The miniDSP Flex (DL) is
a stereo
input, four-channel output
compact high-resolution audio processor.
It features stereo analog inputs, two digital inputs (SPDIF and optical),
USB audio, Bluetooth streaming with LDAC™ and aptX™ HD codecs and automatic
Bluetooth input selection, and an OLED front panel with physical volume
control.
Stereo (two-channel) Dirac Live provides
world-class room correction, while a full suite of miniDSP’s powerful audio processing tools and flexible I/O routing support applications such as subwoofer integration and active speakers.
Info
The miniDSP Flex (DL) implements stereo Dirac Live correction. That is, there
are two channels of Dirac Live correction. After stereo Dirac Live correction, the
audio signal is routed to the four output channels for processing such
as parametric EQ and crossovers.
The miniDSP Flex (with Dirac Live) executes stereo (two-channel) Dirac Live® digital room correction,
from Dirac Research. Dirac Live's mixed-phase
filtering technology will improve the imaging of your system, minimize
the effects of room modes and resonances, and improve dynamics and
clarity.
To accomplish this, the Dirac Live
software steps you through the procedure for taking measurements around
your listening area. It employs a sophisticated analysis algorithm to
make the optimal correction across the whole listening area, not just
at a single point. You have full control over the target frequency
response. Measurements are taken with the miniDSP UMIK-1 or UMIK-2
calibrated acoustic measurement microphone.
In addition to correcting magnitude
response, Dirac Live® corrects the system's impulse response, which
reflects how the system responds to a sharp transient such as a
drumbeat. Reflections, diffraction, resonances, misaligned drivers, and
so on, all combine to smear out the transient. Impulse response
correction is a critical factor for accurate sound-staging, clarity and
bass reproduction.
The miniDSP Flex (DL) has a wide range of applications in home and
professional audio. This section illustrates a few of them.
Figure 1 shows the miniDSP Flex (DL) used as a
digital preamp with subwoofer integration and stereo (two-channel) Dirac Live room correction. The Flex provides accurate
and flexible integration between the subwoofers and the main speakers as
well as comprehensive EQ and tuning options. This is a very
cost-effective way to build a complete audio system that includes all
the benefits of modern DSP technology.
Figure 1. miniDSP Flex (DL) as preamp with subwoofer integration and stereo (two-channel) Dirac Live room correction
Figure 2 illustrates a similar use of the
miniDSP Flex (DL) as a digital preamp, but the output channels have instead been
configured to act as crossover for a pair of active 2-way speakers.
Amplification can be via a multichannel power amplifier as shown, or a
pair of stereo amplifiers.
Figure 2. miniDSP Flex (DL) as preamp and active crossover and stereo (two-channel) Dirac Live room correction
macOS 10.14 Mojave or later, latest version with all updates
At least a dual core i3, i5, or i7 processor, or an Apple ARM processor (M1/M2/Pro/Max)
At least 4 GB RAM
Two free USB 2.0 ports
Internet connection
Info
Intel and Apple ARM processors are supported with separate binaries for miniDSP Device Console.
Info
The macOS version requirement ensures compatibility with the Dirac Live application. The miniDSP Device
Console can run on OS X 10.11 (El Capitan), macOS 10.12 (Sierra) and later versions of macOS.
If you purchased your Flex directly from miniDSP, your software will be
available from the User
Downloads section of the miniDSP
website when your order ships. To access the download, you will need to
be logged into the website with the account you created when purchasing.
If you purchased your Flex from a miniDSP dealer, you will receive a
coupon together with the product. Redeem this coupon at this page:
The User Downloads link is visible from
the drop-down menu at the top right of the website:
Navigate to the miniDSP Device Console section and select the download according to your platform:
Windows, macOS with Intel CPU, or macOS with Apple ARM CPU.
Double-click on the downloaded miniDSP Device Console-x.y.z.dmg image file to open it.
(The version number x.y.z embedded in the file name will change over time.)
Drag the miniDSP Device Console application icon to your Applications folder.
To run the miniDSP Device Console, locate it in the Applications folder
and double-click on it. To make it easier to run in future,
right-click on its dock icon and select Options -> Keep in Dock.
Info
The macOS version of miniDSP Device Console is installed for all users on the computer.
Double-click on the downloaded
MiniDSP Device Console Setup x.y.z.exe installer to run it. (The version number x.y.z embedded in
the file name will change over time.)
Install the miniDSP UAC2 USB Audio driver:
Start miniDSP Device Console.
Navigate to the Settings screen and select Install USB Audio Driver:
Device Console will confirm that you want to install the driver.
Click on the PROCEED button.
Note: we recommend that you accept the default installation location.
The first time you run the
miniDSP Device Console and the Dirac Live application, you may see a
Windows Firewall warning such as the one below. Ensure that "Private
networks..." is checked and "Public networks..." is not checked. Then
click on "Allow access." This warning dialog may appear more than once.
Info
The most recent miniDSP processors require the UAC2 driver to be at least version 5.40. If you have
difficulty with the UAC2 driver or control panel, use Check USB Audio Driver to
make sure that you have the latest version.
Info
The Windows installation of miniDSP Device Console is for a single user. If it needs to be run in more than one account
on your PC, it will need to be installed for each account.
Download the Dirac Live application for your platform from
Dirac Research. On macOS, double-click on the diraclive-latest.zip
download if your browser did not unzip it.
Double-click on the downloaded installer to run it. It will be named
diraclive-latest.exe on Windows and DiracLive v3.x.y Setup Darwin on macOS (the version number x.y
will change as updates are issued). We strongly recommend that you accept the default
installation settings.
Do not run the application from the installer, as it must be started with the
Start Calibration button in Device Console.
The miniDSP Device Console ("DC") is the new configuration and control
software for miniDSP audio processors. It replaces the earlier "plugin" control
software. In contrast to the earlier software, Device Console provides:
A single software program to control all miniDSP processors
Over-the-air (OTA) self-updates
Detection and installation of firmware upgrades for your processor
Purchase and activation of Dirac Live upgrades (where applicable)
If you have not used Device Console with your Flex before,
you will need to perform a one-time migration procedure. See Migrating to DC.
The Discover Devices screen displays devices connected to your computer, or which have a network-accessible
control interface.
Info
For network control, the miniDSP Flex (DL) requires that a miniDSP Wi-DG be connected to its
USB port. Note that network control is for miniDSP Device Console
only – Dirac Live calibration can not be performed
over the network.
Tip
If your device is connected to a Wi-DG but is not visible here, confirm that you have
updated the firmware on your Wi-DG to the latest version.
To connect
to a device for editing and control, click on its name. DC will update to display the
device-specific sidebar on the left and the signal processing control interface in the main area on the right.
The signal processing area is divided into sections for the input channels and output channels.
There is a control strip for each channel. The output channel strips also contain the
signal routing matrix.
If this is the first time you have used Device Console with your
miniDSP Flex (DL), Device Console will check for an existing preset
that was created with the older DDRC-24 "plugin" user interface.
If found, Device Console will automatically import it. This is done on a preset-by-preset basis.
When you are done, click DISCONNECT.
Tip
Once you have switched to miniDSP Device Console, don't use the older DDRC-24
"plugin" user interface to connect
to the miniDSP Flex (DL) any more.
The Template Devices screen displays the complete list of devices supported by Device Console.
This can be used to explore features of a processor that you do not have, or to edit your processor
settings when you are not able to connect to it.
To edit a template device, click on it.
A template device "mirrors" the real device, so that you can edit your settings
while not connected. That is, while you are editing the real device, the
current preset is copied to the corresponding template device whenever you change presets
and when you disconnect. You can then edit those
settings in the template device.
Tip
While editing a template device, the level meters will display a "dummy" signal.
To turn off the level meters while editing a template device, click the Mute button in the left sidebar.
When you are done editing in the template device, click FINISH.
When you connect to your real device again, Device Console detects that you have edited
the template and asks you if you want to use the settings from the edited template.
Or, you can discard the changes you made in the template and revert to the
settings that your device had the last time you were connected to it. This is done on a
preset-by-preset basis:
Info
You will see a similar dialog if you edit the settings of your device using a
different computer or the older "plugin" user interface (the latter is not recommended
once you have migrated to the miniDSP Device Console).
