Omnia SST Audio Processing Software Manual v8.53 C19615044
Omnia SST Audio Processing Software Manual v8.53 C19615044
TelosAlliance.com
User Warnings and Cautions
The installation and service instructions in this manual are for use by qualified personnel only.
To avoid electric shock, do not perform any servicing other than that contained in the operating
instructions unless you are qualified to do so. Refer all servicing to qualified personnel
This instrument has an autoranging line voltage input. Ensure the power voltage is within the
specified range of 100-240VAC. The ~ symbol, if used, indicates an alternating current supply.
This symbol, wherever it appears, alerts you to the presence of uninsulated, dangerous voltage
inside the enclosure – voltage which may be sufficient to constitute a risk of shock.
This symbol, wherever it appears, alerts you to important operating and maintenance
instructions. Read the manual.
WARNING:
The product’s power cord is the primary disconnect device. The socket outlet should be located near
the device and easily accessible. The unit should not be located such that access to the power cord is
impaired. If the unit is incorporated into an equipment rack, an easily accessible safety disconnect
device should be included in the rack design.
To reduce the risk of electrical shock, do not expose this product to rain or moisture. This unit is for
indoor use only.
This equipment requires the free flow of air for adequate cooling. Do not block the ventilation
openings on the rear and sides of the unit. Failure to allow proper ventilation could damage the unit
or create a fire hazard. Do not place the units on a carpet, bedding, or other materials that could
interfere with any panel ventilation openings.
If the equipment is used in a manner not specified by the manufacturer, the protection provided by
the equipment may be impaired.
USA CLASS A COMPUTING DEVICE INFORMATION TO USER.
WARNING:
This equipment generates, uses, and can radiate radio-frequency energy. If it is not installed and
used as directed by this manual, it may cause interference to radio communication. This equipment
complies with the limits for a Class A computing device, as specified by FCC rules, part 15, subpart
j, which are designed to provide reasonable protection against such interference when this type of
equipment is operated in a commercial environment. Operation of this equipment in a residential
area is likely to cause interference. If it does, the user will be required to eliminate the interference
at the user’s expense. Note: objectionable interference to TV or radio reception can occur if other
devices are connected to this device without the use of shielded interconnect cables. FCC rules
require the use of shielded cables.
CANADA WARNING:
“This digital apparatus does not exceed the Class A limits for radio noise emissions set out in the
radio interference regulations of the Canadian department of communications.”
“Le présent appareil numérique n’émet pas de bruits radioélectriques dépassant les limites
applicables aux appareils numériques (de Class A) prescrites dans le règlement sur le brouillage
radioélectrique édicté par le ministère des communications du Canada.”
CE CONFORMANCE INFORMATION:
This device complies with the requirements of the EEC council directives:
♦♦ 93/68/EEC (CE MARKING)
♦♦ 73/23/EEC (SAFETY – LOW VOLTAGE DIRECTIVE)
♦♦ 89/336/EEC (ELECTROMAGNETIC COMPATIBILITY)
Conformity is declared to those standards: EN50081-1, EN50082-1.
Trademarks, Patents, and Licenses
Omnia is a trademark of TLS Corp. All other trademarks are the property of their respective holders.
All versions, claims of compatibility, trademarks, etc. of hardware and software products not
made by The Telos Alliance which are mentioned in this manual or accompanying material are
informational only. The Telos Alliance makes no endorsement of any particular product for any
purpose, nor claims any responsibility for operation or accuracy. We reserve the right to make
improvements or changes in the products described in this manual which may affect the product
specifications, or to revise the manual without notice.
This document and its content are copyrighted by TLS Corporation and may not be copied,
reproduced, or distributed in any form without expressed written permission.
Patent information can be found at www.TelosAlliance.com/legal
Updates
Telos Omnia.9 features and operations are determined largely by software. The Telos Alliance
strives to provide the most stable and feature-rich software available. We encourage you to check for
software updates from time to time by visiting our website or by contacting us directly.
Feedback
We welcome feedback on any aspect of our products or this manual. In the past, many good ideas
from users have made their way into software revisions or new products. Please contact us with your
comments or suggestions.
We support you…
By Phone/Fax
You may reach our Telos Support Team in emergencies by calling +1 216-622-0247. For billing
questions or other non-emergency technical questions, call +1 216-241-7225 between 9:00 AM to
5:00 PM USA Eastern Time, Monday through Friday.
By Email.
Non-emergency technical support is available at [email protected].
By Web
The Telos Web site has a variety of information that may be useful for product selection and support.
The URL is https://www.telosalliance.com/Telos .
SERVICE
You must contact Telos Alliance before returning any equipment for factory service. We will need
your unit’s serial number, located on the back of the unit. We will issue a return authorization
number, which must be written on the exterior of your shipping container. Please do not include
cables or accessories unless specifically requested by the Technical Support Engineer. Be sure to
adequately insure your shipment for its replacement value. Packages without proper authorization
may be refused. US customers, please contact Telos Alliance Technical Support at +1-216-622-0247.
All other customers should contact local representative to make arrangements for service.
Warranty
Metering . . . . . . . . . . . . . . . . . . . . . . .56
Input Scope . . . . . . . . . . . . . . . . . . . . . . 56
Declipped Scope. . . . . . . . . . . . . . . . . . . . 56
Output Scope . . . . . . . . . . . . . . . . . . . . . 56
MPX (Baseband spectrum analyzer). . . . . . . . . . . 56
RF (RF spectrum analyzer) . . . . . . . . . . . . . . . . 57
Left Demodulated Output. . . . . . . . . . . . . . . . 57
Input Meter. . . . . . . . . . . . . . . . . . . . . . 57
Output. . . . . . . . . . . . . . . . . . . . . . . . 57
MPX Power . . . . . . . . . . . . . . . . . . . . . . 57
FM . . . . . . . . . . . . . . . . . . . . . . . . . . 57
Delossifier. . . . . . . . . . . . . . . . . . . . . . . 57
Noise Gate . . . . . . . . . . . . . . . . . . . . . . . 57
Natural Dynamics. . . . . . . . . . . . . . . . . . . . 57
Phase. . . . . . . . . . . . . . . . . . . . . . . . . 57
Auto EQ. . . . . . . . . . . . . . . . . . . . . . . . 58
AGC. . . . . . . . . . . . . . . . . . . . . . . . . . 58
Stereo . . . . . . . . . . . . . . . . . . . . . . . . . 58
Equalizer . . . . . . . . . . . . . . . . . . . . . . . 58
BIYF . . . . . . . . . . . . . . . . . . . . . . . . . . 58
Harmonics . . . . . . . . . . . . . . . . . . . . . . . 58
Multiband 1 & 2 . . . . . . . . . . . . . . . . . . . . 58
SB . . . . . . . . . . . . . . . . . . . . . . . . . . 58
Clipper. . . . . . . . . . . . . . . . . . . . . . . . 58
Micro-MPX (µMPX) addendum . . . . . . . . . . . . . 59
Omnia µMPX™ . . . . . . . . . . . . . . . . . . . . . 59
TRADEMARKS. . . . . . . . . . . . . . . . . . . . . 76
NOTICE. . . . . . . . . . . . . . . . . . . . . . . . 76
UPDATES. . . . . . . . . . . . . . . . . . . . . . . . 76
FEEDBACK . . . . . . . . . . . . . . . . . . . . . . . 76
TECHNICAL SUPPORT. . . . . . . . . . . . . . . . . . 76
ADDITIONAL CONTACT INFORMATION AND SUPPORT . . . . 76
Software Licensing. . . . . . . . . . . . . . . . . . 77
END-USER SOFTWARE LICENSE AGREEMENT. . . . . . . . 77
Software Notice:. . . . . . . . . . . . . . . . . . . 82
Omnia SST and MicroMPX uses several third party libraries:. 82
Creating the Most Exciting and Engaging
Audio Experiences Imaginable
Congratulations on your new Telos Alliance product!
The gang here at Telos is committed to shaping the future of audio by delivering innovative, intuitive
solutions that inspire our customers to create the most exciting and engaging audio experiences imaginable.
We’re grateful that you have chosen audio tools from Telos® Systems, Omnia® Audio, Axia® Audio,
Linear Acoustic®, 25-Seven Systems®, and Minnetonka Audio®. We’re here to help you make your
work truly shine. We hope that you enjoy your Telos Alliance product for many years to come and won’t
hesitate to let us know if we can help in any way.
Introduction
Thank you for purchasing Omnia SST, a fully-featured software-based audio processor for FM
and digital broadcast. SST offers a powerful suite of tools to both repair and process audio.
This document should be used in conjunction with our Omnia SST Quick Start Guide (QSG).
While the QSG gives you a general orientation to setting up SST, especially audio hardware
configuration, these pages are intended as a reference guide, describing what various controls,
meters and elements of SST do. Near the end of this manual, you will also find a guide to
Omnia µMPX® (Micro MPX) Radio’s First Composite FM Codec: High-quality MPX over a
320kbps IP connection.
As is the case with other PC software, it is assumed you have moderate to advanced (PC)
computer knowledge, audio hardware configuration experience and networking skills. In
the case of this very specialized audio software, it is also assumed that you have moderate to
advanced familiarity with broadcast processing, audio signal chains, RDS, FM transmission,
and other systems that may feed or be fed by a computer running Omnia SST.
But whether you are a pro or a newcomer to audio processing, the best approach to learn what
SST can do is to simply install it on a computer, input source audio and start navigating various
screens and trying different settings. Building your own mental model of how different con-
trols and settings can make a difference to your sound will be the best teacher. We encourage
you to experiment, try different settings, alter and make your own presets, and generally
explore. If you get stuck or have questions, feel free to write us at:
[email protected]
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 2
Home Page
Input/Output (Same as “Soundcards” settings panel)
Sample Rate
Selects the sample rate to use for audio input.
Normally, the output sample rate will be identical to the input sample rate. The only
exception is when you are not using ASIO, the sample rate is lower than 128 kHz, and FM
output is used.
