Incorporate Audio into Multimedia PresentationsICPMM44CAICPMM44CAICPMM44CAICPMM44CAICPMM44CAPart 1 - Audio TheoryPart 1 - ...
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Audio theory

  1. 1. Incorporate Audio into Multimedia PresentationsICPMM44CAICPMM44CAICPMM44CAICPMM44CAICPMM44CAPart 1 - Audio TheoryPart 1 - Audio TheoryPart 1 - Audio TheoryPart 1 - Audio TheoryPart 1 - Audio TheoryEducation and Training
  2. 2. PAGE 2Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYIncorporate Audio into Multimedia PresentationsWelcome to the learning resource for Incorporate Audio into MultimediaPresentations from QANTM Australia CMC Pty Ltd. This learning resourcecovers or exceeds the competency in the following qualification/s:CUF30601 Certificate III in MultimediaCUF40801 Certificate IV in MultimediaICA40499 Certificate IV in Information Technology (Multimedia)CUF50401 Diploma of Screen (Animation)AssessmentThose students enrolled in this unit need to complete the assessment itemsas per the Assessment Criteria Sheet.Activities (non-assessable)There are no activities for this unit.
  3. 3. PAGE 3Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYUnit DescriptionThis unit describes the competency required to edit, combine and incorporateaudio into multimedia presentations.Suggested HardwareTo successfully complete this course students will need access to a personalcomputer with soundcard, keyboard, mouse, microphone, headphones and/orspeakers and an internet connection.Suggested Software• Sound Forge 6• Internet Explorer Version 4 or higher or other suitable browser• Microsoft Word or other suitable word processor
  4. 4. PAGE 4Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYIncorporate Audio intoIncorporate Audio intoIncorporate Audio intoIncorporate Audio intoIncorporate Audio intoMultimedia PresentationsMultimedia PresentationsMultimedia PresentationsMultimedia PresentationsMultimedia PresentationsKEY FEATURESKEY FEATURESKEY FEATURESKEY FEATURESKEY FEATURESThis unit is designed to provide you with the knowledge in order tocorrectly identify and understand the terms and features associated withdigital audio. You will also learn how to create, edit and manipulate digitalaudio which is an important aspect in multimedia presentations.• Introduction to the structure and the features of analog anddigital audio• Audio file formats and compression• Audio hardware• Types of microphones and recording tips• How to use Audio Encoder and Decoder ConvertersOVERVIEW OFOVERVIEW OFOVERVIEW OFOVERVIEW OFOVERVIEW OFTHE UNITTHE UNITTHE UNITTHE UNITTHE UNITI C P M MI C P M MI C P M MI C P M MI C P M M 44C44C44C44C44CAAAAAU N I TU N I TU N I TU N I TU N I TIdentify and describeformats of digitala u d i oUse digital audios o f t w a r eEdit digital audioConstruct a digitalaudio track
  5. 5. PAGE 5Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYMenuPart 1 - Audio TheoryIntroduction to Sound .............................................................................. 8Analog Audio .......................................................................................... 8Digitising Audio...................................................................................... 10Sampling and Sample Rate ..................................................................... 10Quantisation and Bit Depth ..................................................................... 11Bit-Rates ............................................................................................. 12The Nyquist Theorem ............................................................................ 13Digital Audio Formats............................................................................. 16Common Audio File Formats .................................................................. 17Streaming and Non-Streaming Audio ....................................................... 20Shockwave Audio ................................................................................. 21Audio Compression ............................................................................... 22Hardware Considerations ....................................................................... 23How Microphones Work ......................................................................... 27Dynamic and Condenser Microphones ...................................................... 27Categories of Microphones ..................................................................... 29Tips When Using Microphones ................................................................. 31Part 2 - Sound ForgeIntroduction to Sound Forge 6 .................................................................. 5Starting Sound Forge from a Shortcut ....................................................... 5
  6. 6. PAGE 6Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYThe Design Window of Sound Forge 6.....................................................6The Work Area........................................................................................ 6The Toolbars........................................................................................... 6ACID Loop Creation Toolbar...................................................................... 7Status or Selection Toolbar ....................................................................... 8Status Bar.............................................................................................. 9Types of Microphones ............................................................................ 10Tips on How to Use a Microphone for Voice Recording ............................... 11Use Digital Audio Software .................................................................. 12How to Record Your Own Voice............................................................... 12DC Offset ............................................................................................. 15Applying DC Offset ................................................................................ 15Editing Audio ....................................................................................... 16How to Copy/Paste a Waveform ............................................................. 16How to Cut a Waveform ........................................................................ 17How to Delete a Waveform .................................................................... 17How to Convert Mono to Stereo ............................................................. 18How to Convert Stereo to Mono ............................................................. 18Converting Mono to Stereo .................................................................... 19Converting Stereo to Mono .................................................................... 20How to Copy and Paste Waveform into Only One Channel .......................... 21How to Copy Waveform From One File to Another .................................... 22How to Copy and Paste Special............................................................... 24Paste Special in a Mono Wave File ........................................................... 24How to Insert Silence ............................................................................ 25
