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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48 views

What are the known ways of changing an audio signal that are not audible to humans?

For instance, I believe we cannot detect global phase shifts. And of course frequencies above ~20kHz or below ~20Hz. I've heard of "frequency masking", but don't know much about it and would ...
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105 views

How to synchronise two audio files using a marker

I'm working on a project on which we use WebRTC and the RecordRTC library to record audio files. I would like to be able to process the audio files server-side to ensure a good synchronisation. To ...
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30 views

Diference between Specific,perceived and perceptual Loudness

i came across many terms describing loudness ! I understood perceived loudness which is what the user perceived and is related to the user. What is specific loudness and perceptual loudness? ...
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74 views

Frequency contour cross-correlation

I'm trying to segment (or extract) different calls from a spectrogram. My current method, calculates a threshold value (variance) and then using RMS can detect voiced/unvoiced signals: $$ Thres = (1 -...
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461 views

audio features extraction using restricted boltzmann machine

I want to extract Audio Features using RBM (Restricted Boltzmann Machine). For this, I am giving the spectrogram (PCA whitened) as an input to the RBM. For each audio file, The spectrogram is a matrix ...
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166 views

Multiresolution discrete wavelet transforms: First steps

I have been looking at trying to use wavelets for multichannel audio dynamic range processor. I am in the very early stages of trying to get decomposition and reconstruction working. In this thesis, ...
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79 views

Length of frames for calculating cepstral coefficients

For audio classification we are using the cepstral coefficients and their derivatives. Which I calculate using the Matlab libary voicebox. I am looking for some literature for how to choose the frame ...
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2k views

Demodulating FM from IQ Samples using Simulink

I am using simulink and have got IQ samples from an RTL-2832 / E4000 usb dongle. I would like to learn how to decode an FM station to audio and attempt this project on simulink. How do I proceed with ...
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16 views

How to design a low pass filter to convert PDM bitstreams into analog?

I have captured two bitstreams of PDM data into Excel. How do I design an appropriate low pass filter (in Matlab) that will convert the PDM bitstreams into analog form that shows amplitude?
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2answers
27 views

Audio keeps echoing continuously when using 2 separate devices for audio and recording on video calls

I had used 2 separate devices for a video call. Desktop for video and audio and phone for recording, since the desktop microphone is not working properly. Unfortunately when I started speaking, my ...
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58 views

How to make ultrasound audible?

Because of interesting spectral phenomena, I would like to shift the >20kHz content of my audio recordings to audible frequencies, so that I can listen to it. I tried the Audacity pitch correction ...
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30 views

Is it possible to open 2 streams from the same microphone on Oboe?

I'm trying to open 2 input streams from the same microphone on Oboe but it's not working. One of the audio inputs I'm opening is like this: ...
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35 views

What would an algorithm that can target different devices be called?

I'm developing an audio filtering algorithm that can be run in different scenarios, e.g as embedded firmware on DSP Codec/Core, as a software library, or as a cloud service. Effectively, it can target ...
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37 views

Reconstructing a signal from multiple filtered copies using overlapping tapered windows

I have an audio signal that I need to process and analyse in blocks. I've now decided to process it using overlapping non-symetric Hanning windows of length M = 9600 and a hopsize of R= M/2. I zero ...
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24 views

How do I change the sampling rate without changing the speed and pitch in Python?

I am building a voice recognition algorithm, and the use case specifies that it should be able to identify the targeted speaker from various audio sources, which means that it must be able to process ...
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1answer
24 views

Error introduced by smoothing filter at the end of a processing block

I'm running analysis on a multichannel audio signal and due to the size have decided to process it in blocks (my computer doesn't have enough memory to process it in one go). Some of the data produced ...
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13 views

What techniques may “State of the art iterative numerical solver for optimally smooth gain reduction” refer to?

What techniques may "State of the art iterative numerical solver for optimally smooth gain reduction" refer to? As given in a description for a particular limiter. But particularly, I've ...
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16 views

MPEG Audio Constant bit rate conversion

I am trying to convert few .wav files to .mp3 format The desired .mp3 format is : I tried with FFmpeg with this code : ...
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23 views

How to scale audio signal from PCM data to a dB SPL value?

I've got a set with MEMS microphones that measure audio signals at ultrasonic frequencies. The MEMS microphones have a PDM output which is then converted into PCM (this is necessary to allow for a ...
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40 views

Generating piano notes using the Karplus-Strong Algorithm

I am trying to get a piano A4(440 Hz) sound from the MATLAB. I have looked at this link https://www.mathworks.com/help/signal/examples/generating-guitar-chords-using-the-karplus-strong-algorithm.html ...
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28 views

Changing Coherence function from float to fixed point

I want to calculate the coherence between two audio signals. After referring a few websites, I found that the coherence is calculated by using the equation $(1)$ which is as follows $$\rm Coherence = \...
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1answer
41 views

Matching input buffer size to opus frame size

I am working on an audio effect that passes it's input though the opus codec. I want the effect to be a VST, so I don't have control of the input buffer size and the sample rate. However, opus ...
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34 views

How to correctly pass filter states from one processing block to the next

I'm processing a 4 channel audio signal and due to memory requirements need to process the signal in an arbritrary number of blocks. To ensure consistent filtering between blocks I've been told I need ...
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26 views

How to deal with different audio formats for audio classification?

