wk5- AUD114 – Digital & analog audio

Digital audio

transducer – mic is transducer

analog signal flow:
spl-mic-mic preamp-console-???- DAW

Analog to digital :
anything digital in signal flow requires processing

ADC/DAC process – anolog to digital or digital to analog
most digital audio devices have built in ADC/DAC’s (eg. rverb/fx unit/delay pedal,etc)

______________________

why digital?

-its easier/convenient

_______________________

ADC
acts like transducer
translates electrically to numeric (1 and 0’s)

analog signal from mixer,mic preamp,etc – sampling – quantisation

sampling – taking portions
measuring analog signal voltage at various time intervals determines by sample rate
sample rates always in KHz
determines number of measurements taken of analog sig voltage/sec
(think of sample rate like camera pixels)

Nyquist frequency – highest freq an ADC can handle is = to half the sample rate (eg. 48khz would have Nf of 24khz)

alias frequency – incorrectly sample resulting in frequencies that were never there being heard in playback (form of distortion- not good)

anti-aliasing filter – most gear has built-in & will role of freq above selected khz (steep freq sound bad)
Oversampling – 1.OS on record 2. OS on playback (creates nicer sounding role off

*soundworks – sound of godzilla documentary 

quantisation –  making things in time/putting something on grid
involves amplitude
assigns binary number to each taken sample
ie. bit depth
bit is 1 or 0 (binary numbers) – basically on or off switch
eg. 16bit ADC would give 16 digit number to each sample taken
greater bit depth = more accurate

quantisation error – sample voltage unlikely to have precise binary value. gets rounded up or down to nearest quantisation value (changes waveform slightly)
lower signals suffer most of Q errors as less bit res is used (its rounded down)

Dither – filtered white noise – white noise = all freq played at same level
dither basically white noise cut around 1-5khz (hides form your ears)
dither reduces Quantisation error

‘what is PCM’ video

multiplexing –  making it interleaved
error processing signal flow

DAW operation/Optimising protools

*See also WK1 AUD113 Tri 2 Blog for more…

Lynda tutorials

Playback engine settings
Buffer sizes – a smaller buffer size will have the least latency (lag basically) – this is most useful as a setting while recording
where as a high buffer size will be more useful when mixing as this allows for the use of more plug-ins (which creates more latency)

ADC Automatic Delay Compensation –
When using a lot of plug-ins in a session your tracks might become out of sync (usually just by milliseconds) when playing back.
ADC (when enabled) takes the track with the longest lag and adds the difference in lag to the other tracks in order to keep them all in time.

If ADC is enabled it will appear as ‘Dly’ in the edit window (see below pic)
ACtivate in OPTIONS- make sure delay compensation is ticked

ADC is becomes more important as your session becomes more complicated with more plug-ins

*MIDI tracks – measure impedance ?
*Instrument track – combo of MIDI and AUX

When would each be used?

Wk1 – AUD113 DAW Operation

Notes:

Lynda.com is good shit.
Dave Pensado (pensados place)- talks to producers
Work flow is super important
Reaper is an underrated DAW
Q-base is on par with protools (but not industry standard
Abelton is fantastic for making music

INFO
Protools-Preferences
(display) – Organize plug-in menu~Catalogue &manufacturer — this allows you faster access to plug-ins if you are looking for a specific manufacturers one
– Organize I/O menu ~ Type & width — Where interface interacts and (width) how many

(Operations) – AUTO backup settings

(editing) – Fade Shapes and level of undos

(Mixing) – Default EQ/Dynamics — can create shortcuts to most used ones so you dont have to find them in drop down menu

(Processing) – AUTO copy files on import —- so you don’t lose tracks!

*Dragging audio onto DAW will automatically convert the sample rate of file to what the session is set to
*Command +shift +I ~ shortcut to open proper import audio window
*option +shift+ I – shortcut to import session data
*playing file in wrong sample rate affects speed and pitch — think of a vinyl record being slowed
e.g 44.1khz file played in 48khz will be sped up (can be used as cheat to increase speed of guitarist in post)
48khz file played at 44.1khz will be slower and lower

_______________________________________

I/O setup (for if you fuck up or someone deletes I/O

set-up / I/O / New Path

Hold option (so it applies to all tracks) select output as whatever new path was named

DONE

______________

Tip:

in laptop system preferences / keyboard

check box for use standard keys

F1/F2/F3, etc can be used as shortcuts to toggle smart tools and slip mode

_______________________

Reflection:

This seems like an enjoyable class and Akshay(?) seems like a good lecturer.
I am looking forward to becoming more proficient in shortcuts and learning wtf a lot of the stuff in protools is.