Dirac Live® is a room acoustic solution that enables users to measure and optimize their sound systems to their preferences and reduces sound colorations introduced by the room. Dirac Live® has two features: room correction, included with every software package, and bass control, an add-on feature available as separate purchase. Improve your studio acoustics without breaking the bank. Dirac Live for Studio belongs to the Dirac Live family of technology and software. It features Dirac’s patented algorithm to digitally reduce the room impacts and enhance the accuracy of sound reproduction, from timing and frequency response to imaging. Purpose-built rooms with ideal acoustic treatments may not need room correction software, but all the rest do. Producer/studio owner Rob Tavaglione takes a look at the latest entrant in the field, Dirac Live, and finds it may be the best. Echo Speech Source Type 4720 – Sound source delivering a calibrated signal that is used by the DIRAC room acoustics software to measure a range of speech intelligibility parameters. Enquire About This Product.

Optimise your room's acoustics

When it comes to setting up a sound system, the biggest impact to the audio quality comes from the room itself. No matter how good or expensive the equipment, a challenging environment can result in poor sound quality.

There are numerous methods of mitigating a room's negative effects, from optimal speaker and subwoofer placement, to using absorption panels, diffusers and bass traps. There's no doubt that carefully installing speakers and subs in the best locations will help eliminate certain room issues, but using panels and diffusers can get expensive and intrusive.

These days the most effective way of addressing the impact of a room is to use some form of room correction software, and one of the most popular is Dirac Live. This particular system has been licensed by numerous mid-range and high-end manufacturers and integrated into their products. These companies include Arcam, Emotiva, Datasat, Focal, Monoprice, miniDSP, NAD, Rotel, Storm Audio, Wisdom Audio, JBL Synthesis, and Thetal Digital.

What versions of Dirac Live are there?

There are two variants of Dirac Live aimed at two different markets: one for the home market; and another for the professional audio market. Both versions share the same Dirac Live room correction calibration tool, with the key difference being that the professional audio/studio version includes an audio plugin, and therefore does not require compatible hardware to store and apply the filters like the home version does.

The version of Dirac Live aimed at home audio market currently has two key features: Room Correction and Bass Control. For both of these features there are tiered offers, leading to variations in Dirac Live's capabilities. For example, when using the room correction feature, end users can choose to upgrade from limited frequency range correction (Limited) to full frequency range correction (Full). For the bass control feature, end users can choose to upgrade from single subwoofer bass management (Limited) to multiple subwoofer bass management (Full).

What does Dirac Live seek to achieve?

Dirac Live aims to enable users to customise their acoustic experience, ensuring that their sound system performs at its best with minimum impact from the listening environment.

Dirac believes that the reproduction of transients is critical for a truly natural and transparent sound. For example, the stereo image is critically dependent on the time-domain differences and similarities between the sound at the left ear and the right ear. As such, Dirac Live's impulse response correction improves the depth, the positioning and distinction of individual voices and instruments.

How Dirac Live basically works:

The end user installs software on their PC or smartphone (available on Windows, MacOS, Android and iOS) to connect to a Dirac Live-capable system over a network. They then follow the on-screen instructions, allowing them to perform the set up procedures, which include selecting the device, selecting the recording device (including any calibration file for the microphone), setting the calibration volume, selecting the speaker arrangement, taking the actual measurements, designing the filters, and exporting them to the device to create the desired sound.

Measure: The user takes measurements of the room and the audio systems with a microphone (either provided with the device or bought from a third-party) by following the instructions in the Dirac Live calibration software. Dirac Live uses several measurement positions (typically 9) in the listening room, making it possible to accurately locate and correct acoustic problems. These measurements are presented in a clear and easy to interpret graphical interface.

In terms of the impulse response, this means optimisation of the direct wave and very early reflections. Generally, at higher frequencies there is less consistent time-domain behavior. Dirac Live room correction automatically finds consistent problems in a frequency-dependent way and corrects for them. After addressing consistent time-domain behavior, there may still be some late reverberation in the room that causes subtle colouration. Dirac Live corrects these in the frequency domain to achieve just the right resolution.

Filter design: The user then adjusts and sets the desired frequency response using the breakpoints and target curve feature. However, the user doesn't have to do anything with impulse response correction, because that is carried out automatically. The software can show the measurements before and after the application of the filter, either for each individual speaker and subwoofer or all together as a system.