Info
If you have more than one
device of the same type, only one of them can be edited using the
template device.
If you have more than one device of the same type or
need to edit your devices from more than one computer, you will need to
manage your presets by exporting them to and importing them from files.
See Importing and Exporting.
These settings are for Device Console itself (and not any connected
processor).
The Modes selection sets the DC to display with a light theme, a dark theme, or with the same theme
as your operating system (Windows or macOS). If you choose the System option, you may need to quit
Device Console and restart it for the change to take effect.
The Install USB Audio Driver feature installs the latest version of the miniDSP UAC2 audio driver.
It appears on Windows only.
See Installation for more information.
The Create Program Snapshot feature is used to help miniDSP support diagnose issues you may
be having with your unit. See
Obtaining Support.
The miniDSP Flex (DL) implements stereo Dirac Live correction. That is, there
are two channels of Dirac Live correction. After stereo Dirac Live correction, the
audio signal is routed to the four output channels for processing such
as parametric EQ and crossovers.
To start Dirac Live calibration, click on the START CALIBRATION button. The Device Console will disconnect
and the Dirac Live application will start.
Info
Be sure to start the Dirac Live
application from within the Device Console using the START CALIBRATION button.
If you open the Dirac Live application by itself, it
will not be able to detect the Flex.
Info
Dirac Live calibration can not be performed
over the network, so the START CALIBRATION button will be visible only if the Flex is connected
to the computer by USB.
Dirac Live room
correction can be turned on and off with the DIRAC ON/DIRAC OFF button. The remote control can also be used as a more convenient way to turn Dirac Live on and off.
To select the input source, drop down the selector menu and select the desired input:
Info
The input source can also be selected
with a remote control.
The
set of inputs that can be selected can be set up by clicking on the CUSTOMIZE SELECTION button. Any input
source can be enabled or disabled here. This is a convenience feature so that unused inputs do not have to
be cycled through when using the remote control. Use this menu to deselect unused inputs:
To import a preset from a file, click the IMPORT button, then navigate to a saved preset file and click
Open. The preset will be loaded from the file. If Device Console is currently
connected to the miniDSP Flex (DL), the preset will also be sent to the Flex.
To export a preset to a file, click the EXPORT button. Navigate to the desired location in the file system,
change the file name if desired, and click Save.
Info
It is not possible to load a preset from the Flex back to the Device Console.
We strongly recommend always exporting your presets to files when they are complete, to ensure that
you don't lose them in future.
Tip
To copy all settings from one preset to another, export the first preset to a file, select the
second present, and then import the file that you just saved.
miniDSP Device Console will check for firmware updates when a device is connected, and notify you if
one is available. To manually update the firmware, click the Refresh DSP Firmware
and Refresh XMOS Firmware button.
Tip
You do not need to use these buttons in normal operation. They are there in case your
Flex has
issues and/or miniDSP Support instructs you to refresh the firmware.
To convert to the standard version (without Dirac Live), click on
Convert to Flex.
After confirming, the device will disconnect and you will need to reconnect to it.
The miniDSP Flex (DL) presets will not be converted to the Flex.
When reconnecting, you can choose to either restore all presets to their default values
or to use existing Flex presets.
Note that this operation is fully reversible: you can freely convert back and forth between
the standard and Dirac Live versions.
Click on "Idle Timeout" to set the idle timeout to a value between 3 and 120 seconds. After this period of time, the
front panel will turn off. It will turn on again on any user interaction.
The front panel and rotary encoder provide access to the key features of the
miniDSP Flex (DL). These can also be accessed with a remote control.
To change the volume, rotate the encoder. The volume changes in
0.5 dB steps. Minimum volume is -127.5 dB and maximum volume is 0.0
dB.
To select a preset, press the encoder knob once, then
rotate until the desired preset number is displayed. Either press
the encoder again or wait for a second. The display will show "Pls
wait..." while the new preset is being loaded, and then return
to normal.
To select an input source, press the encoder knob twice, then rotate
the encoder until the desired input source is displayed. Either press
encoder knob or wait for a second and the display will
return to normal with the new input selected.
To turn Dirac Live filtering on and
off, press the encoder knob three times, then
rotate the encoder knob until the desired selection
(on or off) is displayed. Either press the encoder knob or wait for a
second and the display will return to normal with the Dirac Live
status:
To mute output, press and hold the
encoder knob. When the display changes to show "Mute?," release the
knob. To unmute, repeat this procedure.
To put the Flex into standby, press and hold the encoder
knob. When the display changes to show "Standby?," release the knob.
To take it out of standby, press the encoder knob.
Tip
All these operations can also be performed with the remote control.
DC power. The supplied 12 VDC
power supply includes a set of interchangeable power pins for USA,
UK, Europe and Australia. Fit the correct pins for your country.
Connect the DC plug to the 12 VDC power socket.
Ground lug. This can be connected to an external electrical
ground if it is necessary to ground the chassis of the Flex. This
need is uncommon, and most of the time it can be left unconnected.
USB port for control and audio streaming. Connect to an
available USB port on your computer. This port can also be connected
to a network streamer with USB output for audio playback.
Bluetooth antenna. Screw on the supplied Bluetooth antenna here.
Optical digital input. Connect a digital source using a TOSLINK
optical cable. Sample rates from 32 up to 216 kHz are supported.
Coax digital input. Connect a digital source using an RCA cable.
Sample rates from 32 up to 216 kHz are supported.
Reset button. For use in specific circumstances only.
See the Trouble-shooting and Support section.
Analog inputs. Connect an analog source here using RCA cables.
Typically, the left channel is connected to input 1 and the right
channel is connected to input 2. The maximum input voltage is 2.0
Vrms.
Analog outputs. Connect equipment such as a power amplifier and
subwoofer using RCA cables. Typically, output 1 is the left channel
and output 2 is the right channel. See Section 6.4 for a variety of
other connections. The maximum output voltage is 2.0 Vrms.
Balanced analog inputs. Connect an analog source here. The input
jacks are balanced TRS (1/4" tip-ring-sleeve) connectors, so XLR to
TRS adapter cables can be used to connect to equipment with XLR
outputs. Typically, the left channel is connected to input 1 and the
right channel is connected to input 2. The maximum input voltage is
4.0 Vrms.
Balanced analog outputs. Connect equipment such as a power
amplifier, active speakers and subwoofer here. The output jacks are
balanced TRS (1/4" tip-ring-sleeve) connectors, so TRS to XLR
adapter cables can be used to connect to equipment with XLR inputs.
Typically, output 1 is the left channel and 2 is the right channel.
See Section 6.4 for a variety of other connections. The maximum
output voltage is 4.0 Vrms.
Analog inputs. Connect an analog source here using RCA cables.
Typically, the left channel is connected to input 1 and the right
channel is connected to input 2. The maximum input voltage is 2.0
Vrms.
Digital outputs. Connect to a DAC, a home theater receiver, or
active speakers with digital input. Each digital connection carries
two output channels -- output channels 1 and 2, or output channels 3
and 4. The TOSLINK and SPDIF outputs for each pair of channels carry
the same signal and are both active at the same time. The sample
rate on the digital outputs is 48 kHz (miniDSP Flex (with Dirac Live)).
The miniDSP Flex (DL) has a range of input sources and output connections.
This section provides an overview of the basics so you can
quickly get started with playing audio.
The diagram below illustrates a basic connection of a stereo amplifier and a
pair of speakers to the miniDSP Flex (DL). Since the default settings
of the Flex route the input channels to
output channels 1 and 2 with no crossovers active, you can make this connection
for initial testing and listening.
Connecting a stereo amplifier and speakers to the miniDSP Flex (DL)
Of course, more complex connections are possible using the additional outputs. In that case, the
internal processing will need to be set up using the miniDSP Device Console.
Refer to the Signal Flow and DSP Reference sections for details.
Warning
Ensure all equipment is powered off when making audio connections.
Warning
Before playing audio for the first time, turn the volume down using the front panel encoder.
Increase it gradually after audio playback starts.
The miniDSP Flex (DL) includes the
LDAC™ and aptX™ HD Bluetooth
codecs for high-fidelity Bluetooth™ audio.