  7. 7. PAGE 7Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYHow to Reverse Sound .......................................................................... 26How to Use Normalise ........................................................................... 26How to Fade Sound ............................................................................... 28How to Use Graphic EQ ......................................................................... 30How to Resample Files........................................................................... 31How to Convert to 8-Bit ........................................................................ 32How to Apply Effects to Waveform .......................................................... 33Pitch Bend Effect ................................................................................... 34How to Use Encoder and Decoder Audio Converters .................................. 35Convert Track Files to WAV Files Using Soundforge .................................... 35Adding Effects ....................................................................................... 37Convert WAV Files to MP3 Files ............................................................... 37Convert MP3 Files to WAV Files ............................................................... 38Video and Audio in Sound Forge .............................................................. 40How to Attach Audio to a Video Track ...................................................... 41
  8. 8. PAGE 8Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYIntroduction to SoundBefore we can begin to understand and analyse the specific features of digitalaudio, we need to consider some fundamental questions; what is sound andhow does the human ear perceive variations in pitch, volume and scale?In order for sound to exist, there needs to be a medium, like air or water or asolid object, that acts as the transmitter, a motion or disturbance in thismedium that creates waves and some kind of receiver that detects thesevariations (see Figure 1). Without these three essential parts, sound cannotexist. Therefore, for humans to hear sound, pressure waves or changes in airpressure must be generated by some physical vibrating object, such asmusical instruments or vocal chords. Then these changes are perceived via adiaphragm, the eardrum, which in turn converts these pressure waves intoelectrical signals, which are interpreted by the brain. This type of soundproduction is referred to as analog audio.Analog AudioAn Analog Audio signal can be graphically represented as a waveform (seeFigure 2). A waveform is made up of peaks and troughs that are a visualrepresentation of wavelength, period, amplitude (volume level) andfrequency (pitch).The horizontal distance between two successive points on the wave isreferred to as the wavelength and is the length of one cycle of a wave. Acycle is the distance between two peaks. The period of the wave refers to theamount of time that it takes for a wave to travel one wavelength (see Figure6 on the next page).Amplitude is half the distance from the highest to the lowest point in a wave.If the distance is large the volume level is comparatively loud and alternately, ifthe wave is small, then the volume level is low.The number of cycles per second is referred to as frequency (see Figure 3).http://www.atpm.com/6.02/digitalaudio.shtmlFigure 1Figure 2: Graphic representation of the waveformFigure 3:The greater theamplitude thehigher the volume.1 secondTroughcyclePeakAmplitude1 second!!Frequency!!Frequency!!
  9. 9. PAGE 9Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYFrequency is measured in hertz (Hz), a unit of measure named after HeinrichHertz, a German physicist and indicates the number of cycles per second thatpass a specified location in the waveform.Frequency directly relates to the sound’s pitch. The pitch or key is how thebrain interprets the frequency of the sound created and the higher thefrequency, or the faster the sound vibrations occur, the higher the pitch. If thevibrations are slower, then the frequency is low.Analog signals are continuous and flexible; able to change at varying ratesand size which means that analog audio is relatively unconstrained and unlim-ited. The flexible nature of analog sound, though seemingly positive, is in factits biggest disadvantage, as it is therefore more susceptible to the effects ofextreme changes in audio that cause degradation of sound like distortion andnoise.Originally sourced from:http://www.library.thinkquest.org/19537/Physics3.htmlFigure 5Figure 4Figure 6Figures 4 , 5 and 6 provide a visual guide to the structure of a waveform.
  10. 10. PAGE 10Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYDigitising AudioComputers cannot understand analog information. In order for an analogsignal to be understood by a digital device (such as a computer CD or DVDplayer), it first needs to be digitized or converted into a digital signal by adevice called an Analog to Digital Converter or ADC (see Figure 7). Then forthe human ear to be able to hear a digital signal, it needs to be converted backto an analog signal. This is achieved using a Digital to Analog Converter orDAC.The conversion of an analog signal to a digital one requires two separateprocesses: Sampling and Quantisation.Sampling and Sample RateThe analog signal is sampled, or measured and assigned a numerical value,which the computer can understand and store.The number of times per second that the computer samples the analog signalis called its Sample Rate or Sampling Frequency. While the basic unit used tomeasure frequency or cycles per second is hertz, when sampling audio it isgenerally measured in thousands of cycles per second or kilohertz (kHz).An audio CD, for example, generally has a sampling rate of 44.1 kHz that isforty four thousand one hundred Hertz (or samples per second), while the AMradio has a sample rate of 11.025 kHz or eleven thousand and twenty fiveHertz. The more samples taken, the higher the quality of the digital audio signalproduced.Play the two provided examples, 44100Hz.wavand 11025Hz.wav to hear the difference in soundquality between the two different sample rates.Low sampling rates, below 40 kHz, can result in a static distortion caused bydata loss. This is referred to as Aliasing (see Figure 8). Aliasing can causedigitally reconstructed sound to playback poorly. To avoid the aliasing effectFigure 7: Converting to DigitalAnaloguesoundAnalogue toDigitalconverterDigitalreproduction!SamplesFigure 8: AliasingThis sound wave is 100th of a second long.It has been sampled at the very low rate of 160hz.The red dots are the samples taken.The green line indicates the the sound wave thesamples would generate.As you can see the aliasing is severe in this case.The new sound file would not sound like the original.44100Hz.wav 11025Hz.wavSound waveContinious Discontinious
