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.
2
votes
0answers
107 views
Speaker diarization tool for acoustic analysis
I'm researching certain acoustic prosodic features during live clinical interactions. Because of the setting, the recordings always have cross channel bleed, and in some cases everything is recorded ...
2
votes
1answer
219 views
Making audio clips comparable in terms of RMS
I have a challenging question on how to normalize several audio clips so that they are comparable (similar) in RMS but without the clips clipping (having values over -1 and 1). Is there a systematic ...
5
votes
5answers
233 views
Remove a known wav file from recorded file
I looked at many similar questions but still cannot find an answer to my problem.
I'm playing an audio file N, while recording a man talking (b).
The problem is that the microphone gets the played ...
0
votes
1answer
212 views
File size adjustment using FFT
I have been given the task of doing audio compression using FFT.
The assignment is below.
Below is my MATLAB code:
...
-1
votes
1answer
376 views
How to detect and remove ringback-tone and ivrs voice etc from the beginning of an audio call recording
In an audio recording (say a telephone conv. b/w two people), how would I programatically detect and remove the dial-tone at the beginning of a call using python.
Ex :
As you can see the first 15 ...
1
vote
1answer
532 views
Digital 16-bit PCM audio signal scaling and STFT spectogram amplitude
I'm sampling a 16-bit PCM audio signal at 44100 Hz and I'm trying to understand the amplitude level of both the digital audio signal and its corresponding spectrum. The final goal is creating audio ...
2
votes
5answers
539 views
Interpreting microphone output signal
I have a question about interpretation of the signal produced by a microphone.
The output of a microphone is a voltage level that corresponds to the pressure on the membrane. The part that confuses ...
2
votes
0answers
113 views
Room Impulse Response Inverse Problem
This article here talks about a room impulse response generator which takes the source and destination coordinates, sound speed, sampling rate, room dimension and wall reflection coefficients as input ...
4
votes
2answers
277 views
Relation between samplingrate and frequency
I am working on Fourier Transformation, and applying this for recognizing an audioclip.
I have a 9 second long audio clip of a guitar strumming an A-Minor.
The audioclip has a sampling rate of 44100 ...
2
votes
2answers
174 views
Measuring vibrato rate of a sung note
I'm doing a project where I need to measure the rate of vibrato cycles in a sung note under various conditions. The stimulus will be a several-second-long audio sample of a held a capella note with ...
0
votes
0answers
85 views
Approach to identify someone knocking on a door - preferably not using a neural network
Say I wanted to identify someone knocking at my door using a microphone. I have (or at least I can make) some pretty good examples of what the signal looks like, but of course it will vary from person ...
0
votes
1answer
296 views
How to change x-axis from samples to time?
I need to change the x-axis on this graph from 'samples' to time:
I created the graph using this process:
...
0
votes
2answers
111 views
How to label sound events in noisy .wav files
I'm trying to find a way of automatically identifying birdsong in phone-recorded sound files (1min long). Currently the algorithm I'm using doesn't label all of the events I want it to. (it is ...
2
votes
3answers
75 views
How can a changing signal be Fourier transformed?
Fourier transformation is used to split a periodic signal into frequencies, and calculate the phase and shift for each frequency. But what if we would like to record an audio, which is changing, for ...
0
votes
0answers
42 views
Confusion related to Room Impulse Response equation in a research paper
This paper presents a toolbox to simulate Room Impulse Responses (RIRs) based on Image Source Method (ISM). The equation (1) has a windowed sinc function in it, which is given in equation (2). I was ...
0
votes
1answer
280 views
Best beginner DSP book for developing plugins
I'm a computer science student and I love producing music and I really want to get into developing audio plugins. I'm currently learning the JUCE framework and am fairly comfortable with C++. I'm also ...
1
vote
2answers
136 views
Remove audible click sound from high frequency hopped sound
I have a total sound duration of $T$.
I break the duration to several sub-durations and in each sub-duration, I generate several discrete sinusoidal values by the formula $\sin(2\pi ft)$ with a ...
14
votes
2answers
310 views
When does the laugh track kick in?
