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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Inverse STFT causing unwanted noise spikes?
I have an (audio) signal. I am applying STFT continuously and after applying a certain magnitude threshold I am doing inverse STFT to reconstruct the signal. But I am getting some unwanted mini noise ...
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Detecting background noise from audio time-freq domain analysis
I have a requirement to detect/reduce sidetalk/background noise in real-time audio. I am stuck in how can I detect this from audio time-frequency domain analysis. I am already getting the time-freq ...
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do audio compression algorithms encode channels in parralel?
I'm looking to compress multiple mono sound tracks together, as one track (with loads of channels), however, I was wondering if this is stupid - because I may lose out on real compression, as these ...
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What's the state of the art way to detect / change audio pitch? [duplicate]
Detecting pitch: It seems that selecting the modal frequency in the frequency spectrum is too simple an algorithm for reliable results.
For detecting, and changing, the pitch of a note, what are the ...
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BPSK based audio digital communication system
As part of my diploma thesis I'm implementing digital communication system based on audio transmission in free space (STMF401RE + analog microphone). I'm using BPSK modulation and I've implemented ...
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Confusion regarding STFT phase vocoder-based pitch shifting
I'm currently working on a phase vocoder implementation based on the Short-Time Fourier Transform; it's heavily based on the models described here and here. I have successfully completed the analysis ...
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How are sounds recorded and reproduced (played back)?
Someday I have suddenly got wondered how sounds are recorded and reproduced. I searched a little bit but the information I got was composed of difficult words and contents, so I couldn't understand ...
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How fast is an histogram suppose to decay?
I am working on a software which receives streaming audio data, is expected to process that data on the fly, and then outputs the results.
The module I'm working on is expected to adjust the audio ...
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Effective bit depth of mixed signals
I have three signals of 16bit signed integers. Each signal represents a sine wave of unknown (and variable) frequency / phase in the audible range (20hz-20khz) and the amplitude is anywhere between <...
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Discrepancy in frequency response of biquad filter (in Python)
I have written a simple biquad filter class but when I check its frequency response, I get different results with different methods. If I check it "empirically" (by taking the FFT of its ...
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Prove that exponential sweep sine decay 6 dB per octave
Exponential sweep sine (ESS) is a good excitation signal to measure the impulse response of an acoustic system, such as loudspeaker - room - microphone system. An ESS has the form of
$$
x(t) = \sin\...
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Extract amplitude values from a wav file
I would like to extract the amplitude value from a wav file. Using Python I can extract the raw data into an array. How do I go about extracting the amplitude value of each sample?
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Convert a list of values and times into an audio file
I have two pandas dataframe, one with time in seconds and one with the actual audio data (values from -1 to 1).
The audio data points are not equidistant, this why I need the list of time in seconds.
...
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How can THD (Total Harmonic Distortion) be applied to compare between two signals ( for example between original and encoded signals)?
I would like to compare between two signals (original signal and its encoded version) and plot the scores) as we do with STOI and PESQ.
should we calculate the THD for each single signal separately ...
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Which Audio Codec/Bitrate Relationship delivers best Audio Quality: 128 kbit/s MP3 vs. 192 kbit/s AAC vs. 64 kbit/s OPUS?
Assumed we have those three audio streams available for listening to:
MP3 with a bitrate of 128 kbit/s
...
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What is the best approach to cancel realtime noise from a participant audio for conference purpose?
I am new to audio signal processing and I wanna do real-time noise cancelation for a participant in a conference.
I am capturing audio at 16bit, 16khz, little-endian config. I tried to use the webRTC ...
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How to implement Auto Gain Control (AGC) before and after mixng realtime audio for a conference?
I mix audio(16bit,16khz, little-endian, mono,frame-duration:60ms) packets after dividing the short values by participant count so it never gets clipped. but before that, I need to apply a certain gain ...
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Did Dialup Modem use closed or open loop power/volume control? How did they determine Tx level?
I have been reading about dialup modems, but one thing I cannot seem to find out about is how implementers determined optimal transmit power/volume. Is this part of the echo cancelation framework? The ...
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How do I get the waveform fom the soundcard in VB.NET?
I've written a VB.NET WinForms application that allows me to perform a Fourier Transform to find the amplitudes of discrete frequencies to represent that “strength” as a bar graph. The code for this ...
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0/1 dB normalization using clamp function
I would like to perform dB normalization on audio data to make less loud sound louder and to make unusually loud sound less loud, before feeding them into neural networks.
I came across this function:
...
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Measuring the frequency and impulse response of an audio recorder
I would like to test the frequency and transient/step response of the Zoom H5 recorder preamps. If possible, I'd also like to test the preamps for the built-in (exchangeable) XY capsule module, but I'...
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How to normalise 3D time-frequency sound intensity vector
I've got a 3D time-frequency sound intensity vector that I've derived from Ambisonic b-format signals. W acts as the sound pressure ...
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Audio compressor delay/lookahead
I am making a real-time audio compressor/limiter. I am in little doubt about the delay. What is typical delay times for a compressors or limiters delay/lookahead time?
Most other compressors I see don'...
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Why is a feedback loop "appropriate" design for instruments that don't physically contain such loop?
Why is a feedback loop "appropriate" design for instruments that don't physically contain such loop? Or do they?
See e.g. paragraph 7.7, and Figure 7.9 (shown here):
https://www.cim.mcgill....
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How to detect repeating music?
Suppose you have a music data-file (for example mp3) with a single audio stream which contains an audio time-series for 3 hours, but you want to find out if that audio file (maybe some compilation) ...
