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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21 views
How to convert (I2S Mode) data to “PCM Audio Waveform data”? [on hold]
I'm reading and writing by I2S mode.
For the Real-time sound pre-processing and recognition in ccs,
By selecting just one of the inputs for Left and another for Right (in a single-ended mode), I can ...
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0answers
14 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
31 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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0answers
19 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
22 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 ...
0
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1answer
44 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
16 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
70 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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0answers
16 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
186 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 ...
2
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0answers
65 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
33 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
39 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 ...
1
vote
1answer
37 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 ...
0
votes
1answer
105 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
19 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
82 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
47 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
60 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 ...
17
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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.
...
1
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2answers
71 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
87 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
36 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
48 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.
...
-1
votes
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
58 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
35 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 ...
-1
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0answers
40 views
Are there existing algorithms to detect reverb in a voice recording?
I want to be able to detect if a spoken voice recording happened in a room with a lot of reverb or not. I don't need to be able to extract the exact reverb profile of the room. Ideally the algorithm I ...
0
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1answer
46 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 ...
-1
votes
0answers
36 views
Librosa onset detection detecting too many onsets
So I was trying to use librosa's onset detect function to detect onsets. But on trying it out, this function is returning a lot more onsets than there should be for my audio file. I've included my ...
-1
votes
0answers
14 views
inaudible part of signal in MFC transform
MFC transform eventually captures a good portion of the humanly intelligible part of audio signals. I was wondering at which steps of the transform is the inaudible part of the signal rejected? Is it ...
-1
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0answers
21 views
Audio signal for each video frame
I have an mp4 video. Is there any way to extract the amplitudes of all the audio signals for each video frame in the available video?
That is, suppose there are 1000 audio samples corresponding to ...
-1
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1answer
41 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
193 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
53 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 ...
4
votes
2answers
266 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. ...
0
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0answers
27 views
Sound recognition within an audio signal
I'm trying to do some tests in Android. The phone's microphone is continuously listening, and whenever a balloon is blown, I would like to detect it. There are of course other ambient sounds/noises.
...
-1
votes
1answer
15 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 ...
0
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1answer
57 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 ...
0
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1answer
32 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
votes
2answers
102 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 ...
0
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2answers
82 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, ...
1
vote
1answer
66 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 ...
0
votes
1answer
93 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 ...
0
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1answer
51 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
41 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 ...
0
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1answer
62 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 ...
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0answers
30 views
Make an envelope smoother but keep the sharp attack
I have a signal x for which I would like the envelope. I have taken the Hilbert transform (like in this question):
...
