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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52 views

Generating a high SNR sinusoide with lut on a DSP

I'm generating a sinus by using a lut method on a DSPic33f. My sample rate is 48 kHz, so I saved 12000 of the first value (unsigned int, 16 bits) and use trigonometric formulas to calculate the other ...
2
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1answer
78 views

How to model Tape Saturation (Audio DSP)?

I'm looking for info that would help me to build a Tape Saturation process. I'm working with the WebAudio API. The API provides a number of DSP nodes that perform basic processes, such as a Convolver ...
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1answer
50 views

How to remove background noise from sound file for analysis

I'm working on a project at my university and it involves background noise recognition/classification. I'm very new to this so I'm unsure what to really search/google or where to really begin any ...
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0answers
21 views

Constant Q transform and time synchronisation

I'm looking to use the constant Q transform for onset detection and am having a bit of trouble aligning the time axis of the transform with actual note events. I'm using the CQT transform toolbox ...
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1answer
63 views

Filtering out unwanted square wave (Radio: DCS/DPL signal)

I've need to remove a DPL/DCS signal from a recording. Roughly speaking, a DCS signal is a low frequency (67.15Hz) square wave sent at the same time, over the same carrier, as speech. Because it is a ...
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0answers
34 views

Ads detection using MFCC and DTW

I am doing a project to detect Advertisements from transmission by using clipped segment (slogan) of an advertisement, based on looking at the audio track as follows: Audio Signal $\rightarrow$ ...
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2answers
94 views

Created Matlab echo signal: Is it an actual “echo?” Or am I just manipulating the signal to make it seem like an echo?

This is just an outline of what I did (assume that all my dimensions fit): ...
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2answers
64 views

Recognizing bodily noises (ie, not spoken words)

I was pointed here by a chain of stackoverflow posts. I'm a software developer looking to make a module that recognizes noises like snorts and coughs. Here was my original question on stackoverflow ...
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3answers
94 views

Measuring audio Signal Similarities

i have recorded samples of lungs sound (person breath) signals from electronic stethoscope. how can i measure(in percentage) the similarity between two audio signals (lung sound), the general idea is ...
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0answers
34 views

How to synchronise two audio files using a marker

I'm working on a project on which we use WebRTC and the RecordRTC library to record audio files. I would like to be able to process the audio files server-side to ensure a good synchronisation. To ...
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1answer
46 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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1answer
137 views

Feature extraction for sound classification

I'm trying to extract features from a sound file and classify the sound as belonging to a particular category (eg : dog bark, vehicle engine e.t.c). I'd like some clarity on the following things : 1) ...
0
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1answer
36 views

Is there a way to break an analog sound signal up and send it through two radio chips at double the data rate of one?

Here is a radio chip for audio that sends at a CD-level data rate. Could you sample an analog signal (for instance from a guitar pick-up or microphone) in such a way that you could transmit using two ...
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1answer
55 views

FFT filter to remove cd scratch

I have a .wav file which came off of a cd with a scratch, so there is a periodic "clicking" of about 6 Hz. I was playing around with trying to squash the clicking by taking FFT and filtering the ...
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2answers
44 views

Real-time audio : passing by value

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 ...
17
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4answers
1k views

What are the audio equivalents of images like “Lena”, “Mandrill”, and “Cameraman”?

I don't know what to call these images, but they're very often used in the image processing world as standard examples to show the output of a particular operation. Here's a link to the page ...
2
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1answer
98 views

How do you change pitch with wavelet transforms in MATLAB?

I'm trying to increase the pitch of a wav file using MATLAB. (I don't expect it to sound very good, but I'm trying to figure out how wavelets in MATLAB work.) Here's what I'm doing: ...
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2answers
57 views

noise cancellation of non-repetitive noise

There are a new noise cancellation headphones that almost fully cancel human noise. What kinds of algorithms could achieve that ? EDIT:by human noise , i mean talking. In general, previous audio ...
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2answers
87 views

When should we using moving average in algorithm design?

I'm new to signal processing. And I just read G.720.1. I find there is a lot of moving average used. And I can also recall there is a lot of moving average been used in other audio processing ...
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1answer
89 views

How can I load a wav file, split it in frames, apply different filters do different frames and then recombine the filtered frames into a single file?

I'm working on a wav audio file using MATLAB. What i would like to do is: load a wav file and divide it into frames of the same dimension filter these frames (with a different filter for each ...
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1answer
43 views

How to implement simple noise filter based on the frequency domain information?

Brief introduction: I am working on my university project. My task is to develop application that captures sound from mobile phone and removes/reduces noise (this simpilied version of task). I never ...
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1answer
48 views

how can I analize music as spotify/musixmath/shazam does?

I mean, does exists some state of the art algorithm able to extract music features, classify them and identify similar music or find lyrics? I know that there are many topics or steps involved on my ...
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1answer
51 views

What does it mean for audio to be 'sped up'? How can I undo it?

I very simply have a .wav audio file, but there is a note with it that says "This audio has been sped up by 10x". I am wondering, what does it mean exactly from a signal processing perspective for an ...
3
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0answers
53 views

1/n octave complex smoothing

An excellent answer to this post explains how to do 1/n octave energy smoothing and mentions complex smoothing can be done as well but it's tricky business because of phase wrapping. So how is 1/n ...
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2answers
95 views

Audio distortion while implementing a low-pass filter

I attempted to implement a real-time audio filtering in a callback function using the simplest moving-average filter with coefficients [0.5 0.5] : ...
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0answers
29 views

Speech features that can be inverted?

