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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146 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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50 views

Goertzel filter Poles on unity circle influence of a behavior of a signal at high frequency

I'm trying to apply a goertzel filter on my sound data of 20500 Hz with sampling rate of 44100[samples/sec]. My bit duration which is passed to the goertzel function is returning a very low magnitude ...
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150 views

goertzel or bandpass?

I try to decode freqs of 17k & 17.8k which carry on a sinus of 15ms duration. I try to decode them with a goertzel filter. It seems fine when one of the filters show a magnitude that is around ...
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118 views

Audio output distorted with any FFT Windowing other than rectangular

For an experiment i have a simple application that takes a quadrature signal from a radio and then does the following: Complex FFT -> Inverse FFT -> Cartesian to Polar -> audio out If I tune the radio ...
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147 views

What is the preferred method of aggregating audio FFT data given variable-size frequency bins?

I am coding a program which uses a FFT to process audio data. I need to aggregate the FFT output into three bins ("bass", "mid", and "treb") determined by two arbitrary frequency values. The end-goal ...
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106 views

Formatting issue WAV - MP3?

I have two "sources" of audio: record WAV audio files (44.1kHz, 16bit, stereo) with Linux arecord and convert them with LameMP3Encoder to MP3 files (44.1kHz, 320kbps, 32bit**, CBR)*** record MP3 ...
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140 views

Algorithm used by mironsets

I was curious to know which algorithm(stepwise) is used by the mironsets function of the MIRToolbox library to find the onsets in the audio signal. I am talking about the default functionality when we ...
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2answers
274 views

can MFCC coefficients alone be used to do voice detection?

Aim of my project is to differentiate between cases of silence, single active speaker and multiple ( simultaneous ) active speakers using as basic ( but effective ) logic as possible. Is it possible ...
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72 views

Filter for audio sampler

An audio sampler (in "music" terms because I'm not experienced in dsp) is a software which loads an audio file and allows you to play it at different pitches by slowing it down and speeding it up. ...
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151 views

multipath propagation

i recorded a tone between two devices and applied cross correlation on the recorded and the sent signals. I want to know if there is a multipath effect on my result (on MATLAB). what can i use for ...
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94 views

Normalisation of delay-addition comb filter implementation

I'm trying to write a tempo detection algorithm for analysis of audio samples. I'm roughly following the approach described in Scheirer, the first stages of which give me some very nice beat onset ...
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305 views

Comparing two wav in matlab using NullTest

i have a legacy processed audio wav file. Another UN-processed copy of the source is processed by my algorithm. From both processsing blocks i get processed wav files as output. Now i want to do null ...
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1k 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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132 views

MFCC coeficient comparision

I have two sets of MFCC coeficients which I get with fftExtract utility from two wav audio files. How can I compare this sets (matrix 271x13 and 551x13) if they are similar ? Probably I should use ...
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560 views

3 Band Wavelet Transform In MATLAB

I am currently working on an audio watermarking project in MATLAB. I currently have a code I am using to construct a nxn 3 Band Wavelet Transform matrix. However, when I try to construct a matrix that ...
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140 views

How to measure the short-time change in an audio signal?

The choice of frame length for calculating the Mel Frequency Cepstral Coefficients of a signal, depend on the how much the signal change over a small time scale. I guess that human speech change ...
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1answer
36 views

From pressure to SPL : microphone calibration using sound level meter calibrator and sound level meter

I have a microphone + signal conditioner with adjustable gain and filter. I capture amplified signal using oscilloscope. I'm using this signal to compute equivalent continuous sound level With signal ...
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1answer
108 views

Phase distortion of frequency warping by using Praat

I am trying to do speaker conversion by frequency warping method. I modify the function Spectrum_shiftFrequencies() in Praat by scaling the FFT spectrum of a voice. When one scaling factor is ...
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1answer
967 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
3k views

comparing 2 audio files in c#

I have 2 audio .wav files recorded by the the same person with the same word. Each audio file has a format of 16bit, 44.1kHz and is a mono channel. I would like to know how to compare these 2 audio ...
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1answer
65 views

Counter design for audio signals

I am designing a schematic in which a Counter is to be increased whenever a particular band of frequencies is received at the Audio input. We are receiving the audio input as electrical Signal and I ...
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23 views

How can I append time series data into a finite array with a vanishing point?

I'm not sure what this is called in DSP terms, but I'm looking for a way I can append a new time series data point to an finite array and have it decay slowly over time. The best way I can visualize ...
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1answer
381 views

Preventing clipping in audio signal processing

I'm developing an FPGA upsampler for my DAC project. The last thing I can't figure out is how to prevent clipping which can obviously happen during the band limited interpolation. And what should I ...
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1answer
1k views

Changing voice characteristics so it sounds like someone else voice

I have a recording of speech, and I want to manipulate the voice, so a listener will perceive the manipulated voice and the original one as belonging to two different people. The manipulated voice ...
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1answer
746 views

Invert soundsc function in Matlab

Does anyone know what is the inverse function of soundsc in matlab. I'm playing a sound with a my PC microphone using soundsc, ...
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1answer
104 views

phase amplitude transform as a solution to reverberation

I am trying to use generalized cross correlation with phase transform (GCC_PHAT) on some audio processing indoor, with the existence of low noise. I ve read here that by canceling the magnitude factor,...
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2answers
123 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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2answers
210 views

Linear Predictive Coding and Block Size

I'm experimenting with linear predictive coding as a low-complexity audio compression method, and I'm observing something very odd. I'm compressing a 20 MB mono 16-bit PCM song in different block ...
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2answers
685 views

sound classification

Hello I am trying to do sound classification in matlab. I have different samples of sounds for 2 seconds. How can I proceed with that. The sounds I am using are churchbell, footsteps, trains, sirens ...
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1answer
549 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.