Lectrosonics DMTH4 Data Sheet
DMTH4
Digital Telephone Hybrid
• 3-in/24-out digital matrix architecture
• Fully integrates with DM Series processors
• Telephone, codec and auxiliary inputs and outputs
• Two Acoustic Echo Cancellers - 126 ms tail time
• Line echo canceller - 30 ms tail time
• 6 filters plus compressor on each input
• 6 filters plus compressor/limiter on each output
• USB and RS-232 interfaces for setup and control
TECHNICAL DATA
• Fully balanced audio signal flow through entire system - no pin 1 problem
• Digital I/O ports for "daisy chaining" and to connect other LecNet 2 devices
• Proportional gain auto mixing algorithm with AutoSkew
The DMTH4 integrates telephone lines, video codecs and
external audio sources into the digital bus structure of DM
Series processors so these sources operate as though
they are another microphone or audio input in the sound
system. The unit is much more than just a telephone
interface. Instead, it is a complete DM Series digital
matrix processor, with a 3-in/24-out digital matrix, automatic mixing and comprehensive signal processing on
every input and output. In essence, it simply connects to
telephone lines, video codecs and external audio sources
instead of mic/line inputs and outputs and integrates
seamlessly with DM Series matrix processors.
The primary applications are in sound reinforcement and
conferencing systems in boardrooms, courtrooms,
worship centers, distance learning systems, hotels and
other applications with multiple microphones and loudspeakers. The design represents a milestone in DSP
technology in its basic architecture and in its processing
speed and efficiency.
The challenge in teleconferencing using a sound system
on one or both ends of a conference is to minimize echo
heard at the far end caused by the coupling between
loudspeakers and microphones in the local sound system.
As sound from the far end enters the local sound system
and is delivered by the loudspeakers in the local room, it
will enter the local microphones and be sent back to the
far-end. At the far-end the listeners will hear an echo of
their own speech.
™
- US Patent 5,414,776
The integration of adaptive gain proportional auto mixing*
with an all new proprietary echo canceller provides a
remarkable solution that is as easy to install and set up as
it is effective. Echo-free teleconferencing and clean local
sound reinforcement is provided even in poor acoustical
environments.
The DMTH4 shares the large digital matrix bus with other
DM Series processors to handle a wide range of sound
system requirements from a modest boardroom to large
systems with hundreds of inputs. Multiple units can be
stacked with multiple DM processors to handle very large
systems with multiple phone lines.
Extensive control capability is built into the unit with an
intuitive command structure to allow external control with
USB or RS-232 connections. Up to 128 macros can be
stored in internal memory. Each macro can contain up to
64 commands, with 115 characters in each command. A
built-in macro recorder greatly simplifies the creation of
and use of macros.
*US Patent 5,414,776
Rio Rancho, NM, USA
www.lectrosonics.com
Echo and Echo Cancellation
The fundamental problem with microphone/speaker
acoustical coupling is illustrated below. Far end audio is
delivered by the loudspeakers in the room and the
microphones pick it up and return it to the far end. The
delay through this process creates an echo heard on the
far end.
Telephone
Interface
Far-end
Local
loudspeaker
Local
microphone
Local
sound system
There are several methods used to reduce or eliminate
the echo heard on the far end of the conversation:
• Optimal design in the sound system to minimize the
coupling between loudspeakers and microphones.
• Mix-minus matrix routing.
• Automatic microphone mixing.
• Digital echo cancelling.
Matters become more complex when the sound system is
required to provide both teleconferencing and sound
reinforcement. A gain proportional automatic mixing
process is widely recognized as the optimum solution for
sound reinforcement, but it places significant demands on
an acoustic echo canceller used for teleconferencing.
The matrix mixer enables complex signal routing and level
controls without limitations. The matrix mixing allows
"mix-minus" zoning of microphones and loudspeakers to
decouple them and reduce or eliminate acoustic feedback
and echoes. NOM attenuation is applied by the DSP at
the crosspoints in the matrix, which essentially provides
24 separate automatic mixers, each with its own NOM
mixing bus. Four different mixing modes can be selected
at the crosspoint for each input, so each input can participate differently in each output mix.
The automatic mixing process uses a seamless algorithm that eliminates gating and its ill-effects. Gain is
proportioned among all inputs assigned to a particular
output channel in a seamless and continuous manner
based upon microphone activity. The algorithm operates
in a natural, transparent manner and incorporates an
adaptive AutoSkew
™
process to eliminate artifacts such
as comb filtering and abrupt gating that occur with
conventional automatic mixing schemes. Audio from the
far-end of a conference participates in the local mixing
algorithm just like a microphone in the local sound
system.
Two digital acoustic echo cancellers are provided in the
DMTH4 to further reduce the return of local signals to the
far-end. One operates on the telco connection and the
other is dedicated to the video codec connection. In
conjunction with the automixing process, echoes are
minimized and not heard at the far end.
ERL
ERL (echo return loss) refers to the natural attenuation of
the far-end audio signal as it circulates from the far-end
through loudspeakers and microphones in the local sound
system and back to the far-end. Good design in the local
sound system will reduce the acoustic coupling between
loudspeakers and microphones using physical placement
and mix-minus matrix routing. Depending upon room size
and acoustics, it is often impossible to achieve adequate
decoupling to avoid an echo heard by the far-end during a
teleconference. Thus, other types of processing are
needed to further reduce the return echo.
ERLE
ERLE (echo return loss enhancement) refers to additional
circuits and processes used to further increase ERL.
