The HDPHX plugin can be used when mixing with headphones. It takes advantage of the Haas Effect to provide a delayed and attenuated cross-channel mix in each ear to present the sound that would be heard with a stereo speaker system with speakers placed at +/-30 degrees azimuth from front-center.
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Refined Audiometrics Laboratory has released CLAS (Compressive Loudness Audio Shaping), a tool that provides both progressive phase compensation for bass frequencies, and psychoacoustic bass and/or treble enhancement.
The effect is similar to BBE treatment, but without being “in your face!”. Psychoacoustic compression is similar to many classic compression curves, producing the most boost at low sound levels, and progressively tapering off to no-boost at extreme loud levels. This is more pleasing to listen to than typical linear compression, and accounts for the appeal of many classic analog compressors. The effect, in CLAS, is much like dynamic Loudness Contouring.
CLAS works best on Stereo channels, but it is also capable of running on Mono tracks. The left and right channel compressors are tied together in order to retain stereo field balance.
The progressive bass group delay acts almost independently of the bass and treble enhancement. Bass and treble enhancement are separately applied. The enhancement levels are nonlinear functions of the actual dB boost provided internally, so they are mapped to the faders so that you get the boost you ask for.
You can get a copy of CLAS by sending an email to firstname.lastname@example.org.
Visit Refined Audiometrics Laboratory for more information.
Refined Audiometrics Laboratory has released version 1.45 of their PLParEQ1 (a 1 band Phase Linear EQ Filter) and PLParEQ4 (a 4 Band Phase Lineaer EQ Filter) plug-ins.
Many more changes since v1.23, including:
- restored default behavior as bypass — now precisely a wire with delay.
- added color coding of graph to show Red for phase-warping, Blue for phase-linear.
- added an All-Pass filter as the phase-warping version of the bypass (—— mode). Phase-linear bypass is true neutral and bypass.
- added Cambridge Type 1, 2, 3 to Peaking EQ.
- added simple graph of filter shape to PLParEQ1.
- now FL 5 doesn’t hang anymore.
- stricter conformity with VST standard.
- added ramp down/ramp up when changing quality levels.
- added text-edit boxes on PLParEQ1 so you can directly enter Atten, Freq, Q, and Gain.
- added VST programs (presets) (32 per bank)
For more information and downloads, check the Refined Audiometrics Laboratory website.
PLParEQ1 and PLParEQ4 version 1.23 is now available at the Refined Audiometrics website.
Some fixes and improvements were made:
- Added Stereo/Middle/Side switch to PLParEQ1, and version identifications to both PLParEQ1 and PLParEQ4
- Found and fixed THE bug that causes all the system crashes (a simple typo in a name)
- added fine tuning slider to PLParEQ4
- better internal algorithm approximately triples runtime performance while maintaining previous quality
- fixed Mid/Side filtering
- added fine tuning slider to PLParEQ4
Check the Refined Audiometrics website for downloads and more information.
PLParEQ1 has been updated to version 1.15, which comprises a massive rewrite and bulletproofing against a myriad VST hosts. All runs just great now. Cubase SL 1.06 can run 2 simultaneous instances of PLParEQ1 in Quality level 5 without choking or any discernable dropouts. Can save and restore projects with Quality 5 settings in them.
- Stereo, Middle, Side filtering options, independently for each filter.
- Filter #4 was incorrectly wired by the GUI to respond to Type of Filter #1. Fixed.
- Fixed clicking on parameter changes.
- Added bulletproof thread-locking memory management.
- Fixed interactions between memory allocator and multiple host threads for improved stability (i.e., no more crashes)
- Changed default Q = 0.707 (critical damping), and default Frequency = 1 kHz.
- All starting parameters into reasonable values on initial load.
- Fixed a glaring error that caused a crash when shared input and output VST buffers were passed to a non-block processed filtering operation. Triangular dither to 24 bit output results from double precision internal computations.
- Made minimum block size = 512 samples at low SR, and 1024 samples at high SR. It was found that 256 samples was just too short, too often, to be a very decent filter. Quality 1 => 512 samples at low SR, Quality 5 => 8192 samples at low SR. You may have trouble running live with Quality 4 and 5. But these high quality settings are usable during track bouncing.
- Made the denorm dither significantly below the dither used for conversion from double to float. Noise floors are now down around -149 dB RMS Wide, and essentially a flat (white) spectrum at around -188 dB/root Hz.
- Added dither to all conversions from double back to VST float.
- Change Q slider to use logarithmic scaling. This gives better resolution for the lower Q values. Also increase Atten range to -40 dB
Download PLParEQ1 from the Refined Audiometrics website.
From the Refined Audiometrics Laboratory website:
You can grab a copy of our PC/VST plugin called PLParEQ1 which is a stereo 1-band ultra-high quality Parametric Equalizer filter.
This is the same technology that we utilize in all of our highest quality systems. This plugin comes with absolutely no restrictions on your use of it, no nag panels, no periodic noise insertions, no dongles — absolutely free.
We want you to experience a taste of our technology. And by giving you the very highest quality in a little tool that you can usefully employ, we think you will be convinced of the quality of our other systems too.
- Internal 88.2/96kHz processing at all system-wide sample-rates
- Runs at 44.1/48/88.2/96kHz external sample rates. (probably many others as well)
- 8-fold temporal overlap processing to minimize audible artifacts of time reversal processing
- Elegant windowing of the data on input and ouput ahead of the 8-way temporal overlap. The window used is w[n] = Sin(Pi/2*Sin(Pi/N*(n+1/2))^2).
- High quality Sinc(x) interpolation utilized in upsampling lower sample-rate data to 88.2/96kHz.
- 64-bit floating point computions throughout, with 80-bit intermediate values during filter computations.
- High performance, highly optimized, scientific-grade code compiled to native Pentium code. Portions may make use of MMX found on Pentium III and older CPU’s.
- Blocking of data is avoided when the system sample rate is already 88.2/96kHz and the filtering is purely causal (i.e., no phase-linear processing).
- Blocking artifacts (IMD) are held below -120 dBFS.
- Quality factor is related to the size of the blocks used during blocked data processing (i.e., during upsampled data or during acausal (phase-linear) processing). Quality factor of 1 corresponds to 512 sample blocks, and quality factor 5 corresponds to 8192 sample blocks, where a block is considered to be composed of data samples at the high internal sample rate of 88.2/96kHz. These blocking factors correspond to throughput delays from 5 to 90 ms, with higher quality processing requiring longer throughput delays.
You can grab your copy, along with supporting documentation at the Refined Audiometrics Laboratory website.