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.

Filter by
Sorted by
Tagged with
0
votes
2answers
306 views

can MFCC coefficients alone be used to do voice detection?

Aim of my project is to differentiate between cases of silence, single active speaker and multiple ( simultaneous ) active speakers using as basic ( but effective ) logic as possible. Is it possible ...
0
votes
0answers
13 views

Expand and shrink waveform visualization depending on tempo changes

I am not quite sure if this is the right state exchange site or not. Pls let me know. So, I noticed that Logic and Pro Tools, etc. have this waveform shrinking and expanding effect when dragging the ...
1
vote
1answer
61 views

Produce musical signal from instantaneous frequency

I'm implementing the Yin Algorithm in Python so as to extract the pitch of my voice when whistling or humming. My goal is to produce the Gate (Envelop Control) and CV (Pitch Control) signals which can ...
1
vote
1answer
108 views

Audio: How to measure dynamic range (in sense of dynamic compression/ “loudness war”)

I'm about to calculate some quality measures for audio. One important of them is the dynamic range (in sense of "loudness war"). By which in this context is not meant the SNR but the (naive ...
0
votes
2answers
28 views

Sudden loudness changes producing unwanted noises?

What's that effect called? Is that one of the reasons for having attack and release parameters when implementing synths? And is there a generic method to avoid such sound artifacts? Example: I have ...
2
votes
3answers
1k views

Understanding LPC for Formant Estimation

I went through the Matlab tutorial on Formant Estimation using LPC Coefficients. Though I vaguely understand the details, it's not entirely clear why we need to do this. From http://person2.sol.lu.se/...
0
votes
0answers
17 views

How to design a low pass filter to convert PDM bitstreams into analog?

I have captured two bitstreams of PDM data into Excel. How do I design an appropriate low pass filter (in Matlab) that will convert the PDM bitstreams into analog form that shows amplitude?
4
votes
1answer
115 views

(Newbie question) Calculating the frequency spectrum

I'm quite new to this topic, with some basic knowledge however. I need to analyse an audio signal for different characteristics. Like the audio editor I use (an older Version of Audition, v.3) I would ...
1
vote
1answer
14 views

How do I set window length and shift length for computing STFT on Scipy?

Does nperseg=32, noverlap=16 mean the window length is 32mS with a shift of 16mS? Refer here for more details
2
votes
1answer
339 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.
0
votes
2answers
28 views

Audio keeps echoing continuously when using 2 separate devices for audio and recording on video calls

I had used 2 separate devices for a video call. Desktop for video and audio and phone for recording, since the desktop microphone is not working properly. Unfortunately when I started speaking, my ...
4
votes
1answer
2k views

Compensating Loudspeaker frequency response in an audio signal

I have been working on a project in which I was required to work on the audio signals recorded from the loudspeaker kept in front of a filter. So, to simply explain it: $$\boxed{\rm LoudSpeaker} \...
1
vote
0answers
40 views

Sweep synthesis in frequency domain: How to correctly adjust magnitude?

I'm attempting to synthesize a logarithmic sweep signal to measure the IR of a system. For the most part, I'm following Section 5.2 of the paper Transfer Function Measurement with Sweeps. To generate ...
0
votes
0answers
30 views

Is it possible to open 2 streams from the same microphone on Oboe?

I'm trying to open 2 input streams from the same microphone on Oboe but it's not working. One of the audio inputs I'm opening is like this: ...
4
votes
2answers
125 views

A cubic interpolation function: folkloric copypasta or clever trade-off?

I've been reading on interpolation methods recently and I have come across an implementation of cubic interpolation that is leaving my head scratching. Every other variant and example of cubic ...
0
votes
2answers
147 views

How to graph magnitude plots for basic 4-pole filters?

I have developed magnitude plots for basic one-pole and two-pole filters derived from various Physics PDF's and tutorials I have found online and calculated from: https://www.desmos.com/calculator/...
5
votes
5answers
3k views

Why audio clipping produces harmonics?

