Transform your PC into an LF Generator
Generate Sinus, Pulse, Noise, Sweep and more…
Audiodope is an audio editor. You can load and listen to music files of various formats such as wave, MP3, WMA, etc. You can also edit part or the whole stream with functions like copy, cut, paste, insert and delete. You may modify any part of the stream by applying audio effects and synthesize sound files.
- Sound editing functions such as copy, cut, paste, delete, insert, trim.
- Apply processes like click removal, DC offset correction, compressor, echo, fading, inversion, noise reduction, normalization, pitch scale, pitch shift, reverberation, reversing, sound 3D, tempo, true bass, volume
- Filters like, moving average, high-pass, low-pass, band-pass.
- Special effects like flanger, phaser, chorus, gargle.
- Listen to the audio file with play, pause, stop commands
- Record from any source
- Various tools like, frequency analyzer, tone generator, noise generator, DTMF synthesis, text to speech.
- Process any channel separately or both
- Apply third party VST plug-ins
Free VST audio effect plug-in download for music production, broadcasting and general audio editing.
- Set lower and upper cut-off frequencies.
- Create filter sweeps by automating the cut-off frequencies.
- Add resonant boosts at the selected frequencies to accentuate sweeps.
ISOL8 divides the frequency range into 5 bands.
These 5 bands can be soloed or muted individually. This will help you to concentrate on certain frequency ranges during the mixing and mastering process.
On top you can use ISLO8 as a flexible multi-band splitter with following complex plugin chains. The split signal can finally be mixed together.
Isol8 is originally designed to be used on the master track, but it can also be used on individual audio track busses.
- 5 adjustable frequency bands
- solo/mute function for each band individually
- Linkwitz-Riley crossover filter design
- 24/48dB/Oct filter slope
- multiple filter channel modes (Stereo/Left/Right/Mid/Side)
- multiple monitor modes (Stereo/Left/Right/Mid/Side)
- in-place or centered monitoring
- swap left/right channel
- adjustable output level
- loudness dim function
- keyboard control
- multi channel split
- large and easy to use GUI
- free GUI scaling
- 64-bit internal processing
sTilt is a linear / natural phase and IIR filter which tilts the audio spectrum around a given center frequency
- distortion free processing engine
- adjustable slope from -6dB/Oct to +6dB/Oct
- adjustable center frequency
- 5 quality modes: low, eco, medium, high, max
- 3 filter modes: linear phase, natural phase and zero delay
- unit auto-gain
- switchable clip protection
- sample exact A/B comparison
- stereo / left / right / mid / side channel selection
- flexible channel monitoring
- output gain
- signal mixing
- free GUI scaling
Note: This plugin adds significant latency to the audio path, which is usually compensated by DAW (PDC).
The PC based Soundcard Oscilloscope receives its data from the Soundcard with 44.1kHz and 16 Bit resolution. The data source can be selected in the Windows mixer (Microphone, Line-In or Wave). The frequency range depends on the sound card, but 20-20000Hz should be possible with all modern cards. The low frequency end is limited by the AC coupling of the line-in signal. Be aware, that most microphone inputs are only mono.
The oscilloscope contains in addition a signal generator for 2 channels for sine, square, triangular, sawtooth wave forms and different noise spectra in the frequency range from 0 to 20kHz. The signal can be defined by a mathematical formula as well. The signals are available at the speaker output of the sound card. These can be fed back to the oscillocope in order to generate Lissajous figures in the x-y mode.
- Trigger modes: off, automatic, normal and single shot
- Triggerlevel can be set with the mouse
- The signals of the two channels can be added, subtracted and multiplied
- x-y mode
- Frequency analysis (Fourier spectrum)
- Waterfall diagram (frequency spectrum as function of time)
- Frequency filter: low-, high-, band-pass and band-stop
- Cursors to measure amplitude, time and frequency in the main window
- Audio Recorder to save data to a wave file
- For multi soundcard system, the used card can be selected in the settings tab
- 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)
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
LINGOT is a musical instrument tuner. It’s accurate, easy to use, and highly configurable. Originally conceived to tune electric guitars, it can now be used to tune other instruments.
It looks like an analogue tuner, with a gauge indicating the relative shift to a certain note, determined automatically as the closest note to the estimated frequency.
- It’s free software. LINGOT is distributed under the GPL license.
- It’s really quick and accurate, perfect for real-time microtonal tuning.
- Easy to use. Just plug in your instrument and run it.
- LINGOT is a universal tuner. It can tune many musical instruments, you only need to provide the temperaments. For that purpose, it supports the Scala project .scl format.
- Highly configurable via GUI. It’s possible to change any parameter while the program is running, without editing any file.