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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.

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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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2answers
57 views

Multiple different length Windows for high resolution realtime FFT

I am looking at trying to achieve a 'realtime' (as quick as possible to acquire, process and present data) FFT application to analyse audio. I have setup my app to acquire a set of samples, apply a ...
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1answer
110 views

how to create a harmonic mask from fundamental frequency?

Using an algorithm that gives the fundamental frequency across time, example https://www.mathworks.com/help/audio/ref/pitch.html (or some other one?) how would I then use that, to create a harmonic ...
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1answer
41 views

How to recognize for small video sequence with a large sequence?

I'm working on the idea of recognizing a small video sequence (a sequence of 18 seconds) within a long one (a sequence of 15 minutes), the first idea that come to mind stats that I have to ...
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0answers
25 views

Python scipy firwin2 has a low frequency error when designing arbitrary amplitude filter

I've been trying to design an arbitrary amplitude FIR filter which has a specific attenuation for each frequency. However I get an error at low frequency when comparing the results to my expected ...
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1answer
170 views

Noise removal using adaptive noise cancellation algorithms in real time systems

I am newbie in signal processing. I want to remove noise from real time audio signal using adaptive noise cancellation(LMS&RLS) algorithms. I don't know how to choose reference signal(noise signal)...
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1answer
92 views

How can you 'perfectly' reconstruct phase for transients in a modified audio absolute spectrogram?

I am currently working on the absolute spectrogram of an input audio signal. I am using non-negative matrix factorization to separate out the rhythmic components (bass drum, snare drum and hi-hat) of ...
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2answers
86 views

Modeling the human perception of noise

Add together some sinusoids with irrational or "nearly" irrational frequency ratios, and you get something pseudoperiodic. If there aren't too many, you can hear that it has only pitched components, ...
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1answer
26 views

Bandpass filter for audio wav file

How do I apply a bandpass filter on an audio file which is in wav format? And write back the filtered data in a new wav file for further analysis.
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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 ...
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1answer
211 views

What is the proper way to implement a real time spectrogram?

I am trying to implement a real time spectrogram and I am not sure that I am implementing it the right way. First I decided to use Matlab, I was taking a short recording of lets say 0.1 seconds and ...
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3answers
59 views

Significance of signal energy in audio?

Probably a dumb question but I can't find the answer anywhere. It seems to me that the energy carried by a sound wave should be proportional to frequency squared. For example, the waves $x(t)=\sin(t)$,...
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1answer
59 views

Change durration/tempo of a .Wav file without introducing pops or pitch shift?

I am attempting to develop a similar function as Audacity's Effects>Change Tempo effect. Or more specifically: I am in need of a way to create a funciton in my ...
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3answers
55 views

How can I convert audio to a series of notes using pitch detection?

I have a good pitch detection system set up, and I would like to return a series of notes given an array of audio samples. My current approach is as follows: I have a moving window across the audio ...
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1answer
420 views

Bandpass filtering an audio signal with MATLAB's $\tt kaiserord$: unknown new frequency component

I am trying to filter an audio signal with kaiserord in MATLAB, but I'm having a problem. Here is my code: ...
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1answer
23 views

Reading a Wav File

I need to read two .wav files. One of them is a song, and the other one is impulse response. Then, I have to convolve them. In fact, I have read them by using ...
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1answer
875 views

Are there any realtime voice activity detection (VAD) implementations available?

I'm looking for a realtime VAD implementation like Discord/Mumble/Teamspeak have. My goal is to have a system that let's me use the same VAD for all my VoIP applications, as well as something I can ...
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3answers
535 views

Distribution of energy in frequency bands

On my wav-file (music), I do a short-time Fourier transform (STFT). I have spectrum, which spans from $0\textrm{ Hz}$ to $f_s/2$. I took out the range between $1$ and $1000\textrm{ Hz}$, which I ...
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1answer
272 views

How to make sense of audio file's $\textrm{dBFS}$?

I'm trying to build a song visualizer using the FFT method. When I analyze the audio, I get the $\textrm{dBFS}$ value for each frequency range. From my understanding, $\textrm{dBFS}$ is in the range ...
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0answers
13 views

Interpreting results of multiplying signal and wavelets in the frequency domain

I have followed a lot of stuff on mike cohens website and have managed to make a filterbank of Morlet wavelets and convolve with an incoming block of samples. This of course is a computationally heave ...
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0answers
34 views

Speech processing using matlab

My teacher gave me a project speech processing using matlab. I've completed up to point 6 but I do not know anything about points 7 and 8. Is there anyone who can help me please
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1answer
40 views

“Tail-like” response from digital microphone

I was doing some experiments with audio and wish to get "raw and unprocessed" data, and noticed that all the files had the following profile (just blowing into the mics): Speaking to the microphone ...
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0answers
32 views

Help in identifying low speed image transmission protocol over RF

In the seventies I worked for Muirhead on press fax machines that used to transmit pictures over radio and landline as I recall two FM systems were used a carrier 800 Hz modulated +- 150 Hz and 1900Hz ...
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1answer
105 views

Polyphase decimation filter plot in Octave

I'm using de Soras hiir library for decimation and I'm trying to verify the filter design in Octave. The filter is a polyphase 2x downsampling filter with 96dB attenuation and 0.01 transition band, ...
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4answers
111 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 ...
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1answer
27 views

Similarity in signal plots but dissimilarity in sensed content

I have two audio signals that look incredibly similar when plotted. However, when I play them, they sound quite different. The only difference I can even see in their plots is that one signal looks ...
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1answer
616 views

Verification of all steps needed to extract MFCCs from raw audio data

I need some verification on steps required to extract MFCCs from raw audio as I'm finding that some sources outline them slightly differently. Get raw audio frame Run window function (eg Blackman) on ...
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1answer
2k views

Calculating the SNR of Audio Signal (Recommended Libraries)

Are there any open source packages or libraries available which can be useful in calculating the SNR(signal to noise ratio) of an audio signal. The input will be just an audio signal and I have to ...
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1answer
32 views

Why do AAC files seem to have an inexact bitrate?

