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INVENTING DIGITAL ACOUSTICS

The Optimizer loudspeaker/room optimization technology is at the heart of Trinnov’s products for professional studios, movie theaters, high-end Hi-Fi and home theaters. With its modern approach to acoustic measurements, analysis and processing, it solves the Loudspeaker/Room acoustic equation.
Loudspeaker/ room
optimization
The Optimizer loudspeaker/room optimization technology is at the heart of Trinnov’s products for professional studios, movie theaters, high-end Hi-Fi and home theaters. With its modern approach to acoustic measurements, analysis, and processing, it solves the Loudspeaker/Room acoustic equation.

TRUSTED BY THE MOST DEMANDING AUDIO PROFESSIONALS

The Optimizer is the only acoustic optimization technology that has been recognized by world-class sound engineers, movie theaters, and audiophiles. Other technologies analyze and treat the room and the loudspeakers separately, but the Optimizer analyzes and compensates your reproduction system as a whole. Problems are identified, cross-analyzed and compensated with the most efficient and best-sounding compromise.

ADVANCED FINE TUNING

Sound reproduction is very complex and subjective. Depending on how far you want to dig into its features, the Optimizer is either a straightforward and easy-to-use automatic compensation system or an incredibly flexible and powerful tool, including numerous advanced filters parameters, target curves and manual eq’s, all being recomputable on the fly. By following an iterative installation procedure, the Optimizer achieves results beyond any expectations in record time.

MODERN ACOUSTIC MEASUREMENTS IN 3D

Unlike Old-fashioned real-time analysis and 1/3rd-octave correction methods, the Optimizer uses MLS signals to measure the full impulse response of every loudspeaker in the room. This adds the time dimension to the frequency response and enables the Optimizer to see the full picture of the loudspeaker’s behavior in the room. Trinnov’s 3D microphone identifies the real positions of the loudspeakers in 3D.

CUTTING EDGE ACOUSTIC ANALYSIS AND PROCESSING

Trinnov’s state-of-the-art time-frequency analysis algorithms identify room modes, first reflections, and late reverberation. Every acoustic aspect is analyzed and compensated with a specific technique. All the subtlety of the Optimizer resides in knowing which defects can be corrected with acoustic transparency

ACOUSTICAL GRAPHS

The Optimizer automatically generates complete PDF reports every time a preset is written. These reports can be downloaded from Trinnov processors to a USB stick or Hard Drives via FTP.

PHASE RESPONSE IMPROVEMENTS

The Optimizer improves the frequency response of the loudspeakers, both in amplitude and phase.Trinnov corrects the tonal balance to obtain a neutral timbre for every speaker, working in the time domain to achieve a high-resolution stereophonic image with well-focused phantom sources. The loudspeaker’s sound (including the early reflections) and the room (energy response) are separately equalized, opening up the listening window.

TARGET CURVE: MEET YOUR LOUDSPEAKERS CHARACTERISTICS

The Optimizer automatically defines the filters that will achieve the required frequency response defined by your target curve.This is particularly useful in post-production studios to comply with SMPTE standards (X-Curve). Phase and group delay targets can also be defined, making the Optimizer a unique tool for sound system designers.

MULTI-POINT: BUILD A WIDER LISTENING AREA

Trinnov’s sophisticated multipoint algorithm can take into account the measurements of different positions to perform the optimization. A higher weighting may be assigned to the most important listening position(s), and lower weightings to the remaining points.

ACTIVE ACOUSTICS

The Optimizer starts with acoustic phenomenon that are mostly deterministic, and gradually moves to the ones that are mostly statistics. This automated room correction combines both IIR filters and FIR filters. The IIR filters allow for very accurate equalization in the low range, while the FIR filters work full range.

A SUBTLE APPROACH TO ACOUSTIC PHENOMENON

The Optimizer uses a clever combination of modern techniques to generate complex sets of digital filters and achieve the best loudspeaker/room compensation. Its intelligent acoustic analysis engine automatically computes FIR and IIR filters to dramatically improve the consistency of direct sound against late reverberation. Full-phase, time domain techniques are applied compensating for the loudspeaker’s group delay and very early reflections (deconvolution), while following reflections are left untouched.

