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.
1,656 questions
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11 views
OFDM invalid packets, filtering problems in GNU Radio
I've been working on a custom OFDM implementation and I'm getting a lot of ...
0 votes
2 answers
66 views
How to start building a piano note detection program (simple right-hand recordings, step-by-step approach)?
I’m currently working on my undergraduate thesis, which is related to digital signal processing. The goal of my project is to create a program that can “listen” to piano recordings and convert what it ...
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1 answer
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When buffering a wrapper process at one length, and an internal process at another length, what is the logic behind the delay caused?
In an audio (or likely any data transfer context) processing scenario, at times it is necessary for an internal process which takes a fixed amount of data N to be wrapped by an external process which ...
0 votes
1 answer
42 views
Does Stereo mode use the bandwidth more effectively than Dual Channel mode in SBC codec?
If an earbuds can accept a maximum of 1 Mbps bandwidth connection for audio, can Stereo mode (not Joint Stereo) use the bandwidth more effectively and produce higher audio quality than Dual Channel ...
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4 votes
3 answers
501 views
How can I determine the "almost-period" of a sum of sinusoidals?
I have a sum of sinusoids $$g(x)=\sum_{i=1}^N a_i \cos(2\pi \cdot f_i \cdot x).$$ For all $i$, $\frac{f_{i+1}}{f_i}=2^{\frac 1{12}}$ (like with MIDI). Because the ratio between the periods of adjacent ...
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1 answer
88 views
How to detect repeating patterns in short audio clips (≈4 seconds) using Python and machine learning?
I’m working on a machine learning project where I need to classify very short audio clips (about 4 seconds each) into two categories: repeating song: clips that contain a repeating musical pattern. ...
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1 vote
3 answers
370 views
Low-cost embedded DSP boards for audio processing
What are some low-cost embedded development boards with DSP capabilities suitable for audio processing? Looking for options with good toolchain support for real-time audio applications.
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1 vote
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38 views
Unexpected pitch drops after note release (TarsosDSP/FFT_YIN): algorithm limitation or misuse?
Intro: I'm building a real-time pitch visualizer app for Android using TarsosDSP. The app uses the FFT_YIN algorithm to detect fundamental frequency and maps it in real-time to a vertical piano-roll-...
3 votes
3 answers
612 views
Shape of fft graph obtained from microphone - Android phone
I am trying to visualise the output of an fft. I have taken the microphone output of my Android phone using "audiorecord". The input data is sampled at 32 kHz and I take 128 samples at a ...
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1 vote
1 answer
62 views
Usage of Ramping Functionality in interrupts
This is a general technical query to gather ideas regarding the use of ramp functionality on audio signals within interrupts when developing code for DSP processors. Given that interrupt service ...
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0 votes
0 answers
120 views
Using SDR to capture 2.4 GHz signals from wireless headset transmitter
I've taken up an ambitious task for a small university project. I would like to attempt to intercept the wirelessly transmitted data of my wireless HyperX Cloud Flight headset. I order to achieve this ...
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0 votes
1 answer
124 views
how to achieve autotune pitch shift effect using librosa pitch shift
I was able to create a decent pitch tracker using YAAPT, however, I've been struggling with using librosa's pitch_shift method which I believe uses a phase vocoder internally. How does one achieve the ...
3 votes
0 answers
116 views
Problem with Feedback in LMS adaptive filter for ANC
I am attempting to make ANC headphones, so far I have constructed the headphones and validated that every component works. I have also constructed an FIR adaptive filter using leaky NLMS for its ...
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0 votes
1 answer
50 views
Should the peak level change based on octave banding for a sinewave
I have a 1 KHz tone playing at 85 dBSPL which is monitored on a real time analyzer. I noticed that the level is just about 85 dBSPL on all settings of octave banding, ie, 1 octave all the way to 1/48 ...
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0 votes
1 answer
72 views
ALU Saturation bit causing rand() internal function to saturate
Rand function randomness is getting affected when ALU saturation bit is enabled in ADSP-21594 Series DSP processor. My use case for rand function is to generate white noise. When in the case the ALU ...
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