to design a 4th order band-pass filter with -24dB/Octave?
I would like to implement in software the same isolators that the Pioneer EFX-500 features:
The manual says the following about the isolators:
High-performance 3-band Isolator Function
[Fourth-order ...
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1answer
30 views
Audio Processing Used in Voip Mobile Apps
I am new to VoIP apps and am learning about the various audio processing that are applied in such apps. I have looked at Linphone and ...
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1answer
50 views
About speech preprocessing in mobile phones
I understand that cell phones transmit speech signals in digital form through 2G/3G or other standards. But I could not find anything about the signal processing method lying in between the analog ...
