IT101

1
IT-101

• Introduction to Information Technology

Lecture 6-7
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Overview of Chapters 10-13

• Chapter 10
• Definition of Sound
• Frequency
• Audio Signals

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Definition of Sound

• Vibrations transmitted through an elastic solid
  or a liquid or gas, with frequencies in the
  approximate range of 20 to 20,000 hertz, capable
  of being detected by human organs of hearing.
• Transmitted vibrations of any frequency.
• The sensation stimulated in the organs of hearing
  by such vibrations in the air or other medium.
• Such sensations considered as a group.
• http//dictionary.reference.com/search?qSOUND

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Sound to Signal

• It is the motion of air that we perceive as sound
• To create sound, we must provide a mechanical
  force to the surrounding air.
• In response to this force, the air is compressed
  and the difference in pressure between the
  compressed air and the uncompressed air causes
  the compression to circulate away from the
  source of the sound.
• A common example, when you drop a stone into a
  pool of water.

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Drop a stone into water

• The water is compressed in the vicinity of the
  stone.
• Due to the difference in pressure between the
  area where the stone has landed and the
  surrounding pool.
• The compression begins to circulate outward,
  causing the familiar concentric ripples.

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Frequency

• Frequency is the measurement of the number of
  times that a repeated event occurs per unit time.
  To calculate the frequency, one fixes a time
  interval, counts the number of occurrences of the
  event within that interval, and then divides this
  count by the length of the time interval.
• The frequency, in hertz, is a measurement of how
  rapidly the audio signal is changing.

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Audio Signal

• All audio signals are comprised of a sum of
  different pure tones, each at a different
  frequencyan audio signal can be described in two
  different ways.
• Time-domain description
• Frequency spectrum

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Fourier

• A French mathematician proved that all waveforms
  (musical, speech, i.e.) can be constructed of the
  sum of pure tones.
• The implication of this theory is that every
  audio waveform can be built out of sinusoids at
  certain frequency
• Sinusoidal signals have a frequency of repetition
  that determines the pitch of the tone.

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Frequency Spectrum

• Specifying which frequency components the signals
  contains and in what amount each frequency or
  pure tone occurs.
• The description of a signal is referred to as the
  frequency-domain description.

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Chapter 11

• Sampling
• Sampling Theorem

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Sampling

• The analog audio signal is a time-continuous
  waveform. The waveform is measured, or sampled,
  at periodic intervals.
• These samples form a series of discrete amplitude
  pulses that, together and in sequence, form a
  representation of the continuous waveform.
• http//www.mtsu.edu/djbrown/Digital_Audio_Fundame
  ntals.html

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The Sampling Theorem

• There must be enough samples to faithfully
  reproduce the frequency in question. We could
  take as many samples as possible, but there are
  reasons to limit the number
• Hardware limitations - hardware can only gather
  the samples so fast.
• Storage limitations - the samples must be stored,
  and there is only so much room available to store
  the given samples.
• How many samples are needed? There must be enough
  to accurately reproduce any and all frequencies
  in the original signal.

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History of Sampling Research

• 1915 - E.T. Whitaker devised a proof showing that
  a band-limited function can be reconstructed from
  samples.
• 1920 - K. Ogura proved that if a function is
  sampled at a frequency at least twice the highest
  function frequency, it could be reconstructed
  from those samples.

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History of Sampling Research

• 1928 - Bell Labs engineer Harry Nyquist published
  an article titled Certain topics in Telegraph
  Transmission Theory. In this article he provided
  proof that for complete signal construction, the
  frequency bandwidth is proportional to the
  signaling speed, and that the highest frequency
  is equal to half the number of code elements per
  second.
• 1949 - Claude Shannon unified many aspects of
  sampling, founded that larger science of
  information theory.

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The Nyquist theorem states

• A continuous band-limited signal can be replaced
  by a discreet sequence of samples without loss of
  information, and the original signal can be
  reconstructed from those samples.
• The sampling frequency must be at least twice the
  highest signal frequency.
• More specifically, audio signals with frequencies
  between 0 and S/2 Hz can be exactly represented
  by S samples per second.

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The Nyquist theorem states

• If there are at least two samples for a given
  frequency, that frequency can then be reproduced,
  since there is a sample for the positive and the
  negative portion of each cycle. The reproduction
  process can then recreate both sides of the cycle
  and, therefore, the reconstruct the given
  frequency.
• The highest frequency S/2 that can be reproduced
  for a given sampling frequency S is known as the
  Nyquist frequency
• If a frequency is sampled at less than two times
  per cycle, a different frequency will be
  reconstructed.