In this case, if you use sample rates above 48 kHz, the audio will be downsampled before
processing and upsampled again afterwards. 88.2 and 176.4 kHz are downsampled to
44.1 kHz, 96 and 192 kHz are downsampled to 48 kHz.
The sample rate has a small effect on the latency and CPU load. At 48 kHz there is a bit
more data that needs to be processed, which slightly increases the CPU load. The latency
is a bit smaller because a block of the same number of samples is a bit smaller. If you use
32 kHz or a multiple thereof, the CPU load is a lot smaller - but you should only use this
low a sample rate it if you don't need any audio output above approximately 14 kHz.
If you want to use the FM output with stereo and RDS encoding, and you are using ASIO,
then the sample rate must be set to at least 128 kHz. Samples rates of 176.4 or 192 kHz are
preferred. If you don't use ASIO for the output, the output sound card will automatically
be opened with a high enough sample rate to send out the stereo and RDS signals.
Important:
If you are not using ASIO, you need to make sure that the Windows sound card driver is
configured to use the same sample rate that you have selected in Omnia SST, or Windows
will resample it. This causes audible artifacts and may cause issues with the FM output (or it
may not work at all). Open the properties for the audio card you are using and click on the
“Advanced” tab. Set the “Default Format” to a mode with a sample rate that matches the
one selected in SST.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 3
ASIO
Audio Stream Input/Output (ASIO) is a computer sound card driver protocol for digital
audio specified by Steinberg, providing a low-latency and high fidelity interface between a
software application and a computer's sound card. ASIO bypasses the normal audio path
from a user application through layers of intermediary Windows operating system soft-
ware so that an application connects directly to the sound card hardware. Each layer that
is bypassed means a reduction in latency. ASIO offers a relatively simple way of accessing
multiple audio inputs and outputs independently.
The ASIO control selects whether or not to use an ASIO device driver for your hardware.
These drivers are generally more stable, offer better control over sound cards, and reduce
delay between input and output. It is strongly suggested that ASIO be used if possible.
The ASIO buffer size can be adjusted through the card’s ASIO driver control panel. This
functionality varies between cards. Consult your audio card documentation for more
details.
ASIO (On/Off)
Turns ASIO use on or off. This is a global setting and will enable or disable the ability to
use an ASIO device for inputs and outputs.
ASIO Device ID
Selects which ASIO driver and device to use for ASIO I/O selections. Only one device can
be selected here, though 3rd party drivers such as ASIO4ALL allow additional flexibility in
routing between multiple ASIO devices.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 4
Main Input
Selects the audio inputs that will be used for the main processing path.
Input Device ID
Selects a WDM device to use for input. This setting is overridden when “ASIO Input Left”
is set to anything other than “No ASIO”.
Buffer Usage
Displays the input buffer status. Normally this bar will remain nearly full.
Of course, it is still possible to accommodate lower input signal levels with a large amount
of headroom. With this slider you can adjust the level such that under normal circum-
stances the peaks are at about 0 dB. If peaks are occasionally louder, that's no problem.
Unlike some environments, the signal does not clip at 0 on the input metering.
For best results, adjust the input gain of the sound card mixer before adjusting this
control. Setting gain stages is important in any system, and adding gain at this stage can
potentially increase any noise introduced by the sound card.
Balance
Adjusts for different input levels on left and right channel.
If the left and right channel input levels aren't perfectly equal (which might happen for
example if you're using analog equipment), this slider allows you to correct it. A negative
value means that the left channel is boosted, a positive value means that the right channel
is boosted. A good way to calibrate balance is to input mono audio and adjust this slider
until the input meter levels are visually equal for left and right.
Most other devices (including amplifiers) may reduce the level of a channel instead of
increasing it. The reason to increase the level is that otherwise it's more difficult to show
both the original and the modified (by Input gain and Balance) volume levels - one chan-
nel could be made louder while the other could be made softer.
RC
RC value of the first high-pass filter.
Tilt is often caused by a Resistor and Capacitor (RC) circuit, which acts as a highpass fil-
ter. For example, a sound card might use an RC circuit to remove DC offset from a signal.
Unfortunately, this also reduces the audio quality and makes processing more difficult.
The tilt correction filter in Omnia SST inverts the effect of an RC circuit, thus restoring
the original signal (except for the DC offset).
This slider sets the RC value of the RC circuit to be inverted.
Secondary Input
Selects a second audio input, which can be used for backup (failover) audio or SCA
purposes (including external RDS).
Secondary Input
Global settings for the secondary audio input.
Input 2 On/Off
Enables or disables the secondary audio input.
Input 2 Device ID
Selects a WDM device to use for the secondary audio input. This setting is overridden and
disabled when “ASIO Input 2 Left” is set to anything other than “No ASIO”.
Buffer Usage
Displays buffer status for input 2. Normally this bar should remain nearly full.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 7
Usage
Selects how the secondary audio input will be used
FM Output
These settings control the audio card output used for the FM MPX signal.
FM Output
FM Output (On/Off)
Turns the FM composite (MPX) output on or off.
Device ID
Selects a WDM device to use for the MPX output. This setting is overridden and disabled
when “ASIO Output Left/1” is set to anything other than “No ASIO”.
Buffer Size
Adjusts the amount of buffering or diversity delay to apply to the MPX output. Can be set
up to 12.5 seconds.
Buffer Usage
Displays the status of the MPX output buffer. If this bar is not staying filled, audio drop-
outs will result. Increase the buffer size to ensure this buffer is staying filled.
FM Tilt Correction
As with the audio card inputs, many sound cards introduce “tilt” on their output due to
output filtering and lack of DC coupling. This can cause overshoots and result in lower
modulation (as much as 3-5% or more) as levels are reduced to account for the overshoot
while keeping within appropriate modulation limits.
While ideally an audio card with DC coupled inputs and outputs would be used to avoid
the issue, SST has tilt correction on both input and output to compensate for this.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 9
RC
RC value of the first highpass filter.
Tilt is often caused by a Resistor and Capacitor (RC) circuit, which acts as a highpass fil-
ter. For example, a sound card might use an RC circuit to remove DC offset from a signal.
Unfortunately, this also reduces the audio quality and makes processing more difficult.
The tilt correction filter in Omnia SST inverts the effect of an RC circuit, thus restoring
the original signal (except for the DC offset).
This slider sets the RC value of the RC circuit to be inverted.
Test Signal
Generates various test signals on the outputs for calibration purposes.
Generate Test Tone (On/Off)
Enables or disables the internal test signal generator and replaces the processed audio at
the outputs with the selected test signal.
Available test signals include:
♦♦ Sine
♦♦ Smooth square
♦♦ Square
♦♦ 400 Hz Sine
♦♦ (various) Bessel Null
♦♦ Sawtooth w/pause (for AM)
♦♦ Sawtooth flat bottom (for AM)
Frequency
Adjusts the output frequency of the selected tone. This control is applicable for sine,
square, or smooth square and is ignored for other tone selections with fixed frequencies.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 10
Invert
Inverts polarity on the selected MPX output channel.
Delay
For fine-tuning Adjusts the amount of delay (measured in samples) applied to the select-
ed channel from 0 to 192 samples.
Web/DAB+/HD Output
This section contains controls for a secondary audio processing output, independent
of the FM MPX and low latency outputs. It is typically used for a web stream, DAB/HD,
or processed monitoring where latency is not a concern. The signal patch point for this
output is selectable.
Device ID
Selects a WDM device to use for the secondary audio output. This setting is overridden
and disabled when “ASIO Output Left” is set to anything other than “No ASIO”.
Web/DAB+/HD Output
Controls the output level of the secondary processing output.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 11
Buffer usage
Displays the status of the secondary processing path output buffer. If this bar is not stay-
ing filled, audio dropouts will result. Increase the buffer size to ensure this buffer is staying
filled.
Signal Selection
drop down menu:
Device ID
Selects a WDM device to use for the low latency output. This setting is overridden and
disabled when “ASIO Output Left” is set to anything other than “No ASIO”.
Note:
Using WDM rather than ASIO with the low latency output will add sufficient delay to make it
unsuitable for low latency usage.
Volume
Controls the output level of the low latency monitor output. Since there is no clipping
on this output, the volume may need to be reduced to account for any spikes in the audio
level.
Buffer Size
Adjusts the amount of buffering to apply to the low latency output. Set this as low as
possible to minimize latency while still maintaining a stable output.
Buffer Usage
Displays the status of the low latency output buffer. If this bar is not staying filled, audio
dropouts will result. Increase the buffer size to ensure this buffer is staying filled.
Clean Up
The “Clean Up” sections are used for pre-conditioning the audio prior to processing. This
includes removing hum, de-clipping, restoration of lossy encoded audio, correcting for
azimuth errors in recordings, noise reduction, and restoring dynamic range.
Each block on this page has an “On/Off” toggle, as well as a “Hear” control. These con-
trols allow you to easily listen to the results of each block while fine-tuning the processing.
“On/Off” will enable or bypass each block individually, while “Hear” allows you to hear
the processing up to that point, bypassing all downstream blocks.
The “On/Off” controls are available inside each block as well. Some blocks also contain a
“Delta” control which allows you to hear the effect of that just that block.
Note that toggling these controls affects ALL outputs and WILL be heard on the air. There
is no “Preview” output available. The “Hear” and “Delta” controls can be disabled in the
system settings to avoid inadvertent activation.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 13
Dehummer
The “Dehummer” can be used to learn and remove any constant noise from a source.
For constant noise at specific frequencies, the noise can usually be filtered entirely from
the source audio. For other noise (such as hiss or other broadband noise) it is typically
possible to achieve several dB of attenuation without significantly affecting the source
audio.
Learn (On/Off)
Used to “learn” the noise signature to be removed. With only the noise source applied to
the input, turn the “Learn” control “On” for several seconds. This samples the noise such
that a filter can be constructed to remove it.
