From e40fc849149dd97c248866a4a1d026dda5e57b62 Mon Sep 17 00:00:00 2001
From: Robert Jonsson <spamatica@gmail.com>
Date: Mon, 7 Mar 2011 19:01:11 +0000
Subject: clean3
---
.../plugins/freeverb/CMakeLists.txt | 33 +++
.../muse_qt4_evolution/plugins/freeverb/allpass.h | 47 +++++
attic/muse_qt4_evolution/plugins/freeverb/comb.h | 63 ++++++
.../plugins/freeverb/denormals.h | 38 ++++
.../plugins/freeverb/freeverb.cpp | 166 +++++++++++++++
.../muse_qt4_evolution/plugins/freeverb/readme.txt | 147 +++++++++++++
.../plugins/freeverb/revmodel.cpp | 232 +++++++++++++++++++++
.../muse_qt4_evolution/plugins/freeverb/revmodel.h | 79 +++++++
attic/muse_qt4_evolution/plugins/freeverb/tuning.h | 60 ++++++
9 files changed, 865 insertions(+)
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/CMakeLists.txt
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/allpass.h
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/comb.h
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/denormals.h
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/freeverb.cpp
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/readme.txt
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/revmodel.cpp
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/revmodel.h
create mode 100644 attic/muse_qt4_evolution/plugins/freeverb/tuning.h
(limited to 'attic/muse_qt4_evolution/plugins/freeverb')
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/CMakeLists.txt b/attic/muse_qt4_evolution/plugins/freeverb/CMakeLists.txt
new file mode 100644
index 00000000..0a86454f
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/CMakeLists.txt
@@ -0,0 +1,33 @@
+#=============================================================================
+# MusE
+# Linux Music Editor
+# $Id:$
+#
+# Copyright (C) 2002-2006 by Werner Schweer and others
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License version 2.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+#=============================================================================
+
+add_library ( freeverb SHARED
+ freeverb.cpp revmodel.cpp
+ )
+
+# tell cmake to name the target freeverb.so instead of
+# libfreeverb.so
+#
+set_target_properties (freeverb
+ PROPERTIES PREFIX ""
+ COMPILE_FLAGS "-O2"
+ )
+
+install_targets ( /lib/${MusE_INSTALL_NAME}/plugins freeverb)
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/allpass.h b/attic/muse_qt4_evolution/plugins/freeverb/allpass.h
new file mode 100644
index 00000000..2f27691e
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/allpass.h
@@ -0,0 +1,47 @@
+// Allpass filter declaration
+//
+// Written by Jezar at Dreampoint, June 2000
+// http://www.dreampoint.co.uk
+// This code is public domain
+
+#ifndef _allpass_
+#define _allpass_
+#include "denormals.h"
+
+//---------------------------------------------------------
+// allpass
+//---------------------------------------------------------
+
+class allpass
+ {
+ float feedback;
+ float *buffer;
+ int bufsize;
+ int bufidx;
+
+ public:
+ allpass() { bufidx = 0; }
+ void setbuffer(float *buf, int size) {
+ buffer = buf;
+ bufsize = size;
+ }
+ float process(float input) {
+ float bufout = buffer[bufidx];
+ undenormalise(bufout);
+ float output = -input + bufout;
+ buffer[bufidx] = input + (bufout*feedback);
+ if (++bufidx >= bufsize)
+ bufidx = 0;
+// bufidx = ++bufidx % bufsize;
+ return output;
+ }
+ void mute() {
+ for (int i=0; i<bufsize; i++)
+ buffer[i]=0;
+ }
+ void setfeedback(float val) { feedback = val; }
+ float getfeedback() { return feedback; }
+ };
+
+
+#endif//_allpass
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/comb.h b/attic/muse_qt4_evolution/plugins/freeverb/comb.h
new file mode 100644
index 00000000..22f5591b
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/comb.h
@@ -0,0 +1,63 @@
+// Comb filter class declaration
+//
+// Written by Jezar at Dreampoint, June 2000
+// http://www.dreampoint.co.uk
+// This code is public domain
+
+#ifndef _comb_
+#define _comb_
+
+#include "denormals.h"
+
+
+//---------------------------------------------------------
+// comb
+//---------------------------------------------------------
+
+class comb
+ {
+ float feedback;
+ float filterstore;
+ float damp1;
+ float damp2;
+ float *buffer;
+ int bufsize;
+ int bufidx;
+
+public:
+ comb() {
+ filterstore = 0;
+ bufidx = 0;
+ }
+ void setbuffer(float *buf, int size) {
+ buffer = buf;
+ bufsize = size;
+ }
+ float process(float input) {
+ float output = buffer[bufidx];
+ undenormalise(output);
+ filterstore = (output*damp2) + (filterstore*damp1);
+ undenormalise(filterstore);
+ buffer[bufidx] = input + (filterstore*feedback);
+ if (++bufidx >= bufsize)
+ bufidx = 0;
+// bufidx = ++bufidx % bufsize;
+ return output;
+ }
+ void mute() {
+ for (int i=0; i<bufsize; i++)
+ buffer[i]=0;
+ }
+ void setdamp(float val) {
+ damp1 = val;
+ damp2 = 1-val;
+ }
+ float getdamp() { return damp1; }
+ void setfeedback(float val) { feedback = val; }
+ float getfeedback() { return feedback; }
+ };
+
+
+#endif //_comb_
+
+//ends
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/denormals.h b/attic/muse_qt4_evolution/plugins/freeverb/denormals.h
