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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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261 views

How to add more cowbell?

Remember that Christopher Walken skit on Saturday Night live where he told the band “it needs more cowbell” https://en.wikipedia.org/wiki/More_cowbell ? This site automated it http://www.morecowbell....
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1answer
1k views

How to make pop-free, crisp two-beep ringtone?

I hope this ultra-noob question is not entirely out of place here. I'm trying to make a super simple ringtone: 0.5s silence + 1s "pure tone" + 0.5s silence + 1s "pure tone" + 0.5s silence. By "pure ...
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1answer
7k views

Taking audio frequency and picking out peaks (Peak Detection)

So I have a unique challenge. I want to take an audio file, and pick out the peaks in the samples and plot that onto a graph. Each point based on the amplitude/pitch(?) will be represented as either a ...
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4answers
403 views

Why every musical instrument sounds different than any other while listening?

This is soft question.I am not expert in music. We know that English language has 26 letters (which are again grouped into vowels and consonants) , while language of music is same all over the world ...
3
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1answer
801 views

Lip sync over IP

What are the most used techniques for lip sync (audio/video sync) for conferencing over IP (and related issues with playout buffering)? I could find a lot of paper about audio intrastream ...
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2answers
1k views

Signal Processing/FFT gives very high magnitudes for low frequencies

The script essentially pulls PCM data from the sound card and stores them in a buffer. Then I apply fft and graph it. I am looking through this blog post for some examples and ideas on how to ...
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2answers
1k views

Subtracting Original Audio Signal

here is my scenario: one speaker is sending a chirp, and a microphone is recording it as well as all following echoes. The function of the chirp is already know. My goal: subtract the original chirp ...
3
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2answers
8k views

Simple beat detection algorithm for microcontroller

For the purpose of controlling DIY disco lighting, I'm looking at implementing beat detection in a microcontroller with limited computing power (i.e. Arduino Uno). While detecting all beats would be a ...
3
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2answers
128 views

What is the theoretic difference in the ways to compute the spectrogram?

I have noticed that the spectrogram for a audio signal is often calculated in different ways when the result from the DFT is transformed into real numbers; some tend to compute $|k|$ or $|k|^2$ while ...
3
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2answers
324 views

Linear predictive model convolution

Accidentally asked this question in the general area and was told to ask here, so... I've been trying to develop a lightweight, relatively-fast-to-decode sound compression format for use in my gaming ...
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1answer
277 views

Stereo sound separation into three sources

Whenever we listen stereo songs, one part of the audio like voice hear at the center of the head, and other parts are nearby left ear and right ear. If I represent center part by C, and other parts by ...
3
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1answer
858 views

Common Filter Types for Audio Applications

I'm beginning to write a basic "virtual analog" synthesizer and want to implement a lowpass with variable cutoff frequency and resonance. Does anyone have insight into filter types commonly used to ...
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2answers
1k views

Upsampling PCM audio: from 6 kHz to 8kHz

What would a straight-forward way to convert an audio payload (PCM 16bit) from 6 kHz to 8kHz? I understand that this is an interpolation problem. I fear that a linear interpolation would introduce ...
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2answers
3k views

Hilbert transform filter for audio applications: Using IIR half-band parallel all pass structure

Does anyone have any experience designing wideband IIR Hilbert transform filters with audio applications in mind? I am using the filter for single side band modulation audio effects discussed in this ...
3
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1answer
96 views

Definition of DSP Terms

can someone define/explain briefly (or with some detail-> up to you) the following terms: Block Processing? and what are the differences between it and Sample Processing? (removed) What is a 3D ...
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2answers
247 views

How to detect continuous noise in audio call?