Declipper
Although FM processors typically perform clipping as a final processing stage, many
modern recordings are already clipped before they even reach the listener. The declipper
can account for this and repair clipped audio prior to processing.
DeLossifier
Lossy audio encoding (such as MP3) reduces the amount of data needed to store or
stream audio by using psychoacoustic models to 'throw away' content that the codec
deems 'nearly inaudible'. The trade-off of such data reduction schemes is that they can
cause audio artifacts (especially at lower bitrates).
In addition to the fact that these artifacts are often already quite audible, the processing
that is done by SST (or any other processor) may further accentuate such artifacts, coun-
tering the codec’s assumptions of what is or is not audible.
The DeLossifier attempts to detect and clean up artifacts introduced due to lossy audio
encoding before the audio hits the processing stage.
Pre-Ringing
*Adds considerable latency
One of the typical artifacts in MP3 and other lossy compressed files is pre-ringing. This
means that a sound that should start suddenly has a sort of 'pre-echo'. Instead of starting
suddenly you may first hear it come up softly, and then there's a sudden jump when the
sound was really supposed to start. This gives the typical 'smeared' sound and loss of
‘punch’ on lower bitrate MP3s.
This filter attempts to detect pre-ringing, and reduce it.
Important: this filter uses a look-ahead mechanism in order to determine whether a sound
is caused by pre-ringing or it is actually part of the signal. Look-ahead systems require
buffering that can add considerable latency.
Delta button
Outputs only the difference (delta) introduced by the pre-ringing filter, allowing you to
hear its effect.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 15
Delta button
Outputs only the difference (delta) added by the “Fill Spectral Holes” function, allowing
you to hear its effect.
Azimuth
Attempts repair of azimuth (phase) errors.
Azimuth errors are often present in tape recordings, and also on poorly mastered CDs.
Phase problems cause very unpleasant artifacts when playing a recording in mono or
through a surround system. Even normal stereo playback may sometimes sound a bit
“odd”. Any large phase offset is automatically detected and removed by this filter.
This filter only works properly if audio on the left and right channels is similar. If this is
not the case for a long period of time, the azimuth correction will slowly be reduced.
Noise Removal
The noise gate removes background noise, which is mainly useful for older recordings
from analog source material. Gentle use of this filter can also improve audio from some
CDs by removing quantization noise.
Noise Removal
Global on/off control of how much noise is removed.
Band Settings
Fine-tuning of the amount of noise to be removed per frequency band.
Delta button
Outputs only the difference (delta) removed by the “Noise Removal” function, allowing
you to hear its effect.
Natural Dynamics
In addition to being heavily clipped, modern recordings often lack any dynamic range.
It's much easier for an audio processor to work with music that has some dynamic range
rather than trying to further compress already heavily compressed recordings.
“Natural Dynamics” attempts to restore dynamic range and “punch” to heavily com-
pressed recordings while not altering audio that already contains sufficient dynamic
range.
Delta button
Outputs only the difference (delta) of the “Natural Dynamics” function allowing you to
hear its effect.
Natural Dynamics
Main settings for the “Natural Dynamics” section
Strength
Increases or decreases the overall effect of “Natural Dynamics”.
Bands
Overview of the band splits for the natural dynamics section. There are no adjustments in
this section.
Quick Adjust
Contains a series of controls that manipulate key “processing” components described in
the next section. Quick Adjust acts as a set of “VIP” controls and is useful for when you
want to “tweak” a preset that has just been loaded, without going into all the depth and
complexity of fuller control layers. The “0” points of these controls will be whatever was
loaded via the preset.
Loudness
Trims the amount of drive to the final clipper--this control has the biggest effect on overall
loudness.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 17
Texture
“Master” controls for the multiband section. These will impact how “busy” the audio
sounds, trading openness for aggressive density.
Density
Adjusts master drive to the multiband sections.
Attack/Release speeds
Speeds up or slows down reaction time in the multiband sections.
Stereo Enhance
Adjusts the amount of stereo enhancement effect applied to the audio.
L-R enhancement
Controls the maximum amount of boost applied to the L-R component of the input signal.
Strength
Adjusts the overall amount of Multiband Stereo Boost. Higher values sound 'wider', but
extreme settings may increase the chance of multipath distortion.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 18
Processing
The “Processing” section of SST is where all controls for the main audio processing blocks
are grouped. This includes phase rotation, AGC, stereo enhancement, EQ, bass and high
frequency enhancement, multiband processing, bandpass filtering, single band compres-
sion, and the final clipper.
Phase
Phase Rotation makes the peaks of asymmetrical waveforms more symmetrical. In other
words, if a peak has a high amount of energy in the positive direction and very little in
the negative direction, energy will be added in the negative direction. The same is true of
peaks in the negative direction—energy will be added in the positive direction
This protects the compressors, limiters and clipper against certain types of sounds that
can be difficult to process and tend to cause distortion. Examples are voices (especially
female voices) and instruments like trumpets. These types of sounds are very asymmet-
rical with large peaks in one direction. If there is also lot of low frequency energy present
(such as a song with heavy bass) it can cause additional issues with these waveforms due
to intermodulation distortion.
Phase Rotation
Global settings for phase rotation.
Sound Shaping
Additional controls for phase processing, including adding frequency specific phase delay.
Auto EQ
*Will cause increased CPU usage when enabled
This filter is a dynamic re-equalizer that operates before the AGC in the processing path.
Using the familiar Bands user interface found in MB1 and MB2, it can be configured to
precisely match bands already in use elsewhere. Subject to user settings, this filter will
attempt to keep levels in the bands at a Target level with respect to each other. Spectrum
Correction is not a compressor and does not affect actual levels, overall. Furthermore,
using it does not require any subsequent filters to be altered to accommodate it. If this
filter is boosting the level in some band(s), it is reducing some other band(s).
Without Auto EQ, the multiband sections are responsible for two jobs: 1) Controlling
audio levels, and 2) Dynamically re-equalizing incoming audio. This is usually not a
problem, but it can become one on some material.
Let’s say incoming audio usually produces about 10dB of gain reduction in MB1 and all
bands are roughly equal in the amount of gain reduction they’re doing. Now a song comes
along that has very little high end. The top band of the multiband section will release and
could end up doing almost no work at all. That is suboptimal, because the multiband now
has no margin of error. If the amount of highs drops even further, there’s nothing the
multiband can do to correct it, and the audio will become muddy and dull. That’s where
Auto EQ comes in, to fix material that is imbalanced before it hits any further filtering.
When audio is playing that is properly balanced, per the settings, this filter will have
virtually no effect on the audio. When doing large amounts of correction, the filter should
be unobtrusive. It simply fixes the audio relatively slowly while attempting to be as trans-
parent as possible.
The user interface for this filter resembles the multiband compressor UI, but has some
important distinctions. When the filter is doing no work, you’ll see nothing in the inter-
face (no colored bars). Bars stretching to the right, in shades of blue indicate a boost in
level. Bars toward the left, in shades of red, indicate a cut to the volume in that band.
The individual controls are discussed below.
Band Controls
Effect
Adjusts the effect of the Auto EQ in each band
Effect strength
This filter does not have to fully correct spectral imbalances. Using this slider will make
the filter do less correction. If band 1 is 6dB too quiet, and this is set to 50%, only 3dB of
correction will happen.
Minimum level
The maximum amount of reduction the filter can make for this band.
Maximum level
The maximum amount of increase the filter is allowed to provide for this band.
Levels
Threshold
Sets the relative point at which the Auto EQ will begin to act in a given band.
Channel linking
It is possible that the left/right balance could be incorrect even within a single band. Use
this control to specify how much the correction should be allowed to vary between the
channels.
Band mix
After correction has been applied, it is possible to still increase or reduce the output of the
band.
Bands
Displays the crossover points between bands. This display does not allow modifying the
crossover points.
AGC
AGC (Automatic Gain Control) attempts to make the audio level constant by gradually
increasing and decreasing the gain. If sudden level jumps are present in the audio, it can
remove these transients without lowering the output level too much. This avoids pump-
ing and similar annoying effects.
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Leveler
Controls the overall behavior of the AGC section.
Drive
Adds additional gain to the AGC input to compensate for low input levels or to provide
greater opportunity for low level audio to increase.
Target Level
Sets the target maximum output level from the AGC. This control essentially serves as the
“drive” to the following stages of processing, including multiband.
Ratio
Determines the change in output level as compared to the change in input level. Lower
ratios deliver a more open sound while higher settings provide increased density and
tighter level control.
Window
Sets a window in dB in which the ACG will not increase or decrease gain, providing a
“dead band” during which no processing occurs in this section. Larger values will provide
increased dynamics and a more open, less “busy” sound but will also require downstream
sections of the processor to work harder.
Gate Slowdown
Slows the AGC activity when the input level drops below this value to prevent rush up of
noise or other artifacts. This also prevents large swings in levels between pauses in audio.
Side Chain
An EQ in the sidechain of the AGC to control the frequencies at which it responds. This
can help prevent “pumping” of the AGC with heavy bass or midrange frequencies, for
example.
Graph/EQ
Allows “drawing” an EQ curve for the sidechain, and displays an RTA of the incoming audio.
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Transient Protection
Prevents radical level changes due to sudden increases in incoming audio levels by tempo-
rarily overriding the normally slow-acting AGC with a faster-acting AGC.
Threshold
If the volume suddenly increases by more than this amount, AGC transient protection is
activated.
If you set this level too low, AGC transient protection will be activated too soon and there
will be a loss of dynamics. If set too high, overly loud audio will be let through to the next
processing steps. When transient protection is active, a portion of the AGC meter will
change to a different color.
Slope
Determines the range between the threshold amount, and maximum amount of transient
protection.
Power Bass
Adds additional low end punch to tracks that don’t contain much bass energy—particu-
larly certain older recordings. This results in a much deeper and warmer sound for both
old and new music. On tracks that already contain a considerable amount of bass, there is
little added effect.
Lowpass
Determines the frequency below which bass will be added.
