Digital Audio

Multimedia Systems (Module 1 Lesson 1)

Summary: Sources:
 Basic concepts  My research notes
underlying sound  Dr. Ze-Nian Li’s course
 Facts about human material at:
perception of sound
http://www.cs.sfu.ca/CourseCentral/365/li/

 Computer
representation of
sound (Audio)
 A brief introduction to
MIDI

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Sound Facts
 Sound is a continuous wave
that travels through the
air
 The wave is made up of
pressure differences.
 Sound is detected by
measuring the pressure The human Ear detecting Sound
level at a location
 Sound waves have normal
wave properties
(reflection, refraction,
diffraction etc.)

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Sound Facts
Wave Characteristics
 Frequency: Represents the
number of periods in a
second and is measured in
hertz (Hz) or cycles per

Air Pressure
Amplitude
second.
Human hearing frequency Time

range: 20Hz to 20kHz

(audio) One Period
One particular frequency component
 Amplitude: The measure of
displacement of the air
pressure wave from its
mean. Related to but not
the same as loudness

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Principles of Digitization
 Why Digitize?
 Microphones, video
cameras produce analog
signals (continuous-
valued voltages)
 To store audio or video Time

data into a computer,

we must digitize it by Sound as analog signal
converting it into a
stream of numbers.

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Principles of Digitization
 Sampling: Divide the horizontal axis (time) into discrete
pieces
 Quantization: Divide the vertical axis (signal strength -
voltage) into pieces. For example, 8-bit quantization
divides the vertical axis into 256 levels. 16 bit gives you
65536 levels. Lower the quantization, lower the quality of
the sound
 Linear vs. Non-Linear quantization:
• If the scale used for the vertical axis is linear we say its linear
quantization;
• If its logarithmic then we call it non-linear (-law or A-law in
Europe). The non-linear scale is used because small amplitude
signals are more likely to occur than large amplitude signals, and
they are less likely to mask any noise.

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Sampling and Quantization

Sample
Sample

Time Time

Sampling 3-bit quantization

 Sampling rate: Number of  3-bit quantization gives 8
samples per second possible sample values
(measured in Hz)  E.g., CD standard audio
 E.g., CD standard audio uses 16-bit quantization
uses a sampling rate of giving 65536 values.
44,100 Hz (44100 samples  Why Quantize?
per second)  To Digitize!

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Nyquist Theorem

Consider a sine wave

Sampling once a cycle

Appears as a constant signal

Sampling 1.5 times each cycle

Appears as a low frequency
sine signal

 For Lossless digitization, the sampling rate should

be at least twice the maximum frequency
responses 7
Application of Nyquist Theorem
❑ Nyquist theorem is used to calculate the optimum sampling
rate in order to obtain good audio quality.
❑ The CD standard sampling rate of 44100 Hz means that the
waveform is sampled 44100 times per sec.
❑ Digitally sampled audio has a bandwidth of (20 Hz - 20 KHz).
By sampling at twice the maximum frequency (40 KHz) we
could have achieved good audio quality.
❑ CD audio slightly exceeds this, resulting in an ability to
represent a bandwidth of around 22050 Hz.

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Quantization (Quality ->SNR)
 In any analog system,  Signal to Quantization
some of the voltage is Noise Ratio (SQNR)
what you want to  The quantization error
measure (signal), and (or quantization noise)
some of it is random is the difference
fluctuations (noise). between the actual
value of the analog
 SNR: Signal to Noise signal at the sampling
ratio captures the time and the nearest
quality of a signal (dB) quantization interval
value.
V2signal Vsignal
SNR = 10 log = 20 log  The largest (worst)
V2noise Vnoise quantization error is
half of the interval?
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SQNR Calculation (WC)
 If we use N bits per sample, the range of
the digital signal is: -2N-1 to 2N-1
 The worst-case signal to quantization noise
ratio is given by:
Vsignal 2N-1
SQNR = 20 log = 20 log = N x 20 log 2 = 6.02N (dB)
Vquant - noise 1/2

 Each bit adds about 6 dB of resolution, so

16 bits enable a maximum SQNR = 96 dB.

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Miscellaneous Audio Facts
Typical Audio Formats
 Popular audio file formats include .au
(Unix), .aiff (MAC, SGI), .wav (PC, DEC)
 A simple and widely used audio
compression method is Adaptive Delta
Pulse Code Modulation (ADPCM). Based on
past samples, it predicts the next sample
and encodes the difference between the
actual value and the predicted value.

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Audio Quality vs. Data Rate
Quality Sample Rate Bits per Mono/ Data Rate Frequency
(kHz) Sample Stereo (kBytes/sec) Band
(uncompressed)

Telephone 8 8 Mono 8 200-3400

Hz

AM Radio 11.025 8 Mono 11.0 540-1700

KHz

FM Radio 22.050 16 Stereo 88.2

CD 44.1 16 Stereo 176.4 20-20000

Hz

DAT 48 16 Stereo 192.0 20-20000

Hz

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MIDI
 Musical Instrument Digital Interface
a protocol that enables computer, synthesizers, keyboards,
and other musical devices to communicate with each
other.  Setup:
 MIDI OUT of synthesizer is
connected to MIDI IN of sequencer.
THRU IN OUT IN OUT  MIDI OUT of sequencer is connected
to MIDI IN of synthesizer and
Synthesizer/Keyboard MIDI Interface/Sound Card "through" to each of the additional
(Sequencer) sound modules.
IN THRU  Working:
During recording, the keyboard-
MIDI Module A

equipped synthesizer is used to send
IN THRU MIDI message to the sequencer,
MIDI Module B which records them.
 During play back, messages are sent
Etc.
out from the sequencer to the sound
Typical Sequencer setup modules and the synthesizer which will
play back the music.
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MIDI: Data Format
 Information traveling through the hardware is encoded in
MIDI data format.
 The encoding includes note information like beginning of
note, frequency and sound volume; upto 128 notes
 The MIDI data format is digital
 The data are grouped into MIDI messages
 Each MIDI message communicates one musical event
between machines. An event might be pressing keys, moving
slider controls, setting switches and adjusting foot pedals.
 10 mins of music encoded in MIDI data format is about 200
Kbytes of data. (compare against CD-audio!)

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