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TC Electronic Broadcast 6000 User Manual

Page 52

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52

paGe Head

rOOm sImulaTIOn fOr mulTICHannel musIC and fIlm

• The system should not be limited to simulating natural

acoustics: Often quite unnatural reverb effects are

desired, e.g. for pop music or science fiction film effects.

• The system should be able to render the simulation via a

number of different reproduction setups, e.g. 5.1, 7.1,

stereo etc.

• The system should be modular so that new rooms, new

source positions in existing rooms, new source types or

new target reproduction setups can be added with

minimal change to existing elements.

• The system should be easily tuneable: In our experience,

no semi-automatic physical modeling scheme, however

elaborate, is likely to produce subjective results as good

as those obtained by skilled people tuning a user-friendly,

interactive development prototype by ear.

Fortunately there are a few factors that make the job easier

for us:

• There are no strict requirements for simulation accuracy:

Certainly not physical accuracy (the sound field around

the listener’s head), and not even perceptual accuracy

(the listener’s mental image of the simulated event and

environment). The listener has no way of A/B switching

between the simulation and the real thing, so only

credibility and predictability counts: The simulation must

not in any way sound artificial, unless intended to, and

the perceived room geometries and source positions

should be relatively, but not absolutely, accurate.

• Moore’s Law is with us. The continual exponential growth

in memory and calculation capacity available within a

given budget frame has two effects: It constantly

expands the practical limits for algorithm complexity, and

it makes it increasingly feasible to trade in a bit of code

overhead for improved modularity, tuneability, etc.

• There are physical modeling systems readily available,

which may provide a starting point for the simulation.

3.2 Block diagram
The overall block diagram of the Room Simulator is shown

in fig. 1. As often seen, the system is divided into two main

paths: An early reflections synthesis system consisting

of a so-called Early Pattern Generator (EPG) for each

source and a common Direction Rendering Unit (DRU)

that renders the early reflections through the chosen

reproduction setup. And a Reverb system producing the

late, diffuse part of the sound field. Note that - contrary to

what is normally the case - there is no direct signal path.

The dry source signals are merely 0th order reflections

produced by their respective EPGs. In the following, a

more detailed description of the individual blocks is given.

3.3 Early Pattern Generators
Each EPG takes one dry source input and produces a

large set of early reflections, including the direct signal,

sorted and processed in the following “dimensions”

• Level

• Delay

• Diffusion

• Color

• Direction

The Level and Delay dimensions are easily implemented

with high precision, the other 3 dimensions are each

quantized into a number of predefined steps, for instance

12 different directions. Normally, the direct signal will not

be subjected to Diffusion or Color. The quantization and

step definition of the Direction dimension must be the same

for all sources, because it is implemented in the common

Direction Rendering Unit. Physical modeling programs such

as Odeon [1] may provide an initial setting of the EPG.

3.4 Direction Rendering Unit
The purpose of this unit is to render a number of inputs

to an equal number of different, predefined subjective

directions-of-arrival at the listening position via the chosen

reproduction setup, typically a 5-channel speaker system.

Thus, the DRU may be a simple, general panning matrix,

a VBAP [2] system or an HRTF- or Ambisonics-based [3]

system.

3.5 Reverb Feed Matrix
The reverb feed matrix determines each source’s

contribution to each Reverberator input channel. Besides

gain and delay controls, some filtering may also be

beneficial here.

3.6 Reverberator
To ensure maximum de-correlation between output

channels, each has its own independent reverb “tail”

generator. Controllable parameters include:

Reverberation time as a function of frequency Tr(f)

• Diffusion

• Modulation

• Smoothness

We take particular pride in the fact that our “tail” can

achieve such smoothness in both time and frequency, and

that modulation may be omitted entirely. This eliminates the

risk of pitch distortion and even the slightest Doppler effect,

which tends to destroy focus of the individual sources in a

multichannel room simulator.

Again, an initial setting of Tr(f) may be obtained from

Odeon.

3.7 Speaker Control
This block is by default just a direct connection from input

to output. But it may also be used to check the stereo- and

mono compatibility of the final simulation result by applying

a down-mixing to these formats. Also it provides delay- and

gain compensation for non-uniform loudspeaker setups,

which may also - as a rough approximation - be used the

other way around to emulate non-uniform or misplaced

setups and thus check the simulation’s robustness to such

imperfections.

4. CONCLUSION
The system described above is evidently a very open

system under continual development. At the time of writing

these words, our test system is running in real time on a

multiprocessor SGI server with an 18-window graphical

user interface providing interactive access to approximately

2000 low- and higher-level parameters. However, this is

not the time or place to go into more details. When this

paper is presented at the 107th AES Convention in about