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
2
votes
2answers
3k views

What is the difference between pitch detection and Onset detection?

According to wikipedia Onset detection: Onset refers to the beginning of a musical note or other sound, in which the amplitude rises from zero to an initial peak. It is related to (but ...
0
votes
1answer
628 views

How to detect and remove ringback-tone and ivrs voice etc from the beginning of an audio call recording

In an audio recording (say a telephone conv. b/w two people), how would I programatically detect and remove the dial-tone at the beginning of a call using python. Ex : As you can see the first 15 ...
8
votes
4answers
634 views

How to determine the “variability” in a frequency range?

This is a continuation of a previous question. I'm trying to analyze breathing and snoring sounds, and while I can fairly well detect snoring now, breathing is a bigger challenge. I've learned that if ...
4
votes
1answer
1k views

What is the difference between PSOLA and TDHS time-scaling or pitch-shifting?

What is the difference (if any) between PSOLA (Pitch Synchronous Overlap and Add) and TDHS (time domain harmonic scaling) time-pitch modification algorithms?
1
vote
2answers
3k views

Matching 2 audio files

I'm doing research on how we can do matching between 2 audio files using Android to see if they have the same content or not. After doing lots of research, I have concluded that I must do 2 main ...
1
vote
2answers
1k views

Echo hiding technique

I'm trying to figure out how to use DSP techniques in order to both add artificial echoes to a sound file, but more importantly to be able to detect these echoes. I've been reading about it here: http:...
3
votes
2answers
2k 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 ...
16
votes
1answer
4k views

Find similar Music using FFT Spectrums

I have been experimenting some weeks to find a way to match/find similar Songs in a Library containing different Genres of Music. My first try was to Detect Features like Tempo or how much Bass there ...
32
votes
4answers
11k views

What are good ways to detect signal clipping in a recording?

Given a recording I need to detect whether any clipping has occurred. Can I safely conclude there was clipping if any (one) sample reaches the maximum sample value, or should I look for a series of ...
20
votes
6answers
871 views

What does one need to know in order to build a software synthesizer?

For someone who has some experience in software development, loves his synths and knows the [very] basics of digital audio representation, what are the first steps he should follow in order to program ...
20
votes
4answers
27k views

Libraries for Voice Activity Detection (Not Speech Recognition)

As follow up to my previous question I was wondering if there are any speech detection libraries in existence. By speech detection I mean passing in an audio buffer and getting back an index of where ...
8
votes
3answers
4k views

“Subtracting” one audio stream from another audio stream

I am a long time software engineer but have practically zero experience with signal/audio processing. I am interested in learning about signal processing via a use-case we have for one of our audio ...
8
votes
2answers
8k views

Noise reduction from very noisy audio

I am trying to write an algorithm that would automatically segment a piece of audio with bird calls recordings. My input data are 1 minute-long wave files and on the output I would like to get ...
5
votes
1answer
4k views

Voice Audio Detection algorithm

I have to detect speech intervals in large pre-recorded files in my project. I think, there won't be so much noise in the background (audio will be recordered in the room or even in the studio), but ...
3
votes
2answers
7k views

Eliminate Signal A from Signal B

i have 2 audio signals, both available in time and frequency domain: A: Noise B: Noise(A with minor variations) + Additive sounds (C) i want to eliminate Signal A from B to leave only C. C is for ...
5
votes
2answers
423 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 ...
5
votes
1answer
5k views

Signal filtering using Python

I'm trying to create an application using python that is capable of recording an audio signal and detecting short glitches in the signal. For example, I could be recording a 1000Hz sine wave and ...
3
votes
5answers
2k views

Pitch Detection: avoiding frequency doubling / halving

I am working on a tuning app. I have so far tried 3 different libraries with a number of different algorithms. However, when I compare to other tuner apps, I seem to be getting frequency doubling on ...
3
votes
2answers
232 views

Are there equations for placing sounds in the stereo field using delays?

I'm thinking of the typical stereo widener (haas, binaural) effects here, those that extend from the typical pan knob to create "out of the headphones" -experience. These processors can be automated ...
3
votes
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 ...
1
vote
2answers
1k views

How do I explain a complex exponential intuitively?

What is a complex exponential, explained intuitively? How do I explain to an adolescent a complex exponential function?
14
votes
1answer
1k views

Estimating onset time of a tone burst in noise?

What techniques might one use to estimate the onset time of a sinusoidal tone burst in a noisy signal? Assume the tone burst has a known fixed frequency (but unknown phase) and a very sharp rise ...
9
votes
2answers
616 views

Pattern recognition for temporal data

I'm trying to detect and classify non-speech sounds. Currently, I'm using a series of moving overlapped power spectrums from training sounds as the features I am looking for. When I do analysis, I'm ...
7
votes
1answer
459 views

How to mix two signals without changing the overall loudness?