Common methods are to use automatic mixing and digital
echo cancellation.
Return Loss Enhancement
The gain proportional automatic mixing algorithm* in the
DM Series processors not only provides seamless mixing
for local sound reinforcement without abrupt gating, but it
also contributes significantly to ERLE. The additional
contribution is plotted in the following graph.
Digital echo cancellation is another method of reducing
the echo delivered to the far-end. The concept, described
in very simple terms, is to have the DSP recognize the
far-end audio and subtract it from the transmitted audio to
remove any echo they might hear at the far-end. Sounds
simple, but in a sound system with multiple microphones
and loudspeakers, it is not easy to identify the far-end
audio in the complex mix of local sound, local noise and
the effects of the room on the far-end audio delivered by
the local loudspeaker system. When there is no sound or
noise in the local room, the DSP can do a decent job of
identifying the far-end audio and subtracting it from the
transmitted signal, but this is rarely the case in full duplex
teleconferencing. People talk, laugh and make noise, and
air conditioners and projectors make noise, etc.
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In a simple sound system arrangement, the local microphone can be muted when nobody is talking in the local
room. A simple gated mixer can provide this function.
With no open microphones locally, there is obviously no
return echo signal. This requires that a threshold level be
set high enough to keep the microphone from being
opened by background noise, but low enough to allow it to
open when someone speaks. When the local microphone
is open, a return echo path is created, which is when a
DSP echo canceller is needed. Given the wide variety of
human voices and the dynamics of noise in a meeting
room, a gated mixer is often not the best choice.
Using a dedicated DSP echo canceller on each input of
the local mixer (referred to as “distributed echo cancellation”) is an expensive but effective approach to reducing
the return echo. The process requires the algorithm to
“converge,” which is to identify the far-end audio and
subtract it from the signal sent to the far-end. This requires at least a brief moment when there is very little
local sound or noise, with significant far-end audio present
in the room. If nobody moves and there are no gain
changes made to local microphones and loudspeakers, it
is possible (in theory) to effectively remove return echo,
but this is not a very realistic situation.
The theory behind distributed echo cancelling is that once
the DSP has converged, it can continue to subtract far-end
audio even when the local microphone is open and far-end
audio is present at the same time. If there are any
changes in gain, noise or acoustics in the local space and
equipment, the DSP must re-converge, which requires
another brief moment with little or no local noise or sound,
and significant far-end audio present.
A gated automatic mixer does not change the gain when
the microphone is open, it just turns the channel off and
on abruptly. This helps with distributed echo cancelling
since the microphone is completely muted when not in
use, but it is very “choppy” sounding in the local sound
reinforcement system.
A gain proportional automatic mixer applies the most gain
to the most active microphone with smooth, continuous
changes. This makes it extremely effective for local sound
reinforcement, but the continuous gain changes make it
difficult for the echo canceller to remain converged and
effectively reduce the echoes at the far end.
The DMTH4 in conjuction with a DM Series processor
offers a unique approach to the problems with simultaneous teleconferencing and sound reinforcement. The
patented adaptive gain proportional mixing algorithm
works in conjunction with a centralized echo canceller to
address a variety of issues. The automatic mixer provides
seamless allocation of gain to local microphones through a
mix-minus matrix to reduce background noise and
decouple loudspeaker and microphones, while a very fast
converging DSP echo canceller operates on the composite
transmitted signal being sent to the far end. This combination of processes is possible only with the latest DSP
technology.
The auto mixing algorithm adapts to changes in background noise continuously, and unlike a gated mixer there
are no threshold levels to adjust. A sum of all channels is
the reference signal, each channel level is compared to
this reference and the individual channel gain is adjusted
to apply NOM attenuation. Gain is adjusted continuously
to eliminate audible artifacts that gating and abrupt level
changes can cause. As the common mode noise in the
room changes, all channels are affected equally. The end
result is seamless, adaptive auto mixing that requires no
calibration or threshold adjustments.
Each individual output of the matrix operates as a separate NOM bus, so a particular input can be assigned to
multiple outputs with mix parameters adjusted differently
for each output. In other words, gain and mix mode are
configured independently for each matrix crosspoint,
resulting in great flexibility. Four mix modes are supported: Auto, Direct, Override and Background.
The echo canceller converges continuously when the
level of the far side signal exceeds a minimum level, and
the ratio of the far side signal to local room sound exceeds a minimum ratio. This dynamic control prevents
divergence during periods of silence from the far side
room or in “doubletalk” situations. The convergence takes
place very quickly to keep up with the changes made by
the automatic mixing algorithm and other changes that
occur in the room. Setup is greatly simplified and any
adjustments, such as level changes made with a remote
control system, are accommodated automatically.
The convergence speed is adjustable in the control panel
GUI to fine tune it to a particular situation. Faster convergence times can track changes in the room almost
instantaneously, but the depth of echo cancellation will be
reduced. Slower convergence times take a bit longer to
fully converge, but produce greater echo cancellation.
The ERLE value achieved by the echo canceller is
displayed on the GUI and the effects of altering the
convergence rate will be immediately visible and audible.
An important final note on the DMTH4 is the fact that the
echo canceller will never “diverge” (lose convergence).
This unique algorithm will also converge on a continuous
sine wave, which is especially important when DTMF
tones are present in the room. Since the echo canceller
will never diverge, there is no need for a “panic button” (as
is used in other designs) to generate a noise burst to help
the echo canceller re-converge.
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