Why is it that the same frequency response can sound so different from room to room? The answer lies to a large extent in the time-domain aspects of the measured frequency response, as well as critical differences in the spatial (angular) domain. By addressing time-domain problems as such, and not as minimum-phase spectral problems, frequency response correction becomes more effective. After conducting time-domain correction, Dirac Live treats the spectral colourations in a room. By paying particular attention to the variations across different measurement positions – not just the average – Dirac Live makes sure not to overcompensate.

A loudspeaker’s impulse response affects clarity, detail and all spatial aspects of the sound. Dirac Live corrects the impulse response throughout the listening area, not just at a single point. By focusing on consistent problems across the measurement positions and correcting only these anomalies, Dirac Live can achieve a faster decay time. Typically, the power ratio between the direct wave and the tail (later-arriving echoes) is improved by 6 dB or more, representing a vastly improved overall sound.

A flat frequency response does not necessarily sound neutral due to the ratio between direct and reverberant sound and the frequency-variable propagation loss. Dirac Live can’t remove late reflections, because they change too much between different positions. Therefore, a slight roll-off is often appropriate when listening at a distance. The Dirac Live algorithm suggests a target response appropriate for your listening environment and speakers, which you can of course adjust to your taste using the simple graphical interface.

Export filter: The user then exports the filter they have created into the compatible hardware or audio plugin.

Apply the filter: The filter is then applied to the audio signal via the hardware (Dirac Live for Home) or audio plugin (Dirac Live for Studio), and adjusts the frequency and impulse responses to eliminate the more egregious aspects of the room.

The beauty of Dirac Live is that you don’t have to be a signal processing or computer expert to enjoy its benefits. All of the measurements, processing and correction is automatically managed by the audio calibration software, making it easy for all kinds of users. All you to do is position the microphone and press “Play” – the software wizard will take care of the rest. Advanced users are free to edit the target response as desired using a simple graphical interface. Otherwise, just press “Optimise” for immediate results.

What’s Bass Control and how does it basically works?

Dirac calls this feature Bass Control, instead of bass management, because it is fundamentally different from simple bass management. The Dirac Live bass control feature works by co-optimising all the speakers and subwoofers in the same system. The speakers, subwoofers and the room are all regarded as a complete system.

To ensure that the system’s bass reproduction is accurate, and the performance is the same at all locations in the room, it aligns the phase between the speakers in correlation. In technical terms, Dirac Live Bass Control reduces the spatial variability of the frequency response in the lower frequencies when more than one subwoofer is available. It also reduces any out-of-phase behaviour between paired speakers in a frequency band around the crossover point. The Bass Control also reduces out of phase behaviour between the full-range speakers and subwoofer(s) in a frequency band around the crossover point.

What makes Dirac Live different from the competition?

Dirac Live room correction uses mixed phase technology with advanced impulse response and phase correction. This is one of the main features in Dirac Live’s room correction software and enables Dirac Live to deliver impressive room correction.

There's a look-ahead buffer that enables impulse response correction. In addition, minimum-phase and linear-phase room correction filters can’t physically optimise the acoustic impulse response in a room. At best, they can minimise problems caused by the application of a filter. Room-acoustic responses are non-minimum-phase, which is why Dirac Live uses mixed-phase correction.

Most other room correction systems use minimum-phase, which means they try not to add any time-domain distortion, whereas Dirac Live actually reduces time-domain distortion. Mixed phase technology enables Dirac Live to address some of the key room acoustic problems other solutions struggle to address efficiently. The results are audible in terms of imaging, clarity and bass tightness.

For in depth knowledge about the mixed phase approach, you can read an article written by Dr. Mathias Johansson, CEO and co founder of Dirac.

Traditional EQ systems use IIR (infinite impulse response) filters for the lowest processor usage, and many room correction filters use FIR (finite impulse response) for ease of implementation and sound optimisation. IIR and FIR filters both have their pros and cons. Dirac Live uses a proprietary filter structure that achieves maximum performance at much less processor usage than regular FIR filters. This filter structure also has improved numerical properties, so it doesn’t produce any digital noise, and thus achieves enhanced sound optimisation.

It is also worth to mentioning that although Dirac Live is sophisticated and delivers a high level of performance, it is very easy to run with its intuitive user interface, visual filter design feature and step-by-step guide. This, together with its leading performance and capability, makes Dirac Live attractive to many processor and receiver manufacturers. It also makes it a great choice for consumers of any kind, whether they're AV enthusiasts or average users.

