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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sound classification

Hello I am trying to do sound classification in matlab. I have different samples of sounds for 2 seconds. How can I proceed with that. The sounds I am using are churchbell, footsteps, trains, sirens ...
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441 views

Calculate algorithm's MIPS from cycles?

My audio DSP algorithm processes input sound file (45 seconds, 16 kHz, mono). After simulation is done, gprof profiler tells me: ...
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1answer
200 views

Halving a floating point time series and it's effect on noise levels

Why does halving a floating point time series (audio) cause the noise shown in the spectogram to change? From having a look, all the steps from getting the first half of X, stft() and ...
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164 views

i want to mute a particular frequency in real time audio

i want to mute a particular frequency range in real time audio for eg: in audio signal frequency above 3 kHz should be muted for specific interval of time.
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72 views

Non-uniform resampling - spacing between output samples given speed factors

I'm working on an implementation of time-varying resampling in python. I've got the windowed sinc interpolation itself working nicely, but can't figure out where exactly to sample my input data for ...
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0answers
34 views

Is it possible to detect that sound source is playback of a recording?

I have been trying to find an article where it is described how real-time (or close to real-time) signal processing from a microphone is able to detect that an audio source is playback of a recording ...
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1answer
78 views

Basic sampling excercise - unsure of rationale informing np.arange parameters for time array

I'm extremely new to DSP, so I apologise if it's a very basic question. I'm having issues understanding the following exercise: "Create an array of 201 samples containing a sin wave with a frequency ...
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2answers
101 views

How to minimize peak-to-peak amplitude of periodic waveform through FFT phase shifts?

For one of my music projects, I'm playing back periodic audio signals (by looping single periods). Unfortunately, one of my waveforms sounds too quiet (even at maximum volume). I'm trying to use FFT ...
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2answers
651 views

Removing noise from PCM file [closed]

I want to implement basic noise reduction. I'm recording audio using a microphone mounted on a drone. I would like to filter the audio file from drone noise. What's the best way to do it ? My first ...
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1answer
94 views

Audio texture analysis for visualization

I'm making a complex music visualizer as a personal project. I've already determined that I will be using a series of modified Goertzel algorithms for frequency detection to place the frequencies on a ...
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1answer
244 views

Maximum dB SPL from a digital microphone

Basic sanity-check question: if I have a digital microphone with a sensitivity of -26 dBFS (say, this one), does this mean that the max dB SPL it can handle is 120 dB SPL? My thinking is that if the ...
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1answer
89 views

Does the sound volume affects the MP3 file size?

Let's say I have an X music track in some lossless format (wav?). If I convert it to MP3 format with its original volume and get the file "A.mp3". Then I edit the lossless source and reduce the ...
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2answers
98 views

Approximate the change in perceived gain from applying transfer function to audio signal

I have an audio signal that I am applying an analogue transfer function $H(s)$ to. I would like to approximate the change in gain ($\Delta \text{ Gain}$) as perceived by the listener for an arbitrary ...
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1answer
4k views

How to get the volume level from PCM audio data?

I have a decoded PCM audio data obtained from the Web Audio API. I'd like to generate a chart that represents the volume level (perceived by humans) over time. ...
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1answer
163 views

How do I deepen my voice (I'm a female) to sound like a man in a video for a school project? [closed]

Some extra info. that I think might help or make things clearer: The video has several other people in it also talking (I don't want to change their voices). I have some experience in editing ...
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3answers
259 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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2answers
2k views

What does “Lag” mean in terms of Cross Correlation?

I am working on a program using the Essentia library for audio analysis. I've implemented the Cross Correlation algorithm (here: http://essentia.upf.edu/documentation/reference/std_CrossCorrelation....
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2answers
164 views

Filter away sinusoidal noise properly

I have a stereo music signal corrupted by strong sinusoidal noise that varies over time. Here is the spectrogram of Left channel I plotted with Matlab. As you can see there are 3 or 4 strong ...
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1answer
89 views

How to keep the same power (sonority) between pure sines?

I have a list of sine waves in 1/3 Octave (63,80,100...1k,1.25k,etc) that I want to reproduce with the same power. Right now I just have a global gain that applies to all this freqs, but of course the ...
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1answer
158 views

Calculating gain coefficients for comb filters in Moorer's algorithm

Could someone explain how Moorer calculated the gain coefficients for the comb filters in his reverb? In his book, Udo Zölzer's layed out this equation: $$g = 10 ^ {-3 \left(\mathrm{Reverb \ Time} \...
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1answer
57 views

How to correctly classify voice from other sound events?

I have a huge labeled dataset of several thousand sound events, including human voice, dish washing, things falling to the ground, among others. I need to report when a human voice event takes place....
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0answers
79 views

Processing digital sound in Audiokit to sound like the 1960's

I am looking to imitate 70's analog equipments sound. I was wondering is there a way to create tape saturation / distortion inside Audiokit? I want to add that and randomized tape crackles and ...
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3answers
1k views

Get frequency and amplitude from an audio sample [closed]

I'm wondering how can i get the frequency and the amplitude from a 16 bits sample. I have actually an array of 16 bits signed int and I would like to get the frequency and the amplitude of each sample....
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1answer
1k views

How to convert dB back to manitude and then to rectangular format [duplicate]

I am experimenting with audio (wav files) using Short-Time Fourier Transform (STFT) in Python using scipy.signal.stft. As I ...
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3answers
5k views

How to change speed of audio samples without changing pitch? [duplicate]

This question may be more appropriate on music stack exchange but since sound is a branch of physics and this is more of a technical question relating to sound waves than a musical question I asked it ...
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0answers
72 views

Finding specifications for digital signal processors (DSPs) for given audio application

Note: This topic has been moved to https://electronics.stackexchange.com/posts/345289 since it fits better, please post any answers/comments there. I'm finding difficulties choosing an adequate ...
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2answers
1k views

Self studying, getting a quality spectrogram

So I've been self studying signal processing, and been trying to use Python to get quality spectrograms, quality spectrogram like the one that would be produced with audacity etc. The steps I have ...
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1answer
41 views

What are the differences in overtones for hard vs soft materials?

