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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Standardizing amplitude (volume) of an audio signal at a certain frequency before & after taking the derivative of that signal?
I would like to take the derivative of an audio signal, which as I understand creates a 6 dB/oct upward sloping filter as per this thread: What phase rotation occurs when you take the derivative of an ...
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Saved "wav" file doesn't have a sound
I saved "wav" file with below code.
However, I can't hear any sound from "wav" file.
Is there any solution?
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Implementing IIR and FIR filter using vectorization and SIMD
I am learning fir and iir(biquad) filter implementation for audio applications using C on ARM. Its recommended to use vectorization of input data and SIMD operations for optimization . But i am ...
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Find a specific sample in an audio file
I'm trying to find each occurrence of a kick drum sample in a piece of music. I have a reference sample for this kick. I initially used numpy.correlate to find the locations but ran into issues for ...
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Am I generating audio signal with a given particular SNR value correctly?
I am using this as my reference:
https://www.mathworks.com/help/deeplearning/ug/denoise-speech-using-deep-learning-networks.html
Add washing machine noise to the speech signal. Set the noise power
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Remove sample from audio with different amplitudes
I have a piece of audio with a sample occurring within it multiple times. I wish to remove these occurances. However, the sample has slightly different amplitudes each time. Some of the samples occur ...
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Unable to remove audio noise using weak signal power calculated from FFT as threshold
Question
Please help understand the cause and solution of the problem of unable to remove the audio noise by using the signal power as filter threshold. If the approach is not correct, please advise ...
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What is the best way to sync I/O audio streams to mix them in real-time?
I want to mix loopback stream with audio from microphone in real-time and use the result of mixing for speech recognition.
The project is for Windows, so I use WASAPI and am willing to know what is ...
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Invertible Multi-band STFT (one spectrogram, N different time/frequency resolutions)
I'm working on an ML model that edits audio spectrograms which are then inverted. The problem is too many channels (and so too much compute) are dedicated frequencies that don't really matter much to ...
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How to apply FFT on audio in chunks without introducing distortion?
I have developed a service for simulating hearing impairment based on an audiogram. However, I've encountered a problem: I can only process the entire audio signal without distortions.Unfortunately, ...
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What is the relationship between RMS level and amplitude?
I have an Android recorder project, and I'm trying to implement a waveform display for recording and playing audio.
During recording, I chose to use ...
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are there techniques for mixing signals in real-time?
If I have say, four signals that I want to mix together in real time, in order to avoid clipping, I am assuming the correct thing to do is divide each value by 4 and sum them?
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How to find the "minimum spectrogram" across multiple recordings to extract a common audio signal?
The problem I'm trying to solve is that I'm trying to extract a "pure" version of a one-minute musical track that has been mixed together with voices speaking over it, over let's say 50 ...
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Will open, edit, store from/to lossy audio format significantly degrade quality?
Say I have files only available as "ogg vorbis", 44.1kHz at 128 kbps vbr - which are meant for archival, but not much storage for lossless was available at the time.
These are to be kept for ...
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Beats not detected in clear signal
I am trying to identify beats and tempo to differentiate between signals with librosa. However, when I was exploring the data I found that I cannot find the beats for my fourth audio (audio 4) despite ...
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Audio is sampled at 48 kHz, or 48000 samples per second. Does that mean 48000 bits (or 6000 bytes) of storage?
How does the sampling rate of an audio signal translate into memory capacity?
Does one sample translates into one bit in the memory, and hence 48000 bits of storage needed for a single second of audio?...
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Tools for getting the best out of multiple recordings of the same event
Input: several air audio recordings of a live concert taken from different locations within the venue. Different quality, different noises (e.g. voices of people in the vicinity of the recording ...
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Create a minimum phase filter from an elliptic digital filter
I have the numerator and denominator of a lowpass digital elliptic filter. I know how to create a minimum-phase filter with the same magnitude response using cepstrum technique. But I came across this ...
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Detecting and fixing clipped positive waveforms
I have a large number of audio files recorded using custom IOT devices. When analysing the data I noticed that a portion of them seems to have their positive amplitude ramp up, but the negative ...
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Using DCT to create real-time "levels" animation from microphone input
For context: I'm trying to create a simple "level monitor" animation of audio data streaming from a microphone. I'm running this code on an iOS device and leaning heavily on the Accelerate ...
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How can I use a constant Q transform to detect low frequencies without making the buffer size too large?
I am trying to write code to detect musical notes within a signal. I have been reading up on different methods for extracting frequency information out of a signal to determine which notes were played ...
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Removing a repeated sound from an audio file
I have a pre-recorded sound effect (FX) that will be played every 5 seconds. Together with that, I also play a music track for a few minutes. The result is recorded.
Now, I want to remove the sound FX ...
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--Which dither to use for narrowband speech?
The tl;dr
What dither algorithm should I use when converting 24/48 mono speech to 8/8?
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I'm truncating and downsampling a 24/48 mono audio source containing only speech to 8/8. I won't be doing any ...
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Is it possible to reconstruct the continuous frequency spectra from DFT/FFT?
Given the output of some FFT (lets call it X) with N bins (lets say 2048) is it possible to recover the values in between the bins in the same way that I can recover the values in between samples of a ...
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Audio delay with feedback
I'm writing a simple "audio delay effect", with feedback (the closer FEEDBACK from 1, the more the repetitions ...
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Finding the right phase offsets for minimizing the highest absolute value of a sum of sinusoids
suppose I have a sum of cosine waves. Their frequencies are 500 Hz, 501 Hz, 502 Hz and so on, all the way up to 900 Hz.
