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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1answer
32 views

State-space filters in fixed-point?

I am having difficulties implementing state-space filters in (32-bit) fixed-point. This is because the coefficients have a huge dynamic range, for example (MATLAB code): ...
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0answers
26 views

How to send data without changing its origional form using actual modems?

My modem supports only FH-BFSK modulation. I have a CDMA signal which I want to transmit through modem. Is that okay if i convert my signal to whatever form of modulation supported by modem and then ...
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1answer
75 views

Designing a Chirp in the Frequency Domain

I am trying to create a chirp signal in the frequency domain, by simply setting the magnitudes of each frequency bin to one, and then setting the phase delay function such that the resulting time ...
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2answers
104 views

What is the current “state of the art” in future audio sample prediction, given recent developments in “image inpainting” using deep learning?

In the last few years, there have been many breakthroughs in the image processing world regarding repairing "damaged" images, images with corrupt pixels, or even reversing artefacts from ...
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5answers
111 views

Effect of One Sample Latency in Audio Application

I am dealing with a customized processor that has a processing latency of one sample (20.8 microseconds) at 48kHz Sampling rate . What I would like to know is ...does this latency of one sample have ...
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0answers
23 views

How can I perform sound classification based on spectrograms and waveforms?

I noticed that almost all machine learning enabled sound classification approaches use spectrograms or MFCCs as input for feature extraction. I am new to the area and would like to know if we can also ...
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1answer
36 views

Testing if a signal contains a specific previously recorded signal

I am working on a project that converts radio data and transcribes it into text looking for specific types of radio chatter. For most of the time, the radio signal is dead air. Periodically, there ...
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0answers
11 views

short frame subband coding [closed]

I am implementing a subband ADPCM codec, Fs=16 kHz, frame length 3ms, band1=50Hz-2kHz, band2= 2kHz-8kHz. I used the FIR filter for subband coding and %50 overlap. My reconstructed signal does not ...
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0answers
30 views

Newbie want to self-study to make guitar effect/amp plugin

I am a senior Game developer want to self-study how to create electric guitar amp/effect plugin. I read around and managed to made a simple distortion plugin using JUCE. But because of lacking in ...
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0answers
24 views

Communicate to user that SNR is too low - Speech procesing

I am working in a project to estimate the signal to noise ratio (SNR) for an automatic speech recognition system in order to generate an alert to the user if the noise is too high in the place where ...
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0answers
21 views

how to convert decay time to power law coefficiants

In audio, I am familiar with the definition of t-60, which is a characterization of a room impulse response (RIR) decay time. This characterization demands that the sound will have a 60 db decay over ...
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0answers
20 views

Determine loudness values of certain frequencies when analyzing a spectrogram of a sound [closed]

I load a sound into a sound editor and it shows me a spectrogram similar to this: Black color represents no sound, blueish dots - very quiet sounds, red means quite loud ones and yellow represents ...
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2answers
133 views

DFT of pure sinusoidal wave

I'm writing a program in which you can synthesize waves by adding to a sound's Fourier transform, and then inverse the transform to get the modified sound. In order to do this, I need to know what to ...
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0answers
43 views

Monophonic audio processing [closed]

I need to find a tool written in python that given a monophonic audio file as input, returns all played notes' starting time. Let's say we have the "Twinkle Twinkle Little Star" performed on ...
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1answer
36 views

Which codec and format are Audible downloads?

Files downloaded with the Audible Download Manager are stored with the .aax file extension, but the information I can find about this format seem to vary. Regarding this source, the files are stored ...
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1answer
43 views

spectrogram of a leak in a water pipe

I'm doing a work on the acoustic emission of a leak. I am using a 1.80 m cylindrical tube with water, and the leak is simulated by means of a tap that opens after I have applied negative pressure ...
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5answers
6k views

Why can so little digital information be stored on a cassette tape?

I had heard that tape is still the best medium for storing large amounts of data. So I figured I can store a relatively large amount of data on a cassette tape. I was thinking of a little project to ...
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1answer
26 views

What is the purpose of the recurrence relation in low and high pass (audio) filters?

