Mixer and Input Only Models
ASPEN
TECHNICAL DATA
Mixer and Input Only Models
• Ultra low noise input preamps
• Full crosspoint matrix with 48 outputs
• Simultaneous multi-point 3rd party and
native control
• Unlimited input expansion
• 48 final mixes, ultra-low latency
• TCP/IP Ethernet addressable
• Automatic Master/Slave detection
• Adaptive Gain Proportional Automatic
Mixing at the matrix crosspoints
• Single CAT6 interconnection carries data,
audio and control signals
The variety of mixers available are created by combining
“building block” printed circuit board assemblies:
• 8 input, 12 output mixer board
• 16 channel input only board
• 8 channel input only board
Multiple chassis of any 1RU or 2RU model can be
stacked to expand the number of inputs and outputs
needed for the system design. The ASPEN digital matrix
provides a maximum of 48 total outputs, but there is no
limit to the number of inputs that can be added to a system by stacking multiple units.
Mixer models include:
• SPN812
8 input, 12 output mixer, 1 RU
• SPN1612 16 input, 12 output mixer, 2 RU
• SPN1624
16 input, 24 output mixer, 2 RU
• SPN2412
24 input, 12 output mixer, 2 RU
Input only models include:
• SPN16i
16 channel , 1 RU
• SPN32i
32 channels, 2 RU
Input only units deliver outputs to the digital bus, so they
are always used with a host mixer or conference unit to
provide physical audio outputs.
When multiple units are stacked, Master and Slave units
are automatically detected and configured. All data and
audio from the Slave units in the system is gathered in
the Master, so a single connection between a computer
and the Master allows access to all units in the stack.
The throughput latency of all audio inputs in a stack is
automatically synchronized to maintain absolute signal
phase at the audio outputs.
All models fully support the 48 outputs provided by the
digital matrix, regardless of how many physical outputs
are present on the rear panel. Any physical output can
deliver the signal from any output in the matrix.
Extensive, simultaneous control can be applied through
ethernet, USB, RS-232 and logic I/O ports using the
large command library. Combined with the comprehensive macro library, control options are available to meet a
very wide range of requirements and user interfaces.
The ASPEN control protocol has several key features:
• It is a request-response message protocol
• An error reporting mechanism is provided
• The message syntax is text based
• Messages may carry data payloads
For more information, browse the Help files in the downloadable software from the web site.
Rio Rancho, NM, USA
www.lectrosonics.com
Signal Flow
Every input can be used with microphone or line level
signals with gain adjustable from -10 to +60 dB. Following the analog to digital converter (ADC) processing
stages are arranged in logical order. After processing,
the signal can be assigned to any one or more crosspoints in the matrix for mixing and forward propagation
to the Master unit.
Audio signals and data in each processor are added to
the audio signals and data from the processor below it
and propagated to the next processor above it (forward
propagation) through the 1 Gbps bus. The final total of
all signals and data in a stack are gathered in the Master
and then propagated back to the Slave units (backward
propagation). In this manner, the signal source for each
output on every Slave in the stack can be taken from any
of the 48 crosspoints in the matrix.
Each output has a signal processing chain, gain control
and limiter ahead of the digital to analog converter.
Automatic mixing takes place at the matrix crosspoints
with four different characters available:
• Automatic - normal auto mix activity
• Direct - on at all times; for recording
• Override - dominant in the mixing activity
• Background - subordinate in the mixing activity
• Phantom - for multi-zone mix-minus systems
This unique functionality allows each input to behave
differently at each output. For example, one channel with
a microphone connected can provide a direct signal for
recording at some output channels, participate on an
equal basis with other output channels in a sound reinforcement system and act as the “chairman” microphone
in groups delivered to other output channels.
Mic/Line
Inputs
Mic/Line
Outputs
NRF:
ADC:
DAC:
ADFE:
Noise Reduction Filter
Analog-Digital Converter
Digital-Analog Converter
Automatic Digital Feedback Elimination
This diagram depicts the signal flow through a mixer
model with physical inputs and outputs. Input Only models process and route input signals into the matrix, but do
not have physical outputs, so they are always used with
a mixer or conference model processor.
