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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Band Limited Impulse Train Synthesis

I'm currently working on a piece of DSP/Synthesis software and I have run into the issue of generating band limited waveforms. I have been doing some research to try and find a good solution but I ...
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1answer
22 views

Finding Accurate BPM and Beat Time Values in Audio

I working on writing a program to get the BPM rate and the Beat Times in audio songs but I am having trouble trying to come up with reliable testing strategies for my design.What do I benchmark my BPM ...
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22 views

Cost matrix for syncing audio signals

Is it realistic to, in real time, calculate a cost matrix of two onset location vectors? e.g. process two output(system) audio buffers and get onset locations in vector. calculate cost matrix ...
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1answer
17 views

What is mismatched noise?

Can anyone simply explain what mismatched noise is? And what is the noise that is not mismatched noise? Thank you!
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2answers
42 views

Controlling audio high/mid/low in frequency domain

Is it a realistic mapping to divide the frequency bins in 3 parts, high/mid/low frequencies? How can these bins then be manipulated to control the level of these bands in the time domain? Is it ...
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2answers
28 views

Wavetables and Partials

I am reading Direct Synthesis versus Wavetable Synthesis. The quote below explains code I have read but I don't get why it's a division by the integer multiple, when multiples increase in frequency. ...
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2answers
40 views

Extra energy when altering frequency amplitudes

How can I convert a time signal (audio) into the frequency domain, multiply frequency amplitudes by a vector of scalars, and convert it back into the time domain, without messing up timing information ...
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1answer
50 views

How to get Envelope of Audio Signal

I am having a problem understanding how the author gets the onset envelope of an audio song in this paper. The entire section in question has been quoted below: First the input sound is resampled ...
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0answers
33 views

Convert .mp3/.wav to formula [closed]

I have an .mp3/.wav file. This file was created with unknown to me tool. This tool used harmonic signals for left and right ears. How can I find formulas for harmonic signal from audio file?
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3answers
79 views

How is data compression possible for audio?

I understand that compression works by exploiting certain psycho-acoustical characteristics of our auditory perception, but I'm still confused. I mean ultimately, the waveform must be recorded ...
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1answer
62 views

How can I reduce noise errors when detecting DTMF with the Goertzel algorithm?

Thanks a lot to whoever downvoted my question for no good reason. I wrote an implementation of the Goertzel algorithm in JavaScript and made a demo that detects DTMF dial-tones. You can find the ...
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1answer
52 views

Is there any processing needed to playback audio? [closed]

I am new to audio processing. I wrote a simple real-time playback program (when you speak on the mic your voice is immediately played on the output), however it is not working. I am using a 8Hz ADC ...
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1answer
118 views

Obtaining beat histogram using Tzanetakis algorithm

I am working to make a music player with automatic genre classification application in Java. On Google searches I mostly found the works of George Tzanetakis who describe an algorithm from Music Genre ...
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2answers
86 views

Subtracting Original Audio Signal

here is my scenario: one speaker is sending a chirp, and a microphone is recording it as well as all following echoes. The function of the chirp is already know. My goal: subtract the original chirp ...
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1answer
28 views

what is standard acoustic volume

In Audio Experts book, it's given as Sound pressure meters must be calibrated at factory to a standard acoustic volume. So what doth this ...
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2answers
90 views

Audio signal dither and noise shaping

I'm trying to get a handle on the importance of the error feedback term in noise shaping operation in typical audio dither algorithms. I'm thinking in terms of four signals. The original signal, the ...
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1answer
35 views

Given input and output time series, obtaining filter design

Say I have 5 minutes of input and output audio. One method I know is to do FFT on windows from input and output. Then divide FFT output into bins and find average energy in various bands for all the ...
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0answers
38 views

Moving average on the frequency domain - (Welch's Method?)

Is there a name for a filter that is using a moving average on the frequency domain? I am searching for such a filter to filter out long "pads" (synths) from music. Do any of you know about such ...
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1answer
49 views

Detecting changes between voice and music

I have a bunch of audio files that have both speech and music parts (think radio). For instance, a given file might consist of the following sections: ...
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2answers
71 views

Decompose stereo signal

Basically the same as this question, but it was never really answered: Stereo sound separation into three sources Imagine I have a really trivial signal: Left: 100Hz tone + 200Hz tone + 500Hz tone ...
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2answers
71 views

Audio compression with STFT

Let STFT[frame, freqbin] be the short-time Fourier transform of some audio data. (here frame is the index in time, and ...
2
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2answers
196 views

What does the phase of a FFT bin of an audio signal represent?

I currently run real time audio through an FFT and the library (vDSP) returns magnitude and phase values (polar coordinates) for data in the FFT bins. Does the bin with the highest magnitude always ...
4
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2answers
311 views

Difference between jpg compression and mp3 compression

My understanding of jpg compression is that, if done repeatedly, any image will eventually be reduced to a single flat color, due to the way it deals with neighboring pixels. However, years ago I was ...
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0answers
32 views

AMARI ERROR in Matlab ( FASTICA Comparison)

I am now doing a project comparing non-linearities in FASTICA and the best way to compare them is by using AMARI error. Could I ask how to perform Amari Error after fastICA in Matlab?I tried to search ...
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1answer
48 views

Java amplitude processing

I have some code that collects amplitudes from a complex sound like a song. I now want to plot a graph of amplitude vs time, for the graph plotting I use JavaFX. However I get about 9 million ...
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1answer
53 views

Comparing an audio signal with a previous 'taught/profiled' signal

I'm working on a audio sample recognition project, where first you profile a sound and then run the sound detection function. Currently during the 'profile' stage I'm grabbing copies of the ...
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0answers
15 views

Diference between Specific,perceived and perceptual Loudness

i came across many terms describing loudness ! I understood perceived loudness which is what the user perceived and is related to the user. What is specific loudness and perceptual loudness? ...
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1answer
47 views

How can I find time difference between two same audio signals?