Info
LDAC is an audio coding technology developed by Sony that enables the
transmission of High-Resolution (Hi-Res) Audio content, even over a
Bluetooth connection. Unlike other Bluetooth compatible coding
technologies such as SBC, it operates without any down-conversion of the
Hi-Res Audio content[^3], and allows approximately three times more
data[^4] than those other technologies to be transmitted over a
Bluetooth wireless network with unprecedented sound quality, by means of
efficient coding and optimized packetization.
LDAC is supported on recent Android devices. If your device does not
support LDAC, you can still pair it with the miniDSP Flex (DL), but Bluetooth audio
will use a lower bitrate codec.
aptX™ HD is designed to deliver high definition (HD) audio over
Bluetooth, to help you get the very most from your music.
Pairing your mobile device with the miniDSP Flex (DL) is essentially the same
as pairing it with other Bluetooth audio devices such as headphones or
your car.
To put the miniDSP Flex (DL) into pairing mode:
Press the encoder button on the miniDSP Flex (DL) twice and rotate it until the
display shows "Bluetooth."
Press and hold the encoder and the display will change to
"BTPairing." The miniDSP Flex (DL) is now in pairing mode.
Open the Bluetooth Settings on your mobile device to
pair with the miniDSP Flex (DL). The exact appearance will vary depending on your
specific device, but this series of screenshots illustrates a
typical sequence:
If your device supports LDAC, enable it by using the
gear icon:
When Bluetooth audio is selected as the source, the usual front panel
display will display the source (lower right corner) as one of:
BT(C)
The miniDSP Flex (DL) is paired over Bluetooth.
BT(NC)
No Bluetooth device is paired.
BT(P)
The miniDSP Flex (DL) is in pairing mode.
Once paired, the miniDSP Flex (DL) will receive and play the audio stream from the
paired device. If the Bluetooth input is not selected when you start
audio playback, the miniDSP Flex (DL) will automatically switch to the Bluetooth
input source when audio is Bluetooth detected.
The miniDSP Flex (DL) accepts stereo PCM audio at sample rates of 44.1, 48,
88.2, 96, 176.4, and 192 kHz over USB. The same USB connector is used
for playing USB Audio as well as for configuration and Dirac Live
calibration.
To play USB Audio from a computer, use the
supplied cable to connect the USB 2.0 (Type B) port of the miniDSP Flex (DL)
to a free USB port on your computer.
Alternatively, connect a network streamer to the USB port. For an
example using the Raspberry Pi, see this application note on our
website:
The miniDSP Flex (DL) is USB Audio compliant, so no drivers need to be installed to play audio on macOS.
Open Audio MIDI Setup (in Applications->Utilities) and
click on its name in the list on the left-hand side.
To set the miniDSP Flex (DL) as the default audio output device, right-click
and select "Use this device for sound output":
Note that individual audio playback apps may allow the miniDSP Flex (DL) to
be selected for audio output independently of the system default. They
may also control sample rate automatically.
The miniDSP Flex (DL) sends its four output
channels to the computer over USB. Click on the Input button in
Audio MIDI Setup to see them. These can be used to monitor or record the
output channels of the miniDSP Flex (DL):
To set the miniDSP Flex (DL) as the default
audio output device, open the Settings app and navigate to the
System -> Sound page. Select the Flex with the radio button:
To change the bit depth and sample rate,
click on the rightward angle symbol >. Click the selector next to Format to pop up the sample
rate menu. Sample rates
up to 192 kHz can be selected:
The miniDSP Flex (DL) sends its four output channels to
the computer over USB. These can be used to monitor or record the output
channels after DSP processing:
To avoid Windows resampling the outputs from the Flex, set the sample rate
here to the DSP's internal sample rate: 48 kHz.
If your audio application supports ASIO®
audio drivers, we recommend selecting the miniDSP ASIO Driver in its
settings. This typically allows the application to automatically change
sample rate. For example, in JRiver Media Center:
The current settings of the miniDSP Flex (DL) can be viewed by opening the
miniDSP UAC2 Control Panel from the Windows Start menu (under the
miniDSP Ltd folder). This control panel allows you to view current
settings. In addition, it can be used to set buffer size, although we
recommend that you leave this setting at the default. If you are having
an issue with inadequate output volume over USB playback, check the
Volume tab.
To see the installed version of the UAC2 driver, use the About tab:
The infrared remote control provided with the miniDSP Flex (DL) controls all
key runtime functions.
Standby
Put the miniDSP Flex (DL) into standby and take it out again.
Mute
Mute and unmute audio output.
Volume
Reduce or increase the volume. Each
press changes the volume by 0.5 dB. Holding down a button will
accelerate volume change to 3 dB steps.
Media control
Have no effect
with the Flex.
Dirac Live
Enable or disable Dirac Live filtering. Dirac Live filtering will be effective
only on presets for which Dirac Live filters have been loaded.
Source selection
Cycle through the available input sources in order: TOSLINK, SPDIF (coax), USB, Bluetooth, Analog.
Preset (1 through 4)
Switch to the selected preset. It will take a few seconds for the
preset selection to complete while the processor loads the new filters from
its flash memory into the DSP.
The miniDSP Flex (DL) can be used with a Logitech®️ Harmony®️ remote. The
same functions as the stock remote are programmed into the Harmony
database. In addition, you can set up buttons for direct input source
selection (instead of cycling through input sources).
The procedure is as described in the application note linked below,
except that you should select "minidsp flex" as the device instead of
"shd".
With the addition of a miniDSP
Wi-DG Wi-Fi to USB
bridge, the miniDSP Flex (DL) can be remote-controlled from a phone or tablet
running Android or iOS (iPhone/iPad).
Note that currently the miniDSP Control App cannot be used over the
Bluetooth connection. It will work only if a Wi-DG is connected to the
USB port of the miniDSP Flex (DL).
Warning
If the USB port is being used for iOS/Android app control via the Wi-DG,
it cannot be used for USB Audio streaming.
The diagram below is a simplified representation of the signal flow
through the miniDSP Flex (with Dirac Live):
Signal flow of the miniDSP Flex (with Dirac Live)
The audio signal flows from left to right in the above diagram:
One of the five stereo input sources is selected by the user. Volume
control is applied before any further processing.
The selected input signal is passed through stereo Dirac Live processing.
The processed input channels are fed into a
2-in 4-out routing
matrix that routes/mixes the two processed input channels to the
four output channels.
Each output channel runs a comprehensive set of DSP functions:
parametric EQ, crossovers, level metering, delay, output gain and compressor.
The combination of the routing matrix and the crossover blocks on the
output channels is what gives the miniDSP Flex (DL) the ability to be used in many
different system configurations.
Tip
When using Dirac Live,
first set up the matrix mixer and the output channel crossovers using the
miniDSP Device Console. Then in the Dirac Live app, export the generated
filters into the same-numbered slot. For example, if you set up preset 2 in DC,
export the filter in Dirac Live to slot 2.
Each input channel has a customizable label displayed at the top of the
channel strip. To change
the label, click on it, type a new label (up to 12 characters), and
press the Enter key.
Displays the current signal level in real time, after volume control and channel gain have been applied. (For example, if the
actual input signal level is -10 dB, the volume setting is -20 dB and the channel gain is 0 dB, the meter will show -30 dB.)
Info
The Flex must be connected and online
to display signal levels.
Each input channel can be sent, or routed, to any number of output channels.
This is accomplished with the routing buttons at the top of the
output channel strips:
In addition to routing, the miniDSP Flex (DL) supports mixing, where
input channels can be mixed at different levels to a single output channel. The complete
set of routing controls along the top of the output channel strips is therefore also called
the "matrix mixer."
To mix both input channels to an output, select both of them:
By default, each input channel is mixed at "full strength" – that is, without any attenuation, as
indicated by the "0 dB" label. To mix at different levels, click on the level button and either use the slider
or enter the desired value directly. The gain of the signal being mixed can be
set to a value between -72 and +12 dB.
In this example, both channels are mixed at half level (-6 dB):
This section illustrates a few examples of signal routing. The input and output channels have been
renamed by typing in the labels at the top of the channel strips. We recommend you do the same to
help avoid errors.
