rt_pvc is a:
soundlab.cs.princeton.edu/software/rt_pvc
aur.archlinux.org/packages/rt_pvc
Correlometer is an analog-style stereo multi-band correlation meter.
Multi-band correlation meter is an advanced way to check for presence of out-of-phase elements in the mix. Broadband correlation metering reports overall phase issues and may misrepresent problems present in select spectral bands, while multi-band correlation meter easily highlights problems present in mid to high frequencies that are not easily heard by ear, but may still reduce clarity of the mix. Another application of multi-band correlation metering is phase- and time-aligning of channels and tracks, especially bass and bass-drum pairs, guitar mic and D.I. source pairs, two-microphone stereo recordings, etc.
Correlometer can display 4 to 64 individual spectral bands, with adjustable band quality factor that controls the degree of band’s selectivity. Averaging time of correlation estimator can be adjusted. Correlometer supports side-chain inputs for easy correlation estimation between separate audio tracks.plugins4free.com/plugin/2957/
- 4 to 64 band correlation meter.
- Adjustable band quality factor.
- Adjustable averaging time.
- Side-chain input.
- Resizable user interface.
- All sample rates support.
- Zero processing latency.
- User interface color schemes.
- Retina and High DPI support.
In 5 clicks, DRC for (almost) dummies
Simple, fast and effective software to measure and calculate the amplitude and phase correction and “align” speakers. This software also allows you to measure speakers to adjust them and/or to find the best placement in the room.
Note: DRC stands for Digital Room Correction
www.ohl.to/about-audio/audio-softwares/align
www.ohl.to/about-audio/audio-softwares/align2
Transform your PC into an LF Generator
Generate Sinus, Pulse, Noise, Sweep and more…
vincent.burel.free.fr/download/index
vb-audio.pagesperso-orange.fr/us/products/generator/generator
Collection of audio effects plugins implemented from the explanations in the book ~ “Audio Effects: Theory, Implementation and Application” by Joshua D. Reiss and Andrew P. McPherson.
- Chorus
- Compressor-Expander
- Delay
- Distortion
- Flanger
- Panning
- Parametric EQ
- Phaser
- Ping-Pong Delay
- Pitch Shift
- Ring Modulation
- Robotization-Whisperization
- Template Frequency Domain
- Template Time Domain
- Tremolo
- Vibrato
- Wah-Wah
github.com/juandagilc/Audio-Effects
- 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)
www.holmacoustics.com/holmimpulse.php
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
The Sonalksis FreeG is everything you need in a master fader. A simple, but absolutely indispensable plug-in, 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)
Audio Utility is a free, lightweight, universal general purpose all in one digital channel strip VST plugin offering precise controls over various aspects of an audio signal, from mid/side processing to soft clipping.
Features include:
- input routing with separate pans for the left and right channels, or alternatively a pre mono balance control to decide how the left and right channels are blended for conversion into mono
- gain, and individual left and right channel gains, mutes
- pan and phase inversion of either channel
- width control, along with separate mid and side gain sliders and mutes
- maximally flat minimal phase high and low pass 12db/octave filtering
- optional hard and soft clipping at 0db
- everything is fully automatable with sensible parameter ranges/scaling
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 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
Equalizer APO, REW and Rephase WOW! ~ diyAudio
en.wikipedia.org/wiki/Digital_room_correction
The Audio File 