  11. 11. PAGE 11Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYsampling needs to occur at a high enough rate to ensure that the soundsfidelity is maintained, or anti-aliasing needs to be applied when the audio isbeing sampled. An anti-alias filter can ensure that nothing above half thedesired sampling rate can enter the digital stream ie any frequencies above thedesired frequency are blocked. Be aware that using anti-alias filters may, inturn, introduce further unwanted noise.Analog audio is a continuous sound wave that develops and changes overtime. After it is converted to digital audio it is discontinuous, as it is nowmade up of thousands of samples per second.Quantisation and Bit DepthOnce an analog signal has been sampled, it is then assigned a numeric value ina process called Quantisation. The number of bits used per sample definesthe available number of values.Bit is short for binary digit. Computers are based on a binary numberingsystem that uses two numbers; 0 and 1. This differs from the more familiardecimal numbering system that uses 10 numbers.This two number system means each additional bit doubles the number ofvalues available - a 1-bit sample has 2 possible values; 0 and 1 and a 2-bitsample has 4 possible values; 0 and 0, 1 and 0, 0 and 1, 1 and 1 and so on(See Figure 9).These binary values are defined as its Resolution or Bit Depth. This methodof measurement is used throughout digital technologies. You may already befamiliar with bit depth in digital graphics, where a 1-bit image is black andwhite, a web safe or greyscale image is 8-bit and an RGB image has one byteor 8-bits allocated for each of the three colours and is 24-bits in total.Typically, audio recordings have a bit depth of either 8 or 16-bit and even 24-bit on some systems. An 8-bit sample will allow 256 values, whereas a 16-bitsample will allow 65 536 values. The greater the bit depth, the more accuratethe sound reproduction and the better the sound quality.Bit Depth Possible values12345678248163264128256Adding one bit doubles thenumber of values available.Figure 9: Bit valuesBit: “A fundamental unit of infor-mation having justtwo possible values,as either of the binary digits0 or 1.”Source: The American Heritage®Dictionary of the English Language,Fourth Edition (2000), by HoughtonMifflin Company
  12. 12. PAGE 12Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYAn audio Dynamic Range is the difference between thelowest and highest points of a wave and is measured indecibels (dB). The larger the dynamic range the greaterthe risk of distorted sound. Audio files with a largedymamic range tend to require greater bit depth tomaintain sound qualityAn 8-bit sample, with 256 values can recreate a dynamicrange of 48 dB (decibels), which is equivalent to AM radio,whereas a 16-bit sample can recreate a dynamic range of96 dB, which is the equivalent of CD audio quality.The dynamic range of the average human ear isapproximately 0 to 96 dB (120dB is the pain threshold),so it is no coincidence that the standard bit depth for CDquality audio is 16-bit.Bit-RatesThe number of bits used per second to represent anaudio recording is defined as Bit-Rate. In digital audiobit-rates are defined in thousands of bits per second(kbps).The bit-rate is directly associated with a digital audio file’ssize and sound quality. Lower bit-rates produce smaller filesizes but inferior sound quality. Higher bit-rates producelarger files but are of a better sound quality. Anuncompressed audio tracks bit-rate and approximate filesize can be calculated using the following formulas:When calculating bit-rate it is important to remember that:• 8 bits = 1 byte• 1024 bytes = 1 Kb or a Kilobyte• 1024 kilobytes = 1 Mb or a Megabyte(see Figure 9 to calculate file size.)Calculating an uncompressed CD tracks bit rateCalculating its file sizeSampling Rate x Bit Depth x Number of channels = Bit Rate(in KHz)44.1 KHz x 16 bits x 2 = 1 411.2kbpsSamplingRate(in KHz)= file sizex Bit Depth x Number ofchannelsLength insecondsBits toBytesx /44.1 KHz10,584,000bytes orover 10 MB8 =/60seconds2 xx 16 bits xFigure 9The amount of 1024 is oftenrounded down to 1000 ifstrict accuracy is not required.
  13. 13. PAGE 13Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYThe Nyquist TheoremAudible Frequency refers to the range of frequencies that are detectable bythe average human ear. There is a direct correlation between the sample rateand the highest audible frequency perceived by the ear. The relationshipbetween sample rate and the highest audible frequency is referred to as theNyquist Theorem. The Nyquist Theorem, named after Harry Nyquist, a Bellengineer who worked on the speed of telegraphs in the 1920s, is a principlethat is used to determine the correct sampling rate for a sound.Essentially, the Nyquist Theorem states that a sound needs to be sampled at arate that is at least twice its highest frequency in order to maintain its fidelityor sound quality. Therefore, a sample taken at 44.1kHz will contain twice theinformation of a sample taken at 22,050 kHz. Put simply, this means that thehighest audible frequency in a digital sample will be exactly half the samplingfrequency.Originally sourced from:http://www.csunix1.lvc.edu/~snyder/2ch11.htmlFigure 10: The Nyquist Theorem rules states that awaveform must be sampled twice. The positive peak andthe negative peak must both be captured in order to get atrue picture of the waveform.Average human hearing, at best, covers a range from 20 Hz (low) to20 kHz (high), so a sample rate of 44.1 kHz should theoretically covermost audio needs. It is also the standard for CD audio, which requiresnear optimum sound quality. Therefore, the higher the sample rate,the better the quality of sound that is reproduced. However, this alsomeans that the higher the sample rate, the greater amount of audiodata produced and consequently the larger the file size. This meansthat there is a direct correlation between the sample rate, thequality of sound and the file size of the audio file.An example of how this affects the quality of digital audio is illustratedby the example provided in Figure 10. A music track that has anoptimum frequency of approximately 20 kHz, the highest audiblefrequency perceived by the average human ear, needs to be sampledat 44.1 kHz in order to maintain CD quality sound fidelity. However, ifthe same track is sampled at a rate lower than 44.1 kHz eg 30 kHz,then according to the Nyquist Theorem, the range between 15 kHzand 20 kHz will be lost and therefore the sound quality will deteriorate.The reason for sampling below the recommended rate of the Nyquisttheorem, would be where the sample rate is determined by the
