1 manipulating digital audio. 2 pulse code modulation (pcm) this is a means of encoding the digital...

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Manipulating Digital AudioManipulating Digital Audio

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Pulse Code Modulation (PCM)

Pulse Code Modulation (PCM)

This is a means of encoding the digital signal for transmission or storage

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PCMPCM

Very robust – only the presence or absence of a pulse is looked for

Several signals can sent simultaneously using time division multiplexing

Can contain synchronization and error correction information

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Types of CodesTypes of Codes

There are several possible codes that can be used with PCM

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Error CorrectionError Correction

With analogue audio there is no opportunity for error correctionIf a signal is distorted or disrupted then it is

irrevocably damagedDigital data can contain redundancy

This enables the reproduced data to be checked for errors

Further processing can correct or conceal errors

Strong error correction techniques relax the manufacturing tolerances for CDs, DVDs etc.

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ISBN- Error DetectionISBN- Error Detection

International Standard Book NumberISBN 1-55105-083-8. The hyphens

separate the different parts of the number:•1 : A 0 or a 1 in this position indicates

that the book was published in an English speaking country

•55105 : This is the identification number of the publisher

•083 : Identifies the specific title and edition

•8 : is the check digit

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ISBN- Error DetectionISBN- Error Detection

ISBN is a ten digit numberMultiply the first number by 10Multiply the second number by 9Etc..Sum the resultsThis number should be divisible by 11

0-7167-8321-50-89678-354-10-333-77339-X

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Error Correction MethodsError Correction Methods

Cyclic redundancy check (CRCC)Cyclic block code that generates a parity

check word

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Digital to Analogue Digital to Analogue

DemultiplexerProcessing

(error correction)

Reproduction demodulation

Output sample

and Hold

Output lowpass

filter

D to A Converter

Output sample

and Hold

Output lowpass

filter

D to A Converter

Analogue output (L)

Analogue output (R)

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D to A ConverterD to A Converter

Requires precisionWith a +/- 10v scale a 16-bit converter will

output steps of 0.000305vMain problems are:

Absolute linearity error – deviation from ideal quantization staircase

Differential linearity error – relative deviation from the ideal staircase by any individual step

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Absolute Linearity ErrorAbsolute Linearity Error

Straightness of converter output

Digital input

Analogue output

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Differential Linearity ErrorDifferential Linearity Error

Deviation of individual steps

Digital input

Analogue output

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Zero-cross DistortionZero-cross Distortion

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High-bit D/A ConversionHigh-bit D/A Conversion

Using an 18-bit converter to perform 16-bit conversion

Each 16-bit LSB is represented by 4 Gives better 16-bit conversion than a

16-bit converter

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Sample and HoldSample and Hold

D/A converters may require time to stabilise to the correct output levels

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Low-Pass FilterLow-Pass Filter

Anti-imaging filterRemoves all frequency

content above the half-sampling frequency

Converts the D/A converters output pulse amplitude modulation (PAM) output to a smoothly continuous waveform

Output of sample and hold

Output of anti-imaging filter

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OversamplingOversampling

Analogue brick-wall filters introduce phase shift and distortion

To get around these problems we can use oversampling with a digital filter

The oversampling filter has two tasks:Re-sample at a higher rateFilter through interpolation

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OversamplingOversampling

The original signal is sampled at fs and so has images centred around multiples of fs

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OversamplingOversampling

Re-sampling involves inserting zero valued samples between the original samples

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OversamplingOversampling

The digital filter performs interpolation to form new sample values

The output filtered signal has images around multiples of the oversampling frequency fa

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OversamplingOversampling

As the distance between baseband and sidebands is larger, a gentler analogue filter can remove the images without causing phase shift or other artefacts

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