I've been researching Physical Modeling lately, and so far pretty much everything makes sense from a physical/logical perspective. The one thing that I haven't yet found a proper explanation on is ...
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3answers
3k views

Transmit data through sound between 2 computers (very close distance)

I'm writing an example on transmitting data through sound betwwen 2 computers. Some requirements: The distance is very close , i.e the 2 computers are basically adjacent to each other Very little ...
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0answers
124 views

Speaker diarization tool for acoustic analysis

I'm researching certain acoustic prosodic features during live clinical interactions. Because of the setting, the recordings always have cross channel bleed, and in some cases everything is recorded ...
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1answer
300 views

Making audio clips comparable in terms of RMS

I have a challenging question on how to normalize several audio clips so that they are comparable (similar) in RMS but without the clips clipping (having values over -1 and 1). Is there a systematic ...
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5answers
275 views

Remove a known wav file from recorded file

I looked at many similar questions but still cannot find an answer to my problem. I'm playing an audio file N, while recording a man talking (b). The problem is that the microphone gets the played ...
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1answer
280 views

File size adjustment using FFT

I have been given the task of doing audio compression using FFT. The assignment is below. Below is my MATLAB code: ...
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1answer
489 views

How to detect and remove ringback-tone and ivrs voice etc from the beginning of an audio call recording

In an audio recording (say a telephone conv. b/w two people), how would I programatically detect and remove the dial-tone at the beginning of a call using python. Ex : As you can see the first 15 ...
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1answer
718 views

Digital 16-bit PCM audio signal scaling and STFT spectogram amplitude

I'm sampling a 16-bit PCM audio signal at 44100 Hz and I'm trying to understand the amplitude level of both the digital audio signal and its corresponding spectrum. The final goal is creating audio ...
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5answers
966 views

Interpreting microphone output signal

I have a question about interpretation of the signal produced by a microphone. The output of a microphone is a voltage level that corresponds to the pressure on the membrane. The part that confuses ...
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0answers
157 views

Room Impulse Response Inverse Problem

This article here talks about a room impulse response generator which takes the source and destination coordinates, sound speed, sampling rate, room dimension and wall reflection coefficients as input ...
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2answers
340 views

Relation between samplingrate and frequency

I am working on Fourier Transformation, and applying this for recognizing an audioclip. I have a 9 second long audio clip of a guitar strumming an A-Minor. The audioclip has a sampling rate of 44100 ...
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2answers
221 views

Measuring vibrato rate of a sung note

I'm doing a project where I need to measure the rate of vibrato cycles in a sung note under various conditions. The stimulus will be a several-second-long audio sample of a held a capella note with ...
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1answer
636 views

How to change x-axis from samples to time?

I need to change the x-axis on this graph from 'samples' to time: I created the graph using this process: ...
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2answers
129 views

How to label sound events in noisy .wav files

I'm trying to find a way of automatically identifying birdsong in phone-recorded sound files (1min long). Currently the algorithm I'm using doesn't label all of the events I want it to. (it is ...
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3answers
90 views

How can a changing signal be Fourier transformed?

Fourier transformation is used to split a periodic signal into frequencies, and calculate the phase and shift for each frequency. But what if we would like to record an audio, which is changing, for ...
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1answer
350 views

Best beginner DSP book for developing plugins

I'm a computer science student and I love producing music and I really want to get into developing audio plugins. I'm currently learning the JUCE framework and am fairly comfortable with C++. I'm also ...
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2answers
171 views

Remove audible click sound from high frequency hopped sound

I have a total sound duration of $T$. I break the duration to several sub-durations and in each sub-duration, I generate several discrete sinusoidal values by the formula $\sin(2\pi ft)$ with a ...
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2answers
394 views

When does the laugh track kick in?

One of my weekend projects has brought me into the deep waters of signal processing. As with all my code projects that require some heavy-duty math, I'm more than happy to tinker my way to a solution ...
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1answer
235 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
56 views

Computing real signal with minimum absolute values from even magnitude spectrum

I want to derive a real audio signal from an arbitrary even magnitude spectrum. The phase spectrum affects the values of the signal in the time domain; for example, a phase of 0 for all frequencies ...
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2answers
356 views

How to deal with audio with different length in Audio Classification

Currently, I am trying to work with the Dataset UrbanSound8K to try some Audio classification. And I got stuck in the preprocessing step already. Since the audios are of different lengths, like 4 ...
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3answers
334 views

Audio wave amplitude decreased after FIR filter

I am doing sampling and filtering for a 17KHz sine audio wave signal, the Nyquist frequency is about 20.8889KHz. I found the fact that this audio signal's amplitude decreased dramatically after it ...
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1answer
56 views

Smoothing a Signal which contain White Noise

I'm trying to smooth a signal in order to reduce white noise as a result of changes in amplitude across sequential frames of an audio signal. I've been investigating using a moving average or ...