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* FEATURE REQUESTS AND BUG REPORTS: Completed
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==== Mute Groups in MultiSamplers ====
  KozmykJ
    Is there a way to set Mute Groups as in Hi-Hat open vs closed ?

  LSP Project
    Standard MIDI mapping describes Closed Hi-Hat as MIDI note 42, Opened Hi-Hat as MIDI
    note 46 which are mapped independently.
    Do I right understand that under 'mute group' you mean a case when playback of note 42
    becomes muted when note 46 triggers and vice verse?

  KozmykJ
    Yes that's what I was wondering. Hi-Hat is just the most obvious case. Of course there are
    other uses. 

==== Bookmarks in File dialogs ====
  funkmuscle:
    Also the older versions, you could of bookmarked the dir where IR files are. I can't seem to figure out
    how to do it with the version.
  
  sadko4u:
    Currently because all dialogs were written from scratch, there are no bookmarks currently.

==== Analyzer ====

Sciurius:
  Using the LSP multichannel (x2, x4, …) Spectrumanalyzer I can hold (freeze the graph) of each
  of the channels. But how can I hold all channels simultanesously?
  
  https://discourse.ardour.org/t/spectrumanalyzer-hold-multiple-channels/88489

==== Equalizer ====

unfa:
  Add a knob that allows to simultaneously shift frequency for all active filters in the EQ.

==== Parameter entering ====
  In other subject, it would be great to be able to enter numerical values by double clicking
  on the DB value under the knobs.

==== Equalizer tuning ====
  Jan Lentfer
    it would be very cool if the parametric EQs could import settings exported from "Room EQ Wizard".
    That would make life so much easier :D
    https://www.roomeqwizard.com/
    Thanks and keep up the great work!
    
    The content of a text exported file looks like this:
    Filter Settings file
    Room EQ V5.18
    Dated: 10.01.2018 18:58:28
    Notes:
    Equaliser: Generic
    Average 1
    Filter  1: ON  PK       Fc    42.0 Hz  Gain  -6.5 dB  Q 13.20
    Filter  2: ON  PK       Fc    47.6 Hz  Gain  -3.3 dB  Q 16.77
    Filter  3: ON  PK       Fc    62.0 Hz  Gain   9.0 dB  Q  5.41
    Filter  4: ON  PK       Fc    67.1 Hz  Gain -12.3 dB  Q  5.02
    Filter  5: ON  PK       Fc    86.2 Hz  Gain -15.4 dB  Q  3.01
    Filter  6: ON  PK       Fc     122 Hz  Gain   6.3 dB  Q 15.03
    Filter  7: ON  PK       Fc     132 Hz  Gain  -9.1 dB  Q  6.77
    Filter  8: ON  PK       Fc     150 Hz  Gain  -8.5 dB  Q 10.00
    Filter  9: ON  PK       Fc     165 Hz  Gain   4.3 dB  Q 26.53
    Filter 10: ON  PK       Fc     190 Hz  Gain -19.7 dB  Q  5.03
    Filter 11: ON  PK       Fc     202 Hz  Gain   9.0 dB  Q  5.00
    Filter 12: ON  PK       Fc     247 Hz  Gain  -9.2 dB  Q  5.00
    Filter 13: ON  PK       Fc     258 Hz  Gain   9.0 dB  Q  5.00
    Filter 14: ON  PK       Fc    2073 Hz  Gain  -4.2 dB  Q  5.00
    Filter 15: ON  PK       Fc   10879 Hz  Gain  -3.8 dB  Q  5.00
    Filter 16: ON  None  
    Filter 17: ON  None  
    Filter 18: ON  None  
    Filter 19: ON  None  
    Filter 20: ON  None  

    I thought about an option to "just" read in such a file and set the Freq, Gain and Q values from this file. As far as I understood,
    these are just peak filters. 

==== Spectrum Analyzer ====
  Quirq:
    Back to the main topic, I'd dearly love to see a usable spectrum analyser that looked and behaved like Voxengo's SPAN:
    I use it for mix/mastering analysis; I was giving feedback to a friend for a CD release in December and looked again
    for comparable native analysers but went back to SPAN, even though it was a bit of a pain launching it separately and
    connecting everything up. (Now I've got Carla-rack that would be much, much less of an issue. But still it would be great
    to have something native.)
    I've looked at all the native analysers and none of them seem to come close. It's nice/clear to look at (important when
    you're using it for hours on end), the slope can be varied (useful for that gradual gentle slope down from low to high
    frequencies when things are balanced), but above all else, it's the fact that the display is smoothed and not at all spiky
    and is equally detailed across the spectrum (many analysers seem to be less detailed at the low end than they are for the
    highs). The image above looks like it's probably not on the more smoothed-out Mastering setting and is a bit spikier than
    the settings I use.

  ssj71:
    I think you are exactly right sadko, but Quirq is also right. Even though it's less accurate data,
    a mode like this can help the user to mentally process the data on screen better, even if it's less accurate.
    If you are using it for balance of frequencies, rather than trying to surgically remove or boost this or that
    freq, then I think a "frequency smoothed" mode is a good idea. Really it makes sense to have some independent
    control over smoothing WRT time AND/OR frequency. Especially with a few good presets ("Mastering," "high-accuracy," etc.)
    then it would be an even more useable and capable analyzer.

