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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.

0
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1answer
310 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
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2answers
291 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
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2answers
134 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 ...
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0answers
40 views

Q format for the calculation of second order statistics (power, correlation etc) in fixed point DSP

I am currently working on a fixed point DSP audio system which natively supports 24 and 32 bits. Accordingly, the Q format which I am using for the time domain audio data is Q1.23. Let's say there ...
3
votes
1answer
88 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 ...
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1answer
103 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
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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 $x(t)$...
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1answer
234 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
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1answer
71 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
686 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
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1answer
62 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
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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
122 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
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1answer
242 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
177 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
395 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
65 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
249 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 ...
0
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0answers
133 views

how to make exponential smoothing between transitions in signal?

I'm working on signal echo hiding ,For the watermark data embedding process, the original audio signal is divided into smaller sections in the time domain. By convolving each section with ...
0
votes
0answers
39 views

speech transient separation techniques?

I am trying to separate a noisy speech mixture. I did several tests, for example, try to filter the speech with random broadband noise, harmonic noise, reverb, etc, but I can't seem to tackle the ...
1
vote
1answer
129 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
381 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
32 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
164 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
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1answer
224 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 ...
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0answers
184 views

How to create a smooth graph/waveform from AudioSignal?

I'm pretty new to dsp programming so if my question is bad please forgive me ;) So i want to "design" a smooth waveform from my Audio Signal. Actuall it look like this: //Edit: I played a little bit ...
3
votes
2answers
508 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
444 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
646 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 ...
1
vote
1answer
707 views

How to find Leq for a WAV audio file like an SPL Meter would?

Given an audio file (WAV), I need to find the $\rm L_{A_{eq}}$ (continuous A-weighted equivalent sound pressure level) of the audio file. For this purpose, I am following this workflow: Get a frame ...
1
vote
1answer
52 views

Creating a Phantom Sound through 5.1.2 Speaker Configuration

I am trying to use a 5.1.2 speaker configuration (left front, center, right front, left rear, right rear, left front height, right front height and Low-Frequency) to create a phantom sound at a ...
0
votes
1answer
634 views

Compare two wave files

I have two .wav audio(stereo) files to be compared. The files may vary in frequency or amplitude or phase shift. How to compute the difference between the wave files in Scilab? The expected output is ...
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1answer
285 views

Comparison using cross correlation [closed]

I have 2 stereo audio files to compare. Comparison of files include change in frequency, amplitude, phase shift Code: ...
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3answers
331 views

Why is the first axis of STFT frequency bin

When taking the STFT of an audio signal, the first axis is the frequency bins by convention, and the second axis is how it elapses through time. Is there a reason?
3
votes
4answers
108 views

Can you recommend an efficient encoding algorithm for sound with long periods of silence?

As a side personal project, I'm coding an Android application that I plan to use to record the sounds of our first baby. I'm kind of new to the world of Android development and sound processing, so ...
2
votes
1answer
440 views

Analysing mp3 quality with spectrum, how to interpret the results?

When comparing spectrum images between two different audio sources in order to determine which has better quality, I sometimes come to an impasse. For example, consider the following images from two ...
4
votes
3answers
80 views

Classifing audio signals to detect fault

The basic idea i'm working on is automatic fault detection using audio signals captured from Motor. I have like set of sample audio signal which are recorded when there is no fault and with fault at ...
-1
votes
1answer
116 views

Save Matlab audio output [closed]

I need to save the audio output of a script in Matlab but I don't know how to do it. It is a complex sound, with several pauses of silence with different durations, so I cannot simply use audiowrite (...
5
votes
2answers
195 views

How can I measure the harmonicity of an audio spectrum?

I would like to estimate the "harmonicity" (harmonics-to-noise ratio ?) of an audio signal from its spectrum. What kind of algorithm can I use?
2
votes
1answer
83 views

How to recover signal repeated throughout other signal?

I have an audio signal that contains, at various time offsets, various other short audio signals, often repeated, in addition to other audio content. Here is an example of how that could look like: <...
0
votes
4answers
95 views

Signal and Space/Time

I have been reading a couple of definitions on signal such as: Any quantity measurable through time over space or any higher dimension can be taken as a signal. And this, In the physical ...
1
vote
1answer
56 views

How to extract a signal received from a given angle by two microphones

Please bear with me, as I'm only a programmer, and have no knowledge of signal processing as a discipline. I have an application which generates a short pulse of 12 kHz frequency sound from a single ...
1
vote
3answers
468 views

Avoiding clicks with changing biquad coefficients

I have a DSP application that filters a signal with five cascaded biquads whose coefficients are determined by the Audio EQ Cookbook formulae (Thanks for this work, Mr. Bristow-Johnson btw). Each ...
1
vote
0answers
98 views

Gain of resonating filters

I'm trying to make a music instrument based on a resonator with a high Q, such as Ableton's Corpus and Ableton's Resonator (I think the former is based on a simple 2nd order resonant filter while the ...
2
votes
2answers
549 views

Does a 20 kHz sine wave sampled at 44.1 kHz need to be reconstituted before playback?

I am interested in building a synthesizer but I am confused about the Nyquist-Shannon Theorem and sampling rates. If I sample a 20 kHz sine wave at 200 kHz and plot the data points, the plot looks ...
0
votes
0answers
68 views

Extracting curve of audio file's fft

I have an audio file whose fft looks like this. I want to get the values of this curve over time as in Which algorithms help me do this, which keywords am I looking for? So far, I applied a high ...
0
votes
1answer
555 views

How to overlap two RAW audio signal

I hope I can find some help here. I have started working on a little project concerning voice recordings. What I would like to is: given two audio files in RAW format (or WAV, as long as I can ...
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0answers
49 views

Real-time DAQ (Data AcQuisition) device and filtering of voltage signals in audio frequencies

I'm working on a frequency mixer project in which two analog voltage signals at two different frequencies (each <5 kHz) are considered. I'd like to add the two fundamental frequencies and return an ...