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28 Cards in this Set
- Front
- Back
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PCM quantizes analog audio into steps
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These steps are:
- 65,536 steps for 16-bit audio - 16,777,216 steps for 24-bit audio |
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Dynamic Range
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- The difference in dB between noise floor and maximum output, before distortion
- Wordlength x6 - 6dB of gain per bit |
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Lowest Signal
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- 16-bit audio = -96dB
- 24-bit audio = -144dB - 48-bit audio = - 288dB - 56-bit audio = -336dB |
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PCM quantizing creates
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- Stepped Distortion
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PCM has difficulty
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- encoding signal near the noise floor
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Dither
*(is white noise, equal energy per frequency) |
- randomized noise which averages 0 volts DC (having an equal number of 1's and 0's)
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Dither is added
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- to the analog input stage to toggle or alternate the LSB* in the wordlength
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LSB*
(Least significant bit) |
- represents the lowest level signal at the converter
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Modulating the LSB
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- allows signals at or below the noise floor to be encoded by averaging the dither gain with the low level signal
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Cheap low-pass filters (LPF)
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- create higher distortion, ripple (ringing) and aliasing.
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Ripple/Aliasing artifacts are heard
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- as time-smearing of the audio and short (millisecond) echoes
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Pass Band
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- Unfiltered range below LPF
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Transition band
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- beginning of cutoff frequency to stop band
- steepness determines slope of filter |
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Stop band
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- point at which maximum loss of the filter is reached
- at or below Nyquest frequency to prevent aliasing |
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Aliasing or ripple in the passband is caused by
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- the LPF not attenuating enough in the stop band, allowing excess signal above the Nyquist frequency to fold back (or alias) into the passband
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Aliasing can be prevented
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- with a very steep filter, or a gentle filter at a high sample rate
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For ease and cost, most manufacturers -
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- use cheap filter/high sample rate approach
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Filter distortion issues are off-ten fixed by...
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- Oversampling (A/D) or Downsampling (D/A) processing
- Where the front or back end of the converter operates at double (or greater than) the sample rate |
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EXAMPLE
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- A 44.1kHz source with a 128x oversampling converter is actually sampling at 5.64MHz
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Oversampling...
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- moves the filter distortion above the audible range, which itself is then filtered out with a "decimator" anti-aliasing filter in the downsampling phase
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This LPF decimator...
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- is applied AFTER the oversampling circuit and allows poor filters to not become a liability in the converter's performance
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The biggest impact on our perception of clarity in digital audio...
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- The LPF designs used in different converters - their performance impacts the sound more than higher sample rates alone
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High sample rates sound better
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- because they push the filter's distortion out of the audible band into the area above 20kHz where it cannot effect musical content
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The ideal response for an LPF
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- has to be at or above 50kHz to avoid pre-echo artifacts from being created in the cochlear filter of the human ear
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192kHz sample rates may be unnecessary...
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- as most 192kHz ADC's are not really doing more than a double 96kHz conversion
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This higher sample rate...
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- generates little usable content while introducing extra noise, raising the potential for calculation errors
- and creating storage/processing challenges |
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The use of 88.2kHz and 96kHz rates for sampling and reproduction
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- are justified as they add the benefits of the higher filter setting while maintaining reasonable size files
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The computational simplicity of down sampling 88.2kHz content into 44.1kHz
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- makes it an option preferred by many professionals
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