Today I was talking to my collegue about clipping and he said that clipping produces harmonics in frequency domain as he showed me the clipped signal frequency response. However when I asked him why ...
0
votes
1answer
24 views

Ambient noise levels in desert and recording studio

Are there mathematical models which suggest which of the two - desert or recording studio - is likely to be quieter? I would think a recording studio would be quieter than a desert. Is there a ...
0
votes
0answers
58 views

How to make ultrasound audible?

Because of interesting spectral phenomena, I would like to shift the >20kHz content of my audio recordings to audible frequencies, so that I can listen to it. I tried the Audacity pitch correction ...
2
votes
1answer
155 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 ...
1
vote
3answers
126 views

Modeling the human perception of noise

Add together some sinusoids with irrational or "nearly" irrational frequency ratios, and you get something pseudoperiodic. If there aren't too many, you can hear that it has only pitched components, ...
0
votes
1answer
32 views

How can I filter a noise with pattern type in audio?

I have a set of audio files rescued from a source of phonograph record like a gramophone. Very old family stuff records from my elders back in year 1946. A relative has digitalized these with a ...
0
votes
1answer
2k views

Bandpass filter for audio wav file

How do I apply a bandpass filter on an audio file which is in wav format? And write back the filtered data in a new wav file for further analysis.
1
vote
3answers
60 views

Sampling a low-frequency sound doesn't work at some sampling rates

When sampling a low-frequency sound (like a 55Hz sine wave) I hear an additional high frequency sound at certain sampling rates, but not at others. This happens in Audacity and in my own program. For ...
2
votes
1answer
54 views

Exponential Swept Sine Distortion

I was able to get passable results on fundamental measurements using and exponential swept sine. Now I am trying to get distortion information from the same measurement but am puzzled by the results. ...
0
votes
3answers
839 views

Distribution of energy in frequency bands

On my wav-file (music), I do a short-time Fourier transform (STFT). I have spectrum, which spans from $0\textrm{ Hz}$ to $f_s/2$. I took out the range between $1$ and $1000\textrm{ Hz}$, which I ...
0
votes
0answers
35 views

What would an algorithm that can target different devices be called?

I'm developing an audio filtering algorithm that can be run in different scenarios, e.g as embedded firmware on DSP Codec/Core, as a software library, or as a cloud service. Effectively, it can target ...
4
votes
1answer
57 views

Actual Sample Rate slightly off?

When capturing microphone audio, I noticed that my static buffer eventually overflows. So I timed the samples, and low and behold I'm getting an actual sample rate of 44102-ish Hz (my timer is very ...
0
votes
2answers
184 views

How long does it take for snd_pcm_writei() (ALSA) to send data?

Using a pretty typical call to Linux's ALSA, snd_pcm_writei(pcm_handle, pcm_buffer, pcm_buffer_size) how do you know how long it should take to send this data? What ...
0
votes
1answer
47 views

Algorithms to fix audio feedback loop

I'm a student and I have been assigned to make a presenation on algorithms to prevent audio feedback loop (microphone feedback, that issue when you hear the voice echoed multiple times). I looked ...
1
vote
3answers
188 views

Resample audio signal with a low factors

Currently, we need to handle a piece of an audio signal. We want to resample it with different factor rate (p/q) starting with 1.0002 to 0.9998 with a decrement of 0.000005. But Matlab function cannot ...
0
votes
1answer
24 views

Error introduced by smoothing filter at the end of a processing block

I'm running analysis on a multichannel audio signal and due to the size have decided to process it in blocks (my computer doesn't have enough memory to process it in one go). Some of the data produced ...
2
votes
1answer
98 views

Introductory resources to mathematical properties of “dynamically recalculable” filters in audio/musical equalizers?

