MUSIC NOTE RECOGNITION USING FFT

E.N.V PURNA CHANDRA RAO 1, I. NARASIMHA RAO 2

1
professor & HOD, 2 Assistant professor
Department of electronics and communication Engineering
CMR College Of Engineering & Technology, Kandlakoya, Medchal Road, Hyderabad-501401

Abstract- Signals and sound waves are part of our everyday life. However, music is a
distinct type of signal. Musical sounds each have a certain pitch that we can differentiate
as notes. A song contains basically two things, vocal and background music. Reading a
song on sheet music and then playing it on an instrument is an easy task for any musician.
In this century, computer software has also been designed to do just this. Programs can
create audio files (music we can hear) from sheet music very effectively for a whole range
of instruments. A major problem is that the reverse task, listening to or recording audible
music and then generating the sheet music for that piece, is much more difficult to
complete for both computers and talented musicians alike. Extracting the characteristics of
a song becomes more important for various objectives like learning, teaching, and
composing. The idea of this project is to develop a program that would take an audio input
(a song) and process it, in order to give musical notes as an output.
Keywords: Time-Frequency Analysis; Musical Note; Sampling Frequency; Recording;
Extraction.
I. Introduction various objectives like learning,
Songs play a vital role in our day-to- teaching, and composing. This project
day life. A song contains two things, takes the song as an input, extracts the
vocal and background music. Where features, and detects and identifies the
the characteristics of the voice depend notes, each with a duration. First, the
on the singer and in the case of song is recorded and digital signal
background music, it involves a processing algorithms are used to
mixture of different musical identify the characteristics. The
instruments like piano, guitar, drum, experiment is done with several piano
etc. Extracting the characteristics of a songs where the notes are already
song becomes more important for known, and identified notes are
compared with original notes until the
detection rate goes higher. And then
the experiment is done with piano
songs with unknown notes with the
proposed algorithm. The ability to
derive the relevant musical
information from a live or recorded
performance is relatively easy for a
trained listener but highly non-trivial
for a learner and computer. For
several practical applications, it would
be desirable to obtain this information
in a quick, error-free, automated
fashion. This thesis discusses the
design of a software system that
accepts as input a digitized waveform
representing an acoustical music
signal, and that attempts to derive the
notes from the signal so that a musical
score could be produced. This signal
processing algorithm involves event
detection, or precisely where within
the signal the various notes begin and
end, and pitch extraction, or the
identification of the pitches being
played in each interval. The event
detection is carried out using the time
domain analysis of the signal, where
the problem arises at a different speed.
Pitch detection (nothing but frequency
identification) is more complicated
because of a situation we call
harmonic ambiguity; this occurs when
one pitch whose fundamental
frequency is an integer multiple of
another pitch. The problem is solved
by careful signal processing in both
the time domain signals and frequency
domain signals. The main objective of
this project is to create an aid tool for
learning for Musicians, Producers,
Composers, DJs, remixers, Teachers,
and Music Students. This project can
be treated as a box, where you give
any song as input and get the features
of the song out. This project aims to
propose methods to analyze and
describe a signal, from which the
musical parameters can be easily and
objectively obtained, in a sensible
manner. A common limitation found
in the musical literature is that how
such parameters are obtained is
intuitively satisfactory but, to our
view, not very sound from a signal
processing perspective.
 Fig 1: Note durations
Every song has a tempo or a speed at which
the music is to be played. Tempo is defined
as beats per minute, where a beat is usually
defined to be a particular length of the note.
All notes’ lengths are then given a value,
such as a quarter or a half. This value
determines how many beats that note should
last. Interestingly enough, a beat is usually
defined to be a one-quarter note, and thus a
quarter note is 1 beat, a half note is 2 beats,
and an eighth note is half a beat.
