Abstract

The standard time-frequency representations calculated to serve as features for musical audio may have reached the extent of their effectiveness. General-purpose features such as Mel-Frequency Spectral Coefficients or the Constant-Q Transform, while being pyschoacoustically and musically motivated, may not be optimal for all tasks. As large, comprehensive, and well-annotated musical datasets become increasingly available, the viability of learning from the raw waveform of recordings widens. Deep neural networks have been shown to perform feature extraction and classification jointly. With sufficient data, optimal filters which operate in the time-domain may be learned in place of conventional time-frequency calculations. Since the spectrum of problems studied by the Music Information Retrieval community are vastly different, rather than relying on the fixed frequency support of each bandpass filter within standard transforms, learned time-domain filters may prioritize certain harmonic frequencies and model note behavior differently based on a specific music task. In this work, the time-frequency calculation step of a baseline transcription architecture is replaced with a learned equivalent, initialized with the frequency response of a Variable-Q Transform. The learned replacement is fine-tuned jointly with a baseline architecture for the task of piano transcription, and the resulting filterbanks are visualized and evaluated against the standard transform.

Library of Congress Subject Headings

Automatic musical dictation--Data processing; Musical analysis--Data processing

Publication Date

6-2019

Document Type

Thesis

Student Type

Graduate

Degree Name

Computer Engineering (MS)

Department, Program, or Center

Computer Engineering (KGCOE)

Advisor

Andres Kwasinski

Advisor/Committee Member

Juan Cockburn

Advisor/Committee Member

Alexander Loui

Campus

RIT – Main Campus

Plan Codes

CMPE-MS

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