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Multimedia CommunicationMultimedia Communication

Introduction to Sound

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Sound and digital audioSound and digital audio Sound is comprise of

Spoken word Voice Music Noise

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Sound and digital audioSound and digital audio Sound is based on

Energy Transport medium

Transmitter

Energy Pressure

Transport media

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SoundSound Produced by the vibration of matter. During the vibration, pressure variations are created in the air

surrounding it. The pattern of the oscillation is called a waveform. The waveform repeats the same shape at regular intervals and

each portion is called a period. Sound relies on energy and a transport medium (which is the

air we breathe). Sound is created when changes in air pressure are detected by

the ear drum. Two important characteristics are frequency and amplitude.

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FrequencyFrequency Rate of sound The reciprocal value of the period. Measured in cycles per second Represents the number of periods in a second and is

measured in hertz (Hz) or cycles per second (cps). A convenient abbreviation, kHz is used to indicate

thousands of oscillations per second. (1 kHz = 1000 Hz) Higher the frequency, clearer and shaper the sound Human hearing capability: 50Hz to 20,000Hz

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AmplitudeAmplitude The measure of displacement of the air pressure

wave from its mean or quiescent state. The amplitude could also be described as the

height of a given waveform. Power or intensity of the sound Louder the sound, larger in amplitude Stronger in energy

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Pitch - LoudnessPitch - Loudness Frequency and pitch are related. Pitch is the subjective quality our brains perceive

from frequency. Greater amplitude results when objects are set in

motion with greater force. We derive the relative perception of loudness from

the absolute amplitude of a given sound wave.

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Computer Representation of SoundComputer Representation of Sound The continuous curve of a sound waveform is not

directly represented in a computer. A computer measures the amplitude of a

waveform at regular time intervals to produce a series of discrete numbers.

Each of these measurements is a sample.

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Recording SoundRecording Sound To effectively record sound

Twice the frequency that sound can be heard is needed = 40,000Hz

The higher sampling rate, the better quality At lesser rate, subtle change in pitch and tone will

be lost Effective sampling rate

Human voice: 5,000 to 15,000Hz Music: 40,000 Hz

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Voice and musicVoice and music Consider following three classes of sounds

Voice Talking, not singing

Music Human singing or musical instruments

Sound effect Voice or music More often created by natural events

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Sample RateSample Rate The rate at which a continuous waveform is

sampled is called the sampling rate. Like frequencies, sampling rates are also

measured in Hz. CD quality : 44,100 Hz Radio quality : 22,050 Hz Telephone quality : 11,025 Hz

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Audio Production - Audio Production - Conveying RealismConveying Realism Proximity: As you get closer to the source of a

sound the volume should increase. Environment: Sound should be realistic using

ambience and expected acoustics. Enhanced sound effects: Often sound effects are

required to create a realistic feel in a virtual world.

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Audio Production - Audio Production - Conveying meaningConveying meaning Emotion : dramatic music. Time : Sovereign Hill. Geographic location : didgeridoo, drums. Association : Alarm, Harvey Norman. Signature and continuity : Jaws, 007. Memorability : Star Wars, Dr Who.

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Editing Digital AudioEditing Digital Audio Trimming : Wave Edit deletes silence. Splicing and Assembly : Cut and paste. Fade In/Out : Volume slowly fades down. Noise Reduction : Dolby Digital Noise Reduction. Signal Processing : Signal processing can allow

you to play sounds backwards and insert sound effects.

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Real TimeReal Time A real time process is a process which delivers the

results of the process in a given time span. Examples of real time applications may include

missile guidance systems, nuclear reactor protocols and auto pilot technology.

Multimedia often has temporal constraints that require real time scheduling.

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Real Time Audio (.ram)Real Time Audio (.ram) Using a fast down load process, users with

standard modems can access near CD quality audio and high-quality video normally only available over faster connections.

The buffer decreases sound and image loss on poor Internet connections, improving audio and video quality.

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Windows WAVWindows WAV Created by Microsoft and IBM. It specifies an arbitrary sampling rate, number of

channels and sample size. It has a plethora of different compression formats. It is a standard used on most Windows machines. Uses application specific blocks.

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Sound Blaster VOCSound Blaster VOC This is the format used by the Creative Voice

SoundBlaster hardware used in IBM-compatible computers

It specifies sampling rate as a multiple of an internal clock and is not as flexible as the other general formats.

Portions of silence can be added to a sound file.

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AuAu Sun Audio (.AU) and NeXT. This format became very popular because it is

closely associated with the Unix operating system. Unix is used in most universities as a teaching and

research tool.

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AIFFAIFF AIFF stands for Audio Interchange File Format

and was developed by Apple for storage of sounds in the data fork.