The diagram below shows the connections for a stereo pair of speakers plus a single
subwoofer. This is the simplest type of active system, so is a good introduction to the
advanced features of the miniDSP Flex (DL). It is also known as a "2.1" system.
The stereo amplifier is connected to outputs 1 and 2, and the subwoofer is
connected to output 3. In the diagram, it's a powered subwoofer. (Never connect
an output of the Flex directly to a subwoofer driver or speaker.)
Typical connections for subwoofer integration with the miniDSP Flex (DL)
The routing for this system is:
The left and right inputs are sent to outputs 1 and 2. Both inputs are summed and sent to output 3.
Outputs 1 and 2 will usually have a high pass crossover filter to remove the sub frequencies, and output
3 will have a low pass crossover filter.
The diagram below shows the connections for a two-way active
speaker. Amplification can be either a multichannel amplifier, or a pair of stereo amplifiers.
Typical connections for a two-way active speaker using the miniDSP Flex (DL)
The routing for this system is:
The left input is routed to outputs 1 and 3, and the right input is routed to outputs 2 and 4.
Each output channel will need a low pass filter for the woofers or a high pass filter for the
tweeter.
Click on this button to open the parametric EQ settings window for that output
channel. There are 10 parametric EQ filters on each input channel.
See Parametric EQ.
A delay of up to 30 ms
can be applied to each output channel. To set the
delay, click on the delay button for a channel. The delay value can
be entered by moving the slider or it can be entered numerically. The up and down arrows can be used to change
the delay in 0.02 ms increments.
The gain of each output channel can be set to a value between ‑72 dB and
+12 dB. (0 dB, the default, is unity gain or no change in level.)
To set the gain, click on the Gain value of that channel.
The gain can
be entered by moving the slider or it can be entered numerically. The up and down arrows
can be used to change
the gain in 0.1 dB increments.
Parametric equalization (PEQ)
is a flexible type of
equalization filter.
It can be used to correct for errors in loudspeaker
output, to compensate for acoustic room effects, and to tailor the
overall system response for best sound.
Click on the PEQ button to open
the parametric equalizer settings window:
The miniDSP Flex (DL) has
10 parametric EQ filters on each output channel.
The graph displays the response of each filter in a light color and the
combined response of all
enabled filters as a heavy line. For example, the
curve shown above has a low-shelf boost at 100 Hz, a dip at 500 Hz, and a
high-shelf cut at 5000 Hz.
Change the frequency and gain of each filter by clicking and dragging on the drag-points
(colored circles) on the graph. The full set of parameters can be edited as described below.
Hovering the mouse over any curve brings up an overlay showing the
frequency and the gain at that frequency.
The parametric EQ filters can be set in either basic mode or advanced mode. Basic mode allows
each filter to be specified simply by providing four parameters, while advanced mode requires
the use of a separate design program.
For PEAK filters, this is the center frequency of
the peak or dip. For HIGH_SHELF and LOW_SHELF filters, this is
the frequency at which the gain is half of the set value.
For ALL_PASS
filters, this is the center frequency of the phase shift.
For PEAK filters, this is the gain in dB at the
center frequency. For HIGH_SHELF and LOW_SHELF filters, this is
the gain in dB reached at high or low frequencies respectively. A filter
has no effect if its gain is set to 0 dB. Gain can be adjusted in
increments of 0.1 dB up to +/- 16 dB.
This item is not present for the ALL_PASS
filter type.
Q controls the "sharpness" of the filter. For
PEAK filters, lower Q gives a broader peak or dip, while higher Q gives a
narrower peak or dip. For HIGH_SHELF and LOW_SHELF filters, Q
controls how quickly the filter transitions from no gain to maximum gain.
For
ALL_PASS filters, higher Q gives a steeper phase transition.
Except for filter type, parameters can be edited by:
Clicking and dragging on a slider. Once the focus is on a slider,
you can also change the value with the left and right arrow keys on the keyboard.
Clicking in the value field and typing a new value. Once the focus is in a value field,
you can also change the value with the up and down arrow keys on the keyboard.
Move the mouse cursor over the parameter and use the mouse scroll wheel or equivalent gesture on a trackpad.
The Tab key will move focus between fields; Shift-Tab will move in the opposite direction.
To jump quickly between the parametric EQ window of different channels, enable the
Menu switch at the top right. Jump to a different channel by clicking on the
row of buttons:
Each
channel can be linked to one other channel. When a channel is linked to
another, the PEQ settings of that channel are mirrored to the other.
Typically, corresponding channels on the left and right are linked: for
example, left and right tweeter and left and right woofer.
To link a
channel, enable the
Menu switch at the top right. Click on the channel that you want to link the current channel to.
After confirming, an indicator appears to show the new link:
Info
A channel can be linked to only one other channel. That is, you cannot link three or more channels.
To reset all parametric EQ bands of the current channel to the default settings, enable the
Menu switch at the top right and click the RESET TO DEFAULT button:
Warning
This operation is not reversible, so be sure you mean to reset all bands! Save your settings
to a file first if you need to keep any.
Advanced mode enables custom biquad programming for
almost infinite flexibility in filter implementation.
In this mode, each individual filter
to be specified by its biquad coefficients. The coefficients must be generated by a design program
such as Room EQ
Wizard (REW). The design program must be set
for a 48 kHz sample rate.
Each parametric EQ filter can be individually switched to advanced mode. Paste in the
coefficients in the format shown and click on the PROCESS AND APPLY button for them to take effect.
Click the RESET button to set the coefficients back to their default values.
Info
Switching between basic and advanced mode does not convert between Frequency/Gain/Q and biquad
parameters.
The basic parameters and advanced biquad coefficients for each filter are
stored independently and the selected set is used for processing.
To load the file, select the Menu switch at the top right and click the LOAD BIQUADS FILE button.
The filters are set to advanced mode starting at EQ1.
If the loaded file contains fewer than 10 biquads, then only that number of
EQ filters will be set to Advanced mode. For example, if loading a file with six biquads, EQ1 to EQ6 will be set to
Advanced mode, and EQ7 to EQ10 will not be changed. This can be used to good effect in cases
where manually-set parametric EQ (in Basic mode) is required in addition to parametric EQ developed
with a design program.
If the loaded file contains more than 10 biquads, then only the first
10 will be loaded.
The community-developed biquad calculation spreadsheet allows many
filter types to be calculated, including notch filters, Linkwitz
transforms, and filters with arbitrary Q-factor. Access this spreadsheet
here (requires Microsoft Excel):
Room EQ Wizard is a free acoustic measurement and analysis tool,
available for Windows, Mac and Linux platforms. It includes the ability
to automatically generate a bank of parametric EQ biquads based on a
measurement. These coefficients can be saved to a file from REW and
loaded using the Device Console. Room EQ Wizard can
be downloaded from:
Crossovers "split" the frequency band to send to different drivers. In a
two-way loudspeaker, for example, a low pass filter is used to remove high
frequencies from the signal sent to the woofer, and a high pass filter
is used to remove low frequencies from the signal sent to the tweeter.
When integrating a subwoofer, high pass filters are used on the speakers
and a low pass filter on the subwoofer.
Crossover filters can also be
used to limit low frequency content delivered to a speaker or subwoofer,
to help protect it from over-excursion.
Unlike conventional analog crossovers, the flexibility of DSP allows a
completely arbitrary mix of different filter slopes and types. Filters
can be set at any frequency or disabled completely. This allows maximum
flexibility in matching your crossover to the acoustic characteristics
of the loudspeaker drivers.
Each output channel has independent high
pass and low pass crossover filters. Click on the CROSSOVER button to
open the crossover settings window:
The current channel is displayed in orange, with the others displayed in
grey. Hovering the mouse over the curve brings up an overlay showing the
frequency and the attenuation at that frequency.
The crossovers on each output channel can be set in either basic mode or advanced mode. Basic mode allows
each crossover filter to be specified simply by providing two parameters, while advanced mode requires
the use of a separate design program.
In basic mode, there are two crossover filters on each output channel: high pass
(removes low frequencies) and low pass (removes high frequencies). In basic mode, these two
filters are completely independent and each has its own settings.
Sets the nominal cutoff frequency of the crossover. In actual fact, the
crossover has a more or less gradual transition from "full on" to "full
off," as determined by the filter slope.
