Tontechnikkompendium de en

-- Tontechnik Kompendium--
Sequencing 1 ____________________________________________________________ 7 HD: Peak (Mac OS X)

____________________________________________________ 12 Signal Flow
1___________________________________________________________ 14
The Mackie 8-bus ___________________________________________________________ 18
Physic Basic ____________________________________________________________ 25

Fundamentals of Vibration Theory ____________________________________________ 25 Harmonic oscillations
(sine waves) _______________________________ 26 Fundamentals of wave theory
_________________________________________________ 31 Acoustic
___________________________________________________________________ 33 sound energy quantities
_________________________________________________________ 36
Level invoice: _________________________________________________________ 39 psychoacoustics:

__________________________________________________________ 47
Structure of the ear: ___________________________________________________________ 47 listening (listening
area, Hrfeld) ______________________________________________ 49 Haas Effect:
______________________________________________________________ 51 directional hearing in the horizontal
plane: ________________________________________ 52
HD Recording Basics (ProTools) ___________________________________________ 54

Use Hall: ____________________________________________________________ 55 Tracks: 57
___________________________________________________________________
Tape machine __________________________________________________________ 58

Sound recording _________________________________________________________ 58 path of the tape
____________________________________________________________ 59
Electrical engineering 1 62 ____________________________________________________________
Grundlagen________________________________________________________________ 62 air pressure and water

model ________________________________________________ 63 types of current (voltage types)
________________________________________________ 65 electricity transmission from the power station to the
consumer _________________________________ 67 resistors
____________________________________________________________ 74 series of resistors:
___________________________________________ 75 Parallel connection of resistors:
__________________________________________ 76 charge storage:
___________________________________________________________ 79
Instrumental Acoustics: _____________________________________________________ 81

Time structure: _______________________________________________________________ 82 frequency structure:
__________________________________________________________ 83
Music Theory: ___________________________________________________________ 87

Basics: _______________________________________________________________ 87
Page 2 of 466
The circle of fifths: __________________________________________________________ 88
Effects ________________________________________________________________ 90
Effektkategorien____________________________________________________________ 90 Passive filters
______________________________________________________________ 91 Active filters
_______________________________________________________________ 96 control amplifier (VCA)
_____________________________________________________ 101 Standard (rock) instruments and their
settings __________________________ 108
Microphonic 1 __________________________________________________________ 115

Merge Recipients: (Pressure transducer) _______________________________________ 115 moving coil
microphone ______________________________________________________ 118
Frequenzgang_____________________________________________________________ 122 boundary microphone
_____________________________________________________ 124 microphone positioning
____________________________________________________ 126
MIDI_________________________________________________________________ 128
Messages _________________________________________________________________ 132 System Messages 143
__________________________________________________________ synchronization
___________________________________________________________ 146 MIDI Recieve Modes
_______________________________________________________ 157
Sequencing 2 __________________________________________________________ 162

Audio Features in Logic __________________________________________________ 164 MIDI
____________________________________________________________________ 172
Sampling: 174 _____________________________________________________________ MTK HD Recording

(Pro Tools) ___________________________________________ 180
Configuring a Pro Tools HD system ____________________________________ 180 types of tracks
__________________________________________________________ 181 Modes
_________________________________________________________________ 183
Signal Flow 3 - Studio 4 - Mackie DXB _____________________________________ 187 Signal Flow 3 - Studio
3 - SSL ASW 900+ __________________________________ 193 microphonic 2
__________________________________________________________ 195
Directional differences in the horizontal plane: _________________________________ 195 Spatial stereophonic:
________________________________________________ 196
XY stereophony: _______________________________________________________________ 196 Blumlein arrangement:
____________________________________________________________ 197 MS stereophony:
_______________________________________________________________ 197 Laufzeitstereophonie:
_____________________________________________________________ 200 quivalenzstereophonie:
__________________________________________________________ 201 separator stereophonic:
________________________________________________________ 202
Head Related stereophonic _________________________________________________ 203
Original head microphone ____________________________________________________________ 204 dummy head
_____________________________________________________________________ 205
Sound synthesis _________________________________________________________ 207

Page 3 of 466
Subtractive synthesis _______________________________________________________ 207

Standardwellenformen____________________________________________________________ 207
Additives Synthese__________________________________________________________ 213 resynthesis

_______________________________________________________________ 213 Physical Modeling
_________________________________________________________ 214
Granularsynthese__________________________________________________________ 214
Research______________________________________________________________ 215 Session Procedure

______________________________________________________ 220 Production 1 225
__________________________________________________________
Synchronistaion________________________________________________________ 227
Synchronization ___________________________________________________________ 227 technical
Synchronisation__________________________________________________ 227
Wordclock________________________________________________________________ 231
Signal Flow 3 - Studio 2 - D-Control ______________________________________ 233 236

Digitaltechnik__________________________________________________________
Advantages of digital technology __________________________________________________ 236
Dithering_________________________________________________________________ 243
Framing__________________________________________________________________ 251 Error handling 260
_________________________________________________________
Rundfunkton __________________________________________________________ 266

As the signal comes to the receiver? ______________________________________ 270 Wireless transmission
technology ______________________________________________ 272 self-propelled Studio
_________________________________________________________ 273 digital radio home
________________________________________________________ 280
Signal Flow 4 - NEVE 88 RS _____________________________________________ 282 film and

Fernsehton____________________________________________________ 298
Optical sound _________________________________________________________ 300 FILMPRODUKTION
______________________________________________________ 301
Synchronstudio____________________________________________________________ 308 function True
Diversity ____________________________________________________ 312
Automation____________________________________________________________ 315 E-2 technology

___________________________________________________________ 321
Capacitor __________________________________________________________ 321 Capacitive reactance
________________________________________________ 323
Spule_________________________________________________________________ 325
Coil / coil / inductor 325 ______________________________________________

Bauarten_______________________________________________________________________ 326
Inductive Blindwiderstand_______________________________________________ 327

Page 4 of 466
Coil to AC ________________________________________________________ 327

Effective resistance - reactance ___________________________________________ 329 series resonant circuit 336
________________________________________________________ semiconductor diodes
________________________________________________________________ 338 341
___________________________________________________________________
Switching characters ___________________________________________________________________ 343 ratio of voltage
and Strom_________________________________________________ 347
Netzteile__________________________________________________________________ 349 Transistor

____________________________________________________________ 351
Operation of a transistor (NPN) ______________________________________________ 352
Amplification circuit _____________________________________________________ 357 operational amplifier 359
______________________________________________________
Speakers __________________________________________________________ 369

Converter principles in speakers _________________________________________ 369 Frequency response
_______________________________________________________ 375 Thiele Small
Parameter_____________________________________________________ 376 acoustic short circuit
____________________________________________________ 377
Live art __________________________________________________________ 380

Registration Setup: __________________________________________________________ 381 cable
Livebereich_______________________________________________________ 382 requirement on a PA
___________________________________________________ 384 PA species
_________________________________________________________ 385 coupling:
________________________________________________________________ 389 Feedback / feedback
___________________________________________________ 392 Low-technology
_____________________________________________________ 395
Multichannel _________________________________________________________ 398

Others to multichannel ________________________________________________________ 403
Multichannel Mifkrofonie _________________________________________________ 404

Mehrkanalmikrophonie ___________________________________________________________ 411
Production 2 __________________________________________________________ 412 DVD authoring

________________________________________________________ 416
File structure 417 _____________________________________________________________
Kopierschutz______________________________________________________________ 418
Building and room acoustics: __________________________________________________ 421

Coincidence effect __________________________________________________________ 421 reverberation time T
____________________________________________________________ 423
Schalleistungsmessung______________________________________________________ 425 Building Acoustics
_______________________________________________________________ 426 LEDE concept:
___________________________________________________________ 434
Mastering _____________________________________________________________ 435

Page 5 of 466
Formate__________________________________________________________________ 435 record 440

_______________________________________________________________
Music Business ________________________________________________________ 445 Business

______________________________________________________________ 452
Formelsammlung_______________________________________________________ 458 references Acoustic
____________________________________________________ 466
Page 6 of 466
Sequencing 1
Sequence: Sequence order, series of Something A unit in
the same or similar repeat is. Sequencer:
Unit (now mostly software) for creating, editing and reproducing Sequencen.
Originally not audio but control data or signals that cause sounds are played.
reasons:
Previously: -low memory

-flexibility
Control data / information is used to play back tones from a different source.
Sounds are stored as parameters and as waves. These parameters can be moved in
time and in your Tnhhe and you play in different sounds.
Data throughput:
MIDI Music Instrument Digital Interface interface for electronic tone

generator / Instruments Digital Interface
Information sent over 0 to 1. The combination gives
the commands.
keayboard
(No sound)
SOUND GENERATOR
(Sends control data)
(Including sounds)
A flow of information from here to there.
Transmitter (transmitter)
Receiver (receiver)
Control line received by the receiving device determines a control voltage pitches. The incoming
voltage controls the pitch. control voltage Control Voltage (CV)
Page 7 of 466
Then came synths with sound storage options Midi is a control line and no audio line. Midi is an
internationally standardized interface standard (hardware) and data format for notifying the
control data in real-time and non-real time equipped with this option devices.
Use of Midi:
Midi Keyboard
sound Modules
Midi cable wiring runs in one direction MidiMessage "Note

on" when button is pressed.
MIDI cable
Keyboard
Audio line
sound Modules
speaker
Computerized Midi Sequencing

Midi Keyboard Computer Interface Audio / MIDI Interface sound
computer
Modules
Snapshot Automation: Change the play while recording MIDI events on a timeline
form from rectangular. These bars describe their length, the duration of the note
played and in its height, the pitch of the played sound.
Midi signal has delay in USM cable because traffic and signal flow.
Voices & Co
Polyphonic (polyphonic)
monaural (unanimously)
Power is limited.
32 Trade polyphonic 64
specialist
can simultaneously play so many more votes. Many different sounds simultaneously
have different timbres Multitimbral
many instruments
homophone Equal Voices sounds are played simultaneously.
Page 8 of 466
Hardware:
Sockets and plugs standard 5 pin DIN cable 3 of 5

Poland occupied. Plug sockets always male always
female
serial link:
0.1 are sequentially driven by the cable. Transmission
capacity only 31250 bits / s "Note on" Message
30 seconds.
Midi IN
Midi Out
Midi Thru
Input for MIDI data
Output for the device generated data
Loop jack. 1: 1 copy of the present

at the Midi In
Data (Without Midi Out data)
Virtual channels in Midi Thru sent to different files. Relocation of many processes in the
computer. Simplified Total Recall.
General Midi:
Standardized GM channels which are numbered and attend special instruments. In each unit
equal to the reconstruction of Midi data.
In various sound producers have the sounds in the same standardized sequence were
reproduced in various sequencers. Fixed sound producers 128 is stored sounds.
Page 9 of 466
Logic Pro
Apple Logic Pro 7
linear MIDI / audio sequencer. Works with

midi and audio files. linear because the
timeline Startup via template Midi1.Iso
File format of Logic document .iso = Logic Song Logic Pro
Preferences
Application settings, preferences (complete program)

File sound Settings
Song Presets, Documents settings
The main window Arrangefenster
Window options to open additional windows and documents. Each window has
its own window ledges and settings. Edit, Track, Region, Midi, Audio, Horizontal
View X-axis vertical bars
Tracks so-called tracks Toggle Turn

function switch to SPL Song Position Line Play line
Enter is Play 0 Stop
2 * 0 Advances with SPL to the beginning
Space = Enter, Stop, Enter
, = Pause
SPL is movable with mouse

double Start clock display: 3
clocks
150
beats
vision
Midi ticks
1/3048 assigns Midi files to the clock. Locator:

Marker translated.
Recording or reproducing start and end settings so often used for Loop. Per
mouse / Shorts drag and move. pull backward loops Skip Cycle
Page 10 of 466
Is the marked section not again. Esc: - tool list Select

tool Ctrl Selecting and working tool directly. Edit tool
click track region is created in the selected track

held Alt key Copy when moving Shift
Selektiere specific objects held. apple Z
undo available undo Undo History
window Edit. Open for the region a Editor:
Matrix Editor Key Editor Piano Roll Editor Mark
Region
Edit Matrix Editor or apple 6 color of the attacks

shows how strong the attack was. green
light
Velocity Tools
red
heavy
In Editor
V Changes the velocity of sound function Quantization:

Q Click Hold Move down 1/16 note eg Alt + Shift make
known
New Region is a reference link from the old. I / O Track
under Audio, instrument track
run can be switched under SP.
For I / O input from other sounds for the Midi files choose tone generator. Sampler EXS24
Highlight Backspace Remove
Double click on the track gives me the Mixer View Object

Handling Regions editing
Page 11 of 466
HD: Peak (Mac OS X)

Shortcuts for Peak:
+ .-
zoom in, zoom out
Backspace
delete
create loop
Apple + Shift + (-) =
Selection manipulated by shift Button. Marker

contact Apple + m.
select clip "Apple + mouse click". Apple +
Shift + r
set region
Serves at peak for non-destructive operation by placing it in the
playlist.
There are 3 types of markers
1. marker
. 2 loop markers
. 3 Region markers (non-destructive)
Apple + Shift + p
Calling playlist
About the Contents moves window to the Regions in the Playlist window.
Apple + Space =
Playback with preroll ( Forward) serves as order transitions
check without having the clips which longer always listening

needs.
Ctrl + Space
Preroll function from the playlist window.
Spacebar = Play Selection

with mouse contents Shift
selection manipulating
cursor left
selection beginning
cursor right
selection end
Zoom amplitude = +/- Ctrl Ctrl Left / Right
= scroll further Selects = Spacebar
only being played
Loop define action looping
selection
Set Loop, will be played Loop
markers displaced
Page 12 of 466
delete edit without audio file Alt Backspace Markers are

taken out Apple + M set markers, numbered double change
name options Scroll reproduction apple + click in a
direction
selected until the next marker

Apple + Shift + click Extend the selection to the next marker Shift + click
free selection
Regions
(Destructive non) in a playlist
Create a playlist Apple + Shift + P new playlist

region in a playlist organize window
content
listed audio in the playlist as regions

Regions from the Contents / content window by drag and drop them into the playlist, Region
markers manipulate with Shift-click
Page 13 of 466
Signal Flow 1
A mixer is a signal collectors / distributors
A bus consists of platinum due to which one can merge any old channels
1. Master track: Most stereo,

rare mono
- The volume, which appear on the master is dependent on the fader position
- He is the exit to a 2TK (eg DAT)
- routable touch of a button
- Master not routable
- has an ISR
- has physical outputs
2. subgroups (subgroups):
actual functions as a master, only the subgroup is routable to the master

are the way to MTK, without going through the master
each channel is routable to a subgroup of patients who always goes directly to a MTK
mostly mono, stereo rare

Create a mixdown, Submixungen
Groups:
- a group is no busbar
- it is a collection of volumes
- only the conditions remain the same
3. AUX sends:
- there are mono and stereo (but usually mono)

- physical outputs
- have own master
- No ISR
- is placed with Poti on busbar (routed)
- are not routable
tasks:
1. to effect feed (dry signals)
- POST-FADER, so the effect signal follows the direct signal in the levels
2. for headphone mixes (always PRE FADER and off-Tape)
- so that the headphone mix is independent of the Director-Mix
Page 14 of 466
4. Solo:
- have no physical tap

- There are three different kinds of solos:
1. PFL:
Tap pre-fader, right after In-Gain

for setting levels in order to reach the largest possible signal to noise ratio
so that all signals are electrically identical

allow for maximum signal to noise ratio to sufficient headroom
different operating levels:
- Studio level + 4dBU = 1,228V
- Home recording level = -10dBV = 0,316V
- Broadcasting House standard level = +6 dBu = 1.55V
2. AFL:
- Tap after Fader (sound processing Solo)
- you can simulate with PFL AFL:
1. EQ
2. grind nothing
3. faders to zero (unity)
4. PAN to L or R (for stereo AFL)
3.SIP:
- uses master track as a solo track

- everything is muted except for listening signal
- Special function Solo-Safe
- Solo Safe switched channels are not muted when SIP
Recording:
ToTape way:
Input Gain (leveling in PFL solo)
signal wallbox Channel (Channel Path)

Signal passes through the channel
enters the Subgruppenrouting
8 subgroups Multitracker
(Keeping in 8 subgroups the recorded signals determined.)
Offtape way:
Multitracker Channel (Monitor path) Monitor system what we take. ToTape the intercepts communications
before the band / HardDisk
Offtape the intercepts communications to the tape, which was recorded and back
comes in the channel of the panel. Each channel can

process 2 signals Absorbed in the Multi Tracker (Channel
Path) Recorded interception (Monitor path)
Page 15 of 466
mixdown:
Multitracker
Signals in the channel path
Signals are processed with effects (EQ, Dyn)
any subgroups
2 Tracker
(Signal traces are collected in the master and recorded in 2Tracker

stereo sum) Offtapeweg in recording is the way ToTape mixdown.
Pro track need 2 channels.
Mixer concepts: Split

console:
Edit the mother of all mixing consoles 1 signal per channel strip. Solid
Subgruppenmaster in the master module. You need at least twice as many
channels as traces. Because one which intercepts communications can not
recall in another channel of the track.
inventor Rupert Neve
Inline console :
modern and expensive Mixers 2 signals per channel strip

processed. no fixed Subgruppenmaster, each channel can be
Subgruppenmaster.
Splint console:
A mixture of inline, Split consoles. A split console with

inline features. 2 signals per channel strip fixed subgroups
in the master module no sandwich construction in Split
consoles.
Page 16 of 466
Mixer Status:
Sandwich construction:
Inline consoles
Splint consoles
Line input Tape Return

Tape In
Upper path Channel path Lower

Path monitor path
1. Recording status
. 2 Mixdownstatus
. 3 Input / reverse status
. 4 Fader Reverse ISR:
Insert send and return (effects loop)
Usually a female, but there are also two possible TIP = Send;
RING = Return Patchbay:
All inputs and outputs in a professional recording studio are on it. Central switching unit. There are jack
and TT-Phone Patch panels. In a patchbay are below the inputs, the outputs above. If two points are
superimposed are connected, it's called normalizing. Fully Normalize: In and Out Brake half
Normalization: In Brake, Out Split
Normalization:
Full normalization normalized: Input / Output
Brake
Internal connection is aborted.
half normalization semi normalized: Input Brake /
Output Split
Signal is split in the output and input is disconnected. By default,
ISR is Halbnormalisiert.
Non-normalized:
no normalization, input and output are free for patching.
Page 17 of 466
The Mackie 8-bus
How many
channels
16
subgroups 8 stereo
bar
2
Splint console
- 2 signals per channel strip

- fixed subgroups in the master section, the desk is 8
professional Modular
is the very top of the Mic In:
8 Channel Strips lie on a board

XLR, balanced, ground, Hot, Cold Plus to
ground Short Mic / Line In Mic In switch
mode when not pressed.
Line In:
Jack, balanced, Tip Ring Sleeve

Hot, Cold, mass
Direct Out:
Post Mute and Pre Panorama Post Fader,

unbalanced jack output
+ 4dBu
ISR:
Insert Send Return:

Unbalanced input signal
Unbalanced Balanced jack
Tape Returns:
Balanced TRS switched in

blocks of 8
- 10 dBV / + 4dBU
Phantom Power:
Phantom power switched in blocks of 8. Turn on when

the meter has been turned down. Use only on Mic In.
Trim / Input Gain:
level setting signal to unity gain. Mic In +10

DBu Line In 0 dBU
pad:
Lowering of the signal at -20dB does not

have the Mackie.
Flip:
Status change of the console in 2

different mixer status.
Page 18 of 466
-- T o n t e c h n i k K o m p e n d i u m - -
pressed Flip
Flip ungerckt
Record status
Mixdownstatus
Aux Send section:
- Total 6 aux sends (mono)

-p e r c h a n n e l o n l y 4 a u x s e n d s a v a i l a b l e .
-3 a u x s e n d s a r e s t e r e o
Aux Sends Out
1/2 jack balanced 3-6 jack

unbalanced (+4 dBu)
first 2 potentiometers are Aux sends 1.2. Are regulated

- firmly to +15 in the channel path. Pre Switchable,
usually post fader for dry signals. Monitor Mix B
the next 2 potentiometers control Aux Send 3/4 OR

Aux Send 5/6 Shift switch
unpressed 3-4
depressed
5-6
Pre: Turn Aux Sends 3-6 when pressed in Pre. Source in the aux
send
decides whether aux sends 3-6 are in the channel path or in the
monitor path. unpressed
channel path
depresm
s eo dn i t o r p a t h
E Q s e c t i4o nB :a n d E Q
influence 4 different frequency ranges

1.Band
first
1.
potentiometers
Controller:
Cut
are
/
the
Boost
High
Mids
Controller
2.Regler: Center frequency 500 18KHz
3.Regler = bandwidth Q factor adjustable. left 3

octaves right 1/12 octave 3 controller for 1.Band
Bell Filters
2.Band
1.Regler low / mids cut / boost / -15dB
Page
19
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466
- Bell Filters
- no band width control, fixed Q of 2 octaves

-Semi Parametric
2. controller = Center frequency 45 Hz - 3 KHz
3. tape
1.Regler = High Shelf - <cut / boost controller
shelving filter
Limiting frequency is at all frequencies from 12 KHz
4. band
1 controller = Low Cut / Boost Low

Shelf EQ
Cut-off frequency at 80 Hz
all frequencies from 80 Hz to edit. Switch EQ In =

Switches the EQ on and off Low Cut
passive filter
Low-cut filter
Attenuation of the impact noise from 75 Hz he cuts
off with a slope of 18 dB per octave low.
Mix B section:
Pan pots of the monitor fader volume control for
monitor path level. In Control Room (interception
Matrix)
Mix B switch to Monitor Path listen. Signal return from MTK (Offtape)
Split EQ:
Sets the High and Low Shelf from Channel path in the monitor path directly active during
switching.
Source in Mix B:
Sets the pre-fader channel signal to the mix as from the mix B I make a
pre fader aux send signal. You need a monitor path. The return route is
changed. It comes to nothing, it is sent. but pre-fader signal.
Channel Fader section:

Record status
2 Tape signal
mixdown Offtape
Page 20 of 466
Pan Potis
2 LED's Red LED's OL Overload -> Lights when Channel
overridden. Green LEDs

- 20 dB Illuminates when the channel strip signal or at least
- 20 dB is applied. just
behind the input gain
behind the EQ after the
channel fader
Checks if signal is present or not.
systematically search error: from back to front in the signal flow

Solo Button:
Solo LED
AFL stereo solo
Mute switch:
Fader to 0
Pan left or right EQ from Solo
yellow LED
Signal flow is interrupted signal is grounded. Pre Fader Aux Sends

are not affected by the fader. Post Fader already.
Channel Fader:
100ml Fader at Mackie are logarithmically scaled to 0

affects a resistance at 0 nothing happens above 0 acts
level translator
The 5 buttons next to the faders are routing to subgroups of from channel. Depends on fixed
channel fader. In Mixdownstatus impossible. A pairwise routing a button for 2 buses work function
the Pan knobs.
Master Module:
ISR input
Each bus has 3 physical outputs
1,9,17 jack symmetrical switchable between +4 dBu or -10dBV 3 outputs eg
1,9,17 parallel to. Triple occupancy of buses.
The signal can be routed from the bus in 3 different tracks, depending on which
track is armed 1,9,17.
Page 21 of 466
DRB Dicker Routing Button;

specifies the bus on the stereo track.
Routet the bus to stereo sum. odd buses
on links just buses on law
Assign Mono L / R:
Sets the signal in addition to the unused side of the stereo track. Depending on
DRB DRB must be pressed.
Master section:
Master fader
stereo
ended signal with symmetrical bush 2 physical outputs ->

Balanced XLR
unbalanced jack
balanced XLR 2 Tracker unbalanced jack Safety
copy Main meters
only reliable meters
Master Fader is always set to zero. At 0, no work,
except for the fader Out.
Talkback section:
Voice communication from the director to the people in the studio. Talkback
Mic is set in the Talkback section. 4 button You can route the signal to 4
different ways.
1. Phones and Studio:
Talkback signal is applied to the internal state headphone and aufn Studio Out.
Tape submaster:
Sets the talkback signal to the 8 subgroups. Aux Send 1/2 Talkback
signal goes to aux send 1 or. 2
Page 22 of 466
Master Solo Section:
Master for solo collection. Solo

Level.
Solo and not solo should be identical, because we perceive more at louder signal highs and lows as
centers. Fletcher Manson curve LED Put Solo Light
Monitor Section:
Control Room:
adjusts the volume that one of the monitor system has. Studio
boxes Poti
Monitor speakers CTRL room monitors buttons

decide what one is listening: L / R Mix
actually unnecessary .Abhre ToTape from L / R mix. Mix-B

Monitor system Offtape in Record Status
2TK 2TK in unbalanced jack input.

Offtape mixdown.
External
unbalanced jack input for comparison of reference titles Mono

To check the mono compatibility
Stereo appears somewhat comical if left other phase is as pretty. The angular collapse left
and right and the mono signal is maintained.
Phones section:
headphone amplifiers outputs

unbalanced volume control
Potentiometers (listen to the signal of the musician from)
monitor The monitor section is routed to the headset. Mix-B Sets Mix-B in
the headphones Aux Send 3/4 or 5/6 in the headphone line External
sets the Externalsignal on the headphones.

CD track to a monitoring system of the musician.
Page 23 of 466
Mix B Master:
Master Level for Mix B busbar - <jack unbalanced Pre Fader Aux Sends in
stereo in Mix B Mic or Line In are the channel path is the tape return. Assign
Mix B to L / R Mix Sets the Mix B to the L / R Mix
Doubling of the input channels

Aux Send Masters:
Solo: Aux Send Solo interception and controlling. To

check the signal flow.
Aux Returns:
nothing with aux sends.

Additional business routes. Are Line channels with less features additional
channels with signal line.
are then used if no channels are free.

- has 6 stereo aux returns
- has 12 mono signals
Inputs all jack unbalanced
5/6 have only volume control and go to the stereo master
they can listen to solo.
3/4 can also be routed to the Phone section. Half in addition they have a
balance control
Pan is a mono signal that I differ on 2 rails according chic (input-side control) Balance
If a stereo signal with the output-side control

Left or right, the volume is raised or lowered. Are additional routable. Signal
can be routed out in subgroups.
16 Channels 16 tape
returns 6 Stereo aux
returns 8 subgroups
52 signals can be mixed together
Page 24 of 466
Physic Basic
Audio engineering transmission chain
Sound source -> microphone (transducer) -> transmission medium (sound recording and disclosure, eg
PA or radio) -> speaker (acoustic transducer) -> handset
Fundamentals of Vibration Theory

Definition vibration:
Operation in which a physical quantity in style changes that it resumes the same value after a certain
time.
For periodic
Vibration running resizing in each time zone from always in the same manner.
An acoustic vibration is a mechanical oscillation: A mass moves to the rest position.
Causes of Mechanical vibration:

Restoring force
Mass inertia
Graphical display of the vibration
Vibration representation over time variables
of a vibration
Elongation (y) [m]
O Vibration rash. Time-dependent size. Movements to one side
get positive values moves to the opposite side negative values.
-
Amplitude (A) [m]

O Maximum vibration rash. Distance of the turning point for
rest position
Vibration period (T) [s]
Frequency (f) [Hz] = [1 / s]
O after repeated times to the shape the course of the vibration
O Number of vibration cycles per second

OF=1/T
T=1/f
O For example, 0.01 s = 1 / 100Hz
The shorter the period of vibration, the higher the frequency.
Page 25 of 466
Harmonic oscillations (sine waves)

Vibration on the projection of a circular motion on a flat surface
can be attributed
Phase angle (phase)
-
0 and 180 in the passages through the rest position

90 and 270 when reaching the reversal points
General vibration equation of the harmonic wave is used for calculation of the
elongation at a given time [t]
360 is used instead of 2
point P
y
2 = 3.1415
sin ( ) = Y / r
= phase angle
sin ( ) = Y / A
r = Amplitude
y = A * sin ( )
y = elongation
= ( 360 / T) * T
PM
Depending on the time the indication of the phase angle is given.

y = A * sin ((360 / T) * t)
provided that the oscillation with the phase 0 starts.
y = A * sin ((360 / T) * t + 0)
o zero phase angle

(The phases at the beginning of the oscillation)
Y = A * sin (360 * f * t + 0 )
Evaluate the elongation by the phase angle A = 0.01 m f =

100 Hz t = 2.4 s
= 90
Y = 0.01 * sin (360 * 100 * 2.4 + 90) Y = 0.01 m
Page 26 of 466
Steamed vibration:
In an acoustic vibration part of the sound energy is converted into heat through friction. The resulting
decrease in the amplitude corresponding to the following course:
A3
A2 = A3 = A4 = K = damping ratio
A1
A2
The greater the damping of the oscillation, the greater the damping ratio.
Superposition of oscillations:
Practice:
Superposition of sinusoids
1) partial oscillations with the same frequency f and the same amplitude A of the same phase .
Resulting vibration:
Amplitude A is twice the size, same frequency f, same phase .
resulting
vibration
2 partial oscillations
2) 2 partial oscillations with the same frequency f, the same amplitude A phase opposition both
oscillation having phase difference of 180
There is no resulting vibration extinction position

Page 27 of 466
2 partial oscillations
3) 2 partial oscillations with the same frequency, the same amplitude but out of phase:
2 partial oscillations with the same frequency and the same amplitude, phase resulting vibration
same frequency
The further the phase shift in direction 180 is goes, the smaller the amplitude of
the resultant vibration.
At 90 Phase shift, the amplitude A of the resultant vibration by the factor
(square root of 2) is higher than the amplitude of the harmonics.
At 120 Phase shift, the resultant oscillation the same amplitude as the
vibration part.
At 2 oscillations at the same frequency, the resulting vibration otherwise at different
frequencies is a sine wave, not a sine wave.
Beats:
If the superposition of 2 oscillations with a maximum of 15 Hz frequency difference so arising
beats.
The frequency of the resulting vibration is located in the middle between the frequencies
of the part cycles:
1 fff r +
2
One hears a volume variation, the frequency of the beat frequency (f s) is
specified.
Page 28 of 466
fs= f1 - f2
Example:
fr
f 1 = 444Hz
f 2 = 440 Hz
= + = 442
2440
fs = 444-440 = 4 Hz
444
Harmonic analysis (Fourier analysis)

Decomposition of a complex vibration in harmonic components. Free oscillations and
forced oscillations own:
When a vibratory system is vibrated by a single pulse in vibration, it oscillates at a frequency (f), which
are called natural frequency. This depends on:
-
the restoring force:
the crowd:
- > the stronger the return force, the higher the frequency (f)
- > the greater the mass, the lower the frequency (f)
forced:
If an oscillatory structure (resonator) by a force acting periodically (excitation) is vibrated, it resonates
with the frequency of the excitation (excitation frequency).
The frequency at which a resonator would swing if you'd put him through a single pulse in vibration, is
called resonance frequency. The closer the excitation frequency is at the resonant frequency, the greater
the amplitude of the resonator.
The stronger the resonator is damped, the lower is its amplitude in the range of the resonance frequency.
modulation
Continuous connection of the physical variables of a vibration in the rhythm of another oscillation.
- carrier
=
- modulation =
-
High-frequency signal
Low frequency signal
Amplitude modulation (AM)

The shape profile of the modulation is converted into a change in amplitude of the carrier.
Page 29 of 466
Frequency modulation (FM)

The shape profile of the modulation is converted into a change in frequency of the carrier.
Page 30 of 466
Fundamentals of wave theory

Process of propagation of a physical condition in a carrier medium transverse wave
(transverse wave)
The oscillation direction is perpendicular to the propagation direction. For example, water wave. In this
wave to wave peaks and troughs alternate.
longitudinal wave
The vibration direction corresponding to the direction of propagation. For example, air-borne sound. This
wave is the compression and dilution alternate.
Longitudinal wave represented as a transverse wave
Graphical representation of a wave
Representation of elongation (y) over the distance from the shaft center (transverse wave) sizes of the shaft:
elongation
(Y)
amplitude
(A)
frequency
(F)
wavelength
()
[M]
[M]
[Hz]
[M]
o ( Distance between adjacent particles with the same phase ())
velocity of propagation
(C)
[M / s]
O Speed with which a certain oscillation phase in space
spreads. (Eg the speed of a wave crest or the zone of compression)
speed of sound
is the velocity of propagation of the sound. The
Speed of sound in air at 19.4 C: 343m / s
Page 31 of 466
Depth frequency (f) = high wavelength ()
c = f
High frequency (f) = small wavelength ()
Example: c = 343m / s, f = 150Hz, =
1 5 0
3 4 3
22,9
=
mm
interference
Interference is referred to as the superposition of the waves with the same

superposition of waves can be attributed to the superposition of oscillatio
It is formed from an interference pattern in which there are fixed zones of
wave
Special case of interference when 2 waves converge at the same frequency

vibration movement and wave nodes with erasures alternate.
Standing wave by reflections

If the distance between two parallel walls to each other one-half wav
multiple thereof, so there is a standing wave due to reflections. Examp
4 m Wall distance
1. Standing wave in 4m = =
8m
f =
4 ,88 7 5H
8 3 4 3
Huygens'sches principle
Each point of a wave can be the starting point of a new wave. diffraction
When a wave hits a barrier, behind there is no sharp shadows, the wave is
obstacle. The greater the obstacle, the worse does the diffraction.
If the obstruction is at least 5 times as large as the wavelength, no diffrac
Wave normal of the reflected wave

1 = 2
1
angle
of
incidenceangle
of
reflection
Page
32
of
466
Angle of incidence = angle
refraction
During the transition of sound in a medium low speed of sound (c) carried out a break to the axis of
incidence. During the transition of sound in a medium with a higher refractive c a from the axis normal
carried away.
Waves: spherical
wave:
If the sound is radiated evenly from its sound source in all directions, are formed
spherical waves. Sound field = Each room with sound environment.
All sizes of the sound field are constant on spherical shells around the sound source.
Plane wave:
Sound radiation occurs only in one direction.
All sizes of the sound field are constant in planes transverse to the direction of propagation. With increasing
distance of a sound wave from the sound source, the spherical wave gets increasingly the form of a wave
levels.
acoustics
Acoustics is the science of sound
Disciplines of acoustics:
1. Physical acoustics
Physical fundamentals of acoustics, sound generation, propagation of sound, sound field quantities,
sound energy quantities.
. 2 Technical acoustics
Electroacoustics (techn. Sound generation, sound transmission, noise measurement), room and building acoustics,
flow acoustics
. 3 Physiological acoustics
Page 33 of 466
Structure of the vocal apparatus and hearing, Bioakustik

. 4 Psychological acoustics
Sound perception, perception of loudness, pitch perception, spatial hearing

. 5 Musical acoustics
Sound systems, interval training, instrumental acoustic
classification of sound events
clay
clay mixture
sound
sound mix
noise
pop
Volume -
pure sine wave clay

mixture
two or more tones sound
A sound is a clay mixture, wherein the frequencies of the individual partials are integer multiples
of the frequency of the lowest note
100 Hz
root
1. Harmonic ( Partials o.
Partial tone)
200 Hz
1. overtone
2nd harmonic
300 Hz
2. overtone
3rd harmonic
400 Hz
3. overtone
4. harmonic
500 Hz
4. overtone
5th harmonic
Octave 1:
2
Quinte 2:
3
Fourth 3:
4
Gr. Third 4:
5
sound mix
Superposition of several sounds
noise
-
A mixture of many tones of varying frequency and varying intensity. It has a characteristic
(Rhythmic articulation)
Page 34 of 466
White noise and pink noise

White noise
same sound power in frequency bands with absolutely the same width (for
example in the range of 100 to 200 Hz and 1000 to 1100 Hz) pink noise
It contains equal sound power in the same musical intervals, for example in the
range of 100 to 200 Hz and from 1000 to 2000Hz. Equivalent to an octave.
Broadly equivalent to a music signal
pop
An oscillation of short duration and high amplitude
Classification of sound sources
1. Oscillating plate
intrinsically elastically
eg pools, bell
2. Vibrating Membrane not intrinsically elastic as drum
3. Swinging rod
eg triangle, xylophone
4. A vibrating string
not intrinsically elastically eg guitar
5. Oscillating air column
for example flute, organ etc
Sound field quantities
Speed of sound (c) In gases slightest

c In liquids higher c In solids, the
highest c For air applies: at 0 C
331.4 m / s
+ 0.6 m / s per C
at 20 C = 343.4 m / s
Argon (inert gas) 319m / s water at

15 C 1498 m / s concrete
4500 m / s
maple
lime
5100 m / s
Glass
4560 m / s
4500 m / s
Sound velocity (V) [m / s]

Specified as an effective value
Xeff
Vrms
2
v
general
for V
with the move, the speed, the vibrating particles of the carrier medium to the resting position.
Page 35 of 466
Sound pressure (p) [Pa] (Pascal)
Pressure fluctuation between condensation and rarefaction in the propagation medium. An alternating
pressure is superimposed on the atmospheric air pressure. 1 Pa = 10 Bar
Supersonic flow (q) [m 3 / s]
Volume of air that is moved in alternating directions within 1sec an area of 1 square meter. Q = v S
(Sound velocity surface)
Sound energy quantities

Acoustic energy (W) [J] (Joule)
The energy that has to be applied for the generation of the acoustic vibration.
Sound Power (P) Power [W] Watt

The sound energy which is emitted in one second from the sound source. violin
=
10- 3 W
Trumpet
timpani
0.3 W
10 W
Sound intensity (J) [W / m 2]
Sound power at 1 m 2 its area is
Inverse Square Law
A doubling of the sound pressure represents a quadrupling of the sound intensity (to the square).
In a spherically radiating sound source a doubling of the distance results in a halving of the sound
pressure, but only of the sound intensity.
J1 ~ r
~ rp 1
r = distance from the sound source
Sound energy density
The sound energy density, which also eliminates a cubic meters of space Symbol: w
unit:
J
m
Page 36 of 466
Doppler effect
cf
moving a sound source to a listener, so coming to the signal at a higher frequency than is radiated.
moves the sound source from the listener away, is the
Frequency the listener deeper.
Comb filter effect

If the distance of the sound sources to each other n corresponds, there is an in-phase
superposition and thus to an increase. If the sound source distance from each other + n corresponds
there is anti-phase superposition and thus for cancellation.
Page 37 of 466
Example:
The distance between two speakers to each other is 1 m. Compute

the first 3 extinction frequencies !!!
m =
11
2
cf
=>
=>
21 m =
340
2
Hz f
=>
2m =
=>
f = 170Hz
1. extinction at 170 Hz
2. Extinction:
m =
1 1,5
=>
1
1
,5
m =
=>
0, 67
m =
m =
=>
0, 4 m
340 = Hz f = 510 Hz
, 67
0
3. extinction
m =
2 1,5
=>
1
2
,5
340= Hz f = 850 4Hz

,0
The distance between next boost and cut is equal to the first extinction rate.
Units:
10- 12
10- 9
10- 6
10- 3
10- 2
10- 1
Pico
Nano
Micro
Milli
centi
Dezi
10 12
10 9
10 6
10 3
10 2
10 1
Terra
Giga
Mega
kilo
Hekto
Deka
Page 38 of 466
Level statement:
Def level.:
It is a logarithmic and thus belonging adapted, expressed ratio of quantities of the same unit.
Example of a just perceptible change in volume:
1000mw
+ 260mW
1260mW
+ 26%
(Time 1:26)
+ 322mW
+ 26%
1588mW
Example of a doubling of the volume

1mw
* 10
= Twice as loud
10mW
* 10
100mW
Aim:
A multiplication by a fixed value to be replaced by an addition with a fixed value.
Multiply and divide using powers 100 * 1000 = 100000 * 10

10 = 10 5
Multiplication of powers with the same base can be replaced by adding the exponent 1000:
100 = 10 10: 10 = 10 1
Division = subtraction
power laws
n
0 =0
Page 39 of 466
0=
1
+ NMNM
= aaa
:
1
= aaa
-
= aa
NMNM
nn
10 10 1-
20
31
(N> 0)
aa
nn
10 10 10
- 2
27
= 27 = 3
Logarithm (logarithm)
n=
10
= log
10
bn
100 = 10
2
10 log10=100 log= 2
10
10 3
10
= =log
= 1000 log1010 1000= 3
Logarithmusgesetze
[
lg10
n=
n]
10 lg 1
1 lg 0
==
lg (
lg+ lg NMNM
10 lg100
(
lg+10 lg100
= lg 10 lg+1000100
lg
1
lg (
= lg 10 lg +10 lg 10
lg 1000
= lg lg- 100100
1000
= lg- lg NMNM
lg=10 lg -10 lg 10
lg 100 lg 1100
0
(1 lg)
=- lg nn
=
lg 10
= lg
- 1010
- 2
lg =n lg unu
=lg 10- lg 10
lg 2 lg 10
102
=
Page 40 of 466
Level (Level)
lg
100
mW mW Bel
L
Bel
1
Bel
10 lg
= =1=Bel
10
lg 10
mW mW LdB
lg
10010
10
= =10
dB
General level formula

=
10
lg L
xx
dB
mW
10 mW L dB
2 lg1
lg 10
2 dB
= =3=dB
For power and intensity levels apply: doubling

the value of the size:
Tenfold increase in the value of Size:
Halving:
1/10:
+ 3dB
+ 10dB
- 3dB
- 10dB
relative level
=
10lgxx L dB
X1 is here the reference value
The reference value is not fixed, it can be freely selected. Used to display
resizing. Absolute levels
eg
10
lg 10
1
mW mW
dBm
The reference value is set.

dB is supplemented by an additional, indicative of the USed reference value. Here
m.
Page 41 of 466
Sound power level:
at the relative sound power level is P 1 not fixed:

LP
= 10
lg
PP dB
in absolute contrast, already:

L SWL
lg 10
10
- 12
WP dB
The reference value 10- 12 W is the sound power, which must account for a m surface, so that a 1 kHz
tone is just audible. Examples:
What is the difference in level between 2 sound power of 5 x 10- 8th W and 2 x 10- 2 W?
= 10 2
lg10
LP
5 10
- -8 2
dB WW
= 56 dB
At power levels and intensity levels tenfold applies corresponds +10 dB and a doubling corresponds to
+ 3dB.
Which sound power level corresponds to the maximum sound power of a kettledrum of 10W?
SWL
- 12
lg 10
10 10
WW
dB
13
10 lg 10 SWL dB
= 130 dB
SWL
SWL
electric power level: mark: P

unit: W relative level:
LP
absolute level:
= 10
lg
LP
PP dB
lg 10
mWPdBm
1mW reference value = 10 3
Specified in dBm (m for milliwatt) Example:
An amplifier delivers an output of 40W. Which electric power level in dBm corresponds to?
LP
mW dBm
W
40 10
1 lg
= 10 4
lg10
1 10
dBm WW
= 46 dBm
Page 42 of 466
Sound intensity level: Unit:

J
m
WSPJ
relative level:
LJ
= 10
lg
2
JJ
dB
1
absolute sound intensity level:
L SIL
m WJ
dB
- 12 2
10 lg 10

SIL
Reference: 10- 12 W / m is the sound intensity in a 1kHz tone is just audible. (The index SIL stands for:
Sound Intensity Level) Examples:
A bass speaker emits almost Spherical. He has an efficiency of 2%. The supplied electric. Power is
250W. What is the sound intensity level at 4 m distance from the speaker? required formulas:
ak
=
4
OK
L SIL
for the efficiency
100%
PP
el
for the spherical surface
m WJ
dB
- 12 2
10 lg 10
SIL
Calculation of sound power:

=
ak
ak
100 250% 2 %
WP
2 250%
=
100
= 5%
WWP
Calculation of the sphere:

4 (4
O=) =
16
mm
4 = 201 .
062
m
Page 43 of 466
Calculation of sound intensity:
= =
SIL
= ,
0 062
201 . W m WSPJ
0 lg 10
0249
10
- 12
0249
m
dB m Wm W
= 104 dB
SIL
SIL
SPL:
ls PL = 20lg (p2 / 2 * 10 ^ -5 Pa ) dB SPL
The reference value 2 * 10 ^ -5 Pa is the sound pressure at the 1Khz sound is just
audible. Specified in dB SPL
Doubling of the sound pressure is four times the sound power. Linear Size
It is p = P 2 high / Z 0
p = sound pressure,
P = sound power
Z 0 = Acoustic impedance (resistance to the air

sound propagation
opposes.)
L = 10lg ((P 2 high 2 * Z 0) / ( P 1 high 2 * Z 0) ) dB L P = 20 lg (P 2 / P 1 ) dB SPL: Sound Pressure
Level
Noise voltage level:

L u = 20 lg (U2 / 0.775V ) dB U
Specified in dB U
L V = 20 lg (U2 / 1V ) dB V
Specified in dB V
It is P = (U 2 high) / R
U = voltage
R = resistance P = electrical power
Linear size
The reference value 0.775 V is the voltage that must be expended so that a 1KHz
tone is just audible. Specified in dB U U represents the voltage
The reference value 1 V is the voltage that must be expended so that a 1KHz
tone is just audible. Specified in dB V
v represents the reference value V 1
For sound pressure and voltage level applies:

Page 44 of 466
Doubling the volume with + 6dB tenfold increase

in volume with +20 dB halving - 6dB 1/10 = - 20
dB
Digital level:
In a 16-bit recorder 10000 voltage levels are required to represent a signal.
On how much dB FS the signal is activated? L FS = 20 lg (M / 2 high (n -1)) dB FS
L FS = 20 lg (1000/65535) dB FS = - 16 dB FS
L FS = Level Full Scale
Conversion of the level in the corresponding physical quantity:

For information in Bell L = lg
(X2 / X1) Bel L = lg (X2) - lg
or L = lg (X2 / X2) Bel
(X1) Bel
10 high L = (X2 / X1) X2 =

10 high L * X1
For square sizes apply:

L = 10 lg (X2 / X1) dB L / 10
= lg (X2 / X1)
10 high (L / 10) = X2 / X1 X2 =
10 ^ (L / 10) * X1
Conversions
sound power
dB SWL W
(P)
sound intensity
dB SIL W / m square. (J)
Electric power level:
dB m W
(P)
For linear sizes apply:

L = 20 lg (X2 / X1) dB
Page 45 of 466
10 high (L / 20) = X2 / X1 X2 =
L / 20 = lg (X2 / X1)
10 ^ (L / 20) * X1
Conversions
SPL
sonic strain
Digital level
dB SPL
dB U
Pa
(P)
(U)
dB FS n (number of bits)
IRT norm:
Institute for Broadcasting Technology
- 9dB FS meet + 6dB (Program Level) Funkhaus level

+ 6dB 1.55 V
studio level
+ 4dB 1.228 V
Home Recording
- -10dBv 0.316 V
level
Page 46 of 466
Psychoacoustics:
Structure of the ear:

Inner ear (cochlea)
outer ear
middle ear
-auricle
-ossicles
ossicles Oval window (1st gear)
is behind
-meatus
(Average length =
2.3 cm)
- hammer
going to the atrium staircase
- Ambos
round window (2nd gear) sitting in front

of the scala tympani
-eardrum
(Area =
55 square mm)
-Stirrups with
Stapes connecting
3.2 square mm
-EardrumStretcher,
Vestibuli and tympanic

stairs helicotrema
it provides a
pressure equalization
-stapedius
-
worm gear
Eustachian
(The middle passage)
Tube is the
(Partition for timpani
connection
stairway ) basilar
to throat -Cortisches membrane
with hair cells
- Tectorialmembran cover
membrane protects the hair
cells
-Reissner membrane
(partition for atrial stairs)
Function:
Operations in the outer ear;
ear:
sound bundling
Sound changes depending on the direction of sound incidence
Ear canal:
Quarter-wave resonator
A tube to one side open and the other closed.

In a quarter wavelength fits into the tube, the air vibrates
with a maximum.
(Resonance: maximum amplitude of the resonator) Higher
resonances are at + n * / 2
Page 47 of 466
Calculation of basic resonance: 2.3 cm = .
= 9.2 cm,
3.69 KHz, the frequency with the resonance
f = 340 / 0.092 m = 3.69 KHz

Operations in the middle ear;
Frequency-dependent amplification by size ratio eardrum stapes.
- Leverage the auditory ossicles plate about 20 fold enhancement

-Soundproofing function from 80 Phon volume by the tensor tympani
At high volumes, it is attracted and compressed.
-Stapedius also provides compression at high volumes
-Middle ear is filled with air. Do the same air pressure as have the outer ear.
Atmospheric air pressure. Eustachian tube ensures pressure equalization.
Operations of the inner ear:
Einorts Resonance Theory (1863)

Herrmann von Helmholtz Theory:
The basilar membrane device depending on the frequency of a sound at a particular place by
resonance and vibration. This theory is outdated.
Traveling wave theory 1961:

Georg von Bekesy.
In all 3 courses of the ear is lymph.

There are going waves from the oval window with different frequencies at different
locations of the basilar membrane and cause maximum amplitudes.
The higher the frequency, the nearer the amplitude maximum is at the oval window.
Sound event:
auditory event:
frequency
sensation
sensation sizes
pitch
SPL
Perception of loudness
appeal
Lovely sizes
Page 48 of 466
Listening (listening area, listening area)
If the area by a sound event triggers an auditory event. Frequency range

of 20 Hz - 20 KHz.
The sound pressure range is the frequency we hear Depending
hearing threshold:
has a value of 4 Phon. dB SPL has different values at 4 Phon because you have different at
different volumes volumes to just to hear something (4 Phon) 4 dB SPL louder 1000Hz tone is
as loud as a 72 dB SPL strong 20 Hz tone. The auditory curves are at low frequencies closer
together than at high.
Curves of equal volume:

Fletcher-Manson curve
Robertson Letson curve
UCL is 90 Phon, 130 Phon (pain threshold) Phon is the unity of sensation size. At 4 Phon
the listening area begins.
Frequency-weighted sound pressure level measurement:
filter curves
CBA
C = for large volumes

B = average volume
A = small volumes
dB (A)
dB (B)
filter
dB (C)
Assigns was measured out in which filters. A filter

0-30 Phon
B filters
C-filter
30 - 60 Phon
> 60 Phon
Weave - Fechersches law:

An increase in the sound pressure of 10 dB is perceived as twice as loud.
lukewarmness:
Page 49 of 466
Indicates when a sound signal as double or half runs is perceived. Above 40 dB SPL could this
theory be confirmed below not. Sone indicates when a signal is perceived twice or half as
loud. The ebenmerkliche frequency variation is 1 dB.
Volume and tone duration:
Below 200ms tone duration leads to shorten the sound on one tenth of the original duration of a
decrease in the perceived loudness of 10 phon. After long signals, the loudness perception
decreases.
Adaptation:
Adaptation, adaptation:
The volume drops in a sustained acoustic signal within 2 minutes to 10 Phon and remains
almost constant.
The original volume one takes true again 2-3 minutes after switch off the signal.
tonality:
The tonality indicates when a signal is twice or half as high perceived. Mel scale.
Doubling the Melzahl Doubling the perceived pitch 100 Mel frequency group 1
Bark
Work 2 tones within a frequency group together on the ear a, the volume increase is less than 2
tones of different frequency group.
Ebenmerkliche frequency change:
applies Below 500Hz +/- 1.8Hz is just perceptible

Above 500 Hz applies +/- 0.35 Hz is just perceptible change in frequency.
concealment:
Masking Masking:
Belittled the sensitivity for acting on the ear stimulus with another stimulus.
Mithrschwelle: volume from the well of the slightest sound is audible.
200 Hz tone
16KHz
Page 50 of 466
16KHz
inaudible
200 Hz
audible
200 Hz tone concealed 16 KHz tone, 16 KHz tone Low-frequency inaudible sound
masked high frequency rather than vice versa.
For large volumes high tones of deep shades are more obscured than at low volumes.
Combination tones;
f1 = f2 = 500Hz
400Hz f1 + f2 =
900Hz
obscured because too close to 500 Hz and 400 Hz
| F1 - f2 | = 100 Hz
Frequencies f2 and f1 of 2 tones generate tones sum f1 + f2.

and difference tones | f1 - f2 |
Residual Effect:
Sets up together a sound when there is no tone of only the overtones, as a pitch is perceived, which
corresponds to the fundamental frequency.
The perceived pitch is called residual or virtual pitch.
Haas Effect:
Law of 1 wavefront
Precedence Effect.
1- 30 ms later sound incidence of a page means a maximum of

10 dB stronger sensation of a page.
147 ms are offset by about 15 ms. The listener then locates the stage instead of the boxes. The visual
and audible signal localization must match. Optimal delay 10 - 30 represent ms.
Page 51 of 466
Directional hearing in the horizontal plane:

Intensity differences: For lateral sound from 90 Direction is in language 7 dB level
difference
and 7- 10 dB level difference occur in music.
Timbre differences: When sound from 90 Direction applies below 300 Hz 0 dB level
difference at 2.5 KHz
10 dB, 6
KHz
10 KHz
20 dB
30 dB
Ears distance = 17.5 cm
Detour to the other ear = 21,5 cm time difference: When sound from 90 Direction is the smallest
perceptible 0,63ms skew. Below 100 Hz
no direction detection as Subwoofer

are low frequencies by themselves and can not be located.
100-300 Hz
300 - 1600 Hz
only time differences are evaluated

Duration and intensity differences
Above 1600Hz only intensity differences are evaluated.
Directional hearing in the vertical plane:
Median plane:
Timbre differences for different directions of sound incidence.

Raising of the level:
front 270-550 Hz and
3- 5.5 KHz
above about 8 KHz lift back

750 - Raise 1800Hz and 10 KHz.
Distance perception:
Volume differences
Hall differences
Timbre differences a = b =
b
a
direct sound diffuse sound
The greater the distance from the sound source, the weaker the direct sound with a constant diffuse
sound.
Page 52 of 466
Tone color difference by air absorption which is larger at high frequencies than at low frequencies.
From 10m distance from the sound source, an audible level waste manifests itself. The greater
the distance, the duller. At 1 km
-2dB 500Hz
- 4 dB 1 KHz
- 8 dB 2 KHz
- 10dB 4 KHz
- 32dB 8KHz
Speaker playback:
The listener should make for a stereo reproduction is an equilateral triangle with the speakers, there
are phantom sources between the boxes formed. The listener sitting in the middle and 30 right and left
of his head are the boxes.
Headphone playback:
Localization is in the mind, as the frequency increase for direction detection in the vertical plane is missing.
Page 53 of 466
HD Recording Basics (ProTools)

reference Sequencer
Widespread in the music and advertising production 2

different versions:
1. ProTools LE :
Native working on the computer without additional hardware. Works with
M-Audio and digi design interfaces.
2.ProTools TDM / HD system:
Requires developed DSP cards (Digital Signal Processors) Digidesign

cards only for ProTools TDM systems also require a very expensive
interface.
Plug ins: TDM
systems:
TDM
RTAS -Real Time Audio Suite
(Non-destructive, DSP)
Audio Suite
(Destructive, CPU)
(Non destructive, CPU)
LE system:
RTAS (Real Time Audio Suite)
(Non destructive, CPU)
Audio Suite
(Destructive, CPU)
BroadWave format
Is the expansion of wave files with sensitive information
Good Plug Ins are the Sony Oxford plug-ins.
Last session:
File New Session name Save Start + T

I use Facebook as a hard disk,
When you save your session, a folder is automatically created from it. track create and
name New Shit + Apple + N tracks. Signal flow in the channel ProTools:
1. input
2.Festplatte
3. Inserts (Dyn EQ)
4. Pre fader sends (headphone mix)
7. Post fader sends (dry signals)

9. output
5. Mute
6. fader
8. Panorama
Page 54 of 466
5 inserts per channel Alt + click Select Fader to Default Alt + Plug In Plug In is enabled in all
channels. (Do to all -Alt) Shift + click
individual selection
Shift + Alt + Click

10 sends
Edit selected selection together.
Pre and post fader under the 5 inserts. View Sends AE, FJ shows only the
selected Sends buses are used for subgroups. Create New Track: tap the buses Stereo track
for the return of buses Aux Input Use when buses need to be retrieved New Track Setting the
bus in the input instead of Mic / Line In
Apple + click + Solo Channel Solo Safe is no longer muted.
Hall use:
Create New Line Mono
The inserts of these track Reverb inserting 100% Wet and bus 1 input
is tapped in Aux.
The echoing to channels are routed through buses to the new track. T enlarge R
Zoom
selectable per track between waveform or volumes. Automation with

pen. F8 Grabber Move F7 Selektier Mark, 2 times click Region marked
F6 trimmer shorten piece fades extend in ProTools are destructive.
Fader range is copied and revised destructive. Apple + Fader

movement precise fader settings Keyboard Command Focus.
Regions - Window
Bold Regions include a new audio file. not bold Regions are part of an
audio file. To cut Apple + E
Clear one or more regions from the Region window Delete Shift + Apple + B
Page 55 of 466
Regions menu
Select Unused Shift + Command + U
ProTools can work some nut with monofilament
works internally with monofilament
split stereo files in 2 monofilament.
Import:
1- Shift + Apple + i
Import Files to ProTools
2- drag and drop in the window or pull. 2 types of
selections with ProTools:

1.Timeline selection
Start and end
2. Edit Selection
Connecting two selections Link Edit and Timeline Selection Taskbar Setup Menu ->
Preferences operation
Timeline Selection Follows Playback
Under Preferences ProTools are options and settings. Track height adjustable on
the right side of the Channel attitudinal Button
modes:
Slip mode
F2
F1 region gradually in the grid square Spot
shuffle
Click F3 region, the start and end of the region define Grid F4 Regions
place by clock and time reference.
Move with grabbers in the corresponding grid.
Spot Exercise:
Transient Change of the waveform in a short time, fast attack, eg Tab to Transient
Stepping the cursor to the next transients
Schneidetip: Edit
Strip silence Apple + U
Cuts in the region, the silence out, from -48 dB
Page 56 of 466
Automation:
Set Volume per curve. Bypass the
plug-ins with apple + T.
tracks:
1.Audio tracks
2.Aux inputs
Hard disk is missing, no recording, no playback create

subgroups, create dry signals.
3, Master Fader: Is a volume control for an output

no panoramic no
Sends
has inserts (only post fader)
governs each output
Interface or Bus
On the bus, a signal may be louder al 0 dB FS. no more input. Master Fader has no inputs and
regulates it so that the input is not is clipping. Save: Apple + S | Setup Pref. surgery Enable File
Autobackup
Page 57 of 466
tape recorder
sound recording :
Magnetic tape recording since the 20s used. Now on the decline.
A magnetic material is subjected to a varying magnetic field. Tape heads in the tape
machine
therein are coils
Each electrical, stromdurchfloene conductor forms a magnetic field. Kitchen sink

long coiled conductor current generated
magnetic field, depending on the current direction,
AC voltages (audio signal) are reflected in the magnetic field again. On head, the magnetic tape
is pulled long and is magnetized changeable. Changeable magnetization.
Whenever we have an electrical conductor, voltage is induced. AC voltage is stated
on tape that we can hear. Magnetic tape consists of 4 layers:
1. Back (Haftungsarm, smooth)

. 2 Support layer (tape plastic) thickest layer
. 3 Magnetic layer (very thin 10-25 microns)
. 4 Surface coating (protection against abrasion) in head is thee converted
voltage in a magnetic signal. Because the coil generates a magnetic field.