new file mode 100644
index 00000000..37d3ed27
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/denormals.h
@@ -0,0 +1,38 @@
+// Macro for killing denormalled numbers
+//
+// Written by Jezar at Dreampoint, June 2000
+// http://www.dreampoint.co.uk
+// Based on IS_DENORMAL macro by Jon Watte
+// This code is public domain
+
+#ifndef _denormals_
+#define _denormals_
+
+// this does not work with at least gcc3.3 and -O2:
+// #define undenormalise(sample) if(((*(unsigned int*)&sample)&0x7f800000)==0) sample=0.0f
+//
+// from Laurent de Soras Paper: Denormal numbers in floating point
+// signal processing applications
+// (ws)
+
+#if 0
+#define undenormalise(sample) \
+ { \
+ float anti_denormal = 1e-18; \
+ sample += anti_denormal; \
+ sample -= anti_denormal; \
+ }
+#endif
+
+// from beast-0.7.2 (Tim Janik/Stefan Westerfeld):
+
+#define undenormalise(sample) \
+ do { \
+ volatile float __forced_float = 1e-29 + sample; \
+ sample = __forced_float - 1e-29; \
+ } while (0)
+
+#endif//_denormals_
+
+//ends
+
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/freeverb.cpp b/attic/muse_qt4_evolution/plugins/freeverb/freeverb.cpp
new file mode 100644
index 00000000..0385e887
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/freeverb.cpp
@@ -0,0 +1,166 @@
+//=========================================================
+// MusE
+// Linux Music Editor
+// $Id: freeverb.cpp,v 1.1.1.1 2003/10/27 18:57:03 wschweer Exp $
+// (C) Copyright 2000 Werner Schweer (ws@seh.de)
+//=========================================================
+
+#include "revmodel.h"
+
+//---------------------------------------------------------
+// instantiateFreeverb
+// Construct a new plugin instance.
+//---------------------------------------------------------
+
+LADSPA_Handle instantiate(const LADSPA_Descriptor* /*Descriptor*/,
+ unsigned long /* samplerate*/)
+ {
+ return new Revmodel;
+ }
+
+//---------------------------------------------------------
+// connectPortToFreeverb
+// Connect a port to a data location.
+//---------------------------------------------------------
+
+void connect(LADSPA_Handle Instance, unsigned long port,
+ LADSPA_Data* data)
+ {
+ ((Revmodel *)Instance)->port[port] = data;
+ }
+
+//---------------------------------------------------------
+// activate
+//---------------------------------------------------------
+
+void activate(LADSPA_Handle instance)
+ {
+ ((Revmodel *)instance)->activate();
+ }
+
+//---------------------------------------------------------
+// deactivate
+//---------------------------------------------------------
+
+void deactivate(LADSPA_Handle /*Instance*/)
+ {
+ }
+
+//---------------------------------------------------------
+// runFreeverb
+//---------------------------------------------------------
+
+void run(LADSPA_Handle Instance, unsigned long n)
+ {
+ ((Revmodel*)Instance)->processreplace(n);
+ }
+
+//---------------------------------------------------------
+// runAddingFreeverb
+// *ADD* the output to the output buffer.
+//---------------------------------------------------------
+
+void runAdding(LADSPA_Handle Instance, unsigned long n)
+ {
+ ((Revmodel*)Instance)->processmix(n);
+ }
+
+//---------------------------------------------------------
+// setFreeverbRunAddingGain
+//---------------------------------------------------------
+
+void setGain(LADSPA_Handle /*Instance*/, LADSPA_Data /*Gain*/)
+ {
+// ((Freeverb *)Instance)->m_fRunAddingGain = Gain;
+ }
+
+//---------------------------------------------------------
+// cleanupFreeverb
+//---------------------------------------------------------
+
+void cleanup(LADSPA_Handle Instance)
+ {
+ delete (Revmodel *)Instance;
+ }
+
+static const char* portNames[] = {
+ "Input (Left)",
+ "Input (Right)",
+ "Output (Left)",
+ "Output (Right)",
+ "Room Size",
+ "Damping",
+ "Wet Level",
+ };
+
+LADSPA_PortDescriptor portDescriptors[] = {
+ LADSPA_PORT_INPUT | LADSPA_PORT_AUDIO,
+ LADSPA_PORT_INPUT | LADSPA_PORT_AUDIO,
+ LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO,
+ LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO,
+ LADSPA_PORT_INPUT | LADSPA_PORT_CONTROL,
+ LADSPA_PORT_INPUT | LADSPA_PORT_CONTROL,
+ LADSPA_PORT_INPUT | LADSPA_PORT_CONTROL,
+ };
+
+LADSPA_PortRangeHint portRangeHints[] = {
+ { 0, 0.0, 0.0 },
+ { 0, 0.0, 0.0 },
+ { 0, 0.0, 0.0 },
+ { 0, 0.0, 0.0 },
+ { LADSPA_HINT_BOUNDED_ABOVE | LADSPA_HINT_BOUNDED_BELOW, 0.0, 1.0 },
+ { LADSPA_HINT_BOUNDED_ABOVE | LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_LOGARITHMIC, 0.0, 1.0 },
+ { LADSPA_HINT_BOUNDED_ABOVE | LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_LOGARITHMIC, 0.0, 1.0 },
+ };
+
+LADSPA_Descriptor descriptor = {
+ 1050,
+ "freeverb1",
+ LADSPA_PROPERTY_HARD_RT_CAPABLE,
+ "Freeverb",
+ "Werner Schweer",
+ "None",
+ 7,
+ portDescriptors,
+ portNames,
+ portRangeHints,
+ 0, // impl. data
+ instantiate,
+ connect,
+ activate,
+ run,
+ runAdding,
+ setGain,
+ deactivate,
+ cleanup
+ };
+
+//---------------------------------------------------------
+// _init
+// called automatically when the plugin library is first
+// loaded.
