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Before to install the setup.exe, please make sure that the Net Framework 4.0 is already installed on the PC used. After install the setup.exe and run the app “LPP240A” then the system show the following window and after few seconds the main window...
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Click the preferable connection mode and device model. (In this case, device model shall be “LPP240A”) Note: When you connect under TCP/IP mode, please input the IP address of the device; if under RS485 mode, please input the ID of the device as well.
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Com Select window the found Com ports until found the working one. In order to use the USB connection for remotely controlling the LPP240A, need first set on the unit the communication interface as USB.
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Use the button “Check” to see all COM connected on the PC. In order to use the RS485 connection for remotely controlling the LPP240A, need first set on the unit the communication interface as RS485 and assign an ID to the unit.
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IP address range: In order to use the TCP/IP connection for remotely controlling the LPP240A, need first set on the unit the communication interface as TCP/IP and assign an IP address to the unit, if not used DHCP mode. t To set the TCP/IP interface on the LPP240A the following steps have to be done: 1.
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to edit the device, double click on the name of the device present on the “List ID”...
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(1/2, 3/4) is not included in the copy tool. Default: tool which can be used to bring back the LPP240A to the default setup, which is matching the one got by the Hw reset. All parameters are flatted and the all outputs are in MUTE User (password selection): by default (when released by the factory) the password for locking permanently the LPP240A is set as “000000”.
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Password: this tool is allowing to LOCK the LPP240A by Password. When the unit is locked providing a Password, no control is accessible and to exit this status it is necessary to provide again the password allowing so to UNLOCK the unit.
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This Window is guesting a synoptic image of the several processes available on the Input and Output paths of the LPP240A, so as a series of “frames” for the selection of Input sources and Signal Routings. Input Source frame: The Signal Sources to the LPP240A inputs can be selected in between several options, in alternative to each others.
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2 way system, which in the case of the LPP240A can be identified as being a system processed by any of the 4 Inputs and 2 of the 8 available outputs.
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The way of routing the signal is here in MIXER MODE, giving the LPP240A the possibility to not only assign one input to the desired output, but also to control on any output the level of the assigned input.
In this case, it is possible, given an input level close the the max available, to have internal processes overflow. The LPP240A inner Clip limiting Algorithm is detecting the potential or already occurring overflows and is containing them up to a 12dB Headroom, with a process “distortion free”...
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Input Meters: the input meters show the Input levels from -15dBu up to +15dBu. The Limit Led is indicating if Active the RMS Compressors. The Clip Led is indicating if the Clip limiter Algorithm described here above is active on the Input path. Output Meters: the output meters show the output levels from -15dBu up to +15dBu.
INPUT WINDOW All the processes available on the Input Paths can be edited within the Input Windows. Eq Flat and Bypass Controls: the EQ Flat control is allowing to flat in one shot the 13 bands Parametric Eq. The Bypass control, instead, is allowing to temporarily Bypass the Eq, without resetting the current setting.
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“Adj” with step 1ms or “Fine” with step 10.4us (max input delay = 480ms =163.5meter) RMS Compressor: the LPP240A is implementing a powerful RMS compressor on Inputs and Outputs, which compression coefficient is calculated on the base of tabled values giving a compression step of 0.2dB.
OUTPUT WINDOW All the processes available on the Output Paths can be edited within the Output Windows. Eq Flat and Bypass Controls: the EQ Flat control is allowing to flat in one shot the 7 bands Parametric Eq. The Bypass control, instead, is allowing to temporarily Bypass the Eq, without resetting the current setting.
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“Adj” with step 1ms or “Fine” with step 20.8us (max input delay = 340ms =116meter) RMS Compressor: the LPP240A is implementing a powerful RMS compressor on Inputs and Outputs, which compression coefficient is calculated on the base of tabled values giving a compression step of 0.2dB.
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250Hz. Actually has to be said that the LPP240A is working at a sample rate of 96kHz, but the FIR both on the Input path (the Phase Correction one) and the outputs, are processing with a decimation factor of 2 , so running at 48kHz.
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Once the filter is set, it will be displayed in the window aside the parameters set. In order to make it active and performing on the Signal path, need to load it within the LPP240A clicking on the blinking icon...