CORRECTION OF EARLY REFLECTIONS (DIRECT FIELD)

The Optimizer analyses the measurements in the time-frequency domain to identify Early Reflections. Depending on their amplitude, frequency, direction and arrival time, the Optimizer will compensate for them to a certain extent, or not try to compensate for them. After this process, each loudspeaker’s response is “clean” from the early reflections that it is possible to correct with digital technology. The other reflections are not touched.

CORRECTION OF THE ROOM ENERGY

In this second stage, the Optimizer analyzes the measurements in the frequency domain only (the response of the system in steady state).

Compensation of Resonance Modes (in the low range):

The Optimizer identifies resonance modes in the range where they can be differentiated, roughly up to 300Hz. It applies individual filters to compensate each resonance mode.
Smoothing of the reverberation (in the mid and high range):
The Optimizer analyzes the room’s frequency response, related to the coloration of the room’s reverberation. Another filter is applied to compensate for this coloration smoothly.
All the subtlety of the Optimizer resides in its knowledge of the defects that shouldn’t be tried to correct for without creating even more problems.

In this second stage, the Optimizer analyzes the measurements in the frequency domain only (the response of the system in steady state).

Compensation of Resonance Modes (in the low range):

The Optimizer identifies resonance modes in the range where they can be differentiated, roughly up to 300Hz. It applies individual filters to compensate each resonance mode.
Smoothing of the reverberation (in the mid and high range):

The Optimizer analyzes the room’s frequency response, related to the coloration of the room’s reverberation. Another filter is applied to compensate for this coloration smoothly.

All the subtlety of the Optimizer resides in its knowledge of the defects that shouldn’t be tried to correct for without creating even more problems.

3D simulations

When a loudspeaker produces a wavefront in a room, the walls produce secondary wavefront. At the beginning, it is easy to identify each elementary reflections but after some time, the reflections are so numerous that it becomes impossible to separate them, it is the reverberation.
The Optimizer compensates separately and with different methods the early reflections and the reverberation. Deconvolution provides best results when only applied to early reflections, while minimal phase (or linear phase) equalization provides best results when applied to the reverberation.
When a loudspeaker is placed in free air or an anechoic chamber, only one front wave is produced at the listening position
Let’s consider the first reflection produced by a wall placed immediately behind the loudspeaker. The reflection against the wall creates a secondary wavefront.
When the loudspeaker is producing a single pulse, two wavefronts are produced at the listening spot. When this condition is compensated with deconvolution techniques, the second wavefront is strongly canceled at the listening position, where any other equalization method would fail.
The result of deconvolution leads the loudspeaker to fire a second time after producing the primary pulse and to produce a second pulse whose wavefront is the identical inverse to the wavefront of the reflection. The inversed wavefront produced by the loudspeaker cancels the reflection, and the original single wavefront is retrieved.

«The 3D microphone is amazing; it knows exactly how far your speakers are placed, down to the azimuth and individual height, and you can choose to let the system guide you to move them or let the software do it physically.
There was no way, going forward, I would work in all these different spaces I do without this box.»

 

Daniel Pinder music editor

Thor: Ragnarok, X-Men, The Dark Knight, Pirates of the Caribbean, Captain America

3D
microphone

Speakers localization

The flat response is guaranteed by rigorous quality control and FIR individual compensation filters.
-Every single capsule mounted in Trinnov’s microphones is measured and compared to a benchmark.
-During the final stage, unique compensation files are generated and ensure a perfect consistency between capsules and microphones with a flat frequency response (within +/-0,1 dB) on the 20Hz-24kHz frequency range. Compensation files are available for download from our server or direct transfer over the network onto any Trinnov Processor.

The measurement microphone is one of the most critical components of a loudspeaker/room calibration system. Trinnov’s sophisticated algorithms not only rely on very accurate acoustic measurements but also on the ability to localize speakers positions and to detect early reflection provenance.
Trinnov’s 3D measurement microphone is the result of our extensive research in 3D sound. 
We invested a significant amount of time to design a tailored measurement system that could gather very precise information regarding the spatial position of multiple sources. This is called triangulation. Each of the four omnidirectional capsules is placed at the same distance from each other, forming a tetrahedral pattern. This allows us to localize each speaker’s placement on both horizontal and vertical planes.
-To achieve this challenge, the microphone consists of 4 capsules mounted at the top of thin brass tubes to avoid diffraction.
-The capsules form a tetrahedron figure, ideal for identifying distance, azimuth, and elevation altogether.
-To optimize spatial accuracy, all the capsules responses are analyzed and compared recursively to find the best possible matches of 4 and reach a spatial resolution below +/-2° in every direction.

