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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39 views

Trouble Implementing Huggins Binaural Pitch

I'm trying to make an implementation of the Huggins Binaural Pitch illusion, which is where if you play whitenoise into each ear, but offset one ear by a period $T$ that it will create the illusion of ...
0
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0answers
18 views

Difference between Envelope filters and a unit with variable envelope response

Would a unit with variable envelope response shape a square sound wave into a sine wave? I know that an envelope filter kind of does that through subtractive synthesis. I'm trying to find a way to ...
5
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2answers
816 views

Can you turn a square wave into a sine wave using a low pass filter?

And if it could, would it make the sound of the square wave thinner than before because of losing its harmonics?
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0answers
27 views

How do we detect presence of background noise in audio?

The problem I am facing to solve is detect background hissing or humming sound in any audio stream. The audio stream can be of any type ranging from movies to talk shows. The background noise is ...
0
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1answer
31 views

How is a PCM byte converted to a voltage level?

16-bit signed PCM would have possible values of ~-32,000 to ~+32,000. How is this value converted to a voltage level for the speaker? If a DAC outputs 5VAC, the values would be ~+-5v. How is the ...
1
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1answer
38 views

One-bit depth audio

I have a music player which is only able to play sound with one a bit-depth of one. I can produce this by taking a song and simply boosting the signal $+100\textrm{ dB}$. Is there another approach ...
2
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1answer
148 views

FSK analysis: only ever find one frequency

I've trying to write this software as part of my amateur radio and SDR hobby for analyzing various two-level FSK signals. I'm very new to DSP but have been having a lot of fun so far learning more ...
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0answers
70 views

detect what is noise in audio signal

I'm trying to filter noise from an audio signal. The signal was and is going to be recorded in a city environment which always contains some white/pink noise. At this moment I'm using two options to ...
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0answers
36 views

Low quality mobile phone calls

I am having problems with low quality mobile phone calls when submitting a audio file through it. I am sending the calls with a VoiceXML compatible provider and it seems that the mobile compression is ...
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0answers
23 views

Can I use fpcalc to prove two audio files produce the same sound?

I'm using fingerprinting techniques to detect duplicate audio files on my music library. The first and easy step is to compare the file-size and if they match I use a md5 sum to compare them. The ...
1
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1answer
74 views

Compensating Loudspeaker frequency response in an audio signal

I have been working on a project in which I was required to work on the audio signals recorded from the loudspeaker kept in front of a filter. So, to simply explain it: $$\boxed{\rm LoudSpeaker} \...
0
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1answer
55 views

How to remove wind noise from video file

I've recorded video footage from a recent motorcycle trip using a helmet mounted Go Pro Hero 4. The video is good, however wind noise is drowning out the sound from the engine. Is it possible to ...
0
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1answer
24 views

Cascading FM synthesis operations

I'm trying to understand how to program cascading FM synthesis operations. I can find a good deal of information out there on how to do it on various pieces of hardware, but unfortunately, there isn'...
0
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1answer
43 views

Spectral plot shows more notes than there really should be

I was experimenting with sound analysis lately and from what I see when I plot spectral data of an audio file is that apart from notes that were actually picked there are some other notes with quite ...
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0answers
29 views

Improve $\rm dB$ level after upsampling audio

I am doing upsampling from $44.1\textrm{ kHz}$ to $48\textrm{ kHz}$ on android device, using $\rm sinc$ interpolation. The output $\rm dB$ level is upto the mark for wave file up to $15\textrm{ kHz}$....
0
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1answer
32 views

How to calculate Long Term Average Spectrum of an audio file?