I'm using ffmpeg to encode audio files as AAC at 256kbps. However, the actual bitrate of the resulting file is rarely 256. It's usually a number close to 256 like 241, 255, 257, 254. In one case it ...
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0answers
36 views

Detecting similar behaviour in audio files

Can anyone help, I'm trying to find a solution that would allow me the ability to look at a batch of .WAV files and detect ones which share a similar behaviour in the recording like the image attached....
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0answers
54 views

How enhance amplitude of audio signal without distortion?

After FFT I've set to zero some frequencies and I'd like to enhance the amplitude in order to recover the leak. I'd like to do a normalization in order to avoid distortion. I've done in this way but ...
3
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1answer
574 views

Audio Signal Noise Filter Problem

I'm currently working with audio signals and have a problem: C = A*B + N, where C = recorded signal from microphone consisting of: A = known music file data played on speakers next to microphone ...
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1answer
96 views

Audio/speech detection, verification

I would like to write a program to compare two '.wav' files, one of which is the original played file with human speech and the other is a recorded '.wav' file. The program I wish to write in python ...
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3answers
2k views

How to select the center frequency and filter type for 5-band digital audio equalizer?

I have a small team project: use Matlab implement a Gui 5-band digital equalizer using FIR and IIR filters (2 cases) - The sample rate of all audio files is standard audio 44.1kHz. Coding is not my ...
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1answer
35 views

Generating audio clip by superposing two files

I'm currently trying to generate a set of audio files where some audio events (e.g. dog bark) are immersed in a background audio scene (e.g. crowd noise). I understand that if the event would have ...
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1answer
53 views

Formant Analysis Frequency - What makes a peak value?

from spectrograms, how can you tell which peak is really a formant. Like why is the pink arrow the first formant. And in the spectrogram below, which has fewer samples, what makes a value the peak ...
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1answer
31 views

audio analog input + usb data cable = sound noises

I connect my android smartphone from audio jack output to my pc audio input and get a good analog sound signal. But when I connect this smartphone to the pc with usb data cable the analog sound ...
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0answers
40 views

Finding the input from the impulse response and output

I have $y,h,x$ which are all vectors. From $y[n]=x[n]*h[n]$ which is basically how I got $y[n]$. I also know $h[n]$. I put this through a Fourier transform. Let's assume that the capitalized ...
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1answer
110 views

polyBLAMP anti-aliasing in C++

How should 4 point polyBLAMP work. It does not really seem to dampen aliasing. I have an floating point digital audio oscillator that makes a +1.0f jump at the end of every period. My question is, ...
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1answer
65 views

Hilbert Transformer processing block by block

I am processing audio and trying to obtain a 90º phase shift when processing block by block. I have implemented a Hilbert Transformer (from Lyon's 3rd. Ed. & MATLAB). MATLAB code below (I believe ...
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2answers
244 views

Audio processing. Is it possible to directly access the decoded audio data going into the analog input of a computer

When a computer transcodes an audio file from one file format to another, does the computer first decode it into the raw digital stream (exactly what is fed into the DAC for the audio output) , or ...
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1answer
149 views

Teager's Energy Operator

Does anyone know how to find information on the derivation (rather than the definition and application) of Teager's Energy Operator, both for the continuous and sampled case. I have Kaiser's early ...
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0answers
36 views

Is it possible to analyze this microphone data and what to do if not?

I just created an account here because I really need some help. I'm using a microphone pmodmic3 with a development board called Ublox C030-U201 which uses the mbed OS. The microphone is working for ...
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0answers
6 views

Efficient measurement of audio sound pulse

Here and here are two interesting posts about converting an audio signal recorded from a microphone to SPL. For specific case of pulse recording, with 0.5ms pulse duration with 1s period between ...
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1answer
19 views

extend parts in an existing audio file to make it sound longer without stretching the file

I'm looking for a way to extend certain parts of an exiting audio file to make these parts sound like the where played longer. And I dont want to change play speed or stretch the file. Lets say u have ...
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1answer
28 views

Do Audio Interface manufacturers use A-weighted Measurements to make specs look better?

Recently I came across a reputed music-tech teacher who said audio interface manufacturers use A-weighted measurements to show dynamic range in their specs so that they look better on spec sheets and ...
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0answers
5 views

TEO-CB-AutoEnv implementation on python?

I'm trying to search for an implementation proposed as the nonlinear critical band based TEO auto correlation envelope area (TEO-CB-AutoEnv). If anyone knows of a library or such can you please link ...
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6answers
20k views

Why is telephone audio sampled at 8 kHz?

When did we decide to sample telephone at $8$ kHz? Has this always been the case? Why did we do that? Is it because higher bit rates can't be transferred as quick? And do these reasons still count? ...
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0answers
41 views

Noise Reduction: How can I filter out saw or other tool noises?

I recently came across old recording from the 1930's. Unfortunately not only is the quality low, it's a bunch of carpenters talking as they saw things and make all sorts of other noises. I've tried: ...
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1answer
33 views

how to lower frequency of a soubd file in matlab?

I am new to programming and sound processing, I have been using matlab to process sound files, a few of which are ultrasonic. my sampling rate is 250kHz. I Would like to be able to play the sounds at ...