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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2answers
49 views

Audio template matching - power correlation?

I'm looking at some code that matches audio templates in a longer audio file. The calculation correlates the power spectra of the template and audio file, maximizing over the possible alignments. This ...
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1answer
36 views

Averaging different frequencies

How does one average a group of frequencies? Frequencies taken for example using FFT. That is, given n frequencies over m time intervals (n $\leq$ m), how should one "interpolate" between those ...
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16 views

Audio envelopes: Automatic attack and release?

Can you calculate attack and release settings for an audio envelope (e.g. in a dynamic range compressor) automatically? Are there any known algorithms for it?
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1answer
48 views

Seamless audio loops via Fourier transforms?

I am trying to create short seamless loops of continuous sound. I recorded myself making an "Aaaaah" sound at 44.1 KHz, and cut this recording to a section containing 4096 samples (giving me slightly ...
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0answers
22 views

Locating all sorts of “harmonics” in an audio signal

Harmonics are integer multiples of some fundamental frequency $f_0$. Then there are also inharmonics. But are there general procedures for determining where the harmonics are and what kind of are ...
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1answer
22 views

Is there an existing feedback elimination algorithm that can be implemented in iOS?

The project I'm working on allows a user to speak through a PA system with their iphone as the microphone. Unfortunately, the existing audio units in iOS don't eliminate the feedback between the PA ...
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0answers
20 views

Replicating a visual audio spectrum

I've been developing a web app as a hobby over the past few months which is essentially a real-time Javascript clone of the audio visualizer used by Monstercat. While it looks kind of close to the ...
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0answers
45 views

Real world dsp implementations examples?

Well,this may be basic, but i find you as the best option to answer this. I am electrical engineer and i have taken dsp courses. There is some algorithm that i am trying to implement on audio signal ...
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3answers
72 views

Extreme audio stretching

This tool (Paul's Extreme Sound Stretch) is able to produce extreme timestrech (50x, or even 100000x). Any song can become an interesting sound texture or ambient music (listen to this example). The ...
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2answers
60 views

Creating a flat spectrum filterbank

I'm trying to create a filterbank whose frequency response is generally flat when summed in parallel. Are there any tips to choosing center frequencies and bandwidths for a filterbank of overlapping ...
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1answer
43 views

DFT for audio classification - signals of different lengths

If I have a set of signals each with $n$ points, I can take the DFT of each and then use the resulting frequency domain vectors as features for some classification algorithm. If the signals are ...
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0answers
54 views

how to correct minBLEP sawtooth oscillator DC bias

In my implementation of minBLEP, overlapping minBLEPs result in a DC bias as the frequency increases. This is because each minBLEP correction (i.e. the minBLEP minus the aliased step) has a DC bias. ...
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2answers
66 views

DSP Hardware for Detecting an Audio “Click”

I am working on isolating a specific audio click in a factory type setting. After my initial analysis I determined that a 2-pole High Pass Filter set between 500 and 800Hz would work best for me. The ...
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0answers
24 views

Difference between note based (MIDI) and sampled (FLAC, MP3, etc.) audio formats

I (somewhat) understand sampling and sampled formats, and their relation to the original (analog) signal, but do not quite relate it to note based (MIDI) formats. What actually is a note based audio ...
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1answer
27 views

Voice recovery with high-noise, low-gain

I have a recorded interview with two speakers, in which one voice's gain was much higher than the other (by three orders of magnitude). Increasing the gain does not help, because speaker 2's voice is ...
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0answers
17 views

Multirate filtering attenuates frequencies at overlap

I am trying to build a filter with different frequency bands in Matlab. I would like the filter to be as invisible as possible, but after the filtering, a great amount of frequencies are being ...
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0answers
41 views

Audio block-processing discontinuities

I have an algorithm that looks at an audio frame, calculates a threshold in db-rms, and decides to filter the frame whether or not the threshold is crossed. However, filtering each block is producing ...
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1answer
46 views

Emulate tape delay

I'm trying to emulate a tape delay in a VST plugin, but I'm not sure how to handle the delay changing? I want it to sound like a tape is actually fast forwarding/rewinding. Most digital delays I've ...
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2answers
56 views

Is cutting EQ scientifically different than boosting EQ?

This is a fairly standard belief in the audio production scene that cutting EQ is somehow "cleaner" than boosting EQ. You hear arguments like there being different phase characteristics or whatever. ...
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1answer
40 views

Is there an audio processing library/API that contains all standard features of a digital audio workstation?

I need an audio processing library/API (in any programming language, preferably Java or Python) that is able to do things to sound waves such as: sum/mix sources together, apply EQ, dynamic EQ where ...
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2answers
44 views

Find frequency/phase response of a microphone

I am trying to find the frequency/phase response of two cheap microphones bought off Ebay. Basically, I want to graph the frequency responses and phase responses of both microphones on the same axis ...
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1answer
51 views

Vinyl Crackling Effect

I want to know how can I achieve this effect: https://www.youtube.com/watch?v=DQg9M_47c3A Do I need to use a filter? And what can I use for the crackling, poping effect? Thanks in advance!
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0answers
45 views

Noise reduction on wave file

I have created a simple plot of waveforms using matplotlib of 2 wave files on python. Here is the code: ...
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1answer
33 views

How does spatially white noise differ from diffuse noise

In noise reduction, having read that directivity index is used to reduce diffuse noise and white noise gain reduces spatially white noise, I would like to know the difference between both the noise ...
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1answer
46 views

playing audio continously in matlab

I am in my way to design a function generator using matlab,the point which made me stuck is the challenge of continuous waveform generation code snippets of my attempt ...
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0answers
27 views

Dynamic range compression in the frequency domain?