Fig 2: Frequency table music
II. LITERATURE SURVEY
The Scales are fundamental to all music. In
Ancient Greece, the scales are referred to as
modes. Some commonly known scales are
Major scale, Minor scale, jazz sale, blue
scale, etc. A Raga can be derived from a
scale. It is a unique personality or distinct
flavor which has no fixed rules precisely
which combination. The ascendants and
descendants in Indian classical music are
called Arohana and Avarohana. The home
notes are named Griha Swara, the dominant
is called Vaadi, and Subdominant is called
Samvadi. The Dissonant is called Vivaldi.
The landing notes or resting notes are called
Nyasa Swara. Minimum an octave must
consist of 5 notes at least in a Raga. Notes
sung with 3 or 4 notes in an octave are very
rarely performed. The root note is “Sa” and
a Raga must use “Ma” or “Pa” by default.
Either one of them or both can be used in the
same raga. Other notes are exceptional.
Various combinations can be performed
using the octaves. To expertise in the
concepts and to develop an algorithm, some
more references were viewed and have been
discussed below. In Musical Notes
Identification using Digital Signal
Processing [1] paper, the input is taken as an
audio file and it is processed to extract the
features to identify the note of the song. To
identify the characteristic of the song, digital
signal processing techniques are used and
has been explained. The piano songs are
only allowed as input. The piano songs are
used because the notes are known by us
already, the identified notes and the original
notes are compared until a higher rate is
detected. This method used here for the
identification of notes is more optimized
than the previous methods. We can get the
results by varying the parameters like
threshold values and width, with the time
duration of each note. Thus, it can be used
as a tool for learning the notes of a song.
In pyAudioAnalysis: An Open-
Source Python Library for Audio Signal
Analysis [8], is all about an open-source
Python library that provides a wide range of
audio analysis procedures including feature
extraction, classification of audio signals,
and supervised and unsupervised
segmentation. PyAudioAnalysis is licensed
under the Apache License. This paper
provided the implemented methodologies by
the theoretical background, along with an
evaluation of some metrics of the methods.
Several audio analysis research applications
use the pyAudioAnalysis speech emotion
recognition, depression classification based
on audiovisual features, smart-home
functionalities through audio event
detection, music segmentation, multimodal
content-based movie recommendation, and
health applications such as monitoring
eating habits. SVM regression map the
audio features extracted from the previous
steps to one or more supervised variables.
The library also provides a semi-supervised
silence removal functionality. Based on the
literature survey, future enhancements have
been drafted.
2.1 System Model
A sound can be characterized by the
following three quantities:
(i) Pitch.
(ii) Quality.
(iii) Loudness.
Pitch is the frequency of a sound as
perceived by the human ear. A high
frequency gives rise to a high pitch note and
a low frequency produces a low pitch note.
A pure tone is the sound of only one
frequency, such as that given by a tuning
fork or electronic signal generator. The
fundamental note has the greatest amplitude
and is heard predominantly because it has a
larger intensity. The other frequencies such
as 2fo, 3fo, and 4fo, are called overtones or
harmonics and they determine the quality of
the sound. Loudness is a physiological
sensation. It depends mainly on sound
pressure but also on the spectrum of the
harmonics and the physical duration.
2.2 Musical Notes
Humans can hear signal frequencies ranging
from 20-20 kHz. From this wide range,
some parts are associated with the piano.
Different pianos have different ranges. Each
tone of the piano has one particular
fundamental frequency and is represented by
a note like C, D, ...etc. as shown in fig 1.
The later C is 12 half steps away from the
previous one and has doubles the 2.4 Frequency & Fourier Transforms
fundamental frequency. Hence this portion (FFT)
(from one C immediate next C) is called one A Fourier transform provides the means to
octave. Different octaves are differentiated break up a complicated signal, like musical
by C1, C2, etc. tone, into its constituent sinusoids. This
method involves many integrals and a
continuous signal. We want to perform a
Fourier transform on a sampled (rather than
continuous) signal, so we have to use the
Discrete Fourier Transform instead.

