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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37 views

Generating audio clip by superposing two files

I'm currently trying to generate a set of audio files where some audio events (e.g. dog bark) are immersed in a background audio scene (e.g. crowd noise). I understand that if the event would have ...
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1answer
104 views

Formant Analysis Frequency - What makes a peak value?

from spectrograms, how can you tell which peak is really a formant. Like why is the pink arrow the first formant. And in the spectrogram below, which has fewer samples, what makes a value the peak ...
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Finding the input from the impulse response and output

I have $y,h,x$ which are all vectors. From $y[n]=x[n]*h[n]$ which is basically how I got $y[n]$. I also know $h[n]$. I put this through a Fourier transform. Let's assume that the capitalized ...
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2answers
91 views

Multiple different length Windows for high resolution realtime FFT

I am looking at trying to achieve a 'realtime' (as quick as possible to acquire, process and present data) FFT application to analyse audio. I have setup my app to acquire a set of samples, apply a ...
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1answer
83 views

Hilbert Transformer processing block by block

I am processing audio and trying to obtain a 90º phase shift when processing block by block. I have implemented a Hilbert Transformer (from Lyon's 3rd. Ed. & MATLAB). MATLAB code below (I believe ...
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32 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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2answers
417 views

Audio processing. Is it possible to directly access the decoded audio data going into the analog input of a computer

When a computer transcodes an audio file from one file format to another, does the computer first decode it into the raw digital stream (exactly what is fed into the DAC for the audio output) , or ...
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41 views

Is it possible to analyze this microphone data and what to do if not?

I just created an account here because I really need some help. I'm using a microphone pmodmic3 with a development board called Ublox C030-U201 which uses the mbed OS. The microphone is working for ...
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8 views

Efficient measurement of audio sound pulse

Here and here are two interesting posts about converting an audio signal recorded from a microphone to SPL. For specific case of pulse recording, with 0.5ms pulse duration with 1s period between ...
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3answers
108 views

How can I convert audio to a series of notes using pitch detection?

I have a good pitch detection system set up, and I would like to return a series of notes given an array of audio samples. My current approach is as follows: I have a moving window across the audio ...
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1answer
25 views

extend parts in an existing audio file to make it sound longer without stretching the file

I'm looking for a way to extend certain parts of an exiting audio file to make these parts sound like the where played longer. And I dont want to change play speed or stretch the file. Lets say u have ...
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1answer
31 views

Do Audio Interface manufacturers use A-weighted Measurements to make specs look better?

Recently I came across a reputed music-tech teacher who said audio interface manufacturers use A-weighted measurements to show dynamic range in their specs so that they look better on spec sheets and ...
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TEO-CB-AutoEnv implementation on python?

I'm trying to search for an implementation proposed as the nonlinear critical band based TEO auto correlation envelope area (TEO-CB-AutoEnv). If anyone knows of a library or such can you please link ...
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1answer
42 views

“Tail-like” response from digital microphone

I was doing some experiments with audio and wish to get "raw and unprocessed" data, and noticed that all the files had the following profile (just blowing into the mics): Speaking to the microphone ...
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56 views

Noise Reduction: How can I filter out saw or other tool noises?

I recently came across old recording from the 1930's. Unfortunately not only is the quality low, it's a bunch of carpenters talking as they saw things and make all sorts of other noises. I've tried: ...
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61 views

Frequency modulation audio synthesis: clicks/pops when changing modulator frequency dynamically

I'm very new to DSP and working on a FM VST. It's mostly up and running, or at least it generates FM-like sounds. I want to be able to change various parameters dynamically while audio is generated, ...
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1answer
63 views

Extracting audio from picture of time-domain signal

I saw this nifty art thing where they dress up the time-domain audio signal in some pretty way. https://www.etsy.com/ArtbloxShop/listing/621561709/sound-wave-art-custom-soundwave-print My question is ...
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1answer
196 views

Polyphase decimation filter plot in Octave

I'm using de Soras hiir library for decimation and I'm trying to verify the filter design in Octave. The filter is a polyphase 2x downsampling filter with 96dB attenuation and 0.01 transition band, ...
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1answer
52 views

Frequency analysis : relation between FFT size and sampling

I am currently trying to perform an experiment given an audio signal. I am therefore sampling the audio frequency range (0 to 22kHz) with a 48k sampling rate. When performing a FFT on my signal, I ...
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45 views

Python scipy firwin2 has a low frequency error when designing arbitrary amplitude filter

I've been trying to design an arbitrary amplitude FIR filter which has a specific attenuation for each frequency. However I get an error at low frequency when comparing the results to my expected ...
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1answer
520 views

Audio/speech detection, verification

I would like to write a program to compare two '.wav' files, one of which is the original played file with human speech and the other is a recorded '.wav' file. The program I wish to write in python ...
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40 views

Separating Ultrasonic signal from echo at the receiver

A transducer is acting as a transmitter and a receiver. It keeps on sending pulses which can hit an object placed at 5cm. Echo comes back and hits the transducer while it is continuously sending ...
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247 views

How do I play and record audio at the same time?

