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.

Filter by
Sorted by
Tagged with
2 votes
0 answers
24 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
24 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
108 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
64 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
34 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
32 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
0 votes
0 answers
42 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
88 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
  • 101
2 votes
2 answers
659 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
31 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
28 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
47 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
33 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
34 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
  • 143
2 votes
0 answers
109 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
0 votes
0 answers
22 views

Independent Sound Generative Factors

I am currently in the process of exploring independent sound-generating factors (e.g., Timbre, Pitch, Amplitude, Spacialization, Effects, Instrumentation, etc.) and I'm totally new to this enterprise ...
Yusuf's user avatar
  • 1
1 vote
2 answers
107 views

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

I am recently working on the reproduction of the filtering effect of target play back devices, like phone, or speakers, etc using convolution techniques in MATLAB. I firstly created a function called &...
CPK's user avatar
  • 11
0 votes
2 answers
69 views

Reduce memory usage by bandpassing a signal

When recording an audio signal one can have a high sampling frequency. To reduce the amount of data that is stored, and thereby the power consumption, I head that it is possible to use a high-pass ...
3dSpatialUser's user avatar
1 vote
0 answers
159 views

How to implement STFT for processing audio [closed]

I should start this off by saying I’m a hobbyist and by no means a student. I’ve been reading “The Audio Programming Book”, and attempting to implement STFT on the stm32 based Daisy platform. In order ...
Daniel Lawler's user avatar
0 votes
0 answers
52 views

How do I eliminate background noise?

After I use deep learning algorithm to enhance the speech, the speech will still have a weak background noise.The background noise of this audio has little effect on the calculation of SNR, but it ...
Killuaisaack's user avatar
0 votes
0 answers
75 views

Real time variable speed audio playback with interpolation

I'm trying to correctly implement the real-time variable speed playback of my buffer, to achieve the pitch shifting. I don't need to stretch to preserve the original length. Buffer is calculated with ...
coldmind's user avatar
  • 101
2 votes
0 answers
61 views

How to retune this mixture of 2 close-frequency sinusoids?

I have a signal which is a mixture of 2 close-frequency sinusoids, something like 1320 Hz and 1325 Hz, with an amplitude envelope which is typical for a musical instrument (ADSR). $$s(t) = a_1(t) \sin(...
g6kxjv1ozn's user avatar
0 votes
1 answer
55 views

How can I use interpolation to align audio [duplicate]

I notice that when I connect two audio clips, there is a jump at the junction. How do I use interpolation to connect these two pieces of audio? Thanks for the answers, I've got an answer from my ...
Killuaisaack's user avatar
0 votes
1 answer
58 views

What books describe convolutional neural networks for DSP?

(note: I read the site policy and meta questions about whether such a question is ok. I will be specific) I am looking for books that specifically cover using convolutional neural networks (CNN) for ...
HonestMath's user avatar
1 vote
1 answer
113 views

How to merge audio better?

I try to use a denoising algorithm to process an audio in streaming mode. What I've done is spliting the audio into 0.5s long segments and use denoising algorithm respectively.Then use np.cat or np....
Killuaisaack's user avatar
0 votes
1 answer
73 views

Fixed-point FFT and numeric overflow

I'm working on porting Opus IMDCT/FFT to a fixed point 16-bit DSP with a 48-bit accumulator. I'm trying to do that with 32-bit fixed point numbers to keep the highest possible quality. I'm basing my ...
Giovanni Bajo's user avatar
0 votes
0 answers
44 views

How to detect known audio samples within a stream of audio

If I've got a bunch of short .wav files, 16 bit linear pcm, and an incoming stream of audio that's been run through a voice codec (g.722, g.711, or others) and then converted back to pcm, what's the ...
Matt DiMeo's user avatar
1 vote
2 answers
105 views

Averaging a Signal with a delayed version of itself

I am working on a problem that requires interpolating between two music files. One music file is very similar to a delayed version of the other. I would like the 'average' of the two signals to sound ...
Mason Wang's user avatar
2 votes
1 answer
148 views

BPM Detection using Matlab xcorr function

Objective: Finding the BPM of a short drum loop. Approach: ...
ChalupaBatmac's user avatar
2 votes
1 answer
69 views

What does the frequency of oscillating microphone feedback depend on?

Pointing a microphone to a connected loudspeaker creates a feedback path and usually results in acoustic resonance, which manifests in a sustained oscillation: https://youtu.be/_XTtjZ8aZbc. A similar ...
Theodote's user avatar
0 votes
1 answer
325 views

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

I am trying to understand the spectrum of the sound of a cuckoo. It is 13-second audio clip and its download link is https://file.io/d4nqddrb1NNY Actually, I have a confusion, I have attached a snap ...
DSP_CS's user avatar
  • 1,850
0 votes
0 answers
31 views

How do you handle zeros when logarithmizing? [duplicate]

I'm trying to implement a procedure from a paper. There the authors use two equations to transform a fft transformed music audio file to a cent-spectogram. Equation 1: $$ f_{cent} = 1200 \log_{2} \...
Rocket's user avatar
  • 25
1 vote
0 answers
33 views

Loss of Resolution When Shifting Up With FFT / Phase Vocoder Pitch Shifter (Compared to Shifting Down)