I'm working with audio files and I have to find their long term average spectrum. Although I had an option available in the software Praat but I don't understand the algorithm of finding the spectrum. ...
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0answers
28 views

Synchronization of two audio tracks via Mel-frequency cepstral coefficients (MFCCs)

I am trying to solve the (common) problem of synchronizing (i.e. measuring the delay between) two audio tracks recorded from different sources during a conference talk (about 1h of audio). One track ...
1
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0answers
8 views

STM32 Discovery dev platforms [closed]

I'm still researching this but though I'd post asking about experiences. Looks like st.com offers a variety of STM32 Discovery dev boards for ARM CPUs to, among other things, play with DSP. But ...
3
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4answers
139 views

How to calculate an azimuth of a sound source in stereo signal

Let say we have a stereo recording of only one source. What is a correct algorithm to calculate its azimuth using inter channel time difference and inter channel level difference?
0
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1answer
44 views

Bandlimited PWM

Trying to figure out how to synthesize a Bandlimited PWM in my audio code in C. I have already figured out a bandlimited pulse wave but I can't really figure out how to translate that to PWM. I have ...
0
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1answer
44 views

How do I measure frequency of an audio file into an array?

So I know how to plot fft the audio file within Matlab and a window pops up. My question is, what next? I want to know how to go about measuring the frequency(s). I have two similar audio files, of ...
3
votes
1answer
69 views

How to do continuous signal processing (i.e without windowing)?

I'm working in speech recognition research, and I wonder if there is a way to analyse sound just like how ears do (i.e. without windowing), so for example to take some feature continuously with an ...
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0answers
29 views

How to calibrate two microphones to same frequency range

I am recording stereo with two same microphones but they are physically insulated from eachother and have some additions on sound input side. Not to complicate story, I have a human head simulation ...
5
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1answer
67 views

Bandwidth expansion by computing missing harmonics in music?

In old recordings or highly compressed audio files (lossy), the higher frequencies are lost, and we hear a distinctive muffling effect. (Just like if the song was played through a wall) Is it ...
0
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1answer
46 views

How to overlap and add the frames after applying some modification in time domain

I am trying to apply Dynamic range compression (DRC) in time domain in MATLAB. But I am having problem in overlapping and adding the frames. Here is my work: ...
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2answers
83 views

Problem with Generalized Cross Correlation (PHAT filter)

I have to calculate time delay between two signals.I am using the GCC-PHAT method. First, I cut the signals into smaller parts (~0.5s) and then try to calculate the delay. When I display those two ...
0
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2answers
31 views

I want to know if there is a practical way to find out if multipath propagation is significant for audio in a given environment?

I am employing audio signals generated by a mechanical press for Blind Signal Separation. But, prior to algorithm application, I need to clarify if the signal strengths coming in multipath (except ...
0
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1answer
34 views

Specs for creating sample set?

I would like to create a set of audio samples for playback as a midi-controlled instrument. Thus: a sample for every key of the piano range, with multiple volume timbres for each note. I am able to ...
0
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1answer
48 views

How many DFT bins must I use to get best precision when analyzing melody scale

I need to analyze some signals containing chords. How big DFT spectrum is needed to identify each possible musical note in such a signal sampled at 44100 Hz? I can see that 8192 bins gives some ...
1
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2answers
36 views

Sampling frequency of audio

Based on what we choose sampling frequency when sampling some song? Do we use Nyquist–Shannon sampling theorem, or maximum sampling frequencies are pre-defined, so it will never come to aliasing ...
0
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1answer
70 views

Audio Broadband Noise Detection

I have to quantify the broadband noise of an audio track. I've seen lots of things around and the only viable option to estimate the broadband noise (without having any information about it) is ...
1
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1answer
53 views

Best DSP algoritms for ultrasonic background noise cancellation

I am looking into detection of pressurized gas leaks by means of the broadband hissing sound (white noise) generated from pressurized leaking gas. The instrument will have a broadband microphone ...
0
votes
1answer
74 views

Slope of 2nd order low pass vs band pass Butterworth filters

I have noticed that some manufacturers of audio processing products label a 2nd order Butterworth low pass as having a slope of 12dB per octave, but then also label a 2nd order Butterworth band pass ...
1
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1answer
53 views

Looking for low-noise audio data to benchmark some noise-processing techniques

I'm looking for a source of very low-noise audio data for use in benchmarking various methods of random noise correction methods. My intention is to have a very clean signal to which I can add known ...
0
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2answers
38 views

Combining channels into single FFT left/right vs mid/side

I am creating a spectrum analyser to visualize some stereo audio data in real time. I want to display a single spectrum for the input and a single spectrum for the output. The stereo data is the usual ...
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2answers
50 views

How to do the highest resolution FFT?

