Design Guide for SoundStructure C16, C12, C8, and SR12
iiPolycom, Inc.
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iii
Design Guide for SoundStructure C16, C12, C8, and SR12
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
10
Introduction
The Polycom SoundStructure™ products are professional, rack-mountable
audio processing devices that set a new standard for audio performance and
conferencing in any style of room. With both monaural and stereo acoustic
echo cancellation capabilities, the SoundStructure conferencing products
provide an immersive conferencing experience that is unparalleled. The
SoundStructure products are easier than ever to install and configure and have
been designed to integrate seamlessly with the Polycom HDX™ video
conferencing system and the Polycom Touch Control™ for the ultimate
experience with HD voice, video, content, and ease of use.
The Polycom SoundStructure C16, C12, and C8 audio conferencing devices are
single rack unit devices that have 16 inputs and 16 outputs, 12 inputs and 12
outputs, or 8 inputs and 8 outputs respectively. The SoundStructure SR12 has
12 inputs and 12 outputs and is an audio device for commercial sound
applications that do not require acoustic echo cancellation capabilities. Any
combination of SoundStructure devices can be used together to build systems
up to a total of eight SoundStructure devices and up to one hundred
twenty-eight inputs and one hundred twenty-eight outputs. SoundStructure
products can be used with any style of analog microphone or line-level input
and output sources and are also compatible with the Polycom HDX digital
array microphones.
1
The SoundStructure products are used in similar applications as Polycom’s
Vortex® installed voice products but have additional capabilities including:
•Stereo acoustic echo cancellation on all inputs
•Direct digital integration with the Polycom HDX video conferencing
system
•Feedback elimination on all inputs
•More equalization options available on all inputs, outputs, and submixes
•Dynamics processing on all inputs, outputs, and submixes
•Modular telephony options that can be used with any SoundStructure
device
•Submix processing and as many submixes as inputs
1 - 1
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
•Ethernet port for easy configuration and device management
•Event engine for using internal state information such as muting, logic
input and logic output ports, and an IR remote for controlling
SoundStructure
SoundStructure devices are configured with Polycom's SoundStructure Studio
software, a Windows®-based comprehensive design tool used to create audio
configurations either online (connected to a SoundStructure system) or offline
(not connected to a SoundStructure system). SoundStructure Studio is used to
upload configuration files to SoundStructure systems and to retrieve
configuration files from SoundStructure systems.
For detailed information on how to install, terminate cables, and connect other
devices to the SoundStructure devices, refer to the SoundStructure Hardware Installation Guide. For information on the SoundStructure API command
syntax used to configure SoundStructure devices and control the devices with
third party controllers, refer to the SoundStructure Command Protocol Reference Guide in Appendix A. The SoundStructure Command Protocol
Reference Guide can also be found by pointing a browser to the
SoundStructure device’s IP address.
This manual has been designed for the technical user and A/V designer who
needs to use SoundStructure products, create audio designs, customize audio
designs, and verify the performance of SoundStructure designs. This manual
is organized as follows:
•Chapter 2 is an introduction to the SoundStructure products including the
OBAM™ architecture and details of the signal processing available for
inputs, outputs, telephony, and submix processing.
•Chapter 3 presents the SoundStructure design concepts of physical
channels, virtual channels, and virtual channel groups. These concepts are
integral to making SoundStructure products easy to use and enable
control system application code to be reused and portable across multiple
installations.
•Chapter 4 describes how to use the SoundStructure Studio windows
software to create a design. Start with this section if you want to get up and
running quickly using SoundStructure Studio.
•Chapter 5 provides detailed information on customizing the design
created with SoundStructure Studio including all the controls presented as
part of the user interface. Start with this chapter if you have a design and
would like to customize it for your application.
•Chapter 6 provides information on the Conference Link2 interface and
how SoundStructure devices integrate with the Polycom HDX video
conferencing system.
•Chapter 7 provides information on how to link multiple SoundStructure
devices with the OBAM™ interface.
1 - 2
Introduction
•Chapter 8 provides information on how to install, set signal levels, and
validate the performance of the SoundStructure devices. Start here if you
have a system already up and running and would like to adjust the system
in real-time.
•Chapter 9 provides information on how to use SoundStructure ‘events’
with logic input and output pins, an IR remote, and for options for how to
send commands from SoundStructure’s RS-232 interface to other devices
including a Polycom HDX video codec.
