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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24 views

How to think about beamforming when there's no “space” between receivers?

I'm trying to wrap my head around how does "beamforming" work when there's no physical "space" between receivers. My background knowledge: I've always thought about beamforming as a sort of spacial ...
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17 views

Theory on distortion algorithms? [on hold]

Anyone know references for distortion algorithms (incl. saturation)? Particularly looking for something that characterizes them based on their sound as well as mathematics.
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1answer
39 views

Calculus on sound signals?

I was playing with the idea of performing integration on a sound signal, since I've never heard of where it'd be used. Is it used somewhere? Here's something: ...
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19 views

joining word samples into a sentence

I have a bunch of single-word samples for use in a command-and-control voice interface. These words are played in a certain sequence to compose sentences. It's fast (even on low-powered device), cheap ...
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0answers
24 views

Development board with 4-pos jack and USB

Perhaps I do not know how to research this correctly--I mostly looked at large sites like digikey and mouser--but I can't seem to find a small development board that has: one (ideally two) 3.5 mm ...
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2answers
38 views

best similarity measure for audio classification

I'm doing a project on audio classification, mainly focused on Bird sounds. I would like to know which is the best similarity measure that I can adopt to check test signal with training data. I ...
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1answer
29 views

Using 8 bit samples for speech comparison

I am making a speech comparison app in javascript. I found this dsp.SE post: Speech comparison algorithm for rating on similarities, where I read it is best to use mfcc and dtw algorithms. I ...
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1answer
59 views

Does/can physical audio theory exist?

I've been wondering, Since some sound generating physical phenomena have been physically modelled as physical modelling synthesisers, Does there exist a "physical audio theory" (or scientific ...
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2answers
37 views

How does formant shifting work?

How does formant shifting in synthesizers such as Reason's Maelstrom work? I've been pointed to the PSOLA algorithm among others, but I cannot understand how exactly does it result to formant ...
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1answer
38 views

How to analyze reverb in existing sound recording?

I'm wondering if there is a technique by which to analyze an existing sound recording and extract some details of the reverb and echo that are present (either due to natural environment, or added ...
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1answer
33 views

Efficient real-time pitch detection algorithm

What is the most efficient pitch detection algorithms in terms of computation? I need something to implement on a DSP. The typical signal in my situation would be a mixture of low-level wide-band ...
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1answer
80 views

Are there other filter models besides IIR and FIR for processing digital signals?

I am very familiar with IIR and FIR filters for processing digital signals (audio is my main interest). I am keen to know if there are alternative filters based on different mathematical models to IIR ...
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1answer
45 views

Generating drum sample in programming language [closed]

How would you generate the sample of a drum, say "tom". Using the programming language and it's functions like addition, multiplication, sine, sqrt etc. I tried to generate the sine wave and then ...
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2answers
47 views

Convert one sound sample in different sample where each sample's pitch is a note (do-re-mi…)

For example, I would like to do like in this video: https://www.youtube.com/watch?v=YBhneZuAb5k We have for example a goat yelling sample and from this sample, we can change the pitch according to ...
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1answer
40 views

Acoustical Design considerations for Active Noise Cancellation Project for Air vent

I am working on ''Active noise cancellation for air vent'' project. I am using NI myRIO1900, Audio Technica ATR 3350 omni directional microphones and Tevion Loudspeakers. I have done the programming ...
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0answers
55 views

How to apply a signal to FIR Filter in C/C++?

When I using code to compute FIR Coefficient and Window function, then I don't how to apply this generated filter to a audio data(PCM) signal. What is the procedure of creating a FIR filter and the ...
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0answers
36 views

How to normalize audio data for feature extraction

I recorded the same piece of music from different instruments and I want to compare their timbre by means of feature extraction. As you can see from the image, the recording is composed of different ...
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1answer
32 views

Band limited wavetables

I am a bit confused with the concept of band limited wavetables and the methods such as BLET, BLIP, MinBLEP, polyBLEP etc. As far as I understand certain waveforms contain infinite harmonics, which ...
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2answers
61 views

Finding a signals complimentary sound / frequency

I know about complimentary colors and how to get them, since colors are basically frequency / wavelengths. Is it possible to find a signals complimentary, additive or subtractive frequencies in an ...
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0answers
38 views

Composite signal through audio input

My question might be a bit stupid, but I have no idea if it's possible or not. Would it be possible to get composite input (RCA) through an audio input with an RCA -> 3.55MMadapter on a PC and have it ...
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3answers
63 views

how can i center an audio signal?

I'm trying to reconstruct an audio signal, and below is what i got after rescaling. How can i obtain centered audio signal such as below? SOLVED: ruoho ruotsi's answer addresses my problem :) ...
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3answers
31 views

Select talk not music from podcast

I listen to a podcast with interesting talk separated with horrible music. I'd like to rewrite the podcast automatically to delete this "noise". Edit: the talk and music is not simultaneously, it's ...
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1answer
46 views

How to simulate a stereo recording?

I have a single channel audio recording. I want to simulate the stereo recording when it is (hypothetically) played on a speaker in front of stereo mics (distance between them - say 10 cm). Are ...
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1answer
36 views

Sampling rate conversion for audio using ffmpegwrapper in python

I have many audio files @ 22kHz. I want to convert it to 44.1kHz. Since I have many files to convert, I am looking for an easy solution. I came across ffmpegwrapper for python. I installed the package ...
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2answers
61 views

How many samples per cycle(time period) is actually enough to reproduce the sine wave sound properly?

I have been working on Active Noise Cancellation in real time for Air vent in kitchen. Initially for the demonstration purpose, I am using sinusoidal sound ( 40k Hz Fs,1000 samples, 2 kHz frequency, 1 ...
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2answers
61 views

How does mic combined with loudspeaker work?