The higher the quality of the magnetic particles are smaller in the magnetic layer.
Width of the tape:

inch - 2 inch 1 inch (inch) =
2.54 cm
One can take many parallel signals
so-called tracks.
Track:
separating track
serves for spacing remains unaufgenommen
Number of tracks with tapes: 1 track (mono) 24 tracks

or 32 tracks Standard 24 tracks on 2 inch
Page 58 of 466
The wider the band, the better

magnetization more
interference less
larger scale at records
4 tape heads in the tape machine:
Machine records a timecode track for synchronization in studios a pilot tone is sent for
calibration and tinting.
Path of the tape

1. Reels (bugged)
Arm with a coil (Tension Arm) provides voltage. Band must be
sufficiently stretched, must be good at the heads Between 2 rolls.
the 4 tape heads over
(1. erase head 2 head Sync 3. timecode head 4 Repro head)
5. Tape Cutter
The drive follows then Capston & idler pulley This has a motor and
ensures the capstan. Shut off arms has a switch
says the Capston motor depending on
Tape or no tape when the Capston engine must stop. Idler pulley
recording on the 2 coil.
tape heads:
1. erasing head:
Deletes the tape track when recording a track.
2.Sync head
Used to record and playback is however

optimized for recording.
Can reproduced without transfer and record at the same place
other tracks.
3. Repro head
Plays back
Sync head
Repro head
t [s] = d [cm] / v [cm / s] t

= delay
v [cm / s] = belt speed d [cm] = distance of the heads
Page 59 of 466
Roland Space Echo:
tape speed
Professional level
Inch / inch
centimeter
3.5 inches / s
9.5 cm / sec
7.5 inches / s
19.05 cm / s
15 inches / s
38.1 cm / s
30 inches / s
72.2 cm / s
The faster the belt runs the faster you can take the highs. The smaller the magnetic particles in the
magnetic layer, the better can take the highs.
International film layer German film layer Fast forward (Tail

Out)
Rewound (Head Out)
pre-echo:
comes in before the signal is reproduced .--> arise through overlays

postecho:
begins after the signal reproduction.
International
protected side facing outside. better fast

forward the tape for storage.
German
better rewind the tape for storage. Sensitive magnetic

layer facing outside.
Cut:
Cut is made by listening with rotation of 2 coils point mark at the
playback head.
use scissors stick together
Edit unload button:
When playing the tape is transported and the deck is playing, but not wound on the spool 2.
Cut variants:
90
60
45
Weird cuts have smoother transition. not

recommended for the stereo track. One channel
begins earlier than the other.
reference:
Studer A827 tape machine
Page 60 of 466
Tape saturation:
Above a modulation limit begins the tape to oversteer and compress. The higher the recording level is
turned up, the more the volume changes not because the signal has reached the tape saturation.
(Added harmonics) The distortion ensures Exciter effect and the signal sounds warmer and more
pleasant. harmonic distortion Drum, bass, guitars Typical cases of
tape saturation
Page 61 of 466
Electrical engineering 1
Basics
What is power?
When current flow is defined as the weighted motion of charge carriers,

the latter are usually electron.
The following example explains us something about the structure of an atom. Bohr
model (hydrogen atom):
The core (proton) is positive (+) charged while the electron negative (-) is loaded. Thus the atom is
physically neutral as a whole.
The electron held by centrifugal force and by the Coulomb force (gravitation) on the track.
Both forces are equally strong by circling the electron around the proton.
Elementary charge:
C = Coulomb, As = ampere per second
Character:
1.602 10
* -
19
* = 1.602 10
19
Q elementary =
C
ace
Q is the smallest elementary charge. Any other charge is at least Q elementary otherwise the
multiples.
How much pull electrons through the conductor at 1A? Or. How
many electrons are needed for 1 A?
=*
=
1C n
* C
1, 602 10
- 19
*
6, 252 10
18
Page 62 of 466
x-
Input into the calculator: 1.602 E or EE or EXP + - 19 1
In the case of separation of the charge, a voltage which represents the current flow is formed is responsible. The
pressure at a point (potential) can not be measured, however, can be the difference in pressure measured (voltage).
Air pressure and water model
R = resistance
I = ammeter (current) U = voltmeter
(voltage)
If the conductor thereby getting water from one to another container, perfectly smooth and without
resistance we would have in the case of no resistance. By deformity we have prepared a resistance. U is
the difference in pressure measured (> resistance as high as possible). No matter where the meter is, it
will measure the same pressure difference if. Before or after the curvature
I want the amount of water per time show (l
s)> resistance to a minimum.
The stronger R (resistance) the greater U (pressure difference).
Ohm's law
U [V] = R [] * I [A] U = R * I,
U
U

, RR
I =
U: voltage ( "pressure difference"); Unit: Volt (V) I: current ( "amount of
water per time"); Unit: Ampere (A) R: resistance; Unit: ohms () to:
U=
R=
I=
a)
12 V
2.7 k
4,4mA
b)
45 microvolts
200 milliohms 0.225 mA
c)
0.384 V 120
3.2mA
Calculating method of a, b and c:
a) 12 / 2.7 EXP + 3 =
4 4 10
* -
Page 63 of 466
= 4.4 mA
* = 4.5 10
b) 200 EXP - 3 x 0,225 EXP - 3
0.045 10
* -
shift ENG =
3
CLOSELY
= 45 microvolts
c) 0.384 / 3.2 EXP - 3 = 120 caution

in EXP whether + or -
If we want to convert a smaller unit mA or milliohms in the next larger then - applied.
Conversely, if we want to convert A or in the next smaller size we use +.
Voltage (potential difference)

A voltage is generated by charge separation. Say you have the atoms take away electrons.
eg induction:
An electrical conductor varying magnetic> voltage When a conductor subjected

to a variable magnetic field as in it a voltage is induced.
Types of inductions:
(Or ways of
Voltage generation)
Chemically (battery)
Photovoltaic (solar cell)

Static charging (friction)
Piezo electronics (piezoelectric crystal, piezoelectric microphones)
Heat (thermocouples)
Effect of the electric current

- magnetic effect
- chemical action electrolysis
- lighting effect
- Piezo effect (Speaker)
Physiological effects (effects on the body):
amperage
1 mA
1-15 mA
15-30 mA
30-50 mA
about 50 mA
effect
Tingling in touch with fingers. (Recognizable)
increasing tingling, finally incipient muscle spasm
Crick in the affected person can solve the hands not by included electrical conductors
increasing muscle cramp, dyspnea, cardiac arrest if the current flow is not interrupted, 3-4
minutes suffocate dead by
Ventricular fibrillation, death by seconds or fraction of seconds
Page 64 of 466
Current types (voltage types)

DC or DC voltage:
Direct Current (DC)
Symbol:
Voltage permanently on a value 0 Hz (0
cycles per second)
pulsating DC voltage:
Symbol:
never changes direction because of this DC
(Sinusoidal) AC voltage:
Alternating Current (AC)
Symbol:
eff
UUMax
= 71% 2 from MaxU, or 3dB less
(Rectangular) AC voltage:
Symbol:
UUeff =
Max
(At the same rash between the sine and the square-wave AC voltage, the square-wave voltage is getting
stronger, because she is constantly at the maximum.) The RMS value is a constant value which has the
same effect as the corresponding change size. Example:
If one a given AC voltage to a light bulb on, one can determine that a particular DC
voltage produces the same brightness. The value of this DC voltage is called RMS
AC voltage. With the help of the effective value you can expect to change variables
such as having constant, which is much easier.
Page 65 of 466
AC
DC
Mixed voltage or AC DC offset = DC offset

(DC offset)
The upper broken line, and the lower solid are the
boundaries of the gain at a waveform representation.
The fact that the signal is bordered above already at
the maximum we can no longer signal amplifying
(clipping). For this reason, a high-pass filter is installed
at the frequency 0 to extinguish and to bring the rest
position to 0 in amplifiers.
three phase
Three-phase has 5 lines: - L1 (phase)

- L2 (phase)
- L3 (phase)
- N (Neutral)
- PE (Protection Earth)
If we connect with L1 N thus we get a voltage of 230 V. The same applies for L2 and L3, all 3 phases
form a total voltage of 400 V.
All phases are in a waveform display with 120! Phase offset twisted to each other.
Page 66 of 466
Electricity transmission from the power station to the consumer
FI switches measure how much power are ausgegebne and as much also has to come back again,
from the returning flow from outputted differs, the switch jumps to within a fraction of a second and the
current flow is interrupted.
Physical sizes current

density (J)
Unit:
IAA
[]
AJ
mm
m
(A [mm] is the cross section of the conductor)
Example:
What is the current density at a current of 16.4 A and a conductor section of 2.5
mm?
Solution:
J =
16A
2.5mm
= 6, 4
A
mm
Conductance (G)
Unit:
= S (Siemens)
R []
Example:
R=
a)
a) 35 EXP - 6 1
b) 300 EXP - 3 1
c) 5.34 EXP - 3 1
G=
28.6 kOhm 35 S
b)
3.3
c)
5.34 kOhm 187 S
300 mS
x = 28571.43 ENG = 28.6 kOhm
x = 3.3
x = 187 S
Power (P)
Page 67 of 466
Power is the ability in a certain time to do some work.

PW
Uvia
[ (W) or VA ](volt-amps)
[
[]
] =
* []
P = U * I, Ohm's law by setting up procedures U = R * IP = R * I * I or P =

R * I
div. setting method

P
PR=I* <> = <> = R

<> =
U U
PU
URI= * <> = <> =I
RR
<> =R
<> =U* <>

PRUPR
=
Tasks:
I=
P=
b)
c)
d)
,I =
b)
c)
1W
0.1A 100
10V
d)
4W
0.2A 100
20V
R
20 I
100
2700W I
300
10 I
100
= 0.1A
= 0, 2A
U=
R=
5 mW 3A
2.7 kW 3A
a)
==
9 3A
555,6 1,7mV
900V
300
=
; URI,=U* 300 3 900V
*=
;=*=*=
; = ;* == +* =
At constant resistance voltage doubling leads to power doubling, this results

Leistungsvervierfachung. Example of Verfier fication and Aufga ben:
I=
P=
300mW
b)
800W
d)
a)
b)
P 0.3 I
==
U 12
I =
0, 025A 25mA
=
800
66.1
= 3.5A
U=
25mA 480
12V
3.5A
66,1
230V
97,2W
6mA
2,7M 16,2kV
1555,2W 24mA 2,7M 64,8kV
a)
c)
R=
U 12 R
==
I
0.025
* = 3.5 230 *
; URI =66.1
480
=
*=
*
=
=
* = 0.006 97,2W
*
; PUI=16200
c) URI=2700000
0.006 16200V
16,2kV
d) Now thanks to constant resistance double the voltage and current and quadruple the
performance.
= * =; = * =; = * =
Task:
We bought a resistor labeled R = 2,2K and P = 0.25W.
a) How many volts maximum of rest?
Page 68 of 466
b) What is the power of flows?

a) U
Max
I
b) Max
PR* =
P
0.25 2200
*
23,5V
=
0, 25
2200
0,01066A 10,7mA
=
efficiency ( ) (Small Eta)

How much horsepower comes out when I admit a certain electric power?
(KleinEta)
PW
out [
][
PW
in
* 100%
Tasks:
a) A lamp has a power consumption of 120W on the abgegebne light power

is 9W. What is the efficiency in%?
b) A lamp with 230V and 800W has an efficiency of 6%. What is the light output?
Solution:
a)
b) outP
100%
* =P in
6
100
* 800W 48W
=
Work (W)
W Ws
J (Joule)
=
PW ts]
[
Task:
* []
A television consumes a power of 5W operation in standby. How much is 1 years

standby at an electricity price of 15 cents / kWh?
Solution:
W 365t
=
24s* 0,005kW
*
43,8kWh=
Costs W =Pr* ice 43,8kWh
= 0.15 / kWh
* 6.57
Task:
Solution:
Task:
How long could you operate a heater at the same cost?

W t 43,8kW
==
P
3kW
14,6h 14h36m
=
A recording studio is open 265 days a year. The devices draw 3.696 A from the power
outlet. The units are 10:00 to 03:00 in the operation. How much is it per year when 0.15
/ kWh?
Solution:
P 230V
=
3,696A
*
850,08W
= 0,85kW
W 365t
=
17h* 0,85kW
* 5274,75kWh
=
Cost 5274.75
=
0.15 *791.21=
Page 69 of 466
Task:
An open air concert takes for light 300kW, 273kW and 43kW for the PA for others.
Overall, these devices be shipped 3 days needed every 16 hours / day, the price of
electricity is 0.15 / kWh, which is free of cost?
Solution:
WP =t 616kW
*=
3 16 29568kWh
**=
Cost 29568kWh
=
0.15 * 4,435.20
=
Specific resistance ( )
(Caution: A is the cross section of the conductor specified in [mm])
mm
(KleinRho) m
[] =
mm
* lmm
[]
A mm
[
* l
RA
Specific conductivity ( )
m
mm
(Small kappa)
lm
[
[]
mm
m A mm
mm
*
[
]
Specific resistance and conductivity of materials

material
copper
mm
resistance
0.0179
56
silver
0.0159
gold
0.0222
aluminum 0.027
coal
65
63
conductivity
mm
45
37
0.015
From this table we can see that silver has the lowest resistance and the highest conductivity.
(Note: If a task, the cable length is specified then you have to note that the conductor length of the
double cable length is, round trip)
Page 70 of 466
Task:
Solution:
A power amplifier is connected with a 80m long copper cables, the cross-section of
the cable is 0.25 mm. What resistance has the cable?
* l 0, 0179 160 *R
0, 25
11, 456
This would normally accept what clearly too much for the boxes a resistivity of 4-8 . This leads to
some drawbacks.
The signal which comes to the boxes quieter thereby must turn the amp louder, ie the
amplifier delivers more power and thereby risk of oversteer too high.
-
By extreme amplifying the boxes are a frequency response received which is not necessarily
desirable.
be boxing more by a weak amplifier damaged by the excessive amplification as is of a somewhat

stronger amplifier but which clearly turned down.
Task:
A copper cable has a length of 3m and a cross section of 2.5 mm. What is the
resistance?
Solution:
R =
0.0179 6 *
2.5
0.04296
= 42,86m
Influence of temperature on the resistance
The resistance increases with increasing temperature. Through the chain of atoms can this grid hardly
get through it that electrons through the conductor. The higher the temperature, the greater the
fluctuations in the grid back and forth and the harder it is to get an electron through the conductor. We
differ in 2 different conductors.
1. (PTC> Positive Temperature Coefficient):
resistance increases with

increasing temperature.
2. thermistor (NTC> Negative Temperature Coefficient):
resistance sinks with

increasing temperature.
We also differ in 2 formulas.
Page 71 of 466
1. change of resistance:
R []=
start
[] *
Conversion table Kelvin and C [frame1]

1

[]

*
TKK

R= resistance change
= Temperature coefficient K =
Kelvin T
= temperature difference
2. Resistance at the end:
R
[] =
aim
[] + R []
start
A small table by the temperature coefficient div. Materials.

copper
0.0039
aluminum 0.0038 nickel

0.00015
Task:
The resistance of a copper conductor is at 10 C, 10. What is the resistance at a

temperature of 273K?
Solution:
R []= *
10 0.0039 10 0.39
*=
=+=
Series circuit of resistors
Physical current direction from - to + Technical

current direction from + to In a series circuit, all streams are equal, it follows ...
>
====
PU
in front
in front
* I ges
= *3V 0,2222A 666.7mW

=
= +++
Across each resistor, a voltage may be measured which is proportional to the value of the resistance.
This voltage is called the voltage drop across the resistor. The sum of the sloping of the resistors
voltage is the input voltage.
URI= *
n
=
+++
; The
ges
> includes every resistor
total value is the sum of the individual resistors
=
> * total
voltage
In a series circuit of the total resistance is greater than the largest part of resistance.
Page 72 of 466
Task:
R=
100,
2
R 1= 100,
R 3= 100,
R=
100, connected in series.
4
Total voltage = 25 V.
a) What is the total resistance?
b) What is the current?
c) What is the voltage drop on the resistors?
Solution:
a) R
b)
ges
=
400
U 25 I
==
R 400
0, 0625A 62,5mA
=
* = 0.0625 6,25V
*
c) URI=100
Voltage at)
Task:
(Drops across each resistor same
R 2= 5k,
R 1= 100,
3 resistors in series,
R 3= 2M,
U ges
= 100V.
a) What is the voltage drop of each of the resistors?
b) How much power must the resistors respectively cope?
c) What is the total power?
Solution:
a) gesI =?
I
ges
ges
100
2005100
=
0,000049872A 49,87A
U 1100
= 0.000049872
*
0,0049872V
=
4,9872mV =
U 25000
=
0.000049872
*
0,24936V
= 249,36mV =
U 32000000
=
0.000049872
*
99,744V=
b)
PUI
= 0.0049872
*=
0.000049872
*
248,72nW
1
P2 0.24936
=
0.000049872
*
12,4W=
P3 99.744
=
0.000049872
*
4,97mW
=
c)
PU
ges
ges
* I ges =
99,998 0.000049872
*
4,987mW
=
Page 73 of 466
resistors
Series resistors are always used when a consumer less power must be connected to an
excessive voltage. Sketch:
Example:
A lamp with a caption: "9V 2W" is to be connected to a 12V power

source.
Solution:
==
mpp
1.
.2
.3
.4
lamp
UUU =
in front
PU
in front
2W
in front
I ges
in front
lamp
9V
lamp
ges
in front
lamp
3V
0, 2222A
* I ges
222, 2mA I =
ges
12V 9V
- =3V
=
13.5
= *3V 0,2222A 666.7mW

=
Voltage drop in cables:

As a voltage drop in wires is referred to the voltage loss which is caused by the power resistance.
This voltage loss must be considered when the power resistor or the current is large. 10 m long copper
cable current of I = 13 A
R = (0.0179 * 10 * 2) / (1.5 square mm)

R = 0.2386
U = 238.6 m * 13 AU =
3,103 V
Page 74 of 466
Series circuit of resistors:
R1 R2 R3 R4
+
-
Current direction from - to + is the physical flow direction. Technical

direction of current flows from + to -. In a series circuit the current I is
always the same. Iges = I1 = I2 = I3 = In
Across each resistor a voltage may be measured which is proportional to the value of the
resistance. The voltage is called the voltage drop across the resistor. Uges = U1 + U2 + ... +
Un
The sum of the sloping voltage across the resistors is the input voltage. Un = R * Iges
Potential difference of the two voltages before and after the resistance. Rges = R1 + R2
+ .... + Rn
The total resistance is the sum of the partial resistors. The total resistance is
greater than the sum of the partial resistors. Uges = Rges * Iges
The real voltage source:

Ideal power source supplies the electromotive force. Each real
voltage source has a so-called external resistance (internal
resistance).
Most of voltage drops at the greatest resistance.

The voltage drop across the resistor causes a consumer may not receive the full voltage.
If the resistance of the consumer angngigen substantially greater than the output resistance so the
latter is hardly noticeable. However, if the load resistance is comparatively low, the drop across the
output resistor voltage must be considered. In audio engineering, it is almost always desirable that a
voltage applied to an output voltage (audio signal) as completely as possible drops at the input of the
subsequent device.
Page 75 of 466
This is achieved in that the input resistance of the following device is as large as possible in comparing
the output impedance of the voltage source. Man Speaks then of voltage adjustment.
Microphones dynamic or condenser have an output resistance 150-200

.
Mic preamps have a resistance of 1 k 2 .

A frequency dependent input impedance: At
altitudes it is higher. At depths it is deeper.
Microphone amplifier.
At high frequencies, high output resistance of Mics.
Parallel connection of resistors:
R1
I1
R2
I2
iges
Current can flow through 2 directions by s small resistance to flow more current than the big ones. Iges =
I 1 + I 2 + .. + In
The total power is the sum of the individual partial flows. Uges = U 1 =
U2=Un
In each parallel circuit, the voltage is the same everywhere.
With each new resistance in parallel, the total resistance decreases. The conductivity
increases. G = 1 / R
Gges = G1 + G2 + G3 + .... Gn 1 / Rges = 1 / R1 + 1 / R2 +

1 / R3 + ..... 1 / Rn Rges = 1 / (1 / R1 + 1 / R2 + 1 / R3 + .....
1 / Rn)
Page 76 of 466
The total resistance of a parallel circuit is always smaller than the smallest component resistor.
R1 = 300
R2 = 300
Rges = 149.87
R3 = 170 k
I 1 = Uges / R 1 I 2 =
Uges / R 2 = I 3
Uges / R 3
Huge voltage and very low resistance results in a short circuit.
Voltage divider:
A voltage divider is used to recover from a

larger voltage smaller.
+
Vin
-
R1
U1 greater tension
U2 small voltage
R2 + - Uaus
In a series circuit, the ratio of 2 voltages is equal to the ratio of the resistors where drop it. Vout /
Vin = Raus / clean = R2 / (R1 + R2) Vin = 1.228 V R1 = 5 k
Rges = 22k
Vout =?
Vout = Vin * ((R2) / (R1 + R2)) Vout =
948.90 mV
Vout = 20 V R1
= 100 k
R2 = 300 k
Page 77 of 466
Vin =?
20 V = Vin * (300k / 400k )
Vin = 26.7 V
potentiometer
Poti
is a variable resistor
Switch block diagram:
In reality, a voltage divider is loaded by a greater or

lesser resistance Rload.
+
Vin
-
R1
R2 RLast
If Rload is small, the resistance of the parallel connection of

R2 and R load is significantly less than R2.
Thus, Vout is also lower. We speak

because of a loaded voltage divider.
In voltage divider Vout is output to R2.

In certain cases still approximately the simple formula can also be used for the unloaded voltage
divider. Rule of thumb:
A voltage divider is considered to be unloaded when either Rload is not present at all or if Rload is much
greater than R2. about 10 times greater. Vin = 100 V R1 = 10k
R2 = 10 k
Vout = Vin * (R2 / (R1 + R2))
100V * (10k / 20k ) = 50V when

RLast to come:
Page 78 of 466
R load = 100 k Rges. = 1 / ((1/100 K + 1 / 10th of k ))
R 2 = 9.0909 new k
Vout = (R2neu) / (R1 + R2 new) * Vin

Vout = (9.0909 k ) / ( 10 k + 9.0909 k ) = 47.6 V
Charge storage:
Capacitance C how much charge the capacitor at a certain
In response to the voltage.
save power.
C [F = C / V = A
S/V=S
/ ]
unit
symbol C
Charge per volt
physical flow direction in the case of -
+:
Dielectric Isolated two plates apart.

Prevents sparks umspringen.
The charges are charged by the current of electrons to protons. Q [C] = C [C / V] * U [V]
Charge (Coulomb) capacity (Coulomb / V) Voltage (volts)
C [F] = o [F / m] * r * A [square meters] / d [m]
The electric field is constant o = 8.85 * 10 ^ -12 [F / m]
r = 1 = the material between the plates.

A = (d * C) / (
o * r)
Time constant Tau:
If the time constant that describes when a capacitor is fully charged or discharged.
Page 79 of 466
T = R * CT [s] = R [ ] * C [s
/ ]
After a thaw, a capacitor is charged to 63%. After 5 Tau, a capacitor is charged to

100%. the charges remain in the condenser only by the voltage supply. The
current is discharged from the condenser.
As soon as the capacitor at 5 Tau is 100% full, it is empty again 100% at 5 Tau.
The capacitor represents a frequency-dependent resistance. The faster changes the

flow direction, the greater is the current flow.
Ie, the lower the resistance.
The capacitor thus has a large low-frequency and high frequency low resistance.
The frequency-dependent resistance of the capacitor is known as the capacitive reactance.
capacitive reactance Xc:

Xc [ ] = 1 / (2 * * f [Hz] * C [s /
it frequency dependent resistors can be constructed frequency-dependent voltage

divider. These may be used as a filter or crossovers.
Principle: In a series circuit falls to a large resistance a

Voltage.
Thus, if the reactance is large at a particular frequency falls on the corresponding
component from a large part of the input signal. RC element 1st order
The tension is high at high frequencies and low at low frequencies. Lowpass.
fg [Hz] = 1 / (2 * * T [s])
Page 80 of 466
Instrumental Acoustic:
- a portion of the musical acoustics
- Science of acoustic observation and study ways of musical instruments
Classification of musical instruments according to the type of acoustic excitation oscillation:
1. AeroPhone (air Klinger):
A column of air, which is located in a cavity is, by blowing vibrated (eg: trumpet, trombone, tuba,
recorder, flute, clarinet, oboe, bassoon, saxophone, pipe organ)
2. Chordophone (strings Klinger):
A taut string is by striking, plucking or painting vibrated (eg: guitar, banjo, mandolin,
violin, viola, cello, double bass, grand piano, upright piano, harp).
3. Idiophone (self Klinger):
An intrinsically elastic system is by hitting vibrated (eg triangle, cymbals, bells, cowbell,
xylophone, xylophone, marimba, vibraphone, claves)
4. Membranophones (fur Klinger):
A tensioned membrane is vibrated by striking (eg toms, snare, bass drum, bongo, conga,
timbales, drums)
5. Electrical Phone:
a)
electromechanical instruments (traditional instruments amplified

electronically (eg: E-Bass, Guitar))
b)
electronic musical instruments (newly designed instruments with electronic sound

production (eg: Hammond organ, synthesizer, theremin))
Page 81 of 466
Time structure:
The ADSR (Attack, Decay, Sustain, Release) curve (level characteristic of the sound)
1. Attack (transient effect):
The time from the tonal onset to reach the -3dB point of maximum amplitude (hard
stops, and high notes have a short attack time; soft Anschlage and low notes have a
long attack time)
2. Decay:
A drop in level after the amplitude maximum at the end of the attack phase to the sustain level
3. Sustain:
The area in which the sound of little or no changes subject (eg tremolo, vibrato)
4. Release (Auskling- / release time)

Time from the end of the sustain phase to a drop in level of 60dB compared to the
sustain phase
Page 82 of 466
Frequency structure:
Hllkruve
frequency
spectral
Line spectrum:
graphic representation of the strength of each harmonic. Depends on the note pitch
Envelope of the line spectrum:
Compound of the peaks of the spectral
formant:
a frequency range in which the volume is raised (for example: voice, instrument). Fixed frequency
ranges,
regardless of the fundamental pitch.
based
Instrument Typical resonances.
on
Sngerformant:
at about 3 kHz, the voice gives sustainability and assertiveness

more formants:
Bass Drum
Snare
800-1000Hz
Electric guitar
500-3000Hz
E-Bass
60-100Hz
100-500Hz
Dynamics:
The range between the lowest and the highest sound level, the instrument can achieve.
Page 83 of 466
Dynamic basic steps in music:

-
(quiet)
piano
forte
(according to)
mp
mezzopiano (medium soft)
mezzo forte
pianissimo
fortissimo
(Medium loud)
mf
(very quiet)
pp
(very loud)
ff
in the correct order: PP, p, mp, mf, f, ff
Polar pattern:
the higher the frequency, the stronger (more precisely) the directivity of
The direction factor :

instrument
p
=
pp0
p0
maximum value is 1, a value above the fracture stroke can not be greater than the bottom
eg
0=
1=
Pa p
2 Pa p
==
0
Pa pp
Pa
0=
2 1 ,5
this indicates the factor by which the sound pressure at an angle is different to the main axis of the sound
pressure on the major axis
The direction of degree 2
:
instrument
J
=
JJ
0
J0
this indicates the factor by which the sound intensity at an angle is different to the main axis of the
sound intensity on the major axis
Page 84 of 466
Directivity D:
logarithmically expressed direction factor
20
= lg
Level difference between sound pressure level at an angle and SDP on the major axis
directivity factor :
comparing the acoustic performance of a non-directional and a directional sound source
directional
ung
PP
ger
Non-directional
Ratio of the sound power of a hypothetical sound source that emits non-directional, the actual sound
source radiating directed. Here, the same sound pressure on the major axis is assumed for both sound
sources.
Directivity C:
logarithmically expressed directivity factor
C
10 =lg
there dB
statistical directivity factor
:st
Ratio of the actual existing in a direction sound pressure compared to the sound pressure, the but
Omnidirectional would caused a sound source with the same sound power.
st =
Page 85 of 466
FWHM:
Solid angle range with a maximum of 3 dB drop in level relative to the main axis sound pressure
level as a function of distance from the sound source Sound pressure level instruments
=
SPL
SWL
+ CLL
--
lg 20 11 1
mS
dB
SPL
C = directivity index, S = distance from the sound source
Hall radius H
( Hall distance):
Distance from the sound source in the direct and diffuse have the same level
0, 057
=
st
TV
V = volume, T = reverberation time RT60

at a distance H
the sound source, the entire sound pressure level is twice as big as the sound pressure
level of the direct sound. Since the diffuse sound having all possible phase angles, the level increase on
the pure direct sound 3dB. In half a reverberation radius distance of the level increase is due to the diffuse
sound just 1 dB. At 2 Hall radii from the overall level is almost entirely dependent on the diffuse sound. The
direct sound makes only 1 dB of the total level.
The statistical sound field:

pure diffuse sound
SPL
LLSWL - =
VV0
+lg 10 lg
10
TT+ 14 dB
V = volume, V 0 = 1m, T = RT60, T 0 = 1s
Page 86 of 466
Music Theory:
Basics:
# : Eng. sharp, raised a half, ending in "to" b: eng. flat, diminished by a half, ending
in "it" : resolution characters, advantages and accidentals are resolved

Enharmonic:
Instruments can play still 17 tones called intoning instruments (eg violin, contrabass ...)
If a # or b are located directly in front of a note, it is called these accidentals. These apply to the clock
line.
If a # or b is set immediately after the treble clef at the beginning, so called this sign. These apply to the
entire song and only be lifted by a natural sign . Test question:
Which accidentals there are not as sign? The resolution characters!
G clef: treble clef F-key: bass clef
Page 87 of 466
intervals:
Intervals are pitch intervals Name
semitones
name
verm. second
prime
kl. second
gr. second
kl. third
berm. prime
verm. Terz
berm. second
gr. Terz
fourth
5
berm. fourth 6
verm. Quarte
berm. third
verm. Quinte
Quinte
verm. Sexte
kl. sixth
8th
berm. Quinte
gr. Sexte
verm. Septime
kl. Septime
10
berm. sixth
gr. Septime
11
verm. octave
octave
12
(Tritone)
berm. Septime
Prim, fourth, fifth and octave there are not the greatest, so-called pure.
keys:
Major scales have always between the 3rd and 4th and the 7th and 8th tone semitone. Otherwise, there
are always whole steps.
The circle of fifths:
Page 88 of 466
Mnemonics:
For the major keys with # sign: Go You age donkey

perch fish Cis for the major keys with b sign: Fresh
bread food aces Des song Ces
Kirchentonarten:
7 modes of major key

1.
.2
.3
.4
.5
.6
.7
3rd. 4 and 7.8.
cde ^ ^ c FGAB
de ^ ^ FGAB cd
e ^ ^ FGAB cde
FGAB ^ ^ cde f
^ ^ gave cde fg
from ^ ^ cde fga
b ^ ^ cde FGAB
c-ionic (Major)
2nd. 3 and 6th-7th d Dorian
1.-2. and 5th-6th e-Phrygian
4th. 5 and 7.-8. f-Lydian

3rd. 4 and 6th-7th g-mixolydisch
2nd. 3 and 5th-6th a-Aeolian (minor)
1.-2. and 4th-5th b-locrian
Page 89 of 466
effects
effect Categories
Einschleifeffekte
dry signals
- driven and rear driven with ISR
- Controlled by:
- 100% of the signal is processed
1. Aux sends (always POST)
- Mono effect
2. subgroups
3. Direct Out (no busbar)
- Always 100%
- Back on:
1. EQ on FX
2. FX on tape
3. FX to FX
4. FX on Phones
Control amplifier VCA Alter frequencies dynamikbearb. FX

distorting FX
Time-delayed effects:
- Noise gate
- EQ
- Reverb / Hall
- compressor
- filter
- delay
- limiter
- Distortion, Overdrive
- echo
- Ducker
(- Pitch Shifter)
- Chorus
- expander
- Flanger
- compander *
- Phaser (- Pitch
Shifter)
* Noise Reduction (eg Dolby A and SR)
Page 90 of 466
Passive filter
consisting of: capacitor, coil and resistance
The capacitor is a frequency-dependent resistance, in other words the lower the frequency, the higher the
resistance.
The coil is a frequency-dependent resistance, in other words the higher the frequency, the higher the resistance.
Passive filters can only be lowered and not lifting. There are 2 pieces: high-pass and low-pass filter.
Consists of the following two parameters:
Cut-off frequency (Cut off frequency) [dB] / [Hz]:
The cutoff frequency is the frequency at which already occurred through the use of the
filter has an attenuation of 3 dB (the -3 dB point). This is the point at which the
characteristic curve is linear.
dB
octave
Attenuation of the signal from the limit frequency per Oktavschritt. The steeper
the slope, the worse the phase of the signal is at the cut-off frequency and the
louder the filter resonance. The more coils and capacitors, the higher the slope,
so this is basically dependent on components. This is the so-called filter order /
filter quality (filter quality). Filter Order:
Slope (slope)
per coil and capacitor or there is an order. A filter of nth order has a slope
of
n6
Octave dB
Calculator:
The cutoff frequency is 3200 hertz. A 4th order filter is ground. What is the
reduction? Answer:
a 4th order filter has
4 6
dB
octave
reduction = octave
24
dB
So you count each octave 3200 hertz to 0 Hz
1. 3200 hertz, 2nd 1600Hz, 3. 800Hz, 400Hz 4., 5. 200Hz, 100Hz 6. There are
therefore 5 octave steps. It follows:
24 5
octave dB
120
so
dB
dB
octave
will now still be added to the -3dB point:
octave
123
= 120
dB
octave
The reduction is
123
dB
octave
Page 91 of 466
Filter resonance:
The filter resonance is a transcendence of the frequency at the point where the edge begins (knee).
Analog filters are possible to the 4th order.
1.1. Low Cut / High Pass (low barrier / high-pass) (formerly: subsonic filter / rumble filter)
[DB]
- 3 dB
f [Hz]
Sign in block diagrams

RC 1st order
in
C lowpass
out
RL-1st order 0 dB
in
L
R
out
R = resistance, C = capacitor, L = coil definition:
Attenuates the entire frequency range below the cutoff frequency. Main
application:
Subsonic filter, diplexer (before the tweeter)
Page 92 of 466
1.2. High Cut / Low Pass (high barrier / low pass) (rare: noise filter)
[DB]
- 3 dB
f [Hz]
Sign in block diagrams

RC 1st order
highpass
in
C
R
out
RL-1st order 0 dB
in
out
Definition:
Attenuates the entire frequency range above the cutoff frequency. Main
application:
Noise filter, diplexer (front woofers), antialiasing filter (distortion in digitization)
Page 93 of 466
1.3. Bandpass (combination of Low Cut and High Pass

[DB]
passband
0 dB
unprocessed area between the

cut-off frequencies
-3 dB
f Gu
f Go
f [Hz]
Sign in block letters images

THE DIFFERENCE BETWEEN BAND PASS AND PASS BAND: BAND PASS DESCRIBES THE NOT
EDITED BETWEEN THE INTERNATIONAL FREQUENCY AND BAND PASS DESCRIBES THE
FILTER
in
out
Serial connection of the filter (Low / High Cut), so no notch filter !!! WITH ACTIVE FILTER
ANDERS LRL:
Below this frequency is damped
(Regardless of the position in
Frequency range) Upper

limit frequency:
Above this frequency is attenuated
(Regardless of the position in
Frequency spectrum)
Application:
Telephone voice (300Hz-3kHz)
With the overtone structure of the voice hearing of man can reproduce a voice, although very little is
transferred residual effect
Page 94 of 466
1.4. Band-stop filter
processed region between

the cut-off frequencies
[DB]
stop tape
0 dB
-3 dB
f Go
f Gu
f [Hz]
Sign in block letters images

THE DIFFERENCE BETWEEN BAND STOP AND STOP BAND: STOP BAND DESCRIBES THE
EDITED BETWEEN THE INTERNATIONAL FREQUENCY AND BAND STOP DESCRIBES THE
FILTER
in
out
Parallel connection of the filter (High / Low Cut) !!! f Gu is

always greater than f Go !!!
Application:
Remove Narrowband noise
GENERAL further
filter: brickwall
filters
Remove high grade / quality
tunable filter
Filter with "free" selectable cut-off frequency
Page 95 of 466
Active filters
- can raise (boost; Lower: Cut)
- Cut and boost are limited, so can not lift infinitely / Lower
- Active Components (eg op-amp's, transistors, tubes, VCA ...
Shelving Filter:
[DB]
Lo Shelf Boost
Hi Shelf Boost
+ 3 dB
dB
-3 dB 0
Lo Shelf Cut
Hi Shelf Cut
fG
fG
f [Hz]
There is thereby no single frequency definition:
the entire frequency range below (Lo Shelf) or above (Hi Shelf) the cut-off frequency will be processed
depending on the set cut / boost. 2 phenomena occur here:
1. The slope is dependent on the processing. That is the stronger the processing, the steeper the slope!
2. If Boosted or cutted, an area subjected to machining, which should not be processed!
the shelving filter per se is designed professionally, because it is not working frequency selective!
Page 96 of 466
Bell filter
[DB]
+ 3 dB
dB
-3 dB 0
f Gu
fc
(f 1)
f Go
f [Hz]
(f 2)
Sign in block letters image

definition:
Only a certain frequency range is processed. The closest point of machining is the center frequency.
This is the logarithmic midpoint between f Gu and f Go. The distance between f Gu and f Go determines the
bandwidth (Bandwidth). good is:
1. frequenzselektierender filter (the only)
2. Processing below the upper and above the lower limit frequency. settings:
1. parameters: Center frequency
2. Parameter: Cut / Boost
3. parameters: bandwidth (bandwidth and slew rate are inversely proportional, this means the steeper
the slope, the Narrow band (high quality), and the further the flank, the wider the band (low quality))
Page 97 of 466
The calculation of the bandwidth in Bell Filters:
= B- 2
= c 1
fff
c
Hz
[ ff]
=
2
Hz
[ ]
fff B f -Q
B = bandwidth f c = Center frequency, Q = Quality
Example calculations:
1) f 1 = 800Hz, f 2 = 3200 hertz
ff1
fff2
=3200
[2400 800Hz ]
-=-=
=800
[1600
3200
Hz ]
==
1600
0 2400
=
, 67
===
BfQ
fff
- 2
2.) f 1 = 1500Hz, f 2 = 1800Hz
ff1
1
fff2
==
c
===
BfQ
[300
1800
=
1500Hz ]
-=-=
fff
- 2
1800 1500
= 1643 .
1643 .17
5300
[17
Hz ]
, 48
3.)
What is a half octave above 2800Hz?

calculating the center frequency. Since 2800Hz times 2 = 5600Hz (corresponds to one octave) is obtained:
fff2
==
=2800
[3959
5600
.,, ,
Hz ]
Page 98 of 466
graphic equalizer:
- are x-parallel Bell Filters
- have fixed center frequency and fixed bandwidth
- only possible change is cut / boost -15dB / + 15dB There are two
different forms:
1. Octave band EQ:
- has 8-10 bands

- Bandwidth: jew 1 octave.
2. Terzband EQ:
- 31 parallel belts
- Bandwidth: major third
Parametrics
- Only Bell Filters

- Cut, Boost and Center frequency changeable
1. fully parametric equalizers:
- Bandwidth selectable (adjustable)
- Cut / Boost selectable (adjustable)
2. semi-parametric EQ:
- fixed bandwidth
- Cut / Boost selectable (adjustable)
3. quasi parametric equalizers:
- switchable bandwidth (eg Hi Q, Lo Q, most button)
Furthermore, there is the automatic bandwidth control:

1. Automatic bandwidth control
[DB]
- The more boost, the wider the bandwidth

- Cut the more, the smaller the bandwidth
2. Variable Q (eg: SPL Optimizer)

- The larger the processing, the narrower the bandwidth
f [Hz]
[DB]
f [Hz]
Page 99 of 466
The 4 areas of the frequency spectrum:
Bass (Lows) 20Hz-200Hz:
- Fundamentals in nearly all instruments and voices

- Pressure range of almost all instruments and voices
- Narrowband work, because everything is close together!
Low-mid (Low Mids) 200Hz-1000Hz:

- basically the area that the ear perceives with best
- typical of guitars (about 500Hz-1000Hz)
High-mid (High Mids) 1000Hz-5000Hz:

- The transparency, ie every instrument is audible
- Speech intelligibility, usually about 2000Hz-4000Hz
Heights (Highs) 5000Hz-20000Hz:
- hardly keynotes
- work with broad bands
- high brilliance of sounds
Page 100 of 466
Control amplifier (VCA)
VCA
payload
control signal
control voltage
Logic unit
sidechain
Dynamic processors effects work all by this graph Ratio and Threshold are all
VCA's equal of their importance Threshold (threshold):
Threshold is the point at which occurs a change in the amplification ratio. Either with oversteer or
understeer, the amplification ratio changed. Ratio (amplification ratio):
Relationship in which the signal after exceeding or falling is increased.
All VCA's the below have the ratio of 1: X expander below the farther, the more
expansion all VCA's the crossing have the ratio of X: exceeds 1 compressor the farther,
the more compression
Basically, one has in increasing levels the Attack Time and sloping levels, the Release Time!
off
switch
When falling below the Attack Time

When falling below the Release Time
When the attack time is exceeded

When the release time is exceeded
switch
off
Page 101 of 466
Noise Gate (sub Schreiter):

out
1: 1
1:
in
makes all signals below the threshold infinitely quietly

- to seem to noise (eg Voice breaks)
- Expanders with Ratio 1: is noise gate
- a pure noise gate does not require a ratio controller has 3 time
parameters:
1. Attack Time approx 10ms to 1s:
- The attack time is the time in which after exceeding the threshold in the original ratio of
1: 1 controlled back
2. Hold Time about 2ms to 2s:
- The Hold Time is the time after falling below the threshold the signal is still in the ratio of
1: 1 is processed
3. Release Time approximately 2ms to 4s:
- The Release Time is the time is turned down in after falling below the threshold of the
set expansion ratio (on-time)
Special features of the gate:
hysteresis:
There a level difference to the actual threshold and works as a second Threshold function. Both
thresholds are always working alternately. Basically, the higher the threshold OFF threshold of the gate
and the lower the turn-on threshold of the gate.
[DB]
dB
- 20 dB 0
Turn off
turn on
Page 102 of 466
Link:
The control signal from the master is in addition as a signal for the slave. All parameters can be further
adjusted individually. Mostly this function is used only for the snare to make the Snareteppich audible.
The problem with this is that is that the microphone is set up so that most bass drum with houses, so one
does not have direct attack point of the snare. Therefore, the signal of the snare Fells is ground into the
logical unit of the VCA's from Snare, whereby the gate then only opens when the impact of the snare
Fells is heard. By pressing the "Link" button, or similar, grinds to a this, that is the gate of the carpet is
the slave and the coat to the master.
L in
Lout
VCA
Master
Logic unit
link
R in
logic unit
VCA
R out
slave
Range:
Range is the gate of the controllable damping in the closed state, also referred to as "soft noise gate".
out [dB]
1: 1
Tg
20dB Range
- 20
1:
in [dB]
Page 103 of 466
External Keying:
this controls an external signal, the so-called external key, the useful signal. Similar to the link, only that
this is not about two cascaded gates, but only by one having the external in.
Range
in
out
KEY list
VCA
filter
Logic unit
KEY IN
Ext. In
When the "KEY IN", or similar, is not pressed so operates the gate Selfkeying, so if no external signal
come into it.
Expander (sub Schreiter):
The attack, release, Hold Time are defined just like the Noise Gate. The Expander widens the dynamic,
speak softly signals are made even quieter name for this is a "soft noise gate".
out [dB]
1: 1
Te
0
1: 2
0
in [dB]
Page 104 of 466
Compressor (About Schreiter):

gain Reduction
out [dB]
Tc
3: 1
1: 1
in [dB]
limit 1. Dynamics
2. The loudness is based on RMS
3. make signals miscible
4. signals "fat" make (pressure generating)
5. you can EQen (you can through the attack / release times certain parts of a sound simply compress)
If one makes a signal to "fat", so called pumping this effect. With this you can then hear the level
operation. Because you should never listen to a compressor but that it works, it should be avoided if
possible actually, that he begins to pump. but still is a nice effect.
Actually, all compressors are so-called Downward compressors. These do only quieter, therefore, the
average gain reduction at the output gain has to be turned up.
Attack and Release are in a compressor Fade In and Fade Out.

That is: The lower the threshold, the greater the gain reduction. And the higher the ratio, the greater the gain
reduction.
All About Striders only attack and release and no hold.
The attack time is the time after the threshold is exceeded is adjusted down to the set compression ratio.
The Release Time is the time in which to fall below the threshold in the original ratio of 1: is regulated
back. 1
A special type of compressor is the point of rotation compressor. this has a fixed threshold and only the
input gain can be controlled. Also note that the more input, the lower the threshold.
reduced dynamics
Through compression increases normally the average level - the signal is "louder"
Reduced momentum after
compression and after
Original dynamics
Output Gain
without compression
Reduced momentum after

compression
Page 105 of 466
1. Auto function:
An automatic control of attack / release time which is controlled by the program density. The program
density is the effective value, ie signal per time. A high density program
fast attack, slow release time
A low density program middle attack, medium release time singing is typical of
the Autobots
The Alesis 3630 there is a Peak / RMS switch this feature on or off other
2. hardknee / Softknee:
used in singing, and all long Einschwingenden instruments. For percussive instruments not, otherwise no
pressure develops.
3. Link:
the link at crossers is completely unlike sub Schreitern because it creates a complete parameter link.
The sum of the two control signals running in the logical unit of the master and it controls both VCA's. Is intended
for compression of the stereo signals.
Basically, Threshold, Attack, Release, Ratio are linked. All other parameters may or may not be linked,
depending on the device.
4. ISR in sidechain:
Is the counterpart to external keying to a noise gate. There are different areas of application. Often to
edit a control signal (eg a Equaliser).
Multiband Compressor:
A multiband compressor is a compressor is submitted to a crossover. This all individual bands are
compressed and then summed. Actually intended as a master tool, but you can also take as a snare.
The three standard units are: TC Finalizer, SPL optimizer and the APHEX dominator.
Page 106 of 466
limiter:
The limiter is actually a special function of the compressor. One calls compressors from a ratio of 10: 1
Limiter. When limiter is regulated, ie nothing is louder than the Threshold. The attack and release time
are set as possible as short as possible. transients:
Transients are frequency components which override, eg with percussive instruments special function:
look-ahead limiter
according to the tap of the control signal, the useful signal is delayed by the minimum response time of the
VCA's, characterized shortest peaks can be damped.
Ducker:
The Ducker is a special function of a compressor with a ratio of infinity to 1. The signals above the
threshold are infinitely quieter strengthened. In the end, a ducker is the inverse of a noise gate. Normally,
there is not the Ducker as a single device, but only as a further circuit in the noise gate. Therefore
Ducker also has all the features of the Noise Gate.
Page 107 of 466
Standard (rock) instruments and their settings

!! THE SOUND IS MADE TO THE MICRO !! Drums:
A percussion sounds always only good when it is in a good mood! There must be a deep sonorous
"STUPID" shock, because we recognize that it is in a good mood. First you tune the skins in itself, that
must be connected to each bolt the same train his (Tensionwatch). When skin on the fur near the screw
with a stick, it has to sound the same at each screw. In a drum the skins may sound only in certain sound
steps, ie every Tom, for example only a third away. The higher the sound resonance head compared to
the drumhead, the shorter sounds the drum.
EQ Settings:
The pressure range of the song is about 100Hz (schmallband boost) (subbasses below 80 Hz).
The resonant frequency is slightly above the pressure (schmallband cut). Must actually gone, sounds too
spatially.
The kick of the bass drum is again at about 3 kHz and even at just 4,5kHz (at a middle belt of two boost
apply).
A piece where no bass drum is present does not help, because no one is dancing on it !!!
Compression:
compressing in several stages, ie compressed during recording (easy), and the mix (stronger). Even the
use of several compressors worth it but always series. Basically still only use the noise gate, and only
then the compressor. It always takes a short ratio of 2-4: 1 for the pressure. Each strike must be higher
than the threshold.
Page 108 of 466
EQ Settings:
In singing should always EQen musical little cut / boost, wide bands
1. Fundamental:
Men:
100-150Hz
Women:
150-200Hz
always mittelbandig Boosten
The speech intelligibility is 2-4kHz

Voice over the music intelligibility low, so that the voice is not annoying and vice versa
presence (brilliance): 7-8kHz
Cautious because voice can sound very sharp noise:
10kHz and up
Breathe (not wegmachen sounds otherwise unnatural)
The EQ settings are not so important, because every voice there is a matching Mirkofon. Formants:
Resonant areas that are linked to vowels deepest:

200-400Hz e
u and i
and o 400-600Hz a
800-1Khz (Singformant)
These are things that you can not edit

Compression and Noise Gate:
Noise Gate during recording: Threshold high, so at the slightest noise, the noise
gate rises
Compressor: use Autobots, otherwise medium / medium, low ratio 2: 1 or less, then
compress as much as is
Page 109 of 466
Dry signals Settings: Delay:
originally Tape Delay

The Feedback controls the number of repetitions and is effectively the return control of the output to the
input.
= Time t
Delay
cm cm vd= []ss
In feedback, there are filters (HiCut / LoCut) to get out as the bass increase (Lo) or to simulate natural
echoes (Hi). The voice becomes much clearer, since in reality the heights are absorbed rather than low
frequencies. In natural areas have been above 8 kHz, due to absorption, no signals more (5 kHz are
gone after 3.2s, 10kHz contrast from just 1.2s.
There is an algorithm to the Tape Delay mimics (today it is mostly eh Digital). The delay is a widely used
instrument in acoustics, but must lie exactly on time, so the rhythm. Dotted Delays are common. They
produce triplets (via the TAP function adjustable) special forms: Ping Pong Delay:
Alternating left-right. In this case, the feedback signal is passed the left in the right channel and vice
versa. Multitap Delay:
can vary ms (several individual delays with individual delay (1st series,
2. parallel)
1.
- a single delay block has usually 1232ms

Sum of all delays
- Max. Delay Time
- Starting point of all subsequent delay is dependent on the delay time of
the preceding delays
.2
- no addition of Delays
- all are independent of
application:
- everywhere where one delay times
- to avoid comb filters
- Delay Tower / -line (Haas effect for live performances)
Page 110 of 466
Modulation effects (effects result from the delay):
- Chorus / Flanger (actually the same effect)

The delay time is not perceived, one actually hears only the comb filter. the delay time is
modulated (periodically changed). With the feedback comb filter become stronger / weaker. The
modulation source is the signal itself, the goal is the modulation target
.t The frequency of the LFO is
Modulation speed. A flanger has a large modulation depth and a low speed, the Chorus
reversed. A flanger without feedback is a Phaser.
- Phaser:
The Phaser has several narrowband Bell Filter Cuts. The goal
Center frequency. A wah-wah is ei special case of the phaser and has only one filter.
delay
is the
Feedback Depth Speed
Chorus 10-30ms 10-20% 20% fast / medium flanger

1-10 ms
Phaser
1-10 ms
20-30% 100%
x
slow
100%
slow
doubling 30-40ms
20% fast / medium
vibrato
15%
10ms
nearly
Page 111 of 466
Reverb (Hall):
HERE to simulate natural spaces! The criteria of

a room:
Absorption:
is the conversion of kinetic energy of the particles due to friction and heat
the harder a material is, the higher the degree of absorption
the larger the area of a substance, the higher the absorption
all frequencies above 8kHz be completely absorbed

Hi-Damp and Damp-Lo are the parameters in the Reverb (Also Lo or Hi-Cut)
the bass is always shown better in the room, so do not use too much bass
Diffraction:
when the obstacle is smaller compared to the shaft, the sound is diffracted. The diffraction itself is
relatively uninteresting, because it perceives as diffraction only the direct sound shadow.
reflection:
-
it caused up to 50,000 per second reflections.

it is Dependant on signal when an echo is perceived (Attack Time)
the smaller the space, the shorter is the density (ms)
the smaller the space is represented, the better is the effect device
General to Reverb:
- the larger the PreDelay, the smaller the distance from the sound source seems to the sound source
- be adjusted as more boundary surfaces, the more similar is the detour.
Offset in depth:
-
Predelay:
the shorter, the far
Volume:
the louder the close
Treble: the dull, the far
Hall Share:
the more, the far
Compressor: the stronger the close
There are enough individual points barely audible to tell the brain how deep a room.
Page 112 of 466
Hall itself is very simple and yet aesthetically. Hall should always match, so it must sound good. When
adjust the reverb should always time to stop the song and pay attention to the reverb tail.
Diffusion:
Harte reflections occur on straight surfaces and all reflections are derived in the same direction (0%
diffusion)
Soft reflections occur on uneven surfaces and all reflections are directed in other directions (100%
diffusion) Diffusion is adjustable via the parameter spread.
Room size / reverberation time:
(- Defined world with the RT60)

- is the time until the Reverb sound dropped after switching off the sound source to 60dB at one millionth
of its sound energy.
- This is measured at 1kHz
- ambient noise level usually so high that RT60 is not measurable. There are then approximations (RT30
and RT15 (IRT = initial reverbaration time))
- the longer the reverberation, the greater is the room:
room
Time (TR60)
Language booth <500ms

Cabarettsaal
approximately 0.7 s
Schauspielhaus 0,7-12s chamber

music 1.4s Opera House
1,3-1,6s
concert hall
1,7-2s
churches
> 2s
Page 113 of 466
Possibilities of reverberation:
Naturally:
1 room microphones (beyond the reverberation radius)
"Ambient-Mixing"
Direct and diffuse identical
2. create echo chambers

a room with variable acoustics, which serves only for the creation of the reverb
large, and surfaces can be changed

artificially:
1. Mechanical:
1.1: Hall plate (Plate): free-swinging plate

but has a dimension less as it swings in one direction only
1.2: Gold Plate (films): sounds almost the same, but smaller
1.3: Hall springs (Springs): vibrate the spring is used as Hall

2. analog reverbs (all trash) very loud
noise
3. digital reverb units:
- Multieffects: everything is calculated according to an algorithm

- Quality of these depends on the algorithm, then only the computing power
Folding (convolution):
- to calculate an arithmetic principle complex things faster
Page 114 of 466
microphonic 1
Book Recommendations from
Lutz Nobert Pawe
microphone practice
PPV Medien
Michael Dickreiter microphone recording technique
Hirzel Verlag
Thomas Grne
Microphones in theory and practice
Elektor Verlag
Andreas Ederhof
Microphone book
GC Carstensen Verlag
Jrg Wuttke
Microphone essays Schalltechnik DR. Ing. Schoeps
Dr Gerhard
microphones
Receiver: converts the sound vibration into a vibration membrane. Empfpngerprinzip:
Art fashion as the conversion takes place. done type fashion, as the conversion:
conversion: Converts membrane vibration into an electrical oscillation to Transducer
Merge Recipients: (Pressure transducer)

Only one side of the membrane is exposed to the sound. Have the intrinsic capsule is finished almost
airtight. Only a Kapilarffnung ensures that in the interior of the capsule there is the atmospherische air
pressure.
Whenever the pressure upstream of the membrane deviates from the capsule internal pressure, the diaphragm is
moved. For overpressure the diaphragm moves inward in under pressure outward.
Antenna direction of Mirkofons Studio Projects

C4 Oktava MC 012 (MK 012) The antenna
direction (Polar Pattern)
The antenna direction indicates how high is the output voltage of Mirkofons depending on the
direction of sound incidence. This is specified as a relative voltage level in a pie chart (or
directivity pattern polar diagram).
Page 115 of 466
Bullet:
The antenna direction at a pressure receiver is always ball. Pressure

gradient:
depending incident from which direction the sound, he has to put different size way back to the front or to
the rear of the diaphragm. This results in a phase and therefore to a difference in pressure between the
two sides of the membrane. An open pressure gradient transducer always has the antenna direction
Eight.
Sound of the back results in a Achterkarakteristik phase reversal. Acoustic delay element:
Page 116 of 466
Kidney:
In the antenna direction of the kidney way through delay element is as long as the path around the
membrane.
Wide cardioid:
For the antenna direction width kidney must be gone longer by the delay element as the way around the
membrane. Super and hyper cardioid
In supercardioid and hypercardioid the way through the delay element is shorter than the path around the
membrane.
Richtkarakteris
tik
Engl.
Name
Bullet
omnidirectional 0 dB
wide cardioid
Hypocardioid
Blanking 180
direction
suppression
90
direction
- 4 dB
Off-Axis
monoAufnahmewink
Bndelungsg
el
0 dB
- 10dB
150
subcardioid
widecardioid
kidney
cardioid
1.5
- 6 dB
Theory: - infinity
dB
180
131
1.7
Practice: -20 dB
supercardioid
supercardioid
- 9 dB
- 12 dB
126
115
1.9
hypercardioid
Hypocardioid
- 12 dB
- 6 dB
110
105
eight
Figure 8
Theory: - infinity
dB
0 dB
90
90
1.7
Practice: -20dB
club
praise
2.2
Page 117 of 466
Mono recording angle:
the solid angle range in which you get a maximum of 3 dB level drop towards sound from the front.
directivity factor
The directivity factor is the ratio of the electrical power output in case of frontal sound incidence
compared to diffuse sound incidence higher the directivity factor, the more directional the
microphone is.
directivity factor Is the voltage ratio
unidirectional
bidirectional
Nominal impedance (Impedance rating) output impedance

of the microphone at 1 kHz nominal terminal impedance
(load impedance)
Minimum input resistance, the subsequent microphone preamplifier.
Moving coil microphone ( Moving Coil Microphone)
1.
incoming sound
.2
membrane
.3
Kitchen sink
.4
permanent magnet
.5
resultant signal
At the membrane a voice coil is glued. This is located in the annular section of a permanent
magnet.
Page 118 of 466
compensating coil
The compensation coil is additionally fitted to the voice coil. Is but opposite polarity. This hum can be
wiped detect both coils.
The output impedance is 200 ohms. There is no impedance conversion necessary. The magnet is
composed of:
Aluminium nickel cobalt alloy (Alnico) Neodymium
A piezoelectric transformer as
piezoelectric crystals
Piezo crystals respond to mechanical deformation with a charge transfer, which can be tapped off as a
voltage. Use as a contact pick. Carbon microphone
A sound-permeable housing is closed at the top with a metallic membrane. The case is with carbon
grains (made of Anthracite) filled, is under the carbon grains, the counter electrode. Between the
membrane and the counter electrode, a direct electric voltage is applied. Sound waves are transmitted
through the membrane on the carbon grains. The microscopic changes in position of the particles cause
a modulation of the current flowing through the direct current. Different application requirements can
cater to a certain extent by different grain sizes. THD case is 10-40%
Page 119 of 466
size
condenser Mic
ribbon Mic
Moving coil mic
FL
Electret Kon. Mic 5-15 mV /
about 1 mV / Pa
1-4mV / Pa
Pa
pure Kond Mic 8-25 mV /

Pa
external voltage
> 1 mikroV
<1 mikroV
<1 mikroV
Equivalent noise
12-24dB (A)
15-30dB (A)
15-30dB (A)
frequency response
Kond-DE - very linear Kond
Linear
nonlinear
DGE - linear
Frequency range
Kond DE very large 20Hz-
But limited in bass and treble
But limited in bass and treble
120-130 dBSPL
> 140 dB SPL
Good
badly
No
Yes
20kHz possible Kond DGE

limited in the bass
Max SPL
Transformer sym. 120-130

dBSPL
electr. Sym 125-140 dBSPL
Kond-DE - Best Kond-DGE
transient response
- very good
No
Robust
Sensitivity
Sensitivity (Sensitivity)
Output voltage for a given sound pressure [mV / Pa] field-operating gain
(FB):
The microphone is loaded with a resistor 1K ohm. In parallel, the output voltage is measured.
Measurement signal: 1 kHz tone with 1 Pa sound pressure
field Open circuit voltage (FL)
Measurement of the complete output voltage at idle
Foreign voltage
noise voltage
produced by spontaneous movement of electrons in the components of the microphone transmission
range
Frequency range, in which one max. 3 dB level drop compared to 1 kHz has. Equivalent
noise
Equivalent sound pressure level ((equivalent noise level)
Mic
B (transfer factor) U noise
enhanced gain subway noise
capacitor 20mV / Pa
2 mikroV
61.4
122.8 mikroV
Dynamic 2 mV / Pa
0.5 mikroV
614
307 mikroV
Page 120 of 466
The sound pressure level that produces an output voltage in a microphone, which is the same as
the external voltage. Equivalent noise calculate: applies to IEC standard:
URausch
L noise =
20lg
B
2
10 - 5 URausch
U noise is measured using an A-filter and as an effective value indicated CCIR standard:
U noise measured over CCIR filters, indicated as a pointed peak dB (A) + 10 is

about dB (CCIR)
Signal to noise ratio Signal to

Noise Ratio SNR
SNR = 94 dB - Noise floor
Frequency response:
If you see a transfer curve.

The curve indicates how high is the output voltage as a function of frequency. Is specified as a
relative voltage level. 1 kHz is set to 0 dB. Distortions linear distortions
are level and phase change non-linear

distortion
harmonic distortion
THD (overtones)
THD
Percentage of harmonic distortion total signal in% THD (Total
Harmonic Distortion THD) not harmonic distortion (more frequency
components) SPL: Maximum Sound Pressure Level
The sound pressure level at which a certain distortion is achieved. (Studio: typically 0.5%
THD) (companies that have so: Schoeps, Beyerdynamic, Neumann, Microtech Gefell)
Page 121 of 466
Reasons for the occurrence of harmonic distortion in the microphone
. 2 Transformer is driven into magnetic saturation

. 3 Microphone amplifier is overdriven
. 4 Membrane can no longer follow high SPL
Attenuation in condenser microphones (Pad) dynamics
Distance from Equivalent noise and max. SPL transient

response
How well the membrane can follow the sound signal? The smaller the membrane mass, the better the
transient response. The higher the resonance frequency of the membrane, the better the pulse response.
frequency response
Electrical frequency dependence
Electrostatic transducers are Auslenkungswandler or Elongationswandler. Here, the output voltage

depends on the deflection.
The Electrodynamic converters are Fast converter. Here, the output voltage
depends on the membrane quickly.
Mechanical frequency dependence
lowly tuned capsule: resonant frequency at the lower end of the listening area centrally coordinated
capsule: resonant frequency at the center of the listening area, highly-tuned capsule: resonant
frequency at the upper end of the listening area mass inhibition:
Above the resonant frequency decreases the amplitude of the resonator.

