I.Digital Audio
	A. steps of A/D, D/A conversion
		1.Signal from mixer, preamp, or audio interface, run through low pass 		filter, remove everything below 20 hz
		2.Filtered signal passes through A/D converter.  This samples voltage of 		audio waveform thousands of times a second.
		3.When measured, binary number produced that represents voltage of 		waveform when measured.  Called quantization.  1 and 0s called bits or 		binary digit.  More bits, more accurate.
		4.Numbers stored on recording medium as modulated square wave at 		maximum level.  Can be magnetic on hard disk.
		5.Reverse: binary numbers read from medium like hard drive
		6.D/A converter translates numbers into analog signal made of voltage 		steps
		7.Anti-imaging filter (also low pass, smoothing), smoothes steps 			resulting in original signal by cutting out really high frequencies that	 	result from the abrupt steps.
	B.Bit Depth and Dynamic Range
		1.Audio measured thousands of times a second to create binary 			numbers.  
		2.Longer the word (more bits), greater the accuracy of each 				measurement
		3.Short words, poor resolution, long words, good resolution 
		4.16 bits adequate for hi-fi and CDs, most recorders can use 24. 
		5.More bits sounds smoother and more transparent
		6.More bits, more accuracy, more possible amplitudes can be 			reproduced, dynamic range increase.  
	C.Sample Rate and Frequency Response
		1.rate which A/D converters sample analog signal while recording
		2.Higher sample rate, wider the frequency response of recording.
	D.Quantization Error and Prevention
		1.Software is often 24 bits but CDs are at 16, so last 8 bits need to be 		truncated, or cut off.
		2.24 bit good for capturing quietest parts of music like fades and reverb 		tails
		3.Truncation makes these sound grainy or fuzzy b/c 16 bits is less 			accurate measurement.
		4.Quantization error occurs when amplitude falls in between two 			different bits and it can be rounded, but results in distortion to signal. 
		5.This doesn't occur at higher sound levels
		6.Occurs because A/D converter assigns closest possible value.  The 		right bit, or least
		significant bit, either switches on or off depending on converter rounding 		up or down.
		8.This rounding may lead to faintly audible fuzzy noise in quiet passages
		9.Solved with dither, adding random noise to the lowest 8 bits of a 24 bit 		signal before truncating it to 16 bits
		10.16th bit is modulated with some 24 bit information, average value of 		modulated square wave recovered by low pass filter
		11.Most of 24 bit quality restored, quantization distortion is just a smooth 		hiss.
		12.Noise shaping used to reduce mid range noises and increase level of 		high frequencies
		13.Dithering makes sound more transparent, fades are smoother
		14.Add dither right before converting to 16 bits
	E.Nyquist Theorem, Aliasing, Prevention
		1.Defines nyquist frequency as half the sampling rate.	
		2.Analog signal can be reconstructed from samples if highest frequency 		is less than Nyquist frequency
		3.If analog signal not filtered, aliasing will occur, high frequencies above 		nyquist frequency appear as lower frequencies
		4.Inaudible frequencies turned into audible frequencies
		5.Sample rate is not high enough to accurately represent the waveform.
		6.if frequencies above nyquist freq removed, only waveform that will pass 		through sample points will be original
		7.A/D and D/A converters use oversampling to improve sound
		8.sampling done at higher rate than needed such as 8x 44.1 khz.  then a 		lowpass filter with gentle sloped anti alias filter used.
		9.Results in less pitch shift and less harshness compared to brick wall 		filter
	F.The dBFS scale
		1.0 dBFS is when you have no more room to increase digital recorded 		value
		2.Stands for full scale
		3.Represents maximum possible amplitude
		4.Full scale is highest possible, 0 represents no change from that
		5.Recording preferable at -6 db in case of unexpected peaks
		6.between -12 and-6 in studio because of less sound
	G.Wordclock
		1.Audio devices have internal clocks to set timing on samples
		2.Series of pulses running at sample rate
		3.Transfering digital audio from one device to another requires clock 		synchronization
		4.There is a master clock and slave
		5.Clock can be embedded in digital signal or on separate wire or 			connector
		6.On digital devices, you can select clock source.  
		7.Choices are internal, external, or world clock
		8.internal is A/D converters clock.  External is clock signal embedded in 		another devices signal
		9.Wordclock aligns sample boundaries across multiple digital devices.
		10.Some devices synchronize automatically and you don't need a clock 		source
		11.Uses a pulse or square wave as a reference signal.
		11.systems A/D converter is best master clock, use word clock cable from 		there.
		12.under 25 feet cable should be unbalanced.  over that balanced
		13.Most importantly, keeps sampling times aligned.
	H.Jitter
		1.Accurate conversions require precise sampling intervals
		2.Jitter is instability of timing of samples in A/D to D/A conversion
		3.Nanoseconds change can result in amplitude errors, slight veiling of 		sound
		and low level distortion or noise
		4.Spec of inaudibility is debated.  under 250 nanoseconds or 20…or 250 		picoseconds
		5.Analog noise and crosstalk in recording system can cause noise
		6.also inadequate cables pan pick up noise and hum and introduce 		phase shift, 
		degrading timing of digital signal
		7.Reduce with high quality clock with low jitter specs.  
		8.High quality well shielded cables as short as possible, bad cables can 		cause jitter.
		9.Use audio interface or A/D converter as master clock.
		10.Error is very subtle, unlike clock error.

