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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How to use deconvolution code with an impulse response to achieve the original signal

I am trying to write a code for my thesis to deconvolve a recording with an impulse response so that I can achieve the original audio signal. I have written a simple code so that I can implement this ...
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How do audio readers interpolate?

$124$ Hz sampled at $256$ Hz has "beats", but $128$ doesn't - a phenomenon of this question: Whether these are "legitimate" beats is up to interpretation, but suppose the goal is ...
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CMCIS library LMS does not converge

I have written this code with CMSIS DSP lms filter, but it does not converge, Ive checked input and desiered signal feed to filter but I think both are ok, both are in range of int32_t converted to ...
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2 votes
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Performing Sample-Rate Conversion on Subchunks of an Audio Signal

I have implemented sample-rate conversion (SRC) from scratch in my hobby music player, and it worked fine. However, originally, the music player loaded entire audio files into memory and performed the ...
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What could be the possible reason behind curved signal?

I have a MEMS microphone connected to the microcontroller on a PCB. The microphone collects the data and sends the analog data to the microcontroller in real-time. On the microcontroller, this data is ...
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Find offsets of two audio files with several sections in common

I have two audio files that have mostly identical* content, most of which is just people talking. A typical example would be two files, each approximately an hour long, in which ~55 minutes of the ...
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Negative SDR result for evaluating audio source separation

I'm trying to use eval_mus_track function of the museval package to evaluate my audio source separation model. The model I'm evaluating was trained to predict vocals and the results are similar to the ...
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Can I get audio version of UCSD dataset?

I request to please tell me the source through which I can access audio version of UCSD dataset. UCSD is an video anomaly based dataset. Kindly guide me in this respect. Regards,
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Is there a way to get the "mode" of several recordings? (to remove 1-off noise)

I've got several (4) not-great audio recordings of the same short (a few minutes long) performance, each taken from different microphones. I'd like one high-quality clip of the event. This is... ...
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How were the coefficients for polynomial linear predictive coding derived?

FLAC uses linear predictive coding (LPC) as one of its central compression steps. While it allows for arbitrary LPC coefficients in the "FIR Linear prediction" subframe compression type, ...
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Are common formats and encoders of lossy audio compression idempotent?

That an action is “idempotent” means if you do it several times, the effect is the same as if you do only once. Lossy audio compression by definition loses information. Even if you convert it back to ...
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What *is* pitch-shift?

As I understand it, pitch-shifting (without changing speed) and time-scaling (without changing pitch) are two sides of the same coin, because if we can do one we can get the other through resampling ...
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Signal flow for conversion of audio to Lzeq (sound pressure level time-equivalent)

I'm developing an embedded system which takes in digital audio from an ICS43432 MEMS-microphone via the I2S-bus to compute the sound pressure level over variable ...
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1 answer
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Signal to noise ratio for time series audio data

I am writing some software to benchmark an audio noise filtering method using SNR as an evaluation metric. One software component generates noisey audio by taking a clean audio file and a noise source ...
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2 votes
3 answers
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Delay/Lag calculation using cross correlation in frequency domain

I need to compute cross correlation of two signals using frequency domain. Hence, I am using the below equation for calculation. max(IFFT(FFT(a)*conj(FFT(b)))) ...
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Help on audio filter with FFT on python

I need to take a .wav audio file that's noisy and filter out some noise. I have to do it using Fourier Transform. After some days researching and experimenting, I finally made a working function, the ...
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1 answer
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Correct way to draw zoomable audio waveform

I am trying to implement smooth zoomable audio waveform but am puzzled with the correct approach to implement zoom. I searched internet but there is very little or no information. So here is what I ...
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How to get the high frequency portions of an audio using DWT?

I basically want to use a digital wavelet transform to get the high frequencies of an audio signal and get the time information or really anything that can make me understand where they happen so I ...
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Is it correct to say that increasing the sampling frequency is same as compressing the duration of an audio signal in MATLAB?