Newbie here. I'm familiar with time and frequency domains from math. I think I understand what low and high pass filters do, sonically. But digging into an implementation of ...
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1answer
52 views

Is this statement correct from DSP aspect?

So, In my report I am comparing the related works, In most of the previous work, researchers have used spectrogram as input to vanilla CNN and classify. Where I am using different handcrafted features ...
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0answers
29 views

Which one is better method and why? Manually Handcrafted sound features vs spectrogram + convolution

I am working on classifying different sounds ( not speech or words exactly something like ambulance alarm, police alarm, cough sounds etc) I read few paper which suggested to extract dsp features such ...
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1answer
42 views

Apply filter before or after Downsampling?

Context: I'm trying to do formant estimation. My first attempt didn't work out all that well so I'm trying to follow closely Praat since their formant estimation is remarkably accurate. Question: I ...
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1answer
42 views

How to get rid of discontinuities at the start of a signal caused by the filter

I'm processing an audio signal using an overlap add method.Everything at the block boundaries is fine. However I have a discontinuity at the very start of the first block and I'm not sure how to (or ...
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0answers
24 views

How to identify the vocal folds in a mel spectrogram image?

I am quite new in the dsp and trying to learn How to identify the vocal folds in a mel spectrogram image? My image data looks like this. How to find the vocal folds in this image? I am using Praat ...
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2answers
51 views

How to make spectrogram smooth?

I am quite new in DSP and generating the spectrogram of an audio file. My spectrogram is not smooth and it is showing the quite raw image with pixel values, something like this While I am looking for ...
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1answer
47 views

FFT results are repeating?

I am trying to detect notes out of a musical sound input. My program can be run here. The problem is the patterns in the frequency response is repeating: I suspect that echoing in the room is causing ...
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1answer
142 views

Generating digitized white noise: uniform vs normal sampling

Consider the following two ways of generating noise in the time domain for audio applications: Generate samples from a uniform distribution [-amplitude, +amplitude]...
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0answers
25 views

Expand and shrink waveform visualization depending on tempo changes

I am not quite sure if this is the right state exchange site or not. Pls let me know. So, I noticed that Logic and Pro Tools, etc. have this waveform shrinking and expanding effect when dragging the ...
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1answer
68 views

Produce musical signal from instantaneous frequency

I'm implementing the Yin Algorithm in Python so as to extract the pitch of my voice when whistling or humming. My goal is to produce the Gate (Envelop Control) and CV (Pitch Control) signals which can ...
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1answer
125 views

Audio: How to measure dynamic range (in sense of dynamic compression/ “loudness war”)

I'm about to calculate some quality measures for audio. One important of them is the dynamic range (in sense of "loudness war"). By which in this context is not meant the SNR but the (naive ...
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1answer
128 views

(Newbie question) Calculating the frequency spectrum

I'm quite new to this topic, with some basic knowledge however. I need to analyse an audio signal for different characteristics. Like the audio editor I use (an older Version of Audition, v.3) I would ...
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2answers
32 views

Audio keeps echoing continuously when using 2 separate devices for audio and recording on video calls

I had used 2 separate devices for a video call. Desktop for video and audio and phone for recording, since the desktop microphone is not working properly. Unfortunately when I started speaking, my ...
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0answers
43 views

Sweep synthesis in frequency domain: How to correctly adjust magnitude?

I'm attempting to synthesize a logarithmic sweep signal to measure the IR of a system. For the most part, I'm following Section 5.2 of the paper Transfer Function Measurement with Sweeps. To generate ...
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0answers
58 views

How to make ultrasound audible?

Because of interesting spectral phenomena, I would like to shift the >20kHz content of my audio recordings to audible frequencies, so that I can listen to it. I tried the Audacity pitch correction ...
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0answers
30 views

Is it possible to open 2 streams from the same microphone on Oboe?