Maximum Boost
Controls the maximum amount of boost applied to bass frequencies.
Threshold
Adjusts the detection threshold of the Power Bass section. If the detected amount of
existing bass content is above this threshold, additional bass will not be added.
Power Highs
Similar in function to Power Bass, but for high frequencies. This results in a much brighter
and consistent sound for tracks lacking high frequency energy.
Highpass
Determines the frequency above which high frequencies will be added.
Maximum Boost
Controls the maximum amount of boost applied to high frequencies.
Threshold
Adjusts the detection threshold of the Power Highs section. If the detected amount of
existing high frequency content is above this threshold, additional high frequency energy
will not be added.
Compressor
This is an additional wideband compressor following the Leveler, Power Bass, and Power
Highs
Compressor
Turns the compressor stage of the AGC on or off.
Window
Sets a window in dB in which the ACG will not increase or decrease gain, providing a
“dead band” during which no processing occurs in this section. Larger values will provide
increased dynamics and a more open, less “busy” sound but will also require downstream
sections of the processor to work harder.
Ratio
Determines the change in output level as compared to the change in input level. Lower
ratios deliver a more open sound while higher settings provide increased density and
tighter level control.
Attack
Controls how quickly the compressor will respond to changes in audio level above the
threshold. Smaller numbers will result in a faster attack time. Move the control to the left
to slow down attack and to the right to speed up attack.
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Release
This control determines how quickly the gain reduction value recovers when the audio
level falls below the threshold.
Gate freeze
“Freezes” the action of the compressor gain reduction when audio falls below this value.
Stereo
Tools to enhance the stereo separation.
Delta button
Outputs only the difference (delta) of the “Stereo” function to hear its effect.
L-R enhancement
Controls the maximum amount of boost applied to the L-R component of the input signal.
L-R Delay
Adjusts the amount of delay of the L-R component that is added to the original stereo
signal. When using more delay, the sound appears to 'open up', but extreme settings can
start to sound unnatural.
When set to 0, there is no added delay.
Threshold
Controls the level below which no additional L-R information will be added. This value is
relative to the L+R level.
Strength
Adjusts the overall amount of Multiband Stereo Boost. Higher values sound 'wider', but
extreme settings may increase the chance of multipath distortion.
Equalizer
The equalizer section is fairly self explanatory. It affects the frequency response of the
signal prior to the multiband compressors and subsequent stages. The interface allows
“drawing” the desired EQ curve, and displays an RTA showing the input signal.
Parametric EQ
Allows use of standard parametric EQ controls rather than “drawing” the desired curve.
There are 3 bands of fully parametric EQ, and two “shelving” EQ (Bass, Highs).
The resulting curve will be displayed in the RTA window. When enabled, any changes
previously made by the “drawing” method are discarded.
Delta buttons
Each Bass In Your Face section has a “Delta” button. Toggle this button to hear the effect
of ONLY that section.
Peak Frequency
Controls the center frequency around which each Bass In Your Face section will generate
additional energy.
Strength
The relative “intensity” of the additional generated bass energy. Values close to 1.0 will
give the most “natural” sound.
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Phat Bass
Generates additional bass boost with harmonic energy.
Delta button
Allows you to hear the effect of ONLY the “Bass harmonics” section.
Strength
Determines the amount of bass harmonic energy that will be added
Up to
Determines the frequency up to which the maximum amount of bass boost will occur. Set
the frequency higher to include more mid-bass enhancement. Excessively high settings
can cause distortion on some material.
Slope to
No bass boost will occur at frequencies above this setting. Absolute Highs
Adds high frequency harmonic content.
Absolute Highs
Delta button
Toggle this button to hear the effect of ONLY the “Absolute highs” section.
Multiband 1 & 2
Main
Density
Adds additional gain to all bands to compensate for low input levels or to provide greater
opportunity for low level audio to increase like a typical “Drive” control, but automatical-
ly applies an equal amount of makeup gain to the output. Higher density levels will result
in more activity in the multiband limiters.
Band coupling
Controls the degree to which bands track one another. A setting of 0% allows the bands
to act completely independently. Higher settings force all bands to track more closely
resulting in more of a wideband effect.
Master Gate
Gate Freeze
Stops or “freezes” the release action of the multiband compressors when audio falls below
the selected value.
Gate Slowdown
Slows down the release action of the multiband compressors when audio falls below
this value. This is normally set at a higher level than the Gate Freeze control to create a
window in which the release time slows down but doesn’t stop completely.
Band Controls
G/R Meters
These meters indicate the amount of gain reduction in each band and mirror the multi-
band gain reduction meters elsewhere in the interface. The crossover frequency for each
band is also indicated here.
Speed
This page contains individual attack and release speed controls for each band.
Attack
Controls how quickly each band responds to changes in audio level above the threshold.
Smaller numbers will result in a faster attack time. Move the control to the left to slow
down attack and to the right to speed up attack.
Release
This control determines how quickly the gain reduction value recovers when the audio
level falls below the threshold.
Levels
This page contains controls that affect input and ouput levels in each band.
Density
Operates in the same way as the Main Density control, but for individual bands.
Threshold
Controls the input level above which the compressor becomes active.
Limit
Adjusts the drive to the multiband limiters. Moving the sliders to the right increases
limiter activity and density.
Band Mix
Adjusts the final output level of each band. In Multiband 1, adjusting these controls will
in turn affect the amount drive to Multiband 2. In Multiband 2, these controls will affect
the amount of drive to the final clipper. Bear in mind that since Multiband 2 is the final
processing stage before clipping, extreme adjustments here can cause audible artifacts
from the clipper.
Sound
Controls that affect the gain reduction behavior of each band in the multiband section.
Feedback
Adjusts the balance between feed back and feed forward operation. In feed back mode
(100%), the output level is used for detection. In feed forward (0%), the input level is
used.
Ratio
Determines the change in output level as compared to the change in input level for
individual bands. Lower ratios deliver a more open sound while higher settings provide
increased density and tighter level control.
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Knee
Adjusts the “curve” of the compression knee at the threshold. Higher values will make this
transition softer, lower values will make it more abrupt.
Window
Sets a window in dB in which each bacn of the compressor will not increase or decrease
gain, providing a “dead band” during which no processing occurs in this section. Larger
values will provide increased dynamics and a more open, less “busy” sound but will also
require downstream sections of the processor to work harder.
Detection
Determines whether the compressor will use a peak-based or RMS-based algorithm.
Peak mode will tend to react to a single small spike in the sound. RMS mode responds
more like human hearing does, but may cause pumping with low frequencies.
Gate
The “gate” controls will prevent the multiband compressor gain reduction in each band
from moving back toward zero during periods of low or no audio.
Gate Freeze
Freezes the release action of each individual band when audio falls below this value.
Gate Slowdown
Slows down the release action of each individual band when audio falls below this value.
This is normally set at a higher level than the Gate Freeze control to create a window in
which the release time slows down but doesn’t stop completely.
Bands
This section will display the band crossovers, as well as an RTA of the input and output
signal from the multiband compressor. The grey portion of the bars represents the input
signal, and the yellow portion the output.
Bandpass
The bandpass filter consists of highpass and lowpass filters, will allow only frequencies
above or below a certain point to pass to the processor output.
Highpass
Passes only frequencies above a certain point.
Delta button
Toggle this button to hear the effect of ONLY the “Highpass” filter.
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Highpass frequency
Adjusts the frequency above which audio will pass.
Lowpass
Passes only frequencies below a certain point.
Delta button
Toggle this button to hear the effect of ONLY the “Lowpass” filter.
Lowpass frequency
Adjusts the frequency below which audio will pass.
Singleband
An additional single band compressor which follows the previous processing stages, prior
to the final clipper.
Delta button
Toggle this button to hear the effect of ONLY the single band compressor.
Speed
This group of controls affect the reaction time of the single band compressor.
Attack
Controls how quickly the compressor will respond to changes in audio level above the
threshold. Smaller numbers will result in a faster attack time. Move the control to the left
to slow down attack and to the right to speed up attack.
Release
This control determines how quickly the gain reduction value recovers when the audio
level falls below the threshold.
Window
Sets a window in dB in which the ACG will not increase or decrease gain, providing a
“dead band” during which no processing occurs in this section. Larger values will provide
increased dynamics and a more open, less “busy” sound but will also require downstream
sections of the processor to work harder.
Levels
The controls in this panel will affect the overall input and output levels of the single band
compressor.
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Density
Adds additional gain to compensate for low input levels or to provide greater opportunity
for low level audio to increase like a typical “Drive” control, but automatically applies
an equal amount of makeup gain to the output. Higher density levels will result in more
activity in the single band compressor.
Threshold
Controls the input level above which the compressor becomes active.
Limiter
Adjusts the drive to a limiter integrated with this single band compressor. Moving the
slider to the right increases limiter activity and density.
Sound
These two settings control the overall behavior of this single band compressor section.
Feedback
Adjusts the balance between feed back and feed forward operation. In feed back mode
(100%), the output level is used for detection. In feed forward (0%), the input level is used.
Ratio
Determines the change in output level as compared to the change in input level. Lower
ratios deliver a more open sound while higher settings provide increased density and
tighter level control.
Gate
The “gate” controls will prevent the singleband compressor gain reduction value from
moving back toward zero during periods of low or no audio.
Gate
Freezes the release action of the singleband compressor when audio falls below this value.
Gate Slowdown
Slows down the release action of the singleband compressor when audio falls below this
value. This is normally set at a higher level than the Gate Freeze control to create a win-
dow in which the release time slows down but doesn’t stop completely.
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Clipper
The clipper is the final stage of processing, and does a significant amount of “heavy
lifting” in Omnia SST prior to output. This stage is where adjustments with the biggest
impact on both distortion and loudness are found. Intermodulation distortion is mini-
mized as much as possible. The result is audio that can be extremely loud, yet still quite
dynamic with very little distortion.
Clipper
Main controls for the clipper section
Sound
This panel contains controls that affect the overall texture of the clipper.
