Algorithmix Linear Phase Parametric EQ Orange Every mastering engineer knows that equalization of final complex mixes or orchestral recordings with stereo microphones is very critical and limited.
Any correction of a single instrument or vocal part influences other instruments and can negatively change their sound characteristics. This is because all classic equalizers change the phase of the fundamentals and their harmonics and the phase shift is frequency dependent.
The Algorithmix Linear Phase Parametric EQ Orange does not shift the phase, it only boosts or cuts the amplitude in a given frequency range. Consequently you can apply much more boost without changing the sound character of your recording. Due to our proprietary low-noise filter design in the time domain, the distortions (THD+N) are extremely low ensuring crystal clear, neutral, and uncolored sound.
The Algorithmix Linear Phase Parametric EQ Orange DirectX/VST PlugIn is unique in the world of audio components. Almost all parametric equalizers being used are implemented with filters that accomplish phase shift, e.g. the original signal is remixed with its phase-shifted version. Because the amount of the phase shift is frequency dependent some cancellation or amplification at certain desired frequencies takes place. It works, but with a major disadvantage - the different signal components are spread all over time, so that the time relationship between harmonics in the processed signal is heavily affected. The result is that a nice sharp bass drum becomes slurred and muddy and vocal tracks become strident or brittle. Algorithmix is proud to offer you a true remedy against difficult equalizing tasks, the Algorithmix Linear Phase Parametric EQ Orange PlugIn, a linear-phase equalizer which can be handled exactly like its classic predecessors, but cannot be "heard". The Algorithmix Linear Phase Parametric EQ Orange Orange successfully copes with sonic problems that cannot be solved with any classical equalizer without side effects. You can boost lower frequency regions by even 10dB without mud or slush and with no loss of transients at all. You can remove the sibilance from a vocal with a sharp notch without affecting the whole mix like occurs when using analog or IIR-based digital EQ. Algorithmix Linear Phase Parametric EQ Orange better preserve the time relationships of harmonics in the original waveform. This translates to smoother top end, sweeter and bigger midrange, and a clearer more distinct bottom end as compared with phase-shifted EQ. The resultant sound seems less processed, more like it occurred naturally in the air in front of the microphone. The mixes become clear and transparent, the instruments more defined and realistic. The Algorithmix Linear Phase Parametric EQ Orange works in the time domain. It sounds more analytical than its brother, the Algorithmix Linear Phase Parametric EQ Red, and therefore is specially recommended for difficult mastering and re-mastering tasks on dense mixes. It performs flawlessly with up to 384 kHz sampling frequency and allows extended frequency setup up to 80 kHz, thus being ideally applicable for high-resolution DSD post-processing, including its ultrasonic region.
Features -flawless operation with up to 384 kHz sampling frequency, thus perfectly suitable for high-resolution DSD post- production -equal delay for every frequency, independent of filter settings, perfectly preserving the time relationship of harmonics in the original signal -no phase distortions and therefore no signal dispersion and transient smearing -no unwanted sound coloration; more clarity, transparency, and definition -extended center frequency setup up to 80 kHz covering ultrasonic range in DSD applications -10 bands with five freely assignable parametric filter types: bell, low-shelf, high-shelf, low-cut, and high-cut adjustable Q-factor for shelving and cut filters -extremely low-noise and low-distortion linear-phase filter algorithms -possibility to overlap several low-cut and high-cut filters to get desired brick wall characteristics -very low-noise and low-distortion oversampling technique for getting analog-like filter characteristics when working with low sampling frequencies: 44.1 and 48 kHz; clearly perceptible when equalizing higher frequencies -extremely low noise and low nonlinear harmonic distortion -resizable frequency-response display for high-precision adjustments -parameter editing via numerical fields or directly on the graphical display -automatic latency compensation (in Sequoia and Samplitude only) -complete setup exchange among several simultaneously opened PlugIns -all internal calculations with double floating-point accuracy (80 bits)
System Requirements Minimum configuration (one real-time DirectX/VST PlugIn loaded) Windows 9x/Me/NT4/2000/XP Home or Professional Pentium II 500 MHz 64 MB 1024x768 pixels resolution with true color (24bit or 32bit) any compatible DirectX/VST Host Editor USB port for hardware dongle Sequoia/Samplitude 7.x, 8 allows fully automated PlugIn latency compensation Recommended configuration (more than one real-time DirectX/VST PlugIn loaded) Windows 2000/XP Home or Professional Pentium IV or AMD Athlon XP with at least 1 GHz (resolution low), 3,6 GHz (resolution Xtra) 512 MB 1280x1024 pixels resolution with true color (32bit is usually fastest) any compatible DirectX/VST Host Editor USB port for hardware dongle Sequoia/Samplitude 7.x, 8 allows fully automated PlugIn latency compensation
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