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.
1,187
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148 views
How would addition with a constant in time domain affect my signal in frequency domain?
I have an audio signals from two microphones, and I have to make them equal for equal input.
I.e. calibrate the output of microphones to give the same resulting signal on both of them.
The problem is ...
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168 views
Remove microphone rustle / clothing rustle from raw audio data
Thank in advance for the help.
I have several GB's of raw audio data that I have collected as part of a research study. This audio is divided up into individual samples approximately 30 seconds in ...
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30 views
Improving accuracy of Geortzel using simultaneous frequencies
I have written software to detect the transmission of frequency sequences.
I'm using Geortzel's Algorithm to filter desired frequencies from a data array. The data array is generated from audio ...
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1answer
156 views
Cannot understand the attenuation in signal as the audio bit depth of adc changes
I am working on Active Noise Cancellation Project. I came across this strange problem, where the signal is getting attenuated as the audio bit depth changes and I cannot understand it!
My circutit :I ...
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84 views
Enforcing size to match Convolution using 'same' property & Enforcing Circulant Matrix (Like DFT Based Convolution)
The original problem was from this link about coding de-noising an audio signal.
Because my reputation points disabled me to comment, I have to ask a separate question here.
I don't quite ...
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1k views
Using MFCCs in kNN classification
I'm working on adding MFCCs to my kNN classifier.
Originally my kNN classifier was using the zero-crossing-rate, centroid, bandwidth, skew, kurtosis. This was about 80% accurate for drum tracks using ...
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418 views
I m assigned a project to create karaoke system on matlab. But im a noob in Audio processing and i don't know to approach this problem. Please help
Im simulating a karaoke system. I m giving a mono channel audio as input and i should get instrumental component of audio as output(all vocal component should be removed from audio).
i tried two ...
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31 views
Distinguish between two vocals by their timbre
Right now I'm working on a signal processing project, I'm not quite to the point where I'm implementing the DSP algorithms yet (still working on the low level hardware) although I'd like to get an ...
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211 views
Replicating a visual audio spectrum
I've been developing a web app as a hobby over the past few months which is essentially a real-time Javascript clone of the audio visualizer used by Monstercat. While it looks kind of close to the ...
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417 views
Audio block-processing discontinuities
I have an algorithm that looks at an audio frame, calculates a threshold in db-rms, and decides to filter the frame whether or not the threshold is crossed. However, filtering each block is producing ...
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3answers
853 views
Find frequency/phase response of a microphone
I am trying to find the frequency/phase response of two cheap microphones bought off Ebay. Basically, I want to graph the frequency responses and phase responses of both microphones on the same axis ...
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461 views
Dynamic range compression in the frequency domain?
I'm trying to do frequency-band specific dynamic range compression (like multi-band compression) and was thinking of whether dynamic range compression is doable in the frequency domain or whether it's ...
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172 views
DSP Library IIR and FIR filters
Me and a coworker are currently in the process of making an audio dsp library. We are wondering what some good FIR and IIR filters to use were. I know ALOT of them but what do you guys think in best/...
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466 views
Detailed Analysis of Phase Delay of an Audio Signal
I write this question as a new reference point because it is difficult to find something exhaustive regarding this topic.
I'm talking about the calculus and interpretation of the phase delay of a ...
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66 views
Invertible multi-resolution representation of audio/speech signals
What are the invertible (preferably linear) representations of an audio/speech signal which captures the essence of signal in different resolutions and details. Basically, I am looking for Laplacian/...
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148 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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2answers
138 views
Can software distinguish between different musical instruments
If possible, I would like to write a little script that can identify different musical instruments from just a recording. Apart from variations of amplitude and frequency over time (which can be ...
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155 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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122 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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151 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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1answer
241 views
efficient triangle to sine morphing?
I'm coding a synthesis external for MaxMSP in C.
It's based on Peter Blasser's bounds and bounce concept ( http://petermopar.blogspot.co.uk/2014_04_01_archive.html )- basically triangle waves with ...
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109 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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144 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
282 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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1answer
304 views
Psudocode implementation of true envelope
I am trying to implement the algorithm described in this paper, I'll quote the relevant portion: http://recherche.ircam.fr/equipes/analyse-synthese/roebel/paper/trueenv_dafx2005.pdf
Let $V_i(k)$ be ...
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2answers
54 views
Acoustic record and store devices
I am working on a project involving analysis of bird sounds and I want to record these sounds in slightly remote locations. It is enough if the device can only record and store them (maybe in an SD ...
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1answer
675 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
180 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
73 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
731 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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73 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
179 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
564 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
155 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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0answers
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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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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0answers
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
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
515 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
134 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
491 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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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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141 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
61 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
41 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
36 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
112 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 ...