•Chapter 10 provides information for the network administrator including
how to set IP addresses and how to view the internal SoundStructure logs,
and more.
•Chapter 11 provides the steps for using the Polycom Touch Control with
a SoundStructure system. See the Users Guide for SoundStructure and the
Polycom Touch Control for instructions on how to use the Polycom Touch
Control with SoundStructure.
•Chapter 12 provides example applications with SoundStructure products
including stereo audio conferencing applications, room combining, and
more.
•Chapter 13 provides troubleshooting information and steps including
details on the status LEDs on SoundStructure.
•Chapter 14 lists the Specifications for the SoundStructure devices
including audio performance, power requirements, and more.
•Chapter 15 provides information on how to use the different UI elements
in the SoundStructure Studio software including knobs and matrix
crosspoints.
•Appendix A provides detailed information on the SoundStructure
command protocol and the full command set.
•Appendix B provides detailed information on how to use SoundStructure
Studio’s address book functionality to manage and connect to
SoundStructure systems across an enterprise’s network.
•Appendix C is an audio conferencing design guide. Refer to this section if
new to audio conferencing or would like to better understand audio
conferencing concepts.
If new to the SoundStructure products, it is recommended that the manual be
read starting from Chapter 2 for an overview, chapter 5 to begin using
SoundStructure Studio, and the remaining chapters as necessary to learn more
about using SoundStructure products.
1 - 3
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
1 - 4
SoundStructure Product Family
There are two product lines in the SoundStructure product family - the
SoundStructure C-series designed for audio conferencing applications (the
“C” stands for conferencing) and the SoundStructure SR-series designed for
commercial sound applications (the “SR” stands for sound reinforcement).
While these two product families share a common design philosophy they
have audio processing capabilities that are designed for their respective
applications. As described in detail below, the C-series of products include
acoustic echo cancellation on all inputs and are designed for audio and video
conferencing applications. The SR-series of products do not include acoustic
echo cancellation and are designed for dedicated sound reinforcement, live
sound, broadcast and other commercial sound applications that do not require
acoustic echo cancellation processing.
2
SoundStructure Architecture Overview
This section defines the common architectural features of the SoundStructure
products and then details the specific processing for both the C-series and
SR-series products. Details on how to configure the devices are presented in
Chapters 3 - 5.
All SoundStructure products have been designed with the flexibility of an
open architecture and the ease of design and installation of a fixed architecture
system. The resulting solution has tremendous flexibility in how signals are
processed while simultaneously making it easy to achieve exceptional system
performance.
The SoundStructure processing includes input processing that is available on
all the inputs, output processing that is available on all the outputs, submix
processing that is available on all the submix signals, telephony processing
that is available on all the optional telephony interfaces, and an audio matrix
that connects this processing together. The high-level architecture is shown in
2 - 1
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
1
2
N
1
2
N
Telco
Processing
Telco
Processing
Telco
Processing
Telco
Processing
Matrix
Processing
SubMix
Submix
Processsing
Output
Processing
Output
Processing
Output
Processing
Input
Processing
Input
Processing
Input
Processing
the following figure for a SoundStructure device that has N inputs and N
outputs. The specific input and output processing will depend on the product
family (C-series or SR-series) and is described later in this chapter.
The table following summarizes the numbers of inputs, outputs, and the
number of submixes supported within each type of device. As shown in this
table, each SoundStructure device has as many submixes as there are inputs to
the device.
SoundStructure
# inputs
#outputs
#submixes
C16C12C8SR12
1612812
1612812
1612812
A summary of the different types of processing in the C-series and SR-series
products is shown in the following table. As can be seen in this table, the
difference between the products is that the C-series products include acoustic
echo cancellation while the SR-series products do not include acoustic
2 - 2
SoundStructure Product Family
echo cancellation. The processing capabilities will be described in the
following sections.