  14. 14. PAGE 14Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYtransmission technology. For example, telephone wires and the bandwidthallocated to radio transmission, where low data rates and storage space areconsidered over sound quality.Sampling rates are directly linked to the desired sound quality produced,therefore, different audio types and delivery methods require differentsampling rates (see Figure 11). Many applications do not require a widefrequency range to produce an ‘acceptable’ level of sound quality. The highestaudible frequency in the human voice is approximately 10 kHz which isequivalent to a sample rate of 20 kHz. Telephone systems, however, rely onthe fact that even with the highest audible frequency of 4kHz (a sample rateof 8kHz), the human voice is perfectly understood.Sampling rates for radio broadcasts are also confined within frequencies thatsuit the required quality of the sound produced. AM radio has been broadcastsince the early 1900s and in the 1920s it was allocated to a specificfrequency. Due to the limited technology of the period, in relation to thecapabilities of radio and electronics, the frequencies for AM radio weretherefore relatively low. Edwin Armstrong developed FM radio in the 1930s.His intention was to produce high fidelity and static free broadcasts, thereforerequiring higher frequencies. Although FM radio was available earlier, it was notreally popular until the 1960s.The sampling rate used for CD is 44.1kHz or 44 100 samples per second. Thisrelates directly to the Nyquist Theorem whereby in order to produce highquality sound, the sample rate must be at least twice the maximum audiblefrequency signal. So for a CD to produce audio up to a maximum frequency of20 kHz, which is the upper limit of human hearing, then it requires a samplingrate of 40Khz. The standard sample rate for CD, however, is set at 44.1kHz.Quality Sampling RateTelephoneAM RadioFM RadioCDDAT (Digital AudioTape)8kHz11.025 kHz22.050 kHz44.1 kHz48 kHzFigure 11: Some common sampling rates
  15. 15. PAGE 15Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYDigital Audio Tape (DAT)Developed in the late 1980s the DAT is still used by some sound recordingstudios for both direct recording or data backup. They resemble small cassettetapes in appearance but they have the capacity to record up to 2 hours ofaudio. The DAT recording process is similar to cassette recording but the qual-ity of recording can be compared to CD quality or higher, with 3 possible sam-pling rates; 32kHz, 44.1kHz and 48kHz. DAT Recording is also discussed laterin the Hardware Considerations section of the notes.Stereo and MonoAudio is typically recorded in either Mono or Stereo.A stereo signal is recorded using two channels and when played throughheadphones will produce different sounds in each speaker. This allows for amore realistic sound because it mimics the way that humans hear, thereforegiving us a sense of space.Mono signals, on the other hand, have identical sounds in each speaker andthis creates a more unnatural sound - ‘flat’ sound. This is a majorconsideration when digitising audio, in that it will take twice as much space tostore a stereo signal compared to mono signal.
  16. 16. PAGE 16Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYDigital Audio FormatsAn audio file consists of two main components; a header and theaudio data. The header stores information in relation toResolution, Sampling Rate and Compression Type.Sometimes, a wrapper is also used which adds informationabout things such as license management information orstreaming capabilities (see Figure 12).Digital audio files can be found in a huge variety of file formats but basicallythese files can be divided into two main categories:1. Self–Describing2. RAWSelf-Describing formats are usually recognised by their file extension. Theextension, which is part of the file name, will refer to the type and structure ofthe audio data within the file and it instructs the user and the computer inrelation to how to deal with the sound information.RAW formats are files that are not compressed. They rely on the soundsoftware to correctly interpret the sound file by reading the data or code ofthe header component.File formats are used for different purposes and they vary in terms of file sizescreated. Therefore, when choosing an audio file format, its function andeventual context need to be considered. This is particularly important whenworking with audio files for the web.http://www.teamcombooks.com/mp3handbook/12.htmFigure 12
  17. 17. PAGE 17Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYCommon Audio File Formats1. Wave File Format (.wav)This is a Windows’ native file format for the storage of digital audio data. Dueto the popularity of Windows, it is one of the most widely supported audio fileformats on the PC. WAV files are usually coded using the PCM – Pulse CodeModulation format. PCM is a digital scheme for translating analog data. WAVfiles are uncompressed and therefore have large file sizes. It is a RAW formatthat is often used for archiving or storage. The audio data within the wave fileformat is stored in a chunk, which consists of two sub-chunks; a fat chunkthat stores the data format and a data chunk that contains the actual sampledata.The WAV format supports a variety of bit depths and sample rates as well assupporting both mono and stereo signals.2. Audio Interchange File Format (.AIFF)This is an audio file format that is a standard audio format used on Macintoshsystems, although it can be used on other platforms. Like the WAV file format,the audio data within an AIFF file format uses the Pulse Code Modulationmethod of storing data in a number of different types of chunks. This is aBinary file format that is quite flexible, as it allows for the storage of bothmono and stereo sampled sounds. It also supports a variety of bit depths,sample rates and channels of audio.3. MPEG – Encoded Audio (.MP3)MPEG audio is a standard technology that allows compression of an audio fileto between one-fifth and one-hundredth of its original size without significantloss to sound quality. The MPEG audio group includes MP2, MP3 and AAC(MPEG-2 Advanced Audio Coding).The most common, however, is MPEG 2 Layer 3, which has the file extensionMP3. MP3 compression makes it possible to transmit music and sound overthe Internet in minutes and can be downloaded and then played by an MP3Player. There are several free MP3 Players, but many are not streaming and if
  18. 18. PAGE 18Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYthey are streaming, they use different, often incompatible, methods ofachieving the playback. MP3 files can be compressed at different rates but thegreater the compression the lower the sound quality. MP3 technology uses alossy compression method, which filters out all noise that is not detectableto the human ear. This means that any ‘unnecessary’ information is deleted inthe compression process, which results in a file that is a fraction of the originalWAV file but the quality remains virtually the same. The main disadvantage ofMPEG compression in software, is that it can be a really slow process.4. Real Audio (.RA, .RM)Real Audio is a proprietary form of streaming audio (described later) for theweb from Progressive Networks’ RealAudio that uses an adaptivecompression technology that creates extremely compact files compared tomost other audio formats. The resulting bit rate can be optimised for deliveryfor various low-to-medium connection speeds. Real Audio either requires aReal Audio server or the use of metafiles, otherwise the files won’tdownload and play.Real Audio is a good choice for longer audio clip sounds because it lets youlisten to them in ‘real-time’ from your Web browser and the sound quality