2D/3D
loudspeaker remapping
2D/3D loudspeaker remapping opens new perspectives in high-performance immersive sound, ensuring a unique experience by putting the listener right in the middle of the action, in an incredibly realistic, life-like and holographic multidimensional soundscape.

Thanks to the 3D measurements capabilities of our microphone, speakers original placement can precisely be localized in the room, regarding distance, azimuth, and elevation.
According to this reported information, a more accurate speaker placement can be retrieved.
While 2D remapping deals mainly with the horizontal plane, 3D remapping will manage both azimuths and elevations..
A 3D spatial remapping matrix is computed from the speaker’s coordinates and applied to the input signals in real time to virtually compensate for inaccurate speaker placement according to the content’s format. It guarantees optimum spatial resolution and faithful tridimensional sound staging.

Our unique remapping technology satisfies many functional requirements

Formats Interoperability: the remapping is the only solution to format war, as it ensures inter-compatibility between existing and future 3D audio standards, with an optimal use of speakers and the best possible spatial resolution.
Formats Scalability: If the number of inputs is different from the number of outputs, one could describe the remapping technology as a universal up/downscaling algorithm for faithful 3D audio reproduction.
Architectural constraints: even with format-specific speaker layouts ranging from 5.1 to any possible 3D format, the remapping will compensate for incorrect speaker placement imposed by typical room shapes and provide a dramatic improvement in the reproduction of the soundstage.

Spatial remapping

Spatial Remapping is a technology to adapt multichannel 2D or 3D sound on any loudspeaker layout. An optimal reproduction of multichannel sound is obtained only if the loudspeakers are arranged according to the specified format.
As an example, in the ITU recommendation for 5.1 surround sound, the center speaker is at 0 degrees, the left and right speakers are at +/-30 degrees and the surround speakers are at +/-110 degrees. Unfortunately, this recommendation is incompatible with many listening situations such as homes or location recordings.

The ITU recommendation has been developed to overcome a limitation of multichannel sound, where the spatial environment is described by a mean to reproduce it: loudspeaker at predefined positions and channels to feed them. If the loudspeakers are not at their correct places, the reproduction is incorrect. The Spatial Remapping technology overcomes this limitation of multichannel by providing correct imaging on any reasonable speaker arrangement.

– During a radiation step, a unique acoustic field is associated to the discrete multichannel signals. The radiation is a linear process recreating the acoustic field produced by ideal loudspeaker perfectly respecting a predefined loudspeaker layout such as ITU recommendation. The radiation process provides the Fourier-Bessel coefficients of the acoustic field resulting from the mutual contribution and interaction of all the channels. This is a very powerful step as the acoustic field representation is independent of the original multichannel format and the loudspeaker layout.

– During the decoding step, optimal loudspeaker feeds are derived from the acoustic field according to the Spatial Replay method.

Trinnov’s 3D remapping technology is fully automated thanks to our 3D microphone that measures the actual 3D positions of the loudspeakers. The distance is evaluated within 1cm from the propagation time for the wavefront emitted by the loudspeaker to reach the 3D acoustic probe. The angles (azimuth and elevation) are measured with less than 2? error from the analysis of the orientation of the wavefront crossing the 3D acoustic probe.

FOURIER-BESSEL AND SPHERICAL HARMONICS

The Remapping technology of the Optimizer is based on the ability to calculate the acoustic field that is produced by a set of loudspeakers. This calculation is possible thanks to the Fourier-Bessel decomposition of the acoustic field into a certain number of coefficients that correspond to the spherical harmonics. Just as the Fourier decomposition is commonly used to analyze a signal in the frequency domain, the Fourier-Bessel decomposition can be used to analyze an acoustic fieldin the space domain, by decomposing into a sum of elementary radiation patterns that are referred to as spherical harmonics in mathematics.

