Analog and Digital Sound Representation
Natural Sound - Resonance
It is easy to produce sound that contains a jumble of frequencies: wind, impulse/rattling/vibrating
A resonant cavity is a filter: amplifies frequencies near its wavelength (and multiples), suppresses other frequencies
Most sound-producing things operate in/with a resonant cavity: voice, instruments, etc
Natural Sound — Voice
The human vocal tract
Sound source ("vocal chords") + resonant cavity (larynx, mouth, etc)
Frequency range unsurprisingly similar to hearing range
Natural Sound — Acoustic Instruments
Noisemaker + resonant cavity
Wind: buzzing lips or reed + tube
String: vibrating string + usually cavity
Percussion: impulse + usually cavity
Misc
Pitch adjustment by tension or length; cavity length modification via holes (or slide) — so many choices
Most but not all monophonic: one sound at a time
Analog Sound — Electrical Representation
Represent sound pressure as a voltage on a wire
The classic: telephone
Allows for transmission, processing
Analog Sound — Distortion
Ideally, electric signal exactly represents sound pressure
In practice, the signal path may introduce distortion
Nonlinearity: the signal doesn't accurately track the sound pressure
History: the past signal influences the current signal
We will talk about "harmonic distortion" (THD) at some point
Analog Sound — Attenuation / Amplification
Simplest transformation
Attenuation: Sound out linearly less than sound in
Easy to attenuate in all the obvious ways
Amplification: Sound out linearly greater than sound in
Amplification usually requires electronics
Analog Sound — Speakers
Turn electrical signal into air pressure change
Wire solenoid attached to paper cone like this
Typically in a resonant cavity (speaker cabinet)
Speaker solenoid roughly tracks change in current through the wire
Need wavelength to be long for low frequency: big speaker or pair of separated speakers (long baseline) — "woofer"
- C.f. Huygens's Principle and formula:
Need response time to be fast for high frequency: tiny speaker, maybe piezoelectric — "tweeter"
Analog Sound — Recording
Turn sound into electrical signal: usually with microphone
Microphone varies resistance, capacitance or voltage (reversed speaker) depending on air pressure differential between front and back
Many variations on this theme
Microphones are bad: noisy, nonlinear devices; usually limiting factor in sound chain
Analog Sound — Signal Path
We now know how to build something like a telephone or record player or stomp box:
Use a microphone to convert air pressure to voltage
Maybe process the voltage somehow: store it somewhere or modify it with circuitry
Use a speaker to convert voltage back to sound
Analog Sound — Feedback
"Feedback" is a classic oscillation effect:
Sound coming out the speaker and back into the microphone interacts with speaker + microphone + air as a resonance
The resonant frequency depends on the distance between microphone and speaker
If the loop has net positive gain at some frequency (amplification)…
Analog Sound — Limitations
Representation of analog sound as an electrical signal is potentially awesome: high accuracy in time, can represent very high and low frequencies well
In practice, there are problems:
Any "noise" (unwanted signal) is also very accurately represented
Analog signal storage devices are clunky, and don't work well: records, tapes, etc
Manipulating electrical signals requires complex and expensive and special-purpose electronics
"Audiophiles" love this stuff, so you have to deal with them (could be worst problem)
Digital Sound — Discretization
"Digitizing" analog sound solves our problems:
Somewhat noise-immune
Can be stored in digital memory
Can be manipulated with a simple computer
Audiophiles hate it
What representation should we choose? Discretize analog signal in time and space as "samples"
Simplest to use uniform sampling time interval, binary integer representation of sample values
- High sampling rates and lots of bits is more accurate, but "wasteful" for slowly-varying signals
This is often called "Pulse Code Modulation" (PCM), and is the basis of most ("time-domain") digital representations of sound
Usually PCM is "Linear Pulse Code Modulation" (LPCM): the binary sample values are interpreted directly. Sometimes a function is used to transform the sample values (e.g. A-law, μ-law) to try to use fewer bits with a decent representation: see below
Digital Sound — Nyquist Limit
Sound is a fundamentally frequency-domain (sum of sinusoids) thing: PCM treats it as time-domain
A particularly striking example of this is the "Nyquist Limit"
To make PCM work well, we need to ensure that we don't try to represent signals that vary quickly relative to the sampling rate
Specifically, we need to ensure that frequencies above half the sample rate are not present in the underlying signal (this is a strange way of putting things, but the math checks out)
We will return to this topic throughout the course
Digital Sound — PCM Representation
Sound is represented as sequence of samples: numbers
Usually fixed-width integers: signed or unsigned
Floating point can also be a thing
Units are complicated: usually just normalized to range of values for int and 0..1 or -1..1 for float
There is some specified sample rate in samples per second: note that samples per second is 2× max frequency in Hz, because Nyquist
Stereo (or more), so sample-per-channel, interleaved in the "obvious" way: frames
Digital Sound — Sources Of "Approximation"
Band-limited via Nyquist (approximation in time)
Quantization due to finite representation (approximation in amplitude)
Assumes an idealized sampling clock — clock "skew" and "jitter" is a thing for real clocks
Digital Sound — Digital To Analog Conversion
Need to take a binary number to a voltage
Classic method: direct conversion via R/2R Ladder
Very fast, simple
Accuracy issues are real: bit voltages and component values must be matched pretty exactly
Classic method: Pulse Width Modulation (PWM)
Digital all the way to single-wire output
Arbitrary resolution dependent on timing
Really hard to get the filtering right for audio applications: want super-high pulse rate
Fancy methods: can talk about later if folks are interested
Digital Sound — Analog To Digital Conversion
Convert voltage on wire to binary number
This is the "hard" direction: the DAC tricks aren't invertible
One common approach uses some combination of DACs and comparators to try to make the DAC output match the analog input