  Quirq:
    Thanks, you've very eloquently summarised what I was trying to say. The reason I got into using SPAN was because
    that's what the mastering engineer Ian Shepherd used in his Home Mastering EQ course (which I found incredibly useful).
    When I looked at all the native analysers to put things into practice, they all seem to be built more/just for precision,
    which is fine for most purposes but not as helpful for the broader-brush approach of mastering. 
  
    There are some analysers that are also scaled in a way that I don't find helpful – being marked 100, 200, 200, 400, 600, 800,
    1kHz, 2k etc with appropriate ticks in between is far more useful than 110, 220, 440, 880 etc. I understand why that's done,
    but it's harder to work out quickly where a feature of interest occurs in the spectrum unless it's centred around one of the
    marks. I think as well, having the area under the line makes things easier to read; most analysers do that, but not all.

  Amadeus Folego:
    You can check calf-analyzer as well (https://www.youtube.com/watch?v=TWfqcf-EyUE).

    Something like baudline (http://baudline.com) with LV2 would be amazing.

==== Link order of libraries ====

  The ordering of libraries on the command line follows the same convection as for object files:
  they are searched from left to right--a library containing the definition of a function should
  appear after any source files or object files which use it. This includes libraries specified
  with the short-cut -l option, as shown in the following command:

  $ gcc -Wall calc.c -lm -o calc   (correct order)

  With some compilers the opposite ordering (placing the -lm option before the file which uses it)
  would result in an error,

  $ cc -Wall -lm calc.c -o calc    (incorrect order)
  main.o: In function `main':
  main.o(.text+0xf): undefined reference to `sqrt'

  because there is no library or object file containing sqrt after 'calc.c'. The option -lm
  should appear after the file 'calc.c'.

  When several libraries are being used, the same convention should be followed for the libraries
  themselves. A library which calls an external function defined in another library should appear
  before the library containing the function.

  For example, a program 'data.c' using the GNU Linear Programming library 'libglpk.a', which in
  turn uses the math library 'libm.a', should be compiled as,

  $ gcc -Wall data.c -lglpk -lm

  since the object files in 'libglpk.a' use functions defined in 'libm.a'.

  As for object files, most current compilers will search all libraries, regardless of order.
  However, since not all compilers do this it is best to follow the convention of ordering libraries
  from left to right. 

==== Convolution Reverb ====

  As for the next one? I would like to see your take on a reverb. I've never really found one
  that I am satisfied with in linux. For convolution ardour cannot lot klangfalter lv2, the vst
  version loads but I'm not sure is completely stable. X42 plugins convolver is aimed at short
  impulses and has no controls, IR is unmaintained (I think) and, from experience, can have issues
  saving/loading parameters. For artificial reverb there are some good ones such as mverb but again
  I've never really felt very happy with them, none seem to offer the same kind of control that
  you can get with something like the Valhalla reverbs, epicVerb, or the new fabfilter reverb.
  
  An impulse purchase price of $30-$40 would be fine!
  
  The left input channel is convolved with the left and right impulse response file channels from IR1-A
  and the right input channel is convolved with the left and right impulse response file channel from IR1-B.
  The two output convolutions’ respective left and right components are then summed into a single stereo output.
  This configuration is necessary to take full advantage of true stereo impulse responses. True stereo impulse
  responses are required to be provided as two separate stereo files and loaded into IR1-A and IR1-B (or IR2-A and IR2-B).
  This configuration is typically found in high-end algorithmic reverbs. 

==== Support of ARM Architecture ====
  Chase: 
    was just wondering if LSP-PLUGINS, specifically your trigger-midi-mono, will have a version
    in the near future that will work on the raspberry pi arm7/arm8 ? as a jack standalone and lv2...

==== Compensation delay ====
  unfa:
    I've recently used a-Delay to create a comb-filter effect in my session. 
    It's really great for that kind of stuff, but one thing that could be improved is
    the resolution of the Time parameter.

	When making an automation ramp going from 1 ms to 5 ms over 8 bars you an clearly
	hear the individual steps of the parameter being taken. I wish this parameter could
	have higher resolution so I could make automation of that with not stairsteps, even
	with very slow automation.

==== Slapback delay ====
  Marcel
	I was playing with the Slapback Delay to get some "Giorgio Moroder-like" synth disco bass.
	It is working great, but I have calculate all the time the delays I need for fitting the beat.
	And when I change the bpm I have to set up a new Slapback Delay with other settings.