Bass Shape
The controls in the “Bass Shape” panel help set the “texture” of the bass.
Strength
Unlike most bass clippers, the one in Omnia SST can (but does not have to) create a
square wave when processing strong low frequency content. Setting the “Strength” con-
trol at “0” (0%) will always prevent square waving the bass resulting in a very smooth low
end texture, but one that does not offer the “slam” of a traditional Omnia clipper even on
the initial peak of a bass transient waveform.
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Allow harmonics up to
This control sets the frequency of the highest harmonic produced by the bass clipper.
Higher settings will produce more and higher harmonics and create a more complex and
frequency-rich bass sound.
Distortion
These controls further affect the texture and feel of the bass clipper, and will have the
biggest effects on loudness versus distortion.
Delta button
Toggle this button to hear the effect of ONLY this stage. This will allow you to hear the
distortion being cancelled in the bass clipper.
Clip threshold
The “Clip Threshold” control is the point at which bass clipping will occur under normal
circumstances and is analogous to a traditional “bass clipper threshold” control.
Protection threshold
When bass at the level set by the “Clip Threshold” control would cause audible IM dis-
tortion, it will instead be clipped at the point determined by the “Protection Threshold”
control. Note that if both the “Clip Threshold” and “Protection Threshold” controls are
set to the same value, the bass clipper will operate in all cases in a “fixed threshold” tradi-
tional manner.
Settings
System
General settings for SST
License
License keys for Omnia SST are generated using the request code from this screen, and
are specific to each hardware install. Copy the request code, and paste it into an e-mail to
[email protected]. They will reply with a license key to paste into the
license key field. For evaluation, Omnia SST will run without licensing, although record-
ed messages will be injected into the audio after a period of time.
Request Code
Displays the unique license request code for this installation of Omnia SST. Copy/paste
this code to request a key.
License Key
When you receive your license key, paste it into this field and click “Activate License”.
License ID
The unique ID of this installation.
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Valid till
Omnia SST licenses can be set to expire after a period of time (demo use will cause audio
impairments after 30 days. When that happens, an unlicensed version is not desirable for
on-air demo) Fully licensed versions will show a date far in the future making the license
duration essentially permanent. This field displays the expiration date.
CPU Usage
This panel adjusts overall utilization of the system’s CPU.
Threads
Adjusts the number of CPU threads used by SST for processing. Additional threads can
result in higher quality, but will increase CPU usage.
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CPU Affinities
Manually allocates various components of SST operation to different physical CPUs or
CPU cores if necessary. Under normal circumstances these should NOT need to be adjusted.
LQLL Threads
Adjusts the number of CPU threads used by SST for the low latency monitoring path.
Additional threads can result in higher quality, but will increase CPU usage.
CPU Affinities
Manually allocates the low latency monitoring path to different physical CPUs or CPU
cores if necessary. Under normal circumstances these should NOT need to be adjusted.
Soundcards
This panel allows configuration of audio resources used within SST and will open the
same audio card settings described in the Quick Start Guide for Omnia SST and earlier in
this manual.
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FM Options
This panel contains various controls related to MPX generation in Omnia SST including
pre-emphasis, FM stereo pilot, SCA signals, composite clipping, and RF bandwidth
limiting.
Main
Global settings for MPX output
Enabled (On/Off)
Enables or disables MPX output entirely
Lowpass (On/Off)
FM broadcast is inherently limited in frequency response to approximately 15 kHz. In
many cases it makes sense to limit the audio bandwidth below this point when generating
a composite MPX signal. This switch will enable or disable a lowpass filter prior to MPX
generation.
Lowpass frequency
Controls the frequency of the lowpass filter. The default is 15.5 kHz. This control will be
greyed out if the lowpass filter is disabled.
Pre-Emphasis
Due to noise floor considerations, FM broadcast requires that the high-frequency energy
be radically boosted before broadcast, and then reduced in the receiver. This is done
through a process known as pre-emphasis and de-emphasis. The emphasis curve is mea-
sured by a time constant in microseconds. US broadcasters typically use 75 microseconds
(which provides a more aggressive high frequency boost) while other areas of the world
use 50 microseconds.
Pre-emphasize output
Controls whether not pre-emphasis is applied to the FM output. This control is greyed
out when pilot generation or RDS generation are turned on, since it is generally assumed
you would be applying pre-emphasis to a composite MPX signal which contained a stereo
pilot or RDS carrier.
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Stereo Generator
Global settings for the 19 kHz pilot and 38 kHz stereo subcarrier
Pilot injection
Controls injection level of the stereo pilot.
BS412
In several European countries, FM stations must comply with the ITU-R BS412 standard.
This standard specifies that the total power over any 60 second period is below a certain
threshold. The BS412 limiter in Omnia SST attempts to achieve this without compromis-
ing audio quality.
Since the BS412 limiter performs its power evaluation based on audio input over time,
you will get the best result if the input level is already nearly constant. The BS412 limiter
in Omnia SST knows how to handle pre-emphasis and slightly increased bass. Other
filters in Omnia SST will respond less to bass due to ITU-1770 when BS412 is enabled, but
certain extreme settings may still cause problems.
When BS412 is enabled, try to avoid using bass boost, or excessive boost in the equalizer
of the multiband compressor.
General
Global settings for the BS412 limiter
BS412 (On/Off)
Enables or disables the BS412 limiter.
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Headroom
Determines how much headroom the BS412 limiter has for errors. When lowering this
value, the average output level will drop to some extent. If this value is too low, however,
sudden volume drops can occur.
Drive
This value should be set such that the signal is just above the BS412 target level.
Behavior
Controls the BS412 long-term limiter. The long-term limiter adjusts the level very slowly
based on 20 different algorithms that predict future input. If the amount of limiting is less
than about 1 dB, it should normally not be noticeable.
Attack Shape
Determines how fast the level is dropped if it exceeds the target maximum level in any
60-second period. Reduce this value if the audio tends to change frequently in level. For
poorly controlled music with rapidly changing levels, a value of 2 might be appropriate.
For “normal” music, a value of 3 should work. If your audio has no dynamic range and is
smashed to the wall (or if you happen to broadcast pure tones) go ahead and set the value
even higher. It won’t sound great (unless you happen to like your audio levels smashed)
but it will keep the audio within the BS412 parameters.
HF Limiter Threshold
Sets the threshold at which high frequencies (including pre-emphasis) will be reduced if
they would cause the BS412 limit to be exceeded.
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LF Limiter Threshold
Sets the threshold at which low frequencies will be reduced if they would cause the BS412
limit to be exceeded.
Bass Coupling
Adjusts the “split” between wideband and bass action in the BS412 limiter when the LF
limiter threshold is exceeded. Lower values are wideband (which can result in “pumping”
with bass) and higher values will result in more reduction of bass.
RF Bandwidth Limiter
ITU-R SM.1268 compliance settings. The ITU-R SM.1268 (Stokkemask) standard is a
mask on the RF spectrum. Enabling this mask makes the RF spectrum less wide, which
reduces multipath distortion and improves reception. Using Stokkemask is mandatory in
the Netherlands, but it can improve reception in other cases as well.
Allow overshoots
Allows exceeding the mask by small amounts. Can improve audio quality.
Clipping
Controls for the composite clipper portion of the stereo generator, including SSB stereo
generation. These controls can both increase loudness and improve multipath perfor-
mance.
Composite Clipping
Global settings for the composite clipper
Clip demodulated at
Some radios may have an issue when demodulating peaks beyond a certain point. Adjust
this control to limit the maximum demodulated peak levels. Generally settings between
110% and 130% will have little impact on the audio.
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Stereo Coding
Enables various styles of Single Sideband (SSB) operation in the 38 kHz L-R stereo
subcarrier. This can improve multipath performance in some cases, but some receivers
may not respond well. Experiment with this setting to see what works best in your envi-
ronment.
Normal – Dual sideband
Fully symmetrical operation of the 38 kHz subcarrier. This is the default mode of opera-
tion.
SSB - Standard
“Typical” implementation in most audio processors. Very low frequencies (up to about
150 Hz) are modulated normally, anything higher than that is modulated only on the
lower sideband.
SSB - Smooth 12 kHz
A special implementation to improve SSB stereo operation. When a tone of 12666.7 Hz is
included in the L-R signal, the upper sideband center frequency (50666.7 Hz) is exactly
a multiple of the lower sideband center frequency (25333.3). In the RF spectrum, the
combined energy of these two frequencies creates much higher peaks.
If the L-R signal contains a tone at 1000 Hz, it will be modulated at 37000 and 39000 Hz.
Switching to SSB modulation would create signal only at 37000 Hz, which causes higher
peaks in the MPX spectrum and can be more likely to cause reception problems.
Smooth 12 kHz mode reduces the level of MPX frequencies above 50667 Hz, but leaves
lower frequencies as they are (dual sideband modulated).
SSB – Steep 12 kHz
Places a steeper filter at 50667 Hz in the MPX baseband, but otherwise similar to SSB -
Smooth 12 kHz.
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L-R Asymmetry
Unique to Omnia.SST, this mode allows dynamic adjustment of asymmetry in the L-R
subcarrier. In some cases traditional DSB results in higher MPX peak levels, and in other
cases SSB will have higher peaks. L-R asymmetry will perform “opportunistic SSB/DSB
switching” to minimize the peaks in the MPX waveform. This alone does not gain much
loudness (about .1 dB), but it minimizes distortion prior to the clipper, allowing the clip-
per to be driven slightly harder. The end result is approximately a 1 dB gain in loudness.
Note that when using MicroMPX with this mode, the bitrate should be increased by
approximately 64 kbps (for example, 384 rather than 320).
Maximum asymmetry
Similar to the various SSB modes, adjusts the maximum amount of asymmetry that will
be applied to the L-R subcarrier when in this “dynamic” SSB/DSB mode. As with SSB
operation, settings that are too high may cause performance issues with some receivers.
