PLParEQ
PLParEQ - Our 10-Band Phase Linear Parametric Equalizer
PLParEQ is a 10-band Phase-Linear Parametric Equalizer of the highest quality. Each filter may assume any of many different filter characteristics, and operate in either traditional phase-warping mode, or our phase-linear mode. It uses the same internal DSP core as all of our other high-end products. Audio streams are treated in either stereo or mono. Individual filters can be applied to either or both stereo channels, middle only, or side only.
Phase Linear Operation is achieved by processing your sound in both the forward-time and reverse-time directions through classic filters - all in realtime. This completely removes the phase warping caused by IIR filtering, and applies their roll-off twice. So each filter type becomes two: one for classic IIR filtering, and the other for Phase-Linear operation.
Below is a screenshot from a mastering session by one of our Norwegian Mastering Engineer clients.

(click image for full-size preview)
This PC VST plugin can operate in traditional 32-bit mode (24-bit audio) or as a 64-bit plugin for Cakewalk's new Sonar 5. All internal processing is carried out in double-precision 64-bit floating point, regardless of external host mode.
For audio streams at sample rates below 80 kHz the DSP engine internally upsamples with high-quality Sinc interpolation, applies its filtering, and then downsamples back to your system sample rate.
You can run all of the filters at native sample rates higher than 80 kHz, and forego the internal upsampling conversions. Upper limits on the sample rate (> 96 kHz) are dicted primarily by your computer's capacity, and your need for high quality at the very lowest frequencies (below 100 Hz).
Individual filters can be operated as either traditional phase-warping, or phase-linear. Our algorithms employ blocked processing for phase-linear operation, and produce phase linearity by sending the signal through each filter twice - once in the forward time direction, and then again in the time reversed direction, thereby unwinding the phase back to zero.
The blocks are reassembled using very high-quality windowing and 8-fold temporal overlap. IMD resulting from phase-linear operation has been measured at below -150 dB from peak signal levels. Corner frequencies for filters can be adjusted from 10 Hz to 30 kHz at all sample rates.
Computation proceeds in double-precision floating point throughout the entire DSP core. At the last stage of conversion back to 24-bit audio, we dither with a carefully crafted TPDF dither from our internal 64-bit samples to the 24-bit mantissas utilized thorughout by VST hosts. Our noise floor is typically measured at around -180 dB/Root(Hz). TPDF dither is the best kind to use to avoid noise that correlates with the audio signal. The noise levels may become somewhat higher than produced by other forms of dither, but it remains white.
Our tests indicate that PLParEQ requires about 5-6% of our computer speed capacity, with each additional filter enabled requiring an additional 0.5-1% at the highest quality levels. These tests were performed on a 3 GHz Pentium-4 computer with HyperThreading enabled. Different VST hosts will show varying requirements. Our tests were performed with Mackie's Tracktion 2 as the VST host. Performance on a more modern Pentium IV are likely to show improvements upon these results.
Get the Refined Audiometrics Advantage - The Smoothness of Analog in a Superb Digital Implementation
- Highest Quality Parametric Equalization
- Selectable Phase-Linear or Traditional Filtering
- 10 Bands + 22 Classic Filter Types
- Selectable Stereo, Left, Right, Left+Right-, Middle, Side, or Middle+Side- Filtering
- Output Stereo, Left, Right, Middle, or Side
- Frequencies from 10 Hz to 30 kHz
- Gains from -20 dB to +20 dB
- Q's from 0.20 to 20
- Choice of K-20, K-18, K-17, K-14, or K-12 RMS-Wide Metering
- 7 Different Quality Levels
- Group Enable/Disable for Instant A/B
- Filter Table displays settings in effect for all filters
- Live Graph permits mouse dragging of Frequency, Gain, and Q
- Global Attenuation/Gain control for -20 dB to +20 dB
- Low Rate Audio is Upsampled by 2x, Filtered, and then Downsampled again
- Uses the Highest Quality Data Windowing
- Sinc Interpolation on Upsampling
- Performs 8-fold Temporal Overlap Processing
- Sample Rates from 44.1 kHz to 192 kHz
- Internal 64-bit Processing Throughout
- Samples are Dithered back to 32-bits on Output
- Uses 2-bit TPDF Dither
- All Parameters are Fully Automatable
- Measured IMD and Noise Floor Below -150 dBFS
- 64-Bit Capable for Sonar 5
- PC/Windows VST Plugin
Available Filter Types
- 2-Pole Resonant LowPass, HighPass, BandPass, Band Reject, and AllPass
- 1-Pole LowPass and HighPass
- 4-Pole Resonant LowPass and HighPass
- Classic Low and High Shelving
- Type 1, 2, and 3 Oxford-style Peaking Filters
- 3 dB/octave LowPass and HighPass
- 1/F, A-Weighted, B-Weighted, and C-Weighted Filters
- 6-Pole Notch Filter
We are now up to version 2.24 with product improvements. Purchasers of PLParEQ will receive a free update to version 2.24.
The DLL contained in the Zip file permits anyone to install and use the full capabilities of our 10-band PLParEQ for 30 days. The sign-on dialog gives you the option to purchase a license and register the plugin with us. Registered licenses are entitled to install on up to two (2) computers of their choice. Beyond that, additional licenses are required unless special arrangements have been made with us.
Licenses may be purchased for the sum of $1,000.00 USD, and entitle you to two installations and lifetime support with free upgrades to the code and documentation. We will be continuing to add additional filtering capabilities to this plugin, making this one of your best values in professional audio mixing and mastering tools.

PLParEQ 2.21 -- Try for 30 days before you buy
Version 2.24 adds K-24 Metering. Why K-24? Because a worst-case scenario for output D/A conversion occurs when a long series of sample values are at full negative amplitude, followed by a 2-sample burst at full positive amplitude, followed by another long series of samples at full negative amplitude. When this happens, the output D/A converter would be obligated to produce a peak voltage midway between the two positive samples that is +3.8 dB above full scale. Call it 4 dB.
Hence, in the interest of ensuring the highest possible output quality, one ought to leave a 4 dB margin above the highest peaks in the music. Since the highest peaks occur at 20 dB, and lower, from mean RMS levels, we need an additional 20 dB of headrooom. So, for the very highest quality mixing, one ought to be using the K-24 metering system, which places 0 dBVU at -24 dBFS.
Version 2.24 also fixes some problems that occur on some hosts, notably the Muse Receptor with Version 6 software. And these fixes also remove the dropouts that were sometimes occuring during rapid mouse changing of filter parameters in the graph pane.