SoundStructure
Input Processing
Up to 8th order highpass and lowpass✔✔
1st or 2nd order high shelf and low shelf✔✔
10-band parametric equalization✔✔
Acoustic echo cancellation, 20-22kHz 200 msec tail-time, monaural or stereo✔
Automatic gain control: +15 to -15dB✔✔
Dynamics processing: gate, expander, compressor, limiter, peak limiter✔✔
Feedback Eliminator: 10 adaptive filters✔✔
Noise cancellation: 0-20dB noise reduction✔✔
Automixer: gain shar i n g or ga ted mixer✔✔
Signal fader gain: +20 to -100 dB✔✔
Signal delay to 1000 msec✔✔
Output Processing
1st or 2nd order high shelf and low shelf filters✔✔
10-bands of parametric or 31-band graphic equalizer✔✔
Dynamics processing: gate, expander, compressor, limiter, peak limiter✔✔
Signal fader gain: +20 to -100 dB✔✔
Cross over equalization up to 8th order highpass and lowpass filters, 1st order horn equalization✔✔
Crossover delay: up to 100 msec✔✔
Signal delay: up to 1000 msec✔✔
Submix Processing
Up to 8th order highpass and lowpass filters✔✔
1st or 2nd order high shelf and low shelf filters✔✔
10-bands of parametric equalization✔✔
Dynamics processing: gate, expander, compressor, limiter, peak limiter✔✔
Signal fader gain: +20 to -100 dB✔✔
Signal delay: up to 1000 msec✔✔
Telco Processing
Line echo cancellation, 80-3300Hz, 32msec tail-time✔✔
Dynamics processing: gate, expander, compre ssor, limiter, peak limiter on telco transmit and receive✔✔
Up to 8th order highpass and lowpass filters✔✔
1st or 2nd order high shelf and low shelf filters✔✔
10-bands of parametric equalization on telco transmit and receive✔✔
Call progress detection✔✔
Signal fader gain: +20 to -100 dB✔✔
Automatic gain control: +15 to -15dB on telco receive✔✔
Signal delay on telco transmit and receive: up to 1000 msec✔✔
Noise cancellation: 0-20dB noise reduction on telco receive✔✔
C-SeriesSR-Series
OBAM™ - One Big Audio Matrix
One of the significant advancements in the SoundStructure products is the
ability for multiple devices to be linked together and to be configured and
operated as one large system rather than as multiple individual devices
feature dramatically simplifies any installation where audio from more than
one device is required such as complicated sound reinforcement applications.
OBAM's 'one large system' approach provides many benefits including:
•It is easier to work with the system because all the input signals feed into
the single matrix and all the outputs are fed from the single matrix
1
Requires SoundStructure firmware release 1.2 or higher.
1
. This
2 - 3
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
OBAM
OUTIN
OUTIN
OUTIN
OBAM
16x16
16x1612x12
8x8
8x8
12x12
36x36
•The a/v designer can be more creative as there are no limitations on how
signals from multiple devices can be used together
•The device linking scheme is completely transparent to the designer - all
input signals are shared to all devices dramatically simplifying the setup,
configuration and maintenance of large systems
•It is easier to set up the system with SoundStructure Studio as all inputs
and outputs are viewed on one screen, eliminating the need to configure
multiple devices and view multiple pages
This one big system design approach is the result of the SoundStructure
architectural design and the OBAM high-speed bi-directional link interface
between devices. With OBAM linking, up to eight devices may be linked
together. If there are plug-in cards installed in multiple linked SoundStructure
devices, the plug-in card resources are available for routing to any output
across the system. See the Hardware Installation Guide or Chapter 3 for more
information on how to link multiple devices together.
The one large system design philosophy means that the audio matrix of a
system of SoundStructure devices is the size of the total number of inputs and
outputs of all the component devices that are linked together. Since one
SoundStructure C16 device has a 16x16 matrix, two C16 devices linked
together create a 32x32 matrix and so forth.
The one big audio matrix architecture can be seen in the following figure
where a C16 device is OBAM linked to a C12 device which is OBAM linked to
a C8 device. The resulting system will have 36x36 inputs and 36 outputs
(16+12+8 = 36). In addition to all the inputs and outputs, the submixes of each
device will also feed the matrix allowing the designer to have 36 submix
signals (not shown in the following figure), one for each input that can be used
in the system.
Because of the OBAM design architecture, the A/V designer no longer has to
be concerned with device linking, as multiple SoundStructure devices will
behave as, and be configured as, one large system.
2 - 4
SoundStructure C-series Products
The SoundStructure C16, C12, and C8 devices are designed for audio
conferencing applications where groups of people want to communicate to
other individuals or groups such as in a typical room shown in the following
figure.
