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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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Detecting and fixing clipped positive waveforms

I have a large number of audio files recorded using custom IOT devices. When analysing the data I noticed that a portion of them seems to have their positive amplitude ramp up, but the negative ...
zaza's user avatar
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0 votes
1 answer
58 views

Using DCT to create real-time "levels" animation from microphone input

For context: I'm trying to create a simple "level monitor" animation of audio data streaming from a microphone. I'm running this code on an iOS device and leaning heavily on the Accelerate ...
Joshua Sullivan's user avatar
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1 answer
42 views

How can I use a constant Q transform to detect low frequencies without making the buffer size too large?

I am trying to write code to detect musical notes within a signal. I have been reading up on different methods for extracting frequency information out of a signal to determine which notes were played ...
tjwrona's user avatar
  • 327
0 votes
1 answer
37 views

Removing a repeated sound from an audio file

I have a pre-recorded sound effect (FX) that will be played every 5 seconds. Together with that, I also play a music track for a few minutes. The result is recorded. Now, I want to remove the sound FX ...
dz902's user avatar
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1 vote
0 answers
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--Which dither to use for narrowband speech?

The tl;dr What dither algorithm should I use when converting 24/48 mono speech to 8/8? -- I'm truncating and downsampling a 24/48 mono audio source containing only speech to 8/8. I won't be doing any ...
justaboy's user avatar
1 vote
1 answer
46 views

Is it possible to reconstruct the continuous frequency spectra from DFT/FFT?

Given the output of some FFT (lets call it X) with N bins (lets say 2048) is it possible to recover the values in between the bins in the same way that I can recover the values in between samples of a ...
little_brindle_dog's user avatar
0 votes
2 answers
95 views

Audio delay with feedback

I'm writing a simple "audio delay effect", with feedback (the closer FEEDBACK from 1, the more the repetitions ...
logijaz's user avatar
2 votes
1 answer
43 views

Finding the right phase offsets for minimizing the highest absolute value of a sum of sinusoids

suppose I have a sum of cosine waves. Their frequencies are 500 Hz, 501 Hz, 502 Hz and so on, all the way up to 900 Hz. The important thing here is that the amplitude of the slowest one and of the ...
Petr's user avatar
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1 answer
40 views

What measures are used to identify a faulty speaker?

I'm working on building a mobile app that would play a sound via speaker, record itself and process this recording to identify faults within speakers, what meastures or parameters should be considered ...
Priyank Joshi's user avatar
0 votes
1 answer
36 views

Can a bad quality sound be measured via THD?

I'm trying to create an app that measures the audio quality, this is ideally to filter out blown out or cracking speaker who has different types of distortions, what would be the best measurement ...
Priyank Joshi's user avatar
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0 answers
28 views

Does the print of readframes(n) of wave library return the audio samples in hexadecimal form?

When I print the result of the first audio frame by doing track.readframes(1), I get b'\xfb\xff\xfb\xfe'. My track is stereo so we know that we have 2 channels where each channel has 1 sample. Does it ...
user71782's user avatar
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24 views

Program to compare output and input streams to verify volume

I'm trying to wright a program to compare, two sound signals, output (speakers) and input (mic) streams and verify/adjust the volume. I'm using Microsoft Visual Studio C# forms project with the ...
djangocoder5925's user avatar
0 votes
0 answers
28 views

Why are MFCCs of different signals equal?

What I am trying to do: I am trying to extract features based on the difference between a reference signal (white noise in my case, let's call it ref) and different processed versions of it (let's ...
benczejrobert's user avatar
0 votes
0 answers
11 views

Best practice for classification of parts of a sound file based on content

I have a collection of raw recordings. I also have a text file for each recording with the starts and ends of subparts of interest from the original recording. I want to train a model to classify ...
David Harar's user avatar
0 votes
1 answer
53 views

Minimizing maximum instantaneous magnitude by applying phase shift

Suppose I have the following sum, where all the input variables are positive real numbers while the output is complex-valued data $$a e^{i \left( x + y \right)} + a e^{i \left( x - y \right)} + a b e^...
5-limit_JI's user avatar
1 vote
1 answer
76 views

Zero crossing count vs zero crossing rate: does it make any difference?