I am trying to read and play with an audio signal in MATLAB. I have coded below clear [y,fs]=audioread('guitartune.wav'); sound(y,fs) This audio clip'guitartune....
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Why when ploting the ICLabel in EEGLAB some of the IC topographs look exaclty the same?

I am attaching an image with the output of ICLabel. So you can see that IC5 and IC6 are the same. What does this mean? Thank you in advance:)
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2 votes
1 answer
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How long should be the Room Impulse Response

I am writing a code to calculate the reverberation time of a room. I am new to the topic. I used sine sweep to excite the room and the response was recorded. Then I got the room impulse response by ...
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How to detect acoustic and/or network degradations in speech as early as possible before automatic speech recognition?

I'd like to learn about techniques and approaches to identify speech degradations before they enter automatic speech recognition system and cause false recognitions. For example, I'd like to detect ...
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1 vote
3 answers
81 views

Why mixing two sines sounds like a triangle?

I'm not sure where to post this question that is related to audio synthesis through programmation. More specifically, I'm using a library called synthplayer that implement basic synthesizer in Python. ...
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Detecting whether the signal is increasing intensity or decreasing intensity

I am trying to determine whether the signal is increasing intensity, decreasing intensity, fade-in, fade-out or peak using an audio file. The graph has an x-axis with time and a y-axis with the data. ...
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3 votes
1 answer
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What do I measure in a Sound Sample Buffer to remove noise from an audio file using the Kalman Filter?

I am developing a computer program that removes or reduces the background noise from an audio file using the Simple Kalman Filter. I have implemented the Kalman Filter and a way of obtaining the "...
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Understanding conversion of frequency-warped signal and its orignal signal

I'm reading an article Frequency-Warped Signal Processing for Audio Applications which is talking about frequency warping. An open access version of this artical is available here. Say a signal or ...
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Modeling system with return by LMS adaption

I want to model the accoustic system which have some return from its input. I think return from this system will not damage LTI assignment. To identify system I'm using LMS adoptive filter, by ...
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How is the 'difference' of two audio tracks best computed, in order to maximise instrument isolation?

I'm hoping to improve the results of a simple application I've been working on, which aims to isolate as fully as possible the electric guitar heard (along with drums/bass/vocals) on a single track of ...
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1 vote
1 answer
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Detection of Loopable Audio sample

I have a bunch of samples some of which are loopable [1]. I'm searching for a methodology to programmatically detect them. Any paper or approach is welcomed. Thanks. [1] https://www.wikiwand.com/en/...
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4 votes
2 answers
147 views

Using MATLAB's invfreqz to fitting a audio frequency response, but the result is bad at low frequencies

I am a DSP beginner. I have some audio frequency response (actually acoustic, especially speaker frequency response by the way) curves and I'm trying to fitting them by IIR filters using MATLAB's ...
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1 vote
2 answers
68 views

Inverse STFT causing unwanted noise spikes?

I have an (audio) signal. I am applying STFT continuously and after applying a certain magnitude threshold I am doing inverse STFT to reconstruct the signal. But I am getting some unwanted mini noise ...
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3 votes
1 answer
124 views

Detecting background noise from audio time-freq domain analysis

I have a requirement to detect/reduce sidetalk/background noise in real-time audio. I am stuck in how can I detect this from audio time-frequency domain analysis. I am already getting the time-freq ...
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4 answers
141 views

do audio compression algorithms encode channels in parralel?