How can I find time difference between two same audio signals? Please do not answer cross-correlation or GCC. I am looking up for a new solution that will help me. Thank you.
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2answers
58 views

is there a second order approximation to perceived loudness

Quoted from this As a first order approximation it takes ten times the energy to double the perceived loudness So is there a 2nd order approximation. If so then how much change in RMS value ...
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2answers
155 views

problems calculating phase propagation (in C)

I'm coding a phase vocoder in C for time-stretching audio signals. The only piece that is currently not working is the phase propagation. I take overlapping frames from an input signal, apply a ...
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1answer
98 views

Fourier Transform for frequency analysis

I applied Fourier transform on a sound (PCM generated digitally with sin function) and frequency detection is accurate. However, if I play the sound and record it, and then apply a Fourier transform ...
3
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1answer
126 views

SPL values from FFT of microphone signal

I am trying to get a frequency response vs dB SPL plot for recorded sounds. I have a microphone with a flat response, a sensitivity of 4mV/Pa, and 1 Pa = 94dB. To get the dB SPL, I have the following ...
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44 views

Audio Steganography using Phase Encoding Technique

I referred the below papers to understand the working of Phase Encoding,but the explanation is not proper.Just a few steps have been given in brief. ijcsi.org www.ijarcce.com Can someone please ...
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3answers
68 views

Explanation of binning in frequency analysis post-FFT

I am creating an Android app which records sound for t seconds at a sampling rate of 44.1kHz with a buffer size of 8192 (16 bit mono samples). I need to plot a ...
1
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1answer
95 views

Comparasion of two wav files in Matlab [closed]

Hi I am working on one project and I have some trouble with comparison audio files. I have read a lot of articles about this problem but I couldn't find a good one working. In my project I need to ...
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1answer
36 views

Q value for a high pass filter

How to calculate the Q value for a high pass filter ? Ps: I'm a beginner in signal processing, so please put it in simple terms.
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29 views

Frequency contour cross-correlation

I'm trying to segment (or extract) different calls from a spectrogram. My current method, calculates a threshold value (variance) and then using RMS can detect voiced/unvoiced signals: $$ Thres = (1 ...
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1answer
165 views

How to make this cool Audio Spectrum Visualizer

I want to make/create this spectrum visualization on a physical Led_Bar_Panel(s). Here's what I think I should do: Take sample's in real-time( Sampling freq = Fs KHz, only voice) Do a FFT( As far ...
4
votes
1answer
69 views

Combining time domain and frequency domain data for BPM detection in real time

I would like to write a BPM detector. Does either of the domains provide more information for this type of analysis? What exactly is being looked for in the data to detect onset for BPM detection in ...
1
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2answers
69 views

Preparing audio data for FFT

I would like to experiment with some input from a microphone and am receiving a 2 channel, 512 samples buffer in real time I know the signal could be passed through a low pass filter, and then ...
2
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0answers
41 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 ...
1
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1answer
88 views

Voice Audio Detection algorithm

I have to detect speech intervals in large pre-recorded files in my project. I think, there won't be so much noise in the background (audio will be recordered in the room or even in the studio), but ...
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2answers
92 views

Energy of an audio signal

When energy is referred to as the square summable sequence, is that energy a representation of the power of the signal? How is the energy of frequency determined? E.g. what if an audio signal has ...
2
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3answers
75 views

Detection of wobbling sound with rising notes

In an audio file I would like to find calls of a specific bird specie, one that makes wobbling sound with rising notes. Can you recommend what would be the best audio feature to calculate in time or ...
-1
votes
1answer
30 views

How to check the quality of an audio through an algorithm or software program

Can you please help me with this problem?? I have been working on AM335x sitara evm and my project needs self diagnosis of the captured audio from the mic through the speaker on the same board (we ...
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0answers
47 views

Echo hiding technique

I'm trying to figure out how to use DSP techniques in order to both add artificial echoes to a sound file, but more importantly to be able to detect these echoes. I've been reading about it here: ...
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2answers
148 views

How to analyze audio input/spectrum correctly

I am quite new to Java and Signal Processing, but I was assigned a project dealing in audio processing. My topic given to me is a game that has a component of analyzing a song (any given song) and ...
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1answer
52 views

Pitch correction detection?

What type of signal analysis (if any) might be performed on a recorded music vocal track to determine if pitch correction might have been applied to a performance?
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0answers
39 views

Audio signal source separation with neural network

What I am trying to do is separating the audio sources and extract its pitch from the raw signal. I modeled this process myself, as represented below: Each sources oscillate in normal modes, often ...
0
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1answer
113 views

Get sound from matlab and convert with FFT [closed]

My question is simple but I am new to MATLAB's sound functions so I am unsure what to do. I would like to read the audio (from the microphone) then apply the ...