1. Diaphragm displacement of the pressure receiver
1. It remains constant until the resonant frequency. Above the resonant frequency
drops it because the mass resistance starts.

. 2 Fast membrane of the pressure receiver.
1. It increases to the resonant frequency, because at constant

Diaphragm displacement, the number of vibration cycles per second increases. Above the resonant
frequency, they will drop, because the mass resistance starts.
3. cone excursion of Druckgradientenempfngers
Page 122 of 466
It increases up to the resonance frequency because the phase difference between the two sides of the
membrane becomes larger. Above the resonant frequency to fall off because the
Mass inhibition begins.

. 4 Fast membrane of Druckgradientenempfngers:
Until the resonant frequency it increases, because with increasing cone excursion, the number of
vibration cycles per second increases. Above the resonant frequency remains constant because
the decreasing diaphragm displacement and the increasing number of vibration cycles per
second in balance.
Resulting frequency responses
1. Condenser pressure transducer
1. Deflection response of the pressure receiver

. 2 High matched capsule
. 3 Linear frequency response from 20Hz to 20kHz
. 4 always Omnidirectional
. 5 Measurement Microphones are always condenser pressure transducer
. 2 Dynamic pressure receiver

1. Fast-frequency response of the pressure receiver
. 2 centrally coordinated capsule
. 3 increased bass and treble

. 4 Facial damped
. 3 Pressure gradient condenser

1. Deflection response of the Druckgradientenempfngers
. 2 centrally coordinated capsule
. 3 increased bass and treble

. 4 Facial damped
. 4 Dynamic Pressure gradient

1. Fast-frequency response of Druckgradientenempfngers
. 2 Deep matched capsule
. 3 raised heights
Acoustic Frequenzgangkorrerktur
-
Helmholtz resonator
- Air chamber with openings
- Air in the chamber has a spring action
- Air in the openings a mass
-
corresponds to a spring pendulum
in the range of the resonance frequency of the Helmholtz resonator, the diaphragm
movement is supported.
Page 123 of 466
Sound beam Santander funnel
Through a sound-collecting hopper in front of the membrane high frequencies are bundled, are
low and medium frequencies are unaffected. With dynamic microphones can be lowered only
by passive filters. In condenser microphones
Can be equalized with active and passive filters in the frequency response, because they have a power
supply.
Boundary Microphone
(Pressure Zone Microphone) PZM (Baundary Layer Microphone) BLM basic idea:
The pressure jams during printing receiver is enhanced by incorporation in a large area at low
frequencies.
Typical applications: bass

drum bass instruments
wings acceptance speech
acceptance Theater
Stereo recording (in interfaces AB) benefits

the interfacial microphone 6 dB level increase
more orderly sound field Unobtrusive
Page 124 of 466
Shotgun microphone (Shotgun Microphone)
Interference receiver (pressure gradient transducer with cardioid or hyper-cardioid shotgun +)
low frequencies: Super or hyper cardioid High

frequencies: lobe
Lateral sound is discolored sonically. Use:

Filmton
lavalier microphone
Capes microphone for voice pickup

Bass and 700-800 Hz - overemphasized
heights too weak solution:
Lavalier equalization of
bass
Lowering of 700-800Hz (falls away at lavalier) treble boost
Most condenser pressure receiver microphone

selection:
For loud instruments at close range: moving coil microphones
in instruments with low volume and loud instruments from a distance: condenser microphones PML DL 96 Pearl
Microphones Laboratories
Bass Stressed Instruments: large diaphragm condenser mics height

Stressed Instruments: Small-diaphragm Mirkos
Page 125 of 466
microphone positioning
Close miking
Ambience Miking
Distance Miking
Close miking
Distance decreasing
ambient recording
Sound change (by
Natural sound
Mostly diffuse sound
Nahbeschprechungseffekt and only

small part
Mostly direct sound
Direct and diffuse sound
Reducing akutsischem crosstalk

from neighboring instruments
Instrument order: Acoustic

guitar: condenser
microphone Typical
positioning:
Transition neck in body, about 50 cm distance
Acoustic guitar
Large and small diaphragm condenser microphone
(AKG C414 B-ULS, Neumann TLM 102, Rode NT-2, Shure SM 81 SE Electronics SE 1
A PML DC 96)
Decrease as with acoustic guitar
DI box
Direct Injection
Impedance conversion (from high impedance to low impedance)
signal is balanced Ground lift (mostly switchable) versions: Active -
passive -
Power supply required (battery and / or phantom power)

no power supply needed
Page 126 of 466
Electric guitar amp
Shure SM 57 - height rich, + 6-7 dB at 6-7 kHz Sennheiser

MD 421 -
E-Bass
Beyerdynamic M 88
Sennheiser MD 421 AKG
D 112
Stringed Instruments:
Microphones Bass
Bass Stressed Instruments: large diaphragm condenser mics
cello
or
Viola (Viola)
violin
Height Stressed instruments: small diaphragm condenser mics

Bass Stressed Instruments: large diaphragm condenser mics
Height Stressed instruments: small diaphragm condenser mics
For Violin and Viola: decrease from above with small diaphragm mic. Approximately 1 m For Cello:
directed, approximately
possibly small membrane micro to the area of F holes

50cm distance
In bass
Large-diaphragm microphone, close loss
also possible: boundary microphone on the floor in front of the Bass
Drums: skins
with moving coil mics
pool
with small diaphragm condenser Mikes
Drums and cymbals rays heights perpendicular to the skin, or from pools.
Snare:
top:
Shure SM57 Sennheiser
MD 421 below:
AKG C 414 B-ULS
- Radiating down in opposite phase, so turn for lower Mic phase.
Page 127 of 466
MIDI
MIDI
Musical Instrument Digital Interface
Digital> works serially> 2110011100111> binary Furthermore

MIDI works with messages. Example:
Note On> button> Sequencer to conduct sound

-
Serial
Unidirectional
TR
( T = transmitter,
R = receiver)
x
x
Command structure - Number Systems
decimal system
As a sign of the decimal system using the Dec 137; DEC

101 d 1109 Consists of: 0
099 (now we want 1 Number dranhngen)
100> results
1
23
517, 7 = 1 decimal point, 1 = 2 decimal point, 5 = 3 decimal point

The decimal point are counted from right
4
56789
decimal point
valence
power
of values
decimal point
1000
100
10
10
10
10
1
1
10
-1
Now we want from this number to calculate a decimal value. 1 2 3 9

= 9 + 30 + 200 + 1000 = 1239 9x1
=
3x10 9
2x100
= 30
= 200
Page 128 of 466
1x1000
= 1000
binary
Is made of:
01
As a sign of the binary system is used the following abbreviations:
b 1011, I 1100, 1001 2

Conversions of binary from decimal B1111
= d decimal point
valence
power
b1111
8th
1
1
= d = 1 + 2 + 4 + 8 = 15 1 d
=12
=2
=4
4
8
=8
= d = 8 + 0 + 2 + 1 = d 11 1
b1011
=12
=2
=0
0
8
=8
1001 = d 9 d
7
= B 0111 b
01111111 = d 127 b =
10000000 d 128 xxxx x xxx
x is a binary digit (eng. B inary D i git Bi t = Bit) byte 8 binary

digits = 8 bits = 1
Page 129 of 466
For blocks in bytes MSB and LSB is called directly on an entire block (byte).
Page 130 of 466
hexadecimal
0 1 2 3 4 5 6 7 8 9 ABCDEF Hexadecimal numbers are named with the following abbreviations:
Hex 13A
$ F09 ABD7
H
# 730 (In most Midi # ) 104 1F
decimal point
valence
power
.4
.3
.2
4096
256
16
16
16
16
1.
1
1
16
Conversions from hex to decimal: $ 2 7 = d

= 39 (7 + 32) 7x1
=7
2x16
$ 1C = d
= 32
= 28 (12 + 32) x1
= 12
1x16
= 32
Conversion of hexadecimal numbers into a binary code on the decimal: $ 1 A = D 26 = b 0001 1010
$ A 1 = D161 = b 1010 0001
This example can be seen that it is possible to convert Nibbleweise. MIDI operates generally binary,
but when it is displayed in decimal format then hexadecimal.
Page 131 of 466
messages
A message is a command which generates a sound command =
Or
He makes a sound of a certain pitch = command + data MIDI parameter / is always sent
generally in complete bytes.
status
(Command or opcode)
databyte
3. possibilities of SB Message
SB - DB
SB - 1.DB - 2.DB
The number of data byte are defined with the status.

If the receiver had a status with the definition that 2 DataBytes tuned to arrive so the recipient
knows "Aha, it must arrive 2 Data Bytes". Among other things, the receiver must be able to
distinguish what matters. For this reason, has a status as a 1 bit always 1 and the Databyte a 0th
status
1 .......
databyte
0 .......
For this reason, only 7 bits remain available that makes 128 possible settings.
(128 = 0-127).
Categories of Messages
Page 132 of 466
channels
Construction of a status byte in a message:
In a channel code 0100 it would channel. 5

Due converting from binary to the decimal system would come out. 4 Now Channel 4 is not the reason
because in these channels 1-16 no channel with 0 exists.
Page 133 of 466
Manner as the messages to be sent
The sequencer sends the signal to both receivers. Both Rx receive the same data, but the first Rx only
works on CH1 and the second only to CH2. That is, the first receiver says "Okay, I will respond only to
channel 1 of the rest does not interest me, but I lead him on.", The second says "Hmm I should give to
CH2 eight and the rest does not interest me either I forward this "or the data speak no other receiver to
(CH 3-16) then these are simply ignored.
A MIDI cable asynchronously. (> Untimed)

With a transmission capacity of about 31250 bits / s with a standard deviation of
+ /- 1%.
MIDI does not work like a digital line such as S / PDIF or Optical Cable. These lines are constantly
working. Once we worked connect S / PDIF line and an enormous number of data flowing through the
line. In MIDI, this is not so. If MIDI is connected flows nothing while nothing has been commissioned.
Due to the slow speed MIDI can operate asynchronously. This MIDI recognizes "Aha now I should work
and stop now," one imagines each byte a 1 as a sign of the start bit and renewed 1 as a sign of the stop
bit. For a transfer of a byte so 10 bits are needed. Here is an illustration:
It should be clear that the receiver needs to know simply when to start reading and when to stop.
Transmission of SB, 1 DB, DB 2:
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The duration of the transfer is calculated as follows:

31250 1000 1 30
=
= ms
A sequencer can easily accommodate several things at once and absielen in. 1 But the tone generator
has problems with such a large amount of data because this halt serial works which means that it
receives the data sequentially. With special pots on master keyboard it is possible to activate a
so-called "Running Status".
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Running status
On sending a status byte can then be dispensed with if it would be the same as the above previous
message.
The crossed out in red status bytes are simply not sent because it is not necessary to repeat itself.
channel messages
name
Note Off
SB (Hex)
$9x
SB (Bin)
1000 xxxx
1. DB (range)
Note Number
0-127
2. DB (range)
Note Off Velocity
0-127
Note On
$9x
1001 xxxx
Note Number
poly Pressure
$Ax
1010 xxxx
Note Number
0-127
0-127
Control Change
$Bx
1011 xxxx
Ctrl. #
0-127
Ctrl. value
Program Change
$Cx
1100 xxxx
Prog. #
0-127
0-127
mono Pressure
$Dx
1101 xxxx
Channels
aftertouch
pitch Bend
$ex
1110 xxxx
lsb
Note On Velocity
0-127
Keys aftertouch
0-127
msb
Note Number
Are not divided into separate pitches. In Europe, it may be possible with 12 semitones in an octave.
In East Asia, India, etc. would not be sufficient by 63 semitones in an octave.
Yamaha is quasi standard d60 (1-127) = c3.
What to do with the note number determines only the recipient.
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Velocity
Also referred to as velocity, nothing happens other than that of the filter simply rises and closes
further.
(Digression: A bass has almost no fundamentals, the body creates that not just through the residual
effect of the tone is generated in our brains..)
Note Off
Note Off is not always sent but rarely. One uses Note On with Velocity from 0 per Note On to touch
to produce off. Thereby reducing the amount of data and you can continue to use the Running
Status for this reason you improve that timing.
Not every keyboard can send note off, but each tone generator must note understand Off.
Poly Pressure - Game Help
This function also not all master keyboards.

If one is pressed tast and you then it pushes even stronger one is in the 2nd mode. Among the key
simply is a further sensor which is activated when vigorous press. This pressure is produced as
desired tremolo or vibrato.
Control Change
Is a complex assembled command which is explained on page 12th
Program Change
Is the main reason why studios have switched to computer. One can the channel "verse - Piano",
"chorus - Organ" remote control and change the channel. Example:
We have to change a bass in the Sequencer, now you can send a MIDI command to the effect. Hall
verse = Church Hall, Hall Refrain = Room Hall. That only problem the Program Change is that there
is no 2nd Data byte, this is simply a quirk in Midi.
mono Pressure
At 10 keystroke a Poly Pressure produces 256 signals, this is just too much and does not help 10 times to
produce a tremolo simultaneously for this reason one uses Mono Pressure which is sufficient even thick
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More Match aids

Pitch Wheel band
Adjustable changes of semitones to an octave. (Made was when Jamiroquai bass this octave at
Deeper Underground.)
One can imagine this wheel so it's like working on a guitar when strikes a tone and the string with the
finger pulls up or down, use the wheel to simulate stop this movement. The downside is that the grid is
not exactly fine.
Through an elaborate procedure was ensured that the screening was refined through a kind of
"smoothing values".
The intermediate step 31-32 is again divided into 128 steps.
31 0 31 1
31 2 // / /
31 126
31 127
32 0 32 1
//
Through these values smoothing we reach a total of 16,384 different states of the Wheels.
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A pitch bend works with potentiometers (voltage dividers), which also has a AD / DA converters include
to convert that signal. Because with den16384 options line would that not pack if we played it under half
a second. Very often are not exploited all 16834 possibilities. But the lsb and msb it's just not care as
long as we play it slowly.
Control Change
$ BX> serves to parameters to control remotely. A Control

Change includes 2 Data Bytes.
The first tells the parameter to be changed and the second Transmits the value to be changed.
Some are prescribed fixed and others are individually addressed by the manufacturers.
Festival include:
07: Volume 01 Channel: Mod (Pitch

Bend) 10: Pan etc.
Very often external sound generators have fixed points for the controller value.
MIDI Learn
He knows "the parameter to be controlled by the potentiometer"
Portamento Time
If two keys are pressed at Mono sound generators, then only those who will first pressed, reproduced.
Portamento On:
Switches from erstgedrckten the second pressed button
Furthermore, one can determine the rate of alternation.
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Portamento Time
Portamento On / Off
Expansion Volume
Sustain Pedal
Pan
Continueous Control 0-127
Switch Control 0-1 0 =

00000000 = OFF 127 =
01111111 = ON
The method of smoothing values will also be used for example for fade-outs. 128 gradations for a
Fade Out are easy to make too little just gently around the Fade.
# '= Offset: d32 ($ 20)
> controller couples

msb
LSB
0 1 2 3 4 // 32
33
34
35 //
62
30
63
31
"2 bytes Controlling"
To do that for 20 years only, it is still a rarity. The downside is that the Edit in Cubase and Logic is
not possible.
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A program change is in most cases crap if you only work with a Data Byte, and thus can only control
128 parameters. Let's assume that a sound generator 500 storage locations has.
Now let's assume we want to control program 290th we need to do this rename or
redirect a space of our 128th
We tell him "program 51, you shall no longer control 51 but 290". The downside is that
we can now no longer use 51 unless we address, for example, 50 in order to react to
51st One sees that it allows you to do all the fine broken and is just always limited to
128th
Another way this problem easier and better to solve is that Bank Select.
Let's say we want 29 A and 72 C.

If I now only 72 SPECIFIED, then he will refer to 72 A and not to 72 C. For this reason, we need to
perform a Bank Select. For the exact location we define as follows:
Ctrl: 0 msb
R: 32 lsb
To save this in Sequencer we need a Program Change Bank Select +. Otherwise, you can
independently save.
Suppose that a Korg piano on Program 29 and has in the bank A. In another tone generator, it is not a
Korg piano but as a grand piano, on Bank B Program 29, it would, for example, a Yamaha stage piano.
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Whether we go first Channel 29 put on 72 and then select Bank C or Bank A 29 on Bank C 29 and then
on Channel 72 does not matter.
When Program Bank Select the Sequencer do not know what tone generator were connected are. If a
Yamaha connected tone generator, it may be that the 17 msb and lsb 61 of what is bad. For Logic
does not know this and this then causes problems at the bank switching. EXCURSION: (n) RPN (non)
(N) RPN (non) registrated ParameterNumber
Can be that 128 Ctrl. can not respond to control all parameters.
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system messages
Valid for the entire device, so do not be cross-Channel. All begin
with $ F. reason: 1 000 - 1 110 (must start with 1) Therefore, all
begin with $ F and go from $ F0 $ FF.
System Common Messages
name
SB
1. DB
2. DB
MTCqf
$ F1
value
SPP
$ F2
lsb
msb
song Select
$ F3
song #
Cue Request
$ F6
(Meaning "vocal me times please")
$ F3
A drum machine is a combiner of a sequencer and

a tone generator. This works on the pattern-song
basis. This enables us to create patterns.
ABCDEFG. Combine Patterns for the song would
look like this. DAAABAAACFGFGEA. This would
be an example.
A pattern is a musical pattern that - is repeated

several times - even in slight variations
System Real Time Messages
All system realtime messages have one and the same property. have System Real Time Messages no Data
Bytes!
name
SB
MC (24PPQN)
$ F8
start
$ FA
Continue
$ FB
Stop
$ FC
Active Sensing
$ FE
system reset
$ FF
Start, Continue & Stop Commands are.
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Active Sensing
If we hold down a key and then that cable abstppseln will sound continuously sound as long as no Note
Off message arrives. This may greatly interfere in live mode, if a cable pull from unintentional and jerking
out falls. When Active Sensing is activated, it sends every 250ms an AS signal, when this signal does
not arrive, the tone generator is then made at once. This feature works only with direct connection from
the keyboard to the tone generator when a PC is connected in between, this is not.
system reset
A system reset can define how manufacturers want.

Normally are of all the sounds Controls on normal value and panorama center etc.
system Exclusive
$ F0: SysEx $
F2: E0X
A SysEx message is a basic rule in MIDI which must not be violated.

A SysEx message must so many DB's post as necessary.
The receiver knows characterized not where this message ceases.
What is done with it?

1.
.2
Cross-vendor data, such as extensions of the MIDI standard (loophole with System
Exclusive extension of MIDI light stands etc.)
Manufacturer-specific data so that the manufacturers have the opportunity to
change any data or content to create a custom protocol.
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An ID is determined what is running.
What does it do?

-
OS update
Program parameters dumps, parameters which are played, we send to the sequencer and
recorden this. So we have a backup of our preset parameters.
Universal realtime System Exclusive
MSC - MIDI Show Control $ F0
xDB's
$ 7F
EOX
Start of Universal
SysEx
$ F7
realtime ID
MMC - MIDI Machine Control MTCff
Master Volume / Balance

Universal non realtime System Exclusive
GM In / Out MIDI File

Dump
If the data transfer of MIDI data to an external sampler with onboard sequencers. Sample Dump:
Possibility to send other data via a MIDI cable.
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synchronization
Problem:
That effect unit and the sequencer know just not how long one minute or
second. The time specified by the oscillators is important. The two devices must set the
time from 2 sources. Solution:
$ F8 (System Realtime Message) MIDI Clock (MIDI Beat Clock)
It will be sent 24 instructions between 2/4 notes. This gives us a higher resolution than
a metronome. This is also called "24ppqn". 24ppqn stands for "pulse per quarter note".
At 120 BPM 480 bits / s will be sent.
48 bits per second x 10 bits from Databyte = 480 bits.
Realtime Messages have a high priority, even in front of a Note On command etc.
Programmed clicks
For most tempo changes in a song, the change does not happen just strong. Usually only 1 to 2 BPM
is raised what one occurs as more.
Here one could now produce its own Click the tempo control to do with.
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Synchronous = Together term

simultaneously
If we press play now, then both are simultaneously active and play. The MIDI Clock but only transmits
tempo, so that although both run at the same speed, but when I say the Master Root of clock 3, it is the
slave does not understand, and this will start from stroke. 1
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The position is so uncertain.

For this reason we need the speed / clock-based synchronization. The position
transmission is at the following manner. $ F2 - lsb - msb
SPP: Song Position Pointer The
,
16
This is clear that a song not indefinitely be can. With this resolution, we
get 1024 cycles. 16384/16 = 1024 cycles. So clock 0-1023 on a 1
resolution of the SPP's 1
, This would be sufficient for almost all the albums for

16
However, excess length does not, but in these special cases, you simply increase the resolution.
Once again a summary of what is needed to run a successful synchronization to establish in this area.
$ F2 - lsb - msb
After 8 cycles of msb jumps to 1 because 8 cycles x 16 = 128 = lsb is busy, then it starts again from
0-127.
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"MIDI Clock synchronization"
Application: 2 Sequencer on different computers.

Steinberg it would not work as a slave for this reason must be slave Logic and Steinberg Master for this
reason it is not running asynchronous.
Rewire is a program which the user does not notice directly or sees that it is applied, which
Cubase but works.
Several applications which mutually MIDI data send and then to an audio file.
Software Sequencer + drum machine in one. digression:
ReWire is a software protocol for transmitting control and audio data in real time
between two musical programs. As a rule, a sequencer will speak virtually wired with a
software synthesizer. The direct data transfer saves the detour of importing MIDI or audio
files.
ReWire was developed in 1998 by the Swedish music software manufacturer
Propellerhead. The first compatible products were ReBirth RB-338 Propellerhead and
Cubase VST from Steinberg. 2001 ReWire was significantly expanded in version 2.0 and
has become by supporting all major manufacturers as standard.
There are not many advantages of the MIDI clock synchronization over other options, but a
serious.
MIDI Clock synchronization is very easy to use and is stable! Man must say "Send MC" and the Slave
"Sync to MC" the Master, that's it.
- easy
- inexpensive
- secure and stable
arpeggiator
Is a fingering which is rhythmically played. Logic can be reached via the

Envrioment window on this feature. Works only if the devices used are MIDI
capability.
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Usually one uses the tape recorder as Master.

For a deck has higher flutter and higher shunting. It adapts easily to a PC a tape recorder as a tape
recorder on PC. Another problem with the tape machine is that it does not know which clock number, is
what position etc. turn.
Furthermore, it took a Perfoband which was mechanically coupled to imagery and served for audio
synchronization. This is now used in dubbing studios.
On the tape of a tape recorder is a LTC (Longitudinal Time Code), which is nothing more than a
rectangular audio signal. This LTC Code contains location information.
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The information is divided into hour; Minute; Second and frame.
So the time code is constructed in this format: hh: mm: ss: ff. How can so
much data is sent as an audio signal? The audio signal operates with a
rectangular wave.
A square wave is the most energetic vibration that can exist because halt the amplitude is constantly at
the maximum.
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For this reason, you should stay out of the timecode of a tape from the patch bay. When it comes
also from the mixer or multi-core cables, because that crosstalk is simply too dangerous.
should take the edge track if we record the time code on the tape machine always. Let's say we have
a 24 tape track machine and take on channel 1-6 on, then the time code should be on the 24th
framerates
(Subdivision of the second in pictures) fps
standard
24
Cinema (international)
25
EBU ( e uropean b road cast u nion) PAL / SECAM
30
SMPTE (NTSC b / w)
30d (dropped frame)
SMPTE (NTSC color)
Audio: USA 30 fps

Europe 25 fps
The only problem with this case is the conversion of the time on the clock. Time is not the same
clock, though he knows that minute etc. but how is he to know is which clock this minute?
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The following illustration explains us that problem:
TC offset shifts without tape on the clock position 1. (biite CORRECT) For example, one of the tape machine
to say that the timecode but should run along the immediately click in the sequencer after 01: 00: 00: 00th
1 hour, usually always set and then allowed a buffer of 1 minute, from the start of the time code and
the beginning of the synchronization of a click on the timecode (ie from 59: 00: 00 00 - 01: 00: 00:
00) , BAR Position 1: 1: 1: 1:. Plays at (Sequencer)
The tape machine turning as long until you reach the Info (timecode) has. Task:
Tape with LTC and drums to be recorded to a click. Sync the Sequencer.
(MDA application 24)
First you should adjust the offset.
establish synchronization connection. (Master say he should go put and say slave "run
along now")
Listening to signal to the master
set Click or metronome in sequencer to position (1: 1: 1: 1 :)
Watch TC display, plan
approximate value of the offset entered in the transport field of the sequencer.
rewind and check whether the click at offset start mitstartet.
vary offset until the click in the slave matches the signal of the master or is
successive.
-
Setting the pace of the slaves on the tempo of the master. This is
done manually (control by click / sequencer)
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Preparation: Clear tape, delete 23 tracks when Deleting, on the

24TE record timecode.
In Sequencer we can set a time code generator when the time code is to start and how many fps he
should have. Default offset for the LTC is 01: 00: 00: 00th The remaining 23 tracks should be armed
and put the 24TE on the play mode. The deck starts to recorden and simultaneously track 24 (LTC)
again.
The click is on the ears of the drummer who click the synced slave plays to the master.
If the drummer played any wrong we run the tape and then he is simply at 01: start (as an example)
00: 02: 50th There is also a so-called offset View, this shows another as what is actually there. (He
puts it simply, when in fact 01: 02: 50: 00, View Offset can but 01: 00: 00: 00 show).
One can at the slave and that the Transport and decouple via MMC (MIDI Machine Control) the master
control.
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The procedure of the LTC to MTC conversion
Functionality:
The 8 MTCqf commands are sent in time equal intervals. is the distance from one another 14 frame.
Therefore is the MTCqf = MIDI TIME CLOCK quarterback frame command.
Suppose we fly the band in a frame which is delivered with> not. (CORRECT PLEASE)
The Synchronizer counts the rate of zeros and ones, this stock will be immediately implemented in
the MTCqf.
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VITC:
vertical interval timecode, vertical is also often translated as video but this is wrong. The
time code can be integrated into the video signal.
Often a timeline is displayed in the screen, which is used for synchronization. This is practical for the
setting because you can compare the time data in parallel to a timecode position.
Helical
In the helical scan to fit more data on the same tape, thereby the head must also be inclined, or be
oriented obliquely.
In this process, the head works with a separate read head and a separate stylus, this allows the
playback of a still image. Advantage:
One can in the pause mode, the image read VITC.
With fast reading can not be read, so no pitching possible. In LTC to 10-15%
pitch possible.
Disadvantage:
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For audio signals that is not half as easy pitching. You can not just tell the software in audio signals,
delaying the song!
Because the only parameter that determines the speed of the read operation is the sample rate. must
be driven externally What a World Clock. All audio interfaces with a World Clock In input have the
possibility to control the sample rate from the outside.
Synchronizer
Emagic Unitor 8 MKII

- Multiport MIDI / USB interface (8x in, 8x out)
- reads and writes LTC and VITC (+ converts them also)
- burn in
- "Manual sync" (click in, bpm counter)
- software control (among other things of Logic)
- AMT support, active MIDI transmission
C-Lab Time Machine
Digidesign USD / Sync I / O
Rosendahl Nenosyncs
Stramp Adam Smith, Zeta Three Synchronizer
- to sync the only true box to 2 tape recorders, is extremely expensive, but also to
operate by remote control.
MIDI Recieve Modes

There are 4 different modes.
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Omni On, Poly (. Not transmitted possible for polyphonic, because the channel code
will be ignored and no Multitimbralgert possible) Application: 1 performance of the entire CPU's is
applied for 1 Sound and thus
is the sound of better quality and more complex. The entire CPU power is not
Mode 1:
shared on 16 sounds but it only works for a sound, this makes sense if you want
to be layered stuff and high quality required. Mode 2:
Omni On, Mono As Mode 1, only
monophonic.
Mode 3:
Omni Off, poly
a) polyphonic on a Receiving channel I can define.

b) to polyphonic several (or all ) Channels.
The nickname is "multi-mode". (Is the most common mode for digital sound generators.
Application: appliance Thru chain. Mode 4:
Omni Off, Mono

Monaural on multiple channels.
Application: MIDI guitar, MIDI Bass to each channel's pitch bends to
enable.
Recieve modes can be set on the tone generator.

Also with effect devices and digital mixers to recieve modes can be set. Example:
Reverberation time, Delay etc. which channel where the mixer etc.
Switching of modes can also be remote control. The last controller number to control this function.
= Channel Mode Messages (all modes)
MIDI Implementations table

It is very corrosive every time to look in manuals, where the function of our acquired MIDI device is. The
manual also is not what the machine can not but hold only that what it may. In order to save time and
effort we find out, we use the Implementations table. Because this is first standardized, and includes a
summary everything what can send and receive the device.
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Daisy Chain derailleur
This type of wiring is seen today as a kind of "last resort" to. MIDI therefore works with optocouplers
after galvanic principles. For this reason, you can make MIDI switches without breaking something.
You can pull the plug without damaging anything on the fly. MIDI makes it mind if we join in with in
and out with out. The disadvantage of this optical coupler is that these signal components filtered out.
At 3-4 devices in a chain, it is still in its entirety, in order. To know when it is harmful, does not need
me much to do. There is simply nothing were heard. Well away either a device which stop is the least
important of the 4th or changing the order of 4 units.
Because, if it works, then it works, we will not get a lower quality signal.
Multiport Interface
The current solution is that use of a multi-port interface.
All of the 4 outputs have 16 channels. (Total 64 channels), often one sees the interface as 2 (inputs) x4
(outputs) or as 8x 8x.
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Thru Box (Merge Box)
With such a Thru box or Merge Box, you can split a MIDI signal and send to two receivers.
Only one major drawback, this type of cabling.
If a receiver has a larger concerns and even purely be as it should, then that other is not supplied. For
this reason, the line was divided by priority.
See next figure:
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The data may arrive late, because the signals halt arrive primary and secondary.
The one who has to wait is in the cache. In this case, would in A priority. It is better if the
control of the controller comes too late in this case, as a Note On message from the
keyboard.
Disadvantage:
Local Control
should be set to Off If we use the sound generator in Logic the Local Control function, otherwise we
get comb filtering because we play 2 times with a time interval slightly. This only applies to
keyboards with integrated sound generator.
This function can also be remote control, also applies partially for digital consoles or synthesizer.
"Tape machine principle"
Tape machines principle works with bars and beats, the song Pattern principle has a spontaneous assembly.
Pro Tools, Logic, Cubase operate on the tape machine principle.
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Sequencing 2
Links under the MIDI Thru command, the parameters (standard terms is because GM unit
1).
Port or output port, the ports on the interface. Channel number, you can make all or select between
1-16. Transposition, serves to semitones. All the selected track to halftone, Oktava increase etc. or
decrease. For the bass is often transposed octaves.
Score limit, ideally located close to the keyboard keys exactly to split in half, and then the left half to
the left and right panning one to the right. In Program is the first number which bank, ie Bank Select
and the second changes the Program Number.
Enviroment
The Enviroment in Logic is the programming environment. Also

known as input & output processing. Recording and playback are
controlled here. Which devices are how and where connected.
The enviroment is divided into layers to represent a kind of order and ease of handling. (Layer as,
Click, port, audio, instruments, etc.)
The settings that are made in the enviroment, are also in the suffix. lso stored.
Under the menu item "New", you can just add new objects such as faders, pots etc.
In the transport field is right above the entrance and exit, each transport is firmly held there.
Select synths, but how?

First, we create a new layer and choose MIDI instrument. After a standard instrument in the
Parameter box, we can write the name of the synths in and select an icon as desired. The Port we
provide a way of Synth is also set.
The object in the enviroment is the same as in the Arrangement Window, all parameters are 100% identical.
Page 162 of 466
In a multi-timbral tone generator then selects Multi Instrument. With a double click on the device, you
can make additional settings, you can change the names as numbers those not assign names those
annoying General Midi.
As Key mapping is meant instruments with no tonal pitch, like a drum set, for example.
With a fader in Enviroment can Program Change, Control Change SysEx etc. ship.
The tick box next to the icon will decide whether that object in the track can be selected. Hook, then
he can choose to not hook, then he can not be selected in the track.
It also has the ability to import individual layers.

If we are going to use Global objects then appear in all windows. Let's say we want a change of pace
from bar 5. Then we open the menu Tempo> Tempo as a list. Then we can with a pen set the pace
and tempo. To want to change the tempo too abruptly, we use the function View> Global Tracks>
Tempo. Then go to the clock change on clock and FILLIN manually in the Transport bar that pace
File> Song Settings> Synchronisation (Short Cut ALT-X). MIDI Clock Send
Objective 1 make a hook. For, EDIROL etc.
Step Recording is pretty cool, by MIDI One in Matrix Window can recorden piece by piece.
Song Settings> Recording to lay> new recording in selected region.
> then references to either lay or overwrite halt on that old bar. The Hyper Editor is awesome for
the drums.
The Event Editor is the core of Logic, were in earlier systems that sequencer. In the transport box
you can perform operations and select Events. In SB Sheet, channels, etc. can be chosen or
reverse the half speed of 2 songs.
Or Humanize Velocity, Volume +, - 10, etc. can also select specific. Let's assume we have a
hi-hat over 170 bars,
29 are shit and need a higher velocity, then we can say that all attacks now be aligned with a
velocity of about 80th
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There is one significant feature which is as follows: "Take last catch as receiving."
Each of us knows that we sit in front of Logic, and try what einzuspielen nice, at a time we play a
fantastic take, but unfortunately we did not record ready pressed. With this function we get the last
notes played again and there we have it.
Audio Features in Logic

Backups are incredibly important. Because going to copy and move data solely with the inexplicably
lost or broken. For this reason, one should burn things multiply or copy to another medium. And after
that you should file for 3-4 times continuously to make sure that no data is defective. Logic works
non-destructively to the audio files (except in the editor mode).
Interface for plug-ins

The interface of the effects and virtual instruments called AU (Audio Units). In ProTools RTAS
and TDM interfaces hot. Each manufacturer may use their own names.
The advantage is that this AU are part of the operating system Mac. The AU interface is originally
from VST (virtual studio tech.) From manufacturer Steinberg.
Since version 6 (Logic), this VST can no longer so easy to use plug-ins in Logic. A Russian built
times a wrapper with which you can use VST to AU. This wrapper is called "VST AU wrapper" of
the company Fx pansion. The big advantage of Logic is the coupling of the audio files and MIDI.
Because the operation of the mixer in Logic is the same for MIDI as well as audio. Here is an
illustration which explains the procedure of MIDI in Logic:
The vulnerability of Logic is quite clear that you can not move exactly in the Sample Window
Arrangement. Because Logic scans to MIDI clicks. This is a major point of criticism against Logic.
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The crossfade tool
Such is the symbol:

You simply draw the gefadeten area and lets go of the button.
To delete a fade pushes you the Alt key and click the desired fade. The crossfade will be saved as a file
in Logic. All fades * .aif saved with the suffix. When one plays this file once you hear all Fades which
have been inserted song. Logic bounced virtually fades into an audio file and all fades are appended
sequentially. So that the performance of the computer is not run in the basement. Because the read of
the fade, then do not read 2 audio files, but then when the fade begins switches the computer from the
first audio file on the Fade Files and then on that second audio file, the load remains low. In Region parameter window you can adjust the curve of the fade. Whether an abrupt fade or linear etc.
The HyperDraw not work as safe as a fade. The Hyper Draw created the Fade 1 to 1. During playback,
the fader is moved from the mixer in the system while a fade an audio file is created which were offset
in advance. The crossfade information are always in the first region.
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Equal Power Crossfade
Other editing options we get in the editor. By double-clicking on the

region we come to the editor.
The stereo tracks in Logic 7, but are provided in the track with a region with a stereo logo. Previously,
that so that a stereo file 2 tracks were needed with each Pan extreme left and right. The current
waveform display in this stereo files with a region is such that only the positive excursions from the left
channel and from the channel rights were added in a representation.
The positive range of the L channel is up and the positive range from R channel is in the negative range
of the stereo area.
we get this view in the Arrange Window.

which aufhallten case signals a little longer in the positive range and then the change in the negative
range, we are not seen in this view. The following signals are meant:
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To such to recognize a curve sensible to open the editor and you can see the original curve of the
signal. Furthermore, one can in Editor Preview.
You can also set the over which channel number this Preview is issued. This function is used to
practically hear that signal in bypass mode. So you can through sensible allocation of channel
number that signal in the editor "dry" hear, or with the effects.
In large sessions is 1 always left free. If you want to hear with effects, only then should differ from the
standard and set the channel number.
DC offset
Furthermore, you can remove a DC offset in the editor. Not only in mastering this is important, but also
in the recording of audio files over the mic. By that remove the offset we get a peak gain and a
somewhat greater security of oversteer.
session management
Audio window, the tool is to manage that are used in a session to all data. Man, the Regions, the
watch can be used, and then possibly to optimize the data.
Optimize files
The field of non-used saying is not used in a session is discarded by this function. This is
Destructive.
Audio Object in Enviroment
With an audio object in Enviroment everything is handled what has to do with audio files. Let's assume
we need 70 audio tracks, but we see only 64 in the Arrangement Window, so we open that Enviroment
and even add 6 Audio add objects, then we come to the 70. Enviroment can also busses, aux, master
fader produce etc..
Among other things, you should know that is used in Logic as a channel that word track.
Page 167 of 466
tempo change
Apple Loops Utility is used for tempo adjustment. One
uses elastic Audio> tempo change.
recording time
To extend the time of the recording session you go to the (receiving path) option there you can adjust
the length.
Logic will know from the beginning purely just about how much to expect him. During the recording
operation can be time does not extend, the session is terminated accurate to the second.
project Manager
The project manager is a kind of database, which remembers the files needed to your song and
which fade files belong to which project.
Because not that someone accidentally deletes a snare and wonders why they are no longer heard in the
song. So it is better to delete front open the Project Manager to be sure to be no path accesses that you
want to delete file.
automation
It can also be set up in an Enviroment Automation. About MIDI cable output (wiring in Enviroment).
Previously it was done that way. The object "Monitor" in Enviroment stores when you change
something in the channel strip of the instrument fader messages. These messages are compatible with
anything that they run only in Logic International. Basically this is underwhelming, but you do it now
rather in the region area, or track based. Through the "View"> "Show Track Automation". Possibilities:
1. With a pencil draw the Automation

2. Software Recording Controls
3. control by hardware
Automation modes (above the Pan Controls)
Read =
Off =
touch =
Read, reflect
The last selected fader position is maintained
makes corrections if we have automated what. One knows this mode among
others of the faders at analog mixers, where the faders have so metal caps. These caps
have a high frequency low voltage and when
one with
the finger touches this cap he realizes that and goes into
to write
Mode. Who do we let him go, he will return in mode
"Read".
Furthermore, this function operates so that if we let go of the finger, the
Fader slowly goes to the old value. So happens
these
Movement not jerky and abrupt. This time how long the
fader needs
to reach the old value is called "Ramp - Time".
This can also
Setting manual.
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Write: Existing automation will be overwritten, it overrides all

Parameters, with the newly set value.
Latch: In principle, operates the latch mode as the touch mode. He
reads when you touch it and what changes, then release it again, he remains at the last value
and writes it as long as until the song is over, or you touch the fader. Touch: reads all the
parameters further and change accordingly what the latch mode
does not.
The automation is also possible with the plug-ins. You ask a the touch mode and vernderst your
effect. This saves the parameters and plays dese afterwards in read mode. can be more drawn
automation points to trim, and create copies of them with the Alt key, the Shift key.
Automation tools
One can thus change the range between 2 nodes. This one uses among others for fades. This keeps
you change the curve between the nodes.
Car traces Zoom
The track which one is clicked enlarged. Options
track Automations
> Track automation settings
Move Automation with region, "ask" choose.
Now he asks us every time we move a region in the Arrangement if the highlighted node's or
the automation is to be moved to. Other settings are Ramp Time, Quick Access etc.
With Quick Access, you pitch wheel, for example, that use the master keyboard to individual
parameters like volume Pan to control remotely.
Freeze function
Audio tracks and Audio Instruments are frozen. The performance is optimized thereby. He bounced it
passively and plays it in the background. Normal Mixing is possible with frozen tracks, ie volume, pan,
etc. In order to automate buses must be a hook in the Icon field turn make it in the selection bar appear.
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I / O Settings
Groupings under I / O settings in the channel are possible. Groups can be set, are for common
Record Ready switch etc. An object can belong to only one group.
Overview that Routing

If we receive in Audio> Audio Configuration
MIDI File Export
To exchange data created in Logic with other sequencers, to export it as a MIDI file.
File> Export> Selection as MIDI File

(Previously you should select that what you want and export) Regions
parameter> Convert Loops to Real Data.
Serves to created alias in audio convert data, otherwise you can not use it to create an entire file.
Region Parameters> Apply quantization destructive. In order to halt can stick

all regions.
Then we glue all Regions to, otherwise there are problems with the use of another sequencer.
The format is "1 File Format" saved as. (Distinction of

tracks works) format 0:
Distinction of tracks does not work, all in one file, this format is very rare in
more hardware sequencer.
To export audio files you go to "File> Export> as Audio File".
A major difference between Logic and Cubase comes to MIDI is: Logic's MIDI without an audio track
directly from the track again. Cubase not, Cubase yet produced to the MIDI track, an audio track.
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Main Cubase
Transportation> Metronome Setup Project>
Project Settings Project> Tempo track (tempo
changes)
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MIDI
Audio features of MIDI:

Interface for plug-ins
AU (Audio Units)
RTAS (Real Time Audio Suite)
Pro Tools
VST (Virtual Studio Technique)
Steinberg
The sound generation of MIDI is done internally in Logic, but operates only MIDI Tick accurately and not
sample Exactly
Fades (In / Out / Cross) be made with the Fade tool. Fade Files are created thereby. Curves are
constructed in the region parameter. Crossfade information are always in the first region
EPC (equal power crossfade) means that the transition is no drop in level carries with it (- 3dB point)
Good Function: DC offset! Removes the DC offset in the original rehearsed instruments
Earlier, the snapshot automation was used means you only mixed chorus, then verses etc. together
and put the pieces in the mixdown sequentially together on the 2TK
Track automation:
Read
everything is read
Write
overwrites all automation parameters
to edit previously recorded automation, it is so long
touch
read until a value changed actively. From the time of letting go up to the original value is called
Ramp-Time Latch
similar to touch, only this automation to the final value after
Releasing the controller continues to write the last value
Midi File Export (exchanges with other sequencers) filing Export

Selection as Midi-files ...
Since loops and aliases etc. are unlikely to be read by other sequencers, they must be converted
into proper data
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normalize parameters of the region; Quantisierungspunkte destructively apply; Then the Export is
possible.
Differences, this is still in Format 1, where the distinction works in tracks, and Format 0, where it does
not.
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sampling:
Sample:
Example, sample (gen.)

Scan (techn.)
Sampling rate (sampling rate) [Hz]
. 5 indicates how often an analog signal is sampled in a second.

. 6 Ex. CD 44.1 kHz = 44100 samples per second
. 7 the more samples ( "Sample"), the more accurate the digital image of the
original vibration
8th. Nyqust or Shannon cal sampling, Sampling Theorem
. 9 fs> = 2 x fmax <=> fs / 2
Sampler:
. 2 Main task:
Imitating natural instruments

each instrument can be played by a sampler, also instruments
1. not available (in real terms)

. 2 the musician can not play
- > Sampler is even "musical instrument"
. 3 is a tool tontechnisches
. 4 generates digital audio recordings (samples) and outputs it to any
Time and any pitch again.
. 5 Control of the sampler:
1. the information
1. what pitch
. 2 when
. 3 how long
. 4 which velocity (Velocity)

5. etc.
receives the sampler via MIDI
. 3 Audio is output in real time, immediately. For this to work, this must all
Sample the Ram of the sampler are (case of hardware samplers)
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Functionality:
. 5 a "sound" is picked up
1. HW: similar over Rec button Recording directly into RAM
1. then saving possible.

. 2 SW: recording with any program, then re-loaded into the sampler
. 3 Definition of the original pitch (Root Key) (original pitch) (Original Score)
1. Ex .: E4
. 2 the overlying or underlying pitches are "pitching" by

realized: the sample is played faster or slower.
1. > Pitch change
Problems in imitating natural instruments

Already a few notes above or below the original pitch sounds
. 3 Problem 1:
unnatural.
sample
- Length or duration of the samples varies
1. reasons:
- the formats shift thereby. Formant: characteristic, noisy frequency bands in

the spectrum of an instrument or voice that will always have the same
strength regardless of pitch.
1. Mickey Mouse Effect

Solution 1:
one uses a lot of samples in small intervals
. 4 individual samples must be pitched only slightly up or down.

1. multisampling
. 5 The more samples, the more authentic the instrument will sound
The area is located in the only 1 sample, called Keyzone (mostly)
- > note polyphony!

The assign samples to keyboard areas (Zones) called mapping. To cover transitions between
zones one sometimes uses "Zone crossfades" Problem 2:
Instruments do not sound the same through your entire dynamic range.
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BSP .: Piano: a stronger attack leads to a "brighter" sound as a weak attack
Solution: same pitch will be sampled in several volume levels.

-
Each sample is assigned to the sampler a Velocity Range.
This assignment is called "layering"
If this Velocity Ranges do not overlap, it is called "Velocity

switches "
Again, the samples can overlap. "Velocity crossfades"

"Note polyphony"
The more samples, the more authentic the instrument will sound.
Memory requirements:
Each sample must be loaded into the ram of the sampler to be abspielbereicht.
BSP. Piano: buttons 88
4 layer
20 sec sample time per sample
44.1 kHz sampling rate 16

bit stereo
The space has 1184 MB. This is more than a hardware sampler ever owned. Because space is
always limited, the number of samples is limited and hence the authenticity.
Loops
Loops are in the Sample (defined by user or by the manufacturer) that are played repeatedly. 2
reasons:
save space
To certain instruments (such as strings) playing arbitrarily long "hold" to

can.
Page 176 of 466
Start and end point of the loop should not be heard or noticed, of course. Again crossfades are
possible.
More (re) processing options
- (Start and end time set)

- Filter (High and Low pass etc.)
-
Envelopes (ADSR)
Modulations (LFO)
Sampler need audio, synthesizers use oscillators for sound generation. Otherwise, they are
largely identical.
Sample Library
-
professional make sampling quality samples
often tailored to specific sampler manufacturers (eg AKAI S-1000 format)
Sampler other manufacturers usually have an import function. though

can not all settings are imported.
story
-
1962
Mellotron
-
first "sampler-like" instrument
per key a tape (max 8 sec), 3 Sounds
complicated, error-prone mechanics
1970 "cheap" M400

-
1979
coined the sound of many 70s bands
Fairlight CMI
-
first Sampler
Sampling rate: 24 kHz
Resolution: 8 bit
Keyboard, monitor, light pen, keyboard
Pattern Sequencer
16 KB per sample
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1981
E-MU Emulator
-
1. To some extent affordable Sampler
128 kb memory (RAM)
max sample Lange 2 sec
27.777 kHz, 8 Bit
4 times or 8 times Polyphony
no Midi
49 keys
1985 Akai S612

-
affordable Sampler
19 "format
MIDI
4-32 kHz, max 8 sec Sample Length
12 bit
6 times Polyphonic
128 KB memory
80's, 90:
-
more manufacturers
better Specifications
No major innovations
1999 Nemesys Gigasampler (Software)
1. Sampler with disk streaming technology
direct stream of samples from the hard drive (only a small part needs to Ram
Loading)
high quality sample libraries available
Page 178 of 466
Benefits of SW over to HW
more comfortable user interface
better
greater involvement in sequencer (Total Recall)
multiple instances possible
Benefits of HW versus SW
crash safety
possibly transportable
Main Software Sampler

-
Steinberg Halion (3.x) (not Intel Mac compatible)
Native Instruments Contact (2.x)
-(
E-Mu Emulator X) in conjunction with HW
Apple (Emagic) EXS-24 mk II
IK Multimedia Sapletaule (2)
all with extensive sample library
Technical specifications:
-
Disk streaming
multiple instances
at least 24-bit / 96 kHz
at least 64 times Polyphonic
VST or AU (Apple Audio Unit) or both
Page 179 of 466
MTK HD Recording (Pro Tools)

Pro Tools HD requires special hardware:
1) Core DSP card (at least 1St.)
2) Accel DSP card (0-2 hours), the DSP
cards relieve the CPU
Mixes and partial effects are expected on them PlugIns are TDM
(DPS's), RTAS, Audio Suite DSP stands for Digital Signal
Processor TDM means Time Division Multiplexing
RTAS means Realtime Audio Suite (Native calculation CPU) Audio Suites
Offline ins and destructive
The plug-ins do not have the 96 I / O and 192 I / O can be used and purchased The interfaces must
run in sync and need to be connected via digilink. This is done via an external syncer
For interfaces, only 16 ins and 16 outs are used
Configuring a Pro Tools HD system
Set up Playback Engine Hardware

Buffer Size:
The larger, the more Buffer Size The

bigger, the higher latency
The larger, the slower the view will be greater, the more decreases
the Midi-Timing
RTAS Processors: indicates how many CPUs for calculating RTAS plug-ins are available
CPU Usage Limit: indicates how may take a lot of CPU Pro Tools Number of
Voices:
Voices are simultaneously reproducible mono signals maximum 192
Voices
1 Voice is required as soon as a region in the track's dynamic

voice allocation (switchable)
if behind a TDM is an RTAS plug-in be used 2 Voices Delay Compensation Engine: To
compensate for latencies Most setting Short
Page 180 of 466
hardware Setup
You can adjust the sample clock, ie where they come from (internal / external) sample
rate is adjustable only when no session is open digital format setting (Interface Based)
operating level can be set (+ 4dBu / -10dBV)
I / O Setup
Path:
- we can summarize specific inputs and outputs

- The names of the paths do not say anything about where they go, as you can route all as you want
- used for naming and grouping of inputs and outputs
- you can set up inside a path a Sub Path, ie you click the 5.1 path that was created on, then clicking on
New Sub-Path to create a stereo path there
Surround mixes
Voice must always Mono from the center come, a phantom center is garbage, because if you only slightly to the left
of the image sits, one has the impression that the voice does not come from the front
Audition Path, specifies is cued on which path.
Types of tracks
Audio Tracks: - any format from mono to 7.1
- Audio files are basically mono
- Audio Regions may include
- Regions that comprise a complete Audio File
Aux Inputs:
record or not insert possible Audio Files

Dry signals for internal and external instruments
Subgruppenmaster
Subgroups are Bouncebar
Instr. Tracks: - internal and external MIDI tone generator
- only MIDI recording possible, but you can bring in PlugIns

MIDI tracks: - are nonsense, there are instruments Tracks Master:
- it can only change volume and can use plug-ins
- the inserts are POST FADER
ON MASTER FADER IS A VOLUME CONTROL ON OUTPUT
Page 181 of 466
General and commands
It would in principle extend the audio files and the session to pass because Pro Tools the rest can
automatically add! ALT: means do to all 'ALWAYS
ALT + APPLE: do to all ascending (channel assignment) ALT + SHIFT:
do to all selected
Bypass circuit does not share DSP power, off already
Headphone mix in Pro Tools:
On all channels Send (PRE)

To completely take over channel settings in the SEND: EDIT Copy to send (ALT + APPLE + H)
In Group Sends NOT be moved compensating an RTAS plug-ins will
not run on the DSP's AuxInput always SOLO SAFE
SHIFT + APPLE + I: Audio to Region List ... always COPY Audio Suite are
destructive PlugIns
ALT + APPLE + T: reduce THIN automation points
For complete Fades in a session as a general volume reduction: it is advisable to make the
pre-mastering ZOOM with R + T
Zoom settings can be stored in the markers When

leveling Prefader Metering Auto Input OFF !!
When punching out the input mode out !!!! (ALT + K)
In all groups Editierungssachen effect the group of a loop must be at least 500 ms
In the transport window off the conductors, characterized the tempo can be changed
Page 182 of 466
The modes
Shuffle: the outside cutting the region automatically adds together Slip: move Sample
aligned
Grid: frameshift (adjustable (MM: SS / B: B))
Spot: When the Region click will automatically open the SPOT dialog: settings of the exact start, sync
and stop time by B: B, TC, M: S
Move the regions and selections in slip on shift, apple, Ctrl and Alt in conjunction with + and
- Sync to External TC:
make setup Peripherals Device on Disk Reader allocation: tells
where the audio files are to be Setup Menu
- > Playback engine

10th Hardware Buffer Size
1. Disadvantages of excessive buffer size:
1. PlugIn Automation is no longer as accurate
. 2 Midi is out of sync

. 3 higher latency
best you can set the buffer size as small as possible (default is 512 samples)
RTAS Processors
Standard is all
Determines how many processors can use Pro Tools RTAS plugins
CPU Usage Limit

Determines how much processing power Pro Tools must make maximum use of
Number of Voices
As voice is called the playback capability for a mono track Voices require DSP Power
A Protools HD system can have a maximum of 192 voices. A session can have a maximum of 256 tracks.
Protools has dynamic voice allocation (Voices of unused tracks can be used for other tracks) (happens
in the background)
Adjustment next to the fader in the Mixer (bottom left) (on Button is Dyn) Delay Compensation
Engine
Page 183 of 466
The track with the highest latency is retrieved and the other tracks are then delayed and thus
adjusted.
Delay Compensation is always off in Record Ready tracks. When recording the best off the
compensation. When mixing well. settings
- Off
- Short One can compensate for delays in 1023 samples
- Long One can compensate for delays of up to 4095 samples.
Normally, you short, unless the plugins have too high a latency If it is set to long, is less DSP power
available. DAE (Digidesign Audio Engine) Playback Buffer Size (Heart of Pro Tools)
Controls DSPs, controls the reading and recording audio etc.
Protools is quasi the control surface for the DAE

Logic can these DAE access and can therefore TDM plugins use The larger the buffer, the
more can incidentally happen on the hard drive DAE Error-9073
Disk to slow or to fragmented
Solution: DAE Buffer Size make higher (default is 2) Slow disk, many
sections, many traces -> Big Buffer Hardware Setup
In the selection menu for the input the interface inputs are not visible, but paths. As path (Path) is
called a summary from one to eight or outputs.
Paths are set in the I / O Setup masterfader
In the mixer, a headroom of about 40 db is for all inputs

is the clip border 0dB FS plugins also have a clip limit of
0db With a master fader are the ISRs post fader Session
Folder
The session is in the session folder
Page 184 of 466
The audio files in separate folders The Fade

Files The session backups
Wave Cache.wfm file -> in this file is the graphical representation of the waveform stored
mix Window
A maximum of 32 levels of undo
minimum loop length in Pro Tools are 500 ms. Among no loop is playing. An Absolute marker
refers to a Fixed Time.
A bars and beats marker refers to a specific clock time Time Based Ruler
playlist
When a playlist is called a play order of regions on a track not with in it is automation etc
Edit Modes
Shuffle
sets Regions always adhere to the previous or the following region. (Regions on shock position) If
no region is the shift ends before the region, they will be moved to the front.
Slip mode
Move Sample aligned and position possible. spot mode
Spot is most useful in post production. Once a region is touched with the grabber, the Spot dialog
opens. The region can be moved based on numeric input.
Original time Stamp
If co-written Feiles when recording a wave, therefore knows ProTools where the audio file
originally (actually) belongs.
Sync Point
Used to synchronize HotSpots adding with apple +, or

the region menu
Auto Spot mode in the
options menu
Page 185 of 466
When this mode is selected, a region when selecting is pushed directly to the current timecode.
If no timecode is present, this is 0, and thus the beginning of the session.
Grid
Move possible to the grid. Particularly well-suited for clock-based work. The grid can be set
arbitrarily.
When relative Grid Regions can be moved relatively from the current position to grid values.
The grid is normally determined by the Main Counter, but can be changed synchronization
If Protools to run as a slave to external timecode, menu needs in the Peripherals in sub-item
synchronization, the Sync device can be adjusted. Usually this is the Generic MTC Reader
So then does the synchronization, ProTools must be switched to the online mode.
If Protools will be the master, it must setup menu are set in the Session Setup -> Session
Here, the frame rate is set, among other things.
In the Time Code Settings needs to be selected for transmission of the MTC of the Midi port.
Page 186 of 466
Signal Flow 3 - Studio 4 - Mackie DXB

The Mackie DXB:
is a digital console
is an existing out-shelf components PC

if required components can be exchanged / Improved
through the built-in fans make a sound
Operating System: Windows XP Embedded
Drivers can be updated / Installs

Supports VST PlugIns
Different versions of the DXB:

1) X.200 (SAE)
- up to 72 I / O
- 8 buses
- frame silver, black
2) X.400
up to 96 I / O
24 buses
gold, brown frame

UAD1 card is included
DXB X.200 Overview

Connections:
2x USB (mouse, keyboard)
1x Ethernet (currently has no function)
1x MIDI In, 1x MIDI Out

1x 9-Pin
2x footswitch
audio connectors:
- completely modular system
- 10 slots (X.200)
- User determines the nature and the number of I / O
- future connections Retrofit
- when fully configured relatively expensive I / O
cards:
- 11 slots
- SLOT A: Cards 1-3 (any configuration)
- SLOT B: Cards 4-6 (any configuration)
- SLOT C: Cards 7-9 (any configuration)
- SYNC: Sync Card
Each SLOT-range (AC) must not exceed 24 Ins + 24 outs include standard case
are Sync card and MixOut
Page 187 of 466
Possible I / O Cards
1)
Mic / Line 4 card
- 4 x XLR (Mic)
- 4 x jack (bal.) (For Line)
- no outputs
2)
Mic / Line 8 card
- 2 x DB-25 connector
- 8 x sym. Inputs
- 8 x sym. Outputs
3)
Line Card
- as Mic / Line 8 card, but only for line signals

4)
Digital Card
5)
8 x In and Out but purely digital
T-DIF max. 96kHz (DB-25) (All other to 192kHz

ADAT connection (opt. Toslink)
Assignment depends on sample rate differently
AES / EBU Card
- 8 x In, 8 x Out
6)
FireWire Card
- Max. 24 x 24 x In and Out (48kHz)

- Max. 8 x 8 x In and Out (96 kHz)
7)
MixOut Card
8th)
AES / EBU 2-channel I / O (XLR)
SP / DIF 2-channel I / O (RCA)
2 x 2 speaker terminals (sym.)