SCA
Omnia SST includes dual aural SCA generators. These require a special FM receiver and
are useful for broadcasting closed circuit services such as reading for the blind services, an
IFB/talkback for remote broadcasts, or metering telemetry from the transmitter. Frequen-
cies up to 96 kHz are supported with a 192 kHz sound card.
Use the soundcard options to define audio inputs for these SCA signals.
SCA Lowpass
Applies a lowpass filter to the SCA audio input. 7000 Hz would be a typical value. If this is
set too high, interference to other subcarriers (such as RDS) and crosstalk into the main
channel audio can result.
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RDS
Omnia SST includes a fully-featured RDS (RBDS) generator, capable of the same pow-
erful dynamic RDS functions found in many dedicated hardware RDS generators. This
panel controls all settings related to RDS.
Power Button
Enable or disable RDS generation.
Main
Global settings related to RDS generator operation
RDS Injection
Sets the level of the internal RDS generator. Typically this would be 4-5%.
Quadrature
Encodes RDS in “quadrature” mode. This lowers the total levels of the stereo pilot and
RDS subcarriers which can allow slightly more loudness.
Texts
Defines the text to be sent in the PS field (static or scrolling) and in the RadioText field
(if enabled).
RDS PS Text
The Program Service text (8 characters-can be static or scrolling)
RDS RadioText
Defines the string to be sent in the RadioText field. This feature is not supported by all
radios, but is often a duplicate of title and artist information scrolled in the PS text field.
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Constant Text
To transmit a constant text, just type that text in the appropriate field (PS Text or Radio-
Text) in the interface.
The PS Text must be at most 8 characters (but can optionally scroll) and the RadioText
can be up to 64 characters. Any extra characters are ignored.
Example:
POWER-FM
To show different alternating texts, write all the texts you want to be displayed in the
appropriate field in the interface. Split the different parts with a / character.
Example:
POWER-FM/88.5 FM
Note: The period character will be replaced by a space on radios that cannot display it.
To change the amount of time a text is displayed, put the requested time in front of the
text, as follows:
1.5s:POWER-FM
Displays the text POWER-FM for 1.5 seconds. When specifying a time in seconds, Omnia
SST will automatically make sure that the text is transmitted at least twice so radios can
receive it properly, even if that takes longer than the specified time.
3t:POWER-FM
Transmits this text 3 times. Omnia SST will transmit this text 3 times, then switch to the
next text. Note that it is possible to use 1t here, but it will lead to poor display on many
radios.
Example:
3s:POWER-FM/1.5s:88.5 FM
Displays POWER-FM for 3 seconds, then 88.5 FM for 1.5 seconds.
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Scrolling Text
Due to the slow transmission times, scrolling is only useful for PS text. Transmitting a
PS text takes a short amount of time (2 transmissions * 8 characters / 22 characters per
second) = 0.72 seconds. For RadioText, that's 2 * 64 / 22 = 5.8 seconds, which makes
scrolling useless for that field.
For left scrolling, add a < character at the front of your text. To scroll at higher speeds, add
multiple < characters. To scroll right, use the > character.
Examples:
<THE BEST HITS IN TOWN
<<THE BEST HITS IN TOWN AT HIGH SPEED
3s:POWER-FM/1.5s:88.5 FM/<THE BEST HITS IN TOWN
In the previous example, the first and last characters are displayed only very briefly. There
are two possibilities to solve this.
Adding spaces:
3t:<< THE BEST HITS IN TOWN (ends here)
It is also possible to use different timings for different portions of the scrolling. That
makes it possible to display the start and end for a longer period of time than intermediate
parts.
Instead of using just 2s: which sets the display time for every part of the scrolling, differ-
ent values can be set for different positions.
During scrolling, the first position is called '1', the second '2' etc. The last position is also
known as '-1', the one before that as '-2', etc.
One scenario might be to specify that positions '1' (start) and '-1' (end) display for 1.5
seconds, and in addition that everything in between (2 up to -2) should be broadcast 2
times (the minimum for good reception). That can be done as follows: 1=1.5s,2..-2=2t,-
1=1.5s:
Examples:
1=1.5s,2..-2=2t,-1=1.5s:<THE BEST HITS IN TOWN
3s:POWER-FM/1.5s:88.5 FM/1=1.5s,2..-2=2t,-1=1.5s:<<THE BEST HITS IN TOWN
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 48
Example:
||THIS TEXT IS WORD WRAPPED
If a word does not fit on the display (if it's longer than 8 characters), the remaining
characters are truncated. To avoid that, you can combine word wrapping with scrolling.
Example:
||<TRANSMISSION IS A LONG WORD
Now the scrolling problem from before re-appears: The start and end of the word are
displayed only very briefly. This can be solved using the timing settings described for
scrolling.
So, to display the text, every word 1.5 seconds, but if a word scrolls its start (1) and end
(-1) should be displayed 1 second, and the parts in between (2 up to -2) should be broad-
cast 2 times, that leads to: 0=1.5s,1=1s,2..-2=2t,-1=1s:
Example:
1.5s,1=1s,2..-2=2t,-1=1s:||<TRANSMISSION IS A LONG WORD
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 49
Omnia SST enables you to include this type of data, as long as you can make the text
available in a file. If you have that, there are 4 ways of including data from a file into your
RDS text:
\R"C:\Filename.txt"
Reads raw (unformatted) text from C:\Filename.txt, converts it to capitals, and inserts it
here in the RDS text.
\r"C:\Filename.txt"
Similar to \R, but does not convert the text to capitals. If your input file contains lower-
case characters, some radios may not display parts of the text.
\F"C:\Filename.txt"
Reads formatted text from C:\Filename.txt, converts it to capitals, and parses it as if it
were a part of the filled-in RDS text. Any special characters (<, ||, : etc.) are used for
formatting.
Only use this if you know what you are doing. Unexpected special characters in the file can
lead to text display oddities.
\f"C:\Filename.txt"
Similar to \F, but does not convert the text to capitals. If your input file contains lowercase
characters, some radios may not display parts of the text.
Unless you have a specific reason to do otherwise, you should only need \R.
Example:
POWER-FM/PLAYING/\R"C:\NowPlaying.txt"
The text files are read from disk each time when the sequence repeats. Using a local disk is
recommended so everything gets cached. If you use a remote disk, the process might stall
until it can access it. Do not use a network disk!
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Because of this possibility, there are 10 seconds between the moment when the file is read
and the moment when the contents of the text file start to be transmitted. To avoid that,
place the dynamic part at the beginning of the sequence - it repeats anyway:
\R"C:\NowPlaying.txt"/POWER-FM/PLAYING
Because song names can be long, you'll probably want to use the word wrapping and
scrolling trick described above:
1.5s,1=1s,2..-2=2t,-1=1s:||<NOW PLAYING \R"C:\NowPlaying.txt"ON/6s:POWER-FM
Special characters
As you can see above, some characters have special meanings in the Omnia SST RDS
generator. More specifically, the characters <, >, |, :, / and \ cannot be used in the RDS
messages without special considerations.
If you do need to use any of these characters, put a \ in front of it. So: \<, \>, \|, \:, \/ and \\.
Example:
\<POWER!\>
Info
Various RDS metadata flags describe characteristics of the programming and RDS data
stream. Some of these are not likely to be implemented on many receivers.
RDS PI
Enter the PI code of your radio station.
The PI code controls things like automatic switching to Alternative Frequencies. It's
important to set this up correctly. Here's why:
If you use the same code as another station in your area, you might inadvertently 'steal'
listeners from them (the radio might automatically switch to your station if its reception
is better) or you might lose listeners to them in areas where their signal is better. This is
extremely annoying to listeners.
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If you change your code at a later time, some radios where your station has been pro-
grammed on a preset button may turn the sound off because they 'think' it's the wrong
station, even if reception is perfect. Only reprogramming the preset button helps in that
case - but not all listeners will realize this.
The PI code needs to be entered as a hexadecimal number. The default is FFFF, which
means that it should be disabled - but not all receivers adhere to this.
The first 2 digits of the PI code are generally a country code, but again, many receivers
ignore it. Whatever you do, make sure that the last 2 digits are unique in your area, and
that they are not FF.
That said, PI codes are regulated differently in different countries. Please check with your
appropriate broadcast authorities what you need to do. In the US, the PI code should
generally correspond to your call letters.
A handy calculator can be found here: http://www.w9wi.com/articles/rdsreverse.htm
PTY
The type (format) of program that you're broadcasting.
Many receivers can automatically search for some type of broadcast, such as a news
broadcast. If you don't always broadcast the same type of program and you want to update
this setting, make sure that Dynamic PTY is enabled.
The list of PTY codes is very different for RDS (EBU, used in Europe) and RDBS which is
used in the US. The PTY codes are changed when you change the Pre-emphasis time.
RDS TP (On/Off)
Indicates that the station regularly broadcasts traffic bulletins.
RDS TA (On/Off)
Indicates that the station is CURRENTLY broadcasting a traffic bulletin.
Alternative Frequencies
Define up to 25 alternate frequencies that this station’s programming is carried on. Gen-
erally used in European broadcast networks, and not consistently implemented in the US.
Some radios use signal strength when scanning for AF, others appear to use the PI code.
Only utilize these fields if you understand that receiver behavior with the “AF” feature will
vary depending on a number of factors. Some receivers ignore AF entirely.
Originally, UECP was used over an RS232 connection, but many products are now sup-
porting it via IP. Omnia SST only supports the latter.
Mode (UECP/ASCII)
Selects between control via UECP or ASCII. Many applications now support UECP direct-
ly, but for those that do not, an ASCII mode is supported for sending commands in plain
text.
Port
Specifies the IP port where Omnia SST should listen for RDS encoder traffic.
Specific Address
UECP can send commands via multicasting to multiple RDS encoders. If this feature is
utilized, this addressing information can be used to target a subset of encoders.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 53
Error behavior
Controls what the encoder does should it lose a UECP connection
µMPX
MicroMPX or “µMPX” is an Omnia exclusive, and allows transmission of an entire com-
posite MPX baseband via IP in streams as low as 320 kbps. Omnia SST is the first product
to feature this technology. The Omnia SST GUI includes controls for a MicroMPX server.