The ASPEN architecture allows all signal processing to
be fully enabled at all times without limitations of DSP
resources. With the computer connected to the Master
unit in the system, changes in the setup, filters, etc. take
effect and are immediately audible.
Automatic Master/Slave Detection
Each processor board connects to the other processors
through the upper and lower RJ-45 jacks. The presence
of a connector determines the ranking of the processor
in the stack. If there is no connector in the upper jack,
the processor is positioned at the top of the stack and is
automatically configured as the Master in the system. If
a connector is present in the upper jack, the processor is
automatically configured as a Slave in the system.
Adaptive Proportional Gain Automatic
MIxing Algorithm*
Mix-Minus Sound Reinforcement
An automatic mixer that uses a Gating process turns
channels off and on abruptly as the level passes an
established threshold. While this can be acceptable in a
small sound system, it usually produces “choppy” sound
when used with more than a few microphones. A more
effective approach is Proportional Gain mixing where
each channel level is compared to the sum of all channels and more gain is given to the louder channels. In
this approach, channel levels are adjusted in a seamless
manner to eliminate the abruptness of a gate.
The proprietary algorithm used in ASPEN processors
adds a significant improvement to conventional proportional gain auto mixing. The gain allocation in a conventional algorithm is in direct proportion to the activity at
the microphones, so gain is reduced on channels with
lower levels, but those channels still inject sound and
noise into the final output mix.
Sound levels at the microphones
Sound levels at the microphones
Mix-minus routing through the matrix establishes loudspeaker coverage Zones that function as sub-systems
within the room, each using a dedicated Final Mix to
provide the signal routing. This design approach excludes nearby microphones from the mix sent to each
loudspeaker group to improve feedback stability, noise
suppression and the echo return loss in audio conference connections. In this example, the audio from microphones in the red zone are routed to the green and blue
zones, but not back to the red zone.
As a finishing touch on a mix-minus design, a unique
mixing mode called Phantom Mix can also be employed.
The Phantom Mix Mode
Conventional
Channel levels in the output mix
ASPEN
Channel levels in the output mix
The patented algorithm used in ASPEN processors applies a subtle priority to the channel that has been the
loudest for the longest period of time. This increases the
gain on the dominant channel and decreases the gain
on subordinate channels to a greater degree than a
conventional proportional gain mixer. Only one channel
is dominant, which further reduces background noise,
suppresses recirculating sound in the room and prevents
comb filtering when a single voice arrives at two microphones at close to the same level.
In a meeting space, sound from loudspeakers is not
contained strictly within the defined acoustic zones. In
addition, all of the microphones are in the same overall
acoustic space. Even though a mix-minus routing has
been established, sound from a loudspeaker bleeds into
adjacent zones, is picked by those microphones, and it is
routed back to the loudspeaker where the sound originated. This recirculation reduces ERL and intelligibility,
and can even cause feedback.
The Phantom Mix mode allows multiple zones to participate in an overall room mixing activity, but delivers the
audio signals only to the desired acoustic zones defined
in the mix-minus setup. Microphone channels that are
excluded in the routing are set to the phantom mode to
accomplish this finishing touch.
Final Mix 3
Zone 1 mics in Auto mode
Zone 2 mics in Auto mode
Zone 3 mics in Phantom mode
ZONE 3
ZONE 2
Final Mix 2
Zone 1 mics in Auto mode
Zone 2 mics in Phantom mode
Zone 3 mics in Auto mode
Final Mix 1
Zone 1 mics in Phantom mode
Zone 2 mics in Auto mode
Zone 3 mics in Auto mode
*US Patents 5,414,776 and 5,402,500
ZONE 1
EXAMPLE: The Phantom Mix mode
employed in mix-minus signal routing
SPN 1624
GND
PROG IN
DATECODE
OUTPUTS
+5V
INPUTS
Made In the USA
PROG
OUT
S/N LABEL
Adaptive Proportional Gain Mixing
US Patent 5,414,776
ASPEN PORTS
GND
PROG IN
OUTPUTS
+5V
PROG
OUT
100-240V
50/60Hz 40W
RS-232
INPUTS
ETHERNET
The model SPN1624 houses two 8x12 mixer boards in a 2RU housing.