Dirac is a sophisticated tool to calculate a very wide range of acoustical parameters from impulse response measurements of an enclosure. Through many years of development and customer feedback, Dirac has grown to support many different applications. Due to its user friendly interface it is suitable for field measurements, but it also contains many advanced features that make it useful in a laboratory or research setting.

To capture an impulse response, a stimulus signal is generated by Dirac or an external source. This stimulus signal is played in the enclosure through a sound source, and recorded through a microphone. Dirac then deconvolves the recorded response with the original stimulus to arrive at the impulse response.

Dirac will work with most Windows compatible sound devices. External USB based sound devices, such as the Triton, are recommended. The FAQ lists of some of the other sound devices that can be used with Dirac. Dirac is very easy to use, yet provides all the features you may need for your measurements. The automated sound device loopback test ensures that you will get the optimum results from your system.

Dirac comes with an extensive context sensitive help system and a comprehensive manual that contain many practical hints and tips that will help you to use the tool successfully. Qualified support is available through the worldwide network of Brüel & Kjær backed by Acoustics Engineering specialists.

Features

• Calculates a large number of acoustical parameters.
• Conforms to ISO 3382 and IEC 60268-16 standards and supports ISO 3741 and EN 1793-5 standards.
• Open loop (asynchronous) measurements using a CD or MP3 player to play back an MLS or sweep stimulus.
• Intermittent stimulus for background noise capture.
• Single measurement per position for ISO 3382-3 open plan office measurements.
• The included 1/1 and 1/3 octave band filters are fully IEC 61260 compliant.
• Scale model measurements with air absorption compensation, and auralisation.
• Realtime pre-averaging for up to 1000 recordings. Post averaging for direct impulse measurements.
• Automated measurements reduce the total time required for a measurement session.
• Reverse filtering technique enables accurate short reverberation time measurement, such as in studios and building structures (loss factor).

• Multiple impulse response views with zoom: waveform, ETC, Decay, Frequency and Phase response.
• Waveform editing with unlimited undo.
• Uses standard WAV and ASCII files as well as MLSSA TIM files. Multi-channel import from B&K WAV files (PULSE).
• Calculates and graphically displays statistics over multiple measurements (mean, standard deviation, min-max).
• Plot parameter values versus frequency or versus source-receiver distance.
• Directivity, waterfall and spectrogram plots.
• Parameter graphs can be printed, saved as text files or copied to the clipboard for use in reports.
• Organize measurements in projects and export all parameters for all files in a single operation.
• Automatic notification and download of software updates.
• Runs on Windows Vista, 7, 8 and 10
• Support CoreAudio (WASAPI) and ASIO audio.

Video

The video above highlights some of the new features in Dirac 6.

Screenshots

Click on the image above to see a gallery of a small selection of the screens in Dirac. For a more detailed view of Dirac try the Demo version.

New in Dirac 6.0

• Support for the Echo speech source. The Echo is a small loudspeaker source with a calibrated spectrum and sound level, and a directivity similar to a human speaker. It contains a number of different stimuli that are automatically recognized by Dirac. Using the Echo, accurate speech intelligiblity measurements can be performed much faster.
• Intermittent measurements for Echo speech source and ISO 3382-3. The new intermittend stimulus consists of a conventional MLS or sweep stimulus, followed by a period of silence. The MLS or sweep signal is used to measure the impulse response, whereas the silence period is used to capture the background noise level. Each part of the measurement is stored in a separate channel, and Dirac uses both to calculate accurate speech intelligiblity parameters.
• New parameters for ISO 3382-3: L_P,A,S, L_P,A,S,4, D_2,S, STI, r_D and r_P. Open plan office measurements have never been this easy. Using the new intermittent stimulus, a single measurement suffices to capture all quantities required by ISO 3382-3. The calculated parameters are displayed in clear graphs that optionally include regression lines.
• Graph parameter values versus the source receiver distance. Parameter values are no longer restricted to value versus frequency graphs. Plotting values as a function of the source receiver distance can provide more insight into the acoustics of an enclosure.
• Continuous MTF graphs. Contrary to the modulated noise method, the impulse response method for measuring the modulation transfer function is not limited to discrete modulation frequencies. You can now see MTF peaks and valleys in between the standard frequencies, allowing you to uncover problems that remained hidden up till now.
• Load and save measurement setups in the measurement window. The enormous variation in measurement configurations that are possible in Dirac may seem somewhat daunting at first. With the new measurement presets, you can simply select a predefined setup without having to worry whether everything is set up correctly.
• Multi-channel import (also B&K PULSE files). If you have a PULSE measurement system, then analysing your measurements in Dirac is now much easier. Simply import a multi-channel measurement and Dirac will automatically split the file in single or dual channel results that can be processed and analysed in the parameter graph window
• New level parameters: L_min, L_max, L_peak and L_N with A/C/Z frequency- and F/S time-weighting. Although Dirac is primarily used for impulse response measurements, you can also record signals and analyse them. In a previous version you may already have used the L_eq, L_Aeq or L_Ceq. We have now added a number of often used statistical sound level parameters.
• Display Leq and derived values for measurements with pink stimulus. If you consider that the frequency response of an enclosure is independent of the stimulus that is used to measure the response, it seems strange that you could calculate the measured L_eq values from the impulse response. Yet this is what Dirac will do for you for both white and pink stimuli.
• Improved support for ASIO sound devices. Although the previous version contained some rudimentary support for ASIO sound devices, the new version places ASIO on the same level as the native Windows CoreAudio API. This makes it possible to use a whole range of audio devices for which a CoreAudio driver is not available.
• Automatically add new measurements to a parameter graph. A new option in the measurement window will add every new measurement to a group in the parameter graph window automatically. The graph is then recalculated immediately. When used together with the automeasure option, you can see near realtime parameter updates, and using the statistics option (deviation) you may see the parameter converge.
• Export displayed data (ETC, spectrum etc.). A feature requested by quite a few users is to be able to save the data as calculated for a specific impulse response view. This includes the selected band filtering. The data is exported to a .wav file, but can also be written to a text file for use in f.i. Excel.
• Many more additions and changes. The complete list with new and improved features is rather long, and is contained in the helpfile.