A standalone decoder application is used as the client, and a stand-alone Encoder is also
available . (see appendix later in this document for more information)
Power button
Enables or disables MicroMPX encoding and transmission from Omnia SST.
µMPX
Global settings for the MicroMPX server
Bitrate
The recommended bitrate (and the default) for MicroMPX is 320 kbps. Lower bitrates
will decrease quality, while higher bitrates will increase quality at the cost of bandwidth
use. Bitrates from 300-400 kbps are available, 320 kbps and higher are supported.
Streams
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 54
This section configures the destinations for the MicroMPX streams. Multiple destinations
are supported simultaneously.
On/Off
Enables or disables sending MicroMPX traffic to each destination individually.
IP
Specify the IP address of the host running the MicroMPX client. Multicast and broadcast
IP addresses are also supported (any valid multicast IP, or x.x.x.255 for broadcast to an
entire subnet).
Port
Specify the IP port on which MicroMPX traffic will be sent. The default is port 8854 and
generally should not need to be changed unless some other application is utilizing port 8854.
Error Correction
The MicroMPX server incorporates an advanced error correction scheme which can send
error correction packets along with the audio, with a tradeoff of increased bandwidth
use. These settings adjust how frequently these packets are sent, and how many are sent
in each group. A bandwidth limiter is also available to prevent error correction overhead
from exceeding available bandwidth.
Attempts to limit the bandwidth use (including error correction) below the specified
value. This value will change dynamically depending on the error correction span and
overhead settings, so use this setting to predict approximately how much bandwidth your
MicroMPX stream will ultimately consume when applying error correction.
Presets
Like most other audio processors, Omnia SST includes a system for managing presets.
These presets cover several different categories of parameters including global I/O set-
tings, the “Cleanup” sections, and the processing sections.
Click the “Presets” button on the navigation bar on the access the preset library.
Depending on which portion of the interface you are in, the available list of presets will
change. For example, if you click on “Presets” from the “FM Options” page, you’ll see
IO configuration presets. To see the entire list of presets for all categories, start from the
“Home” screen.
Loading a preset
The preset window is split into several categories: Recent, Processing, and Clean up.
“Recent” will display the most recently loaded presets from any section. The “Processing”
and “Clean up” categories contain presets specific to those areas.
If you attempt to load a preset from an area of the software where there are no presets
defined, you will see a message indicating “You have not saved any presets that contain
configuration settings yet”.
To load a preset, simply click it. The sections available to load from that preset will be
displayed. Click the appropriate button for the section to apply that preset to (Cleanup,
Processing, Full FM preset, All settings).
Note: The currently loaded preset will be highlighted by a yellow frame. If sections from multiple
presets are loaded, multiple presets may be highlighted.
Save Current
Click “Save Current” to save a custom preset based on the current settings. This will save
parameters from all sections simultaneously.
Enter the desired name for the preset, along with any other descriptive data (preset
author, intended preset format, description) then choose to save as a new preset or over-
write the currently loaded preset.
Note: To keep presets from being deleted by mistake, there is currently no way to DELETE a preset
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 56
from within the interface. Preset files are stored in the system’s “Users” directory for the currently
logged in user under “OmniaSST.Presets”. The list of available files is read when SST starts. If
you wish to delete presets that you no longer use, such as iterations spawned in the process of
creating a master preset, use windows explorer to find and delete old presets, then reload the SST
application
Import
Click the arrow to import a preset from a file.
Search
Click the magnifying glass to search the list of available presets.
Close
Click the “X” to close the preset window.
Metering
Metering is available in two forms: A dedicated meter panel displaying all available me-
ters, and a meter banner which displays metering specific to whichever processing section
is being adjusted. The following is a list of what meters are available, and their function is
generally self explanatory.
Input Scope
Displays the input waveform. Audio detected as clipped will be displayed in red.
Declipped Scope
Displays the output waveform of the declipper.
Output Scope
Displays the final output waveform after all processing stages.
Input Meter
Displays the input audio level. Actual input level will be displayed in light green, while
gain added in soundcard configuration will be displayed in darker green.
Output
Displays the output audio level. The light part is after any added gain. The dark part
reflects the level after processing and must not exceed the top (0 dB).
MPX Power
Reflects the MPX power level of the FM signal in dBr.
FM
Displays MPX output level, as well as peak demodulated left and right levels.
Delossifier
Indicates the likelihood that the input audio is from bit-reduced source material (MPEG
L2, L3).
Noise Gate
Shows noise reduction (gating) per band.
Natural Dynamics
Indicates the activity of the “Natural Dynamics” section. Light grey indicates the amount
of expansion, dark grey indicates the total amount of possible dynamic expansion in each
band.
Phase
Displays the amount of phase correction applied by the “Azimuth” section.
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Auto EQ
Gives an overview of the amount of boost or cut being applied to each band by Auto EQ.
Grey indicates boost, red indicates cut, and purple lines indicate the average boost or cut
recently applied in each band over time.
AGC
Shows the amount of gain reduction occurring in the AGC (Automatic Gain Control)
section.For most “Dynamics” sections (AGC, Multiband, Singleband), the action of
both “Compression” (Grey) and “Limiting” (Yellow) are displayed, along with “Gating”
displayed as a horizontal bar along the bottom in purple.
Stereo
The left meter indicates the amount of “multiband” stereo enhancement, while the right
meter indicates the amount of “wideband delayed” stereo enhancement applied.
Equalizer
This display mirrors the one in the Equalizer panel--displaying the EQ curve being ap-
plied, as well as an RTA (real time analyzer).
BIYF
Displays the amount of enhancement applied in each “Bass In Your Face” section.
Harmonics
Amount of extra bass harmonics generated in the “Phat Bass” section.
Multiband 1 & 2
Displays the gain reduction of each band in the the two multiband dynamics sections.
SB
Gain reduction applied in the final single band dynamics section prior to the clipper.
Clipper
Indicates RMS gain reduction in the final clipper.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 59
Encoder:
♦♦ Start Omnia SST.
♦♦ Click on the Omnia SST tray icon to open the web interface:
♦♦ Under Settings -> Sound cards, set the sample rate to 192 kHz.
♦♦ Under Settings -> FM Options, enable MAIN (which enables FM processing), set the
Pre-emphasis time as appropriate for your location and at turn on the stereo generator
(Encode Stereo). You may also want to enable RDS too, under Settings -> RDS).
Make sure that you see the pilot in the MPX display on top of the screen. Audio is also
present in the screenshot below:
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♦♦ Type 127.0.0.1 for the first MPX output, and enable µMPX. Leave all the other set-
tings at their defaults (320 kbit/s, 5.0 sec Keyframe interval, no error correction, no
rate limiting). Note: 127.0.0.1 is the loopback Internet protocol (IP) address also
referred to as the “localhost.” This address is used to establish an IP connection to
another program running on the same computer..
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 62
If it's not already on, Turn on the “power” button above the uMPX settings. Con-
gratulations! You’re now streaming a µMPX signal and are half the way there! Now that
you are encoding a stream, you need to set up a decoder to receive it.
Decoder
The µMPX Decoder is a separate application, designed to provide simple IP and sound-
card configuration for receiving a signal from a µMPX encoder, and is engineered to use
very little system overhead. SST comes with a single license of both µMPX Encoder and
Decoder. Additional, stand-alone licenses are available.
Starting MPX
After configuring your encoder, launch the MicroMpx Decoder executable from the
desktop icon, then go to your system tray and click on the small, blue “u” icon to open the
browser interface:
Note that you may need to change the port number from the default (8080)if you have
SST running on the same machine or the SST interface will open in the browser instead of
the uMPX decoder. Try Port 8081 for the decoder interface.
The µMPX interface will open within your chosen default browser.
There are three choices from the main menu: License, Sound Cards and µMPX:
Within the SoundCards and µMPX screens you will see three windows at the top of the
screen that show:
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 63
Waveform Window
When µMPX is decoding a signal, you will see the waveform of this signal represented in
this middle window.
MPX Screen
When µMPX is decoding a signal, you will see the entire baseband signal of the output
represented in this window.
Home Button
The home button in the upper left of the screen is a convenient way to navigate back to the
three main choices in the main menu.
Warning sign
The triangular warning sign is where µMPX will provide diagnostic information regarding
network issues/outages, or issues regarding your sound card configuration or licensing
information. If the warning sign glows red, click on the icon to see a listing of issues or
errors that µMPX is encountering.
Help is available by pressing the question mark icon in the upper right of your browser,
then clicking on an item you need more information on.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 64
LICENSE
Request Code
Copy your request code and provide to [email protected]. You will be
given an activation code to place in the lower box.
Activate License
Once given your code, press Activate License. Your license ID and expiration date will
appear below.
Fm Output
Choose the 192kHz capable sound card that will be used to feed your FM transmitter or
exciter.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 65
Sound Cards
Use this slider at to adjust the MPX output of your sound card that feeds the exciter or
transmitter.
Use these together to compensate for sound cards that are not DC-coupled or might have
other non-linearity. To ensure the best results, even sound cards advertised as DC-coupled
should be checked for non-linearity before being fed into a transmitter.
To use FM tilt correction with test signals: Turn on both FM tilt correction and test
generator. On an external oscilloscope monitoring the sound card’s MPX output, or using
the onboard , monitor the output of µMPX while choosing Smooth Square from the
dropdown menu of Test Signals available on the right side of the screen.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 66
Test Signals
Turning the test generator on will interrupt any feed from the decoder[GS2] and place test
tones on the output of the chosen sound card. This drop-down menu provides a number
of useful test signals to test the linearity of your sound card and the rest of your transmis-
sion chain.