2 x Phones ports (unbalanced.)
MixOut L / R
SyncCard
- Word Clock (I / O)
- SMPTE (I / O)
At the desk in the SAE following tickets are available:

- MixOut, Sync (which MixOut is not connected to the boxes)
- 3 x Mic / Line 8 card
- 3 x Digital Card
- Speakers are connected to the buses for 5.1 !!!!!!
Page 188 of 466
User interface of Mackie DXB

- 2 x 15 "touchscreen
- 24 faders and 1 master fader
- Fader: Bank Selector
- 1-24
(SAE: signals for recording)
- 25-48
(SAE: signals for mixing
- 49-72
- MSTKS:
- Groups:
- MIDI:
12 Aux Send Mastewege

12 MIDI channels, 12 master Groups
Remote control of sequencers (Pro Tools, etc.)
Construction of channels:
20 METRES
- shows the level of (input, pre- and post-fader)
- Alternating between fader bank and global view by "tapping"
21)
ASSIGN
- internal routing (L / R, buses)

- opens Assigns window
- Choice of input source:
- any input a map
- other strip
- Aux Send
- bus
- Talkback, oscillator
- ISR: - Floating Insert Send
- Floating Insert Return

- Phantom power
- Digital Trim (directly after conversion)
- Direct Out
22)
AUX SENDS
- 12 aux sends:
- 1-8 mono (or stereo)
- 9/10, 11/12 always Stereo 2 Cuewege
- Enable / disable the Aux Sends
- Level via V-Poti
- Pre / Post circuit
23)
DYNAMICS
- 64 channels (max 96kHz)
- Gate / Expander, Compressor, Limiter
- CPU utilization (as DSP) in the right window above
- touch VU meter opens Dynamic Section
- usual parameters, additionally: A / B comparison, copy / save, sidechain
function
Page 189 of 466
24)
EQ
25)
Tap to open it window

4 band EQ fully parametric
usual parameters, additionally: A / B comparison, Morph (a period (0,1-30s)
is faded, Menu
SURROUND
26)
Overview always shows the frequency response
Tap to open Sorround window

simply draw desired position with the finger
Global Surround Mode: 5.1 or stereo
Outputs are automatically assigned to the buses
V-POT section
- Representation of the currently regulated by the V-Pot parameter
- possible parameters in normal channel mode:
- PAN L / R
- PAN F / B
- AUX SEND
- AUX SEND PAN (9/10, 11/12)
- Digital Trim
- Level to Tape (Direct Out Level)
- Selection via POT Controls (right of the TFTs)
- adjustable to other functions menu
real existing controls:

26)
V-POT
27)
SELECT
- selects a channel for editing

- expresses two Select keys simultaneously, you can generate them link (Link
window opens)
28)
ASSIGN
- Function of the key depends on the selection in the Assign button setup range from:
- REC
- L / R channel routes to L / R BUS

- write: record of Automation
- read: Read the Automation
- Setting applies globally to all tracks !!!!!!!!!!!!!
29)
SOLO
- switched channel strip Solo

- supports PFL, AFL and SIP (really) (- Surround in use on
buses AFL only (SAE))
- Several Solo simultaneously Shift
(- Enable Under WINDOWS SETUP MIX OPTIONS Solo Latch,
Page 190 of 466
otherwise only ONE SOLO possible)
- Solo Isolates: for channels to be always listen

- CLEAR: turns off all solos
- Solo Level via potentiometer (46)
30) MUTE
- is after fader, thus mutes when the POST SENDS from also !!!!
31)
FADER
- regulates the channel volume (basic function)
- FADER-SWAP: V-POT function is additionally placed on the fader
CONTROL ROOM SECTION
- Chooses the signal to that go to the monitor system

- SETUP open setup window for the Control Room
amongst other things:
- Selection of the signal source

- NEAR MAIN and have no function
- Listening volume is regulated by potentiometer
- DIM adjusts the volume to a preset value in the SETUP Down
PHONES SECTION
3 signal sources can be selected directly
1) Cue way 1 (AUX 9/10

2) Cue way 2 (AUX11 / 12) CR)
control room mix
- stored via SETUP different templates, and other sources adjusted
TALKBACK SECTION
- internal microphone
- Press Talk to speak
- SET UP:
- Dial internal or external micro
- volume
- Assigning the talkback signal to the CUE-ways
MACROS (47)
- Allocation of 16 freely selectable functions

- WINDOWS SETUP MACROS assignment 1-8 and 1-8 SHIFT
TRANSPORT SECTION
Page 191 of 466
- Keypad for numerical input

- Enter to confirm (49)
- Transport keys and push Wheel: (59-64)
- for drive control of a sequencer via MMC
Page 192 of 466
Signal Flow 3 - Studio 3 - SSL ASW 900+

- actually consists of 3 devices:
- 1x quality DAW Frostend (Split-concept)
- 1x DAW controller (HUI protocol support)
- 1x 5.1 Monitoring Station
- has 8 Track Coach:
- 2x L / R busses (Mix / Recording)
- 2x cue buses, intended for headphone mixes, ie stereo
- 4x aux (FX) buses
- PFL Buss (Mono)
- AFL Buss (Stereo) Post Pan
CHANNEL STRIPS (24, fully analog) per
channel 3 Different inputs
Mic 1kOhm Line 10
kilohms instrument 1
poppy
The higher the impedance, the more height you get usually Per channel:
. 6 48V Phantom Power
. 7 rotator
8th. 20 db PAD
. 9 Flip switch for switching between Mic and Line

Instrument button switches to the instrument input line regulator is used as a gain
The line to be normalized to the patch bay with Protools Outputs

- the frequency response of all inputs is linear to about 100kHz
- Super Analog SSL construction, ie no capacitors in the channel
- Pad switch, flip switch (Line / Mic (down)), phantom power, EQ
- HPF is one of the EQs, so EQ must be pressed in

It consists of 2 parts:
. 4 High-pass filter (EQ LED is red when the high-pass filter is turned on, otherwise
green)
. 5 EQ
1. 4 Band EQ
1. 2 Fully Parametric centers
. 2 2 outer bands are semi-parametric

1. Switchable between Bell or Shelving EQ, standard Shelf
. 2 G-EQ: Turn EQ between G or E Series EQ characteristics
around
1. difference is, how it changes the Q factor in Cut and Boost

. 2 Standard E Series is selected
Page 193 of 466
. 3 Switches To EQ In
. 4 Switch for ISR

1. can also be PRE EQ switch Aux section
. 2 Cue aux paths for the headphones mixes
1. 2 stereo buses
. 2 Per channel only one controller, each channel only A or B possible
. 3 Automatic Pre-connected, and post can be switched

. 4 EFX button: The cue signal is routed to the 8 busses and allows
so additional effect loops
. 3 FX aux sends for effects
1. 4 Mono Fx buses
. 2 Post are connected and can not be switched Pre

. 3 FX 1 can be switched on FX 3, FX2 on FX 4
MONITORING SECTION
1) Old 5.1
2) A MINI
are for 4 different outputs for selecting
3) MINI B
if none of these three outputs has been selected runs Main 5.1, so only three switches. It can also only
be one selected! MISC LEVEL SECTION
-
DIM Level
AFL-Level
PFL Level
Seif-Level (Solo In front raises the selected solo in the foreground)
PHONES Level (Engineers Phones output (front))
MINI-Level (settings of Mini-interception) PHONES
SECTION
- for the Engineer Headphones
stereo buses
There is a record and a mix stereo bus

. 4 both buses have a stereo insert
. 5 Press the total button causes the ISR signal 50/50 with the Dry Signal
is mixed (parallel compression)
. 6 Dyn: Turns Dynamics to a stereo bus. !! Dynamics have linked
become!!
. 7 Comp: Grind a sum the compressor

8th. + 10: signal is amplified by 10dB
. 9 Fader: Selects what the master fader controls either mix or Rec
Page 194 of 466
microphonic 2
Directional differences in the horizontal plane:

Differences in lateral
sound incidence
the natural sound
field
Intensity differences
Differences for locating in

30 direction in the
stereo playback
7dB in language
15-20dB
7-10dB in Music
Timbre differences
<300Hz
0dB
2.5kHz 10dB
6kHz 10kHz
20dB 30dB
Transit time differences
0,63ms
1-1,5ms
Basics for stereophonic:

Main microphone:
Stereo Microphone with 2 capsules or an assembly of 2 single microphones spot microphone:
Additional microphone to volume and Hall differences compensate Spatial

stereophonic:
Optimum stereo imaging at the loudspeaker reproduction. The stereo image is connected to the
reproduction room head-related stereophonic:
Optimum stereo imaging when listening with headphones. The stereo image to the number with the head movement.
The importance of the stereo microphone method by:
- Localization: depending on the recorded intensity differences

- Space: depends on the recorded runtime differences
- Signal depth: depending on the recorded runtime differences mono compatibility:
Depending on the recorded delay differences (in mono mix comb filter effects can arise)
The selection of Mikrofonierungsverfahrens depends on:

- sonic objective
- receiving space
- Dispersion of the sound source (affects the position of Hauptmik's)
- Noise etc. by public
Page 195 of 466
Spatial stereophonic:
1. Intensittsstereophonie: only intensity differences (XY stereophony, Blum
Arrangement, MS stereo)
. 2 Laufzeitstereophonie: only transit time differences (AB stereophony, differences in
Young and old AB)

. 3 quivalenzstereophonie (mixed stereophonic): Intensity and
Runtime differences (ORTF arrangement NOS arrangement stereophonic ZOOM) Both
geospatial and head-related stereophonic:
. 4 Trennkrperstereophonie: intensity, tone and time differences
(OSS method, spherical surfaces microphone, SASS)
head-related stereophonic:
. 5 Intensity, tone and time differences, outer ear distortion

(Originalkopf- or dummy head microphone)
XY stereophonic:
- only intensity differences
- one takes condenser microphones as dynamic microphones would too noisy
- you take 2 individual microphones that are directed, and arranges them on the cross. The two capsules
must be on the same Level. You can also take a coincidence microphone, which consists of a microphone
on the body 2 capsules are mounted above each other directly. Offset angle:
Is the angle by which one of the two microphones at the stereophonic major axis is deflected
angle:
If the angle of both microphones to each other (2 x offset angle) shooting
angle:
If the angular range in which there is a stereophonic distribution of signal sources The recording
angle depends:
the (stereo) The smaller the monaural recording angle of the microphones used, the smaller shooting angle
of the main microphone.
the signal sources the greater the angle, the smaller the recording angle, the drawn further
apart ready. Kidney (big -) Supercardioid Hypercardioid Eight (small pickup angle) If that is too
small opening angles signal sources at the center are displayed too loud. When too large
selected opening angle too quiet.
If that is too small opening angles signal sources are displayed in the middle of dry (ie barely Hall), than
at the edge. When too large selected aperture angles they are dry at the edge, ready than in the middle.
Page 196 of 466
The opening angle should not be greater than the mono recording angle of the microphones
used: Kidney:
80-131
Supercardioid:
70-115
Hypercardioid: 60-105
Night:
45-90
180 viewing angle is from 90 pickup angle 90

acceptance angle is 180 opening angle With 45 offset
angle result: 2 Kidney 180 recording angle 2
hypercardioid 120 recording angle 2 Make 70 receiving
angle
Summary XY:
- good localization (because of differences in intensity)
- small space (for run-time differences)
- bad signal depth (due to delay differences)
- monokompatibel (because no runtime differences for comb filters provide)
Blumlein arrangement:
stereo Sonic
2 Make with 90 viewing angle

70 receiving angle
for XY to the Blumlein is comparatively spatially but has due to the directivity of eight a weakness in
bass
MS stereophonic:
M = mid or mono
Directivity is arbitrary (except thigh) S = side or
stereo
Directivity Eight (left at 90 to the middle microphone) The matrixing in L / R mix via an MS matrix
(usually an external device), but is actually used after the fact and if ever rarely.
Page 197 of 466
MS matrixing on an analog console:

The M-signal is fed into a mono channel (PAN center). The S signal is fed into two channels (PAN once
left and once phase rotated with PAN right (drawing)). The recording angle depends on the
Zumischpegel the side signal. The higher the Zumischpegel, the smaller the angle. Used for basic level
of equality of the two side signal channels:
-
Channel 1 MUTE
S signal in channel (PAN left), fader on U, Setting Levels
POST FADER Channel 2 out via Direct in channel 3
Channel 3 (PAN also left), turn phase fader on U, Setting Levels
Fader Set in channel 3 so that nothing more is heard

Channel 3 span to the right
release channel again

The Zumischpegel the side signals will now only be changed channel 2 !!!
Page 198 of 466
Declaration on sound arrival direction depending on the output stage:
sum
Sound incidence direction
sum
link
pretty
front
for every MS arrangement with a specific Zumischpegel an XY arrangement!
The mono recording angle of the center microphone should not be less than the angular range in
which there are the signal sources.
Summary MS:
- good localization (for level differences)
- little space (missing time differences)
- poor low graduation (missing time differences)
- 100% monokompatibel
Benefits MS over XY:

- Shooting angle is only determined in the post
- as a center microphone, a pressure transducer used (good bass instruments)
- on the stereophonic main axis is no coloration
Page 199 of 466
Laufzeitstereophonie:
-
AB stereophony
Report is prepared in parallel
Microphones arbitrary characteristic
The distance between the microphones is referred to as microphone base
the stereo base can be completely filled until 37cm microphone base
with a smaller microphone basis the transit time differences produced are too low
- The greater the microphone base in excess of 37cm, the smaller the recording angle
- the mono recording angle must not be less than the Stereo recording angle
D=
sin 37 cm
21
D = microphone base, = shooting angle Distinction

is made between:
- small AB:
- Arrangement with small microphones base where the camera angle is
adjusted to the size of the orchestra
- Large AB:
- Arrangement with a large microphone base, which is primarily used for recording of
ambient sound
Page 200 of 466
Recording angle 60
microphone base
74cm
80
100
120
140
160
180
58cm
48cm
43cm
39cm
38cm
37cm
Summary Laufzeitstereophonie:
- poor localization (missing level differences)
- much space (time differences)
- good signal depth (time differences)
- not monokompatibel ((runtime differences result in the mono mix with lateral sources to
comb filter effects; thus:
sound changes
Sources loud in the middle
Displacement of the comb filter curve in moving signal source
quivalenzstereophonie:
- Intensity and time differences are basically available
- 2 directional microphones with offset angle and microphone base needed ORTF
arrangement (Office de Radio Diffusion-Television Francaise):
2 kidney
55 offset angle
17cm microphone base
95 receiving angle
better localization NOS
arrangement:
2 kidney
45 offset angle
30cm microphone base
80 receiving angle
spaciousness
Stereophonic Zoom:
Analyzes of Michael Williams
Shooting angle (Williams curves)
Angle distortion (displacement of the imaging direction)
Volume distribution and Hall distribution (see XY)
Summary quivalenzstereophonie:
good localization (level differences)
much space (time differences)
good signal depth (time differences)
monokompatibel (because in the mono mix with a lateral signal source not only phase
differences, but also level differences occur. Thus, it is merely a ripple in the frequency
response, but am no deep dips on)
Page 201 of 466
Separating body stereophonic:
2 pressure receiver between which a separator is

Runtime differences (through the microphone base between the capsules)
Level differences (by shading

Timbre differences (by the frequency dependence of the shadowing) OSS method
(optimal stereo signal):

developed by Jrg Jecklin
2 diffusfeldentzerrte pressure receiver
16-20cm microphone base
in between a jecklin disk with 30cm diameter and absorbent surface
-
0-30 offset angle

Sound slightly treble in
Summary separator stereophonic:

much space (skew)
very good localization (frequency-dependent level differences)
very good low graduation (skew)
monokompatibel (because with lateral sound not only delay differences occur in the mono
mix, there is a ripple in the frequency response, the increasing frequency is decreasing)
Sphere microphone:
Schoeps KFM 6
Developer: Gnther Theile
Reverberant plastic ball

built 2 diffusfeldentzerrte pressure receiver flush
the capsules sitting sideways with 90 offset angle
90 receiving angle
Page 202 of 466
Crown SASS (stereo ambient sampling system):
2 diffusfeldentzerrte pressure receiver

17, 2 cm microphone base
35 offset angle
Separating body front absorbent, reflective back
SASS-P: with Crown electret capsule
SASS-B: with DPA capsules
90 receiving angle
Head Related stereophonic

Runtime differences (by microphone base of approximately 17.5cm
Intensity differences (by shading)
Timbre differences (by frequency-dependent shading)
Outer ear distortion (locating in the vertical plane) for
three-dimensional imaging: deleted 3D positioning: possible
headphone playback speaker playback
Summary head-related stereophonic:
very good localization
very distinct spatiality
very good signal depth
monokompatibel (because with lateral sound not only delay differences occur in the mono
mix, there is a ripple in the frequency response, the increasing frequency is decreasing)
Page 203 of 466
Original head microphone
Soundman OKM
Electret pressure transducer capsules (diffusfeldentzerrt)
If you put in the ears

versions:
- OKM I:
- Equivalent noise: 36BB (A)
- Frequency response: 30Hz - 16kHz
- Maximum sound pressure (SPL Max.): 114dB SPL
- OKM II K:
- Equivalent noise: 33bB (A)
- additional built PAD 20dB
- OKM II R:
- Equivalent noise: 33bB (A)
- Built PAD 20dB
- and additional PAD 20dB switchable
Page 204 of 466
Dummy head
___
Neumann KU 80 (1974)
1. Artificial Head
with ear canal replica (thereby boost at 3 kHz !!! When playing in the human ear again
increase, ie scrap)
freifeldentzerrte pressure receiver (Play by just over freifeldentzerrte headphones possible)
2. Artificial Head
without meatus
diffusfeldentzerrte pressure receiver
only without the transfer of heights loudspeaker reproduction was possible) ___
Much simplified head shape

Specifications were extremely improved
increased transmission factor of 10 to 20 mV / Pa
increases Max. SPL of 130 to 135
Additionally 10dB PAD
Application:
Metrology, radio plays, musical productions, nature sounds
Page 205 of 466
Additional microphones:
Improving the volume balance

Improving Hall Balance
Any improvement in the localization of the supported instrument (at AB Hauptmikrofonierung)
Mono supports for single instruments
Stereo Support for instrument groups
Problems with comb filter effects:

Solution 1: delaying spot microphone
The distance from the main microphone is 4m to the instrument, the distance
from the spot microphone is 1m to 3m instrument path difference
Consequently, since the sound is flying at about 340m / s:
3 3 1 340
msmttmsm
=
= , 82 ms
8340
So you would have to set a delay of about 8-9ms to prevent comb filter! Solution 2:
The main microphone must be delayed extent that it is 10-30ms behind support microphones (the distance
from the spot microphone to the main microphone must be taken into account)
Problems with small regional units:
Does the listener takes the signal from the microphone support rather true and sees the main microphone
as 1. Reflection! Solution: See Comb Filter
Page 206 of 466
sound synthesis
subtractive synthesis
Principle: a overtones are deprived example spectral components: Basic modulatable, according to the
principle of subtraction synthesis of guidance
Synthesizer.
VCO
Voltage Controlled Oscillator = tone

Control voltage controlled frequency
Standard waveforms
-Sine
- triangle
- rectangle
- sawtooth
-murmur
VCO can run synchronously with other signals
Fourier analysis:
-JB Fourier lead early 19th century to prove that vibrations in an infinite number of sine waves can
be broken down (so-called partial oscillation) with different frequencies, amplitudes and phase
angles. Result of the spectrum of the vibration
A specific calculation method that allowed due to certain constraints, particularly fast to calculate the
spectrum, is the Fast Fourier Transformation (FFT)
Sine :
Is not assembled, but has only one frequency, so no overtones. Thus unsuitable for subtractive
synthesis. Sometimes still present in sub oscillators. Laying sinus under a signal.
triangle
The triangular wave is composed of all the odd harmonics in the amplitude ratio 1: n
The 3rd harmonic 1/9 of the original amplitude The 5th

harmonic 1/25 of the original amplitude
Page 207 of 466
The triangular wave is everyone silent

Rectangle (Rectangle):
The square wave is composed of all the odd harmonics in the amplitude ratio of 1: n. The ratio of the
length of the upper voltage value to the period is called the duty cycle (pulse width). Rule of thumb for
overtone. Amplitudes fall off slowly
3: Harmonious 1/3 of the original amplitude
Rectangle is brighter.
Odd harmonic number Formula for overtone:
The narrower and square structures are, the more harmonic content we tend.
Sawtooth (Sawtrooth):
-The sawtooth wave is composed of all integer harmonics ingen in amplitude ratio 1: n.
Lightest sounds the Sgezahnschwningung.
Noise (Noise):
White Noise: Same energy in the frequency range absolutely equal width (ie same width
in 100 Hz bands).
Pink Noise: Same energy in frequency bands relatively equal width (ie same
Musical intervals)
Because of the logarithmic frequency perception white noise sounds much treble.
VCO module:
The same signal is sent out in this module 4 times with different processing. CV1 Pitch input CV2 pitch
input
Controls: How much is due to signal and affects the
Tnhhe
Out.
PWCV1 Control voltage input 1

Page 208 of 466
PWCV2 Control Input 2 control voltage regulator Audio may be used as a control voltage. If one
goes in with an audio signal, the pitch is greatly beeinflusst- because of the high input voltage in the
frequency image Frequency response is modulated generated noise
VCF
Voltage Controlled Filter Voltage Controlled Filter Multimode Filter that not only Low and High Cut
special can also be other filter) type
Resonance: increase in amplitude in the range of the cutoff frequency
resonance strong: very high peak resonance weak no high top but broad
With very strong resonance (low attenuation) self-oscillation. the filter produces a sine wave at the
resonant frequency. Height can be regulated by Cut Off. Slope (slope) Ex Minimoog, TB 303 18 -.
24 dB per octave optimal sine wave In a lowpass filter
either goes away, quieter or remains.
Keyboard Follow or keyboard tracking to the sound character over the entire keyboard
Way in injury. Depending on clay as the filter operates at cutoff frequency travels with.
FM synthesis
A synthetic method of electronic music FM frequency

modulation
Phase modulation is performed which is expressed sonically. Yamaha DX7
first FM synthesis device
Two oscillators are connected. One can be

controlled by others.
Sets space expected to save the newly developed sounds and presets. Carrier and modulator to
each other always in the fixed time relationship.
ie
carrier
modulator
500Hz
250 Hz.
If the modulator is an integer multiple of the carrier frequency and the tone, arising from carrier frequency
(T) and frequency modulator (M) to T-1M, 2M-T, T-3M.
Page 209 of 466
ADSR envelope
Attack Decay Sustain Release Level
VCO (controls) VCF (cut-off frequency) VCA (Level) VCF (controls) VCO
LFO VCF EC
VCA
Keyboard
(Gate)
EC
Two VCOs modulating mutually VC1 modulated to VC2 and VC2 VC1 modulated metallic
sounds flutter, frequency modulation, vibrato effect of LFO and VCO is vibrato or
frequency modulation. LFO VCO
Vibrato is a special case of the frequency modulation.
Cross modulation: two oscillators modulate each other's frequency frequency

modulation
creating new sounds
Frequency modulation as synthesis form
With digital synthesizers can operate only frequency modulation. LFO VFA (Cut of
Frequency): Auto Wah LFO VCA (Level): Tremolo
Tremolo is a special case of amplitude modulation (in volume)
LFO
Low Frequency Oscillator, not usually used to output audio signals but control voltages. Frequency
range Type: 0-30 Hz Unipolar / Bipolar waveforms.:
- Sine
- triangle
- rectangle
Page 210 of 466
- Ramp Up, Ramp Down (sawtooth)

- Sample & Hold o. Random
Unipolar / Bipolar
Bipolar: Normal Osc. Works bipolar, positive, negative values of the voltage Unipolar: Only zero volts
and positive voltages are accepted. vibrato better bipolar because frequency levels of negatively and
positively to the original frequency
swing.
Audiooscillatoren are bipolar
Distinction between Ramp up, Ramp down Ramp up:

Sounds like Rckwrtsabspielend Ramp down: perkusiv
then drop.
Sample & Hold
Is the opposite of the noise (noise)

Each area is maintained when I LFO faster rotating held it longer, If I turn LFO slower, he is
kept short. (Noise is chopped) LFO module
Additional effect of the LFO's

Ring Modulation: carrier and modulator have no direct component.
Sync
Synchronization of two VCO's.
Whenever the master begins a new period of oscillation, also starts the slave.
If you change the frequency of the slave, thus changes the timbre.
Page 211 of 466
Graphic illustrating the operation of a synthesizer:
External Audio In
VCO
VCA
VCF
Audio
Out
pitch
LFO
Freq cutoff
level
EC
gate
Key
boar d
Page 212 of 466
additive synthesis
Principle: a complete oscillation is from the overtones sounds
mixed together. VCA + EC:
(envelope)
Then you can adjust the volume in each sinus.
eg
32 VCA + 32 EQ
Analog almost impossible to realize (size, price, usability) Advantage:

Extremely flexible sound way disadvantage: Cumbersome
ShowSounds
Rectangle:
Odd multiples of the fundamental. Rectangle not additive but square-wave oscillator better, How do you
create a square wave with an additive synthesizer? Additive synthesis can be good use for imitating any
sound because of its high flexibility. Sinusoidal vibration is generated by gears and a pickup with a
Hammond organ. Due to the mechanics of the pickup picks up the sound and by the wheels it is created.
Hammond organ:
- Draw bars
- Tone wheels with Pickup
resynthesis
Extension of additive synthesis originally developed as an alternative to sampling.
Principle:
Spectral analysis of the sound ie detection of frequencies, amplitudes and phase angles.
Characteristic Spectral Components

- possibly editing
-Additive synthesis using the obtained parameter sound not just once, but
studied more times.
Resynthese- time slicing
Page 213 of 466
Expansion: Time slicing

The spectrum is determined not only once but several fixed points in time. Thus, any sounds that
are reproduced realized change over time.
The closer the time points are beeinander and the more oscillators are available, the more accurate a
sound reproduced.
eg MP3 player uses time slicing
Physical Modeling
Individual components of real or unreal instruments are simulated using mathematical models
and can be linked. The mathematical models to enable realistic sound at the lowest possible
computationally. Nevertheless usually high computational effort. All analog digital synthesizers
are physical modeling. Spits based algorithms directly to sound out without oscillators.
Depending on the models are real-time changes in normally unchangeable parameters
possible.
Versatile and wide technology.

Advantage: sound spectrum from highly realistic to experimental. Disadvantage: difficult
to play, depending on the instrument
eg saxophone on keys.
granular
An existing sound (mostly Sample, selterner Real Time) is in short fragments ( "Grains")
divided. Time Stretching and Time Compression possible.

timestretching divided sound
Gaps in the separations,
Separations with loops of the respective sounds in sections fill.
-Grains can be played in a different order and with different playback parameters (pitch filter).
Settings of playback parameters often by specifying ranges of values,
eg Grain 0-300 to have playback parameters of 300-700 Hz. Program:
GranuLab
Funk Soul Brother Rockafella Skung
A.Ruschkowski
ISBN: 3150096634
Micro sounds
Curtis Roads
Page 214 of 466
Research
Diploma level
Threshold at 80%
BMC certificate diploma degree certificate Degree Bachelor of

Arts (Honors) Bachelor of Arts English High School Honors if you
wrote a special work. Overall, you need 360 Points (720 Ecpts
Points) BMC certificate is important for the Bachelor.
Master Degree by:

Sets Degree advance (number of points) BA is
judged in Council:
1.First rate
2.Upper Second
3.Lower Second
4.Third rate Diploma with 70%
Diplomzeugnis:
Witness in average of 80% 11%

11% theory test practice tests 6%
final production 6% Thesis
33% Practical final exam 33%

Theoretical final test preliminary mark
must be 70%. Total Score 80%
testimony
Diploma I get when the specialized working more than 70% (4.9) skilled work is
important to get an SAE Diploma, made 1 calendar day late 5% deduction for the
technical work note.
Page 215 of 466
Delivery Expose:
Topic of skilled work construction
of skilled work to which specialist
teachers 3 to 4 pages maximum
Should overview of the technical work offer.
Topics Search;
Default: Relevance to sound equipment

Delivery specialist working in an appropriate width and depth. Expose must
meet specifications:
-computer Posted
-Name of student
- Name of specialist Lehres
-Title of Work
-Topic work
-Thesis Outline: What is the work up
-Text Size in percent per point
- motivation
- Name and signature of student and specialist teachers
-date
-Kompilatorisch
Kompilatorisch Composition from various fields
eg acoustics u. microphonic
Unilateral description is not made.
Description of the working sources:
Literature, books
Bibliography
Availability catalogs, Amazon Buy Borrow
from library journals (can be used) Studio
Magazine, Sound and Recording Internet
(no seriousness)
Structure of the Thesis:
- cover sheet
- Table of Contents
-Bulk
-Resort Directory
- investment
Page 216 of 466
-References / Bibliography
-no SAE or Middle Sex Logo

- Name of the student, student number
- Name of the Institute (SAE Cologne)
-Title of Work
- specialist teachers
- Abgebetermin
- Word Count (net)
Cover sheet:
Table of Contents:
Chapter number and heading

subchapter Pages
List of Figures :
Bulk :
Page + Name + Abbildungsnr

Systems CD-Audio examples Recalling ne
installation CD Introduction Main part
Resumee
Resort Directory :
No.
name 1
,,,,,
,,,,,
,,,,,,
in addition to complete work CD with PDF content

2 version for the archive.
Presentation of pictures:
When pictures were used the table of figures is behind the table of contents.
In the figure, the figure number and the source of the picture come. But the source is not in the
source directory,
Formal outline of the main part:

With Introduction Chapter 1 begins.
Decimal: Chapter are classified decimal

1.
1.1
1.1.1
1.1.1.a)
Each chapter and sub-chapter appears in the table of contents.
Page 217 of 466
If you go to an in-depth chapter hierarchy there must be at least 2 sub-items.
Not like that
3. Cheap Audio Equipment
3.1. any
Word Count: 5000 words net +/- 10%:
Topsheet, directories, forewords, plants, headings, footnotes, captions Etc. Do not count it.
From the introduction of the main part to the end of counting.
Typeface etc.
- Helvetica Arial, Time (Standard)

neither italics nor bold color black
12 points
Line spacing: 1.5 fold

Justify and hyphenation enabled.
Number specifying the bottom right cover sheet but not stet it. From page 2
Number of specify up to the Annexes.
-sided printing
- 80g / m2 paper Matt / High Gloss
smtliches which is not one of us, is in thoughts, quotes Direct

quotes
Quotes
taken as complete sentences or sentence structure that does not change the meaning of.
Enhanced Short document:
Blablalbla .. "zitier, zitier" blablablupp Cited section must remain true to

the original. Words, use the following characters take out (...) (...) (The
dog)
(Indicating the author) (!) Exclamation mark Accompaniment of a

quote. "(...) The school (!) (!) Is great (...)"
Page 218 of 466
Primary sources used, otherwise if possible secondary sources. On the same page appeared in the
footnote citation reference to be numbered. Digital audio with (high 12) under the footnote
(High 12) Schulze J.: Digital Technology, 1999 Page 12 (Surname, name, book
title, year of publication, page number) when there is a comparison, then the
footnote:
(High 13) See. See (surname, name, book title, Erscheiningsjahr, page number)
Guide ordered by last name. Complete. Name, completely. Title,

publisher and date of publication.
Internet:
Author Title
specify specify
specify Link
Navigation URL last visited specify date AVL Date, Available or oV without author
Page 219 of 466
session Procedure
Final production:
SSL
Neve
Studios for receiving the final production
Icon
Production process:
need to make final productions SF Studio 2 be at least 70 percent. be issued final production at
least 7 days after the Mixt Ermin.
Tape:
-Optimal receiving tape 4 man at least,

-Genre has to adapt to the sound equipment
pre-production:
of final production is taken into account and assessed. Timing: 10-15

15-18 Recording Mixing Aural Presentation Klangbilder, spatiality
Left / right distribution depth create

drawings.
Enter Klangliche idea of t he whole mix before.
Knowledge of the piece text indication
specify Arrangement in clocks,

Delivery of demo recording
for preproduction demo recording must be set to.

Pre-production is graded with.
quality unimportant
Deadline for completion of all productions is November 17th
remastering:
1. Single Version cut 3:20 min.
2. Original Version
recognizable on the CD label.
Normalization to -0.2 dBFS

Page 220 of 466
fade Out
good fade out is 20-25 seconds. Combustion mode of Audio
CD disc at one Red Book standard
Audio CD Standard
Audio CD pcm
If burning process is complete, again to hear before delivery. 2-second pause
in between must be between the two tracks.
Documentation:
Indication if something is different for preproduction features
and effects.
Final production can only be exchanged by a new production When replacing supervisor
must be informed.
The advance: At
session start;
Set director / studio: Communication

Talkback microphone. On Headphone Amp (Respectively on studio and Loudspeaker) Listenback
microphone (Return Talkback Mic in the opposite direction)
Analog audio recording:

Write down what instrument come in the footsteps of earlier in which lane.
- TC offset frame rate on the last track.
- Adjacent lane next to the time code must remain free because the timecode crosstalk is transferred to the
adjacent lane, because of the sound heads which are next to each other and magnetized. The adjacent lanes
of TC track must remain free (due to crosstalk) therefore recording the timecode tracks on edge tracks
convention, the highest edge track.
The deepest signals are at the beginning Kick the 1.Spur for driving the pulleys, there is a high
frequency attenuation on the edge tracks.
Tracksheet: specifying the instruments to be written on which tracks tape.

16.4 min.
7.6 cm / sec. you should use at no noise reduction.
32.8 min
3.8 cm / sec. with noise suppression system
compander Collapsed during recording, expands when reproducing. Noise Reduction

System:
- Dolby
Page 221 of 466
-DBX
- Telcom / Highco
- Vari
Length of the position refers to the tape speed. Depending on:
level sounds for calibration and matching a tape machine adopts 3-level sounds
at 100 Hz
kHz
- 10 dB 1
0 dB
10 kHz
10 dB
adjust playback equalization: Setting linear

frequency response.
Level sounds you need to adjust the playback equalization. OfftapeMonitoring listening to the recordings headphone mix should also Offtape
interception. use pre-fader signal to the headphones mix. Closed
headphones used.
Crosstalk avoid the headphone mix

Feedback does not occur when
Panel label Test

Counter: Counter of analog tape so
Start and stop point knows.
Headphone Amplifier: headphone mix

Behringer Powerplay HA4700
- Each channel has an Aux Input

- Each channel respectively 2 headphones
- Main Out one can measure the back, jack symmetrically
MTC Output Mode:

Analog tape machine: Input
Input signal is added to the output, monitor system Input signals Sync
rec ready Monitor system input signal non rec. ready Monitor speakers band
signal reproducing head Repro Interception which was taken
optimized for playback

Page 222 of 466
Treble boost, suitable for tape monitoring u. Mixdown
Auto Input (when you select Sync)
Input signal interception in recording During playback

monitor system of band signal for punch in / out
Before punch in and punch out after you hear the recorded signal. During recording, you can
hear the input.
input: entrance
No Input:
corresponds to sync with analog tape machine ie rec. ready.

input monitor system
no rec. ready Recorded Signal listen.

Before and after the recording Recorded Signal During recording
Auto Input:
input
Slate:
Talkback microphone is routed to the oscillator on all MTK buses. Separation of the
strip material by oscillator signal.
Stimmton:
440 Hz recorded on analog tape. To fine-tune the tape speed. Formant displacement due to
incorrect setting.
Bounce in analog tape recording:
1
8th
10
12
11
13
BD SN HH LT MT HAS OHL OHR Bell
Bounce L
Bounce R
Actually should be released on the belt and then the left and right bounce on 2 tracks for bouncing a
track. 1 track space between source and target track. Or next lane panned. to avoid the acoustic
crosstalk from source track to the target track of the tape machine. always target tracks not listen when
bouncing the source tracks.
Recording:
-7dBFS to -4 dBFS recording
Page 223 of 466
-0.2
0.2 dBFS CD normalize gobos Walls for rolling absorbing or reflective overdubs A
method of recording consecutive shots. Guide track (pilot track):
Orientation for musicians overdub process.
Guitars
DI Box (balanced out) Wallbox DI Box (passive) AMP
We want as long as possible the symmetrical connection keep then in an output with a passive DI
BOX.
Connecting a signal from the control room to the receiving space.
For example, loss of electric guitar and amp.
miking:
installation close Sound change, only detail of the instrument sound With multiple instruments in
a receiving space (close miking) observed changes in sound and use directional microphones.
- additionally with gobos (acoustic movable walls) work.
- inserting filters Hide frequency components.
- at 200Hz the fundamental frequency range of an instrument begins.
- Use of Noise Gates.
Mikrofonstativ positioned laterally of instruments. Always set up links. Thus, the screw of the
tripod does not dissolve, but is in the closing direction by the weight.
Page 224 of 466
production 1
- know song structure, key, etc.
- look at tape beforehand to know how they tick.
-CD obtain a reference title
- Instrumentation = Musicians check out what they can, which is not
- Real determine tempo
decides on success and Misserfolg- Key is crucial
a) singers can sing the
b) key suitable for the guitarist.
Arrangement:
- Intro
- Verse / verses / O part
- Chorus / chorus / B section
- Bridge / C-section
- solo
- Prechorus / Refrain B
- Acapella part (only vocal)
- instrumental part
- Outro / Ending
-Coda
Coda Repeat up to a part
-Reterdando
Reterdando Slower becoming middle eight
- <8 bars in the middle Reintro / hook
Set Arrangement in Logic or ProTools already metronome course
mark in the Metronome track orchestra must play another delayed
vivid, realistic
Rock must all play on an impulse. How do you get
keyboard back in the mix: a mixture depth
Slow attacks, long release
Width of a mixture
short attack, short release
Instruments can be widely positioned backwards:
- Pre Delay short

(Reverb)
-Lower highs and lows (EQ)
-turn down level
Instrument far forward position:
- Long Pre Delay
-Compression Settings: short attack, long release
-Raise 3- 5 KHZ
Page 225 of 466
then compress 1.Gaten

Pressure range of the bass drum from 40 to 80 Hz. Except for Bass Drum
and Bass everywhere a low cut inside. Instruments height range:
Hi-hat, overheads
Instruments center area:
Guitar and Vocals
Instruments bass range:
Bass Drum and Bass
sound engineer
-Technical responsibility
- know Microphones
-know equipment
-avoid hum and noise
Producer:
-Musical responsibility
- Manage budget
- Judgments: Hittauglich, Radio is not enough
Marketing is important for the timing of a production charge
between record company and band
A & R and Artist Repartuere
Page 226 of 466
Synchronistaion
Griech :.
Syn = equal
Chronus = time
Synchronistaion = -> Film / sound synchronization technical
Simultaneous flow of equipment
synchronization
synchronization
Localize: ADR (Automatic Dialogue Replacement)
record piece, into the now Part Rework for Ton

Synchronizing a part of the
Piece.
Best Synchronisataitonsarbeit for voice / sound is in Germany German films
are still dubbed again. Lip technique when synchronizing note speech labials B,
P, U, O look note lip sync
At that time band synchronization today ProTools and Neuendo 3 IT

International sound
technical synchronization
equating of 2 or more devices based on a single time base. As the silent film era ended, came
sound and image Synchronization The Jazz Singer (1927)
Allen Crossland
1.Tonfilm time on Nadelton 1922

Voigt Marsol
TriErgon
optical sound
With the help of photocells the sound it shine, the Zelloleut. By differences in light tone could
be generated by 60 Hz-8 kHz optical sound possible.
Page 227 of 466
2 font:
- Sprouts / intensity signature "variable density"
- Pip / Transversalschrift "Variable Area" To date, the
optical sound on the tape on it. "Fallback"
If digital sound fails, the band jumps on analog optical sound

around.
Film sound is kopiert- on optical soundtrack
2 channel, analog optical soundtrack was placed on the copy of the tape. In place of an
optical soundtrack came 2 signals. Stereo: Multi-channel, not only 2 Polyphonic Dolby
Stereo: 4: 2: 4 L, C, R, S matrixing (4 traces) on L, R (2 tracks). C to L / R mixed thereto. S
signal
-3dB is inverted to L / R mix in either +90 degrees or -90 degrees

In Play: L + R
center LR Sites
2 tracks are again applied when played on 4 tracks:

Digital optical sound:
Dolby Digital 5.1 (Surround) AC-3 encoding SDDS 7.1 Sony

Dynamic Digital Sound DTS 6.1
Timecode is on top of image.
Dolby Digital
draws the TC from between Perforationslschern.
DTS. 6.1 timecode is not in the film but in the time code signal
Problem:
Sound and movie on a medium
The film is read back. The tone is read linearly. The sound must be read before
further 20-21 images.
Page 228 of 466
Magnetic tape recording:

24 fps
movie theater
PAL, SECAM 25 fps
29.97 fps or 30 fps (60 Hz frame rate)
NTSC
Film sound is always recorded separately.
By flap picture and sound have their own starting point. By synchronizing Facebook practical
cutting but not enforced, timestretching
PAL Speed u
p
when synchronization of 24 fps to 25 fps.
3: 2 pulldown different change frequencies

Is a process for the conversion of a film signal in a NTSC television
signal.
How to cut images, although timebase. known start and
stop point.
Up to the 80 individual sites have been copied to another medium.
Ways of addressing 1968:
CTL
(Center Tape Length) Each image
is given an impulse. Addressing without time

information.
SMPTE (Society Motion Picture Television Engineers)
1972 first
introduced Timecode
EBU (European Union braodcast)

Various timecode by specialized (24/25 / 30d / 30round) Timecode
HH: MM: SS: FF 23: 59: 59:
23/24/29
Material is ready cut (10: 00: 00: 00) Now comes

the sound
30d / 30round:
if one expects 29.97 instead of 30 is the error. 108 frames / hour

out
3.55 frames too much
Per Minute 2 frames out -120 frames but
every 10 minutes not.
+ 12 frames (108 frames / hour)
LTC = Longitudinal Time Code

rectangular signal
Loudest thing there is.
Page 229 of 466
Edge track comes LTC out intervening prevent a track Place

crosstalk timecode. VITC
Vertical interval timecode Timecode Magnetic tape recording of each image can be
addressed.
Time code can be recorded simultaneously with the video. It can also be read out at
standstill and slower tape speed. LTC is for higher line speeds.
-no audio signal

inaudible (pure video information)
VITC fast rewinding it does not matter with LTC
I am everywhere to address my pictures in a position. It works with both types of
timecode. Videotapes must be pre-coded Timecode already let it read. As is recorded on a
Betacam?
Helical scan recording: tape head rotates faster. This allows high frequencies
are recorded with much better quality. VITC + A3 / A4 +
broadcast sound A1 / A2 are longitudinal tracks (LTC)
A total of 4 tracks are recorded. LTC 80 bits per frame

VITC 90 bits per frame
When SMPTE timecode that every picture an address

Black Burst:
BIMI Seamless Switch
when mixing together all the video signals you need a sync pulse. sync pulse multiple cameras
work with the same clock.
Black Burst Generator
Black image which is transmitted in the clock. Daisy Chain: BB

VTR 1 VTR 2 VTR3
- 75 Ohm impedance
Problem: Einstreuungswege for long lines

Terminator 75 Ohm terminator, otherwise there's cancellation and synchronous error
Star connection:
Page 230 of 466
VTR 1
BB's center
VTR 2
VTR3
No interference because no long line. Not often used because it is

expensive but better. Optocoupler severs the ground connection between
the 2 devices.
Problem: sanding off the edges of the signal. when many devices are connected in series.
Word Clock
Timing of signals on a clock. Sampling frequency and sampling rate Word Clock Multiple units can be
synchronized by the word clock to a specific timing. It is the black burst in the audio field. house clock
In large studio complexes all devices work on a house clock. The receiving
device must be in the synchronization of the master.
Super Clock
Nothing more than a word clock of digidesign. Ext with appliances and internally up to 256
KHz 11.2 MHz clock rate. Super Clock works directly with high sampling frequencies
more accurately. Problem: high frequencies that result in the cable to reflections. Inside
the device there works with 11.2 MHz.
PPL (Phased Locked Loop)
RME produced. Signal having arrives conversion by the A / D its own clock. PPL generates it on the
basis of a control signal the audio signal and synchronized.
Digital transmission formats:
AES / EBU 2 channels
SP / DIF [Self-clocking
ADAT
MADI (includes 28 AES / EBU channels + sync signal)
T / DIF
[
[
requires its own word clock timing
MIDIClock
Midi CLock has 24 pulses per quarter note 24 * /
Sheet pulse
Serves for synchronizing sequencers to tape or 2 sequencers.

Page 231 of 466
- MSP (Midi Song Position Pointer)

Addressing the MIDIClock need to assign each clock an address 1024 cycles at a 1/16
quantization possible.
-MIDIClock [
[
[
-MSP
Synchronize addressing even if stop is pressed
It would take only MIDIClock if you are always running track from the beginning.
- MTC (MIDI Time Code)
External order to integrate into the Midi system.
00: 00: 00: 00 Synchronization of the tape machine to Sequencer, sends time information tape machine
(LTC)
Synchronizer
(MTC)
Sequencer
Edison film perforation
Perforation perforation of film to unroll the pictures by the light. Used to control control
reference and monitoring purposes. To obtain an accurate reproduction of the bands, which
expands and contracts, a 50 Hz pilot tone is generated and recorded.
Page 232 of 466
Signal Flow 3 - Studio 2 - D-Control

Sync Logic via MIDI timecode.
- Peripherals Setup Sync I / O inserting a synchronization tool.
-ProTools Online. Sync I / O is a
timecode
running, standing
Up Sync I / O 2 times Audio Interfaces
Audio matrix of D-Control

X-Mon, interception unit. all analog signals go there by analog audio
portion of the D-Control
Extra unit in the rack is angeseuert D-Control. By D-Control not a

single analogue signal goes out of the talkback signal.
D-Control is connected via Ethernet to the computer.

signal purely Repeater (copies the contents)
power cable
Signal out (without error and loss)
X-Mon is part of the D-Control, it is associated with ProTools interface out of ProTools the
ExMon be aud patched, analog signal. Interface B9-B16 go to the D-Control monitor
system it is 7.1 capable at 8 ProTools outs. 1 Left
1 Output Left Box

5.Ausgang rights Box
5 rights
Dialogue always center otherwise only viewers in the middle in the cinema the dialog perceive
centrally. In I / O Setup in Pro Tools
Always lay a path 7.1 MAIN 7.1

ProTools makes ausganszuweisung automatically for D-Control in 7.1 stereo
path Subpath create:
Way through Subpaths create a stereo or 5.1 monitor system. No second use of
the outputs (B9-B16) AFL / PFL
Solo options PFL non distruktives
Solo
Assignment of solo monitor system by Solo
Path: A7-A8
Outputs A7-A8 fully normalized to D-Control Solo In.

New Track Default Output
all new tracks get the standarmigen

Page 233 of 466
assigned outputs. Meters

8 Main Meters
Define Control Meter Path.

are there appears to which meters exit or entrance Main 7.1 Set (left and right
feet 1 meter 5)
A1-A6 headphone mix outputs buses Delete
default
D-Control is connected to me each unit via Ethernet to the Mac. 2 Fader Units 1 Main Unit
D-Control is an Ethernet Controller keyboard is connected

via USB to the Mav. USB mouse Peripherals menu Ethernet
controllers
3 sections
Eth. tick neutral box
ProTools begins to look at to Ethernet devices. 1 Main Unit
2 Fader Units login transport function Play, Stop, Pause Ethernet more
data stream less clutter 32 channels Pro Faderunit 16
32 he steps fader jump with Nudge Nudge keys 1 1

Channel jump Nudge 8 8 Channel jumps Nudge 32 32
Channel jumps from ProTools on the D-Control workable
A1-A32 or A33-A64
Nudges slidably in channel blocks. Clear clip

deletes all Klippings in Signal + LED display
OWERTY Keyboard Switch Keyboard can be issued. Save switch once change occurs
from the last save lights Save.
Double and session is stored.
Unity menu Calibrating the meter and motorized faders Vegas Fashion lightshow
Page 234 of 466
6 keys with one display
softkeys
Track -Y With Tastendrcuk Track deleten, create, etc. Various ProTools

applications
Snap, Shuffle, shot, Grib, Group, Track, Playlist, Window, Audio Files, Show, Hide can
be operated here. Character Function
- Auto / Input
- Record
- Select Select track

- Solo
- Mute
ProTools
Select key has a second function, they will have a unique function Focus
function
One can choose a channel and switch with pressing the FOCUS key these in the Focus channel.
D-Control in the Main Section still a Focus Channel Strip. Any channels I can bring
in and work in the Focus Channel. Main Section
EQ; dynamic unit
are very high
Each track in the Focus Channel is can be processed with the EQ and dynamics unit from the Main
Section. These are Pulg ins ProTools, you can drive through the D- Control unit, instead to open a
plug-in window extrau on each channel. Saves a lot of time when assigning fixed inserts in the
D-Control. SF3 Sync I / O
Syncing with Logic
Setup Peripherals
synchronization Sync Device = Sync I / O
Then ProTools provide online with Apple J
Session Setup Window: Links below area generator = Sync I / O Select. Then turn
ProTools Online Session Setup Window = links below in the transport sector the generator:
Sync I / O Select.
Page 235 of 466
Digital technology
Depending digitizing the temperature Mreihe =
digitization
With a meter values are measured in sections. In regular intervals Ts,
the voltage is measured. Ts = 1 / fs
Sample rate (number of measurements per second) (time interval between two
successive measurements) The more specific, the smaller the distance, the more data is
produced. This happens at a certain speed the upper limit of a data word width n (bit depth) is
determined.
Since it is measured in the digitization only at certain times voltage, of the voltage waveform between
the times of measurement is not known. The digital signal is scanned in time
So time discrete.
Since when digitizing the voltage is measured only with a certain accuracy,
are rasterized and the possible values recorded. The digital signal is
therefore discrete values. Target: If possible exactly representation of
the analog SignalsLow error, as precisely as possible voltage values.
Advantages of digital technology
1: 1 copy is possible because not affect Strungen in the transmission to a certain degree on the
information content. original
(Digitized)
Values / time continuously
Copy
Values / time discreetly
If a digital signal is processed, it will cost the quality as well as analog. we can work digital non-linear ie
we can jump from back to front. we can work digital non-destructive Signals can be processed without
irrevocable damage UNDO.REDO priced
software
Computers can make the audio device.
Page 236 of 466
- has In and Outs

-Butterfly
- Speciehrmglichkeiten
-additional DSPs
Signal A / D and D / A
Analog In input amplifier Anti Alaising filter
quantizer
encoder Digital Out
Sample & Hold stage
------Recording
Recording and bertragunskanal -------------------------- Digital In decoder
dequantizer
Sample & Hold stage (Degli Tscher)
reconstruction filter Output amplifier Analog Out
antialiasing Highest audio frequency is limited.
If Shanonsche sampling is not considered arises aliasing.
Sample & Hold stage = Samples the analog audio signal. Sampling: (Regular) removal of
voltage values of the analog
Input signal.
Voltages are put off the quantizer. Clock is a switch
Overclocked from the signal, the quantizer assigns in time
Frame.
quantizer
Values from the analog input signal pull in the quantizer. In measuring
instantaneous values are to be fed,
In order for the quantizer has sufficient to measure time, the capacitor holds the value for a
certain time constant.
Compositions of vibrations pulse sequence:

amplitude modulation
pulse train :
A pulse sequence is composed of all integer harmonics ingen in amplitude ratio 1: 1: 1: 1: 1:
1: 1: 1 All harmonics are equal loudly.
Amplitude curve T = 1 / fo
Page 237 of 466
Amplitude Modulation:
Modulations always have a modulator and a carrier
modulator
carrier
AT THE
fT = sine wave
fT + fm = fm = difference tone frequency of

the modulator of the rear sample and hold
stage
The height of the amplitude is dependent on the input voltage of the rear sample and
hold stage is an amplitude modulation before. Carrier: pulse sequence modulator:
Audio signal
input voltage
pulse amplitude modulation
Difference and sum tape
Audio
fs fs fmax
fmax
fs + fmax
fs fmax lowest frequency fs + fmax

highest frequency fs = fundamental
frequency
If fmax is higher, audio tape and differential tape will eventually meet.
Audio difference
Aliazing
Total Volume
Overlap between Audio and differential tape

Page 238 of 466
therefore Shannon sampling theorem cal sampling

fs> 2 * fmax
fmax < fs
half the sampling frequency fs (Nyquist)

If the analog signal before it is sampled is filtered in low-pass, there is no Aliazing. Attenuation at
22.05 KHz Nyquist filter must be extremely steep. 16-bit, for example -96 dB stop band attenuation
problem with steep filters: phasing
filter resonance
Frequency response of the audio signal is Ripple -> Wavy

ripple shall be minimized.
Converters operate with sampling rates in the megahertz range sample rate is
lowered after conversion Bit count increases.
Converter chips continue to work with high sample rates for the analog lowpass Yet they
reduce in digital domain the sampling rate. From digital to analog, a digital low-pass is to be
reinstated. Good converters are characterized by the # input amplifier (preamp) Analog Low
Pass Filter Digital Low Pass Filter Robust Power / Power Good Clock
SACD = 3 MHz sample rate (1 bit converters)
Aperturfehler :
in the sample and hold stage
The longer passes the sampling, the more get the signal more and more a low-pass character.
drop Error
Loss of charge of the hold capacitor over the sample period. If the capacitor has large capacity, it
takes longer when switching between charging and discharging. Aperturfehler Compromise drop
Error
Page 239 of 466
Sample jitter
Jitter: deviation from the nominal clock of the sample and hold stage Sample Jitter = If the deviation off
hook in the sample and hold stage. If it is sampled at a wrong time because of deviation, a false input
voltage is stored. voltage error
Voltage change is high or fast when the amplitude or frequency is very high. As is sampled at more
or a few incorrect times, the sampled voltages are more or less incorrect.
The greater frequency and amplitude of the useful signal, the greater the error. Thus the error signal
generally Noise flow goes down, a high accuracy of the clock is required.
When allowed 16 bit resolution 0.2 microsec
and 18 bit resolution 0.1 microsec. deviate from the accuracy. The higher the bit
resolution, the more accurate the clock must be and the better the filter effect
in order to avoid jitter. The longer the cable are the stronger the reflection effects.
digitization
Quantizer has the task within a sample period
a) to determine the most likely of the input voltage corresponding data value
b) this issue as a binary meter measures and are income from
binary
PCM When Pulse Code Modulation is the value of the data word the level of
Input voltage.
If the value of the data word is proportional to the input voltage is referred to as
linear PCM.
linear PCM
5V
2000
10 V
4000
VST PlugIn works with PCM algorithms used: PCM:

no PCM:
- CD Audio
-WAV File
- Audio DVD
- SVCD
- Internet Radio
Quantization interval Q is the smallest voltage difference to a quantizer

Page 240 of 466
can to recognize and represent a dequantizer or more simply, the grid size of the
quantization.
When converter is the Q no unit of length but the tension. The word length
determines the sizes of Q.
eg
16 bit
16-digit combination of 0 and 1

65536 pieces
The number of Q is calculated by 2 high word

width = 2 n quantizers operate with linear
PCM. They are unipolar
come only 0 and positive.
positive voltage range and zero volts quantizer

always complete.
8.7730 Q gives Q 8
Real quantizer would Rounding, but it makes no difference.
Transfer characteristic:
shows the dependence of the output signal to the input signal
analog In A / D and D / A conversion analog Out
In analog signal and analog signal out of the same signal are compared.
ideal characteristic real

characteristic curve
1 Q length
1Q 2Q 3Q 4Q 5Q 6Q 7Q
Max. Error: 1Qpp (Peak to Peak)
The voltage error can unsubscribe maximum only within a range of a Q. The distance of the real
characteristics of the ideal characteristic is the error and a maximum of 1 Q.
Above Q of error is too large nonlinear distortion. If an error signal from another signal
depends, is called distortion.
Page 241 of 466
Calculation of distortions
-THD
- Signal to Noise Ratio
Specification:
The total harmonic distortion (THD) is the proportion of the non-harmonic distortion of the total signal.
Linear distortion: for example Altered Frequency response (EQ) the proportion of THD you
are in THD (Total Harmonic Distortion) THD:
((Max. Error) / (maximum level)) * 100%

((1Q / 2 (high n) Q) * 100% =
in the digital domain
THD = (1 / (2 n)) * 100% rounding error

in percentage n = word width of the
quantizer
Dezibelgre for the value of the rounding error / quantization Signal to Noise
Ratio
S / E Ratio
Specifies in dB, how great the difference in level between the useful signal and
quantization
is. The larger the value, the better opposite of THD expressed in dB. The greater the distance
between the useful signal and quantization error, the better the S / E Ratio.
S / E = 20 log * 2 (high n)
= N * 20 log (power of 2) = n *
6.02
with every additional bit increases the S / E ratio by 6 dB
Word width n
THD
S/E
1 bit
50%
6.02 dB
2 bit
25%
12 dB
3 bit
12.5%
18 dB
4 bit
6.25%
24 dB
In the quantization (non-lin.) Creates distortions

Average level of distortion
there arise overtones
RMS test value instead of max. Error in sinusoidal signals -1,76dB
Page 242 of 466
dithering
without dithering: quiet signal components disappear in the distortion of the quantization. in quiet places of
the piece.
In Hall, spatiality effect fades, the overtones in the background. Working with the
largest possible word lengths to dither later order. Quantization error depends on the
input signal
therefore distortion
It is desirable always to dekorrelisieren the quantization error (rounding error) from the
input signal.
We make the mistake, regardless of input.
The may be accomplished by that before quantization to the input signal a random signal is added
noise
The noise amplitude should be in the order of the error amplitude. The error is completely
random because of the noise. The quantization error is then only dependent on the total
signal because the signal contains a random component (noise)
can therefore be assumed for identical input voltages different values.
However, the rounding operation is not completely arbitrary, but depending on how close is the input
signal of a scanning stage.
The closer the signal is located on a grid level, the more likely that rounding is there.
Rounding error is not more than 1 or 0.9. That is, it must be added a random number between 0.0 to 0.9.
Only 0.0 leads to rounding down to 8. Any other random numbers lead to rounding up. 9
is rounded in this manner, the number 8.9 frequently, one contains an average of 8.9. So
resolutions are achieved within a Q.
Benefits of dithering:
- The quantization error is randomly distributed over the spectrum (noise)
-Distortions are eliminated.
-Resolution within a Q is possible (average)
- Characteristic is linearized from real to ideal characteristics.
-Noise Flow is transparent by dithering. Today
converter:
- Sample rate in the megahertz range
- Word length of 1-5 bits.
- Reducing the sample rate and compensate for the word length.
- Signal must be dithered vigorously, working with average values.
Oversampling: Works with higher sample rates than indicated.
Page 243 of 466
Sapmlerate (high) + noise, even distortion receiving and converting even at low depths word.
Error has a peak to peak value of 1 Q. noise has a

value of 1 ptp Q.
clipping
Special noise has peak to peak value of 2 Q + useful signal
Dear 1 bit converters used so that the curve / curve remains linear. remains linear
real characteristic curve
In 7-bit converters, the characteristic is not entirely linear. This means that the distance of the
grid levels is not exactly the same size. It takes several quantizer make random error in the
curve. AKM converter chips Multibit converter 123dB Signal to Noise Ratio
Dithering the digital domain:
Dithering at the digital level is always required if the word width has to be reduced. eg 24-bit
session bouncing down to 16 bits. Normal sequencing software works with 32-bit resolution:
truncation
High word length of 24 bits truncation low word width of 16 bits
remaining 8 bits fall off with its contents. They are

cut off.
dithering:
+ (Zufallsgen.)
High word length of 24 bits
8 Bit Cut
low word width of 16
Bit.
The randomly generated 8-digit random numbers consisting of 0 to 1. Adds the 8-digit number
to the 16-digit number and writes from the last 8 bits. Random
8 bit
24 - 16 bit
truncation
,, 01 [11101001]
dithering
Forced round (sound leakage)
,, 01 [11101001]
00110001 Random number 11
[00011001]
16-bit value ends with 11
the last 8 bits are truncated.

Page 244 of 466
by adding the 8-digit and 16-digit number, the information of 8 bits are transmitted to the 16-bit The
value of the 8 bits is rounded and has unnecessarily precise content and can be / separated deleted.
No ultra strong noise shaping when the signal is still being processed dynamically. (Compressors)
only prefer mastering

Noise Shaping
(Reduce word width and noise relocate)

Noise Shaping makes the dither quieter. The noise is shifted to where it is no longer audible.
Outband NS:
We shift the noise above our hearing range. We require high sample
rates as to accomplish that. High sample rate = nohe Nyquist
frequency noise above the 20KHz limit.
Inband NS:
Psychoacoustics is used. Noise is taken out in the areas where our listening area is the most
sensitive and relocated to an insensitive point. nevertheless noise reserves equal energy. UV22
Apogee Noise Shaper, Nyquist dither dither Power Noise Shaper IDF Dither
L2 limiters
There are several Powevarianten:

POW-R1:
1 kHz sine at -80dB (44.1 KHz) with POW-R1
The Big part of the noise energy is the height of the Nyquist frequency Nyquist dither therefore called.
POW-R2:
but in the areas where we hear Wider distribution of the noise well, we have less noise.
Page 245 of 466
POW-R3:
Lowering at 150 Hz the noise. Quietest Noise shape method. At 8 KHz raising. At 12
KHz lowering. Above 19 KHz raising. Outband Noise Shaping
1 kHz sine at -80dB (96 KHz) with POW-R1 noise

increase only at 48KHz
better by the higher sample rate.
Bob Kats Mastering Audio POW-R2

96KHz
Raise of 8-48KHz noise.
POW-R3 96 KHZ raise 14-140 KHz noise. Digital:

Add Mixing together subtracting Mixing together with rotator
multiplying dividing Increase humiliate
16-bit audio and 10-bit fader value at 10-12

bits
4000 different fader
2 high 16 Options for audio word 2 high 10 Options

for fader values
2 high 16 * 2 high 10 = 2 high 26 possible combinations worked My Audio
claims a word length of 26 bits. The program dithers or makes Truncation after
each edit to the word length to be kept low. Floating Point Representation Floating
Pro Tools works internally with 48 bits.
Very high headroom. But all inputs / Inserts work with 24-bit. When a signal goes to
Aux Input or effect, it is runtergedithert from 48 bits to 24 bits. Even if the signal comes
out then with 24 bits. Dithered Mixer with ProTools dithers everywhere instead
truncation to make.
Sounds internally better example for analog summing 24 bits with Dithered Mixer.
Samplitude With global off and settings from dither and dithers in all major areas.
Wahrscheinlichkeitsverteilungsfuntion:
Property Density Functions:
Page 246 of 466
Indicate the likelihood of events. Equal distribution of

probabilities.
Rectangular PDF:
Rectangular probability distribution function, each dither. pay all have equal
probability. Rectangular equal distribution.
Triangular Probability Distribution Triangular PDF
-Q
-Q/20
+Q/2
is bipolar
Values of noise are random sounds in the quantizer at RPDF dither not
particularly good, noise can be improved loudness of the noise is still
dependent on the input signal.
RPDF +
TPDF
RPDF
Triangular Probability Distribution

Ideally dither
-works exactly quiet noise

-the probability distribution does not affect the frequency response.
eg Power Dither / Noise Shape.
Crest factor
Specifies the level difference between peak and RMS. Square wave Crest factor
= 0 dB
Sine wave Crest factor = ((peak) / (square root of 2)) = 3dB measure of how loud
vibration. The volume is not the same waveform. Amplitude response is not the
same waveform.
amplitude response
waveform
Page 247 of 466
RPDF same peak level
sounds louder
TPDF Peak level doubles, but achieves rare.
Digital Transmission:
Between transmitter and receiver is a channel.