ofthe high bandwidth compressions is good. Real Audio players can be includedwith a web browser or can be downloaded from the web.5. MIDI – Music Instrument Digital InterfaceMIDI, or Musical Instrument Digital Interface, is not an actual audio fileformat but rather a music definition language and communications code thatcontains instructions to perform particular commands. Rather thanrepresenting musical sound directly, MIDI files transmit information about howmusic is produced. MIDI is a serial data language, composed of MIDImessages, often called events, that transmit information about pitch,volume and note duration to MIDI-Compatible sound cards andsynthesizers.New Audio File FormatsAACKeeper of the format: the MPEG group thatincludes Dolby, Fraunhofer (FhG), AT&T, Sony, andNokiaSize: Smaller than MP3Extension: *.aac, *.m4aWriter of the format: QuickTime 6x supportsAAC. Other encoders such as Real Networks arestarting to support AAC.File size: AAC files are approximately 50% smallerthan MP3 files.Sound quality: AAC files have a quality betterthan MP3.Comments: AAC files are based on MPEG 4 andhave a better compression and higher quality thanMP3. Apple wants AAC to become the industrystandard audio format.WMAKeeper of the format: MicrosoftSize: Smaller than MP3Extension: *.wmaWriter of the format: Various encoders will writethe WMA format.File size: WMA files are approximately 50%smaller than MP3 files.Sound quality: WMA files have a quality betterthan MP3.Comments: Better compression and higherquality than MP3. Microsoft wants WMA to becomethe industry standard audio format.
  19. 19. PAGE 19Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYMessages transmitted include:• Start playing (Note ON)• Stop playing (Note OFF)• Patch change (eg change to instrument #25 - nylon string guitar)• Controller change (eg change controller Volume to value from 0 to 127)It was initially developed to allow sequencers to control synthesisers. Oldersynthesisers were Monophonic, that is, they were only able to play one noteat a time. Sequencers could control those synthesisers by voltage and atrigger or gate signal that told you if a key was up or down. Contemporarysynthesisers are Polyphonic, enabling them to play many notes at once,which is more complex. A single voltage was not enough to define severalkeys so the only solution was to develop a special language; the Midi. It hasmuch smaller file sizes than other audio file formats, as it only contains playerinformation and not the actual direct sound. The positives of the MIDI are itssmall file size but the disadvantage is the lack of direct sound control.To play MIDI files you need two things:• Either a MIDI plug-in or a MIDI helper application and• A MIDI device, which can take the form of a soundcard,an external MIDI playback box or MIDI keyboard, or a software-based MIDI device, such as the set of MIDI sounds that comes withthe current version of QuickTime.These are the most common audio file formats in the current market but inthe past, computers that had sound capabilities developed their ownproprietary file formats.The following is a list of same of the current proprietary file formats:• .SFR – Sonic Foundry• .SWA – Shockwave• .SMP – Turtle Beach
  20. 20. PAGE 20Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYStreaming and Non-Streaming AudioAudio files, by their very nature, are data intensive, which can result inlarge file sizes; particularly if the audio track requires high sound qualityand needs to be more than a few seconds or minutes in length. Theseconsiderations become of paramount importance when an audio file isincorporated into a web page. Depending on the type and size of theaudio file, a user may experience a long delay between clicking on anaudio link and hearing the sound. This is because the entire audio fileneeds to be downloaded before it can be played. An audio file embededinto a webpage eg a sound effect will be dowloaded into the browser’scache. With other audio files the user will be asked where to save the fileon their hard drive. This method of downloading a complete sound fileand subsequently playing it, is referred to as Non-Streaming Audio.An audio technology called Streaming alleviates this delay in sounddelivery and allows the user to hear the sound immediately or with only aslight pause. It also prevents users from saving copies of the file to theircomputer.Streaming audio uses a buffering system whereby a buffer space in theform of a temporary file is created in RAM or Virtual Memory and theaudio data is transferred to this when the user clicks on an audio link.Within seconds, the buffer becomes full and the audio begins to play.Once this portion of information is used, more audio data is downloadedwhile the sound is playing. Audio data in the buffer is continuallyoverwritten until the file has finished playing.The smoothness of playback of the audio file is directly linked to the ratiobetween data download rate and the data rate required for playback. Ifthe audio data can be transferred as quickly as it is used, then the filewith play smoothly. Another factor that determines the quality of thestreamed sound is the user’s machine and the mode of data transfer. Thefaster the user’s modem, the fewer ‘glitches’ will occur during playback ofa streamed file (see Figure 13).Figure 13: Streaming audiohttp://www.cit.cornell.edu/atc/materials/streaming/definition.shtmlThe Principle of Streaming(A snapshot in time)TimeThe portion in the bufferThe portion you are viewingThe portion on yourhard drive at one timeThe entire streaming audio or video
  21. 21. PAGE 21Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYA 56 kb modem is the recommended minimum speed for streaming audio.However, even a fast modem processes data at a relatively slow rate and thismeans that the audio data needs to be compressed in order for it to bechannelled through the modem to be played back at an acceptable quality.Shockwave AudioOne of the leading providers of streaming audio is Macromedia’s Shockwavefor Director, which also includes an animation player. Shockwave Audio,developed by Macromedia to stream high quality audio over the Internet,uses very sophisticated mathematical analysis to compress audio so that itcan be represented by relatively few bytes of data. This much smaller datastream is sent through the user’s modem; it is then uncompressed in theuser’s computer, converted back into audio and then played back through thespeakers. Shockwave audio is scalable, which means that you can select thequality level to use for the audio playback. A high quality setting, for example,may be too data intensive to squeeze through a modem in real-time. In thiscase, ‘gaps’ may be present in the audio playback.Streaming audio, like Shockwave, may require a Plug-in Player. A Plug-in is aprogram that can be downloaded and installed on a user’s computer in orderto extend the capability of the web browser by allowing a more seamlessintegration of many different kinds of file formats into the browserenvironment. The web browser automatically recognises plug-ins and theirfunctions are integrated into the main html file.