I am trying to make a feedback howling detection program for an audio DSP chip I made a very crude one by counting zero crossings, doing something like this: ...
user13267's user avatar
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1 vote
0 answers
31 views

Bearing vibration analysis from audio signal instead of velocimeter measurements

We currently test bearings on a third-party test machine, which uses a velocimeter and reports through a software front end that outputs a low mid and high band spectrum result, and RMS, Peak and ...
DPC's user avatar
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2 votes
0 answers
48 views

Decoding images from sounds in Voyager golden records

I'm interested in a side learning project in decoding the images recorded as sounds in Voyager's golden discs (https://voyager.jpl.nasa.gov/golden-record/). The instructions to be able to decode these ...
rluque's user avatar
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0 votes
0 answers
61 views

writing from STFT to audio using ISTFT adding harmonics to audio not present before

I'm doing quite a bit of the samething as I wrote in this post here : Shifting frequency peaks such that w[i] moves to w[i/n] using STFT and IFFT I'll quote from it here to present the issue I now ...
Vrisk's user avatar
  • 103
2 votes
2 answers
371 views

Real-Time, efficient Loudness Normalization

I'm working on a Bluetooth speaker system where the same music file results in different SPL up to 10 dBA when played through different devices, such as iPhones and Android smartphones. The system ...
DSP novice's user avatar
2 votes
3 answers
509 views

How can the quality of a resampling algorithm be measured?

I need to write my own resampling algorithm to resample sample and impulse response wav files between 44.1, 48 and 96 kHz. I'm on an embedded platform, so I don't have access to the usual libraries. ...
Tobias Hienzsch's user avatar
0 votes
1 answer
45 views

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?

I have built a Bluetooth speaker using an ESP32. It can stream BLE audio and play MP3/WAV from an embedded SD card. I want to record 44.1Khz res audio from it but unfortunately the ESP32 is not ...
hrithik singla's user avatar
1 vote
0 answers
57 views

Normalized audio cross corelation

I'm trying to cross corelate two audio files to see where they line up. In scipy im doing: correlation = correlate(audio1, audio2, mode='full') Given the graph ...
Ethan Itovitch's user avatar
1 vote
1 answer
59 views

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

I am using FFTs to perform real-time convolution of audio signals on small embedded microcontrollers. I am thinking about testing different/smaller data types to represent the complex bins. I am ...
Tobias Hienzsch's user avatar
1 vote
3 answers
105 views

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

Say we have an audio signal and we do a Fourier transform with block size 2048. Now the outputs correspond to 1024 different frequencies. A spectrum analyzer however doesn't just display 1024 discrete ...
Potheker's user avatar
2 votes
0 answers
39 views

What is the cleanest, most transparent way to rapidly excite a resonant bandpass to a given amplitude (for musical synthesis)?

Background Music audio synthesis can be performed using underdamped resonant bandpasses as harmonic oscillators (set to harmonic frequencies of a given note). These can then be excited when input ...
mike's user avatar
  • 533
0 votes
0 answers
31 views

Differential value range reduction for Huffman encoding

I am trying to understand a piece of coding technology and place it in its context, but I am unable to find any literature on the subject. It would be much appreciated if someone could point me in the ...
Sano's user avatar
  • 1
0 votes
0 answers
22 views

frame size and hop size mismatch with aubio pitch detection algo

I am writing a script that detects the pitch of a short file as part of a larger project. I want to detect using an external call to sox and using the crepe, aubio, and librosa libraries. Aubio is ...
Rich Rath's user avatar
5 votes
3 answers
226 views

Difference between Level, Signal Strength, Amplitude and Volume?

It seems impossible to find a precise answer to this question. I researched for quite some time but different sources come up different definitions, that sometimes disagree or contradict with each ...
MusicQuestions7's user avatar
0 votes
0 answers
67 views

Pitch shifting effects on formants

I am doing a project for uni where i try to change the precieved gender of an audio using dsp techniques on matlab i started with pitch and am trying to move to formants now but i am not sure i fully ...
3izmanique's user avatar
2 votes
2 answers
130 views

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

SPEAKER AS RLC CIRCUIT I read this article here which demonstrates a simulation of a speaker as a simple RLC circuit where the RLC components are in parallel: MY GOAL I am interested in creating a ...
mike's user avatar
  • 533
1 vote
1 answer
209 views

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

I'm working on a vst plugin where a delay is used, the seek position of the delay sometimes changes gradually sometimes abruptly, when sudden changes occur sometimes a clicking sound is produced. Some ...
TiagoLr's user avatar
  • 111
5 votes
1 answer
188 views

"Tone Control" IIR filter coefficients

The documentation for Analog Device's sigmastudio shows some of the formulas behind their biquad IIR filter coefficients. The usual suspects (HP, LP, peaking, ...) as described in RBJ's cookbook are ...
playduck's user avatar
0 votes
0 answers
25 views

Musical Instrument Characterization

I am doing an undergraduate project where I want to first characterize an instrument based on its pitch and timbre and make an instrument recognition system afterward. Pitch characterization is pretty ...
Flash's user avatar
  • 1
5 votes
1 answer
72 views

What algorithms exist for dynamic audio normalization?

