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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16 views
PulseAudio cannot read buffers at consistent rate [on hold]
I want to record my microphone and process in real time, eg. show an oscilloscope view of the microphone signal.
I've tried 3 ways to get the data out of PulseAudio and they all perform the same. ...
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0answers
26 views
Audio callback function for Windows in c++
I'm writing a cross platform c++ based audio library for an app I am creating. For the audio component I am not using any platform specific code. I have a test harness that outputs audio using device ...
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19 views
Remove human noise in background from audio signal [on hold]
I am new to signal processing. I have a set of audio signals which has background noise of humans talking. Is there a way to remove the human voice and save the new audio signal. Can i get the code if ...
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0answers
18 views
Time gating of FFT for audio Measurements or Multi Time Windowing
I am learning more about DSP specifically for audio measurement purposes.
I can an app, where I can create a log sine sweep from start frequency to end frequency over a certain length. I can perform ...
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0answers
13 views
Decoding and playing FM from async RTL-SDR [closed]
I am trying to write a Python code to demodulate an FM signal using pyrtlsdr's ayncio features. As well as, to playback the FM signal with pyaudio using loopback. Here are my codes:
main program:
<...
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1answer
25 views
How to separate silence from frequencies in Matlab
For a project, I was asked to create a DTMF detector on Matlab. I know that the audio sample that will be provided will have 4 tones maximum, possibly less, and that the audio begins with a 0.1 second ...
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29 views
Designing Matched Filters on Arduino
I'm trying to develop a prototype at the moment that includes a sensor to detect how fast someone is typing on a keyboard. Essentially this sensor is a microphone that will be able to detect keyboard ...
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2answers
24 views
audio spectrogram normalization
I am performing language classification from audio signals using mel-spectrograms as my inputs for a ResNet. It works well as long as all of my audio data from different languages is from the same ...
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10 views
What is the difference in the algorithm for noise reduction in SoX and Audacity?
I've looked at this question, and I understand that SoX works way FFT of the noise, then subtracting that from the FFT audio, and then resynthesizing using FFT, but I don't quite understand how this ...
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1answer
11 views
What are the ranges of values for each audio feature/descriptor?
I need to normalize some audio feature measurements I've done with the software Wolfram Mathematica.
Coming from a non signal processing background, I have some difficulties understanding what ...
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0answers
6 views
Wavelet Analysis Magnitude Scale
I am trying to plot magnitude against frequency averaged over a time window.
To do this I make a morlet wavelet Filterbank. And multiply the dft of each kernel by the dft of my incoming data set.
...
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1answer
25 views
Deriving difference equation
I have this block-diagram of Karplus-Strong basic algorithm.
Where $$H_{d}(z) = \frac{1+z^{-1}}{2}$$
The problem occurs when I try to get the difference equation. As I can see from the diagram, that ...
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0answers
18 views
Is it possible to create a causal filter with a positively-sloped phase response?
I am currently prototyping a set of noise-cancelling headphones that utilize active EQ to modify the noise-cancelling signal, allowing the user to modify the frequencies that they hear or don't hear. ...
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11 views
Methodology for log some sweep audio measurement impulse response
I am trying to implement a method for measuring the response of a loudspeaker in a room using a log sine sweep stimulus.
To check my method:
1. I create the stimulus using:
Make recording length ...
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0answers
32 views
Implementing real-time variable fractional delay line in Matlab
I am attempting to implement a fractional delay line in Matlab which operates in real time, uses an object-oriented approach, and whose length can be modulated each time a sample is processed. I have ...
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0answers
16 views
Log Sine Sweep Acoustic Measurements
I am currently trying to understand the Sine Sweep Method for taking acoustic measurements.
I am following this paper:
https://www.thinksrs.com/downloads/pdfs/applicationnotes/SR1_SweptSine.pdf
I ...
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3answers
73 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
65 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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1answer
34 views
Reading a Wav File [closed]
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
94 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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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
37 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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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
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
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
38 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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58 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 ...
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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
66 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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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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2answers
62 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
67 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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1answer
31 views
audio analog input + usb data cable = sound noises [closed]
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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2answers
255 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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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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3answers
61 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
20 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
29 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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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
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
46 views
Frequency modulation audio synthesis: clicks/pops when changing modulator frequency dynamically
I'm very new to DSP and working on a FM VST. It's mostly up and running, or at least it generates FM-like sounds. I want to be able to change various parameters dynamically while audio is generated, ...
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1answer
56 views
Extracting audio from picture of time-domain signal
I saw this nifty art thing where they dress up the time-domain audio signal in some pretty way.
https://www.etsy.com/ArtbloxShop/listing/621561709/sound-wave-art-custom-soundwave-print
My question is ...
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1answer
124 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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1answer
45 views
Frequency analysis : relation between FFT size and sampling
I am currently trying to perform an experiment given an audio signal. I am therefore sampling the audio frequency range (0 to 22kHz) with a 48k sampling rate.
When performing a FFT on my signal, I ...
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0answers
30 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
147 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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0answers
34 views
Separating Ultrasonic signal from echo at the receiver
A transducer is acting as a transmitter and a receiver. It keeps on sending pulses which can hit an object placed at 5cm. Echo comes back and hits the transducer while it is continuously sending ...
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0answers
88 views
How do I play and record audio at the same time?
So this question is more related to Matlab itself.
Some background info: I'm trying out the Exponential Sine Sweep method to obtain the Room Impulse Response with my laptop speakers and microphone ...