We want to detect bad audio quality in audio call. Should we extract this information from amplitude or frequency spectrum ? Is there any technique/transform/filter available for this ? Example: I ...
3
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1answer
7k views

MFCC feature vector from wav file

I am trying to implement a spoken language identifier from audio files, using Neural Network. I need to generate one feature vector for each audio file. From what I have read the best features (for ...
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4answers
2k views

How to calculate an azimuth of a sound source in stereo signal

Let say we have a stereo recording of only one source. What is a correct algorithm to calculate its azimuth using inter channel time difference and inter channel level difference?
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2answers
199 views

“Flanging” in frequency domain?

Flanging is defined as a mix of two identical signals where one signal is delayed in time by a small and gradually changing period, around 10-20 milliseconds. Since delay in the time domain is ...
3
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1answer
951 views

Emulate tape delay

I'm trying to emulate a tape delay in a VST plugin, but I'm not sure how to handle the delay changing? I want it to sound like a tape is actually fast forwarding/rewinding. Most digital delays I've ...
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2answers
1k views

How to identify useful elements of audio recording and ignore lulls?

I assume this is a relatively simple DSP task, but I'm having trouble finding any information on how to approach my problem. Hopefully someone with more experience and brain power can help me out: I ...
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2answers
2k views

Audio signal dither and noise shaping

I'm trying to get a handle on the importance of the error feedback term in noise shaping operation in typical audio dither algorithms. I'm thinking in terms of four signals. The original signal, the ...
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2answers
2k views

what does musical noise sound like and is there a sample clip for it?

Musical residual noise is created when a noisy signal passes through a noise reduction algorithm. For instance, please see the paper: Esch, T; Vary, P: "Efficient Musical Noise Suppression for Speech ...
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1answer
1k views

pitch vs playback rate

Let's say I have an audio sample, whose pitch is known to be C4 (don't wanna go in detail here, let's say I just sampled a tuned piano playing the C4 key). The rules of the game are: I can only ...
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2answers
7k views

algorithm for feature extraction from audio signals

I am trying to build a model for speaker identification, and I understand that the first step is to extract the features from the audio signals that are in my database. I have done quite a bit of ...
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2answers
263 views

Which parameters should I take care of while converting .au to .wav for Genre Classification

I am learning to develop Music Genre Classification models. I have a dataset in which files are in ".au" format, which I need to convert into ".wav". I used SoX (Sound eXchange) to convert it. With a ...
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1answer
2k views

Use MATLAB to Restore a Signal from a Given Degraded Signal Using Tikhonov Regularization

Anyone could share how to develop an application in MATLAB to restore the signal from a given degraded signal using Tikhonov regularization i.e restoring the signal $f$ via solving $$ \min || g - f ...
3
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1answer
443 views

how to correct minBLEP sawtooth oscillator DC bias

In my implementation of minBLEP, overlapping minBLEPs result in a DC bias as the frequency increases. This is because each minBLEP correction (i.e. the minBLEP minus the aliased step) has a DC bias. ...
3
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1answer
158 views

stretch factor in SOLA time stretch methods

I notice that in the general Synchronous Overlap and Add (SOLA) algorithm, we need to obtain the best overlapping start index using, for example, cross-correlation. But won't that start index change ...
3
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1answer
432 views

5.1 Rear To 5.1 Side mixing matrix

I am searching for a matrix definition for converting a audio signal (5.1 with rear) to a audio signal (5.1 with side). Currently I am using the definitions from: https://msdn.microsoft.com/en-us/...
3
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1answer
147 views

Codec Output has “Wavy” Amplitude on Chirp Input

I am sending a chirp that sweeps form 20Hz to 500Hz to an audio codec, CS42L52, connected to an MCU, STM32F405, and I am receiving a "wavy" amplitude output. There is no processing on the MCU, I pass ...
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1answer
3k views

How to model Tape Saturation (Audio DSP)?

I'm looking for info that would help me to build a Tape Saturation process. I'm working with the WebAudio API. The API provides a number of DSP nodes that perform basic processes, such as a Convolver ...
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2answers
1k views

How does Songify work?