SoundStructure Product Family
The SoundStructure C-series products feature both monaural and stereo
acoustic echo cancellation, noise cancellation, equalization, dynamics
processing, feedback elimination, automatic microphone mixing, and more.
All audio inputs have the same processing capability and can be used with either
microphone-level or line-level inputs. Phantom power is available on all inputs.
All outputs have the same processing capability.
A single SoundStructure C16, C12, or C8 device supports 16, 12, or 8
microphone or line inputs and 16, 12, or 8 line outputs, respectively. Up to
eight SoundStructure devices may be linked together (any combination of
SoundStructure C-series or SR-series products may be used together) to build
audio processing systems that support up to one hundred twenty-eight analog
inputs and analog outputs.
2 - 5
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
Telco
Video Codec
Amplifier
SoundStructure
C16
Microphones
Telephony
Playback/Record
Network
PSTN
Network
Favorite Content
SoundStructure Installation
Each SoundStructure C-series device may be used with traditional analog
microphones, with Polycom's HDX digital microphone arrays
1
. For detailed
information on using the Polycom HDX digital microphone arrays, see
Chapter 6.
Typical applications of the SoundStructure C-series conferencing products are
audio and video conferencing where two or more remote locations are
conferenced together. The typical connections in the room are shown in the
following figure.
2 - 6
Before designing with SoundStructure products, the details of the
SoundStructure signal processing capabilities will be presented.
1
Requires SoundStructure firmware release 1.1 or higher.
C-Series Input Processing
Mic or Line
Input
Input to
Matrix
Input to
Matrix
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Non Linear
Processing
Feedback
Cancellation
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
The input processing on the SoundStructure C-series devices is designed to
make it easy to create conferencing solutions either with or without sound
reinforcement. Each audio input on a SoundStructure C-series device has the
processing shown in the following table.
The signal processing follows the signal flow shown in the following figure.
SoundStructure Product Family
Input Processing
Up to 8th order highpass and lowpass
1st or 2nd order high shelf and low shelf
10-band parametric equalization
Acoustic echo cancellation, 20-22kHz 200 msec tail-time, monaural or stereo
Automatic gain control: +15 to -15dB
Dynamics processing: gate, expander, compressor, limiter, peak limiter
Feedback Eliminator: 10 adaptive filters
Noise cancellation: 0-20dB noise reduction
Automixer: gain sharing or gated mixer
Signal fader gain: +20 to -100 dB
Signal delay to 1000 msec
Telco
Telco
Telco Processing
Telco
Processing
Processing
Processing
Input
1
Processing
Input
2
Processing
Input
N
Processing
Matrix
SubMix
Submix
Processing
Processsing
Output
Processing
Output
Processing
Output
Processing
1
2
N
Each analog input signal has an analog gain stage that is used to adjust the gain
of the input signal to the SoundStructure's nominal signal level of 0 dBu. The
analog gain stage can provide from -20 to 64 dB of gain in 0.5 dB steps. There
is also an option to enable 48 V phantom power on each input. Finally the
analog input signal is digitized and available for processing. The digital signal
2 - 7
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
Mic or Line
Input
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Non Linear
Processing
Feedback
Cancellation
Route
C-Series Input Processing
Input to
Matrix
Input to
Matrix
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
is processed by five different DSP algorithms: parametric equalization,
acoustic echo cancellation, noise cancellation, feedback reduction, and echo
suppression (non linear processing).
C-Series Input Processing
Mic or Line
A/D
Parametric
Analog
Input
Gain
Acoustic Echo
Converter
Equalization
Non Linear
Noise
Processing
Cancellation
Cancellation
Feedback
Cancellation
AGCDynamicsFaderDelay
Automatic
Dynamics
Gain Control
Router
Processor
Dynamics
Automatic
Processor
Gain Control
Automatic
Dynamics
Gain Control
Processor
FaderDelay
Automixer
Fader
Automixer
Fader
Automixer
Mute
Recording/
Input to
Matrix
Ungated
Delay
Delay
Input to
Conferencing
Matrix
Sound
Input toMatrix
Reinforcement
Continuing through the signal path as shown in the next figure, the input
signal continues through the AGC (automatic gain control), dynamics
processing, an automixer, an audio fader, and finally through the input delay.