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Audio sample analysis - Comparing two audio samples, the original sample with a reproduction capture sample [duplicate]
Already asked in sound.stackexchange.com but it was suggested by one of the comments that it would be better suited to be asked here.
Introduction
I'm currently working on a project where I'm ...
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FFT on a large number of samples
I have written a C code for a radix FFT transform over 1024 samples. But my main goal is taking audio files, converting them to raw pcm sample values, and then doing FFT on the whole audio file. My ...
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Standard audio format using DWT for compression
Is there any standard audio format that employs discrete wavelet transform to compress signals?
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Composing digital filters
I'm interested in composing filters for realtime audio processing on an microcontroller (MCU). Ideal frequency response is unity as a default, with deviations up and down at specific freq-domain ...
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How timbre shifting is done?
I've recently came across two programs - Morphvox, VCSdiamond
that are able to preform pitch shift, but also timbre shift.
As far as I know the timbre is nothing but the amplitude of the harmonics in ...
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Audio Processing - How to create a spectral pitch display?
I want to create an application that analyzes human voice pitch, but spectrograms are very noisy. However, in Adobe Audition, there is a feature called the spectral pitch display, and it successfully ...
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Why is this ring-buffer jumping?
Central problem (Abstract):
A simple implementation of a two-buffer ring-buffer jumps after filling up completely. What am I missing?
About:
ADC to DAC feedthrough
STM32L476RG NUCLEO as MCU
Intended ...
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Why is voice clipping happening after Dynamic Range Compressor (MATLAB) when applied in real time on chunks of data?
I have been working on real-time audio signals on smartphones (iOS and Android) and my goal is to make voice amplitude stable in a certain range even if a person is talking closer to the mic of the ...
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Multiplying complex signal with its own complex conjugate transposed copy? [closed]
Let input be a real (audio) signal, and let X be its complex frequency domain representation.
What does the output represent when the frequency domain signal is ...
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Remove a full song from audio recording
I have some audio recordings at 16KHz, which only contain music (BenSound Adventure: https://www.youtube.com/watch?v=0H8JTsG1Jtk). Since I know the music and have a separate wav file, I was wondering ...
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Non-Uniformly Partitioned Convolution Implementation
I've succeded in implementing the uniformly partitioned convolution algorithm and now I'm looking to implement the non-uniformly partitioned version. I've had no luck with running parallel threads on ...
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Signal values explode when filtering with bandpass and wiener in Python
I have a five second audio (speech with background noise), which I want to process first with bandpass filter and then with wiener filter to reduce noise. Audio is normalized between [-1, 1] and I ...
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Manually write an audio file
I want to write an audio file (the particular format does not matter to me) by hand, i.e. for each frame I would write manually which frequencies are present at which intensity. How do I do this? Is ...
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What is the name of this plug? (with images)
I have speakers that connect with the plugs shown in the images below to a hifi system, but I would like to connect them instead directly to a computer. Before I can find some kind of adapter to make ...
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How to decode a binary Signal in a wav file into binary - sent here from stackoverflow
I'm doing a puzzle that gets me to call a phone number which produces a load of beeping sounds. Putting this into audacity as a spectrogram produces this:
Zoomed in it looks like this:
This looks ...
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Searching for an algorithm that can detect the stereophonic recording method used for a given stereo recording
I am trying to automatically detect, whether a given stereo recording was done by the X-Y (intensity) or A-B (time-of-arrival) recording method.
Does anybody know, if there is research on the topic? I ...
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Frequency domain processing
I'm a little fuzzy on how the Fourier transform works. It's supposed to map time domain to frequency domain, and vice versa, so my intuition says that if I wanted to make an equalizer, I'd take the ...
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How to remove or smooth the comb filter effect in real time audio signal mixing?
I have been working on a project where I have to mix multiple audio signals of the same source coming from different slave smartphones on one master smartphone in a distributed way. Now I have aligned ...
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Audio, estimate codec's compression ratio only by content
I am about music audios.
These can be compressed by different formats with different degree.
(I.e. I don't mean the dynamic compression).
After being compressed by a large degree (say 96kbps mp3)
the ...
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Eliminate different microphones textures from their recordings
In my setting, I have different microphones recording a same speech signal. For each microphone recording, I can feel a slight difference in the playing sound. I am calling this property of a ...
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Time warping / frequency modulation natural examples?
"Time warping" defined as
$$
x(t) \rightarrow x(t - \tau(t))
$$
where $|\tau'(t)| < 1$ ($=1$ is time stopping, $=0$ is time shift, $>1$ is time reversal).
Where can this occur? A ...
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How to upsample voice properly?
I have a low quality 24 kHz mp3 voice and I'd like to upsample it to 48 kHz for farther mixing with another ...
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Estimating reverberant time (T60) of an IIR filter
I'm learning a method to genenrate pink noise by filtering a white noise in this book, and the author mentioned an estimation of T60 given the coefficients of an IIR filter in the code that
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Fourier coefficient to power
When doing a FT on a 4kHz and on a 8kHz signal
(both 0dB amplitude), the Audition spectrum view shows 0dB for both
freqencies as expected.
But the calculated fourier coefficients for both frequencies
...
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Will a 2-recording approach work to remove ambient sound to isolate insect activity noise?
I am recording insect noise - buzzing, rustling in flowers, munching, etc. with a lav mic on a long skinny stick. My new goal is to remove ambient sound like planes, distant cars, etc.
Poor results (...
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