Communication between transmitter and receiver. Transfer
of low and high level can be: power on, Storm from + 5V,
-5V
reference voltage
Tbit = 1 / fbit
Receiver can distinguish low and high well. Higher than
U.sub.Ref High level Lower than U.sub.Ref
low level
Pretty tough
can Stranflle cope better than analog.
For transmission, we need interfaces.

Digital interfaces:
SP / DIF =
Sony Philips Digital Interface 2 channel

up to 24 bit word length of transmission:
Coaxial = Electrical Transmission
Cable with a conductor and in a screen with RCA

connectors. Up to 96 KHz with 1 cable Up 192KHz
Bitsplitting Toslink: optical transmission
AES / EBU = Audio Engineer Society European Broadcast Union

2 channels of audio to 96 KHz with 1 cable 24 bits per
channel.
Transmission over a balanced cable with XLR plugs.

ADAT:
Alesis has invented a digital 8-track recorder. This has made SVHS
added (videotape) The beginning of the digital recording in
semiproffesionellen Studio. 8 mono and 4 stereo trace
Page 248 of 466
24 bits per channel
working up to a maximum 48 KHz. Optical

transmission Toslink
TASCAM DTRS:
Professional / Interface Devices. 8 tracks

recordable.
4 tracks Longer recording times of up to 48 tracks receivable. HI-8
cassette with 6 devices. has its own interface: the TDIF interface
Electrical interface 4-core cable with multipin connector 25-pole Sub-D
connector with 8 channels max. 24 bit
with Bitsplittung to 2 channels up to 192 KHz otherwise 48KHz.
MADI: Madi 64 channels, 24-bit

Interface: Coaxial cable BNC connector
(RF connector) must be externally clocked from the word
clock.
Optical variants = Profi Opto connector Input - Repeater
(copy) output to 48 KHz
192 KHz at 16 channels Bitsplitting
Advantages and disadvantages of phase optical transmission
advantages
-no electromagnetic interference (interference)

-Galvanic
Galvanic isolation no ground loop
-large distances can Easily transmitted through a repeater.
- redundancies Reserves for equipment during outages.
Disadvantage:
-higher mechanical sensitivity of the glass fiber (fiber optic cable)

Cables are expensive)
To half Sample period offset sent in series.

- parallel to serial conversion
} Two designation
- Multi plexing
Time Division Multiplexing (TDM) zip
system for data
2 aligned to each sent samples are sent to a track. Samples are on a track. Sample
volume is higher. Usual way you get several channels on a transmission path.
as per sample period transferred 512 samples.
Helical Scan:
Problem: magnetic tape must have a very high frequency, played very fast
Page 249 of 466
will. Digital audio signal has frequency in MHz range because very frequently changes
per second between Low and High. Therefore, the magnetic tape must run fast (762m,
1000 sec.) There is a solution. Analog video recording very high frequency. The
oblique video heads describe the tape obliquely with a very high speed and the tape is
fully exploited and incorporated, instead of only at a part inclined head drum by the
oblique recording.
This could be used to record the audio data. (ADAT) drawback: tape is
mechanically loaded through the loop. Opposite of helical scanning is longitudinal
recording. Dash machines put through clips heads ago that are optimized for
recording. Consuming and expensive. Dash (Digital Audio Stationary Head)
Studer D827 MCH
48 tracks in 16-bit 24
tracks in 24 bits.
Bitsplitting:
Audio data generated too much or too fast, are divided by a unit of 2 tracks.
eg
24 bit
16 Bit 1 track 8
bits 2.Spur
eg instead of 2 channels 1 channel summarized} Bitsplitting Dual Wire ( Bit

splitting) 2 cables for 1 channel
Quad Wire
4 Cable for 1 channel
Formatting:
Encoder makes the data for the channel finished and formatted the Kanal.-
1.Kanalmodulation:
Representation of 0 and 1 on the channel. Low and
High levels are codes for 0 to 1.
2.Clocking:
Transmission of the bit clock.
If the bit clock in the signal or externally via a converter clock? Which
signal is to be transmitted.
Page 250 of 466
framing
- Structure of transmission (frame condition )
receiver Channel modulation 0.1 will be sent instead of low and high Where is eg the
first bit of the left channel. we need sync words:
Drawing that say what happens and how it is to process 0.1. Frame has a
1.Synchronwort and
2.Nutzdaten (Audio)
We also need
3.zustzliche data purporting eg =

How many channels we have a total, after
find 4.Checksummen error
5.Fehlerbehandlung
If the error is corrected and deleted, etc. 1
NRZ (L) modulation Level
NRZ preamble for whole class of channel modulation

Non Return to Zero
Definition:
0 = Low 1 =
High
applied within devices at interfaces but never. Also has no clock.
NRZ (M), and NRZ (L)
NRZ (M) Mark A 0 no change, a 1 = Change NRZ (I)

Inverted
Used in audio for CD / DVD, Madi However, with NRZ (I) no clock is
transmitted, no change.
Biphase (M)
Binary FM gold level

- change after everyone bit period Tbit
-0 is not a change within TBit
-1 is a change within TBit
Page 251 of 466
- Because of the golden rule, a clock is present.

-Frequency is higher because it is changed at a 1 within Tbit.
- Change clock font
SPDIF, AES / EBU
LTC SMPTE timecode used.
8 to 14 modulation
- Group code
general
Encoder looks to the Group Code groupings of bits instead of single. raw data word Channel
data word (Greater) EFM: 8 Bit
14 bit
-Better distinction of the channel data words is safer
But EFM is not only Group Code but also a run-length limited code. (RLL) in general: Run Length
Limited (RLL) code
- Certain minimum and maximum distances between changes may
not
are out of range.
The frequency may be higher, but may not be lower.
EFM: On a "1" (AC) have at least two "0" (no change) and not more than ten follow "0".
Codebook of EFM is much larger than other channel modulations. For each
combination of 8 o or 1, a combination of 14 bits must be followed by + 3 merging bits.
Data bits 8 bits dates
Channel Bit 14 Bit dates
In addition to the 14 bits, there are 3 merging bits
They are always packed on 2 consecutive EFM data words. The are used to
assemble the EFM words.
01100101
01100100
8 bits
8 bits
0100100100010
[0 0 1] 000000000100010
14 bits 3 bits 14 bits
NRZ (I)
Piet and Lands 4 T
3T 4T 4T
9T
4T
Alternating with each. 1
Page 252 of 466
Literature: Ken C. Pohlmann
Principles of Digital Audio McGraw Hill T = 1 bit

period
CD production
Due to mass production, the mechanical CD production more quickly (pressing) than digital (burning).
Glass Master Preparation Photoresist everywhere the same thickness glass fiber is everywhere with
paint coated is sensitive to light. Laser (light) is at the recesses exposed (pits) laser is on the increases
unexposed (Lands) CD track is a spiral
50% distribution between lands and pits. Etching bath

makes the footprints. Embossing copy (negative of the
CD)
Data layer is damped purely dominated and aluminum.
Cd reading:
country reflection Pit extinction
Laser spot diameter is wider than egg Pit or country. That is, there will always land with
the reflected darken the pits. The point of impact of the laser is greater than the width of a
pit. Therefore, even when a pit always illuminate a surrounding land mitbeleuchtet.
As between the pit and land exists a difference in height of lambda, is the path
difference of the two beam portions lambda. Thus, the reflected light is released from
in a pit. Pit appears dark.
country
Pit
lambda = 180 degrees Phase cancellation is decisive the difference

between pit and land. Instead of the laser only a pit or a country reads.
Therefore, the point of impact Pit and country must illuminate.
Page 253 of 466
Burned CD:
Upon firing of the laser is so strong that it shines through the dye layer and marked.
From reading the laser is appropriately strong that it is enough to read the labeled CD-RW
poorer difference between pit and land
Dye layer restorable.
For premastering CD-R is not optimal.
Exabyte, DLT with error correction and error log. CD is read from the inside to the outside because the
outside, the risk of damage is greater.
CD 780 nanometer wavelength DVD 650 nanometer

wavelength laser light which is used to highlight the CD
does not always rotate at the same speed. Record
rotates evenly fast.
Constant Angular Velocity (CAV)

Outside it sounds at the same pace, unlike interior.
eg 33 1/3 rpm
CD: Constant Linear Velocity (CLV)
eg 1.2 to 1.4 m / s
leads to a rotation speed of 200- 500 rpm. But where the player knows
how fast to play the CD. Cd Player has its own clock.
44100 pulses the second. Motor rotates with average speed. CD player has a buffer.
This latch regulated with the engine running speed. cache
full
faster motor movement empty

slower motor movement
everything depends on the internal clock,
Page 254 of 466
CD Framing:
CD frame recorded audio information from 6 sample periods: 12 * 16 bit
12 audio data words 24

Audio icon
24 bytes Audio
additional data 1 byte subcode
8 bits = PQRSTUVW
PQ Editing: set the track mark,

Information to track start and track length, end in PQ bit
CD text and copy SCMS are all in the subcode 8 bytes were added Checksums
for error correction 24 audio bytes / Symbol 1 subcode
8 byte checksum 23 bit sync words chosen so that the intentionally against the
channel modulation
breach to become recognized. Cd Player recognizes
then that
there is a sync word.
CD frame has 588 bits
All 6 Sample periods created an audio frame. 44100: 6 =

7350 frame / s
A complete revolution of information until it is repeated 98 Frames 7350: 98 = 75

Subcode block in the second.
75 times the second repeats a subcode block. 1 subcode block

contains 98 frames.
One should not place exactly at the beginning Audio PQ information because the player, the audio signal is not
exactly true, and the piece touches on.
Page 255 of 466
Dear set 0.2 seconds before the start of audio. 10 subcode

blocks beforehand.
AES / EBU and S / PDIF Framing:
Each frame contains audio information for a sample period. That is left once a
subframe and once right subframe
+ Synchronous word ancillary data checksums. Left and right
subframe are each 32 bits long 1 frame is 64 bits long in AES
EBU and S / PDIF. First 4 bits we Preamble. This is a sync
word.
The next 4 bits are the Aux Data
4 bits
If too few audio bits are available, the Aux bits bits are added to the audio.
Bit 8-27 are audio bits.
AUX bit is nowadays 0 if it is not used by audio bits. LSB is front and begins
with 0 at. MSB is behind starts with 1 at.
LSB front so when the Aux bit to come no one prevents it. After 1 Bit V Velocity
Transmitter can hereby inform the recipient, whether it is to signal to analog converting or not.
V == 0 audio ok DA conversion allowed V = 1, no

audio
faulty DA conversion prohibited.
UserData =
User Bit = 0 In most cases,
If left to the manufacturer, as it makes the user bit.

eg CD Text by more auffolgende bits.
Subframe takes half sample period is sent in succession. The left channel has a delay of 1
subframe, half a sample period
Page 256 of 466
subframe Links 32-bit
Frame 64 Bit subframe Right 32 bit 64 bit *
44100 Hz
Parity bit is the simplest form of checksum.

Checksum is a number which is formed by the transmitter from the user data and along with this
transfer. The checksum, error can be optionally detected or even corrected. parity:
Transmitter counts the 1 from the Aux Data to the channel status data. If the number of 1 is even, then
the parity = 0. If the number of 1 s odd, then Parity = 1. The process is called Even Parity
So an even number of 1's is always transmitted, if it should not be error 1
With the parity bits, however, only an odd number of errors reveals.
Channel status block Bit
make 192 consecutive channel status bits in a channel summarized a so-called channel status
block.
This contains the bulk of the additional data for the corresponding channel. mostly the additional
data for both channels are identical.
After 192 sample periods a complete channel status block is accumulated for both the left and the
right channel. 192 sample periods
left channel
right channel
192 sample periods
Which is the 1.Bit and which the 2.Bit? Sequence?

The beginning of a new channel status block is indicated by a special preamble of the left channel.
SP / DIF
3 Preamble
AES / EBU
each having a length of 4 bits
Page 257 of 466
Beginning of a subframe left channel
Beginning of a subframe right channel
Beginning of L-subframes and the beginning of a
neunen
Channel Status Block is but sent only every

192 frames.
As the receiver detects the preamble? y occurs

most frequently.
By rule violation: infringement Biphase (M) (no change after
each bit period)
The intended infraction the Preamble be recognized. information Sample rate /
word width 1 Frame
2 frame ....
193. frame
Left
Y
Y
Right Left Right
Left
Right
Starts after 192 sample periods again with a Z-Preamble. Same channel
modulation Biphase M transmission of audio data Linear PCM
AES / EBU S / PDIF
transmission
Sample rates are identical
Differences lie in the additional data.

Both allow the possibility whether Emphasis was applied.
Emphasis (treble boost )
Treble boost before and after digitization to interference caused to attenuate the treble.
High-frequency noise attenuated by the conversion. Thus, the frequency

response of the useful signal remains linear.
Problem:
AD was sent with Pre Emphasis Filter blank must recognize that this DA DE
Page 258 of 466
Emphasis Filter is inserting. The

settings Digital Out.
Comp or EQ fake because treble boost that we do not perceive. Especially dynamic
edit effects.
In the classical realm Emphasis is often used. Has a sonic advantage.
Differences AES / EBU and S / PDIF:
AES / EBU:
-AES / EBU may additionally transmit different information. Source Name Name of
destination
(Good for digital routers, digital Patchbays)
Time of Day. Includes a counter, intended as a timestamp. Time

information is transmitted.
Has a higher internal resistance than SP / DIF. 110 Ohm
SP / DIF
Has an internal resistance of 75 Ohm. Problem

with the copy of SP / DIF SCMS (Serial Copy
Management System)
At that time, the amplifier should be transmitted via digital SP / DIF. Fearing was built to avoid
the connection a copy to digital robbery.
DAT Digital Audio Tape

1. Copying was possible
. 2 Copying operation was no longer possible. DAT
never made it to the end user. but playing not recording
Transfer of SP / DIF to AES / EBU via a transmission converter 75 Ohm 110 Ohm mass of SP / Dif
on 1 pole of AES EBU place by 2 cables
connect.
Page 259 of 466
Troubleshooting
How do you deal with errors.
1. Error Berke Nung
. 2 error correction
. 3 fault masking
1.Fehlererkennung :
For the error detection you need redundancy

superfluous data.
Redundant data Data that no new information to bring (double data)
eg parity bit (one even and odd number of errors)
controlled redundant data. They are received in the form of checksums as additional information
(more than the information that begins at the receiver)
2.Felerkorrektur:
We need to know which bits are incorrect.
If error can be pinpointed, it can be corrected. This suggests that in the binary and
wrong is right. 01010101010101010101010101
24-bit number
3 times 8 bits
1 and 1 results in 0 1
and 0 gives 1
Problem = If more than 4 bits are false, it seems as if everything right.
3.Fehlermaskierung
1.Muting
2. Extrapolation (holding the last word)
The longer the insert, the fatal error, the shorter is the spot and
the better the inaudible.
3.Interpolation (averaging)
Signal can be reconstructed by 2 spectra examined Additive Synthesis
will.
Better if not to be masked because of the error can be read directly.
Page 260 of 466
Audio restoration:
Clippings are recognized and must be replaced. What will replace it?
Depends on the quality of the algorithms. 2 bits are not to be wrong.
Big mistake:
Burst error (burst error)
Errors in successive inferential data blocks. evil error
uncorrectable.
Definition is dependent on various cases.
Isolated Error:
Isolate Error (uncorrectable errors)
A method that can be converted to several isolates Burst Error Errors. Corrigibility is
increased. interleaving interleaving
Data is recorded or transmitted in a different order. CD data is recorded in a

different order. Interleaving needs a Buffer Bunch of errors is distributed
and the data in various

sequence
reproduced without the error of certain
Put
to read.
Enlarge failsafety for CD points to 2.9 mm which can be corrected. The CD is on it a guide mask, so
that the laser know where to radiate.
This must not be scratched. Without
error correction
Fault masking and affects the sound.
(Block Error Council) BLER
Measure of the frequency of errors,
Data blocks that contain at least one error. In Read Book Specification of
audio CD's in it that BLER shall not exceed 220 C2 errors / s.
Page 261 of 466
C2 error = correctable errors

If too many C2 errors occur more than 220,
Ultra sound masking MHz range
not be corrected only
infrasound m Hz range
Repetition digital technology:
Signal rasterized time to high frequencies at conversion Aliazing Anti Aliazing stage
Sample and hold stage makes signal

Time discrete.
Pulse-amplitude modulation switch is the useful signal at regular intervals Ts.
For each amplitude modulation there are sum and difference tones. They are formed in complex
modulations A. Every time Discrete signal has a spectrum (sum and difference bands) fs> 2 * fmax
Amplitude modulation produced by the switch
the Clock
Anti Aliazing limited the area where the highest frequencies overlap with the differential band.
Each time Discrete signal is a pulse code modulation. The higher the sample rate the lower the
requirement for the filter. Lies in the megahertz range, reason is transformed with a little bit.
(Sample rate is scaled down later and the number of bits increases) more accurate readings,
lower analog filter complexity. In the digital filter then the averaging takes place. After that, it is
scaled down in the quantizer Averaging
Sample rate on standard

scaled down
Bit number increases (Standardized for
Audio formats)
Errors in the sample and hold stage
When the switch in front of the quantizer to stay too long
Lowpass character
ideal pulse train is when the switch scans infinitely short.
Page 262 of 466
capacitor failure
As little as possible to be discharged during a sample period, the capacitor.
Ideally, it should keep the voltage constant.
jitter:
Incorrect dates are sampled noise
no random error
badly
Filter + noise Mistake is made at random, sample jitter occurs

in the A / D conversion on Interface jitter
Occurs during the transition of the data.
Sample Jitter error shows up in the sound and is refusing to be wiped. interface jitter
long data transmission paths, data transmission errors by

Spieglungen in the data stream of the cable.
Transmission Interface jitter unnecessarily when the receiver performs Reclocking procedure (Clock
Regeneration)
Wherever there is a new clock, there is jitter cable. Weiss DCS

Lavry
CD Recording to 88.2 KHz.

Dithering to 44.1 KHz.
Movie 192kHz dithering to 96 KHz. Sample

Rate Convertion
Sample rate does not change after hard limiting.
(Intersample peakes)
Internet
In D / A conversion D / A has not only a lot of headroom
For radio to create a special mix with not so strong because Limiting Radio are already automatically
limited dead.
Sample and Hold stage after dequantizing Degli Paget - Blanked because from where the sample period
reaches the correct voltage. quantizer:
Page 263 of 466
Meter within a sample period.

held input voltage and binary reproduced residual width Q = 2 high
word width
linear characteristic optimally.
but characteristic is stair-shaped, that is always the same distortion rounding error amplitude by 10 pp
dithering:
Distortion is converted into noise = audio signal random portion is mixed with random rounding share
ie the closer the signal of the upper grid level
The closer the signal reaches the lower grid level
round up
round.
Noise Shaping:
One can most of the noise energy outside the audible range shift value of a bit number is passed to
a different number of bits before bit number is reduced. dithering Opposite of Truncation
Digital signal transmission:
receiver
transmitter channel
Lows and Highs are transmitted. Depends on the channel modulation. The channel
modulation provides various condition. Encoder turns raw data channel data
1. as is shown in 0.1 Low and Highs

. 2 clocking Transporting the channel modulation
. 3 Framing: Structural
Receiver must be able to position the
Transfer to recognize.
3. Channel Modulation: NRZ
(L)
within devices NRZ
(I) CD DVD Biphase (M)

SP / DIF, AES / EBU, (self-clocking) LTC SMPTE
From a single Rohdatenbit the channel modulation channel data 14 EFM 3 Logic Bit
Page 264 of 466
CD manufacturing Pit
dark
country bright
Interference, path differences lambda

Takes place a cancellation, the distance between Pit
and country.
CD-framing
MP3 from 256 kbit / s is already transparent.
Preferences Apple +, Advanced
Import (variable data rate)
Page 265 of 466
Rundfunkton
Historical sequence:
Thomas Edison in 1877 has developed first sound recording machine. Phonograph: horn membrane
under the membrane a Zinnwalze Zinnwalze
horizontal
needle
vertical
Sound recording was made spirally on the roller, recording and

reproduction: Nipper his Master, Voice
Record developed Emil Berliner 1887 Phonograph

record +
membrane
the first gramophone

laterally
vinyl record page font

Antriebstechnik gramophone with a governor / speed controller Heinrich Hertz:
Namesake of the vibration measurement. Effect of the electromagnetic waves explored. Knowledge:
Head above 150 Hz vibration Vibration detaches from the conductor without wire
connection you can transfer it to the air in the medium. Valdemar Poulser invented
telegraph in 1898
investigates the foundations of the magnetic recording and discovered.
Le de Forest:
Developed
electron tube
An electronic component which gain and circuit realized. Tube:
Vitreous with vacuum.

This glass body is heated to incandescence by a filament in the glass.
Page 266 of 466
Electrons are released and these charge carriers pass from anode to cathode. Current flow, electron
flow. Controlling the polarity of current flow. more / less by the control grid Reinforcing effect.
Alexander Meisser:
Internal circuitry could generate high-frequency vibrations, Wireless signal
transmission king Wusterhausen Berlin recorded by Radio FM First transmitter 1920
no options
it had to be sent live.
Detector receiver:
Volksempfanger:
Invention of the Nazis had only one transmitter "Reich

broadcasting corporation"
Philip Reizs invented the ribbon microphone disadvantage
emits little tension.

Neumannn bottle:
1 small diaphragm tube condenser microphone low frequency. , Microphone
capsule was constructed like a capacitor. Nickel membrane. today's
membrane Polyester film that is lightly gilded. capacitor
Save and unload cargo

Capsule was coated a charge by an external voltage signal In incident the
capacitance change
by changing the distances of the
Capacitor plates. Change of the incident signal voltage. Is made of
2 tubes and an impedance converter,
Recording medium:
Developed AEG magnetic tape method Radio frequency bias tape was biased with 85 KHz
then came the audio recording.
Magnet plan K4
Developed BASF, laid right polyester film polyester is

stretched before recording.
You could finally record and play back on the magnetic tape.
Page 267 of 466
Grundig TK5
Philips invented 1963 cassettes art Compact

Cassette Tape 1/8 inch 1-7 / 8 ips.
First CD Player Philips 1981

Sony and Philips invented the compact disc.
First Audio recordings were made with UMATIC NTSC. Videotapes digitized and
compatible for all.
44,100 Hz sample rate.
Public broadcasters
Each state was assigned a country broadcaster NDR
Hamburg, Schleswig-Holstein, Lower Saxony, Mecklenburg
Vorpommern North
German R. MDR
Saxony (Freistadt), Thuringia, Saxony Anhalt, 8Leipzig)
Central German RBB
Berlin and Brandenburg area
R.
Radio Berlin Brandenburg WDR

NRW one of the largest stations in this chain
West German broadcast Radio
Bremen
Bremen
H3
Hesse
Hessischer Rundfunk
SWR
Stuttgart, Baden Wrtenberg, Rheinland Pfalz
Sddeutscher R.Br.
Bayrischer Rundfunk, Bayern
ARD
Germany DW
general
DLF
Foreign Ministry
Germany radio
Association of Broadcasters
German wave
(Editors International Language)
Was originally propaganda station. was the
financed. Today very good news channel.
Page 268 of 466
ZDF Nationwide television DLR

Today a very good news channel
Germany Radio
Institute for Broadcasting DAB School of Broadcasting
Technology in Nuremberg Medienakademie for ARD and ZDF
RBT MOT for the devices in the broadcasting
Legal operating technology
financed all this is taxpayers' money. basis Treaties

stations
Your tasks.
cultural, political, sporting point inform.
GEZ 17 euros a month as a taxpayer

makes a total revenue of 850 million euros a month 10 billion revenue
in goes out for broadcasters
for staff
responsible for broadcasting is the national distribution institution
- Monitored
the frequencies of the transfer, the
transmission times of the content transmitter
responsible for control of the private and public broadcasting. Content of the transmitter, which
is sent and what is not is determined by the socially relevant groups.
Page 269 of 466
As the signal comes to the receiver?

Air channel:
receive analog by antennas
Broadband network:
In the 80s, all transmitters were placed in a fed antenna route Kanalweg 30 TV programs 24
radio programs
Transmission via satellite:
-Television transmission transmits also broadcast signals.

Electromagnetic radio frequency waves.
DAB:
(Digital Audio Broadcasting)
Audio is reduced and transmitted in data packets Digital

technology simulcast transmitter.
Send to a frequency with the same shaft at different stations. Applied to a
frequency with any transmitters.
DVBT:
Digital Video Broadcast Television
DRM:
(Digital Radio Mondial)
Data reduction of Mpeg 4 uses all frequencies from shortwave and medium wave Worldwide
Send in FM quality. Saves up to 30 to 40% transmit power.
Radio 2: main
control room:
Central space where all signals, communications and remote control cables arrive and are passed.
With digital computer controlled crossbars
even if the computer fails.
analog manual controls
OB truck:
OB vans
Technical facilities allow of Far recordings and live transmission to the station. Equipped with
extendable antennas.
Page 270 of 466
Transfer the OB truck:

MADI bertragen- About a ladder up to 64 audio channels
less Stranflle transferring image via fiber optic cable bidirectionally
OCW:
Transmission of audio signals over the Ortzubringerleitung. 2 * 2 copper wires

for transmitting
Studio level + 15 dB analog transmission level.
ISDN:
80 interface
1 data channel 2 B channels, each with 64 kbit / s data throughput.
128 kbit / s transfer rate

3 ISDN line 3 * 128 kbit / s
384 kbit / s transmission of audio signals from the OB van to
the main control room
128 KB is enough for a Monophonic voice transmission.
Radio link:
Carrier frequency signal transmission path. Parabolic
antennas with radio.
Radio link can also use to send the OB van.
OB van transmitter
OB van can send in the range of 250 MHz.
The extendable antenna transmits by itself, if not a mountain in the vicinity.
Production Studio, Director, Media

Media:
All devices to record and store audio signals.
phonogram room :
Extra, a space where the electrical and noise-producing devices
stored
will.
self-drive Centre : Developed South West Broadcasting in Baden Baden. engineer
engineer
speaker
All this makes a person.
Page 271 of 466
a space where the speaker can use everything handy. Analogue Self Drive Studio
Sound archive:
All media are required for the operation of broadcasting are stored there. Sound archive has been
digitized. Everything is now stored on large servers.
Bias
About Venue.
Served to suitable formats were dubbed in certain formats not and were sent for the mission.
ZTR
Central berspielungsort.
Foreign correspondent sent worldwide new reports and information on berspielungsort. From there it
is stored on the server of the computer, CA (100 Terra Byte).
Wireless transmission technology

Long wave
150- 185 KHz
medium wave
525 - 1026 KHz
Short wave
(Digital Radio Mondial)
5.95 to 26.1 MHz
Ultrakurzwelle
87.5 to 108 MHz
Analog modulation techniques:

Frequency modulation
amplitude modulation
Amplitude Modulation:
carrier frequency High-frequency signal
frequency
low frequency
Amplitude of the carrier frequency takes over the signal frequency. value of the
amplitude is added to the carrier frequency to. There is an upper and lower sideband.
shortwave Limited bandwidth and quality of the audio signal.
Page 272 of 466
Master control room consists of:
- Digital and analog matrix switchers and switching networks

- documentation machines
-Air check
- Transmitting amplifier, receiver amplifier
- program tuner
- Broadcast limiter control amplifier
- Radio Controlled
All public broadcasters and private broadcasters are required to have
documentation equipment. program tuner
receiver
Limiter, compressor, gate
transmitting amplifier
loudness
Perceived loudness
Overall volume of silence at an average of 40 dB loudness. Digital

compressors Go Ahead Analyser
Checks at the entrance what a signal arrives >
Pump prevents transients

be gripped. clock:
Atomic clock in Braunschweig is operated with cesium. It has an
accuracy of 1 second in 1,000 years. The Asian time is clocked
out.
Self Drive Studio

-Mixer:
4 channels ranging from stereo
1 monaural channel for the microphone 1 channel

stereo music
Working with External Voice processors as pre-amplification.
Remove the channel
control amplifier
Play presets on a smart card.

Aux paths for different effects on voiceover
Micro switches
on / off (Fader Start Controller Start)
regardless of the fader listen PFL signal
Red light at eye level signaled "I'm on the air"
VU PPM indicates peak
short Attack Time
long Release Time
1 ms integration time VU Volume Unit
10ms
1-1.5 seconds
displays mean.
Page 273 of 466
Vu have a bias (Lead) show 3dB on too

much.
shows mono compatibility of the phase position.
good correlation with a stereo signal is at -2 and -4
Gunjumeter
Telephone Hybrid:
Telephone Interfacing device
This device processes telephone signals so that you can connect it to the console. Transfer is used
to electrically insulate Modifies the device potential of the external device to the console, otherwise
hum loop. phone:
Frequency response of 300 Hz 3.44 KHz.
n-1 circuit:
All Events
Matchs need to communicate.
When you add on more people.
Everyone can hear everyone but themselves, because of feedback. plug together modules People
the transfer (especially in the open)
can communicate with each other,
EURO 2008 people are needed. Connection SP / DIF and

AES / EBU
Mass of SP / DIF connecting conductors on the pole 2 of AES / EBU. Transmitting
AES / EBU 110 Ohm
SP / DIF 75 Ohm
Setter on the radio:

Helical Scan: DAT r-DAT
As much tape is exploited by the head drum obliquely recording at high speed the strip. Timing track is
responsible that the head drum and the speed of the belt are synchronized together.
Page 274 of 466
CTL DAT could record with 16 bit 44100 Hz sample rate.

had a SP / DIF and AES / EBU interface
could Timecode record. You could synchronize the device. has a remote control
interface.
parallel remote control: operate with a cable each something much cabling. serial remote
control:
- Sony 9 pin drive function
transmits a timecode information. implemented external
arming via Sony 9 pin ADAT SVHS cassette
16-bit mode
TASCAM HI-8 cassette
Today the hard drives dominate much better the recording procedure.
Vinyl record:
First mechanical sound recording in subscript depth

was the volume frequency of
grooves Frequency scanning 2:
- Moving Magnet Magnet system electromagnetic system

- Moving scroll
moving coil
-Piezzo
Piezzo Scanning Piezzokristallen.
Turntables have a separate ground wire, so that the noise voltage is eliminated.
Moving Coil:
On the needle 2 coils wound will The movement of the needle voltage induction needle
carrier:
much higher sound voltage generating better
signal to noise ratio
Broadcast turntables with plates and saucers analog balanced
outputs high to + 6dB in 1935 began the analog
Magentbandaufzeichnung 15 inches / second 38 inches / second
belt speeds
76 inch / s
Page 275 of 466
Ribbon cutting:
Crossfade
Cut is 75 degrees to the vertical
inch mono (full track) inch stereo Tempo track

0.5mm wide inch two-track
Separating track 2mm wide
It has under it housed a center timecode.

2 inch 24 gauge
Stadardbandmaterial of Music Studios
1 inch 1 inch
8-track 16 track 2
inch 24 track 2 inch
32 gauge
Pre Emphasis Treble boost De

Emphasis Treble reduction Noise
Reduction
M signal goes directly to the output stage
S signal is processed by a crystal oscillator with a 19 KHz pilot tone. Thereafter, the
frequency is doubled to 38 KHz. Modulated signal S + 19 KHz osc. Signal + M signal
multiplex stereo multiplex FM is added completely to the transmitter. hemisphere Air layer
Junosphre strata
Introduction of stereo technology in broadcasting: Initially
2 stations transmit in 2 different frequencies, the stereo signal. Problem not

synchronized: Phase position is inter differently from L and R. MS stereophonic the
oldest recording of stereo signals. M + S left channel
LR S
Page 276 of 466
MS Right channel
(L + R) / 2 M
Also used as a mastering tool. edit signal without filter effect.

We turn L and R signal an M and S signalPilotton Subcarrier
modulation with 19 KHz.
S signal is modulated by the pilot tone as a subcarrier amplitudes at 38 kHz

(upper and lower sideband)
+
M signal at full level
+
Pilot Tone 19 KHz
is transmitted in the VHF range. At receiver Demodulates

filter 19 KHz pilot tone Demodulated signal S M
demodulated signal
lowpass
matrix L and R signal
ARI system
Motorists broadcast information Hintz
Triller
modulated 2350 Hz with 123 Hz
Multiphonics: automatic switching to stations with traffic information. Hintz trills causes that the car
radio is switched to a normal volume. Taken Power-Down Cd player is stopped and it will return to
broadcasting. Includes a small recorder
latest traffic information is retained.
RDS system
Radio Data System
Information display comes from the subcarrier 57 KHz.
A channel list for each transmitter with a different frequency to receive the station at different
ranges.
RDS system
Display max. 8 lines receivable
Identification of the set station by 8 numeric characters.
Blend technique:
Page 277 of 466
Music and Lyrics Ratio of the mixture aperture Regulation of the amplitude of the
respective signal or signals as the listener responds to the mix music is more interesting
Music a little quieter music is uninteresting Music is loud shutter nowadays:
Loop level up and down sat.

VoiceOver Blending:
import original runterziehen value and with spokesperson about it
WORD
MUSIC
crossfade
crossfader
word
music 1
music 2
Specific terms in the broadcaster:

Abmoderation
Abmod
Anmoderation
Anmod
Beginning
Ramp Mod
end
day Mod
Title beginning to Vocal
Ramp
Pad Music
Underscore
fade out
bland
Preview spec. Search For Jobs
cuene
via transmitter
On Aire
program Notes
Teaser
Bumper
indicative
Separation and intermediate Jingle
separator
Page 278 of 466
Exekativ
Stinger
program identifier
Sounder
Station ID
station identification
trailer
Promo
Editorial organizations:
Musikredaktion
Music selection for broadcasts
interviews with artists archive Care
Sports Editors:
current reporting background

reports
Current editorial
provides RvD and managing
editor-date topics Interviews and
Features
newsroom
Agency, news selection presenting
the news
Frequency Modulation:
High-frequency carrier frequency Low

frequency signal routing
If changed, the amplitude, the frequency changes.

KHz
Frequency deviation + - 75
eg carrier frequency = 100 MHz signal

frequency = 75 KHz frequency
deviation: 1075 KHz positive
925 KHz negative
Frequency of the modulated sound signal to be transmitted carrier frequency. In the frequency of
the frequency variation the frequency of the sound signal. In the strength of the carrier frequency,
the volume change of the sound signal is high amplitude and frequency change
+ - 75 KHz
Page 279 of 466
bring CD on the radio:

Register GEMA
LC Code (Label Code), the GEMA

settlement
About a publisher must be sampled music editors. distribution CD

must be in the trade,
arrange broadcasting
France Music domestic share 85% GFK Nuremberg

Society for Communication analyzed the demand of
German listeners.
Sensor on the TV Statistics is extrapolated,
how many to see what time what
GVL Social recovery for ancillary copyright GEMA distribution Reproduction

Rights LC Lable Code Honorary license department at GEMA The longer
the GEMA the more bonus.
A digital radio home

- works with servers
Data packets are sent and received. No analog 1 to 1 transmission.
- data line fiber optic cables
All produced data Reduced to Mpeg. 2
-Sendestudios without analog equipment special PCs and single pair controller. Server are
multiple mirrored 3- or 4 times. DIGA Digital Playout multitrack controller Mixtures of different audio
signals after the one player encoded MPEG 2
then on the server.

numbered on the server and addressed. Prompter
Text display behind the camera
DAVID EasyTrack Editor
Page 280 of 466
Reporter Box =
earlier recordings of reports in an OB van. Today a suitcase with

microphone and headphones connected to ISDN 128 Kbit / s
A, B channel will be together for

sent the main control room of radio station
after transmission management.
1:30 min famous feature length Turbo
- Player of the tracks
Player
Specifying the tracks of the currently

playing song
Adjustable to take place which insertion of the song, cross, etc.
min. 2 screens with waveform display + playlist. Bumper Bet

Get Jingles- sound bites in the playlist.
Separate monitor system ensured without being on air. Database

Manager:
filed by piece by date, name and artist suitable for a flexible

playing request songs.
9 ARD stations + 3 cross-DW / DLR / DLF Technical Broadcast

Formats Terrestrial, FM, Internet control pinch control knives
0dB When the line of 1 mW at 600 ohm drops gives
0.775V (dBu)
Americans say 0 dB 1 V (dBv) stereo signal on a broadcast sound Multiplex

pilot tone technology
Page 281 of 466
Signal Flow 4 - NEVE 88 RS

Outboard equipment:
-dbx
dbx 2 channel soft for sum
- TL Audio 2 channel compressor
- 3 times 192er HD Interface
- 1 time sync
switch when using Pro Tools online
- Drawner compressor Dirty Sound (Drum sounds, overheads)
- TC electronic Reverb 4000 reverb unit
- Lexicon PCM 91
- Lexicon 480 TL
-Effectron
Effectron Delay, Flanger other dry signals
Neve console:
Inline panel 2 signals per channel
5.1 mixtures possible 48 channels -Multi

track buses, 5 stereo mix buses LCR 1
bus for Surround 8 mono Aux buses 4
stereo pairs 2 Cue way
Solo Bus AFL / PFL and Solo in Place
Meters are in VU in the yellow light in the center PPM Peak

Program Meters
Meter character of channel meters adjustable.
Response time at PPM is very short, VU it takes much longer because of the inertia of the meter. PPM meters walk
more slowly down. In short signals the VU meter indicates 10dB too much, because of the flow.
PPM shows the peak level and fall off too slowly. It does not correspond to our sense of hearing.
VU RMS Average level indicator 8 Aux paths have VU meters.
Meters on channels can be changed to VU or PPM. VU's

right scale. PPM is left scale. VU is sluggish, not so level
setting
Page 282 of 466
channel :
Above is the large routing

Input Section: Line, Mic and the bus for Subgruppenkanal) On Line Input
located on the MTK output.
Signal goes both to the Line Ins and in the monitor channel.
- filter : High and low pass

Pull up and use
- Dynamics :
4 Controller for Gate 4 controller for Komp. / Limiter Gate Key (purple color
border patch occupancy gate inputs) SC EQ
EQ purely from the signal out and into the sidechain. Sidechain
complete Dynamic unit (Comp / Gate) Threshold controller
Pulling, control range expanded. Extremely loud and quiet signals gaten.
2.Threshold for HYST
Hysteresis: level differences between opens and closes Threshold.

(Closing Threshold is below the openable
Thresholds
quieter and better gate
as Hold Time Gate
HYST indicates how much dB you to
must it include ffnungsthreshold. THRESHOLD:

Specifies the Schliethreshold. HYST can switch to the
expander with the characteristic 1: 2 (to the right)
Range control: How much dB will be lowered (min 0 - max -60 dB)
Pulled = Fast Attack Time of 50 micro sec. Pressed =
Slow
Release Time control: 10 milli sec - 3 sec adjustable..
Pulled = Inverted (inverted)

Gate switch: gate anmachen Ducker (at External Keying) L / C:
Limiter and compressor:
Linking two CHannel compressors and gain reduction is linked.
Weaker lowering a channel is adjusted to the strong reduction of a
channel. (Always with the right channel neighbors)
Makeup Gain: Output level of the compressor controlled
Pulled: Hard Knee

Otherwise it's soft knee.
Threshold:
Pulled -30 to + - 0 dB Sonst6 -10 to + 20dB

control ranges clip limit at +24 dBu
Page 283 of 466
Until about 14 dBu leveling

- when mixed to the stereo bus to go down with the channel faders (Clippinggefahr)
Reasonable headroom blank.
edit bass and bass drum at about + 5dBu.
- reason Processing speed of 1: 1 to inf: 1
Pulled: Fast Attack 1- 7 ms Otherwise Slow
Attack at 3- 7 ms (program dependent
Attack Time)
- Release Time = 10ms - 3s
Turn hard right is Auto Program Dependent Release Time

(good for vocals)
Opto Compressor: reaction time inevitably Program dependent

Urei LA 2A tube compressor
Principle: LED faces a photo resistor, signal goes through
a lamp through. JE by volume, it is light or dark, the photo resistor arranged
at the lighting and controls Gain Reduction.
Long GR Long Release Short GR
short release
Physical AutoRelease
Aux section : 8 Aux

potentiometers
press switched through Poti and off when pressed again each one can
individually select Pre schalten.- MTK Aux on Multitrackbus route rather than
the
Aux Bus.
Stereo switch 5/6 7/8 make Mono AuXe to stereo signals
EQ section : EQ
switch
4 bands
Mid bands fully parametric bass and treble

semi parametrically Cut Boost
Switching between shelving and Bell High Cue

narrow lowering / raising
Otherwise broad lowering and raising. Cut and Boost with
+/- 20dB for all 4 bands Pre Dyn turn EQ before or after the
compressor.
Inserts: in channel grind on the insert sends. If inserts is to is to what.

From the channel out and the tape return back. Also pre and post EQ switchable. track Master
control for the Bus Outs
Rules only the bus a KanalsPage 284 of 466
YOU Bus Out is made the direct out, only the bus signal from the channel is
sent out. Before GRP key to a group on the bus ausm channel to send
Group / Tape:
Group buses ToTape Tape Tape returns Offtape
Input of the monitor channel adjustable between bus signal and tape return signal. Chop channel
output Signal on Monitor path post fader chip channel input
Signal on Monitor path Prefader
Chop good for additional effects in the monitor path is the same like in
Channel Output. chip Completely
independent Mix
4 switch: Dyn, ins, EQ filters all individually switched into the monitor. When headphone
mixer:
Aux 1.2 to Pre monitor path 1.2 Select,
solo functions
Default = solo in place Cut Mute latch currently all solos remain with
new solos in it I / O all solos fly choice for a new solo out. reset
all fly out Chan /

Safe AFL Solo
Mon. Safe + Safe Chan + Latch = AFL Solo Ret
switch:
Important for global Solo in Place red = Solo
Safe.
green = Channel in solo Safe yellow =
Solo Safe in PFL mode
monitoring section
-Master sum
- level regulator
-speaker Solo
- DIM DIM + Level
- internal 6 Track Mix
- External
- Cue1 and Cue 2 interception, AuXe listen, pairs
Page 285 of 466
-listen several sources together

- Ext Inc
1-12 inserted on Patchbay.
88 RS Recording
Grp Input switch between Mic Line and Group (buses) DIR Direct Out switch
ToTape tap Grp Tape make Offtape signal to ToTape signal.
Determines if You get the bus or purely Offtape signal.
C / O changeover, Puts the respective channel in the mixdown status. SWAP Exchanges the
monitor fader with the CHannel fader.
Practical: In Recording Swapen Adjust Volume of Recordings on the small fader. Large fader in
the monitor path so that the mix is already present later during mixdown (only Fader exchange,
routing remains the same)
-routes
routes 1-24 Channel Routing from large routing
Small Routing: Pull Tape Return signal on Small Fader. Small make fader on
tape, so that we hear the signal Tape Return. All Monitor ins are set to tape,
receivable in the major faders additional mic signals.
Multitrack Monitor Section
Red = Green =
Group Tape
Button for all universally selectable

Mostly tape at recording. Small
Routing on Reassign.
88 RS mixdown
Small fader
for MTK
Large Fader for small routing.
Offtapes go into the entrance of the channel on the Line Ins. Input must
be set to Line In.
Pre Fader and Post Fader tap effects are admixed in connection to the monitor section via the monitor
fader. By CHOP and CHIP monitor path of the canal make for FX. To have the signal in the channel
path and not in the monitor path must be activated in small routing C / O. Feed from Aux Send AuXe
the Multitrack routing.
MTK (SSL EFX)
Instead of the signal goes there towards the Aux signal.
broadcast Mode
Monitor path receives the signal pre-fader with the chip (input) Chop (Output) button If CHipCHop is not
on, Group Tape applies.
Page 286 of 466
The same as the SOURCE switch at Mackie switched pre fader effects into the mix B path.
monitor panel :
Various interception Broadcast Mic Line

Faderswap mixdown
are universal settings between ON and OFF in all channels.
The Neve has 2 line inputs per channel Line A is

locked.
Fader Swap is global interchange of long and short fader. status Lock Lock
for Monitor Panel Status buttons.
Multitrack Meters
What shall show the meter GRP bus
levels
Tape Tape Return Offtape Levels Follow Mon Meters Switchable Grp or tape no
matter what lies Ch Current level (preprocessing) input of channel meters are
directly behind the Gain knob.
monitor Meters
mix Sel Mix one sees through the meter 6 Main Meters L, CCR, S, LS, RS interception:
Mon. Sel Displays the meter what we hear
Mix Sel. + Sel / Mon

LCRS Ls Rs
Balance control (monitor section) = L and R Adjustable.
Centrally localized signal even though you do not sit in the
middle.
TB Out: TB to Cues & SLS Cue 1

Cue 2
SLS
car TB
Slate
all get talkback communication.

Studio speaker
TB is key to the switch until I expressions
AT the beginning of the tape can speak a little extra track info
Page 287 of 466
Neve produced a 30 hertz tone. TB to Cues & SLS must
untwisted
be.
Tip for Cue Out balanced : Cue outs

symmetrical L and R 1 patch cable
Cue Out Left and a Cue Out Right
Tie Line 1 Line 2 Tie
pure-male
connect the headphone amplifier at the tie lines by XLR. Stereo 2:
4 Headphone Amps: Amp

2,3,4 are stereo
2 Listenback inputs. Return TO button to listen to it.

Cue section :
Cue Power-on and search for a source above: Mono or

Stereo:
eg Mix or L / R mix and integration of effects in the headphone mix as EQ or Dyn or dry
signals via stereo aux returns, Reverb Returns. On the Patchbay the Reverb Return are
outputs of the Hall devices.
Console Reverb Returns : Stereo
Outs:
1. Lex Main Out

. 2 Lex Aux Out
. 3 PCM 91 Out
. 4 Reverb 4000 Out
Hall devices are already fully normalized to the console. Aux 1

depends Lexicon Input A Aux 2 depends Lexicon input B
That's 2 mono inputs and 4 stereo outputs they give out.
Patchbayanordnung
Up Red Mic Lines 1-48 receiving space 1

Normalized to Mic In the console
green
Multitrack Return 1-48

Pro Tools outputs normalized to Line In of the console.
Yellow white
ISR send and return are always active
White Orange Group Out

Page 288 of 466
buses Out
Normalized to Pro Tools Input 1-48

Purple Pink
External Keying the Gates 1-48
surround
Large routing route Subgruppenrouting and ToTape signal to L / R. Smaller routing

assignment of stereo busses Out
Master MIX bus outputs
Reassign panels
10 column Outputs 6 lines

Speaker outputs
Individual mix buses are on each Box Outs

Master Mix bus inputs can be split to 2 Mix Bus units for the postproduction.
surround mode
Set surround
Surround button (marking the meter changes)
This surround does need to be activated in the LCR column 3 buttons above.
surround possible
Surround Button Diagonal Mix Bus

Output LCR buttons on.
LCR is an additional bus for panning. Small

Routing:
Ausm Channel in the LCR route, so that the signals in the
surround
Mix are routed

ST Near
narrow divergence, almost mono (dialogue, Center, vocals) wide

divergence broad signal (via L, C, R)
L / R surround switchable by small routing 5/6
Sub not realize by . send signal to LFE on Aux. 1 press Aux 1 to Sub and
untwist Aux send sub signals without panning on a channel.
Sub Channel Edit Pro Tools as an insert on Surround Insert Send Return Phase meter switch.
Correlation shows how the phases are separated. Ratio of the phases in a mono signal is 0 degrees.
One should have a degree of correlation of 0.7.
Page 289 of 466
Headphone mix in Recording Status
Small Routing (Tape Return) key signal 1-2 on Aux to headphone Pre switch and
connect Level + Pan. In cue 1 Enable 1-2 and dine on headphones Aux . Tape
Return 3 / $ 13.14 and are reversed.
EQ in the tape return signal
SMAll monitor EQ in the channel and is processed cues are pre EQ

headphone paths Pre Fader Monitor
If you want that Eq setting in the headphone mix then comes at Cues Post EQ activate,
Although the console post Eq works can be seen in each channel by the O / D button in each channel listening
monitor the signal ToTape nevertheless directed Offtape (officially) For O / D headphones get the signal from Pre
Fader Grp.
create subgroups:
In mixdown Mixdownstatus global Small Fader MTK
Large Fader
Subgruppenrouting
Signals are at Pro on Tools. Large Meters in Tape mode Line In value is displayed. press CH
13,14,15,16, ... to mix turn over small routing Grp small fader Buses
are on.
small fader on Tape
On the routing of Subgruppenkanal is set.
The Subgruppenkanal is on GRP therefore switched this ToTape group signals abgreift-
The routed channels are all switched to C / O in small and routing from Mixbus
1-2 separately. For getting the bus channel routing to the mix 1/2.
Page 290 of 466
Strip of Neve 88 RS
Bus routing:
Here, the buses can be selected
become,
by the red buttons may
determine to whether bus or bus
1-24 25-48 dials
Stereo Bus routing

LCR is used to route the signal to the
LCR bus By Pan Controls to fix the
panning during recording, the button
must be pressed PAN so that the
controller is turned on
Page 291 of 466
Input Section:
-
Line Unity is 0
Mic In (20-70dB gain)
With key C / O choice between mic / line
By pressing the MIC controller switches to the

phantom power on, the lamp lights up red
- 20dB pad on Mic In
Rotator applies to both
GRP button makes the whole channel to a group
Filter section:
-
HPF + LPF
Separated
By pulling on one these (lit lamp) to
Dynamics section:
-
consists of gate / expander (left side) and

compressor / limiter (right side)
KEY brings a sidechain signal via the patch bay in

the section (48 pcs.)
SC-EQ switches the equalizer into the sidechain
gate:
-
HYST is the controller with which the difference

between opening and Schliethreshold is set. If one
continues to rotate the knob below zero, so click one
of these and you have an expander.
With THR of Schliethreshold is meant by pulling the

regulator it is again reduced by 40dB
RGE
(Range)
is
the maximal
Lowering (o to 60dB). roam

Page 292 of 466
one obtains a fast attack time (50 microseconds), pressed a slow (500 microseconds)
-
REL (Release) is adjustable from 10 ms to 3s. By pulling you get a Ducker
There is no hold function, since it has a hysteresis
By pressing the gate switch, it turns on the gate
Compressor:
Pressing L / C to turn on the compressor
By the arrow key, the compressor right will be coupled, however, only the gain reduction
is coupled. One can also quite couple all together
The GAIN control governs makeup gain. By pulling you get HARD KNEE otherwise
SOFT
The Threshold operates from -10 to 20 dB, by pulling the control range is again lowered
by 20dB
COUNCIL (ratio) is from 1: 1 to limiting. Naughty has the Poti an Attack Time of 3-7ms
pulled by 1-7ms (almost)
REL (Release) operates from 10ms to 3 seconds when turning the knob fully clockwise
until it locks, so the release time automatically works
Page 293 of 466
Aux section:
-
to turn on the different ways you have to use the

potentiometers press (lamp)
by pushing the switch PRE, one on each route

PRE-FADER, are not pressed all POST
by pressing the buttons MTK sends one each way on

the MTK OUTPUT
in 2er couples the paths are coupled,

ie that the odd number is the level pot and that
means even number PAN
Page 294 of 466
EQ section:
-
with the INS key inserts can in the channel

be ground
these
are then normally POST EQ, but can be switched

by the PREQ PRE button
-
By PRE-DYN EQ section is switched before the

Dynamic section
4-band EQ
the midrange bands Fully parametric
the two outer bands have only CUT / BOOST and

Freuq settings, but can be set to a high Q value,
normally LO-Q, and the other switch between
shelving further through the switch HI-Q
and
Peak characteristic
be converted
Page 295 of 466
Monitor Section:
-
With
the TRACK-Poti
is
the
Output volume of buses regulated

-
pressing DIR (Direct out), the controller is the main

output volume control
with GRP / TAPE you can choose what the fader

should govern. Unpressed (TAPE) is to hear the
signal off tape, the button is pressed (GROUP), he
refers to the group
O/D
C / O (changeover) switches in the local

Mixdownstatus
SWAP exchanges the Small Fader against the big

fader and vice versa
CHIP / CHOP (Channel Input / Output Channel): the

LED red (output), so get the monitor channel its
signal POST ChannelFader; it lights up green (input),
so get the monitor channel its signal PRE
ChannelFader
Page 296 of 466
Furthermore, Dynamics, reinzuholen Inserts, EQs + filter

individually in the monitor path
AuXe pairs reinschaltbar
RET: this button has three modes: 1. Lights red channel is

solo safe; 2. Turns green Monitor Solo is safe; 3. Lights amber
Monitor and Channel are Solo Safe
Solo refers to SF and LF
Mute applies to SF and LF
FASHION
SEL
The green button switch the output in small routing to the

appropriate outputs
By pressing the LCR key is the faders used for surround

mixing by PAN as Pan Controls
NAR / ST
Main Fader section:

-
through the switch AB can call 2 different mute statuses, by

pressing Cue A cue or B in the Main Section
solo
Mute
Page 297 of 466
Film and TV sound

1930
optical sound
EPRI Standard for Mono optical sound
1933
"King Kong" with first sound effects development of cinema from Europe to
USA filming in Atteliers (giant halls staged the environment)
1935
Woofer to tweeter 500Hz from 500Hz

Fixed with a crossover.
1935
1937
Stereo optical soundtrack by Alan Blumlein
Multichannel sound recordings at Universal "Hollywood" ideals

shooting because Regenlosigkeit.
Radio frequency bias recording on tape was slowly developed.
1938
1940
1953
Magnetic recording sound 35 mm tape by perforation synchronized with the

revolving image
Academy Curve
Pre Emphasis but at 8kHz frequency range was
Over.
"Fantasia" by Walt Disney on 70 mm film
First three-channel multi-channel recording with surround playback 2 projectors
coupled controllable via Pan knobs.
"The Robe in Cinemascope" with Anamorphic lenses 4: 3 16: 9 A man can
perceive sharply at an angle of 60 degrees. Sakkadisches See Anamorphic
Lenses:
Camera recorded broad signal compressed and then narrow to 4: 3 PAL.

1954
1965
1967
1970
1971
1972
1974
"Oklahoma" in Todd-AO 65 mm film strips with 6 magnetic stripe first surround

sound playback.
Dolby NR A Tape
Noise Reduction Noise Reduction with pre emphasis and De Emphasis
"The Graduate" first pop music in a movie erotic
IMAX premiere at the Expo in Japan is based on Big screen, high-seated
audience, high image resolution, viewers can decide where he hinguckt. 70 * 70
mm format Pro frame.
Dolby Model 364 Cinema Unit
X-Curve as a new standard
Dolby Stereo analog matrix system from Kodak

L, C, R Dolby Surround channel mono tape Limited to 7 kHz.
1974
1977
Earthquake in Sensurround (-Multiple subwoofer)
"Star Wars" in Dolby Baby Boom

L, C, R, S and 2 times LFE (Low Frequency Effect) subwoofer only bass
frequencies below 150 Hz.
1979
LFE: All low-frequency components such as shots explosions are reproduced by the
LFE channel.
"Apocalypse Now" of a first appearance on the 5.1 system.
Page 298 of 466
1981
1983
1987
Digital Sound Fluorsid
"Return of the Jedi"

THX standard
Standard fulfills the specific acoustic and electronic condition.
"Robocop" in Dolby SR (Spectral Recording Stereo) Specially
compressed audio format
1988
analog optical sound
IMAX Digital Sound

System 6-channel digital sound L, C, R, Ls, Rs, Top, SW + Too expensive
in productions
Later DTS
HDFS system of Kinoton Studer,

Arri
Full screen width, 2 parallel CD-Roms
Arri is the world leader in lighting equipment 1990

Digital Laser Sound
Audio CD has been converted to an optical disc. Digital audio data and image data, and has
been approved.
1990
1992
CDS (Cinema Digital Sound) Kodak

"Batman Returns" Dolby Digital Spectral digital format
what we hear in the cinema and on DVD
AC3 audio format, sound system in Dolby, binding 356 kB / s for 5.1
1993
"Jurassic Park" in DTS
1993
Audio was leaked separately and purely mixed.

"Last Action Hero" In Sony SDDS 7.1
System This is data reduced to ATRAC.
1997
2002
6.1 (7 Channel) In DTS Encoder Out 2 channel

"Star Wars Episode 1" in Dolby / THX Surround EX (6.1) Center behind
it 2 left and right surround channels in Surround Center was
introduced.
Target: can represent any point in the acoustic space.
DTS EX
The 6.Kanal is not a fixed channel but matrixed from the old system.
film sizes
8 mm film
16 mm film
70
Films are shot out and brought to 35mm due to
blow up.
35 mm film
70 mm film
Cinema Standard Format
IMAX special films Rendering consuming and

expensive (+ 70mm)
Page 299 of 466
optical sound
Narrow and wide light source is exposed to tape. This leads to the recording. Photocell converts
light into electronic voltage and this in turn is Transformed into an acoustic signal. It shifts the
sampling of the sound later than the picture. The transfer is done by pulleys. Sound 21 images
transmitted later as an image. Digital optical sound is written with a laser gun. movie
90-105 minutes
3500m film
For space saving it is played in a roll and wrapped in the other view.
Film is mechanically stressed and worn.
mixture
Dolby Digital SR
D
Dolby stereo
Original Language (Background International) Audio
Language (Atmos, Stereo)
premix
Multitrack music (special effects)
Original dialogue
dubbed language
main mix
music
Atmos effects
-6 channel dialogs are output from the Center Mono

- 6 channel music
-6 channel sound mixing and
encoding
Each film-coated mixture must be present in 2 variants
dub negative on light tone and and

35mm film image
clay
positive copy,
Optical sound may be no sharp edges are created, blurring.
Film are in the dimension of cinema equipment produced otherwise does not sound fine. Film has the studio
the same acoustic characteristics offer as the cinema is ready to reproduce.
Page 300 of 466
FILMPRODUKTION
quest givers
producer
production Supervisor
production manager
Accountant
Author, Dehbuch
Director
Reg.Assistent
equipment
script
supervisor
agencies
clay
Actor, extras from casting

Sound Assistant
Cutter,
mask
Design art
director
Editor
2.Maske
assistant
props
foreign
Avit- non linear
system Editor
Costume
assistant
wardrobe
props
indoors
Gaffer dolly grip

camera
Elektr.Datenbank
Lighting and stage

assistant
Camera Assistant
Material Assistant
Page 301 of 466
Portable Tonaufzeichnungssysteme
NAGRA V
2 tracks receivable FOSTEX
MiniDisc MDP 500 good preamp

outputs good
receivable on the flash card up to 5 hours HDD
Recorder
8 audio tracks with mixer and effects
10 seconds pre-record buffer takes 10 seconds before the recording on HHB PDR 2000
English Supplier of high-quality film and television equipment Aaton Canlar

waterproof Recorder. ZAX Com Deva
Hard disk recorder with 8 channels

Fostex PD4 DAT recorder recordable 2
tracks Fostex PD 6 DVD Recorder Uses
playback device
timecode
Indication on the timecode flap

1. scene
2. Camera number
3. version of the recording
Timecode generators generate Quartz controlled and synchronize the cameras, high swinging in
small parts more precise
Sequential image transfer

Strobing (16 fps) PAL 625 lines
1875 first image transmission with transmitter and receiver line screen. Principle of spiral
image sequence from outside to inside.
Page 302 of 466
1897
Developed the Braun tube electrodes used in tubes and screens Oszilloscopen
The high voltage between cathode and anode electrodes were acceleration and
follow up a Lichtimitierendes formations (white spot where electrodes occur)
19 cm
Unlike the tube for reinforcement purposes. Braun tube as the basis
for television display. Image converter tube (Zwarykin, 1932)
2 Characteristics of light
1958 = first color television NTSC 1. The spectrum of electromagnetic waves. 4 nanometer
wavelength to 7.5 nm wavelength is white light. TV light. It contains all the wavelengths of
visible light (frequencies) (white noise = same sound power in 100Hz bands) (pink noise =
same sound power at intervals of musical octaves)
white light produced by the slit diaphragm a color spectrum.