Parameters

Energy ratios:

• Bass Ratio (based on level), BR(L)
• Treble Ratio (based on level), TR(L)
• Centre Time, Ts
• Clarities, C₃₀, C₅₀, C₈₀
• Clarity (user defined integration interval), C_x
• Definition (Deutlichkeit), D₅₀
• Definition (Deutlichkeit, user defined integration interval), D_x
• Echo Criterion (for music or speech), EC music, EC speech
• Echo Criterion (user defined), EC_n,τ
• Hallmass, H
• User-defined Energy Ratio, ER
• Early Lateral Energy Fraction, LF
• Early Lateral Energy Fraction, LFC
• Inter-Aural Cross-correlation Coefficient, IACC₈₀
• Inter-Aural Cross-correlation Coefficient (user defined interval), IACC_x
• Early lateral sound level, GEL
• Late lateral sound level, LG (GLL)
• Early Support, STearly
• Late Support, STlate
• Total Support, STtotal
• User-defined Support, ST_X,Y

Speech intelligibility:

• Modulation Transfer Index, MTI
• Speech Transmission Index (male, female), STI
• STI for PA systems, STIPA
• Room Acoustics STI, RASTI
• STI for TELecommunication Systems, STITEL
• Percentage Loss of Consonants, % ALC
• A-weighted SPL of speech, L_p,A,S
• A-weighted SPL of speech at 4m, L_p,A,S,4
• Spatial decay rate of A-weighted SPL of speech, D_2,S
• Distraction distance, r_D
• Privacy distance, r_P
• Speech level
• Noise level
• A-weighted background noise level, L_p,A,B

Reverberation:

• Early Decay Time, EDT
• Reverberation Times, T₁₀, T₂₀, T₃₀
• Reverberation Time (user defined decay range), Tx
• Reverberation Time (from best decay sections), RT
• Bass Ratio (based on reverberation time), BR(RT)
• Treble Ratio (based on reverberation time), TR(RT)

Levels:

• Impulse response to Noise Ratio, INR
• Peak to Noise Ratio, PNR
• Signal to Noise Ratio, SNR
• Strength (Level relative to 10 m free-field), G
• Strength (user defined integration interval), G_X,Y
• Early Strength, G₈₀
• Late Strength, GL
• Relative Strength, Grel
• Magnitude Spectrum
• Magnitude Spectrum Pink (-3 dB/octave offset)
• Equivalent Sound Level, L_eq
• Equivalent A- and C-weighted Sound Level, L_Aeq, L_Ceq
• Sound Intensity, I
• Level difference between channels, D
• Minimum sound level (A/C/Z and F/S weighting), L_min
• Maximum sound level (A/C/Z and F/S weighting), L_max
• Peak sound level (A/C/Z weighting), L_peak
• Percentile sound level (A/C/Z and F/S weighting), L_Nx

Reflection & absorption:

• Reflection Index, RI
• Sound Insulation Index, SI
• Sound Power Reflection Factor, Q_w

Other:

• Source-receiver distance

Downloads

Frequently Asked Questions

In general: Yes. Dirac should work with any sound device that supports CoreAudio (WASAPI) or ASIO. Dirac requires that your sound device is full duplex, meaning it will playback and record at the same time. Almost all current soundcards, notebook sound systems and external sound devices are full duplex. The unregistered (demo) version of Dirac, which is available for download, can be used to perform a soundcard test. A successful sound device loopback test implies that Dirac will work with your sound device.

