Dereje Teferi (PHD) Dereje - Teferi@Aau - Edu.Et

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Dereje Teferi(PhD)

dereje.teferi@aau.edu.et

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What is sound?

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Wave parameters
• Sound travels in waves. Waves have the following characteristics

• Wavelength is distance Infra-sound 0 – 20 Hz


sound travels in one cycle Human hearing range 20 – 20 kHz
• 20 Hz is about 16.5 meters Ultrasound 20 kHz – 1 GHz
• 20 kHz is 1.6 cm Hyper sound 1 GHz – 10 THz
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Phase angles

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Fourier Series
 The sine wave is the simplest possible wave.
 In 1807, Fourier proved that any repeating wave form
could be broken down into a series of sine waves.
 The lowest frequency sine wave (in a repeating wave
form) is called the Fundamental.
 The frequencies of the other sine waves are always
integer multiples (2x, 3x, 4x, etc) of the fundamental.
 These are called Harmonics.
 Each harmonic has a different amplitude and phase
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Adding Harmonics

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Sound Fields
directed sound
early reflections reverberation
(50-80 msec)
amplitude

time

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Analog to digital conversion
 Sound waves are continuous
 One needs analog to digital convertor to store
audio/sound in a computer (or any digital device)
 Sampling is required
 The sampled data then need to be quantized

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Quantization
 Quantization in general approximates a signal varying
continuously in amplitude by one whose amplitude is
restricted to a prescribed set of discrete values
 It is a method to digitize the sampled analog signal
 It determines how many discrete digital values should
be used to encode the sampled signals? and
 What analog value does each digital value correspond
to?
 Commonly used quantization methods are Linear (8 bit
and 16 bit) and logarithmic

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The sample and hold technique
 Sample at a specific time intervals
 Hold the sample value until the next sample position is
reached
 This allows quantization
 Then the digitized codes are transmitted via digital devices
as numbers

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Reconstruction of audio from
digitized data

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Increasing the sampling rate

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Low Vs. High sampling rate

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More example
 Real music sample at two different sampling rates
(8KHz and 48KHz)

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Sampling
 Information is lost during digitization especially in the
first example where the samples are small in number
 This is basically due to the sampling rate
 If too much samples are taken, the amount of data will be
huge
 The questions now are
 How many samples should we take between every cycle?
 Should we take the same number of samples for high
frequency as well as low frequency?

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Nyquist-(Shannon) Theorem
• Nyquist Theorem (written by Harry Nyquist in 1928 and
proved by Claude Shannon in 1949) suggests that Optimal
sampling frequency should al least be twice the highest
frequency of the audio to be sampled
• In mathematical terms:
fs > 2*fm
where fs is sampling frequency (the No of samples taken in one second) and fm is the
maximum frequency in the signal

• Although certain limitations are observed, any arbitrary


wave form can be recorded by taking samples at fixed time
intervals.
• The wave form is played back by outputting the same
samples at the same fixed intervals.
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• Some example samples for specific applications are listed in
the table below

Format Sampling Rate Bandwidth Frequency Band


Telephony 8 kHz 3.2 kHz 200-3400 Hz
Teleconferencing 16 kHz 7 kHz 50-7000 Hz
Compact Disk 44.1 kHz 20 kHz 20-20,000 Hz
Digital Audio Tape 48 kHz 20 kHz 20-20,000 Hz

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Sampling rate

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Digitization and playback

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Aliasing
Aliasing occurs when the input frequency is greater
than half the sampling frequency.

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Audio filtering
 To reduce aliasing errors, analog to digital convertors
apply low pass filtering on the input signal with a cut
of frequency of (fs/2)

 the signal is sampled at sampling frequency fs

 Then the sampled signal is quantized into discrete


values

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Sampling rate Vs storage

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Audio Compression
 Compression in general is a mechanism used to reduce the
amount of storage space required by a signal
 The signal could be text, audio, image, video or any
combination of them

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Reading Assignment
 Overview of speech codecs
 PCM (Pulse Code Modulation)
 DPCM: Differential PCM
 ADPCM : Adaptive differential PCM
 etc

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