Page 303 of 466
Additive color mixing
Red Green Blue mix all the colors together.
Colour transfer
Dichriotischer mirror
Are only permeable for certain lights by its mirror surface. (Present in each camera)
Required in the implementation of optical reality is converted into a color film. At
500 nm wavelength light our eyes are on
sensitive.
Technicolor absolute compatibility for black / white TV
Page 304 of 466
RY
matrix
Modulator /
transmitter
BY
Y = 0.3 R + 0.59G + 0.11B
Principle of a CCD
camera system
: F-Bass:System signal. Red Green Blue
transmission.
Composite
Colour blanking signal includes:
Hidden shortly and the next picture can be pushed. This is the sync pulse in the video
Brightness information Color
Information
components
clock information
the even numbers. In between, a dark button so that the image in between that
F- Bass = Green Coaxial Cable
BNC connector
1 frame contains 625 lines and is divided into 2 fields. Only the odd numbers then
Page 305 of 466
Other analog signal transmissions

- composite signal
- Component video cables signal 3 Red,
green, blue
-SVHS
SVHS YC components
S-Video
Y = brightness components C =
color component
3 separate lines Y UV television signal
Y luminance signal UV color

differences signal
HD Ready Component video input 3 RGB
components as image input
Digital Signal Transmission DVI
TV sound
Television broadcasting center
OB vans (Outside Studio) plug HSR / ZGR Studio Movie director / sound control
playout OSL satellite

a) Production b) live c) Turn in live broadcast fiber optic transmission from
OB vans data rate in gigabyte range. Cologne has digitized analog cables
Network DVBT Digital Video Broadcast Terrestrial
Unlock in live coverage:
- via satellite or fiber switched in external image and sound signals.
LMS Library Manager System robotic

Kasettensysteme
Autoplay and switching of video from a large archive. Meanwhile computer server as an
archive used studio camera:
(Three channel image transmission via triax cables)
Central circuit of cameras
CCU (Camera Control Unit) with keyboard 1,2,3,4,5,6,7,8 is served by Central.

Page 306 of 466
BIMI Bildmischer responsible for camera circuit and Umblendung Grass Valley Thomson
renowned video mixer

Sony
MAZ magnetic recording Video Recording System

In the cameras small hard disk recorders are installed that can be retrieved from the CCU to the image
mixer, eg slow motion shots in the cameras themselves are generated by a buffer in the individual
cameras.
Black burst
Clock pulse for synchronizing color video clock signal to

clock all cameras
control room
Control room has individual monitors for each camera then into the mixer Lemo patchbay
Dubbing studio 1950-1980

Always used because foreign films translated into German and need to be synchronized. From
2000, Digital Workstations are increasingly being used with intergrieter image track as Automatic
Replacement dialog systems. Computer system which are operated via PS 24 interface. (Center)
Speaker stands with his back to the camera (easier to synchronize) rough translation
(Dialog book) Includes all voice applications of the film. synchronous writer Then (using the lip
movement the right tone at the
place appropriate image, pay attention to lip sync)
Taken (TC)
Labial note sounds with closed lips M, P, B

The result is a synchronous number, beginning and Endtime code the
takes.
Atomic number and what the speakers say.
Video Working Copy Band / Hard Disk Take list for MRP and voice recordings, take list Lists all takes on
with their length and number.
Odd takes on odd tracks. Just Takes on
straight tracks
Page 307 of 466
ADR system
Processor RS422 serial interface Processor VTR harddisk DAW

dubbing studio
Safety Rec
eg EDICOM II Audio
Solutions
dubbing studio
Voice actors have to speak to the image speaker attributes to repeat as realistic as possible in
the given position of an area. by means of
1 main microphone 1
room microphone
Either with a lot of reflection in the room or not in nature or scenery.

Timecode
A digital time signal which is recorded in analog most cases. It's called SMPTE. It must be recorded
continuously and ascending and are synchronized by a black burst. 2 types: VITC Vertical Interval
Time Code
TV-related timecode between 2 images is a black screen. This is used to
synchronize 90 bits. LTC Longitudinal Time Code 80 bit signal
Interfaces: Balanced XLR interface (+ 6dB)

Asymmetric BNC interface (-10 dB)
Timecode may not be copied if it is recorded. then reads the timecode ReGen Restores this recording
Digital Time format: Time Code 2 different interfaces:
Balanced XLR interface (+ 6dB) Asymmetric BNC

interface (-10 dB)
The analog magnetic recording (MAZ)

control track = dubbing studio between head drum and tape 5,75 m / s
Dolby C noise reduction are provided.
Page 308 of 466
Technical leader: TC
09.59.00.00
video
Color bar CCIR469-4
09.59.40.00
Black Mute
10.00.00.00
program start
Audio
Tone at 1kHz analog 9dB
digital 1 kHz -18 dBFS
Color measurement signal is used for technical calibration systems No beginning at 00.00.00.00 due to
technical problems at the superimposing one period. Therefore, the program starts at 10.00.00.00
oscilloscope = image metrology
Vectoskop = Spatial representation of color dots

-EBU color bars
- Grey stairs (brightness adjustment)
- Test board (geometric values from the image test)
Einmesstechniken the camera image
Beta SP Recorder
4 audio channels
2 analog Longitudinal tracks with Dolby C Other 2 channels in the

frequency modulated blanking Sony 9 Pin to play in a position large and
small cassettes. VHS Video 200
Kasettensysteme
Betamax
recorded Betacam system in Components
Video formats in television

1. Betacam SP
2 + 2 analog soundtracks
2. Digital Betacam
10 bit quantization (converter) and 2: 1 data reduction 4 digital sound tracks
(video signal has up to 5MHz bandwidth)
3. Betacam SX
10: 1 data compression by MPEG 2 4 digital soundtracks SDI A digital

video signal that comes out of a machine
4. MPEG IMX digital
After MPEG 2 4: 2: 2 standard (profile at main level)

5. DVC Pro 50 DV format with a data rate of 50 Mbit / s
6. HDCAM
Digital high-resolution 16: 9 aspect ratio with a resolution of
Page 309 of 466
1920 * 1080 pixels and a data rate of 140 Mbit / s (1080 lines per
second) Non Linear
irregular order
3 machines Video editing suite:
AVIT Master product of digidesign

Player 1 video mixer
section control
recorder
Player 2 Audio mixer

Operating system of the electronic section. crash
Record new band
everything is rewritten black image is recorded as

a sync word
start
cut versions
Assemble Editing
Basic cut versions in 3 machines Video editing suite

insert editing
Precoded cassette can exchange picture content and audio content
Hop off possible without flapping.

Backup light tone always Dolby Stereo Dolby Pro Logic Decoding from stereo to 5.1 system
through electronic manipulation
Wave Field Synthesis.
Waveform within a broadcaster Dolby E
A digital matrix method where I can send 8 audio channels by an AES /

EBU channel. Dolby E Encoder
AES / EBU 3
Dolby E Decoder
Up to 8 channels transferable MAZ Server transmission mixer and satellite

production 5.1 Mix Master Audio, Dolby E, videotape Payout Videotape
Dolby E Dp569 Dolby D Encoder
Dolby Digital MPEG, Video
Page 310 of 466
- Transport of several "programs"

- Transport of metadata (techn. Additional Information)
- up to 8 channels of 20 bits
- up to 6 channels 16 bits
- 25 fps and 29.97 fps operation
-fixed delay of one frame in the encoding and decoding
WIRELESS MICROPHONES
Transmitting by FM.
Wireless technology since the late 50s, Channel: Microphone Amplifier HiDyn
Plus (limiter) Limiter (Low Cut)
Oscillator and modulator
Recipient: Hi Dyn Plus high

Cut
noise reduction
True diversity receiver

Because wireless microphone technology works with very little level, you need a good receiver
technology. Each recipient Demodulationstool is established. 2 receiver demodulators guarantee a
better level than a demodulator.
Antenna technology:
1 to 2 4dB 1 to 3 6
dB
1 to 4 8 dB antenna
ANTENNA MIXER
sharing =
Antenna splitter 2 signals go through a route to their recipients.
Antenna mixer =
With more than 2 antennas, antenna mixer reduce 4 antenna

signals to 2 antenna signals.
Each high-frequency signal is, the more the exponential attenuation increases on the cable. If possible the
short and thickest cable use for an antenna. Coaxial cable with a characteristic impedance of 50 ohms
attenuation not greater than 6 dB at a maximum of 10m antenna length.
hinge
Synthetic fur protects microphone from wind wind force. 4

Schtz also against moisture.
Shotgun Microphones:
Microphones in the direction of the optics as Sennheiser MKH 60 lightwaves THz
Page 311 of 466
radio
MHz
Wireless Microphones:
First set the receiver then the transmitter RF level

check output intercept consider travel indicator
All wireless microphones have to run at a different frequency to the receiver. Frequency deviation in
microphones to +/- 45 KHz RF Field strength microV
RF Hub at KHZ indication in dB

- 20 dB Audio Headroom True Diversity switching (good True Diversity always change the radio frequency)
Function True Diversity

True diversity is a method to increase the signal security, such as in wireless microphone systems.
Here two independent receivers are used, each with its own antenna.
Radio waves, such as light, electromagnetic waves. Just like the light, there are shadows and
reflections. reflect particularly metallic objects radio waves or even absorb them.
The radiated by the transmitter radio waves reach the receiver not only on the direct route. Reflections
are superimposed with the direct signal. Depending on the phase the reflections to increase or
decrease the total signal at the receiver antenna. This effect is exacerbated by movements of the
transmitter antenna. The receiver thus has to do with constantly fluctuating signal field strength.
Especially when moving the transmitter antenna, it is often the case that the signal falls below the
minimum required field strength for interference-free reception. A second receiver, own, offset
antenna is indeed subject to similar fluctuations, but the overlays are occurring elsewhere or at other
times.
Thus, when combining two independent receivers each other - and switched back and forth in case of
falling below the signal strength between the two receivers, so a significant reduction of transmission
errors and thus a significantly higher signal reliability is achieved.
Page 312 of 466
Compander HiDyn plus RF domain 65dB Compressor 2: 1 noise

floor Expander 1: Set 2 squelch function (Noise Gate) off
stations Squelch equipped only at the receiver radio
receiver with a noise gate.
Never turn 2 stations on one frequency. Minimum distance frequency of 400kHz between 2
transmitters. sender MK5021
-RF energy is water always 0 potential. (Do not touch or skin contact)
-Sight between transmitter and receiver
- 4m distance not less than in multi-channel operation.
Directional antennas:
- 10 dB attenuation reverse
+ 3 dB front level gain directivity
electromagnetic waves such as light
For stone and metal, the waves are reflected.
Polarization can be horizontal or vertical but either the transmitter or receiver for horizontal or
vertical position. Antenna Boster AB1
- 10 dB gain with DC Power Distribution
attenuation: Splitter 1: 2 ASP 212 -4 dB
Splitter 1: 3 ASP 113 -7 dB Splitter 1: 4
ASP 114 -8 dB
Cable attenuation should not exceed 6 dB attenuation. This corresponds

to 800 MHz
10 m RG 58 coaxial cable with 7 mm diameter 25 m RG 213

coaxial cable with 10mm average.
Working at paths between stone and iron rather with true diversity receiver
safe reception
Page 313 of 466
intermodulation
Arises whenever two frequencies are used side by side upon thee. Overlap of sum and
difference components of the signal.
400 KHz minimum distance 3. microphones then 700 KHz.
In Ear Monitoring:
Carrier-FM stereophonic track the works.

- directional transmitter and receiver small
Minimum frequency distance from In Ear Monitoring should be mindestes 16 MHz. With multiple
channels Antenna Combiner AC. 1
Others to film and advertising tone :

1) Which switching stages, the transmit switch in the transmit director?
1/2?
3/4,
Tone generator and identifier.

2) How many audio tracks can be found at the DIGI-Beta cam that is not in the video track
are:. 4
3) Absolute Voltage level for digital clip-limit: 15 dBu (not 0 dB FS!)
4) Values for a RG213 cable: 50? and 23 dB down.
5) The city limit line is automatically switched off when the image director "On air" goes.
6) Dolby E has 8 channels of audio at 20 bit.
7) Never are 2 stations on the same frequency, even in operating a disaster

Mikroport conditioning.
8th) Dolby stereo is an analog 4: 2: 4 matrix system.
9) Microphone extension call you Angel.

10) carried synchronization between picture and sound via time code.
Page 314 of 466
automation
2 Digital devices are synchronized by MTC. Master is always
where the audio files are.
Disk, tape and Multitracker
The slow with the winding speed is Master
ProTools synchronization
-first in the transport field online questions under Peripherals / Synchronization Select
whether MTC or LTC port to be set. System Apple +2 Enable Sync 2 Digital Systems
MTC out (1 unit) MTC In (2.Gert) Then it runs.
Synchronization of Neve Console on LTC.

Master is the tape recorder. Synchronizing signal recording to edge track. In addition to the timecode track
any high-frequency or Impulsive signals may lie. Time code is generated on ProTools.
ProTools as master filters Sync I / O Setting, generate output MTC, LTC Out. In the tape
machine, we provide the timecode on track 24th When to use ProTools only way LTC In. LTC
Synchronizer MTC
Play the whole band from the front to the rear. The first time information
begins at 59 minutes. 59: 00: 00: 00 Play and record to tape machine
(Rosendahl) as a Synchronizer
Digital systems have an internal time code. When the time code is
recorded is rewound.
From the tape machine track in the Sync I / O LTC In Pro Tools Sync I / O. From there LTC
out in the mixer in order to synchronize the console. Here ProTools must be on Slave.
Because tape fluctuations there Sync Error in ProTools. You have to change the sample
rate Timecode Chase This is a special function that varies the rhythm on the fluctuation
of the tape machine.
filters Mixers on slave. All converters running at the same clock to avoid jitter.
Word Clock clocked only the sample and hold stage
1. converter
sends
2.Wandler
these must be synchronized
Black burst video cameras
If synchronized from their converter with the Sequencer. LTC signal via XLR cables
on the deck about -7 to -3 dB loud.
Page 315 of 466
In the sequencer set certain offsets.

Each song start and end points must be set. Lead of 10 seconds to allow the sync systems. The
starting points of the different time information must be the same. In between songs always about
30 seconds leave space to meet offset. For most automated systems can not send the timecode in
the PC but the mixer.
Timecode card compatible in analog desk software system.
Timecode converts LTC etc. Automation computer in DOS system.
Automation:
With digital consoles can automate all parameters

Analogously 1. Courage 2.Faderwerte 3.Panorama 4. 5. EQ aux sends are automated.
in very expensive consoles.
automate reverb
Grass Valley, Lavo
individual values in Aux sends email to only those individual words or

Delayen reverben in music productions.
Sooner rather than Cutten has automated courage at certain points.
Analog technology for automation
- VCA technology
- engine technology
VCA fader
Spannungsregulierbarer amplifier.
Fader creates tension. This voltage controls the audio signal control voltage
-----> A / D Auto Computer D / A
VCA (audio)
Audio signal does not pass through the fader but by the VCA (Voltage Controled Amplifier)
Motorized faders
car computer D / A motor (Audio) A / D car computer audio signal goes through the
motorized faders. Motorfader
variable resistor (sounds better for low distortion)

VCA amplifier (distortion)
quality
price
resistance
accuracy
Page 316 of 466
Motorfader better
VCA
worse
expensive
cheaper
not robust
robust
inaccurate
very accurate
Problem with VCA fader is the optical control faders do

not move with.
Autocomp.
VCA
motor fade r
Both
Audio Kinetiks
Mastermix2
no
no
Opti File
Opti File
no
no
Neve Encore
Yes
no
no
Neve Nicom
Yes
no
no
Yes
no
Neve Flying Faders
AMEC Super True
Yes
automation modes
Write: (Record)
Writing absolute changes. Without even write there

is no automation.
Update: (Trim) Writing relative values changes

Relative to an absolute Write passage.
eg +5 dB at any fader position and rotate throughout the fader position to +
5dB high. (Trim from Unity)
Play: (Read) Reproducing the recorded data. No writing possible. Auto Touch and Latch
are special functions of Write AutoTouch: Once change comes, he puncht a. Bypass:
Automation data is decoded, no changes in the signal flow.

Isolate: (only with VCA Pattern)
Signal no longer runs through the VCA, but by the fader. Sounds better,
especially in the Recording)
Page 317 of 466
Isolates and bypass be confused with different manufacturers:

Red LED = Write LED
Green = Read both LEDs =
Update
Automation possible for the switched parameter.
-Write values sequentially
-Piece by piece automate.
Level changes possible Crash Record at Einpunchen (only with VCA) There are 2 special
functions prevent the crash record.
nulling:
After manual punch in / out the Automation expects crosses the originals characteristic.
Only then new values are written or is dropped out of automation recording. Only when the original
curve is taken by the automated curve, a punch in is. If the characteristic is taken again enters the
Punch Out a.
Auto nulling:
Within a set Auto nulling time old and new values are displayed. crossfader
latch
Auto nulling Time / (Return time)
Snapshot Automation
Recording of the panel states.

Save all parameters, photograph, snAppShot.
The snapshots can be integrated into the Dynamic Automation. Snapshots are used
for Total Recall. ProTools it saves automatically when the session ends.
Encore
- Neve computer to
-start Encore
- New Mix: User, client, project, Title
The steps of the Recall functions are displayed based Trees (tree diagrams), which are successively
chained sequentially. transport
system Defaults MCS Preferences Select Timecode Source Both can be
selected External LTC or
LTC from master module
Page 318 of 466
automation Manual or auto Keep Keep (recommended) Fader Lock

Records
- Car Join (Touch)

-Safety Stop (stop, until then is written)
-Record Fade Fader values are for change maintained until
the end
- Record to End
Channel Events:
Aux, faders, etc.
Recall is an independent program.

Preferences
system Defaults
Tree Automation scale steps
Automation switch with Run, is only possible with the key on the desk. As the Encore can
Direct In for panel Set (Neve preamp bypass) Tools Mic Line channel baking tools
Fader Direct specify which channels should bypass the preamp.
Recall function
Total Recall
software Recall press or on the desk the green button "Recall". Create Store Browse
where and under what name to save, then saven. Recall menus: Recall
Load Store
Recall to load it into the console.
To Recallen click the Recall window A.

Recall disclosures on the display follow,
Recall Active
Automation Write in Encore

All + Play
Play all
all Recording
All + Record
Automation Write button on the Rec. System
Control
reboot Console
Reboots the console to the automation computer
Flashing lights indicate is not in the correct position that the fader. Preferences Automation Manual
Keep
Automation delete in the fader List, Note List in order of time Stickies: Tools utilities click Text
Pages Automation step click Run Automation Write Then Stop, Keep pressing (Manual Keep)
and a new automation will be read.
Page 319 of 466
Group make:
Press Master module Group Select button select channel
1. Selected is the master and other channels, then press select. Group verify and at the end press
the Master Chanel Select.

Delete Group:
Must know which Channel Master.
select Master, then deselect the other channels and deselect the end Master Again. off Grp
old group remains, new can not be made.
Channels link (Fader Groups)

Link click Selektiere first since Master then form the other fader groups. Each fader is in the
Fader Group Master. In subgroups, however, the master fader is controllable.
Others to Automation:
11) A VCA faders ...

a) ... wear-free
12) A motorized faders ...
a) ... provides visual monitoring
b) ... has not needed nulling
13) In Mackie d8b can automate:

a) Fader
b) Mute
d) EQ
e) Gate
Page 320 of 466
E-2 technology
The capacitor
Consists of 2 metal plates / sheets where an insulation material in between. Application in
microphones and power supplies as
capacitor
U=
DC voltage source
Capacitor as long as charged to the voltage of the capacitor corresponds to the voltage of the source.
During loading high current and is after the time initially less runs.
Voltage U
increases
from a
certain time
U
Amperage I from
sinks. From a
certain time.
Capacitor stores charge and blocks the direct current after a certain time. Q = C * U
Q = charge [C] Coulomb
C=Q/U
C = constant, the charge determined C =

capacitance of the charge [F] Farrad
Page 321 of 466
Depending on how big C is, more or less charge can be stored. Mikrofarrad 10 -6 Farrad
Nanofarrad
10 High -9 Farrad
Pikofarrad
,, 10 high -12 Farrad
is continued for a time to a capacitor charged. Time

constant? dew
is determined by the components of the time constant
property? = R * C [s] in seconds When a capacitor is fully
charged t c = 5 *?
Loading or unloading
C = 5 uF R = 100 2.5
charging time
ms
t c = 5 *? = 5 * 100 * 5 * 10 high -6
U=
2.5 * 10 ^ -3
Series circuit of the capacitor
1 / Cges = 1 / C1 + 1 / C2 + .... + 1 / Cn
Parallel connection of the capacitor

Cges = C1 + C2 + ... Cn
Capacitor from the AC circuit

The higher the frequency of the voltage source is the faster and discharged When AC.
Energy is constantly pushed to the one and the other plate. reactance because energy
decreases and increases and no fixed resistor
is available.
capacitive reactance
Page 322 of 466
Capacitive reactance
Basically, two or more conductive surfaces which are insulated from one another, to be regarded as
capacitance or capacitor. This includes all cables, conductors and connectors. Two parallel cable or
conductors cause a capacitive reactance when that since the capacity is also very high impedance is
very low at low frequencies. Especially in high-frequency technology, the resulting capacity may affect
the function of a circuit. However, working in high-frequency technology with more complex models and
formulas as described herein.
Capacitor to AC voltage
A capacitive reactance is a capacitor to AC voltage. In a DC circuit, the capacitor acts like an infinite
resistance. Comparable to an interruption of the circuit, except for the brief charging current. In an AC
circuit, the capacitor can be by the current. He acts like a resistor. By constantly changing current
direction, the capacitor is charged and discharged. It is virtually always a current flowing through it (no
flow). The capacitor takes in charge of energy, stores it and releases it during the discharge again. The
energy is pushed back and forth without effect. Therefore it is also called reactive power and the
reactance resistance. In this case, it is the capacitive reactance.
Current and voltage are phase shifted to each other. The voltage lags the current by 90 . It also speaks
of the power of the voltage by 90 ahead of. The waveform is not changed by the condenser. The
reason is the charging and discharging of the capacitor. Whenever the voltage changes, a current flows.
When the AC voltage, the voltage is constantly changing. The power has always reached its peak and
the highest point when the AC voltage changes most. This is at the zero crossing. The current flow is
then ended, when the applied voltage has reached its highest point, when the peak value.
The capacitive reactance can be adjusted using the Ohm's law and the effective values of
voltage and current to calculate.
Page 323 of 466
frequency dependence
The capacitive reactance is affected by its capacity and the frequency of the applied AC voltage. The
capacitive reactance of the capacitor is the greater, the smaller the capacitance of the capacitor and the
smaller the frequency of the applied voltage. The smaller the capacitance, the faster the capacitor is
charged. The current is smaller and thus the resistance is higher. The diagram shows the course of the
capacitive reactance X C as a function of frequency. With increasing frequency, the resistance value
decreases.
Behavior of the capacitor in an AC circuit is calculated, the higher the frequency,

the lower the capacitance. The lower the frequency, the greater is the capacitive
reactance.
Page 324 of 466
The sink
Coil consists of a coiled wire. Application = dynamic microphone,
speaker, Filter induction principle:
When moving a conductor in a magnetic field, there is a tension in him. Exactly the same occurs
when the magnetic field changes.
Each electronic conductor is surrounded by a magnetic field when a current flows through it.
Mark all coils: L
Coil / coil / inductance

A classic coil is a solid body, which is wound with a wire. However, this body must not necessarily be
present. It is used mostly for stabilizing the thin wire. Coils, there are also low-rise and a rectangular
coil. Because there are coils in a variety of designs, it is also called an inductor.
The inductance is the ability of a coil in its own coils to generate a magnetic field by a voltage. It is said
that the coil induces a voltage. The trigger is the magnetic field. Unit and symbols
The symbol is the great L. The inductor L has units .omega..sub.s. The unit .omega..sub.s has the designation
H (Henry).
In electronics, mostly small coils are marked with a small inductance in mH.
Henry
1H
Milli Henry 1 mH
Micro Henry 1 .mu.H
1 H 10 0 H
0.001 H 10- 3 H
0.000001 H 10- 6 H
Nano Henry 1 nH 0.000000001 H 10- 9 H
Inductance voltage
If a coil traversed by a constantly changing stream so formed around the bobbin a constantly changing
magnetic field. Any change in current or magnetic flux produces a self-induced voltage in the coil. This
voltage is directed such that it counteracts a change. A decrease in current leads to the increase of the
voltage, which counteracts the decrease of the current.
Page 325 of 466
inductance
The influence of the coil to the self-induction voltage is specified by the self-induction coefficients. This
is called also inductance. A coil having an inductance of 1 H, if a self-induction voltage of 1 V is
produced at a uniform change in current of 1 A in a second.
The inductance L is a structural size. The self-induction voltage is greater,
the greater the inductance L is.
the greater the change in current I is.
the smaller the time t of the current change is.
switching characters
outdated
outdated with iron core
types
Air coil (coiled wire)

Iron core coil with metal core, RF core or ferrite core
Change of current flow Magnetic field changes
induced voltage
In the coil, the energy is stored in the magnetic field. Storage of the
charge in the magnetic field.
The current need for a time until the climax, it decreases the induced voltage.
Page 326 of 466
is induced voltage in the coil opposite to the voltage of the source (applied voltage) (UL opposite
U)
The ability of, coil to induce a voltage in itself is called inductance L [H]
The time constant Tau determined when the current in the coil is maximum. t L = 5 *?
[S]
Series circuit of the coil : Lges = L1 +

L2 + L3 + ... + Ln
Parallel connection of the coil : 1 / Lges = 1 / L1
+ 1 / L2 + ... + 1 / Ln.
AC circuit (coil)
Reactance (inductive reactance)
Energy is pushed back and forth just like the capacitor. Inductive reactance X L. Coil is
equally frequency-dependent as the capacitor.
Inductive reactance coil to AC
Assuming an ideal coil from, so with a wire resistance of 0 , then one speaks of an inductive
reactance.
As known from coils, they develop an induced voltage when the applied voltage, for. Example, at an AC
voltage changes. The inductive reactance caused by the self-inductance of the coil.
The alternating current in the coil builds a magnetic field on and off. The coil absorbs energy, stores it in
a magnetic field and releases it. The energy is pushed back and forth without effect. Therefore it is also
called reactive power and the reactance resistance. In this case, it is an inductive reactance.
Page 327 of 466
Current and voltage are phase shifted to each other. The current lags the voltage by 90 . It also
speaks of the voltage leads the current by 90 . The waveform is not changed by the coil. The reason
for the phase shift is the self-induction of the coil. The reason is that the self-induction voltage of the
applied voltage counteracts and the current flow is delayed. The inductive reactance can be adjusted
using the Ohm's law and the effective values calculated from the voltage and current.
frequency dependence
The inductive reactance is affected by the frequency of the AC voltage, and its inductance. The
inductive reactance is greater, the greater the inductance of the coil and the higher the frequency of the
applied AC voltage. A larger inductance greater self-induction voltage is induced. It counteracts the
applied voltage and reduce the current and thus the resistance. A higher frequency means a faster
change of voltage or current, and in turn leads to a greater self-induction voltage. The diagram shows
the progression of the inductive reactance X L as a function of frequency. With increasing frequency, the
resistance value increases. The coil has a small DC resistance and a frequency-dependent AC
resistance. The measured values in the following table reflect the back.
Current type
Frequency f Voltage U Current I
induct. Reactance X L
DC / DC 0 Hz
0.1V
8 mA R = 12.5 ohms
AC / AC 50 Hz
6.2V
8 mA 775 Ohm
AC / AC 500 Hz
51.8 V
8 mA 6.4 kOhm
(calculated)
Page 328 of 466
Inductive reactance in the AC circuit

What is the role of the inductive reactance in the AC circuit. Frequency greater X L greater
inductance less X L smaller
Effective resistance - reactance

Inductance = capacitor and coil in an AC circuit effective resistance = Modified not whether
DC or AC circuit
component
character
Resistance in the AC
circuit
resistance
Unchanged
capacitor
Xc
frequency
Constant
dependence
With increasing frequency decreasing

Falling
1 / X function
with increasing
Kitchen sink
XL
Frequency increasingly linear ascending
Page 329 of 466
Passive 1st order filter Passive:

no gain
Order: determines the slope 1st order filter
6 dB per octave
2nd order filters

Filter 3rd order
12 dB per octave
18 dB per octave
RC member and CR member
Case 1 (AC circuit) passive

low-pass filter
A low-pass can through voltages / amplitudes with low frequencies. The passive refers to the lack of
reinforcing member. The diagram shows the course of the output voltage as a function of frequency.
The circuits presented here are only the theoretical consideration. In practice, this can be used only
conditionally. There are similar conditions, such as a voltage divider with resistors.
At low frequencies, the capacitive reactance is large.
The voltage drops constantly on the capacitor.
RC element
For a sinusoidal input voltage U e has the capacitor at low frequencies a large AC resistance. The
resistor R can be neglected. The output is the full input voltage.
At high frequencies, the alternating current resistance of the capacitor is small. The input voltage
drops almost from all over the resistor R.
Frequencies below the cut-off frequency f G deemed transmitted frequencies. When the cutoff
frequency is between U e and U a a phase shift of 45 .
Page 330 of 466
low frequencies Xc is large and Ua Ue High frequencies Xc

becomes small and Ua 0 Whole voltage drops across the
resistor from.
Low pass / high cut off frequency fg = Ua 3 dB less than Ue calculation of the filter. The
phase shift between input and output signal is 45 .
The higher the order, the greater the phase shift.

Passive high-pass filter
A high-pass can be by voltages / amplitude at high frequencies. The passive is the lack of reinforcing
member. The diagram shows the course of the output voltage as a function of frequency.
The circuits presented here are only the theoretical consideration. In practice, they can be employed
only conditionally. There are similar conditions, such as a voltage divider with resistors
RC element
For a sinusoidal input voltage U e has the capacitor at low frequencies a large AC resistance. At
resistance R almost no voltage drops. At high frequencies, the alternating current resistance of the
capacitor is small and the input voltage drops from almost only through the resistor R.
Frequencies above the cutoff frequency f G deemed

transmitted frequencies. When the cutoff frequency is
between U e and U a a phase shift of 45 .
Page 331 of 466
-eludes the highpass

-low frequencies are blocked at the capacitor.
-Xc is very large for low frequencies, thus falls in the resistance are not live.
- low frequency Xc large Ua 0
- high frequency Xc small Ue Ua
RL member and LR member:

RL-member
At high frequencies the coil has a large AC resistance. This falls to her a greater voltage from:
At low frequencies, the alternating current resistance of the coil is very small. At her fall very little voltage
from:
Frequencies above the cutoff frequency f G deemed transmitted frequencies. For cut-off
frequency, the phase shift between the input and output voltage is 45 .
- low frequencies: X L very small Ua = 0 Everything strives

to 0V.
- high frequencies X L very large Ue Ua high pass:
Attenuates the bass, treble to pass through.
Cutoff frequency when RL member

Ua 3 dB below Ue phase shift = 45
.
Page 332 of 466
LR-member
At low frequencies, the alternating current resistance of the coil is very small. At her hardly fall voltage from:
At high frequencies, the alternating current resistance of the coil is particularly large. At her almost falls the
entire input voltage from:
low frequencies:
X L very small Ue Ua
high frequencies X L very large Ua 0
lowpass
Downs are allowed through, cuts the rest.
Circuits in the AC circuit

Impedance: Is the reactance of an ideal coil / capacitor charge losses really
need to be included as well.
Impedance is composed of the reactance and the effective resistance. Impedance is composed
of the reactance and resistance impedance: Z []
Impedance Z
The total resistance of a coil (with AC) is composed of the effective resistance and the
inductive reactance. It is called the impedance Z, impedance or total resistance Z.
The impedance Z can be by means of Ohm's law and the effective values of voltage
and current to calculate.
Page 333 of 466
The impedance Z is the geometrical sum of active resistance and inductive reactance.
Important:
must change variables are added geometrically.
Series connection of R and X c
Ua / Ue is equivalent to X c / Z according to the voltage divider rule ratio is equal to

the output voltage from: Vo = Vi (X c / Z) current I is the same everywhere
Series connection of R and X L
Ratio is equal to the output voltage from: Vo = Vi (X L / Z) current I is the same

everywhere
Parallel connection of R and Xc

- Everywhere is the same voltage U
- Amperage differently
I R = U / R active current flowing through resistance. I C = U / X c
reactive current
Parallel connection of R and X L

- Everywhere is the same voltage U
- Amperage differently
I R = U / R active current flowing through resistance. I L = U / X L Bindstrom
oscillating circuit
- System which autonomously continues to oscillate upon excitation.

Page 334 of 466
Application: oscillators, synthesizers, etc. Radio, filter
An antenna is an open resonant circuit.

Because of this bold assertion and consideration before ignorant readers I want to take a step back and
work your me from the resonant circuit to the antenna. Because it is about the antenna and not to the
resonant circuit here, there is only a brief introduction to the resonant circuit.
Each resonant circuit has an inductive and a capacitive component, which manifests itself as electrical
components capacitor C and coil L. In the resonant circuit, the energy travels from the capacitor to the
coil and back again. Here, in the capacitor, an electric, in the coil a magnetic field. Fields alternate
periodically.
Capacitor is charged (maximum), and outputs the energy again in the coil. This happens until the
magnetic field of the coil is rebuilt. Magnetic field of the coil is again charge from the capacitor
voltage we lower because energy swings back and forth.
Xc
XL
Page 335 of 466
L
R
Ue
Ua
low frequencies Voltage to Xc, X L tend to 0 High frequencies Voltage to Xc, X L tend
to 0 Induced voltage X L opposite to Ue Input voltage.
-Xc
Xc + (-X L) = 0
Xc - X L = 0
Xc = X L
resonance condition
Condition for resonance: X c = X L

The value of the inductor and the capacitor are equal and cancel each other out. It remains the
only effective resistance R. R plays a role. Before the resonant case, the capacitance plays a role,
according to the resonance case, the inductance plays a role. Xc = X L = fo bandpass:
It leaves a certain frequency through to fo. Locks low

and high frequencies R
Series resonant circuit

Vibration operation: charging the capacitor charged to full
Charge transfer to coil
Construction of a magnetic field in the coil to fully
loading
Impedance Z is a maximum of the smallest and the current I.
Induced voltage in the coil (U L opposite U) charge transfer on
capacitor
- Etc.
Low and high frequencies are attenuated. Frequencies to fo to pass.
low frequencies:
Xc very large
Page 336 of 466
X L 0 ShortImpedance very small
high frequencies: X c 0
X L very large
short circuit
Impedance very small
bandstop
Current is greatly hindered by fo otherwise unhindered
Resonance frequency:
Thomson oscillation formula defining the resonant frequency. resonance

condition Xc = X L
Formula for general resonance frequency: fo = 1 / (2 * *
Wuzel from (L * C))
resonant frequency
Formula applies to the resonant frequency of series and parallel resonant circuit f 1 =
lower limit frequency f 2 = upper limit frequency fo = square root of (f1 * f2) bandwidth B =
f2 - f1 Q = fo / B = fo / (f2 - f1) high quality
small bandwidth
small kindness
big bandwidth
Page 337 of 466
semiconductor
Can be used as conductors and as a non-conductor are referred. Solids, which can be considered with
respect to its electrical conductivity, both as leaders and as dielectric (insulator).
main features
Conductivity changeable. Property from outside influence. The conductivity is
temperature dependent.
The smaller the temperature, the smaller the conductivity The greater the
temperature, the greater the conductivity. Silicon, germanium material, for
example. Binding and separation of electrons,
By binding the crystal is held together. Electron bond. crystal

lattice
The electrons are connected and can not move. Above a certain temperature, the electrons
loose out of the bond, current can flow through the voltage applied when the bond is released
and all electrons are free.
Current flow possible by free electrons.
endowment :
Crystal lattice is dissolved, more extinguisher arise, more current can flow,
n-doping (negative)
-Atoms are introduced into the crystal each have one electron more. An electron is
given thereof. Increasing the conductivity of electrons.
n-conductors
p-doping (positive)
-Introducing atoms to crystals that have one less electron.
-Increasing the conductivity (Clear)
- p-type conductors
pn junction
diffusion:
The migration of electrons to the extinguishers is called diffusion. Everything happens only in the
boundary region. Tension is built up by itself in the middle of the permanent magnet. The diffusion
voltage arises by an electric field and preventing a full migration of the electrons.
Page 338 of 466
pn junction (semiconductor diode)
Adds p-type material and n-type material to be together the result is a boundary region between the
materials, the pn junction called. This area is also referred to as a boundary layer. The resulting
device is referred to as a semiconductor diode, in short diode. pn junction without external voltage
Without external influence by voltage or current, but only by the influence of thermal vibrations, the
electrons (free carriers) near the boundary region of the n-type layer in the p-type layer.
In the detailed consideration of the phosphorus atom the free electron travels across the border into the
p-layer and goes there with the aluminum atom, a bond. The phosphorus atom has lost an electron and has
become a positively charged ion. The aluminum atom has one electron more, and has become a negatively
charged ion. In the border region, many electrons move from the n-type layer in the p-layer. The electrons
from the n-layer are recombined with the holes from the p-layer. The wandering of electrons called carrier
diffusion.
By carrier diffusion ion lattice is formed. It is a depleted of free charge carriers barrier layer and is also
called the space charge zone. In this layer, there is a strong electric field that prevents other electrons
hikes. The charge carrier diffusion is terminated when the electric field is large enough to counteract the
force effect of the heat vibrations. The higher the temperature, the wider the space charge zone, the
higher the electric field. Between the space charges, creating an electrical voltage. You will diffusion
voltage U Dif called. She has at 20 C as below: Silicon U Dif = 0.6 ... 0.7 V germanium U Dif = 0.3V
pn junction to external voltage

Below the operating principle of a semiconductor diode is explained, which consists of a p-type and
n-type layer of a. Looking at this pn junction pictorially with its mode of operation in the forward direction
and in the reverse direction, so one can speak of a one-way street for electrons.
Page 339 of 466
Diode in the reverse direction
If the diode is operated in the reverse direction, as is the p-type layer on the negative pole and the n-type layer on the positive pole.
The holes in the p-layer are attracted by the negative pole, the electrons of the n-layer are attracted by the
positive pole. Thereby, the barrier layer, which is also called boundary layer increases. It can allow any
charge carriers through the barrier layer. There can be only a very small current flow through the barrier
layer. Diode in the forward direction
If the diode is operated in the forward direction, the p-type layer to the positive terminal and the n-layer is the negative
pole.
The holes in the p-type layer are repelled by the positive terminal and the electrons of the n-layer are
repelled by the negative pole. The boundary layer is now flooded with free charge carriers. Across the
pn-junction of time, a current flows through the diode. And the diffusion voltage built up by charge diffusion
is degraded.
- Connection of voltage to the pn junction (semiconductor)

-initially the diffusion voltage prevents current flow
-increasing the voltage, the electric field has problems to prevent the electrons in the transition.
- Increasing the voltage greater than U DIF
Electrons pn junction happen.
Current flows U> U DIF
Voltage that occurs when an electric field
builds.
Polarity in the reverse direction
Electric field in the center is wider, since the electrons by the compound
For positive pole walking.
Boundary layer is greater
Locks - no current flow U DIF> U
Page 340 of 466
diodes
pn - Head
Property, reduces the diffusion voltage Current flow (behaves like an insulator)
curve :
- 100V
- 50V
1V
U Dif is 0.7 V for silicon
diode
blocking polarity
Durchlasspolung
diode types
- Leutchtdioden (LED = Light Emitting Diode)
released energy is emitted as light.
In the forward direction the energy released is emitted as light.
Page 341 of 466
operation
The light emitting diode composed of an n-type semiconductor base. Then a very thin p-type
semiconductor layer with high hole density is applied. As with the normal diode, the boundary layer is
flooded with free charge carriers. The electrons recombine with the holes. In doing so the electrons their
energy in the form of a light flash free. Since the p-layer is very thin, the light can escape. Even with
small currents a light emission is perceptible. The light intensity increases in proportion to the current.
LEDs are characterized by the fact that they can shine very bright with a few milliamps of current. The
light signal is bundled or scattered by the lens-shaped form of the head.
LEDs are very sensitive to too large a forward current. Therefore, a light emitting diode must never be
connected directly to a voltage. It is necessarily a current limiting resistor in series with the LED needs.
Alternatively, at fluctuating operating voltage, supply the LED via a FET with constant current.
A current limitation recommends considerate save energy anyway. A light emitting diode is already
burning at a fraction of the maximum forward current. In addition, LEDs do not necessarily radiate at
their full brightness. Most rich only a few mA to produce a sufficient brightness.
Photodiodes:
Photodiodes are semiconductor diodes made of silicon or germanium. The pn junction of the photodiode
is made structurally very accessible to light. In incident light, the electrons are released from their crystal
bonds. In the barrier layer are electrons and holes, that generates free charge carriers. Therefore, the
photodiode is reverse biased. The free charge carriers move from the barrier layer. The reverse current
increases.
Therefore photodiodes are particularly suitable for light measurement, light barriers, positioning
and remote control with infrared (remote control).
diagram
Page 342 of 466
The free holes and electrons increase the reverse current proportional to the light intensity. Reverse current I R and
light intensity to each other linearly. The reverse current changes almost inertia in light change. With increasing
intensity of the reverse current increases. The sensitivity of the photodiode is dependent on the semiconductor
material.
-incident light induces a voltage.

-Gets out a stream
Photodiode operates on incident light as a power source.
Capacitance diode:
The capacitance of the capacitance diode is controlled by the applied voltage. If capacity is therefore
dependent on the voltage. is a non-linear dependency between the barrier Coating capacity and the
blocking voltage. The higher the reverse voltage U R becomes, the wider the depletion layer. In the
Middle by the carrier spacing is greater. The capacity is small. If the reverse voltage U R small, then
decreases the barrier layer. The carrier spacing is small. The capacity is greater.
applications
Replace variable capacitors for the oscillating circuit tuning in radios and televisions.
Circuits for generating frequency modulation circuit symbol
-The capacitance changes as a function of the voltage
Page 343 of 466
In the forward direction the capacitance diode works like a regular diode. In the reverse direction:
power augmentation Barrier layer is greater the more voltage is applied.
Zener diode:
From the breakdown-voltage diodes are conducting.
-Stable, do not break.
-Use: for voltage limitation. Zener effect
and Avalanche effect

The Zener effect is caused by the electric field that above a certain size to disentanglement of the
electrons resulting from their crystal bonds. The electrons lead to the formation of the current I z. At a
certain voltage value, Z0, is the Zener voltage, low impedance, the Z-diode. From the Zener voltage, the
current increases I z
abruptly.
The charge carriers, which were released by the Zener effect are greatly accelerated by the electric
field. This means that more electrons are ejected from their crystal bonds out. The barrier layer is
flooded with free charge carriers. This is called avalanche effect (impact ionization). The Zener voltage
can be adjusted during the production by the doping of silicon crystal in the range 2 to 600V.
In the Z-diode, the Zener effect and the avalanche effect superimposed. This condition is referred to as the Zener
breakdown. The sudden conductivity results in a very high current in the reverse direction. If the current is too large,
the Zener diode is destroyed. Therefore, in the data sheet is always a maximum permissible reverse current I Zmax specified,
which must not be exceeded. Equally important is the maximum allowable power dissipation P dead.
Both limits must not be exceeded and should be known for dimensioning of circuits with Z diode and
be taken into account. If the reverse voltage at U Z0, then the barrier layer is restored immediately. The
area between I zmin and I Zmax named workspace or the breakdown region.
temperature dependence
The temperature dependence of the Zener diode is mainly used in the measurement and control
technology of disadvantage. And applications where a precise voltage is required, making the negative
impact. Why you like on Z-diodes with positive and negative temperature coefficient TK in row. Optimally,
they cancel or it remains only a small remnant. The temperature coefficient TK indicates the temperature
dependence. Sometimes one takes for temperature stabilization and normal silicon diodes. In special
temperature-compensated zener diodes, the manufacturer has already made this interconnection.
Page 344 of 466
RF diode:
-designed diode for the RF range
- switches very quickly
typically up to 1 MHz
Capacity = C [F] Farrad

inductance = L [H] Henry charge
Q = [C] Coulomb
- Optocouplers:
Consists of light transmitter (infrared LED) and the light receiver optical coupler / optical coupler The
optical coupler consists of a light transmitter and a light receiver. As light emitter diodes are used
which emit infrared light or red light.
As light receiver photodiodes, phototransistors, Fotothyristoren, Fototriacs, Photo-Schmitt trigger
and Fotodarlingtontransistoren be used.
The wiring diagram is constructed simply with a light-emitting diode and a photodiode. The
optocoupler there are normally in IC design (DIL) with 4, 6 or 8 legs. Sometimes they are found also in
transistor design.
Page 345 of 466
applications
Optocouplers are always used when circuit components isolated from each other (electrically isolated)
must be. For example, if subsequent circuits may have no effect on previous circuits, or where different
ground covers must be used.
The optocoupler even allows voltage differences up to several 1000 V between input and output.
The optocoupler CNY17F has an isolation voltage of 5300 V AC. For electro-medical equipment
which is a condition!
The transformer:
AC voltages are increased or decreased. Transforms high AC voltage or down (increase or decrease)
- consists of 2 coils On the input side primary coil and the secondary coil at the output. The two coils are
located in the magnetic field (iron core) designs Toroidal - transformer
Rectangular iron core transformer
With a transformer AC voltages up or stepped down. Thus, increased or decreased. However, this
change of voltage leads to a change in the maximum removable current at the output (secondary side)
of the transformer.
If the voltage is stepped down, the removable to current increases. If the voltage is transformed up, the
removable to current decreases. The relationship between voltage and current is inversely proportional
to each other.
The transformer, transformer short, acting on the input, the primary side, as a consumer R for its AC power
source, if the transformer is loaded with nominal load. Unencumbered affects the transformer as an inductor.
The output side, the secondary side, acts as an AC voltage source with source voltage U 0 and internal
resistance R i.
The transformer consists in principle of two adjacent coils, having the same or different number of
windings. On the input winding of a changing magnetic field is generated by the applied AC voltage.
On the output winding, an induction voltage is generated. The magnitude of this voltage depends on
the turns ratio of the primary and secondary side of the transformer.
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Primary side secondary side
Ne Ue
NA
600 50 V 600
UA
50 V
600 50 V 1200 100 V 600 V 50 300

25 V
N = number of turns
Is the number of windings equal on both sides, then the input voltage U e
and the output voltage U A same size. Excluding the losses by the efficiency. One calls this
transformer also isolating transformer. He is only two circuits for safety reasons separate.
Isolation transformers are used for the electrical isolation of the AC voltage from the mains. Exchangers are used for data
transmission and in the measurement and control technology for audio-frequency transmission.
Is the number of windings on the primary side is greater than on the secondary side, the output
voltage is less than the input voltage. Is the number of windings on the secondary side is larger than
on the primary side, then the output voltage is greater than the input voltage. Ratio of voltage and
current
A larger voltage at the output leads to a smaller current at the input. A smaller voltage at the output
enables greater current drain.
Secondary side with a winding
Secondary with two windings
Page 347 of 466
Toroidal Transformers
Ring core transformers consisting of an annular iron core around which the primary and secondary
coils are wound. Toroidal transformers are lightweight, require little space, are more efficient and
have a lower magnetic field radiation. This gives them significant advantages over rectangular
transformers. Toroidal transformer have high turn-on.
Rectangular iron core transformers
Rectangular transformers are widely used. Especially in power supplies and integrated power supplies.
There, the current consumption is not too high. can on a power supply or must even be dispensed with.
The weight of the iron core makes often unpleasantly noticeable and makes a substantial part of the
weight of an electronic device from.
One can assume that the iron core brings 10% energy loss in the transformation. In order to
compensate for the easy 10% more turns wound. This provides you the desired voltage ratio safely.
Transformation of voltages. Currents, impedances, galvanic isolation in the sound equipment is a

transformer as a transformer used for example in mixer input Mic / Line In signal.
Inputs and outputs of transformers to each other at a certain ratio

ratio
Ideally transformer without resistive and inductive charging losses n
= number of turns of the coil ratio: Ns = Np * (Ua / Ua), etc.
Performance Grade: Pe = Pa P = U * I
Ue * Ie = Ua * Ia
Ia / Ie = Ue / Ua is equal to Ia / Ie = Ne / Na
Currents at the transformer are inversely proportional to the tension and turns.
Page 348 of 466
power Supplies
Provides electrical energy equipment or assemblies provided that a different voltage
need or other currents, is thus provided from the power supply.
Ue (230 V)
Voltage conversion in the transformer from 230 V to 9 V)
rectification (AC to DC) smoothing (ripple is taken out) voltage

stabilization (stable against external influences) Rectification:
Wave rectification: (1-pulse midpoint circuit) the forward

direction is transmitted voltage in the reverse direction voltage
is blocked.
Disadvantage: Half the voltage is lost
Bridge rectification:
(Variant: 2 Plus midpoint circuit)
Current path for positive and negative half-cycle is taken into account. Advantage: in
addition, the negative half-wave is utilized
higher output voltage
However, no complete direct voltage takes place, it must be smoothed.
Smoothing :
poled Elektrolythkondensatoren
Voltage rise:
-Elko is charged
-Capacitor
Capacitor emits stored charge Smoothing the output voltage, the larger the capacitor, the
better the smoothing effect.
screening:
- RC elements
Filter ripple
take out ripple voltage,
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Stabilizer:
Voltage regulation: ensures stability throughout the circuit.
Stabilize the circuit against power fluctuations against load
fluctuations to temperature fluctuations
Stabilization by means of Z-Diodes
Voltage regulators.
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transistor
Active semiconductor device is that used for switching and amplifying voltages and currents.
(Transformation Resistor)
adjustable resistor
Consists of two semiconductor crystals consisting of 2 pn ladders Bipolar
Transistors: controlled by current flow.
-work with two carriers (electrons and holes

Unipolar transistors: is controlled by voltage
Function on a carrier
Bipolar transistors consist of

NPN transistor
PNP transistor
Normal transistors have a NPN or PNP layer sequence and are called bipolar transistors. Bipolar
transistors are made of silicon. They also exist in germanium (deprecated) or of mixed crystals, which
are not very often spread. Each bipolar transistor consists of three semiconductor thin layers, which are
superimposed. the middle layer is very thin compared to the other two layers. The semiconductor layers
are provided with metallic connections, which lead out of the housing. The outer layers of the bipolar
transistor are called collector (C) and emitter (E). The middle layer has the designation base (B) and the
control electrode or the control input of the transistor.
NPN transistor
PNP transistor
The NPN transistor is composed of two

n-conducting layers. In between lies a thin p-type
layer.
The PNP transistor is composed of two

p-conducting layers. In between is a thin n-type
layer.
The circuit symbol with the two against each other diodes will be happy
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used to the basic structure of the transistor display. The operation of a transistor can be adjusted so in
reality is not. The reason lies in the changed behavior due to the very thin middle layer of the
transistor.
Voltage and current distribution
This circuit is intended to represent each other, only the current and voltage waveforms and
their relationship. Basically should in I B- and I C- Circuit, a current limiting resistor to be used.
Please note: Here the direction of current from plus to minus true. When PNP transistor, the polarity
of the voltage and current distribution is the other way around. When used it must be only on the
polarity of the operating voltage. NPN transistors are used for positive voltages. PNP transistors are
used for negative voltages. U CE = Collector-emitter voltage U BE = Base-emitter voltage (threshold value) I C
=
Collector current I B = base current
Operation of a transistor (NPN)

In the operation of the transistor is necessary to observe the current direction. Will you explain the
physical principle, then one speaks of the electron current or the physical flow direction (from minus to
plus). It is used in the following embodiment. In circuits and mathematical calculations, the direction of
current is used (from plus to minus). By applying a voltage U BE 0.7 V, the lower diode (principle) is
connected in the forward direction. The electrons reach the p-type layer and of the positive pole of the
voltage U BE attracted.
Since the p-layer is very small, only a small portion of the electron is attracted. Most of the electron
continues to move to the upper boundary layer. This latter is conductive and the positive pole of the voltage
U CE attracts the electrons. It flows a collector current I C.
In conventional transistors slip about 99% of the electrons from the emitter to the collector
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by. In the base layer about 1% of the electrons get stuck and flow from there.
Properties of the NPN bipolar transistor

1. The collector current I C flows only when a base current I B flows. If the
Base current I B changed, then the collector current changes I C. Within the transistor, the base
current change acts as a resistance change. The transistor acts at a base current change as
an electrically controlled resistor.
. 2 The collector current I C is a multiples of 20 to 10,000 times greater than the
Base current I B. This size difference comes from the splitting of the electron flow from the
collector (C) and base (B). This difference in size is called current gain B. It is calculated
from the ratio I C to I B
to calculate.
. 3 The base current I B flows only when the threshold voltage U BE at the baseEmitter path is reached. The threshold depends on the semiconductor material. Usually takes
one silicon transistors, with a threshold value of 0.6 to 0.7 V. It is also germanium transistors
with a threshold value of 0.3 V. By means of an auxiliary voltage U BE can set the threshold value
in advance. This procedure is called the working point. To this set voltage can now control the
collector current of the base current.
. 4 If no base current I B flows, then the transistor blocks. Its resistance in the
Collector-emitter path is infinitely large. The voltage at the collector-emitter is very large. A
base current flows, the transistor becomes conductive. Its resistance has become smaller.
Thus the voltage at the collector-emitter is smaller. More specifically, an increase in the input
(base) in a decrease in the output (collector-emitter). This is called also inverting behavior.
This characteristic is the switching behavior of the bipolar transistor and is applied very widely
used in electronics (transistor as a switch).
. 5 When the voltage U CE is smaller than the voltage U BE, then is the
bipolar transistor in saturation or in saturation mode. This happens when the transistor is
flooded by the base current. The base current is then so large that the maximum current gain is
already reached and the collector current no longer increases.
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In general this has no adverse effects, as long as the maximum base current is not exceeded.
Here, the transistor is destroyed. However, the saturation operation has a negative effect on the
switching behavior of a transistor. In the case of a fast switching operation, if the
collector-emitter voltage U CE must change quickly. Then, the transistor has to be cleared free of
the charge carrier flooding. This will take longer than if only a few charge carriers flow through
the base. This delay has a negative effect at high switching frequencies. Then, the saturation
operation should be avoided.
. 6 The bipolar transistor combines two circuits in itself. The circuit with the
voltage U BE is referred to as control circuit. The circuit with the voltage U CE is referred to
as a work or load circuit.
- many electrons penetrate through the boundary layer but can not continue (p-layer)
- Accumulation of many electrons in the second barrier layers (p layer)
- The application of the large voltage at the collector Uce causes drag over the electrons towards
the collector,
Follow flowing current Ic
Collector current is substantially greater than the base current Ic> I B
A small base current, a large collector current is generated. Depending on the

base current.
Because Uce the electrons are pulled over by the second boundary layer. But before they
have to come boundary layer U B ( I B).
I B Control Circuit Ic Load circuit Uce Voltage between the

collector and emitter of the transistor of the control circuit is
controlled. Context :
Currents: I e = I B + Ic voltages Uc e = U BE
U BC
The basic equation for voltage and currents at transistor

Characteristics:
Input characteristic Io
= f (U BE)
Base current in response to input voltage Differential input

resistance: R BE = U BE / I B
Specifies us like a transistor at a certain Einganswiderstand (modified input signal)

behaves the base current.
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Current control characteristics: Ic

= f (Io)
Collector current Ic as a function of base current Io
Gives us on what the load current happens if we change the curve. The curve gives us the
certain input current to the output stream. Short-circuit current gain?
= Ic / I B
Output characteristics: Ic =
f (Uc e)
Load current Ic as a function of output voltage Uc e

Initially, the electrons are made available for the collector current. At the output levels off and
the number of streams remains constant. Differential output resistance r CE = U CE / I C
Total power dissipation Ptot
emitter circuit common electron for A and output new component is Rc.
Purpose of Rc to limit current of collector.
Rc Limit load current Ic
If there was no resistance, the complete power would drop across the transistor and break.
Operating point setting
Working is the reason and calm state of the transistor with no input signal. Awareness is the
middle Deadlines
Selection of the operating point possible undistorted signal transmission.
Operating normally in the center of the characteristic. Ie flows between maximum and minimum
voltage and currents base current when U BE> U DIF
I B flows only if U BE> U DIF

Not the complete signal is transmitted, we get distortion. Target: the complete signal on the diffusion
voltage transfer. Only then can the transistor through the complete signal. 1 possibility of working point
setting: With the help of the operating voltage U B an additional current Iv is fed to the base via Rv. I B is
the input current. The overall level of the input signals is raised, displaced upward.
Problem working point fluctuations.