I'm manually editing the WAV files for an audiobook. There are often sections where I have to amplify a section, and then amplify sub-sections due to the variability of the quality of the original ...
Robert Rapplean's user avatar
2 votes
0 answers
57 views

Redirecting audio signals coming from a JACK audio connection kit client to MATLAB/Simulink in linux

Description of my scenario My main computer is connected through network as a JACK client to an embedded board working as master with a loopback connection via I2S to another board that acts as a ...
Gabriel Antonio Santos Baez's user avatar
1 vote
0 answers
90 views

In "Mel spectrogram" or "Mel filterbanks", what does Mel mean and why is it often capitalized? [closed]

In "Mel spectrogram" or "Mel filterbanks", what does Mel mean and why is it capitalized ? It doesn't seem to be the name of a person.
f10w's user avatar
  • 111
3 votes
1 answer
131 views

Audio EQ Cookbook Stability

I'm implementing 6 cascaded biquad peaking filters to make a 6-band EQ. In the system I want to calculate the coefficients on the fly (it's just gain thats changing, center frequency and BW aren't). I ...
Michael Beckwith's user avatar
2 votes
1 answer
91 views

low shelving filter frequency response diagnosis

I am building a second order modified biquad filter based on an implementation by Will Pirkle in "designing audio effect plugins in c++". However when I look at my implementation's ...
Brian Barry's user avatar
0 votes
0 answers
53 views

Mixing multiplied audio signals without clipping

I'm building a vocoder on an FPGA, and now trying to mix $N$ signals that are each the product of some carrier signal in one of $N$ bands and an envelope of a modulator in that same band (essentially ...
Andrew Li's user avatar
  • 101
0 votes
1 answer
55 views

Sound having lower rms values but higher loudness.What are the possible reasons?

I have some noise datas which are having lower RMS values. But don't know why they are very loud when I am normalizing and playing them. After normalizing the audio remain still low rms values. But ...
Bisnu Sarkar's user avatar
1 vote
0 answers
58 views

How to identify square wave in audio signal

I'm interested in identifying a square wave signal from recorded audio. This subject may be a complex problem so I would like to present the problem where I'm currently stuck at. I recorded the audio ...
Lim Meng Kiat's user avatar
0 votes
1 answer
98 views

What kind of distortion effect is this?

Here is the image of a plugin distortion I have applied to a sine wave. The sound characteristic is heavily distorted and digital but still good sounding bass wave. Any ideas what kind of distortion ...
TiagoLr's user avatar
  • 111
2 votes
2 answers
680 views

Algorithm to "serialize" impulse responses

New user to dsp.stackexchange here. I'd like to know what's a quick and effective way to take 2 impulse responses and create one that's equivalent to both of these "in series" for audio ...
delt's user avatar
  • 129
1 vote
0 answers
43 views

Is there an open source "Data over Voice" library?

I need to transfer data from A to B using audio as the medium, with a speaker at one end and a microphone at the other. Sometimes simply through the air, but it must also work through legacy phone ...
Björn Morén's user avatar
2 votes
0 answers
39 views

Expanding cross correlation DOA estimation to n microphones

I need to detect the Direction of Arrival of a speech signal using an 8 microphone array as part of a project I've been working on. Currently, I have a working solution in the 2 microphone case: $$G(\...
EPGzuz's user avatar
  • 21
2 votes
1 answer
59 views

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

(Asked on CrossValidated before I found this community) I have audio files and timestamped fragment transcriptions that I have force-aligned with third party software. Unfortunately the software ...
c0bra's user avatar
  • 123
1 vote
0 answers
36 views

Collecting average frequency data of a song/audio recording

I'm a university student, and for one of our classes we have to use TouchDesigner to make some sort of dynamic visualizers. I know how to do this already, but to make the process more scientific I ...
Hugh Mungus's user avatar
1 vote
1 answer
59 views

Mixer and control systems

A mixer in the time domain usually multiplies 2 signals of the time domain , however what does it do in the Laplace domain?Is there a equivalent block diagram of a mixer in the Laplace domain?The ...
Cerise's user avatar
  • 153
2 votes
0 answers
195 views

Detecting the piano notes using python

I am trying to build a small application that is very similar to simply piano, where I am trying to detect the piano notes, my program will be listening and when a key gets pressed it will estimate ...
Sanju Reddy's user avatar

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