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Once the set FIR is uploaded into the LPP240A, the following Icons will become available Allowing to save into a Pc directory the current edited filter for future use, or to upload from a Pc directory, previously saved FIR filters.
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FIR on Output not as Xover filter, using imported coefficients. If the user would like to use an IIR filter as Xover, being necessary in example a cutting frequency below 250Hz, but still having available the FIR in the Xover section, a trick can be used for getting the result.
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Basically, to get Lp, filters from 12dB/Oct, up to 48dB/Oct, need you to set as follow the Q, given the same cutting freq. to all of them: Butterworth 12dB/Oct -----> One Cell / Q1=0.70 24dB/Oct -----> Two Cells / Q1=0.55, Q2=1.30 36dB/Oct ----->...
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Technical Data Analog Input 2 x XLR electronically balanced Analog Output 4 x XLR electronically balanced Digital Input 1 x AES/EBU; Gain 0dBu M inimum Load 150 ohm THD+N 0.001% at 1kHz 0dBu >110dBA Ground Noise -92dBu up to 32 User Presets Device Presets 2 x 24 character LCD display with green LED back light...
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NOTE A FIR WIZARD - User Guide v1.0 The FIR Wizard is a smart and easy-to-use software tool designed for enhancing the audio quality of a loudspeaker by compensating the linear distortions, both in magnitude and phase, through FIR filtering. The tool guides the user through the loudspeaker IR measurement and allows generating state of the art equalization FIR filters, with few essential synthesis parameters, that “invert”...
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Starting Up The image below shows the home page of the FIR Wizard. From the home page, the user can choose to: 1) Start from an existing Wizard project: click the open button at top right of the screen and a file browse window appears allowing to select a Wizard project file (.zwz) 2) Start from an existing IR, that has been already measured and saved from the FIR Wizard: click the “Open existing IR”...
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Measurement Geometry After clicking the “Measure IR” button, the “Measurement Geometry” page is shown and three different geometry set up is suggested. Clicking the buttons at top left of the screen. the three measurement scenarios are shown (vertical section on the left, planimetry on the right). This is just a visualization and reminder for the user that should check every time the correct geometry and choose one of the three options.
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Here below, the three measurement scenarios are briefly described. 1) Loudspeaker system and microphone suspended at half height In this case, the measurement space should be at least 4m x 5m x 4m (L x W x H). The microphone should be in axis with the loudspeaker; obstacles should be at least 2m far from the microphone.
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2) Loudspeaker system and microphone lying on the ground The loudspeaker must be tilted to point the microphone, that should be stricly in contact with the floor. The floor should be very reflecting (smooth), especially beneath the mic: a fine metal sheet under the mic can be useful for this purpose.
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3) Ground monitor in operating position The microphone should be in axis with the loudspeaker at an height of about 1.7m, where the listener's head will stay. So, arrange the measurement choosing one of the three geometry setup and click “Next” button to proceed.
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Wiring Connect an ASIO audio card to the PC where the FIR wizard is running, with its proper connection (USB, Firewire, etc.). Connect the (normally balanced) ASIO card output (XLR or TRS) to the balanced input (XLR) of the DSP device (e.g the LPP Controller). Connect the microphone output to the mic input (normally XLR) of the audio card and turn the phantom power (if needed).
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IR Measurement Clicking the “Measure IR” button at the top left of the screen, the measurement process starts and an activity progress bar shows the activity completion status. Wait in silence for the process completion. After the sine sweep, the Wizard starts the IR computation and the measured IR is plotted on the left graph with its peak centered on zero, while the magnitude and group delay frequency response is shown on the right graph.
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Discontinuities in the group delay view may appear (as the picture above shows near 12kHz), due to the poor phase computation accuracy in the frequency points where the magnitude response has a very deep dip. Now, you can save the IR by clicking the “Save IR” button at the top right of the screen. Choosing to start from an existing IR at the Wizard home page would have lead you directly here.
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As a consequence of the windowing process, the computed IR is not accurate in the low frequency range: it is not possible to estimate the frequency response below the reciprocal of the analisys time interval, due to the time-frequency duality. So, to avoid misleading view, the frequency response plot shows a dotted line below 200Hz.