  22. 22. PAGE 22Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYAudio CompressionCompression is the reduction in size of data in order to save space ortransmission time. Generally, compression can be divided into two maincategories: Lossy and Lossless compression. The main objective of bothof these compression techniques is to decrease file size, however, this is theonly similarity between these two compression types.Text documents can be compressed at extremely high percentages of theoriginal file size eg on average 90% but audio files can only be compressed toapproximately 25 – 55% of the original file size. Although, this compressionpercentage may not seem ideal, it is very useful when reducing audio file sizesthat need to be transferred over the internet or for archiving audio files.Lossless audio compression (eg Monkey’s Audio) is similar in concept tousing a program like WinZip to compress a document or program. Theinformation within the audio file is minimized in terms of file size, whilst stillmaintaining the fidelity of the original data. This means that the compressedfile can be decompressed and still maintain the identical data of the originalfile; with no loss to the audio quality.Lossy audio compression (eg MP3), on the other hand, does not maintainthe identical fidelity of the original audio file and in fact, does not compress allof the audio data. Lossy compression methods analyse the audio data in thefile and then discards any information that seems ‘unnecessary’ in order toreduce the file size. This discarded information is not usually discernible by thehuman ear and therefore does not alter the ‘perceived’ quality of the audio.Any compressor will achieve varied ratios of compression depending on theamount and type of information to be compressed and there are manydifferent file formats available for both Lossless and Lossy audio compression.The web is the most obvious location where audio compression becomes ofparamount importance. Speed and efficiency are the two things that the webrelies on in terms of effective data transfer from the Internet pipeline to theend user’s machine. Therefore, the smaller the file size the faster the data istransferred.
  23. 23. PAGE 23Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYThere are several ways you can reduce the size of an audio file for deliveryon the web. The first and most obvious method would be to consider the lengthof the track. There will be a significant difference for example between 1 minute ofrecorded audio, as opposed to 40 seconds (see Figure 14). The nextconsideration would be the number of channels; does the track need to be instereo or could it be converted to a mono recording. By converting the file to onlyone channel you have already effectively reduced the file to a half of its original sizeand a half of the download time.Another way to reduce the file size is to change the bit depth from a 16-bit track,for example to an 8-bit track. The final way to reduce the size of an audio file is toalter the sample rate. The key in creating digital audio files for the web is toexperiment with the various recording settings, in order to find an effective balancebetween sound quality, performance and file size (See Figure 14).Hardware Considerations1. Video Capture CardsA video capture card is used together with a computer to pass frames from thevideo to the processor and hard disk. When capturing video, ensure that allprograms not in use are closed, as video capture is one of the most system-intensive tasks that can be performed on a computer.Most capture cards include options of recording with a microphone or linelevel signal. A Microphone Level Signal is a signal, which has not been amplifiedand has a voltage of .001 (one millivolt). Not surprisingly, microphones usuallygenerate microphone level signals. A Line Level Signal is a preamplifier and has avoltage of 1.0 (one full volt) generally created by mixing decks, *Video TapeRecorders (VTR), tape players and DAT players etc. If your capture card has theoption, you will be able to decide which type of signal you are recording. Yourcapture card may have two different types of connectors. The microphone inputis usually (except when using Macintosh system microphones) a 3.5 mini jackstereo connector. The line input is usually a stereo RCA connector or sometimes three-pin XLR connector.Figure 14: File sizes for one minute of audio recorded atvarious bit rates44.1kHz22.05 11.02516-bit16-bitmono8-bit8-bitmono10.01MB5.05MB2.52MB5.05MB2.52MB1.26MB5.05MB2.52MB1.26MB2.52 MB 1.26 MB 630KBOriginally sourced from:http://www.beta.peachpit.com/ontheweb/audio/chap1.html* Video Tape RecordersVTRs are professional recording and playbackmachines which use magnetic tape rolls.