Selects the type and slope of the filter. The steeper the slope, the
more quickly frequencies above or below the cutoff frequency are
attenuated. There are three types of filter:
Butterworth (BW)
Available in 6, 12, 18, 24, 30, 36, 42, and 48 dB/octave, Butterworth
crossover filters are 3 dB down at the cutoff frequency.
Linkwitz-Riley (LR)
Available in 12, 24, and 48 dB/octave, Linkwitz-Riley crossover
filters are 6 dB down at the cutoff frequency.
Bessel
Available in 12 dB/octave only, a Bessel filter gives a more gradual
roll-off through the crossover region.
Except for filter type, parameters can be edited by:
Clicking and dragging on a slider. Once the focus is in a value field,
you can also change the value with the left and right arrow keys on the keyboard, or by
using the mouse scroll wheel or equivalent gesture on a trackpad.
Clicking in the value field and typing a new value. Once the focus is in a value field,
you can also change the value with the up and down arrow keys on the keyboard, or by
using the mouse scroll wheel or equivalent gesture on a trackpad.
If the focus is on the Enable button, it can be toggled with the Space key.
Move the mouse cursor over the parameter and use the mouse scroll wheel or equivalent gesture on a trackpad.
The Tab key will move focus between fields; Shift-Tab will move in the opposite direction.
To jump quickly between the crossover window of different channels, enable the
Menu switch at the top right. Jump to a different channel by clicking on the
row of buttons:
Each
channel can be linked to one other channel. When a channel is linked to
another, the crossover settings of that channel are mirrored to the other.
Typically, the corresponding
drivers on the left and right channels are linked: left and right
tweeter, left and right woofer, and so on.
To link a
channel, enable the
Menu switch at the top right. Click on the channel that you want to link the current channel to.
After confirming, an indicator appears to show the new link:
Info
A channel can be linked to only one other channel. That is, you cannot link three or more channels.
By default, the crossover graph shows the crossovers of all four output channels.
The current channel is displayed in orange and the others in black.
If you wish to view only a subset of channels, use the switches to the right of the graph to select
channels to hide. In this example,
channel 2 is the current channel. Channels 1 to 3 are visible, while channels 4 to 8 are hidden:
The crossover settings of channels 1, 2 and 3 can be changed by using the channel selection buttons at the top of the window
to jump between them. As long as you stay on these channels, the set of hidden channels stays the same.
However, if you switch to a hidden channel, that channel then becomes visible. (For example, if you switch
to channel 6, then channels 1, 2, 3 and 6 will be visible.)
The crossover block on each output channel has eight biquads. In Basic mode, four are allocated
to the high pass filter and four to the low pass filter. In Advanced mode, however, the eight biquads
are all specified by their coefficients, in the following format:
These biquads can be used for almost
infinite flexibility (within the limits of eight biquads). For example, you could use two biquads
for a fourth-order high pass filter, and the remaining six for parametric EQ.
Paste the coefficients into the text box, then click on the
PROCESS AND APPLY button for them to take effect.
Info
Switching between basic and advanced mode does not convert between Frequency/Filter type and biquad
parameters.
The basic parameters and advanced biquad coefficients for each filter are
stored independently and the selected set is used for processing.
The community-developed biquad calculation spreadsheet allows many
filter types to be calculated, including notch filters, Linkwitz
transforms, and filters with arbitrary Q-factor. Access this spreadsheet
here (requires Microsoft Excel):
Room EQ Wizard is a free acoustic measurement and analysis tool,
available for Windows, Mac and Linux platforms. It includes the ability
to automatically generate a bank of parametric EQ biquads based on a
measurement. These coefficients can be saved to a file from REW and
loaded using the Device Console. Room EQ Wizard can
be downloaded from:
The compressor reduces the gain of an output channel when the audio
signal reaches the level specified by the Threshold parameter. The
gain of the channel will be progressively reduced as the signal
increases above the threshold, according to the Ratio parameter.
This can be used to limit the power delivered to speakers and thus
reduce the risk of damage from overdriving.
This
screenshot shows an example Compressor setting:
(Note that the compressor algorithm is bypassed by default, so click on
the Bypass button to see the curve as shown here.)
In this example, the threshold is set to −20 dB, so the compressor will
activate when the signal on that channel reaches −20 dB (relative to
full output). The ratio is set to 2, so if the input signal level to the
compressor then increases by 10 dB, the output level will increase by
only 5 dB. If the input signal level to the compressor is at full scale
(0 dB), then the output level will be limited to −10 dB.
Two additional parameters control the action of the compressor: the
attack time and the release time. These two parameters govern how
quickly the compressor activates when the signal level exceeds the
threshold, and how quickly it deactivates when the signal level reduces.
The optimum settings may need to be tuned by ear. For more information,
see the Wikipedia article Dynamic range
compression.
The compressor block is disabled by default. To enable it, click on the Enable Compressor
switch. The graph will change from a dashed grey line (disabled) to a solid colored line (enabled).
Clicking and dragging on a slider. Once the focus is on a slider,
you can also change the value with the left and right arrow keys on the keyboard.
Clicking in the value field and typing a new value. Once the focus is in a value field,
you can also change the value with the up and down arrow keys on the keyboard.
Move the mouse cursor over the parameter and use the mouse scroll wheel or equivalent gesture on a trackpad.
The Tab key will move focus between fields; Shift-Tab will move in the opposite direction.
To jump quickly between the compressor window of different channels, enable the
Menu switch at the top right. Jump to a different channel by clicking on the
row of buttons:
The miniDSP Flex (DL) implements stereo Dirac Live correction. That is, there
are two channels of Dirac Live correction. After stereo Dirac Live correction, the
audio signal is routed to the four output channels for processing such
as parametric EQ and crossovers.
Dirac Live uses from 9 to 17 measurements made at different locations
in your listening room to calculate
correction filters for your speakers
(and integrated subwoofer if you have one). You will take
these measurements using the miniDSP processor, the Dirac Live
application running on your computer, and a miniDSP UMIK-1 or UMIK-2 measurement
microphone.
Info
This software runs on Windows 10 or later and macOS 10.14 Mojave or later. Other versions of Windows or macOS/OS X are not supported.
The diagram below shows typical connections for acoustic measurement
with the Dirac Live application. Usually, you do not need to change any
of the audio connections, as the Dirac Live test signals are generated
inside the miniDSP processor.
Connect the supplied USB cable (type A to type B) from the miniDSP Flex (DL)
to a USB port on the computer.
Connect the supplied USB cable (type A to mini type B or type C, depending on
version) from the
UMIK-1 or UMIK-2 to a USB port on the computer.
System connections for a stereo Dirac Live system
Place the UMIK-1 microphone into a microphone stand and position the
computer and cabling so that there is enough freedom of movement to move
the microphone into the needed locations. A small tripod stand is
supplied with the UMIK-1, but a larger stand with boom arm can be used
if desired. If necessary, a USB extension (up to a total USB cable
length of 5 meters) can be used. In larger spaces, an active USB
repeater may be needed.
For stereo systems, we generally recommend that the microphone be
pointed between the speakers and the "0 degree" calibration file
used. However, many of our customers point the microphone at the ceiling
and use the "90 degree" calibration file. Feel free to try it both ways.
Prior to performing acoustic measurements, optimize your loudspeaker and
listening positions. Start with the recommendations of the manufacturer
of your loudspeakers. Loudspeakers designed for home hifi use typically
perform best away from the walls, whereas speakers designed for studio
use may be designed for use closer to walls or other surfaces. With
Dirac Live, you have more freedom with loudspeaker placement, but the
best result will still be achieved if optimal loudspeaker placement is
used together with Dirac Live.
Speaker and microphone positioning for stereo Dirac Live
You should also experiment with toe-in –
many loudspeakers benefit from pointing directly at the listening
position or even slightly in front. The listening position should be
away from the rear wall, as placing the listening chair or sofa right
against the wall will result in increased early reflections and changes
in timbre.
The first measurement must be taken at
the exact center of the listening area, so that Dirac Live can set
the levels and delays of each speaker. While this will usually place the
microphone an equal distance from both speakers, Dirac Live will adjust
in cases where it is not. Additional measurements are taken at other
positions around the listening area and at different heights from the
floor. All measurement positions should have acoustic "line of
sight" – that is, the direct signal from the speaker to the microphone should
not be obstructed by furniture or other objects.