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers? Particularly, I want to understand What mathematical features (e.g. monotonicity ...
0
votes
2answers
197 views

What Is the Meaning of the Term "Frequency Response'' When We Design a Pitch Detector

This is from Klapuri 2003. What is the "frequency response" and why do we model is as multiplied with the vibrating system we want to get the frequency from?
0
votes
1answer
108 views

Audio texture analysis for visualization

I'm making a complex music visualizer as a personal project. I've already determined that I will be using a series of modified Goertzel algorithms for frequency detection to place the frequencies on a ...
1
vote
4answers
283 views

Signal and Space/Time

I have been reading a couple of definitions on signal such as: Any quantity measurable through time over space or any higher dimension can be taken as a signal. And this, In the physical ...
0
votes
1answer
389 views

Machine Learning - any suggestions to solve Python rounding errors?

I am working with Python to isolate voiced segments from music using the Jamendo corpus for singing detection. Training a model, I break my audio into frames, and have a label (0,1) for each frame. ...
0
votes
1answer
915 views

Verification of all steps needed to extract MFCCs from raw audio data

I need some verification on steps required to extract MFCCs from raw audio as I'm finding that some sources outline them slightly differently. Get raw audio frame Run window function (eg Blackman) on ...
0
votes
1answer
365 views

Dynamic Time Warp through SPTK tool

I'm an IT student and got an assignment to do about Dynamic Time Warping(DTW) using the Speech Signal Processing Toolkit (SPTK) and comparing some words spoken by 2 speakers and finding the ...
3
votes
1answer
36 views

Low Level Audio Engine Audio Data Reformatting - Sample Rate and Bit Depth Conversion

First off, there seems to be a sparse number of low level audio programming tutorials out there so if anyone knows of some good resources I would be very appreciative. Now on to the topic at hand! I ...
2
votes
1answer
814 views

Applying DTW on MFCC frames

I want to use MFCC (Mel-frequency cepstral coefficients) and DTW (dynamic time warping) algorithms to make a speech comparison script. I calculated MFCC and got 13 coefficients for each frame. How ...
2
votes
1answer
1k views

Are there any signal processing libraries that implement TD-PSOLA?

For such a popular and old algorithm, I've had surprisingly little luck finding implementations. I've got it working in MATLAB, but am curious if any other languages have libraries containing it (...
5
votes
1answer
5k views

Demodulating FSK audio in Python

I am trying to write a Python script that can demodulate an FSK modulated audio file and return the data encoded in the audio. The data being transmitted is GPS NMEA strings which are embedded as the ...
0
votes
1answer
46 views

Detect playback position in swept sine sweep

I am wondering what the most precise method is to detect the playback sample position of a sine sweep? The real playback sample position is unknown and an DSP with constraint computing capabilities ...
0
votes
1answer
263 views

extracting MPEG-7 low-level audio descriptors using essentia

According to https://www.ntnu.edu/documents/1001201110/1266017954/DAFx-15_submission_43_v2.pdf, the audio processing toolbox Essentia includes the 100% of MPEG-7 low-level audio descriptors. However I ...
1
vote
2answers
100 views

How to merge 2 WAV files together? Can I take the average of the 2 audio file float values? [duplicate]

I'm trying to do one simple thing on Google's C++ Oboe library: pick 2 recordings and merge both. That is, make as if the 2 were playing at the same time. Oboe works with Float numbers from -1.0 to +1....
0
votes
1answer
182 views

Normalize speech features per dimensions or across dimensions

I am working on pipelines for audio processing (audio features that will go into a neural net to create some embeddings). We are using two features : Filter banks, or MFCC. I was wondering what was ...
0
votes
1answer
68 views

STFT to spectrogram

I would like to know whether I am correct in my understanding of going from STFT to a spectrogram. My goal is to convert a spectrogram back to a wav file. If I have my STFT: ...
0
votes
2answers
170 views

does audio feedback consist of harmonic content?

So sound from musical instruments as well as human voice consists of harmonics. just wondering if feedback from speakers also have harmonics component I have heard that feedback does not consist of ...
1
vote
2answers
94 views

Subtract audio to another audio

suppose I have a sample of a snare and that snare is used in a song,¿ there is an algorithm or tool that detects the sound of the snare and subtracts it, that is to say that cancels its phase or ...

1
2 3 4 5
26