The Macintosh OS includes support for playing and creating AIFF files.

It allows the specification of sampling rates, sample size, number of channels, and application-specific format chunks.

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MPEG AudioMPEG Audio MPEG stands for the "Moving Picture Experts

Group”. MPEG compression is a lossy algorithm which

can achieve high rates of compression without noticeable decreases in quality.

Generally considered a Macintosh based sound format.

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MP3MP3 MP3 is the file extension for MPEG, audio layer 3. Layer 3 uses perceptual audio coding and psycho acoustic

compression to remove all superfluous information. The result in real terms is layer 3 shrinks the original

sound data from a CD by a factor of 12 without sacrificing sound quality.

Because MP3 files are small, they can easily be transferred across the Internet.

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QuickTimeQuickTime This is the Apple standard for time-based

multimedia files. Versions 1.x support moving pictures, sound and later versions support text.

This standard is often associated with Macintosh computers.

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MIDIMIDI Musical Instrument Digital Interface is primarily a

standard for communication between musical instruments.

It stores information about which notes were played in a time-line format.

Often used in conjunction with synthesisers to reproduce musical arrangements.

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AudioAudio

Digitizing sound

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Digitizing soundDigitizing sound Characteristics for recording sound

Amplitude Sound channels (monaural and stereo) Frequency rate

Standard frequency rates for multimedia computer 11.025 KHz (kilohertz = 11.025 x 103 Hz) 22.5 kHz 44.1 kHz

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Digitizing soundDigitizing sound Other frequency sampling rates

Especially lower rates 5 KHz and 8 KHz For network applications Sacrifice some quality Significantly reduce amount of data

Amplitude measurement 8 bits (one byte) for 256 levels 16bits for 65,536 levels

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Digitizing soundDigitizing sound Calculating audio data size

Characteristics Amplitude Sound channels Frequency rates Time

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Digitizing soundDigitizing sound File size calculation

b = bytes (1 for 8 bits, 2 for 16 bits) Amplitude

c = number of channels (mono – 1 channel, stereo = 2 channels)

Sound channels s = sampling rate in Hertz (cycle per second)

Frequency rates t = time (seconds) File size = b * c * s * t

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Digitizing soundDigitizing sound Example: file size for 1 minutes, 44.1 KHz, 16 bit,

stereo sound c = 2 (2 channels for stereo) s = 44,100 (frequency) t = 60 (time) b = 2 (2 bytes for 16 bits)

File size = b * c * s * t =2 * 44,100 * 60 * 2 =10,584,000 bytes

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Digitizing soundDigitizing sound Digital sound characteristics

Frequency Data size Channel Size Comment

44.1 KHz 16 bit Stereo 10.5 MB CD-quality

44.1 KHz 16 bit Mono 5.25 MB Voice or monaural, high quality

44.1 KHz 8 bit Stereo 5.25 MB Best quality for playback on low-end PC

44.1 KHz 8 bit Mono 2.6 MB Good trade-off for voice quality

22.05 KHz 16 bit Stereo 5.25 MB Decent quality

22.05 KHz 16 bit Mono 2.5 MB Good for voice

22.05 KHz 8 bit Stereo 2.6 MB Acceptable for good quality playback

22.05 KHz 8 bit Mono 1.3 MB Thinner sound, TV quality

11 KHz 8 bit Stereo 1.3 MB Lowest quality for music audio

11 KHz 8 bit Mono 1.3 MB Lowest reasonable quality for voice

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AudioAudio

Digital audio systems

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Digital audio systemsDigital audio systems The circuitry for digital audio system is

responsible for Convert the audio signal from microphone or other

source into a digital signal Transferring digital data to computer’s storage media Converting digital data from storage into analog

audio signal, if necessary, amplify for playback

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Digital audio systemsDigital audio systems Digital audio hardware configuration

Audio playeror recorder

Microphone

AudioSpeaker

Digital Audio Circuitry

Data storage

Digital audio file

Analog to digitalconverter

Digital to analogconverter

Amplifier

Line in

Microphone

Line out

Speaker

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Digital audio systemsDigital audio systems Digital audio software support

Playback Translate the audio into sound

Record Capture audio data from external analog source

Stop Halt playing

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Digital audio systemsDigital audio systems Fast forward

Move to a later point in the data stream Reverse

Move to a prior point in the data stream Rewind

Move to the beginning of the data stream

Note many of these functions are similar to controls found on standard

commercial audio tape player

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Digital audio systemsDigital audio systems Sequential access and random access

Sequential access Type must be physically advanced or rewound E.g. tape player

Random access Able to jump to any point in the storage medium and access

data E.g. CD-ROM