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In this page the FIR filter coefficients are computed starting from default settings (or from the parameters set of the FIR Wizard project, when a project file is recalled at the Wizard home page). The two graphs in the screen show the time and frequency response of: 1) the loudspeaker IR (red line), as it appears in the preceeding page.
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Left clicking a point in the plotting area allows to show the cursor coordinates, while right cliking shows the plot zoom menu. Let's now analize in more details the linearization capability of the computed FIR by comparing the time and frequency domain plots of the IR and the IRxFIR curves, for a flat target magnitude. In the time domain what we should ideally get is a unitary pulse centered at 0ms (Dirac delta), that is the TD equivalent to a flat frequency response.
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Looking the chart above, we can immediately note that the IRxFIR (the yellow curve) is much closer to the ideal Dirac delta than the measured IR (the red curve). This is the improvement added by the FIR filter. The slightly difference between the IRxFIR and the Dirac delta is due to the fact that the FIR filter is not working on the whole frequency response, but it is limited to a frequency range from about 200Hz to a MaxFreq (10kHz in this example).
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Let's now describe the user parameters related to the FIR filter generation. Note that, every time a user parameter is changed, the “Calculate FIR” button at the top left of the screen start blinking, meaning the FIR filter needs to be recalculated. Once the button is clicked the synthesis algorithm computes the FIR filter with the new parameters and the corresponding plots are updated.
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2) The frequency response at very high frequencies can be the real one or just what the microphone captures. If you are using a shoddy microphone, results in the high frequency range can be not affordable. This doesn't mean that a satisfactory FIR equalization can't be done on a slightly reduced band: just choose a max frequency value and try listening the result.
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- Smoothing The smoothing parameters allows limiting the FIR filter action in such a way to “correct” the IR in its average behaviour. In other words, when the smoothing is enabled, the FIR filter doesn't apply a punctual IR compensation, but it is limited to correct only an averaged IR response. Smoothing may slightly degrades the compensation capabilty of the FIR filter in axis (i.e.
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Setting up the threshold to 0dB, the smoothing is applied to the whole magnitude response, as shown in the following picture (left side plot, smoothing disabled; right side plot, smoothing enabled) When the “Magn+Phase” mode is selected, the phase smoothing is always applied to the whole frequency range, even if the threshold is not equal to 0dB.
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- Latency The latency parameter allows to control the I/O delay added by the FIR filtering. As a general rule, higher latencies should be preferred where the compensation accuracy is the main target, while lower latency values is often used in live sound system, where the I/O delay is a critical application parameter.
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As a general rule, it is suggested to try different values for the latency parameter and choose the lower one that still gives satisfying equalization results for the particular application, basing on plotted curves and listening tests. - Target Curve To set up a target magnitude curve, click the “EQ Target”...
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The EQ target can be bypassed (or enabled) by clicking the “EQ Bypass” checkbox at the top center of the screen. Clicking the “Flat EQ Target” button, the gain of each parametric filter is set to 0dB to flatten the target curve. Enabling the target curve smoothing allows to reduce unwanted ripples in the magnitude response when low frequency filters are set with high gain and/or high Q.
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To apply the target curve to the compensating FIR filter the “EQ Target” window should be closed and the “OK” button must be clicked in the confirmation popup. In “Phase Only” mode the target curve can't be set. Once the FIR filter is computed with the user defined parameters, the corresponding coefficients can be saved in a file (and recalled from outside the Wizard) or stored in the DSP to start listening tests.
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Verify Once the FIR filter is stored in the DSP, the user has the possibility to verify the equalization result by measuring the new loudspeaker IR with the FIR filtering enabled. To do this, click the “Verify” button at the botton right of the “Calculate FIR” page and the “Verify” page is shown. Clicking the “Verify”...
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NOTE B For importing External FIR Calculated coefficients into the MARANI LPP240A, the file format has to be the following: FIR FILE FORMAT - Standard floating point txt file - One coefficient for each row - Decimal separator: "." - Max coefficient value: 4.0 - Max filter length (i.e.
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