So this question is more related to Matlab itself. Some background info: I'm trying out the Exponential Sine Sweep method to obtain the Room Impulse Response with my laptop speakers and microphone ...
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2answers
166 views

Spatial Audio (Ambisonic) and calculating sound source position

I have 4ch Spatial Audio (Ambisonic) WAV file and I want to calculate sound source position and put it on 2D plane. What I know so far is the following: WAV file consists of signals that support of ...
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49 views

Method for the Localisation of Impulsive Noise in Speech Signal

I'm sorry if this question has already been answered, but I couldn't find anything. I'm working on a program for noise detection and need to locate the position of impulsive noise. The following image ...
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109 views

Error reading a .PCM file

I want to convert a .wav file with a sampling frequency of 44100 Hz to a 16 depth .pcm. I don't why I'm getting those peaks at the beginning of .pcm plot (second figure below 2). If you could ...
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110 views

Dynamic compressor - ratio in logarithmic scale problem

I want to use decibels to describe compression ratio. So for example ratio 4:1 means that every 4 input decibels above threshold causes raise of output only 1 decibel. And I also want ratio to be ...
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252 views

Language detection from audio

I'm trying to do a simple test on language detection on a video input. Basically, given a .mp4 file, I need to process the audio tracks (can have primary and ...
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1answer
96 views

Definition of DSP Terms

can someone define/explain briefly (or with some detail-> up to you) the following terms: Block Processing? and what are the differences between it and Sample Processing? (removed) What is a 3D ...
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49 views

how to lower frequency of a soubd file in matlab?

I am new to programming and sound processing, I have been using matlab to process sound files, a few of which are ultrasonic. my sampling rate is 250kHz. I Would like to be able to play the sounds at ...
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64 views

How do Anti-Aliasing-Filters in audio signal processing work?

I am a little confused about the use of anti-aliasing filters. As on how I am confused, please consider the following task: Consider a simple sampling rate conversion system with a conversion rate ...
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2answers
42 views

What is Causing a Noise Signal in USB Audio Recording?

I am experiencing the following peculiar noise signal when recording from my Behringer Xenyx Control 2 over USB. There is frequency at around 2293 hz as show: And over time, it produces the ...
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32 views

Temporally locating beats in an audio file

Imagine you have a long recording of a guitar player playing an arpeggio at a slow speed. To simplify, assume the tempo is 60 bpm or lower. I want to annotate this recording with the exact moments ...
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154 views

How to use a context window to segment a whole log Mel-spectrogram (ensuring the same number of segments for all the audios)?

I have several audios with different duration. So I don't know how to ensure the same number N of segments of the audio. I'm trying to implement an existing paper, so it's said that first a Log Mel-...
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1answer
69 views

Failed to Get Silence in Audio Streaming

I want to detect silence in audio streaming. I've been following lot of answers from any website and i feel like know how to do that. but i doubt i'm on the right path coz the result doesn't seems ...
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2answers
76 views

Technique to detect pitch after analysing only one pitch period?

does anyone know if there is a technic to estimate the pitch of a given signal by analysing only one pitch period. I'm asking because I read about a program called AudioMotors. https://lesound.io/...
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1answer
295 views

What is the best input for de-noising autoencoder for sound data?

I am currently trying to build an autoencoder to de-noise audio data. However I have not found any good articles explaining about the input to the autoencoder, i.e. feature vector. As in speech ...
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1answer
265 views

polyBLAMP anti-aliasing in C++

How should 4 point polyBLAMP work. It does not really seem to dampen aliasing. I have an floating point digital audio oscillator that makes a +1.0f jump at the end of every period. My question is, ...
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92 views

Decoding data hidden using “echo hiding” technique

I'm working on a little project, which centers around steganography. As part of it, I want to implement the audio echo hiding method, but the problem is, I've never done anything related to digital ...
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1answer
38 views

Modulation scheme for a noisy channel with a complicated transfer function

I want to build an audio modem with a speaker in a room and a smartphone. The channel is quite bad: Far field, so there is potentially significant reverb Typical residential background noise phone ...
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1answer
310 views

Understanding anti aliasing filter

I'm working on a project for the university about the anti aliasing filter. I have a small audio file, which has a sampling rate of 22KHz. Now when I downsample it 8 times, the file sounds weird. So ...
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39 views

Using complex number in non-negative matrix factorization (NMF) for signal source separation

In short, I wonder which kind of spectrum can be modeled using complex number in NMF. And could an imaginary part possibly be a vector? For detail, inspired by audio processing paper that used ...
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1answer
184 views

RF demodulation for dummies

Given a signal that's been FM encoded on a carrier wave, what are the individual steps I need to take to demodulate it, and recover the original signal? I don't have a background in signal processing,...
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29 views

Incorporate another dimension to non-negative matrix factorization (NMF)?

My first data profile has 3 dimensions; atom pattern (I think it is similar to frequency), time, and signal intensity. I can use supervised NMF for source separation straightforwardly. Because the ...
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2answers
348 views

Extending Goertzel algorithm to 24 kHz, 32 kHz and 48 kHz

I'm learning to implement Goertzel's algorithm to detect DTMF tones from recorded wave files. I got one implemented in python from here. It supports audio sampled at 8 kHz and 16 kHz. I would like to ...
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86 views

How do audiobook players manage to quickly speedup/slowdown playback speed without using more memory?

Many audiobook player apps on Android have a reading speed slider which lets us choose the reading speed e.g. 0.5x 0.75x 1.00x 1.25x 1.50x. This is time-stretching ...
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2answers
89 views

Converting speech audio to telephone audio

I am trying to train a machine learning algorithm on telephony speech audio. However, there isn't really enough data for this anywhere that I can find. My solution is to just use speech audio from ...
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1answer
158 views

How to build a realtime simple audio spectogram? [closed]

I'm trying to build a spectogram view of history time of 4 seconds and I'm stuck at the part where you load/draw FFT vectors into the 2D bitmap. I just don't know how to do that, but i got the FFT in ...
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1answer
54 views

How do I warp a 1 dimensional signal in frequency domain?

I want to implement VTLN (Vocal Tract Length Normalization). It is equivalent to warping the signal in the frequency domain according to the following graph. I can't wrap my head around how one ...
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1answer
382 views

Compute group delay of an audio file from STFT

I'd like to reproduce results from this paper in which the authors used Group Delay-grams as an input to the Machine Learning model. The Machine Learning side is not the one that I have problems with. ...