AGC DynamicsFaderDelay
Automatic
Dynamics
Gain Control
Processor
A/D
Mic or Line
Input
Parametric
Analog
Converter
Equalization
Gain
Non Linear
Acoustic Echo
Noise
Processing
Cancellation
Cancellation
Feedback
Cancellation
Automixer
Dynamics
Automatic
Automixer
Processor
Gain Control
Automatic
Dynamics
Automixer
Gain Control
Processor
FaderDelay
Fader
Fader
Mute
Input to
Recording/
Matrix
Ungated
Input to
Delay
Delay
Conferencing
Matrix
Sound
Input to
Matrix
Reinforcement
2 - 8
Each analog input signal is processed to generate three different versions of
the processed input signal that can be used simultaneously in the matrix:
1. Conferencing version,
2. Sound reinforcement version, and
3. Recording/ungated version
The AGC, dynamics processor, and input fader are linked together on all three
audio paths and apply the same gain to the signal paths based on an analysis
of the signal earlier in the signal path.
The automixer processing is only applied to the conferencing and sound
reinforcement signal paths to ensure that there is an 'un'-automixed version of
the input signal available for recording/ungated applications.
SoundStructure Product Family
Mic or Line
Input
Input to
Matrix
Input to
Matrix
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Non Linear
Processing
Feedback
Reduction
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
C-Series Conferencing Input Processing
Parametric
Equalization
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Fader
Automixer
Delay
Non Linear
Processing
Dynamics
Processor
Mute
Each analog input signal is processed to create three processed versions that can
be used in different ways in the matrix.
These three different versions of the input signal mean that, at the same time,
an output signal to the loudspeakers can use the sound reinforcement
processed version of an input signal, an output signal to the video
conferencing system can use the conferencing processed version of the input
signal, and an output signal to the recording system can use the recording
processed version of the input signal. The decision of which of these three
processed version is used is made at each matrix crosspoint on the matrix as
described in the Matrix Crosspoint section below.
Conferencing Version
The conferencing version will be processed with the acoustic echo and noise
cancellation settings, non-linear signal processing, automatic gain control,
dynamics processing, automixer, fader, delay, and input mute. The
conferencing signal path and summary block diagram is highlighted in the
following figure. This is the path that is typically used to send echo and noise
cancelled microphone audio to remote locations. This is the default processing
for microphone inputs when the automixed version of the signal is selected.
Sound Reinforcement Version
The sound reinforcement version will be processed with the echo and noise
cancellation, optional feedback elimination processing, automatic gain
control, dynamics processing, automixer, fader, delay, and input mute. This is
the path that is typically used for sending local audio to loudspeakers in the
room for sound reinforcement. There is no non-linear processing on this path
so that the local talker audio to the loudspeakers is not affected by the presence
of remote talker audio in the local room.
The automatic gain control on the sound reinforcement path is different from
the automatic gain control on the conferencing version of the signal because
the sound reinforcement automatic gain control will not add gain to the signal.
In other words, the sound reinforcement AGC will only reduce the gain of the
input signal. This restriction on the sound reinforcement AGC is to prevent the
2 - 9
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
Mic or Line
Input
Input to
Matrix
Input to
Matrix
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Non Linear
Processing
Feedback
Cancellation
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
C-Series Sound Reinforcement Input Processing
Parametric
Equalization
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Fader
Automixer
Delay
Feedback
Cancellation
Dynamics
Processor
Mute
automatic gain control on the sound reinforcement path from increasing the
microphone gain and consequently reducing the potential acoustic gain before
the onset of feedback.
The automatic gain control on the sound reinforcement processing path will not add
gain to the signal, it will only reduce the gain of the signal.
Recording/Ungated Version
The recording version of the processed input signal is specifically designed to
not include the gain sharing or gated style of automatic microphone mixing
processing. The recording/ungated version of the input channel is typically
used for recording applications or in any application where an un-automixed
version of the input signal is required.
For additional flexibility in audio applications, there are four different
versions of the recording/ungated signal that can be selected through the
four-input router shown in the above processing figures. This selection of
which type of recording/ungated signal to choose is performed on an input by
input basis within the SoundStructure Studio software as described in Chapter
5.
