A tool to create audio processing pipelines for applications such as active crossovers or room correction. It is written in Rust to benefit from the safety and elegant handling of threading that this language provides. Supported platforms: Linux, macOS, Windows.
Audio data is captured from a capture device and sent to a playback device. Alsa, PulseAudio, Jack, Wasapi and CoreAudio are currently supported for both capture and playback.
The processing pipeline consists of any number of filters and mixers. Mixers are used to route audio between channels and to change the number of channels in the stream. Filters can be both IIR and FIR. IIR filters are implemented as biquads, while FIR use convolution via FFT/IFFT. A filter can be applied to any number of channels. All processing is done in chunks of a fixed number of samples. A small number of samples gives a small in-out latency while a larger number is required for long FIR filters. The full configuration is given in a YAML file.
henquist.github.io
github.com/HEnquist/camilladsp
Tag Archives: FIR
BruteFIR ~ Unix Multi-Channel Convolution
BruteFIR is a software convolution engine, a program for applying long FIR filters to multi-channel digital audio, either offline or in real-time. Its basic operation is specified through a configuration file, and filters, attenuation and delay can be changed in runtime through a simple command line interface. The FIR filter algorithm used is an optimized frequency domain algorithm, partly implemented in hand-coded assembler, thus throughput is extremely high. In real-time, a standard computer can typically run more than 10 channels with more than 60000 filter taps each.
Through its highly modular design, things like adaptive filtering, signal generators and sample I/O are easily added, extended and modified, without the need to alter the program itself.
torger.se/anders/brutefir
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BrutefirDRC ~ Add DRC/Loudness To LogitechMediaServer (LMS)
A plugin to use BruteFIR software convolution engine with Slim Devices SqueezeCenter clients for Digital Room Correction. Provides transparent automatic switching of filters for different sample rates. Filter creation can be done with DRC, Audiolense, Acourate or other DRC software.
An optional loudness correction using the digital volume control can be applied. The loudness features uses SoX loudness that is based on the ISO 226 curves.Features:
sourceforge.net/projects/brutefirdrc
- On the fly digital room correction for LogitechMediaServer (formerly SqueezeBoxServer)
- Automatic selection of filter for samplerate
- Loudness correction via digial volume control using ISO 226
- Resampling of filter for samplerate if none exists
- Gapless playback
- removal of silence in music
- Application of ReplayGain with and without loss of bit depth/resolution
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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)
www.holmacoustics.com/holmimpulse.php
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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
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HeSuVi ~ Headphone Surround Virtualizations
This tool imitates the 7.1 to binaural sound effect of many surround virtualizations by making use of Equalizer APO’s convolution filter. Available impulse response that were recorded with activated…
- Dolby Atmos Headphone
- CMSS-3D
- SBX Pro Studio Surround (also found in BlasterX Acoustic Engine & THX TruStudio Pro)
- Dolby Headphone
- Sennheiser GSX Binaural 7.1
- DTS Headphone:X
- Windows Sonic Headphone
- Dolby Home Theater v4 Headphone Surround Virtualizer
- Razer Surround
- Out Of Your Head
- Flux HEar V3
- OpenAL and DirectSound3D HRTFs
- Waves Nx
- and many more!
Features:
- many different headphone surround impulse responses
- powerful graphic equalizer
- equalizer presets for over 1000 popular headphones
- use multiple devices on one sound card
- extensive control over different volume levels
- apply and save the processing onto your audio files
- fully configurable crossfeed
- quickly save & load profiles, even through hotkeys
- supports command line parameters for all options
- rearrange the virtual speakers’ positions
- intelligent stereo upmix
- portable installation with easy one-click updater
sourceforge.net/projects/hesuvi
sourceforge.net/p/hesuvi/wiki/Help
HeSuVi Conversion Tool
This fairly simple tool converts all HeSuVi 14 channel presets into 7.0 formatted _L/_R stereo pairs, for use with the soon to be updated PulseAudio module-virtual-surround-sink, which I’ve updated with a faster FFT overlap-save convolver, eliminated the sample length limits for impulses, and added support for asymmetrical/dual impulse mode.
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DRC Designer ~ Optimize Audio Rooms
Digital Room Correction Designer was created to ease the process of creating and loading room correction filters for use with two channel stereo systems. DRC Designer includes Denis Sbragion’s DRC for creating room correction filters, John Pavel’s Convolver VST for playing music through the filters, and Simple Automated IR Measuring Tool by Denis Sbragion and Edward Wildgoose for creating the impulse response files needed by DRC.
www.alanjordan.org/DRCDesigner
Foobar2000 Setup
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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
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Equalizer APO ~ A Windows System Equalizer
Equalizer APO is a parametric / graphic equalizer for Windows. It is implemented as an Audio Processing Object (APO) for the system effect infrastructure introduced with Windows Vista.
Features:
- virtually unlimited number of filters
- works on any number of channels
- very low latency, which makes it suited for interactive applications
- low CPU usage
- modular graphical user interface
- VST plugin support
- integrates into Voicemeeter (www.voicemeeter.com)
Equalizer APO is best used in conjunction with Room EQ Wizard (www.roomeqwizard.com ), because it can read its filter text file format.
sourceforge.net/projects/equalizerapo
Discussion
Documentation
Downloads
User Interfaces:
Peace Equalizer
EQ APO GUI
mega-switcher
Resources:
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REW ~ Room EQ Wizard
REW is free room acoustics analysis software for measuring and analyzing room and loudspeaker responses. The audio analysis features of REW help you optimize the acoustics of your listening room, studio or home theater and find the best locations for your speakers, subwoofers and listening position.
www.roomeqwizard.com
www.hometheatershack.com/forums/rew-forum
sourceforge.net/projects/equalizerapo
Multi-Sub Optimizer
Convolver – A Convolution Plugin
The Audio File 