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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8 views

Processing of speech signals using AMR-WB codec

I have 2 questions related to AMR-WB codec : AMR-WB codec processes the $50-6400\textrm{ Hz}$ frequency band of the wideband speech signal as seen here. Does it mean than it removes the frequencies ...
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27 views

Extracting features from songs for a beginner [on hold]

I' am working on an machine learning project in MATLAB which requires me to extract features from songs. I'm a complete beginner to signal processing and frankly I'm a little overwhelmed. I' am fine ...
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11 views

MFCC time effect

I am just starting out trying to understand MFCCs in context of an audio recognition (specifically birdsong recognition) problem. I have 10-30 seconds of recorded audio containing bird sounds. Say ...
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0answers
32 views

Audio - TDOA and $\tt xcorr$, find the 'best' peak

I try to locate a mobile device using multilateration. The mobile device acts as the receiver and in the room I have three speakers. To calculate the position I need the time difference of arrival (...
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44 views

How to estimate THD?

I have a signal source that I can digitise and read. It produces mainly a 50Hz sine wave, but there is distortion as well. I'd like to estimate the total harmonic distortion for this signal. I've ...
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0answers
25 views

MATLAB - Stereo signal wav file generation with time and gain changes per channel

I need to create short bursts of sound, ideally clicks, where one channel is out of sync with the other (left and right) either in terms of time or gain. This I need to output as a wav file. How do I ...
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1answer
32 views

Summation of audio tone

Suppose you transmit at a single tone $\omega\textrm{ rad/sec}$ and you have two receiving microphones $i=1,2$ which sample at $r$ samples per second. Assume you have a direct path with coefficients $...
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1answer
23 views

spectral subtraction for stationary noise

my recording was suffering from stationary background noise. I used spectral subtraction to remove it. the noise is reduced in loudness but not eliminated completely. before: http://vocaroo.com/i/...
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1answer
84 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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25 views

Real-time STFT masking

When doing offline audio processing (example : template-based noise reduction) I can use this classic method : $$\boxed{\textrm{Input}}\longrightarrow\boxed{\textrm{STFT matrix}}\longrightarrow \...
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1answer
36 views

Parametric EQ vs multiple bandpass filters

Will filtering a signal through multiple bandpass filters, where each filter is passed the complete signal, and then combining the output have the same result as a parametric EQ? What if the signal ...
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29 views

generate signal with multiple frequency

I'm trying to create a multichannel multi-frequency signal. I've seen some code to generate a signal with only one frequency , but I need to have a signal with two or more frequencies simultaneously. ...
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1answer
51 views

Trouble Implementing Huggins Binaural Pitch

I'm trying to make an implementation of the Huggins Binaural Pitch illusion, which is where if you play whitenoise into each ear, but offset one ear by a period $T$ that it will create the illusion of ...
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0answers
18 views

Difference between Envelope filters and a unit with variable envelope response

Would a unit with variable envelope response shape a square sound wave into a sine wave? I know that an envelope filter kind of does that through subtractive synthesis. I'm trying to find a way to ...
6
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2answers
860 views

Can you turn a square wave into a sine wave using a low pass filter?

And if it could, would it make the sound of the square wave thinner than before because of losing its harmonics?
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0answers
32 views

How do we detect presence of background noise in audio?

The problem I am facing to solve is detect background hissing or humming sound in any audio stream. The audio stream can be of any type ranging from movies to talk shows. The background noise is ...
0
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1answer
36 views

How is a PCM byte converted to a voltage level?

16-bit signed PCM would have possible values of ~-32,000 to ~+32,000. How is this value converted to a voltage level for the speaker? If a DAC outputs 5VAC, the values would be ~+-5v. How is the ...
1
vote
1answer
41 views

One-bit depth audio

I have a music player which is only able to play sound with one a bit-depth of one. I can produce this by taking a song and simply boosting the signal $+100\textrm{ dB}$. Is there another approach ...
2
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1answer
168 views

FSK analysis: only ever find one frequency

I've trying to write this software as part of my amateur radio and SDR hobby for analyzing various two-level FSK signals. I'm very new to DSP but have been having a lot of fun so far learning more ...
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0answers
80 views

detect what is noise in audio signal

I'm trying to filter noise from an audio signal. The signal was and is going to be recorded in a city environment which always contains some white/pink noise. At this moment I'm using two options to ...
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0answers
36 views

Low quality mobile phone calls

I am having problems with low quality mobile phone calls when submitting a audio file through it. I am sending the calls with a VoiceXML compatible provider and it seems that the mobile compression is ...
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0answers
30 views

Can I use fpcalc to prove two audio files produce the same sound?

I'm using fingerprinting techniques to detect duplicate audio files on my music library. The first and easy step is to compare the file-size and if they match I use a md5 sum to compare them. The ...
1
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1answer
91 views

Compensating Loudspeaker frequency response in an audio signal

I have been working on a project in which I was required to work on the audio signals recorded from the loudspeaker kept in front of a filter. So, to simply explain it: $$\boxed{\rm LoudSpeaker} \...
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1answer
63 views

How to remove wind noise from video file

I've recorded video footage from a recent motorcycle trip using a helmet mounted Go Pro Hero 4. The video is good, however wind noise is drowning out the sound from the engine. Is it possible to ...
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1answer
26 views

Cascading FM synthesis operations

I'm trying to understand how to program cascading FM synthesis operations. I can find a good deal of information out there on how to do it on various pieces of hardware, but unfortunately, there isn'...
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1answer
48 views

Spectral plot shows more notes than there really should be

I was experimenting with sound analysis lately and from what I see when I plot spectral data of an audio file is that apart from notes that were actually picked there are some other notes with quite ...
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0answers
29 views

Improve $\rm dB$ level after upsampling audio

I am doing upsampling from $44.1\textrm{ kHz}$ to $48\textrm{ kHz}$ on android device, using $\rm sinc$ interpolation. The output $\rm dB$ level is upto the mark for wave file up to $15\textrm{ kHz}$....
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1answer
38 views

How to calculate Long Term Average Spectrum of an audio file?

