In order to use Dirac Live® Processor pluginPlugin, you will need to download and install two different components: the Dirac Live® application 3 (3.0.x) and the Dirac Live® Processor plugin Plugin (1.34.x). Dirac Live® 3 is used to measure and generate filters, while the Processor audio plugin stores the corresponding filters and processes audio data.
Dirac Processor Plugin compatibility
The Dirac Processor Plugin package requires a plugin host compatible with VST2, VST3, AAX, or AU plugins. Compatible hosts include but are not limited to:
- Logic Pro X
- Cubase 10
- Studio One 4
- JRiver Media Center
- Ableton Live 10
- Pro Tools 11 or later
Download and Install Dirac Live® 3 and Dirac Processor
You can download the Dirac Live® Live and Dirac Processor pluginPlugin here. Make sure that your computer is connected to the internet for licensing purposes.
Open the file files and follow the installation procedureprocedures.
The Processor plugin will be placed in a folder according to the path below, unless you have made changes during the installation process:
- Program Files/Common Files/VST2
- Program Files/Common Files/VST3
- Program Files/Common Files/Avid/Audio/Plug-Ins
- /Library/Application Support/Avid/Audio/Plug-Ins
Start your DAW and locate the audio plugin in the plugin menu. You might need to go through additional steps to activate the plugin, depending on your DAW.
Press the “play” button in your DAW to ensure that the audio stream is active while connected with the calibration tool. It is important that the audio stream is active during the whole calibration process. Here is a 30 minutes mp3 that you can use.
Run Dirac Live®
- Make sure that your computer is connected to the internet for licensing purposes.
- Please see to that your calibration microphone is connected to the computer running Dirac Live.
- Enter your account details in the first screen after launching Dirac Live. Login is required in order to save/restore any project or use purchased features.
Select your device
Dirac Live will start with scanning the network for compatible devices that will store the filters and process audio data (such as AVRs or the Dirac Live processor audio plugin). You can click the “Rescan devices” icon in the upper left corner if you want to refresh the scanning result.
Please select “Dirac Live processor” and proceed. Do not select any AVRs. You want the Dirac Live Processor plugin to function as the device that stores the filters.
Select your recording device
- All available recording devices will appear in this step.
- You can load your microphone calibration file (.txt) by pressing the button “No microphone calibration” and selecting “Load from file”.
- Select the calibration microphone that is connected to your computer.
- In this step we will set up the output volume of each speaker to make sure that the playback volume of all speakers is at an appropriate level. This is an important step, as too low or high volume can result in poor or failed measurements, and excessive volume could potentially damage the speakers.
- It is sufficient that the speakers are approximately within a reasonable range, that is to say, the volume does not have to be at an exact level, since the calibration tool will automatically adjust it.
- Place the microphone in the middle of the listening area.
- Make sure the master output level is at a low value.
- Select a speaker, for instance, front left, and press the “play” button for that speaker. You should now see a sign above the play/stop button that indicates the recording level. If you slowly increase the master output level, you should see the level bar for the selected speaker increases. If not, go back and check the microphone connection.
- Increase the master output until either the level bar for the selected speaker is in the green area, or the output level is loud enough at a comfortable level.
- When the above process is completed, the level bars for all speakers should indicate roughly the same value within 15dB.
Since the filter design algorithm requires that the speakers be measured with a moderate sound pressure level and a noise level as low as possible, it is crucial to do a level calibration of the system before measurement. The microphone should first be positioned in the center of the listening area. This is the "sweet spot."
- Set the Mic gain to 100%, which is the point between the red and blue areas.
- The "Master output" level must be set to a low volume in order to avoid damage to your ears or speakers. If it is not already set to a low volume, drag the magnitude indicator to the lower part of the slider.
- Press the play button beneath the speaker located furthest to the left. The speaker should now play a stimulus in the form of a pink noise or, if the speaker is a subwoofer, short sine sweeps. If you cannot hear the stimulus, slowly raise the "Master output" level until you hear it.
- Repeat this procedure for all speakers. If there is no noise playing from one or more speakers, make sure that your device is configured to the correct speaker configuration and that your speakers are connected to the device. Ensure that the device's firmware is also properly recognizing each speaker.
