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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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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1k views

discrete Haar wavelet transform, fast and efficient method?

I'm working on my own implementation of the discrete Haar wavelet transform, I understand the wavelet theory and how to construct the Haar matrix of size N to perform the transform, but obviously ...
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504 views

Audio Separation of .wav signal [duplicate]

What I am attempting to do is take a signal (.wav audio file) which contains calls from different Bat species. Here is a graph representation of the signal: What I have done so far is: 1) Emphasise ...
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131 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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490 views

Complex Cepstral Transform on Audio in MATLAB

Hello I am trying to computer the Complex Cepstral Transform of an audio signal. I am using wavread to get an array of data type double precision. However, when I ...
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0answers
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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0answers
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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2answers
60 views

Parallel equalizer, working for boosts but not cuts

I am trying to replicate the behaviour of a parallel equalizer. The approach is to use a parallel sum of bandpass filters like so: $Response = 1 + bandpass1 * (gain1 - 1) + bandpass2 * (gain2 - 1) + ...
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1answer
28 views

How to find out that a wav file is edited or original in Python? [closed]

I want to verify that if a audio (.wav or any format) file is original or edited. How can it possible in python? Here, "edited" means modification of audio file using software. For example, for an ...
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1answer
32 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
32 views

audio analog input + usb data cable = sound noises [closed]

I connect my android smartphone from audio jack output to my pc audio input and get a good analog sound signal. But when I connect this smartphone to the pc with usb data cable the analog sound ...
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1answer
105 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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2answers
2k views

Audio signal processing

I am trying to code audio signal processing software in java. Purpose: It will use audio signal processing library to filter out noise from the raw PCM based input (.wav) of heartbeat recording. ...
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1answer
946 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
64 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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2answers
84 views

Real-time audio : passing by value [closed]

In real time audio, one cannot program the way they usually do: with new delete or even malloc: it introduces glitches when creating memories. By if I pass by ...
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3answers
134 views

Does 1 kHz sine tone means $\sin(2(1000)\pi t)$ or $\sin(2(500)\pi t)$?

Does 1 kHz sine tone means $\sin(2(1000)\pi t)$ or $\sin(2(500)\pi t)$?
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2answers
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Can I slow down an audio file by halving the sampling rate?

A question came up, and as a layman I am not sure how to appropriately find an answer; bear with me. Given a 2 minute audio file, say a .wav file sampled at 44,100 Hz, I am curious if it is possible ...
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2answers
476 views

Levinson Method

i have just one question about Levinson, my question is: can we apply this method to determine the coefficients for FIR filter? if yes, please help me ! this is an example: ...
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3answers
144 views

Removal of gradually-increasing dc component (offline, zero-phase)

A program I'm writing produces a signal with a dc component that increases over time. I need some kind of post-process which can remove the dc content, without affecting the phase or magnitude of the ...
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1answer
2k views

What does the graph of audio signal tells? what are negative samples? [closed]

What does the graph of audio signal tells? what are negative samples?
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2answers
86 views

how to remove noise from heart sounds?

i have .wav files acquired through putting android cell phone MIC directly on chest of person.For feature extraction, i have to made them noise free. These recordings were taken in rooms with no noise ...
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3answers
262 views

Features Used for Instrument Recognition?

I'm building a neural network for instrument recognition based on the IRMAS data set. However, I'm a bit at a loss for what features to use. MFCC seems to be quite popular but extracting this per ...
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1answer
57 views

How do I efficiently process 20+ minute audio tracks? [closed]

I have built something that reads in audio files and processes the information for features. Sometimes the audio is rather long, and obviously I cannot keep all of this information in RAM. How can I ...
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1answer
58 views

Can someone please help me out with audio, sampling frequency question

I really having a hard time to tackle the following question. It is asked to load a ".mat" file using load file.mat. And the result is two variables: Fs varaible ...
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1answer
490 views

How to detect and remove ringback-tone and ivrs voice etc from the beginning of an audio call recording

In an audio recording (say a telephone conv. b/w two people), how would I programatically detect and remove the dial-tone at the beginning of a call using python. Ex : As you can see the first 15 ...
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1answer
190 views

Save Matlab audio output [closed]

I need to save the audio output of a script in Matlab but I don't know how to do it. It is a complex sound, with several pauses of silence with different durations, so I cannot simply use audiowrite (...
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1answer
359 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: ...
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1answer
545 views

Getting the average pitch of a wave in C#

I want to separate out the components that I need to evaluate the pronunciation of a user, so I am dealing with the energy/stress/volume and intonation or pitch. Focusing first on the pitch, is ...
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1answer
40 views

How to separate silence from frequencies in Matlab

For a project, I was asked to create a DTMF detector on Matlab. I know that the audio sample that will be provided will have 4 tones maximum, possibly less, and that the audio begins with a 0.1 second ...
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1answer
24 views

Is there a software tool or web service to detect problems in speech recordings?

I am looking for a software/library or a web service/API that can detect problems in short samples of recordings of speech. For example echo, clipping, pop, background noise... Do you know of any ...
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2answers
483 views

Amplitude ranges for different bit-depths

In Python, I am using the library Soundfile (which uses the library libsndfile) for digital sound processing. I am working with different sound files coming from different internet databases, and ...
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1answer
238 views

Synchronizing 2 seperate speech audio signals

I have two speech signals: one with a man reading a passage, and one with a woman reading the same passage. I want to synchronize these signals so that, for each word, their readings line up. The ...
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1answer
46 views

Problem in Processing .aif file in Matlab

I am trying to use the following crossfading code in MATLAB. I use audioread() to open the audio signals and I convert both of them to mono. When I use .wav files, ...
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1answer
368 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
67 views

Programming a peaking filter (and how do I FFT?) [closed]

I want to get into audio signal processing. I know how to generate a tone, but I have no idea how filters work. What do I do to a set of samples to boost/cut a frequency with adjustable bandwidth? ...
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1answer
742 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
408 views

Confusion when applying Hamming window to signal

I am writing some code which takes a signal of length $n$, and processes it in 20 ms chunks with 5 ms of overlap... So basically I have a buffer of 20 ms worth of sampled data, and I am trying to ...
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1answer
102 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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1answer
41 views

audio classification based only on first few seconds

I record audio notes on my recorder, and each belongs to a certain category ("work", "parenting", "miscellaneous", etc.). At the beginning of each recording, I say the name of the category, then stay ...
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2answers
162 views

processing 16bit audio files [closed]

I currently have a 16bit mono file. I would like to read this data to retrieve only the data chunk part and not the entire file with the headers so that I can apply fft on it. I tried finding ...
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2answers
209 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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1answer
333 views

Comparison using cross correlation [closed]

I have 2 stereo audio files to compare. Comparison of files include change in frequency, amplitude, phase shift Code: ...
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2answers
674 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
542 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.