Each UMIK-1 and UMIK-2 measurement microphone is individually calibrated to
ensure accuracy. To download the unique calibration files for your
microphone, go to the
UMIK-1 page
or the
UMIK-2 page.
Enter your microphone's serial number. It is in the form xxx-yyyy and
labelled on the microphone.
Two calibration files are downloaded:
The regular or "on axis" calibration file will be saved to the
computer with a name that is the same as the serial number e.g.
"7001870.txt". This is the one that you should use when pointing the
microphone towards the speakers.
The second calibration file has "_90deg" in the name e.g.
"7001870_90deg.txt." This corrects the frequency response when the
microphone is oriented at 90 degrees to the sound source, such as
when pointing the microphone towards the ceiling.
If you have used the miniDSP Device Console to set up routing, crossovers or parametric EQ on
the output channels, check that you have the correct preset
selected before starting Dirac Live. This is because the Dirac Live test
signal originates in the Dirac Live blocks and therefore
passes through the routing matrix and the output channel
processing. Refer to the diagram in the Signal Flow Overview
page to refresh your memory if needed.
For example, suppose that you have set up preset 2 with routing and
crossovers for a subwoofer. Before starting Dirac Live with the
START CALIBRATION button, select preset 2 from within
the miniDSP Device Console or with the remote control.
If you haven't plugged in your UMIK-1 yet, plug it in now.
Tip
Be sure to plug in the UMIK-1 before starting Dirac Live.
Check that UMIK-1 is set as the default recording device¶
Open the Windows Control Panel and select
the Sound pane then the Recording tab. Check that the UMIK-1 is set as
the default recording device. If it is not, then click on it and then on
the Set Default button.
Open the Windows Settings and select
Security, then Privacy then Microphone. Check that "Allow apps to access
your microphone" is turned on.
Allow apps to access your microphone
Scroll down and check that "Allow desktop
apps to access your microphone" is also turned on, and that
DiracLive.exe is listed with the applications that are allowed to access
the microphone:
The heightened security in recent versions
of macOS can cause some difficulties with the Dirac Live application
accessing the UMIK-1. The Dirac Live application should request
microphone access when it first runs. A dialog like this will appear:
When it does, click OK. You can
confirm that access has been granted by opening System Preferences, then
going to the Security&Privacy pane, selecting the Privacy tab, then
clicking on Microphone at the left:
The application "Dirac Live" should appear here. You will typically have
additional applications to which you have given microphone access.
To start Dirac Live calibration, click on the START CALIBRATION button
The Device Console will disconnect
and the Dirac Live application will start.
Info
Be sure to start the Dirac Live
application from within the Device Console using the START CALIBRATION button.
If you open the Dirac Live application by itself, it
will not be able to detect the Flex.
Info
Dirac Live calibration can not be performed
over the network, so the START CALIBRATION button will be visible only if the Flex is connected
to the computer by USB.
The Dirac Live application works better
when you are logged into your Dirac account. The first time you start
it, the option to log in or create an account appears.
You can log in here if you have a Dirac account. If you do not have an
account, you can create one by clicking on "Create or manage your
account." This will direct you to the Dirac Live website.
If you wish, you can click "Proceed without logging in" to perform a
calibration. You can return to this tab later and create an account by
selecting Login from the hamburger menu.
The following functions are enabled only if you are logged into your Dirac
account:
Load a previously saved project. You can only load an existing
project if you are logged into a Dirac account. If you don't log in,
you will be able to save your measurements as a project, but you
will not be able to open them again.
Other optional or paid-for features. miniDSP always provides the
"full" version of Dirac Live with its products, so you do not need
to log in to use the full version of Dirac Live if you have a
miniDSP product.
The only device visible here will be the miniDSP Flex (DL).
If necessary, close the help overlay that obscures
the device icon. Click on the device icon to select it and proceed to
the next tab.
Once the output device has been selected,
the main tab menu appears. This menu indicates which tab you are
currently viewing. Each tab has buttons to navigate to the previous and
next tabs. You can also navigate directly to a tab by clicking on its
name in the menu.
When events happen, a notification appears
to the top right of the window. Normal notifications are in green, with
warnings and errors in black and orange:
To close a notification, click on the "X" at its top right.
The help overlay appears on each tab the first time you open it. You can
scroll to read all of it. Note that the recommendations in the Help
overlay are generic -- that is, they do not necessarily apply to the
specific device that you are using. For example, it may say "If your
HiFi device has a microphone..." -- this does not apply to the miniDSP Flex (DL).
To close the overlay, click on the icon at
its top right:
To open the help overlay again, click on
the question mark icon:
The navigation buttons to the lower left and lower
right of the main display area link to the next and previous tab, using
the order in which a complete calibration is normally performed.
To save the project, click on the hamburger icon and select Save
project.
To load the project, click on the hamburger
icon and select Load project.
Info
If you save the project and you are not logged into your Dirac account, you may not be able to load that project again. Be sure to log into your Dirac account before saving the project.
Use this tab to set the output level and microphone recording level.
For best calibration results, ensure that
the environment is reasonably quiet. Low frequency noise in particular
can disrupt the calibration.
Check that the Master output slider is set low.
Click on the triangular button for the Dirac 1 channel and gradually
increase the Master output slider. You should hear pink noise
playing from the left front speaker. Continue to increase volume
until it is at a moderate level, such that your voice would have to
be raised to converse with someone sitting next to you.
Increase the Mic gain slider. Setting it so that the level meter reads between
-20 and -30 dB will generally give good results.
Click on the triangular button for the Dirac 2 channel and confirm
that the level is around the same as the first channel.
As a safety feature, output volume is
limited to -32.0 dB. If you try to raise it higher, a lock icon will
appear. To raise it higher, click the lock. It will "unlock" and then
you can raise the slider further. The lock icon will automatically lock
again after a few seconds.
Once you have set the input and output levels, click the Proceed to
Select Arrangement button.
Info
Note: there will be only two channels shown, even if you have used the Device Console
to set up routing for a subwoofer. This is because the miniDSP Flex (DL) is a
stereo Dirac Live processor.
Therefore, it treats each speaker combined with the subwoofer as a single channel. There is no
separate Dirac Live channel for the subwoofer.
Tip
While the measurement technique used by Dirac Live is quite robust, low-frequency noise (traffic, HVAC, machinery, aircraft, storms) in particular can adversely affect measurement accuracy. A high level of ambient noise can degrade signal to noise ratio and prevent the algorithm from analyzing the test signal properly. Minimize the effect of any external noise, ensure that measurement signal levels are adequate, and/or choose a suitable time for performing measurements.
The Select Arrangement tab selects the type of seating arrangement
to calibrate for. Each arrangement has a different number of measurement
positions.
The menu bar shows the selection choices:
Use the menu to select the arrangement that is most appropriate for your
listening area. They are described more below.
There are no hard and fast rules on which arrangement to use, so feel
free to experiment by calibrating for more than one arrangement and
auditioning the results. Up to four different sets of
correction filters can be loaded into the miniDSP Flex (DL).
After selecting the seating arrangement, click on the Proceed to
Measure button.
This arrangement is optimized for a single
listener but also caters for additional listeners. There are thirteen
measurement positions: the same nine as the tightly-focused arrangement, plus an
additional four positions extending out to the left and right:
This arrangement is optimized for multiple
listeners. There are seventeen measurement positions: the same nine as
the tightly-focused arrangement, plus an additional eight positions that extend
the listening area out to the left and right:
The left side of the window displays a
visual guide to the measurement positions. The position that will be
measured is shown in white. To display the name of a position, hover the
mouse over it -- the position will be highlighted in blue and its name
displayed. For example:
While the visual guide indicates a suitable set of microphone positions,
these can be varied to suit individual circumstances. It is, however,
important that the first measurement, named "Main Position." is taken in the
center of the listening area, as this measurement is used to set the
level and delay of each channel.
Prior to running the first measurement,
you can optionally set a short delay before the measurement starts. To
do so, click on the small clock icon on the right side of the Measure
selected position button and choose the delay:
With the microphone in the
center of the listening area, click on Measure selected position.