Does anyone know how does the "Songify" app algorithm work? There are 2 inputs: speech (or amateur singing), and underlying music. Somehow the speech is altered and mixed into the music. Which DSP ...
3
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1answer
822 views

Equal loudness curves: any good numerical approximation equations?

What might be, if any, some good numerical approximation functions to the ISO 226 family of human perceptual equal loudness curves? Is there a polynomial fit of reasonably low degree?
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2answers
868 views

change the frequency component of a spectrum in an audio file

I am working on an audio dynamic range compressor using Matlab. After calculating the FFT of the audio file, I use several bandpass filters to amplify specific bands separately. What I want to do then ...
3
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1answer
325 views

Real-time single speaker recognition

I'm developing an application for PC that records microphone data in real-time, and plays it back after applying some effects. I'd like to apply these effects only when a single known speaker is ...
3
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1answer
481 views

From a 44kHz audio signal, how could I extract 32 evenly spaced frequency bands?

Assume I have an array of floating points that contain the last 8192 samples played. How would I convert that array to 32 frequency bins that are spaced from 0Hz - Nqyst?
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1answer
425 views

Am I handling offline FFT correctly?

I need some help clarifying FFTs and what they represent. I have a buffer containing compressed audio. Due to limitations, I can't handle the full uncompressed audio but can decompress small segments ...
3
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1answer
202 views

How can I calculate the entropy of a signal that's not independent from itself?

I really don't know how to word this better, sorry if that title is confusing. Basically, I'm wanting to use the noise from an AM radio audio signal to generate random numbers. To do this effectively,...
3
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1answer
349 views

Detecting a low frequency cue in audio recording

I am hoping to find a linux program or library for a language like perl or python (open to ...
3
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1answer
176 views

Acoustical Design considerations for Active Noise Cancellation Project for Air vent

I am working on ''Active noise cancellation for air vent'' project. I am using NI myRIO1900, Audio Technica ATR 3350 omni directional microphones and Tevion Loudspeakers. I have done the programming ...
3
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1answer
170 views

Accuracy of IFFT(FFT(X)) unit transformation

I want to use FFTs to do seriously accurate interpolation on band-limited data. To do that, I need to get a handle on the fundamental accuracy of the FFT() and IFFT() algorithms available. My idea ...
3
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4answers
611 views

Removing Bounded Noise from an Audio Signal

I'm currently trying to remove some noise from a "contaminated" audio file. From what's given, I know that the noise is isolated to a specific frequency band - that is, from ~1400 Hz - ~1800 Hz. The ...
3
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1answer
135 views

Controlling one signal with another one (Rocket engine effect - Subtractive synthesis of sound)

I am programming some sound effects in Java and exporting them into .wav files. Currently, I am trying to program a rocket engine sound effect. I want to do it in the following way: The sound of a ...
3
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1answer
114 views

Compressing A Range Of Decibel Values to One “Representative” Value

Don't have any formal dsp experience - sophomore in college. Working on constructing a 25 band bark spectrogram for psychoacoustic analysis of audio FFT data. To do this, I'm grouping all FFT ...
3
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1answer
781 views

Skype's SILK Codec

Let me start by clarifying the terminology that I will be using in the following. A codec packet is the compressed data which an encoder returns when it is called once. An RTP payload can consist of ...
3
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1answer
608 views

Audio Signal Noise Filter Problem

I'm currently working with audio signals and have a problem: C = A*B + N, where C = recorded signal from microphone consisting of: A = known music file data played on speakers next to microphone ...
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1answer
1k views

Automatic Gain Control for voice

I am tasked with designing the logic for an automatic gain control. It won't be a traditional AGC in that it has to run on a computer and continuously modifying the volume is not possible in the same ...
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3answers
4k views

Output of KISS FFT interpretation

I'm receiving PCM data trough socket connection in packets containing 320 samples. Sample rate of sound is 8000 samples per second. I am doing with it something like this: ...
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1answer
173 views

Estimating harmonic loudness

I'm working with data that looks something like this: ...