I am working on a project writing a realtime FFT / phase vocoder based pitch shifting algorithm in C++. In development, I notice that shifting down has much better audible pitch tracking and ...
Emmett Palaima's user avatar
1 vote
1 answer
59 views

Adding mono sources to Ambisonic tracks

I have an ambisonic multitrack with several spatialized sound effects (I'm using 1st order ambisonics). However, I'd like to add some mono sources (e.g. narration) to my project. My initial thought ...
Sergio's user avatar
  • 111
2 votes
1 answer
78 views

How to simplify several cross correlations used for feature extraction

I have a system where I cross correlate recorded audio signals with several smaller generated signals to extract feature vectors that later I use in a kNN matching. The system works the way I want it ...
besabestin's user avatar
2 votes
2 answers
75 views

How to Dereverberate Speech taken in an Auditorium with Reverberation Time of 3.8 to 4 seconds

I'd like to cancel echoes from a Talk recorded in a large extremely reverberant auditorium. It's unintelligible as recorded, and I'm hoping to make it intelligible by echo cancellation. Audio was ...
philwalk's user avatar
  • 121
1 vote
1 answer
60 views

Measure of how well sounds fit together in a composition

Are there any approaches to gauge how well a sound may sit together with other sounds in a mix? I’m thinking about something similar to picking instruments that might go well together in a band. ...
Bjorn Thor Jonsson's user avatar
5 votes
4 answers
2k views

What is the audible level for digital audio dB units?

What is the audible level for digital audio dB units? I was looking at a soundwave from an mp3 file. I am using the "Sonic Visualizer" app available on Linux. The amplitude is between ~ -0.1,...
user2183336's user avatar
2 votes
1 answer
41 views

Dominant Speaker Selection in a Proximity Cluster of Smartphones with Arbitrary Positions (Not static)

I am working on an algorithm for dominant speaker selection in a closed proximity cluster of smartphones (iOS and Android) with arbitrary positions. The smartphones act as clients, and one of them ...
Khubaib Ahmad's user avatar
1 vote
2 answers
137 views

IIR/FIR equations for custom frequency response

Given a frequency response plot where gain is 0dB up to Fc and after Fc gain increases linearly by factor K (example chart below): I am seeking how to arrive at its IIR/FIR equations so it can be ...
JCFS's user avatar
  • 11
1 vote
0 answers
90 views

Properly FM Demodulation & Display baseband FFT in QT using QTCustomPlots

I am having trouble trying to display the baseband signal because it looks incorrect. How do I properly display the baseband signal? Should I be doing a float to complex transformation? Am I doing ...
TheDaChicken's user avatar
1 vote
1 answer
62 views

How should I go about calculating sines or cosines without the errors introduced solely by the transcendence of pi?

This may be irrelevant for many people but I'm asking this question because I don't need to do these things in real time, I don't care if modifying a 1-minute sound recording takes 5 minutes to ...
5-limit_JI's user avatar
1 vote
0 answers
51 views

What can I do to make synthesized speech segments add up to an exact duration?

I'm implementing a project which interacts with a Text To Speech library. It generates speech segments and then returns the waveform, which can be manipulated/saved according to my needs. Those ...
gabriel_tiso's user avatar
1 vote
1 answer
63 views

Handling asynchronous MCU I/O peripherals

Technical Background I'm making an audio processing board with an onboard microcontroller that takes in data via I2S from an ADC, applies digital filters, and outputs a stream of the same samplerate ...
sevenskiesrain's user avatar
1 vote
1 answer
40 views

What is the rationale for division by 8 during ADPCM audio compression?

I am studying ADPCM and while I understand the concept I cannot find an explanation for some of the calculations in available implementations such as the one outlined here: ...
banus32's user avatar
  • 11
0 votes
0 answers
55 views

Half-band FIR filter is not filtering as expected

I recently discovered that one may be able to further optimize an FIR filter processing time by skipping calculations of coefficients whose values are zero, providing that the filter is about halving ...
aybe's user avatar
  • 284
0 votes
1 answer
70 views

Why does extending the time domain of a signal with a fft introduce a fade out and fade in?

I was playing around with FFTs and decided to try extending the time domain of the original signal. I did this through taking an FFT to the frequency domain, doubling the amount of samples by ...
Rain's user avatar
  • 123
2 votes
1 answer
132 views

Frequency response measurements by audio signal FFT

I am trying to measure the frequency response of different audio canals, e.g. tubes, hollow pipes etc. My experimental setup looks like this: I generate an audio signal with a loudspeaker (Visaton WS ...
Scaty's user avatar
  • 23
1 vote
1 answer
109 views

STM32H7 I2S to TLV320DAC32 Setup Issues

I designed a board with an STM32H7, an ADC, and a TLV320DAC32. I'm in the stages of just getting a straight audio passthrough working. I am now getting signal output but it's all jumbled, and not at ...
Michael Beckwith's user avatar
0 votes
0 answers
63 views

For PCM audio samples, can values like max RMS and frequency magnitude be estimated?

I am using OpenAL and a loopback driver to grab PCM samples of whatever audio is being played on the computer in realtime. The PCM data is 1024 samples of 16 bytes (C# ...
McGuireV10's user avatar

1
2 3 4 5
32