The function that provides the resulting acoustic field from the input signals is called a “radiation matrix”. In a pseudo math notation: Input Signal * Radiation Matrix = Acoustic Field

REAL RADIATION MATRIX

The Optimizer first computes the radiation matrix of the real system, in other words the radiation matrix that corresponds to the measured loudspeaker positions. This is possible because the Optimizer knows the exact positions of the loudspeakers in 3D.

This radiation matrix for the real system allows to calculate the actual acoustic field that is produced by the measured loudspeaker placement.

IDEAL RADIATION MATRIX

On the other hand, the Optimizer can calculate the radiation matrix for the reference placement, because the loudspeaker positions of the reference placement are, by definition, clearly defined.

This radiation matrix for the reference system allows to calculate the ideal acoustic field that would be produced if the loudspeakers were positioned correctly, according to the reference placement.

REMAPPING MATRIX

The last stage is to find out the additional processing that should be applied to the input signal in order to obtain the ideal acoustic field from the measured loudspeaker system. This is done by inverting the Real Radiation Matrix:

Remapping Matrix = Radiation Matrix of the ideal system * (radiation matrix of the real system)-1

CONCLUSION

This Remapping Matrix is computed once (after the loudspeaker positions have been measured) and applied in real time to the input signals to compute the output signals that should be sent to each loudspeaker in order to obtain the reference acoustic field.

Note: in the case where the number of inputs is different from the number of outputs, one could describe this remapping technology as a universal downmixing/upmixing algorithm for 3D audio reproduction.

See Trinnov’s AES Convention Paper 6375 for a detailed explanation of louspeaker remapping.

Loudness Metering /
Smart Meters
The SmartMeter is a software option available for Trinnov’s Pro Audio processors: the ST2 Pro, the D-Mon Series, and the MC Series. It is available as a stand-alone metering solution or in addition to Trinnov’s acclaimed loudspeaker/room Optimizer.
TIME CODE-AWARE LOUDNESS METERING

Most typical loudness real-time loudness meters require to measure a project from beginning to end without rollbacks or time jumps to obtain its Integrated Loudness level. Even if pausing is possible with most instruments, this method is not ideal.
Trinnov unique time code-aware SmartMeter takes loudness metering to a new dimension. The measurement automatically starts and pauses following the playback system, continuously providing consistent values, whether the operator jogs, shuttles or rewinds through the project.
AES134 Convention e-Brief: Timecode-Aware Loudness Monitoring: Accelerate Engineers Everyday Workflows
Another huge advantage of Time Code synchronization is the possibility to TRIGGER REAL-TIME ALERTS and update them as the mix is adjusted. Alerts thresholds can be configured to meet requirements of many organizations and companies.
Having Loudness and Peak values stored and time-stamped also makes session recall relevant. Projects can be paused, shared over a network and resumed in other studios in the actual very same conditions.
Trinnov powerful platforms allow Loudness and True Peak measurements to be activated independently for multiple user-defined sources.
Multichannel platform users can customize their workspace and save custom layouts in presets.

COMPREHENSIVE SET OF INSTRUMENTS

SmartMeter offers a comprehensive set of instruments to make sure your mixes can be broadcasted or delivered.
Four complementary Loudness Meters (EBU R128)
Peak Meter (sample-peak)
Quasi-Peak Meter (DIN 45406)
True-Peak Meter (EBU R128)
Real-Time spectrum analyzer (1/3rd octave)
Surround Sound Analyser: Level, Phase and Interchannel Phase correlation meters.
2-channel audio vectorscope including a stereo correlator and true peak meter
Monitoring Controller with source switching, DRC, and Downmix activation.

FUNCTIONALITY

Multiple Sources
The SmartMeter is designed to integrate seamlessly into your broadcast or post-production workflows.
The SmartMeter V3 supports MULTIPLE SOFTWARE SOURCES with INDIVIDUAL MEASUREMENT and input format.
The number of sources is limited only by the number of physical inputs and the processing power of the platform.
Industry standards from mono to 7.1 are supported as well as custom formats up to 24 channels.

