Digital Theory
Digital Theory
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Sampling Rate Comparison
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Sampling Rules
- Nyquist Theorem: the upper frequency limit of a digital recording is one-
half the sampling rate. CDs use a 44.1 kHz sampling rate, so their
frequency response extends to 22.05 kHz.
- Oversampling: sampling an audio signal at a higher rate than needed to
reproduce the highest frequency in the signal. Ex. sampling a 20 kHz audio
signal at 8 times 44.1 kHz is called “8x oversampling.” This process is
followed by a digital lowpass filter and a gentle-slope analog anti-alias
filter. Result: less phase shift and less harshness compared to a steep,
“brick-wall” analog filter used alone.
Digital Recording
- Reduces noise, distortion, speed variations, and data errors
- Because digital playback head reads only 1’s and 0’s, it is insensitive to the
magnetic medium’s noise and distortion
- During recording and playback, numbers are read into a buffer memory
and read out at a constant rate, eliminating speed variations in the
rotating media.
Recording Medias
- Hard-disk drive records on magnetic hard disk
- Compact disc and DVD recorder record on an optical disc
- Memory recorder records onto a Flash memory card
- Sampler records into computer memory
File Formats
- Recorded as a wave (.wav) file or Audio Interchange File Format (AIFF) file.
- Both formats use linear PCM encoding and no data compression
- CDs (Compact Discs) are 16 bit/44.1kHz --- also referred to as “Red Book
Format”
Digital Recording Level
- In a digital recorder, the record-level meter is a peak-reading LED or LCD bar graph
metere that reads up to 0 dBF
- In a 16-bit digital recorder, 0dBFS means all 16 bits are on (1) at the waveform
peak, and all 16 bits are off (0) at the waveform trough
- In a 24-bit digital recorder, 0 dBFS means that all 24 bits are on or off
- “Over” indication means that the input level exceeded the voltage needed to
produce 0 dBFS, and there is some short-duration clipping of the output analog
waveform
- Setting recording level: aim for -6 dB maximum so that unexpected peaks don’t
exceed 0 dBFS
Clock
- each digital audio device has a clock or internal oscillator that sets the
timing of its samples
- Clock signal is a series of pulses running the sampling rate
- When you transfer digital audio from one device to another, their clocks
must be synchronized; one device must provide the master clock and the
other must be the slave
- Clock comes embedded in the digital signal, or comes on a separate wire
or connector as a word clock signal
- Internal, External or Word Clock
Digital Audio Signal Formats
- AES/EBU: professional format that transfer 2 channels of digital audio over 110-
ohm shielded twisted-pair cable with XLR connectors
- MADI (Multichannel Audio Digital Interface): professional format that transports
28, 56, or 64 channels up to 24 bits/96 kHz on 75-ohm coaxial cable or fiber-optic
cable; used to link large mixing consoles to digital multi-track recorders
- S/PDIF (Sony-Philips Digital Interface): 2-channel consumer or semi-pro format;
uses a single 75-ohm coaxial cable with RCA or BNC connectors, or a fiber-optic
cable with TOSLINK connectors
- ADAT Light pipe: send 8 channels of digital audio on a single optical cable with
TOSLINK connectors; data transfer up to 24 bit/48 kHz or 24 bit/96 kHz with half
the number of channels
Digital Audio Signal Formats
- TASCAM TDIF (TASCAM digital interface): DB-25 connectors; TDIF sends 8 channels
of digital audio in and out ona single cable
- USB (Universal Serial Bus) and FireWire: standard Mac/PC protocols for high-speed
serial data transfer between a computer and an external device, such as a hard
drive, USB drive, MIDI Interface, or audio interface
- AES/EBU, S/PDIF, and ADAT Lightpipe are self-clocking systems (clock is
embedded in the signal and marks the start time of each sample.
- MADI, TDIF, and ADAT Sync carry a separate word-clock signal on a separate
connector or wire
- FireWire and USB signals don’t have clock information; reclocked at the receiving
device
Dither
- When you save a 24-bit audio file as a 16-bit file to transfer to CD, those last 8 bits
are truncated or cut off. Result may be a grainy static sound at very low levels. This
distortion can be prevented by adding low-level random noise (dither) to the
signal
- Example: 24-bit resolution can accurately capture the quietest parts of a musical
concert: very low-level signals such as the end of long fades and reverb tails.
Truncation of that signal to 16 bits makes those low-level signals sound grainy or
fuzzy, because 16 bits is a less accurate measurement of the analog waveform
than 24 bits. The fuzzy sound (called quantization distortion), doesn’t exist at
normal high levels.
Jitter
- the unstable timing of samples that occurs in A/D and D/A conversion
- Any change between the sample times creates amplitude errors -- small changes in
the audio waveform’s shape -- resulting in a slight veiling of the sound (low-level
distortion or noise)
- Accurate A/D and D/A conversions rely on the clock precisely sampling the analog
signal at equal time intervals
- One cause of jitter is analog noise and crosstalk in the recording system; they
affect the switching times and switching threshold of the clock, causing frequency
modulation of the clock; also affect analog filters and oscillators used in the clock’
s phase-locked loops
Jitter
- Also caused by inadequate cables; pick up hum and noise, and introduce phase
shift and high-frequency attenuation which degrade the timing of the digital signal
- Jitter doesn’t occur during real-time data transmission over FireWire or USB
because they are clockless systems
- To reduce jitter:
- use high-quality clock sources with low jitter specs
- use high-quality, well-shielded cables designed for digital signals, and as short as possible
- keep analog and digital cabling separate
- use your audio interface or A?D converter as the master clock; don’t drive it from an external source
Digital Transfers
- When sending digital audio signal in real time from one device to another, they
must be set to the same sampling rate
- In a digital transfer, the sending device is usually the master, and the receiving
device is the slave
- OK to send a lower bit depth signal to a higher bit device
- CDs must be converted down to 16 bit and 44.1 kHz
- WAVES or AIFFs copy a perfect clone of the original file; file transfers much faster
than the real-time playback of the signal
- Flawless digital transfers:
- computer from one hard drive to another
- Ethernet, USB, FireWire, or the INternet
- Between computers or flash memory cards
Digital Audio Workstation
- DAW: a computer running recording software with a connected audio
interface such as a sound card
- Allows you to record, edit, and mix audio program entirely in digital form
Sources
Practical Recording Techniques Text Book