HOLMImpulse ~ Frequency & Impulse-response Measurement


  • Measure Impulse-response
  • Measure Phase-response
  • Measure Frequency-response
  • Measure Harmonic Distortion (THD)
  • Make bandwidth limited measurements with no phase-errors
  • Compare measurements
  • Import Frequency-response & Impulse-response
  • Export Frequency-responses & Impulse-responses
  • Use open formats for saved files (zip, wav/flac, txt)

holmimpulse

www.holmacoustics.com/holmimpulse.php

Foobar2000 & MathAudio Room EQ


The frequency response of your room is very different in every point of the room. “Good” equalization in one point can worsen the sound in the neighboring point. Single-point equalization is not reliable and cannot be used in a professional room correction system. On the other hand, multipoint equalization is not simple; both the volume and the phase of the testing signal are very different in different points of the room. Simple averaging is not applicable. MathAudio Room EQ applies a state-of-the-art multipoint correction algorithm which ensures the best possible improvement in every point of the listening area.

  • Corrects deficiencies of room acoustics (multipoint compensation).
  • Corrects acoustic imperfections of speakers.
  • Avoids the pre-echo (pre-ringing) problem of conventional convolver-based room correction systems. The absence of pre-echo ensures the neutrality of the sound.
  • Works in rooms, halls and outdoor venues of any size.
  • Works with zero latency. Perfect for live performances and studio monitoring.
  • Doesn’t delay the audio track when playing video.
  • Corrects both amplitude and phase components of frequency response.
  • Quells resonance peaks of frequency response while leaving the deep notches.
  • Avoids the overcompensation which happens in conventional linearizing room correction systems.
  • Manually adjustable level of compensation allows one to reach the maximum transparency of the sound.
  • Supports full range of sample rates from 44,056 kHz up to 384 kHz. All sample rates are supported without resampling to avoid any possible loss in quality.
  • Includes a custom target curve feature.
  • Applies 64-bit signal path throughout.
  • Works with USB measurement microphones (e.g. MiniDSP UMIK-1 or Dayton Audio UMM-6) or standard measurement microphones (e.g. NADY CM100 or Dayton Audio EMM-6).
    Supports microphone calibration files.
  • Applies a patented method of frequency response correction.
  • Freeware Foobar2000 Component

mathaudio-room-eq

mathaudio.com/room-eq

FreeG ~ Fader Plug-in


The Sonalksis FreeG is everything you need in a master fader. A simple, but absolutely indispensable plug-in, the FreeG extends the functionality of a typical audio host by providing a large-format style fader, along with ultra accurate metering. FreeG helps to maximise workflow with extended, customizable metering, and multiple control features and settings.

When working with digital audio, a lack of fine metering and extended signal flow control in the host is a common issue. FreeG helps to improve workflow in the host by providing these features in the insert chain. By delivering enhanced control of amplitude, phase and pan (stereo version) in the insert chain as well as fine metering, with user configurable industry standard ballistic options, the freeG is the perfect master control.

Features:

  • Long-throw fader
  • Ultra accurate metering
  • Multiple metering ballistic types
  • Peak and RMS readouts
  • Pre/Post metering options
  • Selectable Pan-Law
  • ‘Fine’ mode for sensitive calibration of the master fader
  • Mono-compatible for use in tracking situations
  • VST and Audio Unit (32 and 64 bit)

FreeG

www.sonalksis.com/freeg

foo_dsp_subwoofer ~ Foobar2000 Linear Phase Filter


Extracts bass from the source signal using a linear phase low-pass filter and sends it to the LFE channel.
Features:

  • fast fir processing
  • adjustable filter shapes
  • delay control
  • mono, stereo, 3.0, 4.0, 5.0 input support

foobar2000.org/components/view/foo_dsp_subwoofer
HA Forum Topic

Linear phase is a property of a filter, where the phase response of the filter is a linear function of frequency. The result is that all frequency components of the input signal are shifted in time (usually delayed) by the same constant amount (the slope of the linear function), which is referred to as the group delay. And consequently, there is no phase distortion due to the time delay of frequencies relative to one another.

~ en.wikipedia.org/wiki/Linear_phase

rePhase ~ Loudspeaker Phase Linearization


rePhase is a FIR generation tool for building fully linear-phase active crossovers with arbitrary slopes.

It also let you manually compensate for the phase shifts of your loudspeakers and existing crossovers, be it active or passive.

Once generated, the FIR can be applied using any hardware (openDRC, miniSHARC, Najda, …) or software (foobar, convolver, JRiver, …), stereo or multi-way convolution engine.

sourceforge.net/projects/rephase

Additional resources:

Equalizer APO, REW and Rephase WOW! ~ diyAudio