I am trying to figure out speech features that can be inverted to recover the audio back with no loss or as less loss as possible. I am currently using the power spectrum which is completely ...
2
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2answers
52 views

Using FFT to Unit Test a Music Player

I'd like to validate changes to our app music player in our Continuous Integration unit tests (more like integration tests, really). My thoughts were to start with an audio file that contains only a ...
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1answer
42 views

High energy coefficients in 1d DCT for audio

I've been reading a little bit about how DCT works in image compression using the 2D transform, however I don't know how this applies to audio using a 1D transform. Suppose I have an audio segment, I ...
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1answer
79 views

how to eliminate frame boundary discontinuity?

I'm applying at each 2048 samples the pitch synchronous algorithm to pitch scale some audio files, everything seems to occur well, but I've noticed some glitches, artifacts, clicks between every ...
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0answers
25 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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2answers
64 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 ...
0
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1answer
57 views

Pitch Detection: HPS not working

I'm trying to code the HPS (harmonic product spectrum) algorithm and the problem I'm facing is that using my guitar, the fundemental frequency (Here it's 82Hz) just isn't there, so is the 5th Harmonic ...
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0answers
36 views

how does a karaoke work? [duplicate]

in audio signals the extraction of the background score , isolating the speech is known as the karaoke. if i use a simple band reject filter which will eliminate the spectral components in (0-4)KHz, ...
0
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1answer
32 views

Audio Beat Divide

How exactly is beat division applied to a signal? Does the rhythm need to be detected or can just the onset be manipulated? For rhythm detection is it better to look for patterns in the frequency ...
0
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1answer
52 views

How to frame an audio sample?

I am trying to calculate the mfcc of an audio sample of 5sec duration. How can I frame this audio sample. Is there any criterion for selecting the frame size and number of frames? What all change will ...
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2answers
51 views

how multi track recording is performed? [duplicate]

I have a doubt regarding multi track/multi layer recording of audio signals. in recording studio each instrument or chorus,playback will use a independent layer/track..so after recording a all tracks ...
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0answers
28 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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0answers
19 views

ADPCM: Theory and HIghest Quality Algorithm(s)

Two questions. What is the highest audio quality ADPCM algorithm(s)? IME ADPCM is quite simple, but would I be correct to assume that a multi-sample algorithm (where the previous two samples are ...
2
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2answers
95 views

Low-Complexity Audio Compression?

What options exist for discrete audio compression that is good quality, moderate compression (e.g. 3+:1), and low complexity (e.g. hundreds of processing cycles to encode each sample without the use ...
0
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1answer
75 views

Does audio upsampling create noise/artifacts or degrade the signal?

I've read that upsampling performed in digital music playback can color the sound, produce artifacts, etc. For example, an audio file ripped from a CD might be 44.1Khz/16-bit, and then upconverted to ...
0
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1answer
96 views

Band Limited Impulse Train Synthesis

I'm currently working on a piece of DSP/Synthesis software and I have run into the issue of generating band limited waveforms. I have been doing some research to try and find a good solution but I ...
0
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1answer
40 views

Finding Accurate BPM and Beat Time Values in Audio

I working on writing a program to get the BPM rate and the Beat Times in audio songs but I am having trouble trying to come up with reliable testing strategies for my design.What do I benchmark my BPM ...
0
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2answers
48 views

Controlling audio high/mid/low in frequency domain

Is it a realistic mapping to divide the frequency bins in 3 parts, high/mid/low frequencies? How can these bins then be manipulated to control the level of these bands in the time domain? Is it ...
0
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2answers
35 views

Wavetables and Partials

I am reading Direct Synthesis versus Wavetable Synthesis. The quote below explains code I have read but I don't get why it's a division by the integer multiple, when multiples increase in frequency. ...
0
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2answers
53 views

Extra energy when altering frequency amplitudes

How can I convert a time signal (audio) into the frequency domain, multiply frequency amplitudes by a vector of scalars, and convert it back into the time domain, without messing up timing information ...
1
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1answer
84 views

How to get Envelope of Audio Signal

I am having a problem understanding how the author gets the onset envelope of an audio song in this paper. The entire section in question has been quoted below: First the input sound is resampled ...
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3answers
89 views

How is data compression possible for audio?

I understand that compression works by exploiting certain psycho-acoustical characteristics of our auditory perception, but I'm still confused. I mean ultimately, the waveform must be recorded ...
-1
votes
1answer
113 views

How can I reduce noise errors when detecting DTMF with the Goertzel algorithm?

Thanks a lot to whoever downvoted my question for no good reason. I wrote an implementation of the Goertzel algorithm in JavaScript and made a demo that detects DTMF dial-tones. You can find the ...
0
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1answer
58 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 ...
0
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1answer
172 views

Obtaining beat histogram using Tzanetakis algorithm

I am working to make a music player with automatic genre classification application in Java. On Google searches I mostly found the works of George Tzanetakis who describe an algorithm from Music Genre ...