I'm looking to compress multiple mono sound tracks together, as one track (with loads of channels), however, I was wondering if this is stupid - because I may lose out on real compression, as these ...
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How to get frequencies from MFCC

So i recently learned how to graph a MFCC, but I am unable to understand what each axis means, and I know you will be like "coefficients", and "frames", but like coefficients of ...
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What's the state of the art way to detect / change audio pitch? [duplicate]

Detecting pitch: It seems that selecting the modal frequency in the frequency spectrum is too simple an algorithm for reliable results. For detecting, and changing, the pitch of a note, what are the ...
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BPSK based audio digital communication system

As part of my diploma thesis I'm implementing digital communication system based on audio transmission in free space (STMF401RE + analog microphone). I'm using BPSK modulation and I've implemented ...
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1 vote
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Confusion regarding STFT phase vocoder-based pitch shifting

I'm currently working on a phase vocoder implementation based on the Short-Time Fourier Transform; it's heavily based on the models described here and here. I have successfully completed the analysis ...
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4 votes
2 answers
269 views

How are sounds recorded and reproduced (played back)?

Someday I have suddenly got wondered how sounds are recorded and reproduced. I searched a little bit but the information I got was composed of difficult words and contents, so I couldn't understand ...
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1 vote
1 answer
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How fast is an histogram suppose to decay?

I am working on a software which receives streaming audio data, is expected to process that data on the fly, and then outputs the results. The module I'm working on is expected to adjust the audio ...
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1 vote
1 answer
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Effective bit depth of mixed signals

I have three signals of 16bit signed integers. Each signal represents a sine wave of unknown (and variable) frequency / phase in the audible range (20hz-20khz) and the amplitude is anywhere between <...
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Discrepancy in frequency response of biquad filter (in Python)

I have written a simple biquad filter class but when I check its frequency response, I get different results with different methods. If I check it "empirically" (by taking the FFT of its ...
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3 votes
1 answer
251 views

Prove that exponential sweep sine decay 6 dB per octave

Exponential sweep sine (ESS) is a good excitation signal to measure the impulse response of an acoustic system, such as loudspeaker - room - microphone system. An ESS has the form of $$ x(t) = \sin\...
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Extract amplitude values from a wav file

I would like to extract the amplitude value from a wav file. Using Python I can extract the raw data into an array. How do I go about extracting the amplitude value of each sample?
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1 vote
3 answers
96 views

Convert a list of values and times into an audio file

I have two pandas dataframe, one with time in seconds and one with the actual audio data (values from -1 to 1). The audio data points are not equidistant, this why I need the list of time in seconds. ...
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1 answer
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How can THD (Total Harmonic Distortion) be applied to compare between two signals ( for example between original and encoded signals)?

I would like to compare between two signals (original signal and its encoded version) and plot the scores) as we do with STOI and PESQ. should we calculate the THD for each single signal separately ...
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1 vote
1 answer
142 views

Which Audio Codec/Bitrate Relationship delivers best Audio Quality: 128 kbit/s MP3 vs. 192 kbit/s AAC vs. 64 kbit/s OPUS?

Assumed we have those three audio streams available for listening to: MP3 with a bitrate of 128 kbit/s ...
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What is the best approach to cancel realtime noise from a participant audio for conference purpose?

I am new to audio signal processing and I wanna do real-time noise cancelation for a participant in a conference. I am capturing audio at 16bit, 16khz, little-endian config. I tried to use the webRTC ...
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2 votes
1 answer
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How to implement Auto Gain Control (AGC) before and after mixng realtime audio for a conference?

I mix audio(16bit,16khz, little-endian, mono,frame-duration:60ms) packets after dividing the short values by participant count so it never gets clipped. but before that, I need to apply a certain gain ...
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1 vote
1 answer
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Did Dialup Modem use closed or open loop power/volume control? How did they determine Tx level?

I have been reading about dialup modems, but one thing I cannot seem to find out about is how implementers determined optimal transmit power/volume. Is this part of the echo cancelation framework? The ...
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0 votes
1 answer
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How do I get the waveform fom the soundcard in VB.NET?

I've written a VB.NET WinForms application that allows me to perform a Fourier Transform to find the amplitudes of discrete frequencies to represent that “strength” as a bar graph. The code for this ...
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