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Sampling Intervals (Rate)

• The sampling rate is the rate at which samples
  are taken from an analogue waveform during the
  analogue-to-digital conversion process. In order
  to prevent aliasing, the sampling rate should be
  at least twice the bandwidth of the signal being
  sampled. In communications systems, such as
  satellite communications, this means that the
  minimum sampling rate is limited by the bandwidth
  of the source signal.
• http//satellite.argospress.com/ysamplrat.htm

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Sampling Frequency (Period)

• When an analog signal is digitized at an
  inadequate sampling frequency (in effect, when
  the Sampling Frequency slider is adjusted to
  values exceeding a few pixels per sample), events
  are missed and a phenomenon known as aliasing
  develops. Aliasing can result not only in the
  loss of important high-spatial-frequency
  information but also in the introduction of
  spurious lower-frequency data
• http//micro.magnet.fsu.edu/primer/java/digitalima
  ging/processing/samplefrequency/

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Converting

• Converting between sampling rate and sampling
  period
• - fs 1/Ts
• - Ts 1/fs

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Oversampling vs. Undersampling

• Undersampling
• The information in the signal is aliased into a
  new form, lose the information in the original
  signal.
• Oversampling
• Sampling at a rate higher than the minimum rate
  required .

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Chapter 12

• Conversion Process
• Quantization

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The Conversion Process

• The process of converting analog signals into
  digital representations involves creating a set
  of discreet values that correspond to the
  original signal. Amplitude measurements are taken
  at specific time intervals, and those
  measurements are converted into digital data.

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The Conversion Process

• The process is like placing the signal onto a
  grid. The x-axis is time and the y-axis is
  amplitude.
• The process of taking measurements at specific
  time intervals is called sampling, and the
  process of assigning a value to those
  measurements is called quantization.

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Quantization

• After sampling, the modulated pulse chain is
  quantized. Quantization is the process of
  assigning a discreet value to each pulse.
• The pulse is measured against a scale of discreet
  quantities. Its amplitude is assigned to the
  closest quantity.
• The value is then given a binary number that can
  be stored or processed. The most common code for
  such storage and processing is Pulse Code
  Modulation (PCM).
• The assignment of the value is an approximation -
  the measurement is "rounded off" to the closest
  predefined quantity. The greater the number of
  quantities, the closer the measurements will be.
• The number of steps available in a binary system
  is equal to 2n, where an is the number of bits in
  the word used to represent the sample.

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Analog-to-Digital Conversion (ADC)

• This is a sample of the large number of
  analog-to-digital conversion methods. The basic
  principle of operation is to use the comparator
  principle to determine whether or not to turn on
  a particular bit of the binary number output. It
  is typical for an ADC to use a digital-to-analog
  converter (DAC) to determine one of the inputs to
  the comparator.
• http//hyperphysics.phy-astr.gsu.edu/hbase/electro
  nic/adc.html

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Digital-to-Analog Conversion (DAC)

• When data is in binary form, the 0's and 1's may
  be of several forms such as the TTL form where
  the logic zero may be a value up to 0.8 volts and
  the 1 may be a voltage from 2 to 5 volts. The
  data can be converted to clean digital form using
  gates which are designed to be on or off
  depending on the value of the incoming signal.
  Data in clean binary digital form can be
  converted to an analog form by using a summing
  amplifier. For example, a simple 4-bit D/A
  converter can be made with a four-input summing
  amplifier. More practical is the R-2R Network
  DAC.
• http//hyperphysics.phy-astr.gsu.edu/hbase/electro
  nic/dac.htmlc1

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Quantization Noise

• With complex audio, like music, the error is
  spread over a larger amplitude and dynamic
  spectrum, and manifests more as noise.
• Unlike analog noise, quantization noise only
  exists when signal is present, since it is the
  result of quantization error.

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Quantization Error

• There is error involved with quantization, due to
  the approximation of the level being quantized.
• Error size will be a maximum of /- half the
  amplitude of one quantizing interval. More bits
  make smaller intervals, and error is smaller as
  well.
• Since the quantizing interval is constant, the
  maximum error is also constant.
• As the signal level gets smaller, the
  signal-to-error ratio gets larger. Therefore, the
  error is more apparent with lower levels of
  audio.

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Chapter 13

• Analog Phone System
• Digital Telephone System
• Cellular System
• Satellite Telephone

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Analog vs Digital Phone Systems

• Analog is the process of taking an audio or video
  signal (in most cases, the human voice) and
  translating it into electronic pulses.
• Digital on the other hand is breaking the signal
  into a binary format where the audio or video
  data is represented by a series of "1"s and "0"s.
  
• Simple enough when it's the deviceanalog or
  digital phone, fax, modem, or likewisethat does
  all the converting for you.

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Terms to Know

• PSTN
• Public Switched Telephone Network
• AMPS
• Only system in US until 1977
• Uses 800 MHz
• GSM
• Group Special Mobile
• Originally a European standard
• Redefined to represent Global System for Mobile
  Communication
• AMPS GSM differ so that both systems may
  operate simultaneously

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Continued

• FDMA
• Frequency division multiple access
• TDMA
• Time division multiple access

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Analog Phone System

• POTS ( Plain Old Telephone System)
• Components
• Microphone
• Receiver
• Transmission System
• Switching System
• Avayas Analog Phone System Features
• Single-Line Analog Phone
• Handset Volume Control
• Ringer Volume Control
• Tone Dialing
• PBX Message Waiting
• Redial
• Flash
• Set Hold (with LED)
• Data Jack

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Digital Telephone Systems
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Satellite Telephones
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Cellular Systems

• 3rd Generation of Cellular Systems

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Next Class

• Please Read
• Chapters 14 -15
• Completed Homework Assignment 5