The important thing here is that the amplitude of the slowest one and of the ...
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What measures are used to identify a faulty speaker?
I'm working on building a mobile app that would play a sound via speaker, record itself and process this recording to identify faults within speakers, what meastures or parameters should be considered ...
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Can a bad quality sound be measured via THD?
I'm trying to create an app that measures the audio quality, this is ideally to filter out blown out or cracking speaker who has different types of distortions, what would be the best measurement ...
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Does the print of readframes(n) of wave library return the audio samples in hexadecimal form?
When I print the result of the first audio frame by doing track.readframes(1), I get b'\xfb\xff\xfb\xfe'. My track is stereo so we know that we have 2 channels where each channel has 1 sample. Does it ...
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Program to compare output and input streams to verify volume
I'm trying to wright a program to compare, two sound signals, output (speakers) and input (mic) streams and verify/adjust the volume. I'm using Microsoft Visual Studio C# forms project with the ...
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Why are MFCCs of different signals equal?
What I am trying to do:
I am trying to extract features based on the difference between a reference signal (white noise in my case, let's call it ref) and different processed versions of it (let's ...
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Best practice for classification of parts of a sound file based on content
I have a collection of raw recordings. I also have a text file for each recording with the starts and ends of subparts of interest from the original recording. I want to train a model to classify ...
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Minimizing maximum instantaneous magnitude by applying phase shift
Suppose I have the following sum, where all the input variables are positive real numbers while the output is complex-valued data
$$a e^{i \left( x + y \right)} + a e^{i \left( x - y \right)} + a b e^...
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Zero crossing count vs zero crossing rate: does it make any difference?
I am trying to make a feedback howling detection program for an audio DSP chip
I made a very crude one by counting zero crossings, doing something like this:
...
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Bearing vibration analysis from audio signal instead of velocimeter measurements
We currently test bearings on a third-party test machine, which uses a velocimeter and reports through a software front end that outputs a low mid and high band spectrum result, and RMS, Peak and ...
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Decoding images from sounds in Voyager golden records
I'm interested in a side learning project in decoding the images recorded as sounds in Voyager's golden discs (https://voyager.jpl.nasa.gov/golden-record/). The instructions to be able to decode these ...
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writing from STFT to audio using ISTFT adding harmonics to audio not present before
I'm doing quite a bit of the samething as I wrote in this post here :
Shifting frequency peaks such that w[i] moves to w[i/n] using STFT and IFFT
I'll quote from it here to present the issue I now ...
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Real-Time, efficient Loudness Normalization
I'm working on a Bluetooth speaker system where the same music file results in different SPL up to 10 dBA when played through different devices, such as iPhones and Android smartphones. The system ...
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How can the quality of a resampling algorithm be measured?
I need to write my own resampling algorithm to resample sample and impulse response wav files between 44.1, 48 and 96 kHz. I'm on an embedded platform, so I don't have access to the usual libraries. ...
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Is there a Mcu/ DSP IC which can help me convert wav to MP3 (44.1 Khz sampling) in real time and store it into an embedded SD card?
I have built a Bluetooth speaker using an ESP32. It can stream BLE audio and play MP3/WAV from an embedded SD card. I want to record 44.1Khz res audio from it but unfortunately the ESP32 is not ...
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Normalized audio cross corelation
I'm trying to cross corelate two audio files to see where they line up.
In scipy im doing: correlation = correlate(audio1, audio2, mode='full')
Given the graph ...
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What are the theoretical maximum values of the complex bins in an unscaled FFT of size N?
I am using FFTs to perform real-time convolution of audio signals on small embedded microcontrollers. I am thinking about testing different/smaller data types to represent the complex bins. I am ...
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How are the results of a fourier-transform interpolated to cover the entire frequency range?
Say we have an audio signal and we do a Fourier transform with block size 2048. Now the outputs correspond to 1024 different frequencies. A spectrum analyzer however doesn't just display 1024 discrete ...
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What is the cleanest, most transparent way to rapidly excite a resonant bandpass to a given amplitude (for musical synthesis)?
Background
Music audio synthesis can be performed using underdamped resonant bandpasses as harmonic oscillators (set to harmonic frequencies of a given note). These can then be excited when input ...
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Differential value range reduction for Huffman encoding
I am trying to understand a piece of coding technology and place it in its context, but I am unable to find any literature on the subject. It would be much appreciated if someone could point me in the ...
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frame size and hop size mismatch with aubio pitch detection algo
I am writing a script that detects the pitch of a short file as part of a larger project. I want to detect using an external call to sox and using the crepe, aubio, and librosa libraries. Aubio is ...
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Difference between Level, Signal Strength, Amplitude and Volume?
It seems impossible to find a precise answer to this question. I researched for quite some time but different sources come up different definitions, that sometimes disagree or contradict with each ...
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Pitch shifting effects on formants
I am doing a project for uni where i try to change the precieved gender of an audio using dsp techniques on matlab i started with pitch and am trying to move to formants now but i am not sure i fully ...
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Laplace transform of this simple parallel RLC circuit? (For audio speaker simulation ...)
SPEAKER AS RLC CIRCUIT
I read this article here which demonstrates a simulation of a speaker as a simple RLC circuit where the RLC components are in parallel:
MY GOAL
I am interested in creating a ...
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How to reduce clicking in an audio signal with seek jumps?
I'm working on a vst plugin where a delay is used, the seek position of the delay sometimes changes gradually sometimes abruptly, when sudden changes occur sometimes a clicking sound is produced.
Some ...
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