TIMECODE SYNCHRONIZATION

When an LTC input is enabled and detected at 100% +/-5% playback speed, the system automatically starts the measurement. Numerous information are recorded and TIME STAMPED.
The measurement instantly stops if the operator rewinds, shuttles or jogs through the project and is resumed as soon as the play speed is back at 100%, providing accurate values all along different mixing stages, including Integrated Loudness and Loudness range which both rely on a constant knowledge of the entire program content.

SESSION RECALL

As a direct consequence of the Time Code synchronization, measurements remain consistent with all the other elements of the project.
As for video, audio or console automation, Trinnov implemented SESSIONS RECALL.
The whole project, including every sources measurement, alerts, and user settings can be saved and recalled to resume mixing in another room and at a different time.

DYNAMIC ALERT ENGINE

Session-specific thresholds can be chosen from the standard ALERT PROFILES (EBU R128, ATSC A/85, CST RT17, ARIB TR B32…) or user-defined to comply with delivery requirements of most organizations and companies.
Overrunning these thresholds trigger visual alerts and REAL-TIME EVENT LOGGING. The event log is constantly updated and can be filtered to display the alerts of a specific source and type of alert. Users are notified about LTC drops and therefore potential measurement errors.
Complete PDF REPORTS are saved with the session, providing project information, the compliance status of each source with regards to the selected alert profile and the list of alerts with their respective time codes.
Alerts can also be transmitted to a server as SNMP TRAPS.

CUSTOM LAYOUT

The Multiview-Mode is a customizable display for MC Processors that gives the user great flexibility on how to arrange and access any of the 15 instruments and menus on a 4×4 split interface optimized for touch screens.
Except for the DRC module, every instrument can have four independent instances simultaneously displayed in the four views.
Layouts can be customized and saved in presets

INSTRUMENTS

The SmartMeter V3 includes a comprehensive set of Loudness metering instruments implementing momentary, short-term, integrated loudness measurements, as well as loudness range and True Peak, as specified in the EBU R128 recommendation. It also complies with USA CALM act Requirements
Most instruments of the SmartMeter V3 share the following features

  • Source selection panel if the instrument is activated for multiple sources
  • Real-Time alerts according to the session-thresholds – as values are rewritten, visual alerts and event logs are updated
  • Loudness instruments display the current gating status of each source
  • Time Code, locking status and LTC drop alarms
  • Multiple independent instances in MultiView display
ULTIMATE LOUDNESS METERING SUITE

The Loudness Timeline, the Loudness Meter, and the Loudness Overview offer different views to monitor loudness levels.
Please note: all measurements can be performed as specified in ITU-R BS.1770-2, which the EBU R-128 extends even further or without Gating for specific “Anchor” section measurement.
The Loudness Timeline specifically displays the Short Them Loudness in the form of a history graph and allows zooming and different scrolling modes. Gated areas are highlighted when the measurement is paused.
The Loudness Meters combines bar graphs and numeric values.
The Loudness Overview provides measurements as numeric values for up to 5 independent sources in a single window. 20 sources can be monitored using the MultiView display.
The Loudness Statistics instrument displays and updates the distribution of momentary Loudness values of the program in real-time in the form of a histogram. The Target and current Integrated Loudness levels are overlayed as vertical lines, and color codes are the same as in the Timeline. The M Stats includes a range selection tool that highlights corresponding values in the Timeline, providing the most helpful instrument to adjust a mix and comply with recommendations without loudness leveling.

TRUE PEAK, PPM, QPPM

The SmartMeter offers a variety of peak-metering Tools:
True Peak Meter (EBU R128)
A Peak Program Meter (sample-peak)
A Quasi-Peak Program Meter (DIN 45406)

VECTORSCOPE, SURROUND ANALYSER

The SmartMeter provides a Surround Analyser to control multichannel audio signals amplitude and correlations.
The Vectorscope is 2-channel audio phasemeter including a stereo correlator and a true peak meter.

MONITORING CONTROL

SmartMeter v3 includes a Monitoring Controller with volume control, solo and mute of any output.
It also includes source switching, each associated with DRC and Downmix profiles.
GPIO commands can also be sent to third-party equipment.

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