One of my weekend projects has brought me into the deep waters of signal processing. As with all my code projects that require some heavy-duty math, I'm more than happy to tinker my way to a solution ...
-1
votes
1answer
141 views
Synchronizing 2 seperate speech audio signals
I have two speech signals: one with a man reading a passage, and one with a woman reading the same passage.
I want to synchronize these signals so that, for each word, their readings line up. The ...
0
votes
1answer
54 views
Computing real signal with minimum absolute values from even magnitude spectrum
I want to derive a real audio signal from an arbitrary even magnitude spectrum.
The phase spectrum affects the values of the signal in the time domain; for example, a phase of 0 for all frequencies ...
1
vote
2answers
214 views
How to deal with audio with different length in Audio Classification
Currently, I am trying to work with the Dataset UrbanSound8K to try some Audio classification. And I got stuck in the preprocessing step already.
Since the audios are of different lengths, like 4 ...
1
vote
3answers
226 views
Audio wave amplitude decreased after FIR filter
I am doing sampling and filtering for a 17KHz sine audio wave signal, the Nyquist frequency is about 20.8889KHz. I found the fact that this audio signal's amplitude decreased dramatically after it ...
0
votes
1answer
48 views
Smoothing a Signal which contain White Noise
I'm trying to smooth a signal in order to reduce white noise as a result of changes in amplitude across sequential frames of an audio signal.
I've been investigating using a moving average or ...
0
votes
1answer
357 views
Converting Google's text-to-speech voice to, say, Justin Bieber's voice
We are trying to convert text to an artist's voice. For example, we want input text in Justin Bieber's voice. We are converting text to mp3 files in Python using gTTS (Google Text-to-Speech). We can ...
-1
votes
2answers
333 views
What is the amplitude range of a spectrogram?
Is there an upper and lower bound to a spectrogram representation of an audio signal? The range of the amplitude values of the audio signal is from 0 to 1.
2
votes
2answers
138 views
How to detect continuous noise in audio call?
We want to detect bad audio quality in audio call.
Should we extract this information from amplitude or frequency spectrum ?
Is there any technique/transform/filter available for this ?
Example:
I ...
3
votes
1answer
94 views
Finding good feedback coefficients for basic reverb
I'm currently working on some reverb algorithm code for the Gamebody Advance to get a bit of that ambient feeling for the music instead of having plain dull sound. A few challanging restrictions are ...
0
votes
1answer
139 views
Frame overlapping
I 'm reading the following article from NVIDIA which involves some audio processing a the beginning : link just here (part 3.2)
I don't understand the following part :
In practice, we use 520ms ...
0
votes
1answer
65 views
Extract same in phase signal from 3 different sources with shared delayed noise?
If I have 3 signals coming from 3 microphones with noise $n$ delayed like this:
$S_1(t) = x(t) + n(t+\phi_1)$
$S_2(t) = x(t) + n(t+\phi_2)$
$S_3(t) = x(t) + n(t+\phi_3)$
Their shared feature $x(t)$...
0
votes
1answer
300 views
Normalizing audio waveforms code implementation (Peak, RMS)
I have some audio data (array of floats) which I use to plot a simple
waveform.
When plotted, the waveform doesn't max out at the edges.
No problem - the data just needs to be normalized. I iterate ...
0
votes
1answer
74 views
Phase distortion of frequency warping by using Praat
I am trying to do speaker conversion by frequency warping method. I modify the function Spectrum_shiftFrequencies() in Praat by scaling the FFT spectrum of a voice. When one scaling factor is ...
0
votes
1answer
754 views
Delay-and-sum beamforming: implementing steering in C
I am currently trying to understand how delay-and-sum beamforming works.
I have found this website which explains the process of this technique very well in which the following C code is provided:
<...
0
votes
1answer
63 views
Trouble understanding an article about MFCC
I'm willing to apply the same audio preparation that this article page 3.
However, I don't understand the final part :
...
0
votes
1answer
49 views
Does the audio signal in the time domain change much after a RMS from one person to another?
I would like to train a deep neural network to perform lip syncing given an audio input like in this article.