The Triton sound device was designed specifically for use with Dirac, so we can highly recommend this device. Another fine device is the Edirol UA-25 which provides a phantom power supply. Lower cost devices that work well are for instance the Behringer UCA202 and the ESI MAYA44 USB+. For scale model measurements the E-MU 0202 USB and the Terratec DMX 6Fire USB can be used. Please note that there are many more devices that work perfectly with Dirac. Use the demo version to test Dirac with your sound device.

No, you do not need to perform a loopback test on your sound device in order to use Dirac. However, working with Dirac becomes much easier once your sound device has passed the test. During the loopback test procedure, Dirac figures out how to control your sound device hardware. It also finds the optimum settings for the volume controls. Finally, the test procedure will correct the loopback frequency response of your sound device if required. You could of course set the volume controls yourself before every recording. However, it is almost impossible to reproduce the optimum settings for each measurement session.

There are a number of possible causes for a sound device calibration to fail:

1. The problem could be as simple as an incorrect connection between the selected input and output. Also make sure to use a (short) shielded cable, to minimize interference.
2. Not all sound devices support all sample frequencies. Many current sound devices have the hardware laid out for a single sample frequency (often 48000 Hz for AC'97 codecs), and use sample rate conversion algorithms for other sample frequencies. Sometimes these algorithms produce results that are unusable for Dirac.
3. Some sound device manufacturers make near perfect hardware, but manage to write drivers that make the sound device almost useless. Switching drivers may help in this case.
4. The load on your processor may be too high. The calibration procedure uses a lot of processing power. Running compute intensive programs in the background may prevent Dirac from completing the calibration procedure. In particular, we have seen problems caused by virus scanners running in the background.

The minimum system requirements are a 1 GHz CPU, Microsoft® Windows Vista ,7 or 8, Microsoft .NET framework 4.5, 500 MB of available disk space, a WSVGA resolution 1024 x 600 (WXGA 1280 x 768 or higher recommended), a CD-ROM drive and a full duplex sound device with support for CoreAudio (WASAPI) or ASIO. A typical current laptop would therefore be more than adequate.

To perform single channel parameter measurements according to ISO 3382 or IEC 60268-16, you can use a type 1 sound level meter, meeting the IEC 651 requirements, and equipped with a line output. Normally, single channel parameters can also be approximated using lower cost omnidirectional electret microphones or sound level meters. The measurement of LF requires either an additional bidirectional microphone, a.k.a. a “pure pressure gradient” or “figure-of-eight” type, or a switchable omni-bi-directional type. The latter is particularly useful if you have only one measurement channel available, e.g. when using a random noise source, which will occupy channel 2. The measurement of LFC or Sound Intensity requires a matched omnidirectional microphone pair at a fixed distance, such as a sound intensity probe. The measurement of IACC requires a head simulator.

Bidirectional microphones (for LF or LFC measurements) are supplied by Schoeps, AKG, Neumann and Sennheiser. A switchable omni-bi-directional microphone is supplied by Neumann. An example of a sound intensity microphone probe (for LF, LFC and Sound Intensity measurements) is Type 3519 from B&K. Examples of usable head simulators (for IACC measurements) are the HMS III Artificial Head from Head Acoustics, the KU 100 from Neumann and Type 4100 from B&K.

The onboard A/D converters in a digital head simulator will provide a THD+N that is probably superior over any analog head simulator with external sound device, because the analog microphone signal path is highly optimised. This very low noise configuration is useful for music recording purposes. However, for room acoustic measurements, where minimum SNR values are normally much higher and controllable, it is not really necessary to use a digital head simulator.

You can use wireless transmission of input or output signals under certain conditions. The transmission channel may be equipped with a compander, i.e. analog level compression at the transmitter and complementary level expansion at the receiver. This can for instance be found in some wireless microphones and may cause allowable noise.