I wish to do the FFT of streaming audio data coming into a computer. I am just interested in frequencies from 80 Hz up to 1000 Hz but with a frequency resolution of 3 to 4 Hz and no more. Basically ...
0
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1answer
47 views

Why is this FFT code not working to extract frequencies from a wav audio?

I'm quite new to audio processing and i'm trying to extract frequencies from a wav file. As research papers stated i put the wav audio into a byte array and segmented and trying to apply FFT to get ...
1
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3answers
73 views

Performing DFT of streaming audio problem. Is there a limit?

I am trying to write software that will perform the discrete fourier transform of real time data coming from the microphone into the sound card on a computer. I am using Java with the javax.sound APIs....
4
votes
2answers
161 views

How to deal with low fundamental when using AMDF for pitch extraction?

I am using the Average Magnitude Difference Function to estimate the fundamental frequency of a quasi-periodic audio signal. The AMDF is defined as $$ D_n = \frac{1}{N-n}\sum_{k=n}^{N-1}|S_k - S_{k-n}...
2
votes
1answer
161 views

Demodulating FSK audio in Python

I am trying to write a Python script that can demodulate an FSK modulated audio file and return the data encoded in the audio. The data being transmitted is GPS NMEA strings which are embedded as the ...
1
vote
1answer
90 views

How to remove 50 Hz frequency noise from a PCM file?

Are you kind to tell me the odds or the means by which should I grab a clean signal with the $~50 \mathrm{Hz}$ carrier removed using just programming? I managed to make a program in Visual-C that ...
1
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2answers
76 views

How to do 3D audio?

There is the concept of audio played at specific locations relative to the listener, such as in games. What are the details of how this is accomplished in stereo or higher number of channels? I know ...
0
votes
1answer
71 views

how to detect short pause in a speech with noise?

I would like to count the number of short pauses in a speech and know the duration of the pauses by loading the signal into Matlab but I am very new to Matlab. Can anyone help and give me some ideas ...
-1
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1answer
33 views

How can I read PCM linear file in Octave? [closed]

I have a pcm linear file (16 bits) collected with a sampling rate 8000 Hz. I want to read the file in octave. Whenever I read it using: ...
0
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0answers
38 views

Adding two audio files at a predefined SNR level

I have two speech audio wav files. file1 is primary speech, file2 is supposed to be interference. I want to add them so that ...
0
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0answers
42 views

Unable to generate and verify a sound wave of frequency 200 Hz

I am trying to generate a sound that contains low frequency in the range 100-500 Hz. I then record the sound using a microphone and generate a spectrum for the recorded signal using ...
1
vote
0answers
36 views

Applying DTW on MFCC frames

I want to use MFCC (Mel-frequency cepstral coefficients) and DTW (dynamic time warping) algorithms to make a speech comparison script. I calculated MFCC and got 13 coefficients for each frame. How ...
0
votes
3answers
96 views

Convolution and windowing using a buffer - how do I do overlap add?

I'm reading chunks of audio signal (1024 samples) using a buffer, applying a Hanning window, doing an FFT on it, then reading an Impulse Response, doing its FFT and then multiplying the two (...
1
vote
1answer
95 views

identification of sounds within audio signals

I am looking for methods to identify sounds within audio files. I was hoping to find some ready-made user-friendly software to do this but haven't found it yet. I am familiar with Matlab and some C++...
0
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2answers
53 views

Smoothing an LFO Sawtooth Wave

I'm trying to smooth a Sawtooth wave coming out of my LFO. Effectively the transition from 1.0 to -1.0 in a 1 sample step is causing clicks and pops if I modulate my audio signals amplitude with ...