I have two audio signals that I want to mix at various mixing ratios. Initially, I simply went for something like $y(t) = \alpha \cdot x_1(t) + (1-\alpha) \cdot x_2(t)$ where $\alpha$ is the ratio ...
7
votes
2answers
6k views

Loudness of PCM stream

I'm receiving PCM samples through stream which is one channel, 8000 samples per second. I figured out way of finding out loudness by calculating 'force' of signal by adding abs value of every next ...
6
votes
1answer
9k 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 ...
5
votes
6answers
1k views

Find Short Clip of Audio Within Longer Clip Of Audio

I need help figuring out a way to automatically synchronize two audio files that are both different recordings of the same source. More info below. This example is from a concert, where there is only ...
3
votes
2answers
43k views

Removing noise from audio using Fourier transform in Matlab

I'm trying to remove noise from an audio file. This audio file contains speech as well as constant pink noise. I know that I have to use the Fourier transform to convert to the frequency domain then ...
14
votes
2answers
1k views

What bandpass filter design will yield the shortest impulse response?

Designing a simple 2nd order IIR bandpass Buterworth filter with a centre frequency of 500 Hz and a 1 octave bandwidth gives me the following frequency response ... Now, if I take the impulse ...
10
votes
5answers
880 views

Mathematical question that comes out of using bilinear transform

So this is related to the Cookbook and I tried solving it maybe two decades ago, gave up, and was reminded of the unsolved problem. But it's pretty damn straight forward, but I still got slogged down ...
9
votes
1answer
6k views

Measuring time delay of audio signals

Before anyone yells at me I fully realize that this question has been asked numerous times. I assure you that I've read through the existing questions and answers but I am still confused about part of ...
9
votes
1answer
3k views

Synchronizing Two Audio Tracks

I would like to implement an algorithm that synchronizes two audio tracks. The two tracks are very similar; you can imagine they are two recordings of sound (not necessarily music) from two ...
8
votes
1answer
7k views

1/n octave smoothing

Given a frequency response obtained with FFT, I would like to apply a 1/n octave smoothing. What filter should I be using and how? Maybe someone could point to a good reference (a paper or book on the ...
4
votes
1answer
2k views

How does Adobe After Effects generate its “audio spectrum” effect?

I'm trying to replicate the "audio spectrum" effect from Adobe After Effects. An example can be seen in this video: Obviously, it has to be some variant of a fourier transform, but I've tried taking ...
4
votes
3answers
4k views

Detection of wobbling sound with rising notes

In an audio file I would like to find calls of a specific bird specie, one that makes wobbling sound with rising notes. Can you recommend what would be the best audio feature to calculate in time or ...
3
votes
1answer
4k 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 ...
3
votes
1answer
280 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
votes
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 ...
2
votes
2answers
97 views

How does mic combined with loudspeaker work?

There is variety of loudspeakers on the market that have built-in microphone so that they can produce sound and perceive it at the same time (and do it surprisingly well). How they managed to ...
2
votes
2answers
1k views

Problem with Generalized Cross Correlation (PHAT filter)

I have to calculate time delay between two signals.I am using the GCC-PHAT method. First, I cut the signals into smaller parts (~0.5s) and then try to calculate the delay. When I display those two ...
1
vote
3answers
1k views

How to add pitch-shifted version of signal to itself?

I want to write an algorithm to add pitch-shifted version of audio signal to itself (like Electro-Harmonix POG does). And I need to do it in realtime. How to implement it? My guess is: break signal ...
1
vote
2answers
202 views

Understanding DFT-ODD operation in Gardner Efficient Convolution paper

As a newbie to DSP I am trying to understand and implement an efficient convolution engine as per JAES V43.3 1995/03 - Gardner - Efficient Convolution without I/O Delay. I'm pretty much there with a ...
1
vote
2answers
621 views

Distorted output of delay VST effect in C

I'm getting some odd results with some code I'm working on. Help me with the implementation of this delay effect? The code takes an input, adds delayed input to it and then passes it to the output ...
7
votes
1answer
323 views

How do Phons relate to loudness?

Let's say I have the very same signal stored in 2 .wav files. And lets say I process each file using a function, and it is established that one file is 20 phons ...
7
votes
1answer
5k views

Plotting the magnitude response of a biquad filter

I've put together a multi-band audio EQ using biquad filters. I'm getting the coefficients using the methods from the RBJ cookbook. Now I want to plot the curve showing the magnitude response. I'm ...
5
votes
1answer
633 views

RMS values above 0 dbFS?

I wonder why several RMS meters I'm using for music production show RMS values above 0. For example, when using an input signal with a constant DC of 1 (0 dbFS), I get these results, although ...
5
votes
0answers
623 views

What is EAQUAL ? How does it relate to PEAQ?

This wiki post on wiki.hydrogenaudio.org says that it is very similar to PEAQ and also a part of ITU-R BS.1387 recommendation ? How reliable it is for determining audio quality. Are there any other ...
4
votes
2answers
417 views

How can I measure the harmonicity of an audio spectrum?

I would like to estimate the "harmonicity" (harmonics-to-noise ratio ?) of an audio signal from its spectrum. What kind of algorithm can I use?
4
votes
1answer
773 views

$1/n$ octave complex smoothing

An excellent answer to this post explains how to do $1/n$ octave energy smoothing and mentions complex smoothing can be done as well, but it's tricky business because of phase wrapping. How is phase ...
3
votes
2answers
3k views

Which filter for an audio equalizer

I’m new to DSP and I have to do an audio equalizer in c++. I did a lot of research about it and tried some stuff in the last month, but I’m a little overwhelmed with all those informations and it’s ...