Fig 3. An octave of a piano

III. Proposed Methodology

2.3 Digital Signal Processing for Music

When a sound wave is created by your
voice (or a musical instrument), it's an
analog wave of changing air pressure.
However, for a computer to store a sound
wave, it needs to record discrete values at Fig 4: Block diagram of the proposed

discrete time intervals. The process of methodology

recording discrete time values is called The audio signals stored in the computer are

sampling, and the process of recording different in format, sampling rate, the

discrete pressures is called quantizing. number of bits, or the original audio signal

Recording studios use a standard sampling containing sharp noise, which can affect the

frequency of 48 kHz, while CDs use the rate processing effect. At the same time, the unit

of 44.1 kHz. Signals should be sampled at of audio processing is in a frame, so before

twice the highest frequency present in the the feature extraction, the original audio data

signal. Humans can hear frequencies from needs to be pre-processed: into a unified

approximately 20-20,000 Hz, which format, pre-emphasis, segmentation, and

explains why common sampling frequencies windowing framing.

are in the 40 kHz range. The envelope of a sound can be measured in

four ways:
1. Attack – The attack is the portion of the
envelope that represents the time taken for
the amplitude to reach its maximum level.
Essentially it is the initial build-up of a
sound.
2. Decay – Decay is the progressive
reduction in the amplitude of a sound over
time. The decay phase starts as soon as the
attack phase has reached its peak. In the
decay phase, the signal level drops until it
reaches the sustain level.
3. Sustain – The sustain is the period during
which the sound is held before it begins to
fade out. Many instruments do not contain a
sustained phase.
Sound with long sustain (and no attach or
delay)
4. Release – The release is the final fade or
reduction in amplitude over time.
Fig 5 : Attack, decay, sustain, release graph
Every sound is different. Percussion
sounds start suddenly, then decay and
release quickly as no more energy is being
applied to sustain the sound. A bowed
string, on the other hand, may build up with
a slow attack, sustain for a short period and
then release. The concept of hearing
envelopes relies upon the root-mean-square
values (RMS) values of amplitude and not
peak-to-peak amplitudes. High peaks in the
signal will not necessarily make an
instrument sound loud unless the amplitude
is sustained for some time. Short peaks tend
to contribute to the character (timbre) of the
sound rather than the loudness.
Edge Detection: We need to determine the
edges of the audio wave to identify if a note
was played or not. Edge detection includes a
variety of mathematical methods that aim at
identifying edges. There are many methods
for edge detection, but most of them can be
grouped into two categories, search-based
and zero-crossing based. The search-based
methods detect edges by first computing a
measure of edge strength, usually a first-
order derivative expression such as the
gradient magnitude, and then searching for
local directional maxima of the gradient
magnitude using a computed estimate of the
local orientation of the edge, usually the
gradient direction. The zero-crossing-based
methods search for zero crossings in a
second-order derivative expression
computed from the image to find edges,
usually the zero-crossings of the Laplacian
or the zero-crossings of a non-linear
differential expression. As a pre-processing
step to edge detection, a smoothing stage,
typically Gaussian smoothing, is almost
always applied.
Convolution: Convolution is a dimensionality, to produce a third array of
mathematical operation on two functions (f
and g) that produces a third function (f*g)
that expresses how the shape of one is
modified by the other. The term convolution
refers to both the result function and to the
process of computing it. It is defined as the
integral of the product of the two functions
after one is reversed and shifted.
Convolution is a mathematical operation
that expresses a relationship between an
input signal, the output signal, and the
impulse response of a linear-time invariant
system. An impulse response is the response
of any system when an impulse signal (a
signal that contains all possible frequencies)
is applied to it. A linear time-invariant
system is a system that a) behaves linearly,
and b) is time-invariant (a shift in time at the
input causes a corresponding shift in time in
the output).
Fig 6: Convolution Process
The integral is evaluated for all values of
shift, producing the convolution function.
Convolution provides a way of multiplying
together two arrays of numbers, generally of
different sizes, but of the same
numbers of the same dimensionality.
Fast Fourier Transform (FFT) : The "Fast
Fourier Transform" (FFT) is an important
measurement method in the science of audio
and acoustics measurement. It converts a
signal into individual spectral components
and thereby provides frequency information
about the signal. FFTs are used for fault
analysis, quality control, and condition
monitoring of machines or systems. This
article explains how an FFT works, the
relevant parameters, and their effects on the
measurement result. Strictly speaking, the
FFT is an optimized algorithm for the
implementation of the "Discrete Fourier
Transformation" (DFT). A fast Fourier
transform (FFT) is an algorithm that
computes the discrete Fourier transform
(DFT) of a sequence, or its inverse (IDFT).
Fourier analysis converts a signal from its
original domain (often time or space) to a
representation in the frequency domain and
vice versa.

Fig 7: View of a signal in the time and
frequency domain
In the first step, a section of the
signal is scanned and stored in the memory
for further processing. Two parameters are
relevant:
● The sampling rate or sampling frequency
fs of the measuring system (e.g. 48 kHz).
This is the average number of samples
obtained in one second (samples per
second).
IV. Simulation Results
Fig 8: Various steps involved
● The selected number of samples; the block
length BL. This is always an integer power
to the base 2 in the FFT (e.g., 2^10 = 1024
samples)
From the two basic parameters fs and BL,
further parameters of the measurement can
be determined.
Bandwidth fn (= Nyquist frequency).
This value indicates the theoretical
maximum frequency that can be determined
by the FFT.
fn = fs / 2
For example, at a sampling rate of 48 kHz,
frequency components up to 24 kHz can be
theoretically determined. In the case of an
analog system, the practically achievable
value is usually somewhat below this, due to
analog filters - e.g. at 20 kHz.
 (a) (b) (c)
Fig 9: Data set songs recording out information (a), (b), (c).

(b) (b) (c)

Fig 10: Spectral Analysis of The Output for database audio music.

V. Conclusion up to eighth notes, a high success rate in

In this paper, we successfully designed and
implemented an effective and user-friendly
music note recognition system with FFT.
The target users of the system are not only
the people practicing music or learning to
play a musical instrument but also
professional musicians who cannot waste
their time figuring out the notes of an audio
sample. We have achieved a high precision
for hit detection for reasonably paced songs
note detection when single notes were
played at a time, and a very precise note-
length or tempo determination for all notes
which were detected and analysed.
However, there are a few shortcomings of
our project such as poor note-hit detection in
busy or fast songs, overlapping note-hits
causing single, held notes to appear as
multiple shorter notes, and faults in an
instrument and non-ideal behaviour that can
cause anomalies in output. Finally, our
system is not real-time. We can't implement
our process of musical note recognition as
the instrument is playing in real time. In
fact, through our MATLAB-based code, we
first input the sound file and it takes around
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