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.
1,148
questions
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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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1answer
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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2answers
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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0answers
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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1answer
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 ...
0
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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 ...
0
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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
42 views
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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votes
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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votes
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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votes
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:
...
-1
votes
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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votes
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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votes
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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votes
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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votes
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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votes
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 ...
-1
votes
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 ...
-1
votes
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? ...
-1
votes
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, ...
-1
votes
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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votes
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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votes
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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votes
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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votes
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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votes
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:
...
-3
votes
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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votes
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.