  24. 24. PAGE 24Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORY2. Metering and MonitoringYour capturing software should also allow you to see agraphic representation of sound levels – it shoulddisplay meters. There are different types of meters,which use a variety of measurements and colourcodes. Regardless of metering systems used, youshould always use the meter to ensure that theincoming sound does not exceed the recording abilitiesof the capture card. Unlike analog systems, whichdue to the electrical nature of the signal and therecording medium, allow for sounds to be recorded atlevels that clip or peak, digital systems don’t allowfor this. Digital recorders can only record levelswithin their range capabilities. If the incoming levelexceeds the maximum level, clipping (distortion) willoccur. The result of this is distortion of the digitalsound when played back.3. Sound Cards and Sound ConsiderationsA sound card is a peripheral device that attaches tothe motherboard in the computer. This enables thecomputer to input, process and deliver sound. Soundcards may be connected to a number of otherperipheral devices such as:• Headphones• Amplified speakers• An analog input source (microphone, CD player)• A digital input source (DAT, CD-ROM drive)• An analog output device (tape deck)• A digital output device (DAT, CD recordableCD-R) (see Figure 15)CD Player, Cas-sette, VCR etc.Line InMicrophoneLine OutSpeakersHead phoneJoystick/MidiAdapter PlugLine-outLine-inStereo Amp. etc.Figure 15: Back of device shown
  25. 25. PAGE 25Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYThe core of the sound card is the audio processor chip and the CODECs. Inthis context, CODEC is an acronym for COder/DECoder. The audio processormanipulates the digital sound and depending on its capabilities, is responsiblefor converting sample rates between different sound sources or adding soundeffects. Although the audio processors deal with the digital domain, at somepoint, unless you have speakers with a digital input, you will need to convertthe sound back into analog.Similarly, many of the sound sources that you want to input to your computerwill begin as analog and therefore need to be converted into digital. A soundcard therefore needs some way to convert the audio. DACs (digital to analogconverters) and ADCs (analog to digital converters) are required to convertthese audio types and many audio cards have chips that perform both ofthese functions. They are also known as CODECs due to their capability toencode analog to digital and decode digital to analog.The other factors that can influence the functionality and usability of the soundcard is the Disk Driver, along with the number and type of input and outputconnectors (see Figure 16).4. DAT RecordingDAT (Digital Audio Tape) is used for recording audio on to tape at aprofessional level of quality. A DAT drive is a digital tape recorder with rotatingheads similar to those found in a video deck (see Figure 17). Most DAT drivescan record at sample rates of 44.1 kHz, the CD audio standard and 48 kHz.Recording on DAT is fast and simple. It is as simple as choosing what youwant, setting the levels and pressing record. DAT has become the standardarchiving technology in recording environments for master recordings. Digitalinputs and outputs on professional DAT decks allow the user to transferrecordings from the DAT tape to an audio workstation for precise editing. Thecompact size and low cost of the DAT medium makes it an excellent way tocompile the recordings that are going to be used to create a CD master.http://www.tweakheadz.com/dat_recorders.htmFigure 17: DAT recorder designed for hard disk recording,editing, digital signal processingFigure 16: RCA Connectors for PC/MAC
  26. 26. PAGE 26Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORY5. Mini Disk PlayersMiniDisc was developed by Sony in the mid eighties as portable equipmentthat combine the storage qualities of CD with the recordabilty of cassettes.They are very cost effective and run on power or on re-chargeable batteries,which last for approximately 14 hours of play time.While CD-ROMs and DVDs use optical technology and floppys and hard drivesuse magnetic technology MiniDisc uses a combination of both to record data.Therefore care should be taken to protect minidisks from strong magneticfields. Just like a computer’s hard drive, the audio data is recorded in digitallyand in fragments - this is called Non-Linear recording.MiniDisc’s use sample rates of 48Khz, 44.1Khz or 36Khz. They usescompression to enable them to record the equivalent to a full sized CD on tothe 64mm disc. This compression is called ATRAC (Adaptive TransformAcoustic Coding) incorporates noise reduction and has a compression ratio of1:5. Similar to MP3 it reduces data by only encoding only frequencies audibleto the human ear6. MicrophonesComputers that have built in microphones are not usually considered to behigh-fidelity devices. When dealing with audio production, the adage ‘garbagein garbage out’ applies. In essence, nothing can fix poorly recorded sound. Ifyour audio is going to be compressed, or its sample rate and bit depth arereduced, then it is very important to record clear, dynamic sounds. Choosing agood microphone is very important. There are a variety of microphonesavailable on the market, each offering different sound qualities that areoutlined in the following section, but firstly, let’s discuss how microphoneswork.Figure 18: Sony Mini Diskhttp://www.dealtimeshopping.com/DT_a19/mini_disk_player.htm
