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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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3k views

Verifying MFCC Final Result

I have created a Octave code for MFCC calculation for speech recognition, which takes a wav file as input and then gives the MFCC as the output. But the final results are a bit confusing, so I am ...
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893 views

STFT overlap add method - What am i doing wrong?

i am trying to implement a short time fourier transform of an audio signal to do some filtering (subtract another stft signal using spectral subtraction). I am using a hamming window function with 50% ...
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542 views

What is EAQUAL ? How does it relate to PEAQ?

This wiki post on wiki.hydrogenaudio.org says that it is very similar to PEAQ and also a part of ITU-R BS.1387 recommendation ? How reliable it is for determining audio quality. Are there any other ...
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128 views

Voice Biometrics (aka Speaker Recognition) - Standard accuracy rate? State of the art tech?

I'm working on a voice biometrics project and am looking for some guidance on two key issues: First, does anyone know if a standard exists for voice biometric system accuracy? I have been unable to ...
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101 views

Can you apply an audio filter/effect to only part of an audio signal and not get audio glitches?

I'm entertaining the idea of making a music-creating application. I'm a total newb at signal processing. The question that concerns me most right now is, if I apply a filter/effect such as convolution,...
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393 views

Why are the frequencies associated with the STFT bins not representative of the energy?

I'm experimenting with a 30 Hz sine wave and I want to extract the frequencies of this, and have implemented the algorithm to do so. This gives a correct reading, well, gives 29 Hz. I'm now wanting ...
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0answers
8k views

numpy.correlate and autocorrelation; audio signal

I want to calculate delay between an input and an output audio signal of my audio processing system. The input and output signals are available as signed 16 bit integers. To try out, I tried the ...
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301 views

choosing windowing function for audio signal processing

Currently I am working on chord recognition, and get confused on several aspect. One of them is choosing windowing for reduce the spectral leakage. from what I read, hanning/hamming function is a ...
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68 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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105 views

Frequency translation with FFT

I'm new to DSP and I would like to make an SDR receiver. For this I want to shift bins in the frequency domain in order to bring to 0Hz any portion of the base band spectrum (I know that there are ...
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0answers
152 views

What are F0 counters in Speech signals? What is represented by F0?

I can't find any dedicated definition for Frequency F0 counters. Can anyone tell me what does F0 exactly mean, I am confused about that.
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94 views

Is it possible to “equalise” a signal by deconvolving the impulse response of the room in which it is to be played?

I am using a test sweep with a flat power spectrum and linear group delay (Optimized Aoshima's Time Stretched Pulse) to measure a room's frequency response. Having obtained the impulse response of the ...
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0answers
57 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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97 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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0answers
103 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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0answers
641 views

What does Fast and Slow Time Weighting signify for real-time audio signal processing?

I have a streaming audio signal that I am recording in real time. To make the explanation easy, say the signal that I receive is calibrated and has a sampling frequency, $F_s = 44100\textrm{ Hz}$. I ...
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0answers
730 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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153 views

Alternative to scipy maximum_filter to find peaks?

I'm using scipy maximum_filter together with binary_erosion to find peaks in my spectrum. I just wondering if there are any ...
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0answers
85 views

Low-rank approximation of the STFT matrix : good audio compression technique?

I recently heard about this interesting math tool: With a matrix $A$, compute the SVD ($ A = U \Sigma ^tV$ factorization, where $\Sigma$ is a diagonal matrix containing the "singular values") Keep ...
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203 views

1/f noise generation

Recently I was reading a paper about the pink noise generation. But I don't have any background of this field. I am confused with some part in the paper. First, the Voss's algorithm generated a ...
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0answers
226 views

MFCC help: Same sound, different coefficients?

Problem: I'm getting totally different MFCC coefficients for identical and near identical sounds like several finger snaps or several desk taps. What am I missing? I thought since the timbre doesn't ...
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0answers
169 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 ...
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0answers
482 views

Audio watermark embedding text attack resistant

I'm developing an audio watermarking algorithm which embeds text (binary form) in DWT-DCT coefficients. The goal of the algorithm is to hide song information (like artist, song, album,...). Everything ...
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0answers
110 views

Complex Cepstral Transform and Audio Watermarking

I am doing research in audio watermarking and have come across the paper that uses the complex transform to embed the watermark in order to improve the robustness and inaudibility of the embedded ...
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40 views

FIR Audio Equalizer Crackling Noise

I've implemented a few FIR filters into a song equalizer, but there seems to be a crackling that I'm unsure how to fix. Heres a video for reference: https://www.youtube.com/watch?v=IlzWRYpv0rk. Noise ...
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52 views

Artefacts from playing ultrasound through PA system

I encode data in audio by energising 5 frequencies between 17.5kHz and 19.5kHz: I play the ...
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0answers
345 views