I am working on an audio classification problem statement to classify between two audio classes. I have collected samples from jotform, they are providing audio widget to collect .wav audio but it ...
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1answer
66 views

Popping sound after using bandpass filter

I'm trying to create guitar Wah Wah effect (basically band pass filter with dynamic frequency range). I'm using Steinberg SDK for it. I refactored this code to work in real time and it looks like this:...
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28 views

Spectrogram parameters piano sound

I have to do 'inverse engineering' with piano sound and find sequence of frequencies in it. I have to use spectrogram() and adjust parameters but my problem is that I don't know how. I set window ...
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13 views

Masking threshold in psychoacoustics

I am reading the paper Adversarial Attacks Against Automatic Speech Recognition Systems via Psychoacoustic Hiding here. In Section III-F, they explained how they keep perturbation below the masking ...
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2answers
50 views

Conceptually confused by LPC for speech: Do we synthesize by the inverse filter (FIR)?

I've been following the process of using LPC to analyse the speech and then synthesize the speech by swapping out coefficients over time. I'm purely concerned with pitched vowels at this point. ...
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1answer
44 views

Phase difference calculation between two wave files

I am doing comparison between two wave files, How to calculate phase difference between those files? Second wave is generated by making some changes in the first one .I have tried calculating phase ...
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29 views

How can I detect the presence of echo in an audio?

The presence of echo during a phone call can be extremely irritating. Echo consists typically of the speakers sound being picked up by the microphone with a certain delay (10-200ms). One of the causes ...
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18 views

Can someone help me this db transform applied on melspectogram?

I have read some audio processing codes written in Python : ...
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33 views

Hide audio inside other audio using FFT

How can i hide audio inside other using FFT, I need to know how to begin and what to do ? the first image is for the first audio and the second which i want to hide
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11 views

Spectrogram vs MelSpectrogram

I am a university student in computer science and during a project, I run into the Mel Spectrogram. Doing some research, I found that the MelSpectogram is essentially a spectrogram where the distance ...
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15 views

Classifying audio files by their instruments

I have a small set of sound files where each sound file contains a note or chord played on some instrument. Some of those sound files may be generated by the same instrument and I want to detect that ...
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40 views

Remove aliasing caused by scanning a wave file at arbitrary playback rates

I am writing a sound generator that needs to scan through audio samples at an arbitrary rate. When the sample is played at it's original rate, it sounds fine. But when I change the playback speed, it ...
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22 views

Remove known sound from signal

I'm currently getting FFT values from the music playing from a user's PC. However, my application also generates noise which interferes with the FFT data as my app's sound is also coming out of the ...
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23 views

Transform-domain mixer for audio codec (e.g. Opus) using only addition and multiplication

If this is the wrong StackExchange forum, please tell me where else I should post this question. Is it possible to create a transform-domain mixer for compressed audio streams using only summation ...
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29 views

Detecting Vowels using formants in an audio file

I'm new to signal processing, but I have a project where I need to identify the start and end times of vowels that occur in an audio file (wav file of one sentence). From what I've read, vowels can be ...
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1answer
24 views

Ambient noise levels in desert and recording studio

Are there mathematical models which suggest which of the two - desert or recording studio - is likely to be quieter? I would think a recording studio would be quieter than a desert. Is there a ...
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1answer
32 views

How can I filter a noise with pattern type in audio?

I have a set of audio files rescued from a source of phonograph record like a gramophone. Very old family stuff records from my elders back in year 1946. A relative has digitalized these with a ...
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41 views

Simulating packet loss in audio

I have loaded a wav file into a Python Numpy array. I would like to "perturb the signal" such that the audio sounds akin to audio streamed over a poor internet connection. How can I mutate the ...
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18 views

Is audio normalisation required for a speaker recognition/speech classification task? How to do it?

Not entirely sure if this Stack or a ML-related stack would be a fit for this question, but I focus on the signal processing here on this question, so I decided this one I'm developing a speech ...
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55 views

What cause frequency change of the acoustic signal (3 kHz to 10 kHz)?

I am working on Acoustic Resonace testing in which I excite the sample and then record the sound (3 kHz to 10 kHz) by microphones. I do few times repetation of the same measurement in series (one ...
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39 views

How to measure quality of an audio/music recording compared to the original

I have 3 wav files: The original recording of a viola note A noisy re-recording of the original clip A different noisy re-recording of the original clip I'm looking for a way to show which of the ...
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1answer
48 views

pyo: Retrieve amplitudes of audio signal frequenciy ranges

I'm playing around with audio signal processing for the first time and just to start somewhere, I decided to use the Python library pyo. My goal is to analyze an audio input signal from the microphone ...
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30 views

Time Frequency bin

I am doing research in audio signal processing for DOA estimation. I came across the term Time-Frequency bin. I have gone through several articles but none of them specify a TF bin rather it specifies ...
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21 views

How can I calculate time gap in different waves

I want, calculate the time gap between two signals .but that signal has frequently changed the frequency range. I want to combine with these two signal including phase shift I have a circuit for ...
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1answer
52 views

How do I find the transfer function of a speaker by analysing the audio produced by it?

I have the original audio signal(MATLAB GENERATED) and the recorded audio signal (anechoic conditions). The setup consists of a DAC with a power amp plugged into the speaker. The audio is recorded via ...
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29 views

Audio in Image,Graph or WaveForm Representation

Our task is to feed the audio data into the deep learning tensor flow model in the form of image representation(or graphical,waveform). Question is what is the best way of representing audio in image ...
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2answers
180 views

How long does it take for snd_pcm_writei() (ALSA) to send data?

Using a pretty typical call to Linux's ALSA, snd_pcm_writei(pcm_handle, pcm_buffer, pcm_buffer_size) how do you know how long it should take to send this data? What ...