The miniDSP Flex (DL) will generate a test signal, audible as a
frequency sweep through the left speaker, then the right, then through the left again.
Warning
If your speaker sounds distorted or strained during the measurement,
click Abort this measurement immediately. Go back to the Volume Calibration
screen and lower the output volume.
While the measurement proceeds, a time-domain graph of the captured
audio signal is overlaid on the screen. This graph is related to the
magnitude response but is not the same display. Its purpose is to verify
that the recorded signal level is in a suitable range.
On completion of a successful measurement, the window will update with a
graph of the measured frequency response:
If the measurement is successful, a check mark will be shown on the
visual guide and the next measurement position will be highlighted.
If the application indicates that the measurement was not successful,
you will need to take corrective action. The most common errors are
related to signal level:
The measurement signal is too low to ensure a clean capture.
The measurement signal is too high and the audio signal has exceeded
the maximum level (clipping). This is shown in red on the time
domain display.
In either of the above cases, go back to the Volume Calibration tab
and adjust Master output or Mic gain. Then re-run the
measurement.
To take the next measurement, move the microphone to the highlighted
location -- labeled "Next" on the seating visual -- then click on
Measure next position.
If the measurement produces an error that the signal was too low or
high, go back to the Volume Calibration tab and adjust Master
output or Micgain. Return to the Measure tab and then click
on Remeasure selected position. You do not need to redo measurements
that you have already successfully completed.
Repeat until all positions have a
successful measurement:
Tip
The measurement positions should be well spread out over the
entire listening area so that Dirac Live can acquire a good set of
measurements that capture the acoustic behavior of the room. Typically,
you should spread the microphone positions over a circle with a diameter
of at least a meter (three feet) and vary the microphone height from the
central position by at least 30 cm (one foot) up and down. Placing all
measurement positions too close to each other may result in
"over-correction" that will sound dry and dull.
A different set of measurement positions other than those indicated by
the visual guide and the above guidelines can be used if necessary. For
example, if the displayed measurement position doesn't provide acoustic
"line of sight" from the speaker to the microphone, then adjust the
measurement position so that it does. The important thing is to ensure
that the measurement positions are spread out over the whole listening
area and that the microphone is moved a sufficient distance vertically
as well as horizontally.
In some cases, such as when the listening area is very close to the
loudspeakers or the loudspeakers have a very narrow dispersion pattern,
the size and in particular the height of the measurement area can be
reduced, to avoid discrepancies caused by varying output response from
the speakers themselves.
Once the full set of measurements has successfully been taken,
Dirac Live has the acoustical information it needs about your
loudspeakers and listening room to create the correction filters.
The Filter Design tab shows the
frequency response of one or two channels and the target curve. The
following example shows the left channel in mauve and the default target
in yellow/black:
The target curve – shown in yellow in the above graph – is the response
you desire from the speakers after correction. This will be an average of different
locations around the listening area.
The default target curve provides two main controls: the bass shelf, and the treble
roll-off. Simply grab the horizontal lines to move the level of the bass shelf and degree
of treble roff-off (or gain, if you desire).
The grey regions to the left and right of the graph are excluded from
magnitude correction. You can adjust
the frequency range of correction by dragging the
"curtains" to either side of the graph.
For example, you may not wish to correct the bass to such a low frequency, so drag
the left curtain up to, say, 30 Hz. Or, you may be happy with the inherent
in-room response at higher
frequencies, so you can set the upper curtain frequency region to limit correction to
the modal region (up to 300 Hz, in a typical room).
Dirac Live
There are two channels listed towards the
top right of the window, in a single group:
Each group has its own target curve and range curtains. If you wish to have
different target curves for left and right, you can separate the two
channels into separate groups. To do this, click on "Left" or
"Right" and drag it out of the darker box.
The graph can be zoomed in and out by
using the mouse wheel, or with zooming gestures on a touchscreen. To
move a zoomed graph, left-click and drag with the mouse, or use the grab
and move gesture with a touchscreen. To zoom out to 100%, double-click
on the graph.
There are several controls at the right of
the graph:
Spectrum
Display the average of all magnitude responses. If the "Measured"
checkbox on the left is on, the average of the uncorrected measurements
is displayed. If the "Corrected" checkbox on the right is on, the
average of the response after correction is displayed.
Group colors
If checked, the graph colors are the same for all channels in a group.
This can be individually selected for the measured and corrected
responses.
Spread
Display the range of corrected responses
as a lighter highlight on the graph. For example:
This can be individually selected for the measured and corrected
responses.
Target curve
Display the target curve for the displayed group.
Curtains
Display the correction frequency range for the displayed group. This is
shown as "curtains" to the left and right of the graph.
Detected Range
Display two cursors indicating the frequency range within which Dirac
Live has determined the speakers to have reasonable output. Note: at low
frequencies, environmental noise can cause this estimate to be
incorrect.
To convert the target curve to use a set of control points, right-click
and select "Convert shelves to control points":
The target curve is converted to the equivalent but with control points,
shown as dots on the target curve. The curve can then be edited as shown here:
To switch between the shelving controls and the control points, use the
icons located with each group on the right side of the window:
You can save the current set of groups,
target curves and curtain frequencies by clicking on the Take
snapshot button:
Snapshots will by default be named as the
current time. They can be renamed for easier selection later on. Click
on the right side of the button and then on the pencil-shaped icon.
Enter the name and press the Enter key:
To select a saved snapshot, click on the right side of the button and
then click on the desired snapshot.
Info
It is not currently possible to delete a snapshot.
To reset a target curve and the range
curtains to the default, drop down the menu from the hamburger icon and
select Set default target curve, then select the group to reset:
Target curves can be saved and loaded
individually for use at a later time or in other projects. To save a
target curve, drop down the menu from the hamburger icon and select
Save target curve, then select the group:
To load a target curve, drop down the menu
from the hamburger icon and select Load target curve, then select
the group:
Care should be taken to create a target curve that works well with your
speakers and room, as well as suiting your personal preferences. Small
changes to the target curve can have significant effects on the tonal
quality of the system, so it is important that you experiment with
different target curves.
If you initially don't achieve a satisfactory result, ensure that you
have spread your measurements over a sufficiently large area and with
sufficient variation in height. The following guidelines will help you
understand how to adjust your target curve.
All loudspeakers have a natural low frequency roll off. Setting the
target curve to boost the region below the speaker's natural roll off
frequency may result in overdriving the speakers, especially with
smaller home loudspeakers and depending on your listening habits. A
system with capable subwoofers integrated into it, however, will support
much more low-frequency output.
The target curve is the desired measured response of loudspeakers in a
room, in contrast to measurements made of a loudspeaker during its
design under anechoic (measured in free space) conditions. While
high-quality loudspeakers are usually designed for a flat on-axis
anechoic response, these same speakers when placed into a listening room
will tend to have a downward-sloping or "tilting" response at high
frequencies, due to the effects of limited dispersion at high
frequencies and greater acoustic absorption.
A completely flat in-room response is therefore usually not desirable
and will tend to sound thin or bright. Start with a target curve that
follows the natural behavior of your speakers in your room, and then
experiment with greater or lesser degrees of tilt in the treble region
to obtain the most natural timbral balance.
A completely flat response at low frequencies, with complete elimination
of peaks due to room modes, may sound light in the bass. Typically, a
slight increase in the target curve below 100 Hz will give a more
balanced sound.
In some cases, it may be helpful to adjust the target curve to follow
dips in the measured response. This can occur where, for example, the
listening area is very close to the speakers and the measurements
exhibit a dip caused by the vertical response of the speakers
themselves. Experiment with a target curve that follows the dips. You
may also wish to try a different set of measurement positions.
Below is an example impulse response
display. This example shows the original impulse response at the top and
the corrected and delayed impulse response below.
There are several controls at the right of
the graph:
Impulse response
Display the impulse response corresponding to the average of all
magnitude responses. If the "Measured" checkbox on the left is on, the
impulse response corresponding to the measurements is displayed. If the
"Corrected" checkbox on the right is on, the impulse response
corresponding to the corrected response is displayed.
Group colors
If checked, the graph colors are the same for all channels in a group.