I want to train my neural network only on Obama speeches like done in the article and ...
1
vote
1answer
135 views
MPEG Layer I: use of scalefactor and quantizers
I'm studying MPEG compression on the book "Introduction to Data Compression" by Khalid Sayood.
In the figure there is a paragraph piece about the subject.
I didn't understand a few things:
use and ...
0
votes
1answer
266 views
Dynamic Time Warp through SPTK tool
I'm an IT student and got an assignment to do about Dynamic Time Warping(DTW) using the Speech Signal Processing Toolkit (SPTK) and comparing some words spoken by 2 speakers and finding the ...
1
vote
0answers
196 views
What is the algorithmic latency of STFT processing?
I was reading this paper, and in section 3 I got a bit confused. It claims that the algorithmic latency is equal to synthesis frame length. If the overlap between two consecutive frames is 50%, the ...
0
votes
1answer
435 views
Measure the amount of drift in video and audio frames
I setup a stereo system using 2 GoPro Hero Session Cameras, since I need a larger baseline than the one of the Dual Hero System.
This unfortunately means the two cameras are not ...
1
vote
1answer
34 views
Why does this video's audio change depending on the playing device?
Yesterday I noticed that if I played this video from my smartphone on speaker, one could barely hear anything. On the other hand, if I used earphones I could hear it perfectly.
Then, I tried with ...
2
votes
1answer
68 views
What techniques/algorithms can I use to reduce noise when recording audio?
I'm making a handmade guitar chords dataset by recording 20480 samples of a 44.1kHz microphone to later on make a chord classification using CNN.
But there's a problem... I'm making this dataset on my ...
0
votes
1answer
277 views
Filter problems in python
I am recording a 30 second 1000Hz Sine wave, saving it to a list, splitting the list into 5 second sublists and I would like to apply a notch filter to each sublist to check when the signal drops out ...
5
votes
1answer
2k views
Signal filtering using Python
I'm trying to create an application using python that is capable of recording an audio signal and detecting short glitches in the signal. For example, I could be recording a 1000Hz sine wave and ...
1
vote
1answer
133 views
unsigned audio formats, does DC offset matter?
I'm a little confused about DC offset in audio files. If I have a signal that's not using the entire dynamic range, does the DC offset matter at all, or does it need to be centered around some zero ...
2
votes
2answers
425 views
Room impulse response 'noise'
I am working on getting room impulse response(RIR) and the first order reflections of a room.
For now I am stuck at getting proper RIR.
I followed Joseph's work and implemented on my own.
My setup ...
1
vote
1answer
33 views
Is it necessary to scale a signal for quantization noise analysis?
I am doing a series of tests to analyse the round-off (quantization) noise performance of single and two stage biquad cascades (direct form 1), at different sampling rates. The system I am simulating ...
0
votes
1answer
183 views
CODECS used for VoIP calls
I am trying to self-study audio codecs and I am trying to understand the difference between CODECS that come under AMR-WB, SuperWB and EVS vs. AAC-LD (MPEG-4). When is the former used and when is the ...
1
vote
1answer
298 views
Filtering Wind Noise From Audio Signal
I have an audio signal where most of the signal of interest is above 400 Hz but it is corrupted by heavy wind noise. What type of filter would you recommend to try to remove the wind noise considering ...
3
votes
2answers
564 views
Generating inaudible sound waves using MATLAB
I want to generate inaudible sound waves (above 18kHz) with 48kHz sampling rate using MATLAB.
Generating a sound wave with 18kHz frequency makes it inaudible.
I've inserted the data in the 18kHz ...
1
vote
2answers
538 views
how to use image filters for spectrograms?
Using MATLAB, I can import an audio signal and use some FFT function to get the spectrogram into the workspace. My question from here, is how to then process this spectrogram with image filtering ...
2
votes
0answers
722 views
What does Fast and Slow Time Weighting signify for real-time audio signal processing?
I have a streaming audio signal that I am recording in real time. To make the explanation easy, say the signal that I receive is calibrated and has a sampling frequency, $F_s = 44100\textrm{ Hz}$.
I ...