Cocktail Party Problem — audio source separation

I am trying to solve the Cocktail Party Problem. I am trying to separate these two mixed audio files: mixed 1 [WAV] mixed 2 [WAV] into 2 separate audio files that contain the 2 original sources, ...
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65 views

Subtracting audio signal emitted - trying to use spectral subtraction to localize moving objects

I am a Software Engineer without much signal processing background and currently spending and experimenting to get use to it. My scenario: Assume a speaker and a microphone array. A speaker emits an ...
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0answers
72 views

low-frequency amplitude with phase modulation of low Hz carrier

I am writing an FM/PM synthesizer, using Java. For the most part it works pretty well. What I am curious about is an attempt to use a low frequency tone as a carrier. By low frequency, I mean in the ...
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111 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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0answers
27 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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0answers
59 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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0answers
175 views

What is the algorithmic latency of STFT processing?

I was reading this paper, and in section 3 I got a bit confused. It claims that the algorithmic latency is equal to synthesis frame length. If the overlap between two consecutive frames is 50%, the ...
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97 views

Gain of resonating filters

I'm trying to make a music instrument based on a resonator with a high Q, such as Ableton's Corpus and Ableton's Resonator (I think the former is based on a simple 2nd order resonant filter while the ...
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0answers
49 views

Real-time DAQ (Data AcQuisition) device and filtering of voltage signals in audio frequencies

I'm working on a frequency mixer project in which two analog voltage signals at two different frequencies (each <5 kHz) are considered. I'd like to add the two fundamental frequencies and return an ...
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0answers
105 views

ICA for blind source separation clarification question

I'm currently trying to implement FastICA for blind source separation from scratch. The code below does not generate W, the umixing matrix, correctly. When I matrix multiplied the outputs ...
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0answers
164 views

Applying a frequency filter in dB onto an audio wav file in the frequency domain

I am working on a university project consisting of a C program supposed to apply a correction of the isophonic curves to an audio PCM file. So far, I have managed to code the most part of it (read/...
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0answers
117 views

What is advantage of Logarithmic Sampling?

Logarithmic Fourier Transformation This paper starts off in the abstract saying that the given audio signal is sampled logarithmically. We also know that the human ear has a sort of a filter bank ...
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0answers
80 views

Fastest way to subtract known music tracks from live videogame audio stream

I want to isolate video game "action" sounds from the video game soundtrack. Given all ~20 songs, I'd like to filter them out from any live audio stream during gameplay. What is a fast and robust way ...
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494 views

How does the size of a spectrogram relate to window length and frame rate?

I'm trying to understand the methodology from section 4 in this report about audio classification but I have some trouble understanding how they pre-process the data (dsp is not my area). $30$ ...
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246 views

removing noise from audio

I have an audio file, in the range of 4 kHz to 120 kHz. I want to remove the lines that are not harmonic and are recognized as environment noise. the energy of the signal in this interval for each ...
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0answers
221 views

What FPGA development tools recommended for building audio music synthesizers?

I want to experiment with FPGAs to build audio synthesizers. They would end up being manufactured in quantities of less than 100. I have very little knowledge of FPGAs, but I thought LabVIEW looks ...
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0answers
82 views

difference between logged power spectrum and power spectrum?

What is the difference between a logged power spectrum and power spectrum... Well the log part. So can a normal power spectrum be converted to a logged power spectrum just by taking the log of the ...
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0answers
71 views

Can MFCC consist any zero values?

After extracting audio features using some library, I just realized that sometimes it (MFCC) might or might not have zero values. At what time/event does MFCC could have zero values? Because it's (...
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0answers
72 views

Determining the probability of two signals, with added independent noise, being the same?

I have the first 100ms of two signals, each with a noise independent to that signal. I want to compare the two signals and determine the probability that these two signals are (minus noise) the same, ...
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0answers
113 views

Feature recognition within a single song

I want to write an algorithm in C++ which is capable of identifying specific features within a single song, e. g. the sound of a drum which is played 100 times during 5 min. State of the project So ...
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0answers
520 views

Detecting delay and sample drift in audio files (wav, mp3, etc.)

In my application, I have an audio recording with background music. I know which song is playing in the background and have a copy of the original song (wav, mp3, etc.). The goal is to align the ...
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0answers
165 views

I have 24-bit data in two's complement 32-bit integers. I need Q31 data

I have been trying to learn about Q31 format (as well as Q15), but I am having trouble understanding how it works. So my CODEC is giving me 24-bit two's complement data. I have converted this data ...
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0answers
152 views

Two way Audio crossover MATLAB code using Bessel

Could MATLAB's besself function be used in designing an Audio crossover (Tweeter and Woofer) with highpass and lowpass filters? Most likely the same with this ...
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294 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 (...