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Author: | Haghparast, Azadeh |
Title: | Low-Latency Pitch-Bending Technique for Audio Signals |
Publication type: | Master's thesis |
Publication year: | 2006 |
Pages: | 66 Language: eng |
Department/School: | Sähkö- ja tietoliikennetekniikan osasto |
Main subject: | Akustiikka ja äänenkäsittelytekniikka (S-89) |
Supervisor: | Välimäki, Vesa |
Instructor: | Penttinen, Henri |
OEVS: | Electronic archive copy is available via Aalto Thesis Database.
Instructions Reading digital theses in the closed network of the Aalto University Harald Herlin Learning CentreIn the closed network of Learning Centre you can read digital and digitized theses not available in the open network. The Learning Centre contact details and opening hours: https://learningcentre.aalto.fi/en/harald-herlin-learning-centre/ You can read theses on the Learning Centre customer computers, which are available on all floors.
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Location: | P1 Ark S80 | Archive |
Keywords: | audio effect correlation function digital signal processing digital filter onset detection pitch-shifting resampling time-scale modification |
Abstract (eng): | A new high-quality low-latency pitch-bending algorithm is presented. The algorithm consists of three main sections: pitch-shifting algorithm, onset processing algorithm, and gain control and equalization. The pitch-shifting algorithm is based on the resampling and time-scale modification method. However, the previous time-scale modification methods could not achieve the required latency and quality for the system. A new time-domain time-scale modification algorithm has been designed, which is called the Normalized Filtered Correlation Time Scale Modification (NFC-TSM) method. In order to modify the time-scale of the signal, some parts of the signal are repeated or discarded. The best splicing point is searched in the normalized low-pass filtered signal using the Average Magnitude Difference Function (AMDF) as the correlation function. In order to obtain the feeling of realistic playing, the onset processing algorithm is used. It detects the onset of the input signal with the required latency and performs time-synchronization with the new onset. Gain control and equalization are used to simulate the gain and timbre of the physically pitch-shifted signal. The presented algorithm can be exploited in the applications that are using audio effects particularly. Because of being low-latency, it can be used in real-time performance of music. |
ED: | 2006-06-14 |
INSSI record number: 32014
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