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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640 views

Record and playback has echo/feedback

I didn't mess around with the code here, it was straight from their website, I just want to see if it would work. However, whenever I run it, it has feedback in it, sometimes I can't hear anything at ...
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1answer
178 views

Complex numbers in $\tt ifft$ of an MMSE amplitude estimator

I am trying to reconstruct a signal from a noisy speech using an MMSE algorithm proposed long time ago by Ephraim and Malah (1984). After going through the algorithm, I got a matrix A which represents ...
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1answer
72 views

resolving clipping audio issues

I have implemented a pre-emphasis filter by the following (pseudo) code: a = 0.5; s1[0] = s[0]; for (n = 1; n < N; n++) { s1[n] = (s[n-1] * a) + s[n] } The ...
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1answer
723 views

How can I enforce aliasing with Audacity?

I would like to hear how aliasing sounds like. How can I get this with Audacity? (I was told that I should adjust the Fourier coefficients in Audacity. I have no idea where I can do so / how I should ...
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0answers
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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1answer
178 views

Help me understanding this source code of a Phaser

I tried to understand the following source code of a Phaser: ...
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3answers
558 views

Choosing the best measure for machine learning on a Spectrogram

I am working on an application to classify Talas(a piece of music with onsets). For that, I have the spectrogram of a Tala, which I want to compare with the test Tala, or compare 2 Talas. For this I ...
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1answer
61 views

Segmenting a sound file before each of its attacks

Like the title says... I want to split audio file where a single instrument is played (not so fast)... I just need to detect the attack phase of the waveform to know where a played note begins. Can ...
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0answers
150 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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0answers
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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0answers
304 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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1answer
74 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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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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1answer
1k 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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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
508 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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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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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
35 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
34 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
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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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
964 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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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
165 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
52 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
500 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
150 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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1answer
41 views

How convert Given DBFS into DB in iOS?

I am getting DBFS value of an audio ranging from -160 to 1. I am looking for to create a SPL meter in iOS to show actual DB values of audio capture through mic in device. Using recorder class, I'm ...
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2answers
110 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
282 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
194 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
368 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
546 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
43 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
27 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
503 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
255 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
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
68 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? ...