2. Possibility of working point setting
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Working point setting by voltage divider

By decreasing current through R1 and R2 we continued our high input voltage. Voltage divider R1 and
R2 ensures offset voltage. Advantage: -przisere operating
-Filtering out "noise voltage"

- R2 is connected to ground Problem:
Strong temperature dependence
Operating point stabilization
With a temperature-dependent resistance NTC (Negative Temperature Coefficient) Instead R2 is a

NTC.
Function: with increasing temperature the resistance becomes smaller. temperature
increases
R2 is smaller
I B is small (it drops less voltage)
sinks
Power is small
temperature
Transistor gets cooler I B is
control current
Problem signal is still a Geichspannungs Offset By Boxing out it generates a buzz why coupling
capacitor capacitor
to interference voltage and DC filter out from the input signal
exit
filtered Offset DC out

DC Moderate Abkappung of switched before or after assembly. There is a phase shift
between input and output by 180 degrees.
Darlington circuit
Component, which consists of two bipolar transistors combined in a single housing. Advantage:
Substantially greater current gain than single transistor Single transistor has current amplification
factor of the currents I B = C / I B
Total gain: B tot = B 1 * B 2
Page 356 of 466
amplification circuit
Class A amplifier:
Advantage: characteristic of a Class A amplifier is highly linear
almost distortion-free linear transmission THD + N performance disadvantage: -stndiger
poor efficiency
- Amplifier becomes warm by constantly high power, about 50% of energy is lost in heat
energy.
Efficiency: ratio of power output to power input. Efficiency = P output / P Up
high quiescent current, not very effective
Class A amplifier is a push-pull amplifier called because only one transistor in the circuit is present.
Class B amplifier push-pull amplifier
because 2 transistors assume the power transfer. 2 transistors are connected in parallel
in the signal path and there is a resistance to. After the resistor the signal U A output.
function
Case 1: positive half wave:
Go to NPN and PNP output and passes no current. Transistor 1 amplifies

power transistor 2 blocks current flow
Case 2: negative half wave
Go to PNP transistor and is output amplified. NPN transistor no

current. Transistor 2 amplifies power transistor 1 blocks current
flow.
Each of the transistors takes over a half-cycle of oscillation positive and

negative
Input voltage remains linear and ends with operating voltage on the positive and negative half-wave.
Benefits Class B amplifier;
higher efficiency, no quiescent current, because the non-active transistor always locked.
-more output power with less loss of heat energy

Page 357 of 466
- is not so warm
disadvantages Class B amplifier:
- Crossover distortion greatly because transistor constantly the performance takes from others.
- larger circuit scale applications Power amplifiers, devices where low power
is available.
Class AB amplifiers
Requirement: combining the advantages of the first two circuits Class A low
harmonic distortion and better sound Class B performance. Improved efficiency
Realizable by DC sources:
Crossover distortion is prevented.
If you pass the positive to the negative half-cycle, the base is held in DC without it falls back to 0
V and will rise again from 0.7 V.
Quiescent current holding transistor always
open. Hardly audible transient distortions
Advantages: -low crossover distortion drawback:
slightly impairs
0.7V is mostly about silicon. because only from 0.7 V voltage silicon is
conductive.
Summary :
Class A. high power consumption, low harmonic distortion
Class B complex, lower power consumption, but crossover distortion Class AB Compromise
Characteristics of amplifiers
Power:
Power Rating: (RMS power "root mean square") Maximum permissible
continuous output power (pink noise)
RMS power:
Maximum continuous power (measured with a sine signal) Obsolete was canceled
music performance : Briefly achievable maximum power, 1kHz sine wave,

(outdated)
Page 358 of 466
rated capacity RMS = (Umax / (square root of 2)) (Watt)

Frequency response:
Frequency response of the amplifier
Delivery via the amplifier influence on the frequency response of the signal.
SNR:
(Signal to noise ratio, signal to noise ratio)
Measure the quality of a useful signal. Distance from noise to useful signal. Preamps
very important
snr (in dB) = 10 log (useful signal / noise)
Typical values go off at 90 dB. The higher the
quality of the amplifier.
THD :
Measure of the strength of the harmonics that occur in non-linear distortion, relative to the total signal.
Even-numbered harmonics: harmonic distortion
Odd distortions are discordant distortions unpleasant
for the ear pleasant
Clipping (distortion)
soft clipping Tips are rounded in transistor technology as with tube compression.
operational amplifier
(OPV, OP) Operational Amplifier (OPAMP)
Function: one of the main amplifier in electrical engineering
Diagram: 2 inputs, 1 output, +/- characterize properties of the inputs

+
-
non-inverting input
inverting input
Symmetric power supply of + 5V and -5V

Typical values for the supply voltage of +/- 5 V +/- 12 +/- 15 V V
Advantages over the transistor
-OP has no built-in voltage / threshold voltage as the other amplifiers such as 0.7 V.
-lower noise
Page 359 of 466
Function (ideal OP)

Case 1: voltage is greater at non inverting input.
Output voltage value to Vcc + (operating voltage)
Case 2: voltage at the inverting input is greater
Output voltage value goes to Vcc
1st case
2nd case
negative operating voltage
positive voltage difference

negative voltage difference U d is the voltage
difference at the inputs U d = U + - U U a = V * U d = V * (U + U) V = amplification Dimension Size
a factor
Basic circuit of the OP:
comparator:
It compares the input voltage to the ground potential function:
- Ue positive Ua Vcc +
-Ue Ua negative Vcc
Measuring instrument to distinguish whether signal is positive or negative.
Applications such as A / D converters
Last bit is set if signal is positive and not when it is negative. "Sign bit". Dissolve with 15
bit and the sign bit is determined whether positive or negative voltage.
Page 360 of 466
inverting amplifier
The inverting amplifier is operated with parallel-to-voltage feedback. For this purpose a part of the output
voltage across the resistor R 2 to the negative input () the OP returned. The input voltage U e overlying the resistor R 1 at the negative input of the OP on.
The non-inverting input (+) is connected directly or via a resistor to ground.
By inverting operation, the output voltage is at a positive input voltage as far into the
negative, so that the point S is always close to the zero potential (0 V).
The point S is called a virtual zero. It is based on the ground potential to approximately
zero. Gain factor v U
The voltage gain V U only depends on the external circuitry of OP! The inverting amplifier circuit
reverses the sign of the input voltage. From + U e we you a or from -U e is U + a, multiplied by the two
negative feedback resistors. The following applies:
Input resistance r e
The input resistance r e of the inverting amplifier by the resistor R 1
certainly. He stressed the source.
Page 361 of 466
Output resistance r a
The output resistance r a of the inverting amplifier is very small. The circuit acts as a voltage source.
applications
A deficiency of this amplifier is the relatively low input resistance. It can with the resistor R 1 be
determined. At high gain, the resistance has R 2
have an excessively high value. However, since a gain factor V U of 1 is possible, the inverting amplifier
can be used as a filter circuit and analog operational amplifiers.
1
RRV
2
- 1 ()
VUU
Non-inverting amplifier
1 ;
VUURRV
This circuit of the non-inverting amplifier has a

series voltage negative feedback.
In the non-inverting amplifier, the input signal to the
output signal is in phase.
The non-inverting amplifier is used for
used applications that require a very large input impedance and very low output impedance.
The circuit is suitable as an impedance converter, AC amplifier and a high-impedance voltmeter for
small DC voltages. Due to the low output resistance it is also suitable as a DC voltage source. In the
non-inverting amplifier of the non-inverting input (+) is connected to the input signal and the output to
the inverting input (-) coupled back (feedback). In the negative feedback, the output voltage change
of the input voltage change will help prevent. The voltage U PN is therefore very small.
Gain factor v u
Without additional measures, the voltage gain is V U greater than or equal to 1!
Page 362 of 466
Input resistance r e
The input resistance r e the non-inverting amplifier is a very high impedance (10 M ....). Almost
infinite.
Output resistance r a
The output resistance r a the non-inverting amplifier is very low. The circuit acts as a voltage source.
Use as impedance transformer

If you couple all the output voltage to the input (the R 2 = 0 , R 1 =
infinite), then the circuit operates with V U = 1 as an impedance converter (resistance)
converter.
The input impedance is almost infinite. And the output resistance is about
0th
Impedance transformer / voltage follower
-based on the non-inverting amplifier, but = -R1 0

R2
Impedance converter allows the signal unimpeded through (neither amplification nor attenuation)
properties infinitely large input resistance. Output resistor to 0V.
At the entrance, a high resistance is present.
Adder or inverting summing

Vi1 ... 3: input Re1 - 3 mixer,
fader
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Rn = masterfader Ua =
total output
Problem = If Re1 is affected, the other Re2 and Re3 with affected. Virtual mass stabilizes the circuit. If
Re1 is greater, Re2 and Re3 be smaller. Advantage of the virtual ground.
Stabilized interconnection of multiple inputs.
Simplified structure of a mixer Ua = - (Vi1 +

Vi2 + Ue3) Re1 = Re2 = Re3 = Rn
The summing amplifier is a special application of the inverting amplifier. It also speaks of the adder or
the Umkehraddierer.
Each of the input voltages supplies a current share, which came together in the virtual zero point
S and the resistor R K generate a voltage drop.
special case
Are the input resistances equal to the resistance R K, so the input voltages are added. The summing
amplifier provides an output voltage U a,
the sum of the input voltages equal. Because of the basic circuit of the inverting amplifier, the
output voltage has a negative sign.
Differential amplifier / subtracter
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In the differential amplifier or subtractor of the operational amplifier is connected at both inputs
with signals.
If all resistors are equal, then the circuit at the output is the difference between the two input
signals. That is, the differential amplifier subtracts the two signals from each other.
The inputs of the arithmetic circuit burden the signal sources. This creates miscalculation. To counter
the output resistances of the sources must be low. Are the sources negative feedback operational
amplifier circuits, then this condition is likely to be met. If it is high-impedance signal sources are
impedance converter to switch before the inputs. e 1 to ground: Non-inverting amplifier
e 2 grounded: Inverting Amplifier
Both inputs used (see circuit)
Circuit dimensioning With R 1 = R 3 and

R2= R.4
and no gain at R 1 = R 2 = R 3 = R . 4
Increases the difference between the inputs
Differential amplifier is used to make an unbalanced signal from a Symmetrical signal.
Page 365 of 466
DI boxes eg
Active filters
Active: potential gain

Advantages: Waiver of inductance (coils are expensive and prone to interference)
Higher slope better to realize than with passive filters. Reinforcing V may be
greater than 1. Easy cascadable
Disadvantage:
Requires power supply
Low pass (integrator)
- Frequency-dependent components are installed.

- these are mostly in the negative feedback support: V =
(-R2 / R1) R2 replaced by C.
Xc = 1 / (2 * * f * C)
Viewing with increasing frequency:

Xc is smaller
Attenuation of high frequencies lowpass
Highpass (differentiator)
Capacitor and resistor are reversed.

: V = (-R2 / R1) R2 replaced by C. Xc = 1 / (2 *
* f * C)
Xc becomes small
R1 is ersetzt- by C
V is greater
High frequencies pass through, low locked highpass
The result is a 2nd order filter when you combine 2 of these circuits together.
Page 366 of 466
Balancing and symmetry from

Advantage of symmetric signals: less interference
Balancing on DI Box Direct Injection.

2. Possibility of balancing:
Passive: -teuer, because good transformers (transformers)
Susceptible to magnetic interference
Not negligible distortion (especially in older instruments) No amplification, no power

supply required. Galvanic isolation
Have symmetrize and desymmetrieren active in both directions: OPERATION
only one direction possible
- Affordable, no expensive transformer
+ Signal can be directly amplified.
Others to E-Technik2 : P ARALLELSCHALTUNG the capacitor X C a capacitive reactance:
XRZ
=
XRCC
11) In an LC-S IEBSCHALTUNG (- 18dB) is a high-pass as TS CHALTUNG implemented

1) T RANSFORMATO r. Power conservation (!):
12212
=12
IUUIUIUPP
1
2) disadvantages of 1-P ULS S CHALTUNG compared to a 2-P ULS S CHALTUNG

Half of the power
Greater smoothing capacitor
3) An NPN T RANSISTOR is a BIPOLAR T RANSISTOR

5) You need one F REILAUFDIODE, when a transistor relay
want to turn
Page 367 of 466
6) capacity of a K APAZITTSDIODE decreases with increasing blocking voltage
4) The two inputs of OP-A MPS ring Inverting & Non-Inverting

7) N SE inverting V ERSTRKER:
1 ;
VUURRV
8th)
inverting V ERSTRKER:
RRV
2
- 1 ()
VUU
9) C LET- B amplifier ...

produce no crossover distortion
both transistors to be permeable shortly before before
do not need coupling capacitors
Page 368 of 466
speaker
speaker
Voltage brings membrane to vibrate

electrodynamic loudspeaker
Converter principles in speakers
Electrostatic:
membrane
counter
electrode
Electrode
area S
C = L*C*S / s
R
Electrode
spacing s
Electrostats be used as a mid-range and tweeter in the HiFi range. Advantage: the
membrane vibrates evenly back and forth drawback: susceptible to moisture.
A membrane of goldbedmpfter plastic film together with a counter electrode a variable

capacitor.
This is with a DC voltage of 1000 - 5000 V charged. The DC voltage is superimposed on
the useful signal. By changing electronic approaches, the membrane vibrates.
Page 369 of 466
Advantages: uniform diaphragm actuator, good transient response. Disadvantage:
moisture sensitive
Midrange and treble in the HiFi range.
Voltage of a Transfer of 230 V stepped up to 1000V 5000V
Commitment:
An electromagnetic transducer
Power line field
air gap
N
Effective area S in
the air gap
winding
In front of a permanent magnet which is wound around a spool, bringing a soft iron armature which is
coupled to a membrane.
At the coil, the useful signal is applied, thereby the strength of the magnetic field changes which acts on the
magnet. Advantage: high efficiency
Efficiency o = (Pak / pel) * 100%
Ratio of how much the electric power is supplied as an acoustic performance. Disadvantage:
poor sound use = miniature speaker
magnetostrictive transducers
Used for underwater sound.
Page 370 of 466
Ferro Magne Tischer body changes in the magnetic field extending them. It wraps a coil at the
desired signal is applied to an iron body.
Ion plasma converter
membrane Dimensions
The larger the mass, the worse the transient response, the smaller the
membrane, the better the pulse response.
By two electrodes, an arc is generated, which replaces a membrane. No mass movement

(membrane mass) uses air movement of the useful signal. Very good transient response.
Two electrodes a high frequency voltage is applied.

It burns an arc. The high-frequency voltage is superimposed on the useful signal. This changes the
temperature of the arc, so that the extension of the ambient air. The two electrodes are located in
an air chamber, the radiation
If by a horn. Advantage: very good
transient response
Disadvantage: low Maximum sound pressure is released ozone.
Piezzoelektrischer converter
Instead tap off by deforming a voltage can be deformed by tension a Piezzokristall and stimulate
the diaphragm so. Capacitive: low frequency, high resistance
High frequencies, low resistance speaker
type without crossover.
When a voltage is applied to a piezoelectric crystal, it deforms. The Piezzokristall is coupled with a
membrane through which the sound emission takes place.
It is suitable only as a tweeter because its resistance behaves like a capacitor.
can be connected without crossover.
An electrodynamic transducer:
1. A ribbon transducer:
If used only as tweeters, because a small diaphragm displacement can take place.
Distance between north and south pole of the magnet is too large. Weak
magnetic field in the ribbon.
An aluminum ribbon is stretched between the two poles of a permanent magnet.
If you put on the ribbon to the desired signal, so an alternating magnetic field is built up around the
ribbon around and it affects a changing force to the ribbon.
Page 371 of 466
Use: Tweeter
Advantage = Very good pulse response
drawback: Poor efficiency
2. Isodynamischer converter
On a plastic membrane, a loop-shaped conductor strip is placed attenuates This is located between the
two perforated side of a permanent magnet. advantage
good efficiency
disadvantage small membrane deflection use:
treble
3. moving coil transducers:
Coil is located in the spiral of the permanent magnet. Coil is centered

attached to the back of the membrane.
A voice coil which is directed to a diaphragm, is located in the annular air gap of a permanent
magnet. Electrodynamic Cone loudspeaker
spherical surface tweeter
An electrodynamic cone loudspeaker
1. Drive system (converter):

From the magnet rear front pole voice coil
2. membrane Cone shaped (elektrodynmaisch) with dust cap

3. membrane suspension Outside suspension
Bead (external mount) spider

(always suspension) centering of the coil in the
air gap.
Provides together with the bead, the restoring force
low restoring force
high compliance and vice versa
Drive system (converter):
Magnet with rear and front pole plate, voice coil material: Alnico,
Neodymium (better efficiency)
Efficiency: from 0.3 to 2% less than 1% is bad
Transformer constant BL
Decisive for the force action is B * LB = magnetic flux density (from

the permanent magnet)
Page 372 of 466
L = length of conductor (coil) variable factor

Strength of the desired signal which I create.
Interaction of the current with the transformer constant B * L causes the transmitter Put the
electrodynamic acoustic transducer; F = B * L * I Force acting on the coil. F force
B magnetic Fludiche L
conductor length l
Stroomstrke
The number of turns in the air gap has to stay constant by the diaphragm moves with the
movement of the coil.
Underhung coil
The depth of the air gap is greater than the length of the coil. disadvantage Efficiency
weaker, shorter coil lower resistance. advantage
linear frequency response
Overhang coil
The depth of the air gap is less al the length of the coil.
Number of turns is constant. If the number of turns decreases, the converter constant is small.
Membrane deflects off no longer good. disadvantage

poor frequency response
advantage
better efficiency
Beading
voice coil
developable cone
Membranffnungsangle
Membrane must be fully up as evenly as possible and move back.

Page 373 of 466
What factors are dependent on the constant back and forth motion
-
membrane stability
membrane thickness
Density of the membrane material
Diaphragm aperture angle
Avoid flat membrane as 180 degrees is rather unstable.

membrane curvature
Not developable membrane
stable membrane by curvature of the bead.
Diaphragm diameter
The larger the diaphragm, the more unstable it is.
Frequency response:
Depending on the radiation impedance of the air
(Resistance which opposes the air the membrane movement.)
Reactive component: resistance to air movement
Active component: resistance to the sound radiation Zr
Zr increases to f, in the lambda = diaphragm diameter (Zr proportional f ^ 2) Zr remains constant

above the f, when lambda = diaphragm diameter (Zr proportional to 1
depending on the membrane Fast

below the resonance frequency fs of the loudspeaker diaphragm takes the velocity v increases with increasing
frequency. v proportional f Above fs takes v with increasing f from v proportional to 1 / f
Dependence of the sound performance of an electrodynamic cone loudspeaker of Zr and v Pak proportional to Zr *
^2
Page 374 of 466
Frequency range
f <fs (low f)
P ak proportional to Zr * v ^ 2 it is
Zr proportional to f ^ 2 v
proportional to f
Pak proportional to f ^ 2 * f ^ 2 = f ^ 4
- 12dB per octave at low frequencies
fs <f <f at lambda = membrane permeation (mean f) Pak

proportional to Zr * v ^ 2 it is
Zr proportional to f v ^ 2
proportional to 1 / f
Pak proportional to f ^ 2 * ^ 1 / f ^ 2 pak
proportional to 1
Performance remains about constant frequency f at lambda =

diaphragm diameter <f (high f) Pak proportional to Zr * v ^ 2 it is
Zr is proportional to 1 V proportional to
1 / f Pak proportional to1 * 1 / f ^ 2 Pak
proportional to 1 / f ^ 2
- 6 dB per octave at higher frequencies,
The transmission range is from f to f at = diaphragm diameter
Problems with the sound radiation of a loudspeaker (moving coil loudspeaker)
Concentric modes circular natural oscillations problem at high

FrequenzenAt the upper end of the transmission range, the vibration can be instable membrane circular spread
on the membrane surface. This wave is partly reflected at the bead. Has nothing to do with
overloading of the speaker.
rather
unstable membrane carrier membrane Broadcast with a time
delay.
Doppler distortion:
Arises when a speaker has to radiate to large frequencies. High + depth together
for example, 100 Hz and 10 kHz, are one period of oscillation of 100 Hz in the same time he needs 50
vibration passages for the other frequency of 10 kHz.
Page 375 of 466
Vertical displacement of 10 kHz tone due to the simultaneous emission of the 100Hz tone.
Beam, a speaker at the same time a deep and High tone from, so it is important for the high tone to
flutter.
Better distribute the frequency responses on different speakers.
THD:
Membrane may no longer be broadcast. Superpositions of overtones produce flattened when
overloaded. hum Additional bertne when overloaded and difference tones arise if I have to play
multiple sounds simultaneously.
Intermodulation distortion
Additional sum and difference tones when overloaded.
spherical surface tweeter
Drive system: moving coil transducer sound

radiation through dust protection
Use: midrange in diameter. 37mm or 50mm

Tweeter with diameter 10mm or 25mm
Horn Speakers
In front of a spherical surface tweeter is mounted a pressure chamber whose opening is smaller than the
membrane area. This leads to an increase of the sound pressure. For directional radiation a horn is placed in
front of the pressure chamber.
Thiele Small Parameter

Resonance frequency f s is an unincorporated Chassi (speakers without housing) Total Q TS Aquivalenzvolumen
Vas
Impedance maximum at the resonance frequency of the loudspeaker.
Caused by the mutual induction. We have not only the applied voltage but also a coil which is induced
in the coil. So the offset voltage produces a resistance.
Largest Movement in the coil
the highest reverse voltage at fs.
At the resonant frequency of a moving coil loudspeaker has a maximum impedance. This is caused by
mutual induction.
you have to make only a resistance measurement to determine fs.
Page 376 of 466
Resonance must be damped. The overall quality Q TS is an indication of the resonance
Q TS
damping.
Statement read numeric value without unit.
What is the average sound pressure compared to the sound pressure in the transfer area.
Q TC = Overall quality of the speaker in a built-in box, the effect on the

quality and the resonance frequency. Q TC = 0.71 optimum frequency
response Q TC> 0.71 exaggeration of the frequency response of Q TC < 0.71
A flattening of the transfer curve
Better transient response if something smaller value of Q TC
Aquivalenzvolumen Vas
Air volume has the same resilience as does the cone suspension. Indication in liters
Small Vas
Large Vas
little flexibility
great flexibility
Acoustic short circuit

Sound radiation is the front anti-phase sound emission backwards
Tuscaloosa is the extinction
Acoustic short circuit at long wavelengths with membrane diameter of level drop of 3dB
If depths phase opposition cancel each

wavelengths of 60 cm 583 Hz including 6dB per
octave attenuation.
Baffle:
Critical baffle diameter = gives -3dB level drop Therefore baffle has a length of 3.8 m
have to get around a clean transmission range.
Closed housing
-Problem: cavity resonances (standing waves)
Page 377 of 466
-Solution: change geometry.

No parallels pit walls, absorbing material of the box. Cabinet resonances have to be
attenuated Battens, struts High density of pit walls, concrete Marble, slate
Closed housing
fs and the quality increases.
Shift of the resonance frequency fs to fc. Thiele Small

Parameter
fc = fs * (square root of (Vas / V B) + 1)
Displacement of the total Q TS to Q TC

Q TC = Q TS * ( Root out (Vas / V B) + 1) V B = Boxing
volume
Resonant frequency and the quality of the installed state.
bass reflex
The case of a bass reflex is a Helmholtz resonator Its resonance
frequency is tuning frequency (frequency of the Helmholtz resonator is
supported) radiating to the back
works on resilience of the air pipe
Due to the air mass, there is 3dB increase in resonant frequency, since the signal in the mass 90 degrees
out of phase ist.3 dB increase in the tuning frequency. Tuning frequency aware of the resonance frequency
including acoustic short circuit and -18dB per octave attenuation of the membrane (Chassi) fc When tuning
frequency + 3dB compared to a closed box.
(-12 DB octave reduction through loudspeakers + - 6dB octave acoustic short circuit)
Transmission Line Box
Similarly, the bass reflex but with quarter-wave. Vote on the length of the
quarter-wave mounted behind the diaphragm.
1/3 on the side of the chassis with fiber material (buffer)
Bandpass enclosure
Closed housing gets a Helmholtz resonator

Helmholtz resonator is intended to cover a range of the resonance frequency of the diaphragm. serves to
radiation (but excluding) the bass range.
Page 378 of 466
The high frequencies are pulled out.

Emission takes place only in the area of the Helmholtz resonance frequency.
Rated power Pn
Max. Power in continuous operation testing and noise from 300 h. 1 min. and 2
min. from, PMPO Endurance
important for individual pulse to the maximum loading.
Efficiency = (Pak / pel) * 100%

Other details = SPL at 1 Watt power supplied in 1 meter distance measured.
Double sound pressure quadruplicate efficiency directional
factor Directivity.
Directional factor
comparison Sound pressure on the major axis with the sound pressure on the sides of
Value between 0 and 1

Level drop information.
Directivity:
How strong in dB decreases the level.
Beam angle
Angle range where max. 3 or 6 dB drop relative to the main axis gets. -3 Distance with 3 dB drop
-6 Distance with 6dB waste
Other speakers to
14) An acoustic short-circuit is a level drop by superimposing the and forward
behind the sound emitted in the uninstalled chassis.
15)
QQQ
ES
MS
TS
+MS QQ
ES
16) Objects of the spider, the centering of the coil in the air gap of the
Permanent magnets, and the return force.
17) It is not true that the overhang coil is as long as the gap.
Page 379 of 466
Live art
Safety is always at the forefront Troubleshooting: From back to front Mic last
rules Light and audio conductor laid out separately (separately)
- DMX with the light wire transferred not to the audio conductor
- embarrassed Tonstrom on audio conductor
otherwise I build a capacitor, depending on the distance between the plates. Sound and light cables can be
60cm possible distance or more at the same time availability of light and sound lines.
to crosstalk of light to avoid If the cross, then possible at a 90 degree angle on clay,
Sound and light should never be the same current cycle, phases 1 phase
light tone. Otherwise light sprinkle.
Sound Current should be possible on a mass, otherwise hum at 50 Hz. Stage power
belongs to the sound current amplifier, etc. ground lift Ground Lift DI Box Mixer DI box Effektgert
Galvanic isolation between the mobile and fixed parts of the PA. PA Public Address
Only the portion of the boxes, the sonicated audiences.

multicore multicore cable
Page 380 of 466
Registration Setup:
8th
9
10
10
8 FOH
1 = PA boxes:
The Standing sideways so that the speakers are acoustically decoupled from the
stage, it may not sound bridge type rigid connection, otherwise oscillates the stage
with .--> Low Cut
2 = Power Amplifiers:
As close as possible to the pits short signal as little power loss

in the cable
before it in the boxes
is broadcast. Thick cable far pathway.
3 = 4 = Monitor Amplifiers
Wedges
5 = Drum Fill
6 = Side Fills
7 = -Front Fills
8 = FOH
to set the monitors in the Off-Axis: cardioid,

hyper cardioid, cardioid
Drum monitoring Speakers at the ear level and as close as possible to it
on the ear. Fill should be on the side of the hi-hat.
Sonication the whole stage because the small monitors are addressed and
radiate narrow. can play stereo. Play Mono so let the instruments left or right
are not up or down. Stage separate. Inverse stereo. FOH Cross Monitoring X
-Y (10 to 2 am is very wide stereo)
Midrange and treble reinforcement for the audience in the middle of the stage
Sound engineer, lighting technician, Mixer Stands on the acoustic axis of

symmetry (exactly in the middle) In the simplest case monitor mix is a
bout the FOH
operated via Aux paths. Until 5 ways you go about FOH mixer otherwise own monitor
panel only for Monitor Mix. Monitor console on stage practically very good
communication, The only one allowed to say something about the monitor mix is t he
singer. The more professional the band, the less important the Monitor Mix
Page 381 of 466
Option 1: Only stage sound via Parallel Split to the amplifiers

Connect the FOH system.
option 2 Direct out of a panel to another panel
In both cases the level of a mixer affects the level of the other mixer.
option 3
best possibility
Splitter, split box
By means of a transformer
Transformer coil = 2 face each other at the entrance. Coils are all the
same when the number of turns is equal.
2. coil is the magnetic force from at 2.Spule. Splitter is galvanically
isolated .--> DC is filtered out. Coil induces magnetic field in
alternating on the splitter.
3 ways available, eg WDR then docked on to the grit box to the

mixer. Optocouplers
Splittbox
10 = Stagebox Wallbox without wall. Laying Kabelweg trip over Cable under the stand triangle clearance for the
mic cable when moving.
Cable for live applications
XLR 1 ground 2 hot 3 Cold

Symmetrical Line Signal
Balanced Mic signal DMX light
signal
Semi-professional volume cable with 3 wires without jacket
AES / EBU digital mixing console wiring pawl
Mono, Stereo Small, large,
very small bulk = 6.3 cm =
3.5 cm Mini Micro = 1.7 cm
Tip = Hot
Tip = Hot
Ring = Cold
Stereo = mass
Sleeve = Ground
From balanced to unbalanced is 6dB level loss

-Instrument cable normally unbalanced
- Line signal connection semiproffesionell (patch bay at the FOH position)
- headphones
Tip = Left
Ring = Right
Page 382 of 466
- ISR cable
Line signal
-Aux
Cinch
unbalanced Tip Hot

ring Mass is for a player device uses the S / PDIF
signal routing.
Speakon cable
proffesional Boxenkabel
Extremely Reliable 2-8
Pole
1 + 1-1 + 1- etc. no jacket
-Plugs
Plugs are screwed Multi Pin Connector
Canon ITT Multi pin connector but there's also 2-8 Pins
312
Harting
4 Kant plug
Sub D
25 pole Sub D Digidesign interface Sub D
E-DAC
25 Pol
Toslink
visually if there are flexible fiberglass, it is the new routing are in

the live area.
BNC
2 core
MADI
Multichannel management
live
Page 383 of 466
Request to a PA
-Clarity and transparency (you should be able to hear the music)
- Improving the acoustic balance, if possible the same sound pressure
everywhere in the concert hall.
Correspondence between auditory and visual localization
Mastery acoustic complicated three-dimensional shapes (reverberation
time charge)
Long reverberation time is not good beschallbar due to long reflections trick up
possible softly as many speakers. avoid echoes Boxing does not depend on hard
reflective surfaces.
Small PA Great PA
Klein PA :
Up to 1000 watts one speaks of a small PA A BOX covering the

entire frequency range. Power mixer Mixer with amplifiers space
for people to 200 people.
Basically: So few signal as possible to the PA because interference
modulation occurs. Kick and voice simultaneously complex there
Rippled again
indistinct PA
guitarists:
in small clubs guitar amp turn on the artist and not the audience
- as quietly as possible
Bass:
- Bass not on the system, only through the monitor do not hear by PA. Singing at
the monitors all go quietly to the PA.
Great PA:
From semi kW per site is large PA
Back then you could stand no 100 kW system, because the amplifiers could not hand
over the power.
Today 120 dB A volume monotonous music has to be loud. Mixer and external
power amplifiers
How big a system should be a rule of thumb = 10 watts apiece

100 People 1000 Watt
Gymnasium = 2kW per page
Rock 4kW per page
not bring investment to the limit Lieber take something stronger investment and untwist a little less.
Open Air takes about twice as much power:

A distinction is made between centralized and decentralized PA species.
Page 384 of 466
PA species
Central: Audible and visual localization agree. Decentralized: Optical and acoustic
localization can match
but need not.
Central PA types:
1.Zentralbeschallung: Mono system, centrally, above the stage, the system with the best
Speech intelligibility at all. No duration differences. Conveniently with high

ceilings. Localization point: center stage Not recommended for music
reproduction. It is mono. Good for Sprachsale Aulas interface principle:
The box must be flush with the wall inlets. The entire wall is resonated.
Due to the pressure accumulating effect. Standing waves reflect, when the diaphragm diameter encloses
the frequency, there is something for high frequencies
Pressure storage. We make the wall a huge membrane through the

Boundary Microphone.
2. Frontal PA: A speaker system on either side of the stage.

Usually the boxes are on stage height. Stereo system locating on the
stereo basis. System Mobile PA Fair Eclectic Concerts, shows, panel
discussion Uneven distribution of the sound pressure
forward much pressure behind
little pressure.
2 ways to fix:
1.Delay Lines
2. boxes flying in the air, hanging from the ceiling.
Usually basses do not fly only mid and treble Three advantages flying
Boxing:
1. uniform sound pressure
. 2 no shading of the audience
. 3 Horizontal and Vertical boxes alignment
. 4 Make no visual obstruction is.
If less than 80% of stage one sees arise should the organizers for tickets.
Page 385 of 466
Distributed sound types

1.Nahfeldbeschallung:
Any sound for near field (front fills in mono) plug of the acoustic hole
through highs and mids. In the case of this sound is a secondary PA
2. armchair PA:
In each sitting a loudspeaker installed No optical
and acoustic localization comb filter problem
Delay the other seat of the speaker is a little
louder to a speaker in each row.
3. Ceiling sound
Very uniform sound pressure on a wider range comb filter are connected
acoustically transparent delayed. Unfavorable at high ceilings .. low with
low ceilings. Muzaks sprinkler systems.
4. Derived PA
there is the balancing with or without runtime ceiling sound. Is always a
secondary sound source, because it follows a different sound.
-Optical and acoustic localization votes do not accept. Follow PA with
maturity approximation: Prefer despite Rear boxing stage front locations can.
Rule of thumb: 3ms per meter delay. At 25 m distance Boxing delay by 75
ms Here the sound must be 10 dB louder from front to back.
Delay line must be maximal10 dB louder when maximum 1- 50ms comes later.
Above 50 ms of the Haas effect is no longer effective
Echos
Over delay Signal is delayed longer than it needs. Haas effect

was draufgeschlagen to 75 ms.
The shorter the Attack time, the shorter the Haas effect, the faster we hear a Echo.Boxen:
Delay Tower, Delay Line
Page 386 of 466
Other sound types: Diagonal PA

- Typical Disco PA
Throughout the area you stand in the stereo triangle Deflected that the
boxes are never completely out of phase with each other.
SFA - PA Selected Function Array

Additional Pa in the PA, where only individual parts of the program will be
played in turn, including voice over a loudspeaker (singing)? But if it is not
played on the main PA. Specific settings for voice and the rest from the main
PA. Very good
Inside the boxes controlled by buses and routed from the
voice in the outs.
Up to this point you can all sound types together combining except the last
Beschallungsart.
Delta stereophony:
All support speakers included Delay. One needs for each one Sim.lautsprecher with One Delay Time of
3 * 3 The simulation speakers are active only when it is spoken.
Internal communication: Stage Manager must FOH engineers and lighting technicians
Talk.
Followers need voice communication
about
Interkom
Local operator Organizer. Gofer. Assistant Job Internal

Communication:
Problem: Live there are very many radio links Security
- Micros
- Paramedics etc.
Speaker:
Speakers are most important Quality focus in Live area

Studio microphonic
Everything Electrodynamic transducer at high frequencies Piezzowandler

Problem: treble sounds too sharp. If feedback is buzzing, bad for the ears. For
the bass, there are motor boxes Motors transmit the signal as possible thin
membrane choose so that inertia does not occur. You can up to 30 inches
speaker diaphragm. high wavelength
Transfer. Go to 50 Hz, motor drives the membrane.
Page 387 of 466
Power Distribution: Bass 4 70% energy content

Facial 2 20% energy content highs 1 10%
energy content
Bass have more energy than heights so you need more energy in the basses to them identical display as
midrange and treble.
Frequency and power management:
out speakers with frequencies that they may pose.

eg bass speakers from 60 Hz. Controlled by the controller.
Typical Chassigren: Subs:

-15 inches
- 18 inches deep /
middle: 15 inches
12 inches
10 inches
High Medium: 1-2 inches
Tweeter: less than 1 inch and less
rather obnoxious.
Tweeter: manufacturers with exceptional Chassigren
What can break with chassis Membrane, bead,

Kitchen sink DC voltage in the coil makes them hot.
at clipping
stack:
Vertical Group of boxes different remit

stacked as woofers, midrange and treble speakers boxes 3 boxes each other
cluster
Horizontal Group of Boxing same remit as 5 middle boxes.
Page 388 of 466
Coupling:
2 identical speakers side by side if half their wavelength for all frequencies the distance of
boxing wraps. Both boxes have the dual SPL.
If boxes 2 juxtaposed and play them the same signal, so couple the boxes for all frequencies
whose half wavelength the distance of loudspeaker surrounds together, it sounds louder and
better
+ 6dB more
For bass almost always almost never used
1,7mm
for heights, because 20 Khz
Line arrays
Coupling of all boxes by Transversallcher.
Due to this, the coupling will work on high frequencies. Trick in Line Arrays
always cluster but no full-range speakers.
Mostly flown At the top of the back

At the bottom of the front
in banana shape
Line arrays only operate at high speakers more than 5kW line arrays, there are not
in clubs. Nexo Line array reference with the right training
Only Nexo worthwhile for training not other manufacturers.
Topic cross to right of use in terms of music:

GVL statement as guitarist or as an instrument player of a band
for this performance you get your money back
From the GVL. Exploitation rights of artistic activity you get back of the invoice Focus is
actually driven in the orchestral Gema GVL discharged.
Page 389 of 466
Transducer / speaker manufacturers in the Live area
D & B leader PA boxes Eastern Acustic

W o r k s NH
e xi go h E n d ( H i p H o p )
Dynacord
first major company PA
E- Voices first line Chassi manufacturer
JBL bad
B e a d s o f t a n d ssoqfut i sc ho ya t e d m e m b r a n e
M e y e r s oBu on xd e s w i t h t h e d r i e s t b a s s o n t h e m a r k e t .
F i r s t c o m p a n y w i t h t r a p e z o i d l ae l s sb opxaersa lal d
e vl awnat lal gs e
No standing waves.
H K m i t t e l k l a s s e b e a r a b l e b oRx oe cs k T nu rRb oo lsl o u n d
B o x ems i d d l e - h e a v y
emitter types
D i s t i n g u i s h i n g t h e b o x e s w i t h r e s pFer cetq u
t oe nt ch ye i tr o dn ier se cat ri e
v i tr yo t a t i o n a l l y d i r e c t e d .
higher the more directed. Typical frequency directivity.
1 . B r e i t a b s t r a h l e n d e i n dBi ov xi d w
u ai tl h roaudt i ad ti or er sc t i v i t y
2 . T h e s o u n d N ca or lr uo m
wn
, high boxes
horizontal wide, vertically narrow directivity is tu

the horizontal one another
horizontal wide.
For music extremely comb filter Lastig, because

shifts at the edge of the vertical extinction.
it comes to the vertical narrow
Shadowing.
3. funnel or horn speaker
b u n d l imn og r e v o l u m e i n t h e m a i n d i r e c t i o n b y t h e h o
before the pressure chamber. Counteracts against
Particularly at low frequencies. The greater the la
aperture, the lower the frequency. More sound pre
emission direction.
Horn speaker for the bass: There are

3 types:
1 . F r o n t L o a dHe od r nB ooxp e n i n g f r o n t
2 . F o l d e d H o fronl dBeodx H o r n b o x
Low boxing greatness, nevertheless big horn
3 . R e a r L o a d fer do mB obxe h i n d l o a d e d b o x .
Only the Rear sound is derived in the

Horn. 5% of the electric charge is
converted into sound pressure. Conver
General -> 50% front
50% to the rear
Page
390
of
466
In certain frequency range of the rear sound

is delayed so that there is no phase shift.
Advantage of Rear Loaded Box Greatest efficiency,
very narrow Frequency Response No directivity
Horn loudspeakers from the middle :
are usually Frontloaded boxes because one works with much less energy.
Horn speaker of heights :
20 kHz is clearly directed than 10 KHz tone.
Constant Directly horns that have a consistent viewing angle over the entire frequency range of the
horn to vary widely addressed.
Multizell Horn
Horn is distributed in different chambers, the movement area of the chambers are differently inclined.
For example, 5 KHz goes out 15KHz is smaller and is on passage
Cracked, often until it is as far radiated as 5 KHz. Attenuation of sound

pressure characterized at high frequencies. Horn is heavily distorted
high frequencies high Brennungsverlust.
Diffraktionshorn
Each point of a waveform was the starting point of a new wave wine. Due to the special designs are all
frequency dependent on the mounting position. bent at 90 degrees or 180 degrees. A quarter length
same radiation at all frequencies.

Problem no further transmission range advantage
no energy loss.
Short and Long Throws
Short Throw: Bass reflection throwing wide but not far. Long throws
only horns Throwing wide but not wide.
eg Short throws for front Long

Throws for rear
Monitor speakers
Usually you have active or passive 2-way systems. directed relative narrow viewing angle
possible Feedbackarm possible transmission range as wide as possible monitor mix Bassarm
Page 391 of 466
wedges
actually typical monitor speakers

as robust as possible
possible high rear attenuation Audience to hear as little as possible. Very good monitor
speakers have a PTC resistance before the tweeters. No filter That is not an internal capacitor or
coil.
The warmer it is, the greater the resistance.

the resistance- is louder The colder it is, the smaller, the warmer. The louder, the
quieter the tweeter.
Feedback protection Feedbacks are high frequency normally.
Contra Portal Speakers

Come out of the theater art portal
stage opening
Speakers opposed to the stage opening Contra
Stage effect speaker

Serve the acoustic positioning reference Speakers at the door
makes the door Sound locating reference
Feedback / feedback
Signal flow is created in a feedback arrangement, when the second
signal is louder than the 1.Signal The quieter the sound source the
more likely the feedback. If all Feeden would there noise. The signals of
the frequencies must be equal phase.
The pitch of the feedback depends on the distance of the box up to Micro. The shorter the
distance, the higher the frequency, and vice versa.
Error Monitor not depend on hard reflective surfaces or no drums

on the monitors Compressor
also quiet signals louder
Feedback Hazard
No Compressor on the monitor mix.

Not in the compress channels but in the subgroups
Signal splitting 15
16
monitor
FOH mix
15, 16 through direct outs in the FOH mix and Monitor Mix
Monitorbox not in Off Axis

Micro not in the box.
No strong EQ settings for the monitor mix
Page 392 of 466
countermeasure
installing a graphic EQ in monitor path.

Aux Send graph EQ
monitors Priming
In Bhnenmic pipes
Graphic EQ on the monitor as long untwist to feedback to occur and maintain.
Then search frequency and turn down the graph EQ.
If you have to turn down half of the bands, something is wrong. Smart Tools
Software (FFT analyzer) Pengiune meters
Measurement Microphones ICDE (ball)
Condenser, pressure transducer
use Feedback Killer Sabine and

Behringer bad.
Shure and Sennheiser good addition to tuning in the monitors. In Threshold, there is a defined
value All frequencies above rated as feedback and steamed.
In a microprocessor, the feedback frequency is passed to a filter and lowered. Very limited Therefore,
in addition to the graph EQ for highly mobile Mics. Other methods:
installed in the monitor in the Aux path pitch shifter.
Not the same sound but no more feedback. Between 1/8 or quarter tone. Signal short delay with a
delay. The distance from the microphone box to change by approximately a few milliseconds (rather
with vocals).
In Ear Monitoring
-Headphone mixes (the best are of Hearsafe) Grommet earplugs where

you speaker is included.
-Basses are better represented by bass radiator by vibration
(Alternatively for limited bass response of in-ear monitoring)
Advantages: a sound source less
No feedback
Always the same setting
Disadvantages: trust to engineer must be large, high accident risk
No sound pressure level
no air is moved. Other feeling
Locating difficulties not hear
the audience
One need microphones so that the artist hears the audience. When in Ear default effects on
the signal at the receiver Integrated limiter to minimize signal you should be able to hear only
the monitor mix of artists as monitor engineer.
-additional spark gaps

Page 393 of 466
amplifiers
-2 types of amplifiers
a) pre-amplifier (raises signal to operating level)
b) power amplifier
a) pre-amplifier
in possible little noise, so high Signal to Noise Ratio to have. It raises the level of a voltage.
voltage amplifier
1 level
Broadcasting House standard level
+6 dBu
2.Pegel
studio level
+4 DBu
3.Pegel Home recording level
- 10 dBv
b) power amplifier
1) One-hundred-voltage technology
2) Low-technology to 1) ELA Technology

Electric Acoustic conditioning (70 Volt technology) for durchsage
systems
Conferences, etc. all hundred volts technology constant 100 volts are sent
from the output stage to the box. P = U * I
current flow
-The less current is sent out, the smaller is the current

resistance.
-Much lower power resistance
- Much longer cable runs without loss of performance.
- high voltage delivered by less
amperage.
To accommodate voltage down to use them, there are
transformers in order to transform down or back up.
Adapting transmission therefore only parallel impedance and
performance differences are balanced in the transformers Amp
(voltage adjusted it)
Without 100 Volt technology performance would be processed in the

cable. (Extremely easy to maintain safe) coil induces an alternating
current. The higher the frequency Nevertheless stronger operates the
magnetic field, low-pass CR member has high-pass character capacitor
has 100 Hz to 7 kHz Frequency response
Page 394 of 466
Low-technology
-Audio
Audio signals are voltage Adapted transformer voltage drops in resistors = Voltage adjustment, the
input resistor is much greater than the output resistance. Distortion-free signal is ensured because the
Einganswiderstand many times smaller than the output resistance. Power resistor =
Einganswiderstand equals the output resistance. At live events Boxes and amplifiers adapt. If load
resistance is 8 ohms to 4 ohms Einganswiderstand amplifier Resistance is greater about half power beranpassung
If load resistance 4 ohms to 8 ohms Einganswiderstand amplifier blows overwhelmed because more
power than possible required
Parallel connection of resistors: Rges = R1 + R2 + R3 + .. Rn series of

resistors: 1 / Rges = 1 / R1 + 1 / R2 + 1 / R3 series circuit at the output stage
coupling better
no Leistungsverlust-
Calculate how thick must be the cable at the maximum length and max. Power q = (2 * L * f * (100
-n) / * v nq = cable cross section [mm] L = length of the cable [m] = specific resistance P =
power [W]
n = permissible power loss [%] U =

Voltage [V]
Operation selector
1. opportuni. stereo Both amplifiers operate independently 0 = Configuration
2. opportuni.
2 bridge circuit
to 1 chance
parallel Bridge
left input channel is set to both amplifiers. (Y split of the

final stage) power doubling twice the performance for a
mono channel L or R.
For Option 2 Mono Bridge, as if left input signal successively both

would pass through power amplifier blocks, 4-fold volume between
the two + Poland, the signal is tapped.
Page 395 of 466
If the amplifiers are connected in parallel, it is clarified internally that it rests at

Leistungsvervierfachung the same internal resistance.
Protection circuits for amplifiers:
Overheating protection:
- 1.Khlkrper
- 2.Lfter
- 3. Temperature protection fuse
- 4. Biegmetallsicherung
2 metals having different temperature coefficient Will's warm,
the securing bends.
Good amplifiers regulate quieter and does not disconnect from.
5. Water cooling (most effectively)

Adjust 6. Fans
Overvoltage protection
- Short Fuse
If the line on the ground completely runs short Therefore short fuse before
the final stage outputs overdrive protection (DC) Headroom 10 dB
-
Because the transients that are 10 dB louder he usually
has introduced a 10 dB headroom.

In power amplifiers, a limiter is installed
whose threshold is just below
The clipping limit.
Controllers installed a limiter with SendInput. Comparator instead of a VCA. If the output
exceeds the reference level, input level is completed. Only when the output actually overridden.
DC backup power amp channels
separate fuse
Page 396 of 466
Crest factor
Average deviation from RMS

Ratio of the mean value of all peak level to the average of all levels of the amplifiers are designed for
particular crest factors. music Crest = 10 dB language
Sine
= 7 dB
= 3 dB
The less dynamic nevertheless the crest factor THD

Proportion of non-linear distortions of the total signal How many new
overtones produced the component.
Damping factor:
Describes how strong would distort the final stage. Relationship between
internal resistance and the rated load impedance. This must be maximized.
Rated load impedance of the resistance must be at least is otherwise the device is warming up.
Mostly run the 200 kOhm. Working with very high resistance and are transformed by transformer up
and down, thus not transfer the effects on signal.
Page 397 of 466
multichannel
Dolby noise reduction method A invented. Dolby Stereo
made possible.
Dolby Stereo is an analog Matrizierungsverfahren with 4: 2: 4 channels. L + R = C L R =
S
Magnetton
Vormagentisierung strip material 1943 (of Braunmhl / Weber. Magnetton has ne time
superseded optical sound. Benefits = better quality = disadvantage expensive
1952 Paramount Cinema format:
Wide screen format with 4 projectors, 3 projectors for image

4. Projector for sound.
Initially 7 Languages
Problems with synchronicity relatively quickly disappeared because it was too expensive and impractical.
1953 20th Century Fox:

Cinemascope is anamorphatisches broadband procedures with special lenses image is projected. 35 mm roller
and 4 audio tracks (L, C, RmS)
1955 Todd AO
Cinemascope similar but with 70 mm film role but with more tracks. (L, LM, C, RM, R, S) Too
expensive
60 years he was the collection of Dolby Stereos in pop music film "The
Graduate" with Dustin Hoffmann with pop sounds. New Hollywood
Young Guard of directors, the films made in Hollywood.

Apocalypse Now Francis Ford Coppola Star Wars
George Lucas
Sound design, make sound to image
Page 398 of 466
Term sound design from the period.
Sensurround:
representing 1975 Universal "Earthquke" action with sound infrasound. Up to 15 Hz

deep. 6 tracks on 70mm 4 tracks for 35 mm.
80 :
Dolby Stereo 1982 (4 channels L / M / R / S) Dolby
Stereo in Bad
Noise reduction by the playback. 1986 Dolby

Prologic
6 channels
Prologic 2
5.1
6.1 and 7.1
1992 Batman Returns
Dolby SR-D
Not yet 5.1 with 16 kHz Figure

Analog optical sound is on it as a fallback always on the copy.
Dolby Digital 5.1

Came with Terminator 2
Format: Movies and DVD's 3/2/1 6: 6: 6

transmission channels
Even light tone to copy it between Perforationslscher encoding of Dolby
Digital
AC-3 (Audio Coding)
Compression ratio to 10: 1384-448

kbit / s
5.1
192 kbit / s
(Stereo / 2 channel)
LFE channel is limited to 168 bits. DTS has a greater dynamic. Dolby Digital
has prevailed.
Dolby Digital EX
1999 Star Wars Episode 1 6.1 (+ CS) 7: 6: 7 rear center surround is mixed on the rear 2
channels on it
and matrixed. Discontinued because lack of space on the film roll.
Dolby Digital Plus HDTV (HD-DVD)
6 Mbit / s data rate which is 13.1

THX no Sound
Page 399 of 466
DTS Digital Theater System:

1993 Jurassic Park
1996 DVD
5.1 / 6.1 ES Matrix

Today 6.1 ES Discrete
3/3/1
7: 7: 7
now DTS HD 6 Mbit / s optical sound nut timecode to copy sound even on CD advantage more
space will be compressed with APT-X 100 882 kbit / s
DVD: Coherent Acoustics 1.4 Mbit / s

DTS: HD 6 MBit / s
SDDS Sony Dynamic Digital Sound
1993 Last Action Hero Based

on Todd AO 7.1 8: 8: 8 5/2/1
Coding: ATRAC (Adaptive Transform Acoustic System)

1.4 Mbit / s
DTS: HD 6 Mbit / s
Other multi-channel process Mpeg

2 Audio:
Up to 8 channels of 2: 2: 2 to 8: 8: 8 Max
compression ratio. 12: 1 6 channels 5.1 400
kbit / s
Mpeg 2 AAC (Advanced Audio Coding)
Dolby E:
Encoding format for a transmission mode 8: 2: 8 (5.1 + Stereo +
2 channel sound) data rate 1.92 kbit / s
PCM (Pulse Code Modulation) sampling

analog to digital audio sampling
Uncompressed audio
Page 400 of 466
MLP (medium Lossless Packing)
2: 1 but lossless
1.4 and 9.2 Mbit / s
DSD (Direct Stream Digital)

SACD 2.8 MHz 1-bit converter
Dummy head stereophony 2ch

2 channel
one of the oldest 2 channel Stereophonien
quadraphonic 70
4 discrete channels
Binaural Room Scanning (BRS):
ITU 77 S
To what I hear on the headphones just to listen to a plant. In order to make possible for head movement,
I need the Adaptive bersprechungs- compensation.
Wavefront synthesis:
A sound source of any kind should be in any position the same place. At any point of the
diffuse and direct sound should be the same. Realizable by speaker arrays on CPU.
Controllable in all positions. Realizable scans by a sensor of the movement and all the speakers
adapt to this.
2 + 2 + 2 method
L, R Ls, Rs L, R
10.2 procedure
(Front Speaker / 3 Surround Speaker / 2 Subwoofer + praise and R above) even plastic
diffuse sound audible.
Speaker Placement for ITU (International Telecomunication Union) RBS 755.1 (3/2) so that a
mixture sounds the same everywhere in the room.
Page 401 of 466
3 at the front (L / C / R) 5 behind (L, LC, C, RC, R)
Bipole and dipole speakers:
Work opposite to each other and extinguishers off. The aim for the rear speakers by more
surround sound and diffuse fraction. Are the side and rear boxes of cinema. If front isolated
acoustically by phase shifting of the boxes L / RC.
X-Curve cinema 2000Hz everything from constant as per octave 3 dB drop. THX is a seal of
quality for cinemas Founded in 1983 by George Lucas. prerequisite Speaker Type, cinema
acoustics, speaker positioning etc. LFE + 10dB Loud 78 dB level in the sweet spot. dialogue
normalization
- 31dBFS for dialogue normalization

EBU 90 Guidelines for a mixture
1. Quality of the front sound image:
In front sound all sound sources are genuine and are easily determined (width,
localization, stability, distribution)
2. lateral and rear sound quality:

Side and rear sound becomes balanced. Same points of the front sound image homogeneity of surround
sound.
3. spatial impression
The sound content appears in an appropriate spatial environment (direct / indirect)

4. transparency
The details can be clearly perceived (clear / cloudy)

5. distribution
The sound sources in the sound appropriately distributed (Front / Rear) dynamics / degree
(course / compressed)
6. timbre
Characteristic sounds accurately reproduce the direct sound Des reverberation.
(Garish, dark, bright, warm)
7. Error-free and free from noise

Mixture should be free of noise and noise to be, not to be compressed and bergemastert.
8. Whole main impression of the mixture:

Only satisfied when all 7 have been met.
Page 402 of 466
Others to multi-channel sound
1) Dolby Surround is a de- and Encodierungsverfahren for H EIMBEREICH.

2) Ton Dolly digital method is the 35mm film IN BETWEEN of the
Perforation.
3) 7 speakers with Dolby Digital Surround EX and DTS ES: L / C / R, LS / CS / RS and
LFE.
Page 403 of 466
Multichannel Mifkrofonie
Movie
music
center : dialog
L / R Rear : Acoustic environment
center : Soloists, formation of

phantom sources
L / R Front : Picture of instruments produce
phantom sources
L / R Rear : Gravity, space
main emphasis : Localizability lap recording

level differences
Main emphasis: Spatiality, recording of runtime

differences.
L / R Front Offton, acoustic environment
Coincident: both capsules are close together.

This is about how to create with main microphones a surround mix.
Mono and stereo compatibility
How good is the downmix from 5.1 to L / R stereo. How well do you
create the mono or stereo downmix. Depending on the following criteria.
mono compatibility
How well can be created without causing sound deterioration by a comb filter Mono down mix.
stereo compatibility
How well can be created without causing sound deterioration by comb filtering, and taking into account
the appropriate recording angle a stereo down mix. The shooting angle of L / R will be smaller if I leave
as the shot angle L / C and C / R C-Mic.
process
Stereo + C
Oldest method of surround recording technique. 3 channel LCR front. All conventional stereo
configuration is complemented by a center microphone.
Page 404 of 466
This must have a certain distance to the stereo microphone. 2 m above the stereo microphone is
the center microphone. Bigger Stereo surface by C-Mic. Without it sounds wider than with. Center
microphone is omnidirectional. Good for music because of the spatial extent.
stereo downmix:
Only the stereo configuration
Mono downmix:
2 variants
a) only Center Mic with a treble boost for 2 m high position
b) L and R front to mix signal
ABC method:
3 omnidirectional microphones
Center microphone 37 cm forwards. Stereo recording angle of L / R must be as small as
possible. Total recording angle
Mic distance a in cm
Mic distance b in cm
100
87.5
158.5
130
74
128
140
64.5
105.5
160
57
88
Small Gesamtaufnahme angles are better. Center microphone is always 37 cm from the connection axis L
to R microphone. The greater the distance L to R, the smaller the total recording angle.
Calculating the partial pickup angle d = 37cm

/ (sin (1/2 ))
d = microphone base LC and CR and is the partial pickup angle. Good for
music DVD music production.
stereo downmix
We need the C-Mic to mix with 10-30 ms interval to L / R for spreading the shooting angle. It
must be far enough away to avoid comb filter and close enough to avoid echoes.
Mono downmix
Only Center signal used
Page 405 of 466
INA 3 methods:
Ideal kidney arrangement with 3 microphones Volker Henkel and Ulf Herrmann. Combination of 2
equivalent arrangements (LC and CR)
Recording of level and delay differences. The two part attachment bracket must encounter in the
middle just to each other without gaps and without overlapping. Gesamtaufnahme angular distance a
Mic
Mic distance b
Triangle height t
100
69
126
29
120
53
92
27
140
41
68
24
160
32
49
21
Level differences and time differences exist, suitable for movie sound and music
stereo downmix
Similar to ABC to small shooting angle if only L / R inputs are available. Therefore center with 10-30 ms
is mixed to enlarge the shooting angle.
mono downmix
Up 131 degree total angle recording only use C- Mic.

With a larger overall shooting angle L / R front are delayed with this mixed.
surround microphonic
United AB 2-3 m mic base aligned (to 10 m possible) 2 kidneys opposite the front assembly. Time
interval for front assembly 10-30ms Haas effect is to be exploited, so that will not affect direct signal
components of the front microphone locating. Locating only towards the front due to utilization of
hatred effect.
in 3.4m to 10.2 m because the signals arrive Aufstellu8ng therein to 10-30ms later surround
microphones. No delay is necessary.
INA 5 methods:
A rear area is pulled apart.
Front section is compressed. Ideal kidney arrangement with 5 microphones. Fixed configuration for a
360 degree detection. If used for film sound
around
Locating.
Good localization and level and time differences for film and sound.
Page 406 of 466
Stereo downmix:
L / R alone would be a small camera angle showed C with 10-30 ms Delay to be mixed. Surround
signals are mixed with a time delay. L / R front with admixture of 10-30ms delayed Center and L / R
surround signals.
Mono downmix:
C signal with delayed addition of L / R and surround L7R.
Brauner ASM5 + SPL Atmos 5.1
5 Brauner VM! Capsules with fernumschaltbarer directivity in INA 5 arrangement in combination with
the surround mixer. SPL Atmos 5.1 level and time differences for film and music.
OSIS 321
Optical Space Imagesound
Optimal Surround. Localization and sound. Extra for music recordings. Level of the
center very low -10 to -15 dB. Optimum space, localization and sound / sound. Front
assembly:
Image- disc with 32 cm in diameter with 2 pressure receivers with 20cm microphone base.
Shotgun in the Image pane installed to support the soloist. This produces good localization and
level differences, Good sound for the DE.
Surround arrangement:
Space disc 28cm in diameter with 2 rearward kidneys are more spatiality. 19 cm microphone
base.
3.4 m to 10.2 m distance from the front assembly for music.
Stereo downmix:
Both signals L / R front.

One can possibly after L7R surround and mix yin this case without delay.
Mono downmix:
L / R front can be mixed with mono possibly L / R surround.
Holophone (Rising sun Productions):
Elliptical separating body. Total 8 EN are in this body incorporated DPA4060 capsules.
Frequency-dependent level differences and time differences. 3 Front capsule signals and 3 surround
capsule signals
Page 407 of 466
1 capsule lowpass signal: the differences in level timbres differences and time
differences for film and music.
Stereo downmix:
L / R Front possibly supplemented with the time-delayed further signals capsule.
Mono downmix:
C signal possibly supplemented by delayed further capsule signals.
IRT Cross;
Parts Cross developed by Gnther Theile.

If a surround atmos microphones for music to complement a 3 channel front assembly in 3.4 to 10.2 m
distance. L / R front of the front assembly is extended, the IRT Cross a C- microphone mixed together
for film sound. 4 kidneys with a Mic base of 20-25 cm and 90 degrees opening angle and 90 degree
partial recording angle. Parts with part recording angle may overlap to 10 degrees.
Hamasaki Square:
4 laterally aligned with Make 2-3 m microphone base and 3.4 to 10.2 m distance from the front of the
assembly Hamasaki square. One would only receive diffuse sound. Front L / R of the front assembly
of Hamasaki square are mixed together. Creates distinct space and is used for music.
OCT system:
Developed by Schoeps. Optimized Cardioid Triangle
Simple OCT Front System:
1 kidney + 2 supercardioid. Kidney is 8 cm from the connection axis Left - Right Mic. L and R Mic
are aligned laterally. Mic base L / R 40-90 cm. Mic base L / R
40cm
Recording angle 160
50cm
60cm
70cm
80cm
90cm
140
120
110
100
90
OCT Front System + ball

Ball is to compensate for the bass boost in L / R mix.
Expansion of the OCT front system in a pressure sensor on the front with carbon microphone via low pass at 40
Hz is added the signal of the DE to L / R.
Page 408 of 466
OCT Front System + 2 ball
Advanced OCT Front System with 2 EN with several meters Mic base, which are mixed to L / r front via
a low pass at 40 Hz.
L and R signal yield a fairly small shooting angle.
Stereo downmix:
L and R front complemented by time delayed C signal.
Mono downmix:
C signal possibly supplemented by L / R front.
OCT surround system:
OCT Front System expanded to include even a rear aligned AB arrangement with 2 kidneys (Mic base
of A- arrangement is always 20cm higher than the base Mic L / R front) record Lateral phantom
sources. This can be mapped and temporal phantom sources.
Fukuda Tree:
5 kidney + 2 DE
2 EN to be L / R Front mixed pronounced venue for music.

Stereo downmix:
L / R front with delayed mixing and possibly the surround signals.

Mono downmix:
C signal with delayed admixture of other signals.
Filmton specifically:
Double MS arrangement:
Designed v. Schoeps (Schoeps WSR)
DMS splitter
2 kidney + 1 night. The DMS Splitter provides: 2 front

kidney, 2 times 1 time Eight rear Kidney Center: front
kidney
L / R Front: MS of front cardioid and figure eight L / R
surround: Ms from rear kidney and eight.
Sphere microphone:
An eight is directed forward. 2 laterally directed MS arrangements.
Page 409 of 466
L Front and L surround and front R and R Surround. Center must be mixed together.
Spherical surface with 2 GB on each of which a forward Eight is. This results in a left and a
right-facing MS arrangement. The C signal is mixed with a matrix of Grenzon L / R front. Processor
DSP4 AFM 360 accepts the matrixing. Perfect stereo and mono compatible .--> for movie and music
SoundField SPL D22B:
4 kidney tetrahedral arranged A format: 4 single kidney signals from the A format is created
with the appropriate processor the B format. W ball
X Forward-looking Eight Y leftward
Eight Z downward Eight
If you mix a ball and a night together, resulting in a kidney. W + X = forward facing
kidney center WX = rearward kidney MS of (W + X) and YL / R Front MS of (WX)
and YL / R Surround
MST method:
Center side Depth: 1
ball (W)
1 Eight showing forward (x) 1 night,

pointing left (y)
Matrixing as the B-Format SoundField SDS 422B 3 Eight is actually
meaningless.
MMAD:
Multichannel Microphone Array:
Design by Michael Williams. 5 kidney for 360 degrees positioning arranged.
Others to Mehrkanalmikrofonie:
Page 410 of 466
Mehrkanalmikrophonie
4) INA5: Ideal kidney arrangement with 5 microphones.
5) Coincident multichannel methods: MTS

6) Hamasakiquadrat works only with A FRUITS!
7) Locating at ABC method via L AUFZEITUNTERSCHIEDE
Page 411 of 466
production 2
Good bands sound always equally well without power fluctuations. Always the
same quality.
grooving constant to slow things is quite difficult. Count In: 2/4 then 4/4
a total of 2 bars
How I reckon a quarter of Delay: 60 / bpm
Number
In 3/16 -Y down expect to 16th and then multiply the time. 3
Surround: DVD Live ACT : L / R -Stereopaar everything has to be there Do not miss anything, because it is so only on
all plants
is compatible.
C bass drum, snare, bass solo part, Beat Vocal, Solo Instrument
dry
L / R surround Hall rooms, PA, size of the stadium, audience. Locating from behind is
hard, because the ears are pointing forward.
POP DVD's : L / R surround Percussion, loops, backing vocals, Backing Guitar. Backing Strings Center Bass
drum, snare, Main Vocal, Bass, Vocals, Instruments Front Dry L / R Front
everything in the center with effects (reverb) to L / R speakers.
Surround primitives on the Front Speaker. In 5.1 we must

do things by halves either entirely on one speaker or not
at all
but not a bit
not clear to the listener
confused.
it comes
Classical music:
Front L / R entire orchestra Rear L / R room
shares the hall
Page 412 of 466
center is almost not used.
Orchestra with Hauptmikrofonieverfahren: ORTF or AB plus Sttzmikrofonie for all

Instruments.
Full links
1. Violins
Left
2. Violins
center
3.Holzblser (clarinet, vgott, Oboe, English Horn, Flute)
Right
4.Bratschen
Quite right Celli Behind this orchestra Basses, impact testers, Brass, Trombone, Tompeten, Tuba.
2 Mics front main microphones 2 Mics rear

main microphones for each microphone spot
microphones.
Skirt:
center Base drum, snare, solo instrument, vocals dry phasing important in
Mehrkanalmikrofonie Extinction Very good live DVD mixture Pink Floyd
Pulse
Page 413 of 466
Music History:
1. The beginning:
Companies featured turntable ago to sell records

Problem no plates Time of live music
Live bands were brought into the studio .--> downfall of Live Business Produced plates
Plates sold.
2. First band Live musicians were the first bands like Pink Floyd and ACDC
By Studio Music Live business is broken.
3. invention of the CD early 80s.