Instead of a parallel port dongle, you can also use a USB dongle. Please ask your local Brüel & Kjær representative.

Dirac Room Acoustics Software Type 7841

The loopback dynamic range or Impulse response to Noise Ratio INR, averaged over the octave bands from 125 Hz through 4 kHz, is typically about 60 dB. With a good sound device, the loopback INR is about 50 dB at 31.5 Hz, increasing up to 96 dB at 4 kHz and up, and 80 dB on average. Sound studios typically show 60 dB, and concert halls 50 dB.

MLS and sweep will normally result in the same impulse response, but the methods differ in the effect of system irregularities, such as click noise, system variations during a measurement, distortion in the measurement chain, etc. With MLS these effects result in parasitic energy, time-distributed as noise over the impulse response. With sweeps, these effects result in parasitic energy, time-lumped as e.g. small sweeps in the impulse response. Unlike noise, the energy packages can often be removed very easily or have hardly any impact on the derived acoustical parameters. On the other hand, recognising impulse response details may be easier if the impulse response is affected by random noise rather than by a nonrandom but unknown parasitic signal.

One cause is mentioned above, and has to do with the way parasitic energy is converted into noise using MLS rather than sweeps. Another cause is that a filtered MLS signal has a higher peak to rms ratio than a filtered sweep signal. This allows the power amplifier to produce a higher rms level from a sweep than from an MLS signal.

The imported file is processed as if it were measured at the actual settings. Therefore, the imported file MLS/Sweep/Capture length and Pre-Average value should match the corresponding values in the Measurement window. Also, the imported file sample rate should match the one in the sound device setup window.

Dirac Room Acoustics Software

The following data can be exported:

• The original impulse response (or any other opened .wav file), by saving it in Dirac as a .txt file and opening it as such in Excel.
• Any parameter table in the Parameter window, including statistical data over several impulse responses, by saving the table in Dirac, and opening it as .txt file in Excel, or through copy & paste using the clipboard.
• All (or selected) parameter values for all measurements in a project, using the 'Save Project Data' menu item of the Parameter window.
• Any single impulse response parameter table in the Parameter menu, by saving the table in Dirac and opening it as .txt file in Excel.
• Any of the impulse response views: Enery-Time Curve, the Decay Curve, the Magnitude Spectrum or the Phase Spectrum.

With an External Impulse measurement, the graphically displayed signal and noise simply reflect the signal produced and the system noise present during the measurement. Therefore, the displayed energy ratio equals the real ratio. System noise includes acoustical and electrical noise.

With a non-Pink+Blue filtered MLS or lin-Sweep measurement, under certain conditions the graphically displayed noise relates to the real noise as follows. The ratio of the total energy of the file and the total noise energy (with the same file length) equals the ratio of the received signal energy produced (plus system noise energy) and the system noise energy present during the measurement. The mentioned conditions are: Pre-Average = 1, time invariant system and no significant signal distortion. The displayed ratio is basically proportional to the Pre-Average value, but practical acoustical systems may vary slowly in time.

In any other case, there is no one-to-one relation between the displayed and the real noise.

You can move the total impulse response rotation-wise using the Rotate command in the Edit menu. This will in most cases not affect the calculated parameters. For some parameters such as the Strength (G) and related parameters, the source-receiver distance is essential, and rotate should not be used.

The red line indicates the start of the impulse response, as used in parameter calculations and determined in conformance with ISO 3382. You can influence this starting point, hence the calculated parameters, by setting the 'Minimum Source-Receiver distance' in the Measurement window. Normally this option is used to skip response peaks caused by crosstalk between output and input lines.

It is possible to get any energy ratio, by setting the appropriate ER integration intervals in the Parameter Setup window.

The most important difference probably stems from the fact that Dirac is developed by users wanting a user-friendly tool. The best way to experience the differences, is by trying out demo versions.

Dirac may not yet contain your favorite feature. Given enough interest, and provided the feature fits within the 'philosophy' of Dirac, we will implement it. Just send us a description of your favorite feature, and we will consider it for inclusion in one of the next versions of Dirac.

Dirac Room Acoustics Software

No, currently Dirac is only available in an English-language version.

Please contact your local Brüel & Kjær representative for purchasing information.

Bruel & Kjaer Dirac Room Acoustic Software

Examples

Ordering

Worldwide sales and marketing are handled exclusively by Brüel & Kjær. To order Dirac you must contact your local Brüel & Kjær representative.
If you have technical questions regarding Dirac, you may contact us directly.