	Would it be possible to add a delay mode  by which you can set delay based on synced beats instead
	of time and distance? Would be nice to have a " synced beat" with the rotary knob you could increase
	in steps of 1/16 up to e.g 64/16.

==== Multiband compressor ====
  Marcel:
    I have one question though. I don't see a multi-band compressor.
    It that something still in the pipeline? Would be great to have one.
    Keep on the good work, and of course when I like the plugins I will make a donation. 
    
  sadko4u:
    I also thought that non-crossover multiband compressor is lacking linux audio. Crossovers do phase
    distortion at split points. Instead of this multiband compressor could be implemented as N-band
    equalizer with dynamically changing gain for each band where the gain is controlled as compression gain.
  
  sysrqer:
    That sounds excellent sadko4u, that's actually what I had in mind, I don't think there is anything like that for us.
    
  sysrqer:
	Being able to move the bars on the graph with the mouse would be fantastic.

==== Sampler ====
  robb:
    with your LSP Sampler, is it possible to have it obey note-off? Ie. If i release the key, the
    sample stops playing? Currently, it seems that a sample plays through to the end each time, no
    matter how long the keypress is for... 
    
  Julian:
    Is it possible to configure samples so that it stops playing when receiving a midi note off? So
    far I haven't been able to.

==== Equalizer ====
  Norlick:
    Any way to make the EQ frequency display focus on a smaller range? Most of my EQing is
    within 5dB of change so on the current spectrum it looks like very miniscule changes.

==== Convolvers ====
  Anton Stankevich
    I often use a Calf Eq 5 after reberb effect. It will be nice to have some parametric equalization for reverb wet signal.
    This feature can help to give more presence for vocal with reverb for example.

==== UI bugs ====
  Tom Moody:
    2. Oscillator, VST:
      The controls underneath the dropdown menu in "Oscillator Function" are missing. You can switch between waveforms in the menu.
      The  Oversampling function does not appear under "General."
      Everthing else in the "General" section works well. So does the Waveform graph.
      Screenshot: https://www.tommoody.us/images/dec17/LSP_VST_oscillator.png
    3. Oscillator, LV2:
      The LV2 has similar issues to the VST but they can be corrected (somewhat) by closing and reopening the plugin interface.
      The main ones are: The controls underneath the dropdown menu in "Oscillator Function" freeze, and don't change when you
      select a new waveform. Sometimes "Oversampling" appears, sometimes it's blank. 

  JohnR
    I just tried 1.1.0 JACK version of the plugins. In the trigger plugins the Instrument menu
    item doesn't open a file chooser, so samples can't be loaded.
    In the IR plugins the file chooser works fine and I can load IR files without a problem.
    I use cwm window manager on Debian 9 amd64. I don't have any LV2 hosts on my system (I'm
    using non-DAW which is LADSPA only), hence the need for the standalone JACK versions.

==== Delays ====
  jonetsu:
    The benefit with using delays is that there are no tails to manage. Also, those delays can be colorized by
    EQ (dull is farther, bright is closer) and adjusted accordingly with compression. The compression has to work
    in the same way as the EQ, not against.

  sadko4u:
	Your explanations helped me to understand what another plugin we are currently missing in linux audio. I think that
	plugin that allows to simulate set of slap back delays combined with equalization will be very helpful while doing
	spacial processing. What do you think about it?

==== Lookahead option ====
  Some dynamics plugins feature lookahead, some don't. As it is a very useful feature
  (eg. on a sidechain gate, having the gate start closing 1ms or so before the trigger makes it even more
  effective) and the code is there already, I feel it would be a welcome addition.

  Seconding this request, lookahead would be incredible and make the dynamics plugins very hard to beat.
  
==== Limiter issue ====
  Tom Moody:
    On the limiter issue:
    1. I am using Ubuntu Studio 16.04, Ardour 5.6. No Calf plugins (one reason I tried out LSP is I kept having
       Ardour crashes and am trying to have a "life without Calf")
    2. The limiter doesn't crash Ardour. I have had the limiter GUI freeze but I can't duplicate this at the moment.
       Possibly this was an issue with Ardour 5.5 -- I'll let you know if it happens again.
    3. If the limiter GUI is open, when I open the export dialog, the limiter GUI stays on top of the export window.
       To view the export window, I have to move the limiter GUI to one side. When I export, the export window shows
       no moving progress bar, it just sits there until the file is exported, then the "analysis window" pops up on
       conclusion of the export. There is no movement in the limiter GUI window during the export, either, until the
       very conclusion of the export, then a little squiggle of audio appears in the graph, just at the fade.
    It's a minor issue. One thing I wanted to mention, on a positive note. I've been using the PSP Vintage Warmer plugin
    on Windows as a maximizer for a final, admittedly non-expert master. I am gradually moving everything to Linux, and your
    limiter in "boost" mode is an excellent replacement for the PSP. Much appreciated.
