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.
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Separating Ultrasonic signal from echo at the receiver
A transducer is acting as a transmitter and a receiver. It keeps on sending pulses which can hit an object placed at 5cm. Echo comes back and hits the transducer while it is continuously sending ...
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26 views
How do I play and record audio at the same time?
So this question is more related to Matlab itself.
Some background info: I'm trying out the Exponential Sine Sweep method to obtain the Room Impulse Response with my laptop speakers and microphone ...
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2answers
45 views
Spatial Audio (Ambisonic) and calculating sound source position
I have 4ch Spatial Audio (Ambisonic) WAV file and I want to calculate sound source position and put it on 2D plane.
What I know so far is the following:
WAV file consists of signals that support of ...
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38 views
Method for the Localisation of Impulsive Noise in Speech Signal
I'm sorry if this question has already been answered, but I couldn't find anything.
I'm working on a program for noise detection and need to locate the position of impulsive noise. The following image ...
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1answer
31 views
Error reading a .PCM file
I want to convert a .wav file with a sampling frequency of 44100 Hz to a 16 depth .pcm.
I don't why I'm getting those peaks at the beginning of .pcm plot (second figure below 2). If you could ...
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34 views
Dynamic compressor - ratio in logarithmic scale problem
I want to use decibels to describe compression ratio. So for example ratio 4:1 means that every 4 input decibels above threshold causes raise of output only 1 decibel. And I also want ratio to be ...
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18 views
Language detection from audio
I'm trying to do a simple test on language detection on a video input. Basically, given a .mp4 file, I need to process the audio tracks (can have primary and ...
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1answer
69 views
Definition of DSP Terms
can someone define/explain briefly (or with some detail-> up to you) the following terms:
Block Processing? and what are the differences between it and Sample Processing?
(removed)
What is a 3D ...
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1answer
20 views
how to lower frequency of a soubd file in matlab?
I am new to programming and sound processing,
I have been using matlab to process sound files, a few of which are ultrasonic.
my sampling rate is 250kHz.
I Would like to be able to play the sounds at ...
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1answer
33 views
How do Anti-Aliasing-Filters in audio signal processing work?
I am a little confused about the use of anti-aliasing filters. As on how I am confused, please consider the following task:
Consider a simple sampling rate conversion system with a conversion rate ...
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2answers
28 views
What is Causing a Noise Signal in USB Audio Recording?
I am experiencing the following peculiar noise signal when recording from my Behringer Xenyx Control 2 over USB.
There is frequency at around 2293 hz as show:
And over time, it produces the ...
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28 views
Temporally locating beats in an audio file
Imagine you have a long recording of a guitar player playing an arpeggio at a slow speed. To simplify, assume the tempo is 60 bpm or lower. I want to annotate this recording with the exact moments ...
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How to use a context window to segment a whole log Mel-spectrogram (ensuring the same number of segments for all the audios)?
I have several audios with different duration. So I don't know how to ensure the same number N of segments of the audio. I'm trying to implement an existing paper, so it's said that first a Log Mel-...
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1answer
32 views
Failed to Get Silence in Audio Streaming
I want to detect silence in audio streaming. I've been following lot of answers from any website and i feel like know how to do that. but i doubt i'm on the right path coz the result doesn't seems ...
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1answer
29 views
Technique to detect pitch after analysing only one pitch period?
does anyone know if there is a technic to estimate the pitch of a given signal by analysing only one pitch period.
I'm asking because I read about a program called AudioMotors.
https://lesound.io/...
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20 views
What is the best input for de-noising autoencoder for sound data?
I am currently trying to build an autoencoder to de-noise audio data.
However I have not found any good articles explaining about the input to the autoencoder, i.e. feature vector.
As in speech ...
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0answers
47 views
polyBLAMP anti-aliasing in C++
How should 4 point polyBLAMP work. It does not really seem to dampen aliasing.
I have an floating point digital audio oscillator that makes a +1.0f jump at the end of every period.
My question is, ...
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24 views
Decoding data hidden using “echo hiding” technique
I'm working on a little project, which centers around steganography. As part of it, I want to implement the audio echo hiding method, but the problem is, I've never done anything related to digital ...
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1answer
24 views
Modulation scheme for a noisy channel with a complicated transfer function
I want to build an audio modem with a speaker in a room and a smartphone. The channel is quite bad:
Far field, so there is potentially significant reverb
Typical residential background noise
phone ...
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1answer
53 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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0answers
19 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
76 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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25 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
192 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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68 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
34 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
41 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
38 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 ...
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1answer
119 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
20 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
83 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
48 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
62 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 ...
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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
72 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
94 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
40 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
52 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
66 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
60 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
39 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
59 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
73 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
52 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
372 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
58 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
271 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
16 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
66 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
37 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 ...