I'm working with audio files and I have to find their long term average spectrum. Although I had an option available in the software Praat but I don't understand the algorithm of finding the spectrum. ...
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0answers
33 views

Synchronization of two audio tracks via Mel-frequency cepstral coefficients (MFCCs)

I am trying to solve the (common) problem of synchronizing (i.e. measuring the delay between) two audio tracks recorded from different sources during a conference talk (about 1h of audio). One track ...
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0answers
9 views

STM32 Discovery dev platforms [closed]

I'm still researching this but though I'd post asking about experiences. Looks like st.com offers a variety of STM32 Discovery dev boards for ARM CPUs to, among other things, play with DSP. But ...
3
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4answers
143 views

How to calculate an azimuth of a sound source in stereo signal

Let say we have a stereo recording of only one source. What is a correct algorithm to calculate its azimuth using inter channel time difference and inter channel level difference?
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1answer
44 views

Bandlimited PWM

Trying to figure out how to synthesize a Bandlimited PWM in my audio code in C. I have already figured out a bandlimited pulse wave but I can't really figure out how to translate that to PWM. I have ...
0
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1answer
46 views

How do I measure frequency of an audio file into an array?

So I know how to plot fft the audio file within Matlab and a window pops up. My question is, what next? I want to know how to go about measuring the frequency(s). I have two similar audio files, of ...
3
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1answer
70 views

How to do continuous signal processing (i.e without windowing)?

I'm working in speech recognition research, and I wonder if there is a way to analyse sound just like how ears do (i.e. without windowing), so for example to take some feature continuously with an ...
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0answers
30 views

How to calibrate two microphones to same frequency range

I am recording stereo with two same microphones but they are physically insulated from eachother and have some additions on sound input side. Not to complicate story, I have a human head simulation ...
5
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1answer
67 views

Bandwidth expansion by computing missing harmonics in music?

In old recordings or highly compressed audio files (lossy), the higher frequencies are lost, and we hear a distinctive muffling effect. (Just like if the song was played through a wall) Is it ...
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1answer
50 views

How to overlap and add the frames after applying some modification in time domain

I am trying to apply Dynamic range compression (DRC) in time domain in MATLAB. But I am having problem in overlapping and adding the frames. Here is my work: ...
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2answers
97 views

Problem with Generalized Cross Correlation (PHAT filter)

I have to calculate time delay between two signals.I am using the GCC-PHAT method. First, I cut the signals into smaller parts (~0.5s) and then try to calculate the delay. When I display those two ...
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2answers
37 views

I want to know if there is a practical way to find out if multipath propagation is significant for audio in a given environment?

I am employing audio signals generated by a mechanical press for Blind Signal Separation. But, prior to algorithm application, I need to clarify if the signal strengths coming in multipath (except ...
0
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1answer
34 views

Specs for creating sample set?

I would like to create a set of audio samples for playback as a midi-controlled instrument. Thus: a sample for every key of the piano range, with multiple volume timbres for each note. I am able to ...
0
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1answer
51 views

How many DFT bins must I use to get best precision when analyzing melody scale

I need to analyze some signals containing chords. How big DFT spectrum is needed to identify each possible musical note in such a signal sampled at 44100 Hz? I can see that 8192 bins gives some ...
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2answers
42 views

Sampling frequency of audio

Based on what we choose sampling frequency when sampling some song? Do we use Nyquist–Shannon sampling theorem, or maximum sampling frequencies are pre-defined, so it will never come to aliasing ...
0
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1answer
72 views

Audio Broadband Noise Detection

I have to quantify the broadband noise of an audio track. I've seen lots of things around and the only viable option to estimate the broadband noise (without having any information about it) is ...
1
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1answer
55 views

Best DSP algoritms for ultrasonic background noise cancellation

I am looking into detection of pressurized gas leaks by means of the broadband hissing sound (white noise) generated from pressurized leaking gas. The instrument will have a broadband microphone ...
0
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1answer
77 views

Slope of 2nd order low pass vs band pass Butterworth filters

I have noticed that some manufacturers of audio processing products label a 2nd order Butterworth low pass as having a slope of 12dB per octave, but then also label a 2nd order Butterworth band pass ...
1
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1answer
54 views

Looking for low-noise audio data to benchmark some noise-processing techniques

I'm looking for a source of very low-noise audio data for use in benchmarking various methods of random noise correction methods. My intention is to have a very clean signal to which I can add known ...
0
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2answers
45 views

Combining channels into single FFT left/right vs mid/side

I am creating a spectrum analyser to visualize some stereo audio data in real time. I want to display a single spectrum for the input and a single spectrum for the output. The stereo data is the usual ...
0
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2answers
51 views

How to do the highest resolution FFT?

I wish to do the FFT of streaming audio data coming into a computer. I am just interested in frequencies from 80 Hz up to 1000 Hz but with a frequency resolution of 3 to 4 Hz and no more. Basically ...
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2answers
65 views

Why is this FFT code not working to extract frequencies from a wav audio?

I'm quite new to audio processing and i'm trying to extract frequencies from a wav file. As research papers stated i put the wav audio into a byte array and segmented and trying to apply FFT to get ...
2
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3answers
78 views

Performing DFT of streaming audio problem. Is there a limit?

I am trying to write software that will perform the discrete fourier transform of real time data coming from the microphone into the sound card on a computer. I am using Java with the javax.sound APIs....