chapter 6

Chapter 6Digital Audio Technology

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Objectives

• Understand the physical and mathematical basis of sound waves

• Recognize the amplitude, frequency, and phase properties of sound

• Apply the three-step process for audio digitization • Understand and apply the Nyquist Sampling

Theorem

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Objectives (continued)

• Recognize the need for digital audio compression• Explain how music files can be compressed without

significant sound quality degradation • Understand the difference between lossy and lossless

compression techniques• Become familiar with popular digital audio formats

like MP3, WMA, WAV, AAC, and AIFF

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Recording Sound

• The phonograph, invented by Thomas Edison in the late 1800s, was the first device created for sound recording and playback

• The tin-coated cylinders of the early phonograph were ultimately replaced by vinyl records

• Another early analog recording technique from the late 1800s relied on magnetism and electricity

• The magnetic recording approach was adopted by the music industry, and magnetic tapes replaced metallic wires as a recording medium

• This type of recording technique was later used to store music on audiocassettes, but its popularity diminished in the wake of CDs, flash drives, and alternative storage media

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Recording Sound (continued)

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Creating Sound

• Sound is caused by physical disturbances of air molecules

• This transfer of energy among molecules creates a mechanical wave of energy called a sound wave, which propagates away from the source of the disturbance

• The creation of sound waves is usually compared to the formation of water waves

Information Technology in Theory

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Creating Sound (continued)

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Converting Between Sound and Electricity

• To digitize sound, electrical systems must convert mechanical sound waves into electrical sound waves—in other words, into an electrical audio signal

• Microphones capture sound waves and convert them into an electrical form, while speakers convert the electrical signal back into sound waves

• The electrical fluctuations correspond to the pressure fluctuations of the sound wave


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Converting Between Sound and Electricity (continued)

• This electrical signal can then be converted into a stream of bits by passing through an audio digitizer

• Such digitizers are found in sound cards embedded within computers and in chips within Cellular phones

• To understand the process of audio digitization, you must understand some additional properties of sound


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Pure and Complex Sounds

• Sound may be classified as either pure sound or complex sound

• Almost all the sounds we hear, from human speech to music to a barking dog, are classified as complex sounds

• One example of a nearly pure sound is the tone produced by a tuning fork

• A pure sound is a signal that varies in a sinusoidal manner


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Pure and Complex Sounds (continued)

• By contrast, a complex sound captured by a microphone fluctuates in a nonsinusoidal and irregular manner

• Although most sound is classified as complex, we will first discuss properties of pure sounds and then address complex sound properties


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Pure and Complex Sounds (continued)


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Amplitude, Frequency, and Phase

• Amplitude (A) is the magnitude of the signal at a given instant in time (t)

• Period (T) is the time a wave requires to complete a single cycle; it is measured in seconds (s)


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Amplitude, Frequency, and Phase (continued)

• Frequency (f), measured in hertz (Hz), is the number of cycles a wave completes in one second (s)

• Phase difference describes the alignment of two waves along the time axis and may be measured in degrees


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• kilohertz (kHz) 103 = 1000 Hz (thousand)

• megahertz (MHz) 106 = 1,000,000 Hz (million)

• gigahertz (GHz) 109 = 1,000,000,000 Hz (billion)

• milliseconds (ms) 10-3 = 0.001 seconds (1/1000 seconds)

• microseconds (μs) 10-6 = 0.000001 seconds (1/1,000,000 seconds)

• nanoseconds (ns) 10-9 = 0.000000001 seconds (1/1,000,000,000 seconds)


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Frequency Composition of Sound

• Pure sounds, in simple terms, can be considered the basic components that make up complex sounds

• Each pure sound component can differ in terms of frequency, amplitude ranges, and phase differences

• A spectrum analyzer separates frequency components and displays this information on a graph called a frequency spectrum, with vertical spikes indicating the frequency components


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Frequency Composition of Sound (continued)


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Frequency Composition of Sound (continued)


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Digitizing Sound

• The digitization process takes place in circuits called analog-to-digital converters (ADCs)

• Digital cellular phones have ADCs that digitize the audio signal generated by their microphones

• Sound cards and cellular phones contain digital-to-analog converters (DACs)—circuitry that converts streams of ones and zeros into a corresponding electrical audio signal


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Digitizing Sound (continued)


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Three-Step Process of Digitization

• Sampling• Quantizing• Encoding


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Three-Step Process of Digitization (continued)


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Sampling


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Quantizing


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Quantizing (continued)

• How do we calculate the available voltage values? Two parameters are required:– Audio signal dynamic range– Number of bits we are willing to use per sample

• 2Number of bits = Number of voltages that can be represented


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Encoding


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Nyquist Sampling Theorem

• How do you know how often to take samples so that they adequately represent the original signal?