Test signals include: Smooth Square Wave, 400Hz Sine Wave, 5023.1 Bessel Null Tone,
6360.8 Bessel Null Tone, 8665.6 Bessel Null Tone, 13587.0 Bessel Null Tone, and 31189.4
Bessel Null Tone. Sawtooth with Pause and Sawtooth flat bottom are designed to be used
for instances of µMPX that may be feeding an AM modulator.
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RC Slider
Using the “Smooth Square” signal, adjust the RC slider as necessary to achieve square flat
top waveform at frequencies of 30Hz, 400Hz and 10kHz on the test signal generator to
the right
MPX Menu
To decode a signal, choose your incoming port number, (default port is 8854) and toggle
the “power” button in the upper left hand side of this section.
Delay Slider
Use this control to adjust up to 9 seconds of buffering delay. This may be useful in situa-
tions where extra buffering is needed, or if you are trying to time align signals.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 68
Backup Player
µMPX Decoders include a simple backup playout system. This system allows a single
MPX audio file to be played in the event that the encoded stream feeding the decoder is
lost. Source audio files must be composite audio (MPX) format. µMPX checks if the file
is valid, i.e. contains only one channel at 192kHz. It gives an error if you try to put in a
file that does not comply with this. The backup audio file should be stored on the same
PC that is running the decoder application. The path location and name (eg; c:\audio\
backupfile.wav) of that file should be placed in this window (above). The file is played
in a loop. When the stream comes back and results in properly decoded audio (so at least
one keyframe was received), the backup file will fade out and the stream will fade in. If the
stream drops out again, the backup file playback will resume playing from where it was
last stopped.
Backup timeout
This control specifies the silence threshold level and timeout before switching to the
backup file.
Choose a value by moving the slider between 1-120 seconds of silence before the Backup
File begins playing straight to the output of the µMPX decoder’s chosen sound card.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 69
In this case the IP address of the decoder is 172.22.16.101 Streaming over a LAN
(without going through firewalls or port forwarding) is the next logical test to
demonstrate you are sending and receiving the µMPX signal.
(You will also need this info if you plan to use the public internet!)
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♦♦ Returning to your PC running Omnia SST, enter the IP address of the PC you want to
stream µMPX to, as in the example below:.
♦♦ Now, from the second (external) PC, run the Decoder application just as you did for
the internal test. Assuming you are running on a LAN, you can setup the decoder in
the same way as above, except you are now sending the µMPX signal over a network.
Assuming your sound card is properly set on the second machine, you should now be
getting an MPX signal out from your audio card, and you’re done!
♦♦ With enough bandwidth, you can send µMPX signals wherever you have a clean IP
connection. Whether you are sending over a LAN, a WAN, using a VPN, or even if
you are using the public internet, a single Omnia SST system encoding µMPX can be
delivered to multiple µMPX decoders on different machines at different destinations.
Your distribution can be as simple as a single point to point connection, or a “one to
many” setup with dozens of decoder endpoints. µMPX harnesses standard internet
protocols, so possibilities are gated only by what your network infrastructure and IT
skills will support. Note however that each Decoder will require a separate license.
PLEASE NOTE!
While it may be possible to use the public Internet with µMPX, we can make no guarantees
that performance, bandwidth or up-time will be adequate for your needs.
Low budget outfits that have few or no STL choices may consider the Internet to be perfect-
ly adequate. Major operators with real revenue on the line may consider the Internet a last
resort backup link only. Results will vary. The risk is on you.
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♦♦ If your Decoder machine is on the internet, you’ll need a public IP address, and you’ll
probably need to do some port forwarding. The easiest way to find your public IP
address is to use a service such as whatismyip.com, whatismyip.org, whatismyipad-
dress.com, …:
Once you find the correct menu, set things up such that the port number that you are
using on the sender (by default, port 8854) is forwarded to the local IP address (see
ipconfig above) of your decoder pc.
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In some routers, if you want to map a port to a different external port number than
what you are using internally (for example, if you need a port that is already in use, or
if you want to send multiple µMPX streams to the same network), you might need to
use a ‘Virtual Server” instead of “Port forwarding”. In the example above, you may not
be able to enter what a certain port should map to, so if you want map ports 8854,
8855, 8856 and 8857 to different decoders on your local network, you may need to use
Virtual Server instead.
For comprehensive notes of how to setup port forwarding on nearly any router imag-
inable, go to http://portforward.com/router.htm.
In our example, the 10 recovery packets are generated immediately for every 100 packets
that are sent, and - depending on your network - it could be that sending them all at once
to the other side will fail and the network will drop some of the packets. Because of that,
the “Rate limiter” will throttle the speed at which packets are sent (both for the normal
and the recovery packets). Omnia SST automatically calculates the minimum bandwidth
needed for both normal and recovery packets. Recovery packets are slightly bigger than
normal packets so it’s beneficial to allow some headroom. In our example, settings of 360
kbit/s allow for a 320 kbit/s audio stream with decent forward error correction.
Please note that for these settings, at least 100 packets must have arrived on the other end
before reconstruction can start. So if the first packet is dropped, it will take time before it
can be reconstructed. The decoder must have enough buffer latency for it to still be able
to use the recovered packet - if the audio at that point should already have been played,
you’ll still have a gap in your audio. The current version of the software sends 94 packets
per second, so for every 94 packets or so you need to add a second of latency. You can set
the latency (in seconds) for the decoder with option -d. So, in the example above with
10 recovery packets per 100 normal packets, set the latency to at least about 1.5 seconds
(with -d 1.5, see below). The extra half second is there to reduce network jitter. Decoder
default latency is 1 second. Generally, the higher the latency and buffering you use, the
fewer problems from dropped packets. The inherent tradeoff of more error correction is
greater signal delay.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 75
One final consideration for latency vs. bandwidth. Packet loss tends to occur in bursts.
So, the same number of recovery packets often provides the same “ballpark” amount of
recovery, regardless of the number of packets over which they are calculated. So, 32/4 and
64/8 will give you the same network bandwidth, 64/4 gives you less network bandwidth
with almost the same quality of recovery. But, you need to allow more time on the decoder
end for the data to be recovered. 64/8 has a much smaller chance of losing audio, but the
tradeoff is incurring more latency.
Latency
The default latency is set to 1 second. If you don’t use error correction at all, you can
attempt to set it lower. We have seen good results with values down to 0.2 seconds
(example below), but results depend on many variables outside the control of SST, most
importantly, your network topology and the quality of the hardware which you run the
encoder and decoder on.
Backup file
If the connection drops, the µMPX decoder can switch to playing a backup audio file that
you can provide.
--backup-file /path/to/file.wav points to the file (which must contain 192 kHz MPX data,
in WAV or FLAC format). You will need to place this file in the same directory as the
decoder.
--backup-timeout is the time in seconds of dropped signal before the backup file is trig-
gered.
If multiple dropouts occur, the file will keep playing in a loop, picking up where it left off
after the last dropout.
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TRADEMARKS
Omnia Audio, Omnia SST, µMPX, and the Omnia logo are trademarks of TLS Corp. All
other trademarks are the property of their respective holders. Software functions and
specifications may change without notice.
NOTICE
All versions, claims of compatibility, trademarks, etc. of hardware and software products
not made by TLS Corp which are mentioned in this manual or accompanying material are
informational only. TLS Corp makes no endorsement of any particular product for any
purpose, nor claims any responsibility for operation or accuracy. We reserve the right to
make improvements or changes in the products described in this manual which may affect
the product specifications, or to revise the manual without notice.
UPDATES
Omnia SST software is routinely updated with new versions to add features and fix bugs.
Check the Telos Alliance web site for the latest. We encourage you to sign-up for the email
notification service offered on the site.
FEEDBACK
We welcome feedback on any aspect of Omnia SST or this manual. In the past, many good
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Software Licensing
END-USER SOFTWARE LICENSE AGREEMENT
This Agreement is a legal agreement between you (either as an individual or an entity,
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1. GRANT OF RIGHTS: TLS Corp. grants you only those rights expressly granted to
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OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 78
3. CONFIDENTIALITY:
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OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 79
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OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 81
The maximum liability of TLS Corp. to any person, firm or corporation in connection
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governmental charges in addition to freight, insurance and installation charges and
import or export duties.
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 82
Software Notice:
Omnia SST and MicroMPX uses several third party libraries:
ASIO by Steinberg
The above copyright notice and this permission notice shall be includ-
ed in all copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EX-
PRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MER-
CHANTABILITY,FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN
NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY
CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,
TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE
SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
-----
spdlog - Copyright (c) 2016 Gabi Melman.
The above copyright notice and this permission notice shall be includ-
ed in all copies or substantial portions of the Software. THE SOFTWARE
IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FIT-
NESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR
OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARIS-
ING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTH-
ER DEALINGS IN THE SOFTWARE.
-----
fmtlib - Copyright (c) 2012 - 2016, Victor Zverovich. All rights re-
served.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
-----
libsodium - Copyright (c) 2013-2017
THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRAN-
TIES WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE
FOR ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY
DAMAGES WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETH-
ER IN AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION,
ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS
SOFTWARE.
-----
OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 84
-----
pffft - Copyright (c) 2013 Julien Pommier ( [email protected] )
FFTPACK license:
Copyright (c) 2004 the University Corporation for Atmospheric
Research ("UCAR"). All rights reserved. Developed by NCAR's
Computational and Information Systems Laboratory, UCAR,
www.cisl.ucar.edu.
-----
libvorbis - Copyright (c) 2002-2015 Xiph.org Foundation
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
-----
libflac - Copyright (C) 2000-2009 Josh Coalson
Copyright (C) 2011-2016 Xiph.Org Foundation
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
-----
Intel IPP - Copyright © 2017 Intel Corporation.
Use and Redistribution. You may use and redistribute the software
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Third party and other Intel programs. "Third Party Programs" are the
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-----
libsndfile - (C) 1999-2017 Erik de Castro Lopo
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OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 88
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OMNIA SST USER’S MANUAL V8.53-DECEMBER 2017 | 90
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