This can be individually selected for the measured and corrected impulse
responses.
Show latency
Show the impulse responses including actual delay through the miniDSP Flex (DL).
Separate curves
Show the measured and corrected responses on separate graphs. See the
previous page for an example.
Dense grid
Display additional grid lines on the graph.
Cursors coordinates
Display a vertical cursor line where the mouse is, together with the
time delay and impulse response value at the cursor position.
To export a filter into a slot, click on the slot. At the
right, enter the name and description and click Export filter:
A circular progress indicator will indicate
that the filter is being downloaded to the miniDSP Flex (DL). When filter export
is complete, the UI will return to the Measure tab so you can adjust
the target curve to create a different set of filters. Return to the
Filter Export tab to view the list of loaded filters:
Check your device volume
Check your device volume before you return to normal playback. The Dirac Live app
may have reduced the device volume as a safety feature.
To overwrite a filter slot, select a slot
with a loaded filter and click on Replace existing filter.
Info
If you have set up any back-end processing (subwoofer, active speaker, etc.), be sure to load the filter into the same slot as the preset selected when running the Dirac Live calibration measurements.
To audition different filters while still
running the Dirac Live application, use the filter selection checkboxes
in the sidebar to choose the filter set to listen to:
Info
Selecting a filter here will also select the matching
preset in the back-end processing set up by miniDSP Device Console.
If you have used the older "plugin" software with your miniDSP Flex (DL), you will need to
perform a one-time migration procedure to use it with miniDSP Device Console.
Follow the procedure described below. Please be sure to perform all
steps in order.
Tip
If you are having trouble using your device with Device Console, we recommend
that you start
again from step 1 and make sure that every step is done in order.
1. Check that you have the latest version of Device Console¶
Start miniDSP Device Console. Go to the Settings page and click on CHECK NOW:
If a newer version of Device Console is available, you will see a popup offering to
download and update to the latest version. Click YES.
In the Settings page of Device Console, click on Install USB Audio Driver:
Device Console will confirm that you want to install the driver.
Click on the PROCEED button.
Info
You may already have the latest version of the driver installed. However, we recommend that you always
reinstall when migrating to DC, to ensure that this is the case.
Connect your Flex to your computer and power it on.
It will appear on the Discover Devices page
of Device Console. Click on it.
If the Flex needs a DSP Firmware update, this window will appear:
If this message appears, click on Update Now. (If it does not appear, you do not need a
firmware upgrade and can jump straight to the next step.)
After the firmware update completes, the Flex will restart.
Tip
In some cases, the Flex will not restart properly. If you experience odd symptoms, such as Device Console not returning to the Discover Devices page, the
Flex not appearing on the Discover Devices page, or the DSP Firmware update screen appearing again,
power cycle the Flex and restart Device Console.
Locate your Flex on the Discover Devices page. Click on
it. You will see a message like this:
(If you do not see this message, you do not need to upgrade the firmware and you can skip ahead to the next step.)
Click on the Download button to download the Migration pack containing all necessary files.
Alternatively, you can download it at any time by clicking on this link:
After downloading, if your browser did not automatically unzip the downloaded file, unzip it manually
by double-clicking on it (macOS) or right-click on it and select "Extract All" (Windows).
The firmware in your Flex must be upgraded for it to work properly
with Device Console. This is a one-time operation that must be performed with the
firmware upgrade utility contained in the Migration Pack.
Check that the Flex is connected to your computer via USB
and that it is powered on.
Start the firmware upgrade program.
For Windows, navigate to:
XMOS Firmware ⟩ Firmware_Upgrade_Tools ⟩ Windows ⟩ miniDSPUAC2Dfu
Double-click on the miniDSPUAC2Dfu.exe program to run it.
For macOS, navigate to:
XMOS Firmware ⟩ Firmware_Upgrade_Tools ⟩ Mac
Double-click on the miniDSP USB DFU.app program to run it.
The firmware update utility will start:
Click on Browse and select the firmware file from the XMOS_Firmware folder of the migration pack. It will have a name like "miniDSP_Flex_XMOS_v1.59_up.bin." (The version number embedded in the file name may change.)
Warning
Do not proceed if you cannot locate the correct firmware file. The name of the firmware file must begin with "miniDSP_Flex_XMOS" or you could damage your Flex.
Click on the Start button.
You will see a progress bar as the upgrade proceeds.
Warning
DO NOT DISCONNECT THE USB CABLE OR POWER FROM THE Flex WHILE FIRMWARE UPGRADE IS IN PROGRESS. DOING SO MAY “BRICK” YOUR Flex.
Once the firmware upgrade completes, you will see a message that the upgrade completed successfully.
Click on Exit.
Now that your Flex firmware has been updated to a version that is compatible with
Device Console, future firmware upgrades can be done directly from within Device Console.
Existing presets must be imported manually into Device Console. Locate your Flex on the Discover Devices page and click on it. Device Console will offer this prompt:
Click on Initialize to default settings.
Then use the import function of Device Console to import the existing presets one by one.
These can be found in your Documents folder, at the path:
If you are experiencing any connectivity issues, the first thing to
do is double-check that the miniDSP Device Console and your processor's firmware are up to
date.
The Dirac Live application use network ports 11113, 11115, 11117, 5000,
and 8080. The last two in particular are needed for the bridging app to
communicate between Dirac Live and the miniDSP processor.
If the miniDSP processor is not detected:
Double-check the USB cabling from your computer to the processor. If
you are using extenders or hubs, remove them and make a short direct
connection. Try different USB ports on your computer.
Check that your computer's firewall isn't blocking any of the ports
listed above.
Check that other applications aren't using these ports.
a. On Windows 10, open a command prompt and type the following
commands:
If there is no output, the ports are not in use. If there is output,
then try to close the application that is using them. Typically,
software that provides a local web server is likely to be the one
blocking port 8080.
Some customers have reported that changing
the following setting enabled Dirac Live to find their processor. Go to
Settings -> Privacy -> Other Devices and enable "Communicate with
unpaired devices."
If you get no sound while on the Volume Calibration tab of Dirac Live,
first make sure that you increase the volume slider in the UI (the
leftmost one).
If there is still no sound, double-check that output cabling is
connected to the right connectors on the rear panel and to the correct
inputs on downstream equipment. Make sure that downstream equipment
(e.g. amplifiers) is not muted, turned down, or turned off.
If you get no sound while playing music, double-check the following
items. Depending on the specific processor, you may be able to do this
with the aid of the level meters in the miniDSP Device Console (quit the
Dirac Live app before clicking on the Connect button in the Device Console).
Check which input source you have selected.
Check the connection and cabling between the source(s) and the
processor.
Check the master volume setting.
Check that the processor is not muted.
Check your Routing and make sure that you are routing inputs through
to outputs.
If your processor has crossovers, check your crossover settings. (In
particular, some processors have a default high pass filter set at 1
kHz on some channels as a safety measure for 2-way active speakers.
If using these channels for subwoofers, the high pass filters must
be bypassed.)
Check that output cabling is connected to the correct connectors on
the rear panel and to the correct inputs on downstream equipment
(e.g. amplifiers).
Check that downstream equipment (e.g. amplifiers) is not muted,
turned down, or turned off.
If the soundstage width is very narrow, check that you have not summed
left and right input channels and routed the sum to left and right
speakers. This will result in a mono signal and hence a narrow
soundstage.
If the soundstage is "weird", check the routing and crossover settings
and make sure that left and right are consistent. If you are
implementing an active speaker, double-check all connections on the rear
panel.
If audio playback is distorted, you may have too much gain internal to
the DSP. Dirac Live can apply up to 10 dB of gain, so the output level
should be kept lower than -10 dB to guarantee that there is no
distortion with a full-scale input signal.
This issue typically occurs when volume control is being done downstream
of the miniDSP processor. If so, set the processor master volume at -10
dB, and then use the downstream equipment to control volume.
a. The firmware versions obtained from the Display Versions button in miniDSP Device Console.
b. A clear explanation of the symptoms you are seeing.
c. A description of troubleshooting performed and your results.
d. (Recommended) For quickest resolution of your issue, include the current program snapshot.
This can be obtained from the Device Console Settings screen.