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Nyquist Sampling Theorem (continued)

• The minimum number of samples per second (the sampling frequency, or fs) required to perfectly reconstruct the analog signal should equal at least twice the value of the difference between the signal’s highest frequency component (fmax) and lowest frequency component (fmin)

• The theorem is represented by the following equations:– fs ≥ 2(fmax-fmin)

– fs ≥ 2BInformation Technology in Theory

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Standard Sampling Rates

• In the public switched telephone network and cellular systems, analog voice signals are sampled at a rate of 8000 Hz, or 8000 samples per second

• The standard is to assign 8 bits per sample• CD quality music is sampled at a rate of 44.1 kHz• The standard is to assign 16 bits per sample


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Quantization Error

• Quantization error is the difference between the actual value of the sample and the value to which the sample is rounded off

• It can be reduced by assigning more bits per sample during the ADC process


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Digital-to-Analog Conversion of Audio

• The recovery phase takes place in the DAC, which is present in any device associated with digital audio, such as an iPod, CD or DVD player, Play Station Personal (PSP) device, or a cellular phone

• If a signal is sampled at a rate that is lower than the Nyquist rate during digitization, the reconstructed signal is said to undergo aliasing into a new form


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Digital-to-Analog Conversion of Audio (continued)


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Digital Audio Compression

• Compression significantly reduces the size of digital audio files using techniques that are transparent to users

• Without compression, users would quickly run out of space on their hard disks and flash drives and be frustrated by time-consuming file downloads


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Audio Compression Requirements


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Enablers of Compression

• The limitations of human abilities, such as hearing, enable the compression of audio information

• Compression techniques fall into two categories: lossy compression and lossless compression


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Enablers of Compression (continued)

• Digital audio formats compress audio by actually eliminating some information, but not enough to be detectable by human hearing

• The process of compressing information in consideration of the limitations of human senses is called perceptual coding


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Simultaneous Masking


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Popular Digital Audio Formats

• Moving Picture Experts Group Audio Layer-3 (MP3)• Advanced Audio Coding (AAC) • Windows Media Audio (WMA) • Audio Interchange File Format (AIFF)


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Example of Digital Audio StorageProblem—Confirm that 20,000 compressed songs can be stored on an iPod that has a hard-disk capacity of 80 GB. Assume an average of 4 minutes per song and that the iPod encodes the songs using 128-Kbps AAC formatting.

• 80 GB = 80 × 8 × 230= 687,194,767,360 bits

• 687,194,767,360 bits/128,000 bits per second = 5,368,709.12 seconds

• 5,368,709.12 seconds/60 seconds per minute = 89,478.4853 minutes of digital music

• 89,478.4853 minutes/4 minutes per song = 22,369.621 songs

• This number confirms the iPod’s advertised rate of accommodating 20,000 songs


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Summary• Sound waves are created by the mechanical disturbance of air

molecules• Sound waves can be captured electrically by a microphone as an

analog audio signal, which can be digitized using an analog-to-digital converter

• Audio digitization is a three-step process of sampling, quantization, and encoding; there is an important trade-off between the accuracy of this process and its associated cost

• Sound is categorized as complex or pure• Complex sounds comprise frequency components that vary in

terms of their amplitude ranges, frequencies, and phase differences


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Summary (continued)• The minimum sampling frequency for digitizing

audio signals is at least twice the value of the highest frequency components; aliasing occurs with a sampling rate that is less than the Nyquist rate

• Digital audio files consume considerable storage space and bandwidth, but they can be compressed by exploiting the limitations of human hearing and the built-in redundancy within information

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Summary (continued)• The two general approaches to compressing

information are lossless compression and lossy compression; some compression techniques use a combination of these two approaches

• In lossless compression, none of the original information is lost

• In lossy compression, some information is discarded, such as frequencies

• Some common digital audio file formats include MP3, AAC, WMA, WAV, and AIFF

chapter 6

Documents

sound cards

electrical sound wavesin

mechanical sound waves

tuning forka pure sound

digital audio technology

electrical systems

electrical form

properties of pure sounds