There is variety of loudspeakers on the market that have built-in microphone so that they can produce sound and perceive it at the same time (and do it surprisingly well). How they managed to ...
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2answers
104 views

DSP for computer engineers

DSP is more kindred with electrical engineering than with computer science, how ever there are topics of DSP like MP3 encoding, JPEG compression and sound analysis can be understood by a computer ...
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1answer
47 views

How do delay lines create so many effects?

I was trialing this audio plug-in from SonicCharge: https://soniccharge.com/permut8 (there's a program demo in the site) It claims to be based on a delay line and uses some logic to modify it: ...
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1answer
91 views

The Bode frequency shifter

Anyone know if the Bode frequency shifter has been studied somewhere. I just used an audio plug-in called EchoBode from SonicCharge and this is claimed to use a frequency (pitch) shifting algorithm ...
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2answers
79 views

How is the Leslie speaker simulated?

What are Leslie speaker / "rotary" style of digital audio processors doing? I think a simple one would model: Delay between speaker and a pair (stereo) of microphones. The tonal change when the ...
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2answers
98 views

Are there equations for placing sounds in the stereo field using delays?

I'm thinking of the typical stereo widener (haas, binaural) effects here, those that extend from the typical pan knob to create "out of the headphones" -experience. These processors can be automated ...
4
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1answer
69 views

Calculating Beam pattern, Directivity of microphone array

As answered in Relationship of Fourier Transform and Spatial Directivity Pattern the relation between directivity pattern and weights is give. But I am bit confused, I have Beamformer weights ...
4
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0answers
59 views

Accuracy of IFFT(FFT(X)) unit transformation

I want to use FFTs to do seriously accurate interpolation on band-limited data. To do that, I need to get a handle on the fundamental accuracy of the FFT() and IFFT() algorithms available. My idea ...
4
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1answer
93 views

How does Adobe After Effects generate its “audio spectrum” effect?

I'm trying to replicate the "audio spectrum" effect from Adobe After Effects. An example can be seen in this video: Obviously, it has to be some variant of a fourier transform, but I've tried ...
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31 views

Is there a way to determine what pitches are present at a given time in a music sample?

I'm sure it would be complicated to do so, but it seems the programs like Melodyne can separate notes in a chord. How complicated would it be to determine with any accuracy what pitches are being ...
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3answers
65 views

Are there libraries for extraction of sound wave features?

I am looking help for my project for which I need C++ (or any other language) libraries useful for extraction of sound wave features like frequency, loudness, pitch and orientation. Using these ...
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2answers
52 views

Audio silence on 1.0

I have audio files with silence on 1.0 (max amplitude). I attach an example of audio waveform. I don't know why it is on 1.0? How can I convert it to have silence on 0 (as usual)? I noted that at the ...
6
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1answer
85 views

How to create a sine wave that transitions through frequencies and returns to phase

For an auditory psychological experiment, I need to create a sine wave, such that at a given point the frequency of the wave will rise smoothly, and descend back to base frequency. I need it to return ...
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1answer
111 views

Best algorithm for time-scaling

what is the best algorithm for time_scaling of a speech signal in which I've seen two of them and I don't know which one to use : Psola:Pitch synchronous overlap and add. time scaling modification ...
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1answer
33 views

Why is the name “Phase vocoder”?

I've read that phase vocoder is the STFT, but done in windows (i.e. using windowing). Why is it called "phase vocoder"?
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1answer
32 views

Windowing function squared?

Reading this paper about WSOLA (Verhelst, Roelands) At page 2 there's a "time-warped STFT's synthesis function" involving $v(n)=w^2(n)$ which is a windowing function. A windowing function squared. ...
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1answer
56 views

Real-time resampling pitch-shifting (without corrective time-stretching)?

Is real-time resampling pitch-shifting impossible (without corrective time-stretching, such as WSOLA)? I've been told that it is. Real-time resampling pitch-shifting would mean applying resampling in ...
2
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0answers
22 views

Alternative to scipy maximum_filter to find peaks?

I'm using scipy maximum_filter together with binary_erosion to find peaks in my spectrum. I just wondering if there are any ...
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2answers
57 views

Removing Bounded Noise from an Audio Signal

I'm currently trying to remove some noise from a "contaminated" audio file. From what's given, I know that the noise is isolated to a specific frequency band - that is, from ~1400 Hz - ~1800 Hz. The ...
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1answer
55 views

How to eliminate noise that has the same frequency of my signal

I have an audio signal , and I want to remove noise from it. I have applied a bandpass filter in order to remove frequencies that are not part of mi signal. But if there is background noise (some ...
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1answer
76 views

Finding peaks in an Audio spectrogram

I'm using gist.github.com/sixtenbe/1178136#file-peakdetect-py to find maximum peaks in my spectrogram. I used ...
2
votes
1answer
82 views

What is the difference between PSOLA and TDHS time-scaling or pitch-shifting?

What is the difference (if any) between PSOLA (Pitch Synchronous Overlap and Add) and TDHS (time domain harmonic scaling) time-pitch modification algorithms?
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1answer
52 views

How does PSOLA pitch-shift work in real-time?

How does the PSOLA pitch-shifting algorithm manage to remove or add audio frames in real-time without creating gaps? Wouldn't extending the "windows" (it's like in FFT windowing) to longer than the ...
2
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1answer
40 views

How does “image synthesis” work?

I'm interested in how does "image synthesis" in audio synthesisers such as Metasynth work. Image synthesis is when a picture is read into a program and "interpreted" into sound. But how does it ...
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21 views

Audio mixing from 2 Microphones using java

Here's my situation, 1, I'm getting audio data for playback from network in byte array format 2, When I speak using 2 microphones near by to each other, I can hear what I'm saying but the problem is ...