I'm trying to open 2 input streams from the same microphone on Oboe but it's not working. One of the audio inputs I'm opening is like this: ...
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2answers
135 views

A cubic interpolation function: folkloric copypasta or clever trade-off?

I've been reading on interpolation methods recently and I have come across an implementation of cubic interpolation that is leaving my head scratching. Every other variant and example of cubic ...
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3answers
60 views

Sampling a low-frequency sound doesn't work at some sampling rates

When sampling a low-frequency sound (like a 55Hz sine wave) I hear an additional high frequency sound at certain sampling rates, but not at others. This happens in Audacity and in my own program. For ...
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1answer
54 views

Exponential Swept Sine Distortion

I was able to get passable results on fundamental measurements using and exponential swept sine. Now I am trying to get distortion information from the same measurement but am puzzled by the results. ...
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0answers
36 views

What would an algorithm that can target different devices be called?

I'm developing an audio filtering algorithm that can be run in different scenarios, e.g as embedded firmware on DSP Codec/Core, as a software library, or as a cloud service. Effectively, it can target ...
4
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1answer
58 views

Actual Sample Rate slightly off?

When capturing microphone audio, I noticed that my static buffer eventually overflows. So I timed the samples, and low and behold I'm getting an actual sample rate of 44102-ish Hz (my timer is very ...
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1answer
47 views

Algorithms to fix audio feedback loop

I'm a student and I have been assigned to make a presenation on algorithms to prevent audio feedback loop (microphone feedback, that issue when you hear the voice echoed multiple times). I looked ...
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2answers
114 views

How to merge 2 WAV files together? Can I take the average of the 2 audio file float values? [duplicate]

I'm trying to do one simple thing on Google's C++ Oboe library: pick 2 recordings and merge both. That is, make as if the 2 were playing at the same time. Oboe works with Float numbers from -1.0 to +1....
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1answer
36 views

Low Level Audio Engine Audio Data Reformatting - Sample Rate and Bit Depth Conversion

First off, there seems to be a sparse number of low level audio programming tutorials out there so if anyone knows of some good resources I would be very appreciative. Now on to the topic at hand! I ...
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0answers
18 views

Pitch analysis Audio Unit [duplicate]

I am developing an audio application for macOS that I would like to use to analyse the notes in a given section of music. I am happy to work from an analysis of the frequencies in a given section then ...
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1answer
17 views

How do I set window length and shift length for computing STFT on Scipy?

Does nperseg=32, noverlap=16 mean the window length is 32mS with a shift of 16mS? Refer here for more details
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1answer
27 views

Order of operations when creating a spectrogram

Signals noob here! I'm dipping my toes into your world by trying to recreate the Shazam algorithm, and I have a quick practical question: When creating a spectrogram, is it better to take in a whole ...
4
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1answer
69 views

How to do Frequency Scaling on an Audio File

Please excuse me if my terminology is wrong. I'm from a music production background and have no experience in signal processing. I was wondering if it was possible to stretch out the overtones (...
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0answers
38 views

Reconstructing a signal from multiple filtered copies using overlapping tapered windows

I have an audio signal that I need to process and analyse in blocks. I've now decided to process it using overlapping non-symetric Hanning windows of length M = 9600 and a hopsize of R= M/2. I zero ...
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0answers
26 views

How do I change the sampling rate without changing the speed and pitch in Python?

I am building a voice recognition algorithm, and the use case specifies that it should be able to identify the targeted speaker from various audio sources, which means that it must be able to process ...
0
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1answer
24 views

Error introduced by smoothing filter at the end of a processing block

I'm running analysis on a multichannel audio signal and due to the size have decided to process it in blocks (my computer doesn't have enough memory to process it in one go). Some of the data produced ...
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0answers
13 views

What techniques may “State of the art iterative numerical solver for optimally smooth gain reduction” refer to?

What techniques may "State of the art iterative numerical solver for optimally smooth gain reduction" refer to? As given in a description for a particular limiter. But particularly, I've ...

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