II.EQs and Filters
	A.Filters
		1.filters cause rolloffs at frequency extremes.  reject everything above or 		below.
		2.filters named for steepness.  -6 db attenuation per octave is first order, 		12 db per octave is second order, etc.
		3.Lowpass filter-Passes frequencies below a certain level, gets rid of 		ones above this frequency.  Also high cut filter.
		4.Bandpass filter-FIlter that passes a band or range of frequencies, cuts 		everything else outside of this.  combine high and low.
		5.Highpass filter-passes frequencies above certain level, cuts 			everything below 
	B.Parameters/types
		1.simplest eq has bass and treble control. 
		2.three band eq has boost/cuts to lows, mids, and highs.  
		3.sweepable eq, allows you to set center frequency and control boosts 		and cuts
		4.parametric eq-contol center frequency, boost/cut, and bandwidth
		5.Q or quality factor is center frequency/bandwidth.  low Q: wide range, 		high Q: narrow range.
		6.graphic equalizer-outside mixing console, controls adjusted, positions, 		graphically show frequency response.  can be used on monitors or 		patched.
	C.Shelving and Peaking
		1.Peaking EQ-shapes response like a hill or peak when set for boost
		2.shelving EQ-frequency shape represents a shelf.  Done for high or low 		shelves.
	D.Uses
		1.improve tone quality, make instruments sound better.
		2.create an effect, extreme eq to create interesting sounds
		3.reduce noise and leakage, get rid of certain bands with high Q
		4.compensate for fletcher monsoon, restore bass and treble when 			recording loud groups
		5.make pleasing blend, two similar instruments make them distinct
		6.compensate for mic placement, if you have to mic where it doesn't 		sound best
		7.remix a single track, two instruments, w/e
		8.improve tonal balance of mix-make songs sound similar, or more 		commercial.
	E.better to use good mic placement than have to eq.  only use eq if you have to.

III.Dynamics Processors
	A.Dynamic Range concepts
	B.Compression settings
		1.Turns down volume if signal gets too loud
		2.Reduce gain when signal hits threshold
		3.Loud notes made softer, compressor reduces dynamic range
		4.Uses: consistent volume, prevent clipping, special effects on drums, 		vocals, guitars, etc
		5.Compress only individual tracks that need it
		6.ratio  or slope-change in input compared to output.  
		7.typically ratio is 2:1 to 4:1
		8.knee-transition point between no compression and compression.  		softer knee is low
		 compression for low level signals and higher ratio for high level signals 		for a more natural 
		sound than a fixed ratio.  hard knee is abrupt.
		9.typical threshold is -10 db
		10.attack time-how fast compressor begins reducing gain when signal 		hits threshold typically .25 to 10 msec
		11.release time-how fast gain returns to normal after level falls below 		threshold again.
		12.quick release good for lots of volume changes.  however, creates 		pumping sound.
		13.make up gain-increase output level of compressor by amount of gain 		reduction.  brings up level of quiet parts.
		14.side chain: connect to equalizer, compress only sibilant sounds on a 		track or breath pops
		15.limiter-10:1 ratio
	C.Noise gate
		1.also called expander.  remove noises during pauses, reduce gain 		when level falls below threshold.  helps clean tracks up and shorten 		decay.
	D.Slap Echo vs Echo vs Doubling
		0.delay-holds signal in memory and plays it back again.
		1.echo-50 msec to 1 sec repetition
		2.slap echo-100 to 130 msec
		3.repeating echo-fed back many times, timed to music
		4.doubling-23 to 30 sec.  fuller sound to instruments.
	E.Chorus and Flanging
		1.chorus-wavy, shimmering.  15-35 sec.  sweep delay time to cause 		pitch to bend.
		2.flanging-0-20 ms.  signals combine and have phase interference, 		causing peaks and dips
		3.comb filtering, very colored filtered tone. 
	F.Reverb Parameters
		1.reverb time-rt60, time it takes for decay to 60db below originlal level.		2.pre-delay-short delay before onset of reverb to simulate what actually 		happens in rooms, as well as adding clarity between the original signal 		and reverb signal.
		3.helps to separate reverb from main instrument sound
		4.density-how closely echoes are spaced together
		5.damping-set amount of reverb at certain frequencies
		
IV.Signal Flow
	A.Mixer Features
		1.Analog, digital, software, control surface.
		2.input: faders, eq, aux, pan pots, channel
		3.output: master faders, group faders, output
		4.Mic inputs: mic, line, track -> mic preamp  -> selector switch ->  inserts -		> fader -> eq -> direct out
		5.
	B.Inserts vs Auxiliary
		1.aux/side chain: send some of input signal to equipment outside mixer.  		pre or post fader.
		2.pre-fader for monitors
		3.post fader aux for amount of effects.  turn up send knob to send more of 		signal to aux.
		4.decide if you want a part or all of your signal edited.
		5.dry vs. wet signal, side chain
		6.compression mostly an insert unless you're doing parallel compression 		for drums, etc.
	C.Pre vs Post Fader
		1.pre to monitors.  post for effects.  
		2.whether or not something is effected by changes in the fader volume.
	D.Cue mix
		1.Aux track sent to musicians during tracking.
		2.All tracks are sent here usually.
		3.Usually pre-fader.