I'm trying to do frequency-band specific dynamic range compression (like multi-band compression) and was thinking of whether dynamic range compression is doable in the frequency domain or whether it's ...
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1answer
27 views

Matching two video conversations

I have two video where two person having conversation. They begin recording different times. I want to match these two videos' frame numbers, i.e. By matching audio. How can I do that?
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0answers
33 views

DSP Library IIR and FIR filters

Me and a coworker are currently in the process of making an audio dsp library. We are wondering what some good FIR and IIR filters to use were. I know ALOT of them but what do you guys think in ...
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1answer
80 views

How to simplify the Fourier Series Using an Approximation?

I have a signal, $f(t)$. I know a function that can be used to generate this signal, such that I can determine its Fourier series. I want to express this Fourier series in simpler terms so that the ...
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2answers
70 views

How to generate realistic echo to sounds?

I guess the most straightforward way of adding an artificial echo to a sound signal is to add its time-delayed version to the original signal. However, I am not sure how realistic that would sound ...
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1answer
41 views

How to fix audio clipping in Adobe Audition?

I am editing audio from video that was captured from the sound board. The audio was recorded too loud since someone turned up the board and and the bass and vocals sound scratchy. I have brought the ...
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1answer
29 views

Cancelling the acoustic echo that occurs with a variable delay and volume

I am trying to build a software that records audio to a server and then broadcasts that audio into the same room that the microphone is present. This means that the audio signal is uploaded to the ...
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1answer
73 views

How to interpret these different Fourier analysis of this audio signal?

This is my first dive in DSP. I would like to familiarize myself with frequency analysis. I have two audio tracks which should be digitized at 16bit-44.1kHz and 24bit-192kHz (music, presented as a ...
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0answers
37 views

Detailed Analysis of Phase Delay of an Audio Signal

I write this question as a new reference point because it is difficult to find something exhaustive regarding this topic. I'm talking about the calculus and interpretation of the phase delay of a ...
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0answers
19 views

Should I normalize my input vector when using a matched filter?

I am using a matched filter to recognize an audio sample within a noisy input stream by cross correlating my sample with the input array and checking if the maximum crosses over my threshold. ...
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1answer
46 views

Using Chebyshev polynomials in wavetables

How are Chebyshev polynomials used in wavetables for audio synthesis? I dont understand how to store these inside wavetables? How does the synthesized audio result in matching to the correct index ...
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0answers
38 views

Advice for creating differential microphone array

For a school project I'm required to create a differential microphone array that can filter out background ambience, so a clear voice signal can be recorded in noisy environments. The plan was to use ...
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0answers
42 views

How to calculate the frequency shift of FMCW transmitted signal and received signal using Matlab?

I generate a FMCW chirp signal between 18k-20kHz using Matlab, then sound the signal through the speaker in laptop, then record the reflected signal through the Mic in the laptop. What I want to do is ...
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0answers
29 views

Invertible multi-resolution representation of audio/speech signals

What are the invertible (preferably linear) representations of an audio/speech signal which captures the essence of signal in different resolutions and details. Basically, I am looking for ...
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0answers
19 views

Representing and compartmentalizing FFT of audio sample into user friendly spectrum (need direction)

I am using the Apple accelerate framework to do some audio FFT analysis. Here is what I have so far and I want to make sure that my methodology is correct and I am indeed displaying appropriate data ...
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0answers
9 views

What's horming frame in AMR-WB code?

I find a lot of word to talking about "homing frame" in the AMRWB spec and code. So what it is?
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3answers
85 views

EQ - Calculate the power per Octave of a signal through FFT

I'm working on a audio Equalizer, and I'm not sure if the way I obtain the power per octave of my signal is good. Here is the code I have : (I use the third octave bands, which is initialized earlier ...
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0answers
33 views

Use SNR for Wav signals

I have implemented an audio steganography program in MATLAB. One way to evaluate your steganography method is to use SNR. The formula I found for SNR calculation is mentioned below. The problem is, ...
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1answer
50 views

Adaptive volume normalization on a stream of PCM data

I am writing a program that receives a continual stream of PCM data from various sources. Some sources have different volumes than others. The PCM data is always a live continuous stream of 16 not ...
0
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1answer
44 views

Division by zero in Levinson–Durbin recursion?

I am doing a LPC analysis of a speech signal using the autocorrelation method. To calculate the LPCs I am using the Levinson–Durbin recursion. In my literature the error is initialized with the first ...
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2answers
42 views

phase, magnitude, audio file and information

I've been doing some thinking lately. I know that for image files a larger portion of information is contained in the phase of the signal. How does that go for the audio? Does the same hold or is the ...
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0answers
25 views

How to detect impulse noises in audio recording?

How do you detect impulse noises in audio recordings of speech? By "impulse noise" I am thinking about noises like: Typing on a keyboard A door slamming Somebody bumping into a table Tapping on a ...
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0answers
26 views

How can I convert a FM discriminated signal to binary data using software

I currently have FM discriminated data in a .wav file. How can I convert it to binary data (+0.5 == high bit, -0.5 == low bit) given a baud rate.
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1answer
51 views

Possible to implement simple IIR filter with greater than 6 db/octave slope?

I am using the following IIR code as a low-pass filter in a software project, and it works great. However, I would like the slope of the filter to be much steeper as the one here seems to be very ...
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1answer
100 views

Determining The Sparsity or Density of a Chroma Vector

I have a range of songs where I have took an 'unfolded' chromagram computed from a series of short-term windows. Specifically, for one song, I have one vector which contains the amplitudes of the ...