All records were produced again as a CD and you
made
The double conversion. Fraunhofer Institute MP3 invented File exchange Napster
-Y CD trade went Kaputt downloads.
Large record companies began to only great bands
finance.
At the time, Bravo Hits were sold and the money you have
brought out new bands. But by downloads broke the deal.
Record companies less budget Music level was worse
Each manure came into the market.
Today life the record companies still. Of the Stars From then -Y no
money for new production
Entertainment area has in the years to be a problem because the good artists extinct and increasingly
lost due to the rapid and cheap production of musical value of music goes
Record companies go bankrupt all
no bands
4. New ways
for money in the music business to be large firms from other
integrate areas and the artists buy on their home page advertise their product to make. These
companies have enough money Such as Coca Cola musicians in future employees of Coca Cola
and Levis and his Sony you get paid by them and not get the money through the CD sales. Music
will always exist .--> Dies never count out what the future
the middle class disappears Whether on television or
Music:
Soon the collection of AUTHENTICITY OF MUSIC comes

ARTIST THE EMOTIONS VERMITTELN
THE MARKETING MANAGER that are now sitting in record companies superseded by MUSIC
MANAGER are again distinguish good music from bad music. Without the pressure of money problems.
Page 414 of 466
It is in the future to convey emotions about playing music not to fast Geldmacherei
Only the true music will survive only by real artists CDs to buy.
for advertising purposes of billion company radio stations do not hit, they play hits. It is our chance to
exploit the Digital quality and home recording standard is so high, but lacking quality in music will change
in the future. The mixture is not a static setting in the solo mode of each track The mixture is an act
which is driven by automations in each track.
Intro must convey emotions through other settings as the main part and outro and chorus,
It shows the average consumer and begins to mix the suffering.
Page 415 of 466
DVD authoring
CD Redbook: pure audio format Tot
Table of Content
Cd specialized for Redbook, is only for audio. So you need another book to
realize a CD-Rom. YELLOW Book
Yellow Book / Orange Book: Iso 9660 / Joliet Multisession rewriteable CD's Blue Book (1995) Separation
of Data + Audio
74 650 MB
740 MB 90 Mb is output for error correction

is on each CD pregroove strikes which projects are the storage of data. If one is fired just over it is,
because pregroove Cd has taken in the 90's like a bomb
1. Multimedia CD
. 2 Superdentity CD
Film industry has got wind of wanted to avoid that again a normal format emerges as VHS.
1995 Founded DVD Forum
Voluntary association of 213 companies to
Industry standards to set. Digital Versatile Disc Versatile Versetail
Even tighter track Prepare the pregrooves in the future color of
the laser Blue Red, Green, Yellow, Blue Violet
Length of the wavelength of light.
UDF
(Universal Disc rear)
File system that is used for the DVD DVD 5
but 4.7GB effectively are 4.3
SS (Single Sided / Single Layer)
(AC-3 Dolby Digital)

DVD 10 2 * 4.7 GB (4.3) DS / SL
Page 416 of 466
DVD. 9 Single Sided Double Layer (DL) 8.54 GB

7.9 Gb effective
2 grooves are provided on the DVD with different depths. DVD 19 DS / DL 2

times 8.54 GB DVD 14 DVD 18 DVD Plus
file structure
Video_TS
Transscript Stream
Audio _TS
VOB (Video Object) (max-1 GB in size) image Sound1, Sound2, image (max. 1GB in size)
multiplexing so that the sound and the image can run simultaneously with the DVD jumping. Laser
must be able to read all linear. Data rate of 10 mbit / s up to 8 audio tracks (not channels)
Max. 5.1, eg 5.1 German, English 9 times Camera (Multi
Angle) Max. 9 times the cameras switch back and forth.
Audio TS
If -Y brought in the DVD Audio
Page 417 of 466
copy protection
Macrovision:
CSS
(Content Scrambling System)
manufactured by the press shop Regional

Coding:
1.USA / Canada
2.Westeuropa / Japan / Middle East / Africa Egypt

3. Southeast Asia / Korea / Taiwan
4. Australia / New Zealand Central and South America
5, Eastern Europe / Russia / Africa / India
6. China
7. Reserved
8th International Territory

A DVD can be enabled for one or more codes, a DVD player can only be unlocked for a
code of a territory.
DVD Studio PRO

Bitbudgeting:
fits EVERYTHING ON THE DVD (calculate)
DVD Video DVD 5 120

min. Movie
37,600
5.1 AC3
* 0.96
2 stereo
36- 0.96
CAPTION 1500 / Language Different data rates Convert gigabyte megabit per
second
Gigabyte on megabit / 120 minutes 5.01 mbit / s
deducted 0.928 5.01

4.08 Mbit / s
5.01 Mbit / s
5.1 + 2 stereo has a fixed data rate of 0.928 Mbit / s

5.01 Mbit / s 0.928 Mbit / s = 4.08 Mbit / s
formula to calculate:
Preferences Apple +, orientations Colors and alignment of

objects. encoding Adjusting Mode 1 pass
Page 418 of 466
Bitrate or
1 Pass Variable BitRate bitrate
Preset 2 Pass involves analytical file
image is even faster.
method Encoding when creating

Background encoding
for weaker PCs created when exporti

stronger PCs faster
Function keys F3 rise window Function keys

secure WINDOW open arrangement Function key in
Manage arrangement Add.
palette Choose from different styles and objects which are inserting inspector
One of the main work area

No matter what I click, it appeared in the Inspector and I can adjust and change it there
as well.
3 main parts of the DVD Authoring menu
traces Slideshows
Links available under the bar list. representation toolbar

Adaptation
customize toolbar Standard symbols retrieve.
Check if everything is working what I set:

click Motion males and the program displays run properly if everything Motions. Simulate Simulates
DVD authoring with Fernbedienungsfuktions controller. Internal designation of the buttons so that they
can later recognize.
eg movie button
Play
Ctrl. clicking popup menu Menu simulate Everything About

Inspector regulated
Color, type of font, size, shadow, etc.
Template can reinladen via templates top right.
reinladen Videos Create a track under tracking link.
In inspector a start action set to have a starting point for navigation.

Page 419 of 466
Create DVD menu with levels behind each button own image, start in Photoshop with the
background image, button is Play not selected selected not selected 3 Wallpapers 2 wallpaper.
Not selected selected selected. For each state, the Button set. make Chapter Menus Menus.
stack
individual mono channels of 5.1 mixture in the stack and all send the import as a 5.1
Stream ...--> Import in the voice menu to
5.1 German
5.1 English and

so on
Page 420 of 466
Building and room acoustics:

Literature Tips:
Wolfgang Fasold: soundproofing + room acoustics in practice.

Publisher of Construction
F.Alton Everest + Master Handbook of Acoustics. McGraw-Hill

H.Herr, GRSinambari. Engineering acoustics, Vieweg Verlag
we know 2 waveforms
longitudinal Vibration direction corresponding to the direction of propagation
(Sound connection: air, water spreading)

transverse waves Vibration direction transverse to the direction of propagation 2 waveforms
are added:
Dehnwelle: If the waveform causes the sound propagation in rods. Sound propagation in
rods -> change of compression and expansion.
Bending wave: Transversal propagation
Sound propagation in plates, interaction between weak and strong

longitudinal transverse component.
coincidence effect
In a plate, it can lead to a spread of bending wave whose propagation velocity of the thick and the
material of the plate depends. Smaller air sound obliquely to the plate and the velocity of
propagation of an oscillation phase of the air along the disk surface corresponding to the
propagation velocity of the plate's bending wave as it comes to coincidence. The transverse
movement of the plate is maximum. Coincidence of Schalleinstrahlwinkel dependent.
The larger the angle of sound incidence, the deeper the coincidence frequency. The lowest frequency in
the even coincidence occurs is limiting coincidence frequency. In the field of this frequency, there is a
decrease in the action of sound on the wall. Different limiting coincidence frequency with varying wall
material and wall thickness.
Page 421 of 466
Room modes:
Standing waves by reflections with parallel shafts in a room. Wall distance of 1/2
lambda or a multiple thereof.
On the walls maximum pressure, in the middle of the minimum pressure at lower order waves.
1 = OK -> a minimum
2nd order -> two minima
3. Procedure -> three minima
The number of pressure minima corresponding to the order of the standing waves.
3. kinds of modes:
1.Axiale modes (standing waves between 2 parallels walls)
2.Tangentiale Mode (standing wave between 2 times 2 parallel walls)
3. Oblique Mode (standing wave between 3 times 2 parallel walls) With increasing
frequency, the number of standing waves is increasing.
Natural frequency density:
Number of natural frequencies (room modes) that is attributable to a 1 Hz bandwidth. They

should be at least 3 natural frequencies per 1 Hz at 1 KHz. This is the case above 15 cubic
meters of room size. The sound travels in 15 ms back 17 m distance.
Flutter echoes:
emulate Periodic series of echoes in parallel walls. If the wall distance greater than 17 m can
hear a series of echoes If the wall distance is less than 17 m so gets the flutter echo tonal
character. JE smaller the distance from the wall, the higher the flutter echo. Wall distance /
speed of sound -> frequency of flutter echoes.
Sound absorption coefficient alpha

Alpha = Yes / Je
Yes -> absorbed sound intensity The incoming

sound intensity alpha = 1 -> complete absorption
alpha = 0 -> no absorption Sound absorption is
frequency dependent plywood is a very good bass
absorbers. www.DEGA-Akustik.de
Equivalent sound absorption area A:

the area with alpha equal to 1, the same amount of sound absorbed as the whole surface of the
room.
Page 422 of 466
Reverberation time T
was first investigated by Wallace Clement Sabine. Sabine'sche

reverberation formula: Ts = 0.163 s / m * V / A
Volume, reverberation time and can determine the equivalent sound absorption area.
less for alpha equal to 0.3
high frequencies air absorption is much stronger than for deep. Question of the
optimal reverberation time: for language max. 1 sec.
Music Church music -> 3.4 seconds

Chamber music no less than a sec about 2
seconds.
Depth Freuq. longest reverberation times
Ratio of the reverberation time of the depths frequencies to Middle called Bass Ratio.
(125-250 Hz for the reverberation time at 500-1000 Hz) BR = (T125 + T250) / (T500 +
T1000) Language max.1
Music -> to be 1.1 to 1.3 longer.

measurements:
Reflecting poor room:

Uses for measurements in the free field as possible wants
to measure only direct sound.
Wedge-shaped absorption material which 1/4 of the wavelength of the frequencies extending into the wall. If
lined with wedge-shaped sound absorbers with 0.85 m length. Measurements can thus be possible to 100Hz.
Hall space:
for measurements in the diffuse field

sound-reflecting curved wall surfaces.
Suspended curved reflections.
Page 423 of 466
Measurement Microphones:
Always condenser pressure transducer
For measurements up to 20 KHz, we need fourth inch / eighths of an inch membranes. For
measurements in room acoustics to 8 KHz taking a 1/2 inch diaphragm for measurements in building
acoustics 4kHz you take a 1-inch diaphragm.
Pistonphone
small pressure chamber with a piston which is driven with a motor. This generates a certain
constant sound pressure. In the pressure chamber is the microphone capsule. If can thus the
output voltage of the microphone determine To calibrate a microphone. ie for the determination of
sensitivity thus the gain of the microphones is determined.
Measurement of sound absorption coefficient alpha
- Measurement in a reverberation room

About the differences in the reverberation time is determined the absorption coefficients measuring the reverberation
time with and without the test object.
The surface of the specimen should be at least 10 square meters. Determination of Alpha on
the difference between the reverberation time.
Measurement Kundt tube:

The pressure differences in the pipe depend on the proportion of reflections. Microphone probe is
introduced into the tube.
A loudspeaker broadcasts a sine wave into a tube, where the opposite end of
the test object is. The length of the tube corresponds to 1/2 lambda. it creates
a standing wave.
The higher, the lower the pressure differences in the tube. The measurement of pressure
differences is carried out with a microphone probe.
Time rated sound pressure level measurement
Averaging the sound pressure level SPL over a certain time Slow = 1 second
S -> dB (AS) averaging in time of a second Fast = 125 ms F -> dB (AF) pulse
= 35 ms I -> dB (AI ) peak 50 micro second P -> dB (AP)
On value measurement method
A value over the entire listening area.

VAT assessment process:
Level is displayed on multiple frequency bands.

Listening area is divided into individual frequency bands whose level is measured.
Page 424 of 466
Sound power measurement

For the measurement of sound power, there are 3 methods
1.Hallraumverfahren;
Measurement at different points in space.

One can only just measure the sound pressure and calculate the sound power. Measurement
of sound pressure level at different points in the diffuse field (echo chamber) and the
determination of sound power over the
Sound pressure level. The SWL is calculated on the common SPL.

2.Hllflchenverfahren:
In Engineering method measuring in an anechoic chamber is done. In an envelope with one

meter distance from the sound source. Measure from different directions the sound pressure of
the sound source to 1 meter distance. About the common SPL of SWL is calculated.
3.Vergleichsverfahren:
Measurement in any room .On an envelope of sound pressure level of a reference sound
source with known sound power level and the Prfschallquelle is measured. About the level
differences of the SWL is calculated.
Reverberation time measurement:
Frequency independent measurement as with RT 60th
There are 3 ways to measure the reverberation time frequency-dependent:
1. Fast Fourier Transform (fast Fourier analysis)

at any given time a sound broken down into the individual components,
Sinuschwingung, level difference, phase angle. If this discrete analysis is carried out
in real time. Problem: Noise is measured with. Measurement errors caused by
background noise.
Time Delay Spectronerty

Zeitverzgerungsspektometrie
Measure for each frequency, the reverberation time
By narrowband frequency bands. Background noise is cut out. Measurement of individual

frequencies accurately. Measurement with pure tones over narrowband bandpass filter. This
little background noise is taken. Techron TEF Time Energy Frequency
Page 425 of 466
MLS measurement:
Maximum Length Sequencer
A computer generates a noise whose frequency content are precisely known at any time points.
The measured signal is compared with the generated noise. Old signal components that do not
originate from the computer are eliminated. DRA Laboratories MLSSA Maximum Length Sequenze
System Analyser
Building Acoustics
A branch of acoustics, which deals with the sound insulation in buildings. It is necessary to address
airborne sound insulation and to structure-borne noise.
Airborne sound insulation:
Airborne sound brings walls to vibrate. Interior and exterior of

airborne sound insulation of a cabin. Sound propagation on main
road (on the part)
side roads (Flankenweg, shafts or Rohdurchfhrungen etc.)

sound reduction index R
Laborschalldmmma: Letter R Specifies only the

sound insulation of the wall. Sound on the main
path. R = 20 log (P1 / P2)
p1 incident sound pressure p2

transmitting sound pressure
Bauschalldmmma R '
R 'sound reduction in the installed state, ie with byways R' = 20 log ((p1) / (P2
+ P3) p1 incident sound pressure
p2 on Hauptwegbertragener sound pressure p3 on

Nebenwegbertragener SPL
Measurements in building acoustics go no higher than 3.5 KHz.
Page 426 of 466
Bauschalldmmma reviewed R'w
The measurement curve is compared with a reference curve. The reference curve is shifted so far
that the sum of the shortfalls in the measurement curve at Terzbandmessung 32 dB at
Oktavanordnung 10dB. R'w is the value of the shifted reference curve at 500Hz.
Impact sound:
In determining the impact sound is necessary to make a definite impact sound. With a standard
hammer, a precisely defined footfall generated. In the room below the impact sound is measured.
Impact sound Li
Sound pressure level in the receiving room.
Standard impact sound:
The absorption of the reception room is taken into account. Ln = Li - 20 log

(A / A0) dBSPL A = 10 square meters
It takes into account the absorption of the space with the standard footfall.
Evaluation of the standard impact sound Ln, w
analogous to R'w the measured curve with a reference curve is compared.

Impact sound reduction .DELTA.L
Reduction of the standard impact sound through a floor covering or a suspended ceiling.
Impact sound reduction Lw

Reduce the rated standard impact sound through a floor covering or suspended ceiling.
Walls:
Single-tier wall:
You have 3 different frequency ranges Very deep
frequencies -> not important - so high
Area of the mountain's mass Act:
ie a doubling of the mass per unit area gives + 6dB sound. Doubling the frequency is + 6dB
sound
Page 427 of 466
Limiting coincidence frequency:
In the field of this frequency we have a significant drop of the frequency response.
- coincidence area
(In the limiting coincidence frequency, there is a break in the sound insulation).
If the cut-off frequency above 2 KHz is unproblematic, because the sound insulation is
very good there. -> Bending points wall in the middle region is the cutoff frequency
problematic. Rigid walls -> limiting coincidence frequency is less than 200 Hz. We
want that the limiting coincidence frequency is in the good range. Bending soft:
plasterboard max. 18mm thickness
Chipboard up. 16mm thickness.
Rigid: concrete or masonry with a minimum of 150 kg / per square meter basis weight.
Sound insulation of a single wall deteriorates when eg reinbaut a door.
Bivalve wall:
Mass-spring system can be sure to specify a resonance frequency. At the resonant frequency the
amplitude of the shell in the receiving room is maximum. Doubling the frequency is 6 dB more
sound insulation. Resonant frequency considering all three principles.
Below the resonance frequency corresponding to the sound insulation of a double wall of a single wall
with the same mass per unit area. At the resonant frequency, the sound insulation of the double wall is
even worse.
Above the resonant frequency, the sound improves, but with 18 dB per octave.
Resonant frequency should be as low as possible.
should be below 100 Hz.
Page 428 of 466
The resonant frequency depends on:
-Mass, mass per unit area (in the sum of the two shells)
- The distance between the two shells to each other
-Depending on the dynamic stiffness of the insulation layer, between the shells. Examples of
bivalve walls with a resonant frequency <100 Hz. Dynamic stiffness lower
increases the resonant frequency.
1.a massive wall with bending-flexible furring and 5-10 cm layer of air between the shells.
2. Two soft bending shells with 10-15cm layer of air between the shells. The less mass,
the more air we need between.
It makes sense at 2 Bending turnout walls in different lengths to have, because at different limiting
coincidence frequencies have, instead of the same. Decoupling of the stator mill with the walls with
furring Kauchuk: The wall of the internal space of outer space.
Floors: screed
types:
Composite screed: Screed is in liquid form on the raw

Ground is applied. less improvement of airborne sound
insulation.
Separating screed:
On the raw soil a foil or tar comes cardboard on the floor.

Acoustically no difference to the topping.
Floating floor:
resilient insulating layer of mineral fiber sheets or mats, on foil or

tar paper, it screed.
it may sound there is no bridge.
Bottom plate on springs with approximately 5 Hz resonant frequency.
Blanket:
-Pladur on battens.
-suspended ceiling with abhngdrhten and vibration isolators.
doors:
Vulnerability: mass of the door panel
Seal in Trfals lower

termination
Page 429 of 466
Sound insulation of doors massive door 25-30 dB

Tubular particle boards with sand filling 35 -40 dB bivalve door
with about 30 kg / square m 35 dB. with additional
Minearlwollfllung planked about 40 dB with steel sheet.
50 dB
Upon completion:
-stop threshold
-Hump sill with towed rubber
- Sound Ex is ausgefahren.- when closing the door
glazing:
For Studio discs double glazing (or triple glazing) 2 separate disks in separate walls
and separate frames. The discs are tilted: -Verhinderung of standing waves
-slanting slices prevent reflections running
back toward the sound source.
Air conditioning and ventilation systems:
unfavorable: 1 Air conditioning for Studio and Control Room together.
(It should be selected in this case long power paths and this is equipped with absorption
silencers.) Low: 2 separate air conditioners.
Soundlock:
Link program between Studio, directed and outdoors.
Room acoustics:
A branch of acoustics that deals with sound propagation in rooms.

Technical sound absorbers:
- passive absorber
- active absorber
Page 430 of 466
Passive absorbers:
porous absorber
pored fibrous materials in the entering of the sound and in which sound energy is
extracted, which is into heat through friction Transformed.
-Acoustic foam (convoluted foam, pyramid foam) textiles (carpets, curtains,
upholstery)
- mineral wool
- Acoustic plaster.
The absorption is at a maximum at a quarter wavelength distance from the wall, because there the sound
velocity is maximum. 1/4 wavelength and its multiples.
Porous absorbers are used primarily for Hhenabsorbtion. If tubes or boxes filled with
mineral wool (bass trap) so the absorption may be extended up in the bass range.
Active absorber:
Are resonators beyond the sound by resonance energy. If therefore called

resonance absorber. Absorber for the bass range:
Plate absorber / plate transducers:
Plate resonators.
Rigips plywood or hardboard are mounted on longitudinal battens with at least 60 cm bar spacing in
front of a wall behind it. Resonant frequency depends on mass per unit area m [kg / sq m] is how
many kilograms per square meter surface the plate weighs. Wall distance [cm] formula:
fo = (600) / (square root of (m * d)) [Hz]
eg m = 5 kg / m d = 5 cm fo = 120
Hz.
At 120 Hz, the absorption will be the strongest, the further away it is, the weaker.
Page 431 of 466
Eckplattenabsorber:
The plate is mounted in a corner, the distance to the wall behind lying is d
different resonance frequencies -> breitrandigere absorption in the bass r
the plate in a corner with different wall distance. This broad-brimmed bass
Absorption in the vicinity of the centers:
Mid-absorber:
-P e r f o r a t e d p l a t e a b s o r b e r p e r f o r a t e d p l a t e r e s o n a t o r s
-S l i t p l a t e a b s o r b e r . S c h l i t z p l a t t e n r e s o n a t o r
-H e l m h o l t z r e s o n a t o r s a r e a i r c h a m b e r o p e n i n g .
Air in the chambers has a mass, spring action of the air. Borungen an
slots are an air mass. -> Behind, we have a confined volume of air -> spri
of the air.
Plates with Borungen or slots which are mounted on a batten in front of a
fo [Hz] = 5400 * square root of (p / (l '* d))
p = perforation percentage .--> what percentage of the area make the

Perforations made. l '= l
+ 1.6 r
Mndugskorrektur l '[cm]
l p l a t e t h i c k n e rs si
s the radius of the holes. [cm]
d = wall distance [cm]
eg a perforation of 10%
a muzzle correction = 2 cm r = 1cm d
= 5cm
f0 5400 * square root of (10 / 3.6 * 5) =
5400 * square root of (10/18) 4024

Hz
l '= 2 cm + 1.6 cm = 3.6 cm
Correction of the air mass holes by 1.6 cm radius.

Diffusion:
Diffuse scattering from textured surfaces over about 2 octaves of bandwidth. P

resonators simultaneously absorbing in the central region
and diffuse scatter.
Page
432
of
466
Schrder diffusers:
Arranged in a line shafts with different shaft depth. It even differs to
Primitive Root Diffuser -> Interspersed asymmetrically
Studio area, receiving space usable

Quadratic Residue Diffuser -> stray symmetrical, evenly in all directions. Number of slots is
always a prime number +1
Below the cut-off frequency depends on the depth of the deepest pit. Where: Shaft width =
lambda
Upper limit frequency depends on the width of the wells. Where:

Shaft width = 1/2 lambda.
One should the relationship between depth of the well and width selected so that you get a scatter
range of not greater than 4 octaves. The scatter should be selected not more than 4 octaves. In
practice :
RPG diffuser system:

Diffractes: broadband diffuser
3 QRD in a f = (340
/ 1.28) = 265 Hz Hz In a simple
QRD
Omnifusor: two-dimensional QRD scattered to the sides and down. Abfusor: combination of
diffuser and absorber and which is inserted in the receiving space Trifussor
Trifussor: combination of QRD, absorber and reflector.
Requirements for the control room:
Initial Time Delay Gap (Gap ITD) Delay time of 1.Reflexionen compared to the direct sound at the
listening position should be 10-30 ms. Size: space should be greater than 50 cubic meters be
(sufficient diffusivity) and ITD Gapvon 10-30ms). Geometry:
The space should be symmetrical and have no parallel walls. Reverberation time:
Page 433 of 466
The reverberation time RT 60 should lie between 0.2 and 0.4 seconds. In addition, should the
reverberation time in the middle frequency range between 200 Hz - to be 8 KHz consistently,
Ambient Noise: must not be more than 25 dB (A). Diffusion:
Monitor position: 0 - 7 degrees in height
Monitors should not be obscured.
LEDE concept:
Live End Dead End
In the front area where the monitors the sound is constructed are very absorbent. Front area
absorbing (avoid to short IT D- Gap) Rear range reflective (best with diffusers)
Reflection Free Zone:
The walls in the front part are angled so that the first reflections are derived from the back and only
from there to ITD Gap of 10 - 30 ms access to the listening position.
Requirements for the receiving space:

To the accommodating space requirements
avoid parallel walls. Standing waves, flutter echoes. Flexible room

acoustics
eg by Triffusor, with gobos. or with mobile

Wandpads.
Control room optimal acoustics

Receiving space depending on the music, more reflective or more absorption.
With a large reception space can LiveArea and Dead Area. Reflective and
absorptive area in the recording room. Room sizes for distant miking: greater
than or equal 40 square RT 60: for language shorter reverberation times. 0.3
to 0.6 s
for music
0.6 to 1.1 s
Page 434 of 466
mastering
PM Studio artistic Ttigk.
Mastering consists of premastering
mastering
Presswerk Technically
Always pay attention to dynamics Classic has more momentum
Pop and Disco tracks do not need much momentum
CDs must we write RedBook be compatible.

Clicks, noises and remove everything is made in the technical field
ie in pre mastering.
Bob Katz Mastering Audio good book
It is important to know: YOU HAVE GOOD MIXES GET.
YOU CANT POLISH SHIT.
Communication is very important - to exchange> consultation with the artists Dear
issues with the artists accept requests
obtain information.
take notes, listen Protocols lead on mastering steps.
How should the added sent Material:
formats
CD - data - Audio DAT
Tapes
Hard Disk
DVD
No master tapes / Endmixe
Master tapes have next to the audio and the PQ data on it. Data includes information about the
track length over track start and end of how long are the breaks, how many tracks are on it, and
different codes. Only tapes have the PQ data master tapes, all others are master tapes.
SPECIFIED date of bands, from whom, studio production. phone number
Sound engineer, producer,

Technical specifications': How much word width of the audio signal
Sample rate of the audio track name

with the length.
Page 435 of 466
Usually one makes a template, where all things are on it, then you need only add
things.
Analog Tapes: necessarily play: level tones: our system needs
be played at the same level as there. System

is calibrated.
We need the same level as the mixers appropriately has.
Specifying the level sounds important
100 Hz, 1 kHz, 10 kHz. at 0 dB on

the VU - meter
Dispensing the tape speed:
15 ips
38 cm / s
30 ips 76 cm / s
At a high belt speed has a high signal to noise ratio. Dolby SR Spectral Recording proffesional
noise suppression system DBX
an other proffesional System
TELCOM for broadcasters, inch must be the
tape for mastering
Pop music and disco music one does not need noise suppression system. as like classical
because more momentum. ribbon colors Red Ribbon Mono
Red White Stereo Red White Black stereo

TELCOM
Mastering Format:
Mastertapes include PQ data. TOC Table of Content U-Matic PCM

1610, 1630
PMCD Pre Master CD must have high quality Mitsui Exabyte DVD Audio Multichannel
sound DLT Tapes (Digital Linear Tapes)
Umatic (PCM 1610. 1630)
This is the first format which has the digital interface allows.
Page 436 of 466
Consists of a VCR and a processor. Converts the analog signal into a Pulse Code
Modulation, transform.
VCR
PCM
The PCM signal has a very wide frequency range. Records

with Longitudinal.
Super susceptible system belts cause problems. Basic
structure written to tape header in these basic structures will
be written in. The tape is from front to back with timecode
everything is in real time.
Specification of Umatic:
The tape format: U-Matic Professional NTSC standard 30 fps.
NTSC standard 30 fps Non Drop Frame Drop

Frame 29.97 fps. PAL 25 fps 24 fps film
coding:
linear encoding two's complement

Dickreiter daily, Wikipedia.
linear format Sony PCM 1610, 1630 Emphasis,
de-emphasis to get a better signal to noise ratio.
Deemphasis is performed in the CD player if
Emphasis is detected. 15 micro second Emphasis 50
micro second Emphasis control times of Emphasis
number of channels: 2
Modulation: The entire audio content including Lead In and Lead Out
must be written down.
Lead In:
Before the modulation begins allowed to empty an area to do this for

security reasons. Most 2 minutes idle
then begins to audio.
The must not be shorter than 30 seconds. This Lead In area, the PQ
data is written. On the analog track is written. 2 tracks Digital 2 Analog
tracks.
Page 437 of 466
On the first track, the PQ codes, on the other track of the time
code signal is on it. Timecode must be continuous and counting
upwards. That must be from front to back.
Synchronicity must Phase Coupled be recorded for sampling

frequency
The time code must have a level of 100 +/- 50 nanoWb / m Weber
per meter is the unit for the magnetic flux. We can 50-150 disqualify
our level.
Sonic Solutions was the first digital workstation for mastering: Sadie DAW
for mastering. Hardware, associated software. Internal 64-bit processing.
all processes running on the processors.

Fewer losses during audio.
Red Book says that the track length but not less than 4 seconds. The pause length
with changing preemphasis He needs time to adjust.
When changing from pre-emphasis on non Emphasis must be a pause of seconds.
Very first break must always be 2 seconds. Our digital headroom

must not be below -2 dB FS.
Normalize is the tips to 0 dBFS bring each other in the same proportion. Limit is adjacent the
dynamics. Normalize to 0 level by 0.1 down.
Desire: 24 bits are a must and 44.100 KHz.

48 KHz to 44.1 KHz
badly
quantization
88.2 KHZ to 44.1 KHz
no problem no
quantization. Redithering at the very end runterdithern to 16
bits. When working with heights better analog EQs Surgical
better with a digital EQ. Red Book says 99 tracks may be on a
CD. Subindex may again be only 99th
Page 438 of 466
How long is a CD of the season Single CD

80 mm 8 cm Normal CD 120 mm 12 cm
maximum playing time
78 minutes 30 seconds
Single CD 22 minutes maximum playing time. BLER
block Error
with more than 220 errors that must be corrected
in the second. not readable.
220 C / s
According to Red Book it should be less than 220 per second correction,
Solopress does not accept CDs with more than 50 c / s. E32 System can not
correct the error.
System makes, Mutes we hear non> Drop Out
C1 (if not corrected error) continue to C2 E 32 after the second Corrector he do to because
no third Correctorblock exists. BERS <5
burst Error
these are errors that can not be corrected. According to Red
Book, they must not be more than. 5
Exabyte
looks like a small Hi-8 cassette

pure data transmission. Write in Double Speed in double
speed and read. PQ data hot DDP Files Wikipedia Disc
Discription protocol here. The things are going with time
saving formats must be 8500 EXB compatible format is
uncompressed recording. Data rate is at least 400 kilobytes
per second. Maximum approved tape length is 112m. fits it
over 80 minutes.
Page 439 of 466
vinyl record
Recording on record: 1877 Thomas
Alva Edison
Record was then a wax cylinder.
subscript
Gramophone records - 78 rpm / revolutions per minute N 78 1888 Emil

Berliner
invented the Page signature.
1931 Alan Blumlein cross font

Jump from the mono recording in the stereo recording coils above the
needle for L, R channel
45
45
Even then equalized from 500 Hz 6 dB per octave. Today LP

rotates with
33 rpm
Single disc rotates at 45 rpm

The voltage at the head is inversely proportional to the signal voltage at low frequencies.
Normal groove: 120 microns wide

Rillenverrundung: 45 microns wide
Tip rounding the needle wide at 65 microns.
Micro groove:
in mono recordings 55 microns

Rillenverrundung: 4 microns with stereo
recordings: 40 microns
Rillengrundverrundung: 4 microns. LP M
ST
33
single
33
M ST 45
45
small plates have 7 inch 18 cm Maxi
have single
10 inches 25 cm
Page 440 of 466
have LP
12 inches 30 cm
Made a record:
2 cutting process
1.Lackschnittverfahren :
Modulation of grooves made on a nitrocellulose lacquer. Resist layer on plate. the grooves are
engraved on this paint with a hot specialty wire from outside to inside.
paint film
is made and electroplated electrically. It depends a copper layer. 1 copy of the paint film
is the father, producing several fathers.
several fathers more mothers are made. Paint copy (positive)

father (negative) Mother (positive) son (negative)
In a panel's there is a signal to noise ratio of 35 dB. Mix coal dust to to the signal noise
to improve distance. The board held longer.
2.DMM Direct Metal Mastering

Invented by the company cut TELDEC in the 80 he direct years
on metal,
With a diamond coating is engraved. Achieve a signal to noise
ratio of 60 dB. Shall sound very good.
On a steel plate with 8mm thickness, an amorphous layer of
copper is upped. The diamond cuts the groove from outside to
inside.
Page 441 of 466
Sectional characteristics:
Low frequencies can destroy our paint film.

At low frequencies, the groove can be so small that the needle out of the groove jumps out.
+ 20
seconds
Characteristic in the records
50 Hz 3180 micro
500 Hz - 318 micro seconds

2120 Hz 75 micro second
Amplifier to compensate
+ 10
0
- 10
Characteristic on the plate
- 20 0
50
500 2120
15000 Hz
Everything happened in recording before the needle

Everything happens when you play for a needle How is the
average characteristic defined: 50 Hz 3180 micro seconds
500 Hz - 318 micro seconds 2120 Hz 75 micro second
78 rpm disc has a diameter of 25 cm - 30 cm Single is 17.5 cm

tall
LP can be up to 27 minutes on a side.
For a private radio Mastering:
It is primarily about the loudness.
Good Maximizer leave despite heavy compression a certain
momentum left.
Page 442 of 466
Monitoring unit:
consists of stereo vision

device
Correlation Meter
Compatibility meter
Show us with stereo recordings, the instantaneous phase difference between the coherent
signals of the two stereo channels left and right as Korerelationsgrad on.
Ratio is displayed in degrees - -1

Coherent:
The signals have the same source.
It nevertheless phase differences may emerge. The left and right signals are
coherent when they come from the same sound source.
It can frequency-independent level and delay differences occur. Even if the
signal is phase-rotated by 180 degree by Lin and right it can be coherent
anyway.
Kolle ration of a stereo signal the degree of relationship of left and right is independent of the level.
We must always make mixtures Mono Compatible.
Compatibility: A signal is then compatible if in its stereo playback the recording sounds not
noticeably worse than a resultant under comparable conditions mono recording.
When the signal to 0 degree, it shows that we have no signal. No signal,

incoherent signals or a channel missing. at +1 If then it shows that the signal is
mono. It is a mono signal.
-1
+1
broken
signal
no
signal
Mono
Channel
polarity.
In mastering a record sure that everything is 200 Hz Mono. Then pay attention that at 10
KHz everything soft sounds.
Page 443 of 466
Stereo vision device
shows us the intensity and the stereo width.
material
DAW EQ MBC eg lower 120 Hz and 3 dB. Bass clearer

Limiting
Once you Converts then then one last time convert and remain analog digital in an
analog chain as, limiting the end.
1. listen to
. 2 do check with the artist and notes
make picture of dynamics, sound, correlation
. 3 Edit with effects and EQ,

. 4 At the end of the DAW make cuts and breaks.
a. Cuts always make the zero crossing.
b. listening watch on long Fades always on headphones Zoom
whether phases agree.
let 9 Frames position at the beginning, 0.3 seconds at Offset setting (marker) At the very end
in the DAW Redithern .--> 24 bits to 16 bits. Dumping is the process of burning.
Page 444 of 466
Music Business
author
Spiritual author of a work
artist
Artist / Performer, one who embodies everything on stage
A&R
Artist & Repertuare
Ensures that new songs come. Care for the band. Contact the band.
Back Catalog reference or computer catalog

Lists all Unfortunately and performers at a record company.
Tape transfer agreement:
Band remains a higher royalty rate in production. Only the Master

gets the record company.
Artists Treaty:
Artists heard the record company. Production part of the record company. Artist
gets ultimately less profit rate.
Processing: An amendment requiring approval of an original.

Rock version or hip hop version.
Music you should not change without approval processing.
boat paint
Illegal live recordings of live concerts Bootlags
Catering:
Supply the artist with food and beverages
Cover: Last requiring approval variant of the original,

requires processing authorization.
CI:
Corporate Identity
Uniform production identifier, logo, sign

CD:
Corporate design, extended to the Internet
hyping:
semi-illegal positive influence on the success of one artist. Through Acquisition

of own panels.
Artist Agent:
In the name of the artist through the counter runs and worried concerts, but no
signature has permission.
Page 445 of 466
Artist Manager:
1.Business Manager
Communication with companies and sponsors, etc. advertising

company, not Unterschriftbevollmchtigt.
2. Personnel Manager
Caregiver of the artist, Unterschriftbevollmchtigt.

Artists secretary:
Employee / he the artist gofer.
Lead Sheet: piece of paper, harmony structures of a song that everyone
Musicians can play.

Lable:
Smallest version of a record company, making music and selling CD's
Merchandising:
Merchandise sales, artists receive a percentage of it.

Option:
Prerogative of contract after a certain contract period. Franchise contract

extension.
Parody:
Humorous version of an original.
Performing Fees:
Fees for Self

plagiarism
A copy of the original, plagiarism
Plastic Strap: If someone stands on the stage but nothing of the piece of it is.
Not even the vocals.
Promoter:
Advertisers for a band that makes advertising for a band
Pos:
Points of Sail. sale
Poi:
Point of Information. Posters, displays, stickers localities
Shops, displays, large display
Pseudonym: code name of an artist or author robbery Pressing: A publication of a
phonogram, in violation of
Copyright. eg same CD printing presses in Poland
New production of a track with samples from the original processing
required authorization.
refund: Payments of Gema. Royalties from the publisher to the author. Returns:
Remix:
Return of unsold recordings from the record store to the sales.
Page 446 of 466
Rider:
contract Annex
Annex of the technical equipment.
Technical Rider:
Roadie:
Technicians live with the band on the road is.

Specializing in a particular instrument or territory. Light, PA,
Guitar (backline)
Road Manager: or tour manager:

Person in charge of the tour of the artist. Knows: Hotels,
Stages, organizer etc.
royalists:
Percentage of shareholding of CD sales.
Semi Proffesionals:
People can not live 100% of the work as a sound engineer or musician, part time.
Signing Fees: funds the artists at time of contract in addition as confidence Bonus
get.
Sleeper
sub-publisher:
If a hit after a certain time is only a hit. due to a certain reason.
Partner of original publisher mostly abroad.
Traditional: A Gemafreies work what you can edit freely.

where the author is at least 70 years dead.
Transpose: Transpose song in a different key. Variation:

Requiring approval modification of a variant of a work.
Organizer: Local organizer: organizer for a local event.
Tour Operator: contract with an artist over several

Events.
scrapping:
Distribution:
Remaining copies of a phonogram is sold in a strong price reduction.
Company brings the plate from the press shop on behalf Releases into the
record.
Publishing company:
Is the company with the composers and lyricists
cooperates. Intellectual copyright.
Page 447 of 466
Artist Manager demo package Record label (A & R Manager) Record (A & R Manager) (Take
artists or not) Meeting Publisher listen to demo Publisher angagiert composers and
lyricists Publishers done songs
Song 1,2,3 be taken.

30% Composer, Lyricist 30%, 40% Publishing Distribution
Budget 20,000 Studio Producer musicians Mastering Promotion
20,000 Studio cost of 3 songs. Studio per day

800 9 days 7200
1 day
2 day
Day 3
Song structure, click Guide Track
day 6
Drums and bass of the 3 songs are recorded.

draufspielen guitars on Arrangement
record vocals Arrangement
Backing vocals, guitar overdubs
7/8/9 day
Every day a mix of three songs
4/5 day
For 3 songs: Producer:

2700 700 Mastering
External Remix ...
1,500
sound engineer
3600 9 days a 400 Euro
Musician
Drums 600
500 Bass
2 days guitarist 2 x 500 3 days
keyboardist 3 x 500
4 days male and female backing vocals 350 each
Page 448 of 466
Executive producer: company, donors, sponsors Producer
Music and Content Responsibility Responsible for

compliance with the budget.
Line Producer:
Financial controller
monitors their progress of production.
Radio promotion
Instore promotion
Promotion:
TV promotion
PrintPromotion
cost about 500,000 - 1 million to commercialize an artist professionally.
Society for Musical Performance and Mechanical

Reproduction.
GEMA :
paid composer, writer, publisher and editor. Headquartered in

Munich and in Berlin.
PHO VR clearing of Phonographic Reproduction. Discos, pubs

businesses
Companies with music on hold,

Radio, TV, Internet
Live events PHO-VR
GEMA
composer
60%
30%
writer
arranger
publisher
40% for Instrumental

40% for normal song
30%
If the agent also integrated in Gema that 12tel rule: Composer 4 / 12tel
Texter 3 / 12tel
Editor 1 / 12tel
Publisher 4 / 12tel
GEMA abroad
USA:
BMI / ASCAP
England
PRS (Phonographical Write Society)
Holland
BUMA / STEMRA
France SACM
Page 449 of 466
Hardware manufacturers have to pay a GemaPauschale, ZP revenue:

800000 Gema fee / 13% revenue Gema 50 admission fee 25 annual
fee per member 3 types of membership:
1.Angeschlossenes Member
at least 5 songs published
2.Auerordentliches Member
3. Full Member
30,000 based within 5 years.
GEMA does not protect our things, is responsible for only the revenue,
Protect can be the stuff at the notary. The CD

store, he makes a certificate.
GEMA differs between Large and Small legal rights. Large law:
Even as author with an organizer arrange the revenue for an
event
Small law
GEMA takes over the revenue regulation,
GEMA member only as a composer and lyricist.
A band can not become a member, only the individuals in a band. As GEMA members are
allowed to do no GEMA free music. Music may be changed only with the consent of the
originator. GEMA differentiates between serious and popular music. E-Music: film music,
etc. U-Music: music charts, etc.
E-music gets 12-24 times higher payment than Light music GEMA income subject
to income tax. GEMA Music is subject to the author's death 70 years GEMA.
GEMA to the artist a pseudonym. Various billing spade:
Page 450 of 466
Performance rights public performance by musicians

public performance by phonogram public performance
by loudspeakers public performance of color films /
Display
1)
2)
Radio Television
reproduction right
broadcasting rights
1.Film
2.Video
3. broadcasters
4.Private duplication,
Rental.
GEMA protected in addition:
- Blank tapes, CD Blanks,
-Rental of video films
-Rental by libraries
- Internet
Billing Information:
Once a year Metro
Music
April 1
E-Music
1st of May
Radio and TV
July 1
October 1
Loudspeaker reproduction
Movie and TV 1 July with Nahverrechnung on October 1 video carriers

1.january
PHO-VR
January 1 with Nahverrechnung on 1 April and 1

June mN on November 1
Page 451 of 466
Business
GEMA: collected money where the company wants to earn money through the value of music. PHO-VR
Label must pay to GEMA, the number of production. Only by pressing.

100000 Singles * 40 Cent
LP
* 80 Cent
DJ, transmitter, TV need to fill a list of songs they play for GEMA GEMA is only responsible for the
intellectual authorship.
Green registration form of a piece with GEMA : No.1 Work Title /

song name No.2 genre / style of music
No.3 playing time (minutes / seconds)
No.4 whether other authors must be considered

Author can decide to whom the copyright is owned processing
authorization must be added to you. No.5 Composer is entered No.6 Cast
Instrumental Instrumentation in pop, rock, classical
No.7 Lyricists
b) Another Testierung in other languages
No.8 Verlagsangabe the publisher No.9 Editor Note
printing No.10 subtitles
Title with additional names so that the Gema assigns the.
No.11 phonogram with artist signature and

title is registered.
Member Login with GEMA:

Request to join Author
Contract you have to fill whom one wants to be GEMA member. 50
admission fee, 25 annual fee
1.
First name and surname
.2
Address / Tel.
.3
fiscal domicile
Pseudonym Pseudonym
.4
Page 452 of 466
.6
Date of birth
nationality
.7
Main and secondary activity
8th.
Membership of a collecting society

Details of the training.
.5
.9
5 criteria to GEMA to be a member: Login GEMA that GEMA

money has what belongs to you.
1.The following works have been published in print
2.Folgende works have been published on sound recordings and video carriers
3. The following works have been performed in public.
4. The following works are on the radio, was broadcast on television.
Music has been written 5. For following talkies.

GVL
founded Society for Recycling and performance rights
Vinyl Foundation to compensate for the live musicians as the live
music broke through the studio era.
GVL
Electing the musicians relinquish their rights for a piece, ancillary copyright cede
for a production get for each piece that they have co-founded refunded with a
voucher for 30% of their money.
One can
producer
Musician
sound engineer
GEMA
GVL
power money
be
a person can do anything and get paid for everything, write invoices to itself
einquittieren. To keep an overview. SCHIZO
GVL assigns a LABEL CODE to record producers.

If the phonogram have no label code, it may not be broadcast on the radio. But it must be done a
broadcasting contract. GVL pay money to the Tontrgerhesteller.
if a label has 3 Acts LC multiplied by an Act of the label to the time and duration Distribution.
GVL phonogram
GEMA composers and lyricists, publishers ZP

(Office for private dubbing right)
Page 453 of 466
pay GVL
GEMA
Artist
Percentage of the CD, Honorary sold
guitarist
purchased, authorized payment of GVL, GVL
Composer
lyricist
GEMA, copyright
Contracts:
3 contracts:
Contracts decide: Who heard the

name? Band is a company GBR
Everyone has to pay for the other in a GBR. GBR contract must
proceed at a limitation of liability, it limited the contract to a specific
area.
Shareholders' agreement of a musical band with limited liability
-Shareholders HIDE for operating a pop group.

- Persons listed with indication of all members
- Establishing a music group with limited liability. Giving the name,
place of residence, Perso-number etc.
GBR, GDBR civil law BGB Civil Code 1 Name, state, subject
Purpose of the band, Who owns the name, What's on dissolution of the
band called In games of a member in another band then ..what?
Agreement that all are satisfied. 2 Duration
Treaty with today's date to a specific time 3 posts
PA is one whom Transport

accepts who? Technicians etc
......
Instruments financed who and how .... 4
Representation:
Banking, who will take it at least 2 shareholders

have represented,
Page 454 of 466
regarding making 181 transactions with themselves and persons below 5 Profit and
loss
How is income classification regulate
6 Vote
Unanimous decision of decisions eg eviction, take care in Family
Alliance ... voting 7 Termination of Membership
Disbanded
The Company is not dissolved by the withdrawal of a partner. If someone goes out
his profit is divided among all. The death of a member do not occur, the heirs as a
member. 8 competitive activities
If band members in another band playing conditions
must be agreed upon. 9
Other, Law and Jurisdiction: Where you do not go to court
in compliance. Verbal agreements have been made, Any
changes must be in writing. What is not written, does not
count. Severability Clause:
Should certain terms of the contract null and void or unenforceable shall not affect its so will
the validity of the remaining provisions of the Treaty.
The parties will be void or ineffective provisions replaced by other provisions, which
economically hedge the original purpose.
HIDE for a personnel manager:

Management contract:
1 Name and purpose of the subject,
Manager performs the following tasks, preparation of contact with the

media, establishing contacts with the organizers must be noted that the
manager must do everything. 2 proxy
Artists must the manager the authority to give the order Signing can. With veto
right of withdrawal
Contract with veto power:
Within 10 days of nothing negative belongs, then the contract is.
Page 455 of 466
3 Remuneration:
Manager usually gets 15% - 20% remuneration. In some

industries, the managers get 60% because they have invested
everything. 4
Running time:
Runtime length with option
Option 1 year before the Treaty extension, earlier, later ... 5

Others:
,,,, Signature.
Artists Treaty:
1 Name, purpose, object

Artists ensures songs,
Record company takes over the production
and the composition etc ... 2 excludable:
exclusivity:
The artist will sing only for this company, exclusive bond over
adhesion contract 3 Recording and publication:
Per year
so many LP, CD
2 video clips
who looks for the producers who

book the studio
when artists his song from its fault does not finish, he must pay the
production itself. 4 Transfer of rights:

All recordings with their creation the company include artist. get percentages.
unlike the tape transfer agreement.
5 Guarantee obligation:
Artists must respect this treaty guarantee that he hindered by other commitments. 6
Remuneration
What percentage gets the artists of publications, samples ... advertising etc.
7 Billing and Payment;

Every six months is charged and within the next 3 months will be paid.
Page 456 of 466
8th
Power disturbances
alcoholism
Drug Addiction Artist
Things restrict its performance, writes bad songs contract is terminated. 9
video carriers
Artist poses in their own video clips as an actor
available 10 Contract
Contract with the associated options 11 Governing
Law and Jurisdiction
Salvatore clause 12 Other

agreements
Group clause:
If one wants to go out in a group, he must wait until his band contract expires, unless it comes
to the unification of the record companies by compensation percentages, fees, etc.
A normal studio construction has cost approximately 27,500
to build an established studio approximately 95 000 A normal

club has monthly costs of about 1200 An established club has monthly
cost of about 9,400
Page 457 of 466
formulary
Physics - Acoustics
Phase angle (phase)

-
0 and 180 in the passages through the rest position

90 and 270 when reaching the reversal points
General vibration equation of the harmonic wave is used for calculation of the
elongation at a given time [t]
AY
tf + 0
sin360
( =
Instead of 360 can be used 2

Damping ratio of vibrations
A3
A2 = A3 = A4 = K = damping ratio
A1
A2
beat
1 fff r +
2
One hears a volume variation, the frequency of the beat frequency (f s) is
specified.
fs= f1 - f2
Example:
fr
f 1 = 444Hz
f 2 = 440 Hz
= + = 442
2440
fs = 444-440 = 4 Hz
444
velocity of propagation
Depth frequency (f) = high wavelength ()
c=f
High frequency (f) = small wavelength ()
Example: c = 343m / s, f = 150Hz, =
150 343 =
29
2,
mm
Page 458 of 466
Sound velocity (V) [m / s]

Specified as an effective value
Xeff
Vrms
2
v
general
for V
Sound pressure (p) [Pa] (Pascal)
Pressure fluctuation between condensation and rarefaction in the propagation medium. An alternating
pressure is superimposed on the atmospheric air pressure. 1 Pa = 10 Bar
Supersonic flow (q) [m 3 / s]
Air volume that is moved in alternating directions within 1sec an area of 1 square meter. Q = v S
(Sound velocity surface)
Inverse Square Law
A doubling of the sound pressure represents a quadrupling of the sound intensity (to the square).
In a spherically radiating sound source a doubling of the distance results in a halving of the sound
pressure, but only of the sound intensity.
J1 ~ r
~ rp 1
r = distance from the sound source
Multiply and divide using powers 100 * 1000 = 100000 * 10

10 = 10 5
Multiplication of powers with the same base can be replaced by adding the exponent 1000:
100 = 10 10: 10 = 10 1
Division = Subtraction
Page 459 of 466
Power laws 0 0 =
n
0=
1
+ NMNM
= aaa
:
1
= aaa
-
= aa
NMNM
nn
10 10 1-
20
aa
nn
10 10 10
- 2
31
(N> 0)
27
= 27 = 3
Logarithm (logarithm)
n=
10
= log
10
bn
100 = 10
2
10 log10=100 log= 2
10
10 3
10
= =log
= 1000 log1010 1000= 3
Logarithmusgesetze
[
lg10
n=
n]
10 lg 1
1 lg 0
==
lg (
lg+ lg NMNM
10 lg100
(
lg+10 lg100
= lg 10 lg+1000100
lg
1
lg (
= lg 10 lg +10 lg 10
100
lg 1000=lg lg 100
- 1000
= lg- lg NMNM
lg=10 lg -10 lg 10
lg 100 lg 1100
0
(1 lg)
=- lg nn
=
lg 10
= lg
- 1010
- 2
lg =n lg unu
=lg 10- lg 10
lg 2 lg 10
102
=
Page 460 of 466
Level (Level)
lg
100
mW mW Bel
L
Bel
1
Bel
= =1=Bel
10 lg
10
lg 10
mW mW LdB
lg
10010
10
= =10
dB
General level formula

=
10
lg L
xx
dB
mW
10 mW L dB
2 lg1
lg 10
2 dB
= =3=dB
For power and intensity levels apply: doubling

the value of the size:
Tenfold increase in the value of Size:
Halving:
1/10:
+ 3dB
+ 10dB
- 3dB
- 10dB
relative level
=
10lgxx L dB
X1 is here the reference value
The reference value is not fixed, it can be freely selected. Used to display
resizing.
Absolute levels
eg
10
lg 10
1
mW mW
dBm
The reference value is set.

dB is supplemented by an additional, indicative of the USed reference value. here m
Page 461 of 466
Sound power level

=
10
lg p
PPL
relative level without fixed reference value
dB
SWL
WP dB
lg 10
10
- 12
10 -
The reference value
12
SWL
absolute level with a fixed reference value
W is the sound power, which must account for 1 square meter area, thus
on
1 kHz tone is just audible. This is specified in
dB wherein
SWL for Sound Power
SWL
Level is. Ex .:
How large is the level difference between

2
LP
8th
W and
-2
10 2 W
WW
) dB
=
(10
10 2 10
lg 5- -8
= lg 10 10 4
10 5
dB
= 56 dB
Which sound power level corresponds to the maximum sound power of 10W Timpani from?
L SWL =
- 12
lg 10
10 10
lg 10
=
10 1
1 10
= lg 10
= 10
= 130 dB
WWdB
WWdB
-1
SWL
SWL
13 12
dB
SWL
SWL
Electrical power level character:

P
Unit:
=
10
lg P
PPL
Indication of relative level
dB
mW PL P
lg 10
1
dBm
The reference value corresponds to 1mW
Summary for absolute level
10 - W
This is specified in [dBm] (m represents mW)

Page 462 of 466
An amplifier delivers an electrical output of 40 W. To which electrical power level in dBm, this
value ????
LP =
mWdBm
W
40110
lg
lg 10
=
10 4
1 10
- 3
WW
dBm
= 46 dBm
Sound intensity level
sound intensity
10
lg J
JJL
SPJ
Level by area
m W]
2
Indication of relative level
dB
mw JL
dB
- 12 2
10 lg 10
Summary for absolute level (SIL = Sound intensity
SIL
level)
reference value
10 12 m
is the sound intensity at which a 1 kHz tone is just audible. The
Disclosure made in SIL
efficiency:
Sphere:
ak
PP
el
4=r
O
Sound intensity level:
L SIL =
100%
lg 10
10
m WJdb
SIL
12
2
SPL
P=
zp0 2
P = sound power p =
sound pressure
z0 = acoustic impedance sound

pressure [p]
Sound power [P]
linear size
square size
Page 463 of 466
20
lg p
PPL
relative level
dB
dB SPL
L SPL
lg 20 10 2-
Pa P
dB
-5
10 2 Pa is the sound pressure, in which a 1 kHz tone is just audible.
reference value
Specified in dB SPL
Voltage level (dBV and dBu)
RUP
P = electrical power [W] U =

Voltage [V] R = resistance []
dBV
=
20
IVlgPL U
dBV
Reference value is 1 V
dBU
LU =
20
0 lg , 775
VP dBU
Reference value is 0.775 V
Umrechnug the level in the corresponding physical quantity. For details in Bel
10
10=
XXXX
1
For Quadratic sizes (power, intensity)

L
10
10 2 1 2 10
10=
XXXX
1
For linear sizes (sound pressure, voltage)

L
10
20 2 1 2 20
10=
XXXX
1
Digital level (Level Full Scale [LFS])
FS
=ML n
2 lg 20- 1
dB
FS
Page 464 of 466
M = number of voltage levels N = currently used in

quantization bit
Units:
10- 12
10- 9
10- 6
10- 3
10- 2
10- 1
Pico
Nano
Micro
Milli
centi
Dezi
10 12
10 9
10 6
10 3
10 2
10 1
Terra
Giga
Mega
kilo
Hekto
Deka
Page 465 of 466
Reference values Acoustic

size
designation
reference value
Sound power level
Electrical power level

Sound intensity level
10 -
12
dB SWL
10 -
dB m
10 -
12
SPL
2 10
voltage level
log
10
dB SIL
- 5
dB SPL
dB V
0,775
dB U
20
Sound power level (dB SWL)

10 -
The reference value
12
W is the sound power, which must account for 1 sq.m.,
so that a 1 kHz tone is just audible. This is specified in
dB wherein
SWL
SWL
stands for Sound Power Level.
Electrical power level (dB m)

The reference value corresponds to 1mW
10 - W
This is specified in [dBm] (m represents mW)
Sound intensity level (dB SIL)
reference value
W
10 12 m
is the sound intensity at which a 1 kHz tone in hearing
becomes. It is stated in SIL
Sound pressure level (dB SPL)
reference value
-5
10 2 Pa is the sound pressure, in which a 1 kHz tone in hearing
is. Specified in dB SPL
Page 466 of 466

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Available Formats

-- Tontechnik Kompendium--

Sequencing 1 ____________________________________________________________ 7 HD: Peak (Mac OS X)

Physic Basic ____________________________________________________________ 25

Level invoice: _________________________________________________________ 39 psychoacoustics:

HD Recording Basics (ProTools) ___________________________________________ 54

Tape machine __________________________________________________________ 58

Grundlagen________________________________________________________________ 62 air pressure and water

Instrumental Acoustics: _____________________________________________________ 81

Music Theory: ___________________________________________________________ 87

The circle of fifths: __________________________________________________________ 88

Microphonic 1 __________________________________________________________ 115

Sequencing 2 __________________________________________________________ 162

Sampling: 174 _____________________________________________________________ MTK HD Recording

Sound synthesis _________________________________________________________ 207

Subtractive synthesis _______________________________________________________ 207

Additives Synthese__________________________________________________________ 213 resynthesis

Research______________________________________________________________ 215 Session Procedure

Signal Flow 3 - Studio 2 - D-Control ______________________________________ 233 236

Rundfunkton __________________________________________________________ 266

Signal Flow 4 - NEVE 88 RS _____________________________________________ 282 film and

Automation____________________________________________________________ 315 E-2 technology

Coil / coil / inductor 325 ______________________________________________

Inductive Blindwiderstand_______________________________________________ 327

Coil to AC ________________________________________________________ 327

Netzteile__________________________________________________________________ 349 Transistor

Speakers __________________________________________________________ 369

Live art __________________________________________________________ 380

Multichannel _________________________________________________________ 398

Multichannel Mifkrofonie _________________________________________________ 404

Production 2 __________________________________________________________ 412 DVD authoring

Building and room acoustics: __________________________________________________ 421

Mastering _____________________________________________________________ 435

Formate__________________________________________________________________ 435 record 440

Music Business ________________________________________________________ 445 Business

Previously: -low memory

MIDI Music Instrument Digital Interface interface for electronic tone

(Sends control data)

A flow of information from here to there.

Midi cable wiring runs in one direction MidiMessage "Note

Computerized Midi Sequencing

homophone Equal Voices sounds are played simultaneously.

Sockets and plugs standard 5 pin DIN cable 3 of 5

Input for MIDI data

Output for the device generated data

Loop jack. 1: 1 copy of the present

linear MIDI / audio sequencer. Works with

File format of Logic document .iso = Logic Song Logic Pro

Application settings, preferences (complete program)

Song Presets, Documents settings

The main window Arrangefenster

Tracks so-called tracks Toggle Turn

2 * 0 Advances with SPL to the beginning

Space = Enter, Stop, Enter

SPL is movable with mouse

1/3048 assigns Midi files to the clock. Locator:

Is the marked section not again. Esc: - tool list Select

click track region is created in the selected track

Edit Matrix Editor or apple 6 color of the attacks

V Changes the velocity of sound function Quantization:

Double click on the track gives me the Mixer View Object

HD: Peak (Mac OS X)

zoom in, zoom out

Apple + Shift + (-) =