The four ungated versions are described in more detail below:
1. bypass version
2. line input version
3. conferencing version
4. sound reinforcement version
Recording/Ungated - Bypass
The recording/ungated-bypass version has no input processing other than a
fader gain control, input delay, and input mute. This version bypasses the
automatic gain control and dynamics processing as shown in the following
figure. This version can be used when it is important to minimal audio
2 - 10
SoundStructure Product Family
Mic or Line
Input
Input to
Matrix
Input to
Matrix
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Non Linear
Processing
Feedback
Cancellation
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
UNGATED - Bypass
FaderDelay
Mute
Mic or Line
Input
Input to
Matrix
Input to
Matrix
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Non Linear
Processing
Feedback
Cancellation
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
UNGATED - Line Input Processing
Parametric
Equalization
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Mute
Mic or Line
Input
Input to
Matrix
Input to
Matrix
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Non Linear
Processing
Feedback
Cancellation
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
UNGATED - Conferencing Processing
Parametric
Equalization
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
FaderDelay
Non Linear
Processing
Dynamics
Processor
Mute
processing on an input signal. This version of the signal has no acoustic echo
cancellation processing and will consequently include any acoustic echo signal
that may be present at the microphones.
Recording/Ungated - Line Input
The recording - line input version includes equalization, automatic gain
control, and the dynamics processing as well as fader gain control, input delay,
and input mute as shown in the following figure. This processing path is
typically used by line input signals such as program audio, and hence the
name line input path.
Recording/Ungated - Conferencing
The ungated conferencing processed input includes the acoustic echo and
noise cancellation as shown in the following figure. This path is typically used
for recording of conference microphones as it includes all the acoustic echo
cancellation but not the automatic microphone mixer processing.
2 - 11
Design Guide for the Polycom SoundStructure C16, C12, C8, and SR12
Mic or Line
Input
Input to
Matrix
Input to
Matrix
Parametric
Equalization
A/D
Converter
Analog
Gain
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
Automixer
AGCDynamicsFaderDelay
Fader
Automixer
Automixer
Delay
Automatic
Gain Control
Non Linear
Processing
Feedback
Cancellation
Dynamics
Processor
Dynamics
Processor
Fader
Input to
Matrix
Automatic
Gain Control
FaderDelay
Dynamics
Processor
Delay
Mute
Recording/
Ungated
Conferencing
Sound
Reinforcement
UNGATED - Sound Reinforcement Processing
Parametric
Equalization
Acoustic Echo
Cancellation
Noise
Cancellation
Automatic
Gain Control
FaderDelay
Feedback
Cancellation
Dynamics
Processor
Mute
Recording/Ungated - Sound Reinforcement
Finally, the sound reinforcement recording input includes the echo and noise
cancellation and optional feedback elimination processing as shown in the
following figure.
All three versions (conferencing, sound reinforcement, recording/ungated) of
the input signal processing can be used simultaneously in the matrix. The
conferencing version is typically used to send to remote participants, the
sound reinforcement version is typically used to send to the local loudspeaker
system, and the recording version is typically used for archiving the
conference audio content.
C-Series Matrix Crosspoints
The audio matrix is used to create different mixes of input signals and submix
signals to be sent to output signals and submix signals. Matrix crosspoints gain
values are shown in dB where 0 dB means a signal value is unchanged. For
example, a crosspoint value of -6 dB will lower the signal gain by 6 dB before
it is summed with other signals. The matrix crosspoint gain can be adjusted in
0.1 dB steps between -100 and +20 dB and may also be completely muted. In
addition, the matrix crosspoint can also be negated/inverted so that the
crosspoint arithmetic creates a subtraction rather than an addition. The
inversion technique may be effective in difficult room reinforcement
environments by creating phase differences in alternating zones to add more
gain before feedback.
Matrix crosspoints associated with stereo channels have a balance or pan to
control mapping mono to stereo channels, stereo to mono channels, and stereo
to stereo channels.
The three different versions of the input processing - the ungated,
conferencing, and sound reinforcement - are selected at the matrix crosspoint.
The SoundStructure Studio software allows the user to select which version of
the input signal processing at the matrix crosspoint. As will be shown in
Chapter 4 Creating Designs, the different versions of the input processing will
be represented with different background colors in the matrix crosspoint.
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