GND
PROG IN
SPN 16i
INPUTS
DATECODE
+5V
Made in the USA
PROG
OUT
100-240V
50/60Hz 25W
S/N LABEL
RS-232
ETHERNET
ASPEN PORT
The model SPN16i houses a single input only board
Specifications
Audio inputs
All inputs are digitally programmable-gain microphone to line level differential inputs. Either side
can be grounded or left floating. The cable shield shall be connected to ground.
Max. input level:
20 dBu
Gain:
0 dB to 56 dB, programmable in 8 dB steps
(the analog gain is automatically selected by selecting
the input gain)
Input impedance:
8 kΩ differential mode, 2 kΩ common mode
Phantom voltage:
48 V
Dynamic range:
102 dB
EIN:
-127 dBu (20Hz – 20kHz, unweighted)
THD + noise:
0.01%
Audio outputs
All outputs are floating transformerless differential outputs. Either side can be grounded or left
floating. The cable shield shall be connected to ground.
Nominal level:
0 dBu, channels 1-8
0 dBu, -20 dBu, -40 dBu, channels 9-12
Headroom:
20 dB
Output impedance:
< 50 Ω, all outputs, at all attenuator settings
Dynamic range:
105 dB
THD + noise;
0.01%
Latency
Single-board:
64 audio samples = 1.333 ms
System:
64 + 6 * (total number or boards – 1) audio samples =
1.333 + 0.125 * (total number or boards – 1) ms
Monitor output (1/4” headphone jack)
Signal:
any of the 48 final mixes
Output power:
50 mW (<50 ohm impedance recommended)
Filters
All filters, including the noise reduction filter (NRF), have zero processing delay.
Noise reduction filters:
Adjustable 6 to 35 dB on every input
Tone control stages:
4 per input channel
Parametric EQ stages:
8 per output channel
ADFE: 8 per input channel
Configurable as Static or Dynamic
Filter types
Low Pass:
Butterworth (6, 12, 18, 24 dB/octave)
Bessel (6, 12, 18, 24 dB/octave)
Linkwitz-Riley (12, 24 dB/octave)
Additional parameters: frequency [Hz]
High Pass:
Butterworth (6, 12, 18, 24 dB/octave)
Bessel (6, 12, 18, 24 dB/octave)
Linkwitz-Riley (12, 24 dB/octave)
Additional parameters: frequency [Hz]
Low Shelving Butterworth (6, 12, 18, 24 dB/octave)
Bessel (6, 12, 18, 24 dB/octave)
Additional parameters:
frequency [Hz]
boost/cut [dB]
High Shelving Butterworth (6, 12, 18, 24 dB/octave)
Bessel (6, 12, 18, 24 dB/octave)
Additional parameters:
frequency [Hz]
boost/cut [dB]
Peaking EQ (parametric)
Parameters:
frequency [Hz]
bandwidth [octave]
boost/cut [dB]
Internal Signal Generator:
Swept sine:
Modes:
single sweep, continuous sweep
Waveforms:
sawtooth (up or down), triangle
Sweep rate:
linear, logarithmic
Parameters:
start freq, stop freq, level [dBu], sweep time [sec]
White noise:
Parameter:
level [dBu]
Pink noise:
Parameter:
level [dBu]
Tone (sine wave): Parameters:
level [dBu], frequency
Power Requirements: 100-240 VAC, 50/60 Hz
Power Consumption:
SPN16i:
25 Watts
SPN32i:
45 Watts
SPN812:
20 Watts
SPN1612:
40 Watts
SPN1624:
40 Watts
SPM2412:
45 Watts
SPNCWB:
35 Watts
SPNTWB:
50 Watts
SPNDNT: 15 Watts
Dimensions:
1RU models:
2RU models:
Weight:
1RU models:
2RU models:
1.75 x 19.00 x 7.70 inches
3.50 x 19.00 x 7.70 inches
3.64 lbs., 1651 grams
5.73 lbs., 2600 grams
581 Laser Road NE • Rio Rancho, NM 87124 USA • www.lectrosonics.com
(505) 892-4501 • (800) 821-1121 • fax (505) 892-6243 • sales@lectrosonics.com
6 Ausust, 2014
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