- After you have confirmed that all speakers play stimulus, then play the stimulus from the furthest left speaker again. Ensure the sound pressure indicator indicates sound being played.
Adjust the Master Output to a comfortable or slightly louder-than-normal listening level for Measurement and then proceed.
If you receive a Signal-to-Noise Ratio error during Measurement, you will need to increase Master Output or decrease Mic Gain at the Volume Calibration stage.
If you receive a Clipping error during Measurement, you will need to decrease Master Output at the Volume Calibration stage.
Remember that volume of the stimulus should never hurt your ears. There is a lock on the Master output slider for safety reasons. However, if you need to raise the volume into the red zone and are positive that your system can handle it, press the red lock that appears above the slider. You should now be able to drag the slider into the red area.
There are three options available, please select “Studio”.
- Make sure there is a clear line-of-sight between the microphone and speaker.
- Make sure that there is no background noise (tv, radio, AC, etc.) while doing the measurements.
- Remember to keep the microphone still, preferably using a stand or similar.
- A sweep will be played in turn through each speaker and a final sweep will be played through the first speaker once more.
- The main measurement position: This is where you head is most likely to be, and it is the most important position to be measured correctly.
- Spacing and order: In general, the measurement positions should be 40 – 60 cm apart, however the exact placement or order of the remaining positions is not crucial, and the position layout only provides a rough guide for you.
- Number of measurements: You can do fewer measurements than the full set before proceeding to Filter Design, but we recommend that you complete the full set for the best possible audio optimisation results.
- Your project will be automatically saved, however, we strongly recommend that you always manually save the project as well. To save the project, please select “Save project” in the hamburger menu (upper left corner).
Potential problems and workarounds during the measurement
- Clipping: If the level is too high during the measurement, the signal will be clipped, and the measurement will be therefore terminated. Please return to the Volume Calibration step to reduce the gain of the corresponding speaker or master volume of the system.
- Low SNR (signal-to-noise ratio): If the level is too low during the measurement, it is difficult for the application to discern the signal from background noise. Please return to the Volume Calibration step to increase the gain of the corresponding speaker or master volume of the system.
- Lost samples: the buffer size might be too low in your DAW settings, please increase your buffer size.
Edit target curves of frequency response
After completing the measurements, Dirac Live will automatically generate suggested target curve(s) for frequency response and the corresponding correction filters that are intended to enhance your sound system as much as possible. You can make fine adjustments to your preferences. The frequency response correction will be performed in real time so that you can see the correction result immediately and continuously make more adjustments.
- Add/delete control points by right clicking on the curve
- Modify the curve by dragging the control points
- Modify the frequency range that Dirac Live will compensate by dragging the curtains (left/right). The dotted lines indicate the automatically detected sound thresholds. Outside of these limits no correction will be applied, i.e. the audio signal will not be adjusted in the frequency area on either side of the curtains (shadowed area).
- Custom target curves (.txt or. targetcurve format) can be loaded, see option in the hamburger menu (top left).
- With the “Take snapshot” function, you can create a snapshot of the current state – if you make any changes to a target curve, it is possible to switch between different snapshots, without having to save/load projects.
- Zoom in/out by using pinch zoom on a touchpad or scroll wheel with a desktop mouse, panning can be achieved by hold+drag.
- Select different display options to your preference in the lower right corner.
Inspect impulse response correction result
Under “Filter design”, you can switch from “set target” to “impulse response” to view the impulse response correction result. Impulse response affects clarity, detail and all spatial aspects of the sound. Dirac Live’s unique impulse response correction can greatly improve staging, bass and clarity.
There is no need to adjust the impulse response yourself, as Dirac Live automatically optimizes it for the best possible result.
The last step is to export the filter for a listening test. Select a slot and save under the desired name (there may be an auto-generated name, which can be replaced). When export is complete, the application will return to the Filter Design view. Remember to save your project before exiting the application.
After the filter is exported, the filter will automatically appear in the audio plugin. No other actions are needed to apply the filter to your audio stream.
- You can save the project at any time by selecting “Save project” in the hamburger menu (upper left corner).
- Depending on your device, changing the configuration may clear the filter list or switch to a configuration-specific filter list. You should never change the device configuration while performing a room correction.
- You can always get context-aware help by pressing the question mark “?” in the upper left corner.