  27. 27. PAGE 27Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYHow Microphones WorkMicrophones work by converting real sound waves into electrical audio signals.They have a small light material inside them called a diaphragm. When soundvibrations travel through the air they reach the diaphragm, which causes it tovibrate. This in turn causes an electrical current which is sent out to a mixer,preamplifier or amplifier for use. Microphones are generally classed by howthe diaphragms produce sound.Dynamic and Condenser Microphones1. DynamicSometimes called a Moving Coil Microphone, this microphone works on anelectro-magnetic principle – that is, a coil of wire moving within the flux of amagnetic field to produce a small voltage. Dynamic microphones consist of afine coil of wire attached to a pressure sensitive diaphragm. This coil issuspended in a permanent magnetic field and as sound waves hit thediaphragm, the coil moves within the magnetic field thereby producing anelectrical signal. The microphone, because it generates its own signal(voltage), does not require a battery.Dynamic microphones are not as sensitive as higher-grade condensermicrophones and they have a reputation for being reliable and hardy which iswhy they are used frequently in live performances where they can take therough handling as well as more powerful sound. They are also relativelyinexpensive and have a ‘warm’ sound quality (see Figure 19).2. CondenserThis microphone works on a Capacitive or Electro-Static Effect. Thecondenser microphone is essentially a capacitor with one of its plates beingmovable and the other plate fixed (back plate). Once again, a diaphragm(which is bonded to the movable plate of the capacitor) is used to sensechanging air pressure - sound waves. As the air pressure changes it impactson the diaphragm, the gap (insulator) between the movable capacitor plateFigure 19: A Dynamic microphone
  28. 28. PAGE 28Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYand the fixed plate changes. This changing insulator alters the capacitivereactance (Xc) and thereby alters the current flowing through the capacitor.Condenser microphones require batteries to drive the capacitor circuit.Condenser microphones have smooth, sound quality and a clarity anddefinition not usually found in dynamic microphones. Another advantage isthat they can also be miniaturized, making them especially suitable for clip-onuse (see Figure 20). Both dynamic and condenser microphones can bedesigned to be directional or omnidirectional.3. Phantom Power for Condenser MicrophonesA power source is required to produce the charge on to the capacitor of aCondenser microphone. This power source may be provided by either aninternal battery, a permanent charge on the microphone’s diaphragm or by anexternal ‘phantom’ power supply.Phantom power is the supply of power through the ground cable of an XLRcable. The voltage of Phantom power supplies ranges from 9 volts up to 48volts. The power can enter the cable from a number of sources; from abattery pack which is an alternate source to the mains power; a phantompower box, which is like an intermediate component between a mixer and amicrophone that just puts a charge on the ground cable, or a mixer that mighthave a button that enables the phantom power source through the XLR cable(see Figure 21).Other types of microphones include Electret Microphones, PlaintalkMicrophones, Ribbon Microphones, and Carbon Granule Microphones.These can all be further researched on the Internet.Figure 20: A condenser microphoneFigure 21: PM4 - Phantom Power Adapter forCondenser Microphoneshttp://www.samsontech.com/products/productpage.cfm?prodID=118&brandID=2
  29. 29. PAGE 29Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORYCategories of MicrophonesMicrophones can be used for many different recording uses but unfortunatelythey cannot always pick up sound from different directions. The way in whicha microphone detects sound is known as its pickup pattern. The standardsof the pickup pattern are:1. Omni-DirectionalOmni-Directional Microphones pick up sounds from all directions. Theywork well, either pointed away or towards the subject, providing that themicrophone is at equal distance. Other factors that have a bearing on howwell the microphone maintains its omni-directional characteristics, is itsphysical size. The body of the microphone blocks the shorter high-frequencywavelengths that arrive from the rear; the smaller the microphone bodydiameter the closer the microphone can come to being truly omni-directional.(See Figure 22).Typical Uses: Used for vocals because of their lack of proximity effect,picking crowd noise at a football match or as lapel microphones fornewsreaders, which allows them to keep looking directly at the camera ortelereader.Figure 22: An omni-Directional microphone
  30. 30. PAGE 30Multimedia — Learning Resource — Version 3, November 2003 Copyright © 2003 QANTM Australia CMC Pty LtdICPMM44CA - INCORPORATE AUDIO INTO MULTIMEDIA PRESENTATIONS PART 1 - AUDIO THEORY2. Uni-DirectionalUni-Directional Microphones are best at detecting sounds from one directionie directly in front. These microphones are generally long and rod shaped withgrooves on the side. This allows sound coming from the side to either passthrough without reaching the pickup or cancel each other out. (See Figure23).A slightly modified pickup pattern is also found in specialised Uni-Directionalmicrophones. These are the Shotgun and Cardioid (Supercardioid andHypercardioid) microphones (See Figures 24 & 25).• Shotgun Microphones are more directional in that they can pick upclose perspective sound, with less background noise, from a greater distance.• Cardioid Microphones are less sensitive to sounds from behind,than they are to the sides and front, which is why they are favoured for stageuse. There are two types of the Cardioid pickup pattern. These are calledSupercardioid and Hypercardioid, which have limited ranges of pickup. (SeeFigure 25 on previous page).Typical Uses: Good for noisy locations to hone in on sounds such as in aninterview at a sports game. Good for drum andinstrument applications.3. Bi-DirectionalBi-Directional Microphones pick up sounds in two-axis -from two opposite directions. This is known as thefigure-8 pickup as, when viewed from above, thepattern resembles a figure-8 (See Figure 26).Typical Uses: Generally used in interview situations.Most stereo microphones can be used as bi-directionaldevices.Figure 23: Uni-Directional microphonesFigure 24: A Shotgun microphonehttp://www.micsupply.com/festivariandelight.htmFigure 25: A Cardioid microphoneOriginally sourced from:http://www.aes.harmony-central.com/109AES/Content/Earthworks/PR/Z30X.htmlFigure 26:A Bi-Directional microphoneA T T H I S P O I N T AA T T H I S P O I N T AA T T H I S P O I N T AA T T H I S P O I N T AA T T H I S P O I N T A T T E M P T A S S E S S M E N TT T E M P T A S S E S S M E N TT T E M P T A S S E S S M E N TT T E M P T A S S E S S M E N TT T E M P T A S S E S S M E N T SSSSS 1 T1 T1 T1 T1 TO 8O 8O 8O 8O 8PPPPPARARARARART A OFT A OFT A OFT A OFT A OF T H E A S S E S S M E N T C R I T E R I A S H E E TT H E A S S E S S M E N T C R I T E R I A S H E E TT H E A S S E S S M E N T C R I T E R I A S H E E TT H E A S S E S S M E N T C R I T E R I A S H E E TT H E A S S E S S M E N T C R I T E R I A S H E E T

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