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FreqTweak - Realtime audio frequency spectral manipulation
Written by Jesse Chappell <>

Please see the file COPYING for license details.
For building and installation instructions please see the INSTALL file.

**** Description ****

FreqTweak is a tool for realtime audio spectral manipulation and
display. It provides several algorithms for processing audio data in
the frequency domain and a highly interactive GUI to manipulate the
associated filters for each. It also provides high-resolution spectral
displays in the form of scrolling- raster spectragrams and energy vs
frequency plots displaying both pre- and post-processed spectra.

It is an extremely addictive audio toy, but I hope it has value for
serious audio work too (sound design, etc). The spectrum analysis is
pretty useful in its own right.

**** Features ****

Freqtweak supports manipulating the spectral filters at several frequency
resolutions (64,128,256,512,1024,2048, or 4096 bands) depending on your needs/
resources. Overlap and windowing are also selectable.

The GUI filter graph manipulators (and analysis plots) have selectable
frequency scale types: 1x and 2x linear, and two log scales to help with
modulating the musical frequencies. Filters can be linked across multiple

The current processing filters are described below in the order audio is
processed in the chain. Any or all of the filters can be bypassed. The state
of all filters can be stored or loaded as presets.

    ** Spectral Analysis -- Multicolor scrolling-raster spectragram, or energy
      vs. freq line or bar plots... one shows pre-processed, another shows

    ** EQ Cut/Boost -- Your basic multi-band frequency
      attenuation. But you get an unhealthy number of bands... Note
      that this EQ is not intended for mastering purposes, it allows
      for (and doesn't protect against) highly irregular
      filtering. Two versions, one does only frequency gain cut, the
      other boost.

    ** Pitch Scaling -- This is an interesting application of Sprengler's pitch
      scaling algorithm (used in Steve Harris' LADSPA plugin). If you keep all
      the bins at the same scale, it is equivalent to Steve's plugin, but when
      you start applying different scales per frequency bin, things quickly
      get weird.  For highest quality results (at the expense of transients) use
      larger FFT (>= 1024 bins).

    ** Gate -- This is a double filter where a given frequency band is allowed
      to pass through (unaltered) if the power on that band is between two dB
      thresholds... otherwise its gain is clamped to 0.

    ** Delay -- This lets you delay the audio on a per frequency-bin basis
      yielding some pretty wild effects (or subtle, if you are careful). A
      feedback filter controls the feedback of the delay per bin (be careful
      with this one). This is basically what Native Instrument's Spektral-
      Delay accomplishes. Granted, I don't have all the automated filter
      modulations (yet ;). See their website for audio examples of what is
      possible with this cool effect.

    ** Limit --	This is very harsh brick wall limiter on a per-bin
		basis.  It is not very pleasant, but can be interesting.

    ** Compressor
          This is a massively multiband compressor.  It will not
          behave quite like a normal time-domain compressor because of
          the inherent block processing of the FFT.  Each frequency
          bin has its own compressor complete with Threshold, Ratio,
          Attack/Release time, and makeup gain.  Again, this is *not*
          suitable for mastering applications!

    ** Warp  --	  This one is a little different, both axes represent
		  frequency, and the identity matrix is unaltered
		  audio.  Changing the value (height) of a bin,
		  reallocates the energy at that frequency to the new
		  frequency bin represented by the height of the bar.
		  For instance, if all bins are the same height, all
		  the frequency energy is added to a single bin.  This
		  is a sensitive filter, the Log frequency scale is helpful here
		  (it affects both axes).

 Modulators to an filter can be attached from the Modulations Window
 (Control->Modulators...   Ctrl-M).  Add a modulator by clicking
 on the Add Modulator... button and select from the choices.  To
 attach a modulator to a filter, click on the Attach... button on the
 modulator panel and pick a filter.  You can modulate many filters
 simultaneously.   The text entry fields can be used to exactly set
 the slider values, by pressing enter/return after entering the number.
 The following modulators are
 currently implemented, with more to come soon.

   ** Rotate  --  this will continually shift a filter horizontally at a constant
            definable Rate, wrapping when it reaches the edge.  The edges
            are definable with the Min and Max Freq controls.

   ** Rotate LFO -- the same as the above, except the shifting rate oscillates via LFO
            with its own Rate and Depth controls.  Currently there are sine, triangle,
	    and square waveform shapes.  The frequency range that the modulator affects
            is definable with the Min and Max Freq controls.

   ** Value LFO -- shifts the values up and down with an LFO.  The depth control
            here is percentage
            of total value range.  The frequency range that the modulator affects
            is definable with the Min and Max Freq controls.

   ** Randomize --  randomizes the bin values between the given value bounds
            (as percentages of total range).  Again, the frequency range
            that the modulator affects is definable with the Min and Max Freq controls.

**** Requirements ****

    * JACK [] -- providing realtime low-latency audio
      interconnection and delivery. JACK requires the ALSA Linux sound
      drivers so you'll need those too.

    * FFTW [] -- for speedy FFT processing
	(compiled as single-precision) Supports v2 or v3.

    * wxWindows (wxGTK) [] -- the GUI toolkit I've
      chosen to use. It should work with the most 2.2.x, 2.3.x, and 2.4.x.

    * libsigc++ 1.2  -- this library is usually already on recent systems
          but can be found at [ ]

**** Misc Usage Tips ****

    * Left button click/drag to draw filters. If Control is down, the y-axis
      is fixed at the last cursor location (to draw nice horizontal lines).
      If Control *and* Alt are down you can draw nice arbitrary lines.

    * Right button drag to move filters around in space. The filters wrap
      around the left/right edges unless you hold down Control. Dragging with
      both left and right buttons down moves both primary and alternate
      together (on Gate).

    * Holding Shift modifies the alternate filter (on double filter graphs
      like Gate) for the previous operations.

    * Middle-button pops up frequency axis menu.

    * Ctrl-Alt right-click resets a filter to default values.

    * Shift-Ctrl-Alt Left-Drag zooms in on the y axis.
      Look at the status bar to see the values for the cursor
      itself and the values of the filter at the cursor's
      frequency.  Shift-Ctrl-Alt Right click-release resets
      the Y-zoom to full.  	      

    * The B and BA buttons mean Bypass and Bypass All respectively.

    * The L and LA buttons mean Link and Link All respectively.

****  Here is an example of using freqtweak with an alsaplayer feeding it
      and output going to speakers (alsa_pcm:out_?) without using a JACK patchbay:
   Start freqtweak first with this command line:
      freqtweak -i none,none -n ft  &

   [ you will see some jack errors, ignore them.. they are intentional ]

   Start alsaplayer like so:

     alsaplayer -o jack -d ft:in_1,ft:in_2  &

**** TODO ****

    * Fix known bugs
    * Support reordering of processing modules
    * Automated filter modulation (via plugins?)
    * Plot performance optimizations (OpenGL?)
    * Whatever else the users want :)

  Jesse Chappell <>
  Last modified: Sat Oct 12 13:02:59 EDT 2002

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