+//---------------------------------------------------------
+
+void _init()
+ {
+ }
+
+//---------------------------------------------------------
+// _fini
+// called automatically when the library is unloaded.
+//---------------------------------------------------------
+
+void _fini()
+ {
+ }
+
+//---------------------------------------------------------
+// ladspa_descriptor
+// Return a descriptor of the requested plugin type.
+//---------------------------------------------------------
+
+const LADSPA_Descriptor* ladspa_descriptor(unsigned long i)
+ {
+ return (i == 0) ? &descriptor : 0;
+ }
+
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/readme.txt b/attic/muse_qt4_evolution/plugins/freeverb/readme.txt
new file mode 100644
index 00000000..2c1349a3
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/readme.txt
@@ -0,0 +1,147 @@
+readme from original freeverb-source:
+==============================================
+
+
+Freeverb - Free, studio-quality reverb SOURCE CODE in the public domain
+-----------------------------------------------------------------------
+
+Written by Jezar at Dreampoint - http://www.dreampoint.co.uk
+
+
+Introduction
+------------
+
+Hello.
+
+I'll try to keep this "readme" reasonably small.
+There are few things in the world that I hate more than long "readme" files.
+Except "coding conventions" - but more on that later...
+
+In this zip file you will find two folders of C++ source code:
+
+"Components" - Contains files that should clean-compile
+ ON ANY TYPE OF COMPUTER OR SYSTEM WHATSOEVER. It should not be necessary
+ to make ANY changes to these files to get them to compile, except to make
+ up for inadequacies of certain compilers. These files create three classes
+ - a comb filter, an allpass filter, and a reverb model made up of a number
+ of instances of the filters, with some features to control the filters at
+ a macro level. You will need to link these classes into another program that
+ interfaces with them. The files in the components drawer are completely
+ independant, and can be built without dependancies on anything else.
+ Because of the simple interface, it should be possible to interface
+ these files to any system - VST, DirectX, anything - without changing
+ them AT ALL.
+
+"FreeverbVST" - Contains a Steinberg VST implementation of this version of
+ Freeverb, using the components in (surprise) the components folder.
+ It was built on a PC but may compile properly for the Macintosh with
+ no problems. I don't know - I don't have a Macintosh. If you've
+ figured out how to compile the examples in the Steinberg VST
+ Development Kit, then you should easilly figure out how to bring the
+ files into a project and get it working in a few minutes. It should
+ be very simple.
+
+Note that this version of Freeverb doesn't contain predelay, or any EQ.
+I thought that might make it difficult to understand the "reverb" part of
+the code. Once you figure out how Freeverb works, you should find it trivial
+to add such features with little CPU overhead.
+
+Also, the code in this version of Freeverb has been optimised. This has changed
+the sound *slightly*, but not significantly compared to how much processing
+power it saves.
+
+Finally, note that there is also a built copy of this version of Freeverb called
+"Freeverb3.dll" - this is a VST plugin for the PC. If you want a version for
+the Mac or anything else, then you'll need to build it yourself from the code.
+
+
+Technical Explanation
+---------------------
+
+Freeverb is a simple implementation of the standard Schroeder/Moorer reverb
+model. I guess the only reason why it sounds better than other reverbs,
+is simply because I spent a long while doing listening tests in order to create
+the values found in "tuning.h". It uses 8 comb filters on both the left and right
+channels), and you might possibly be able to get away with less if CPU power
+is a serious constraint for you. It then feeds the result of the reverb through
+4 allpass filters on both the left and right channels. These "smooth" the sound.
+Adding more than four allpasses doesn't seem to add anything significant
+to the sound, and if you use less, the sound gets a bit "grainy".
+The filters on the right channel are slightly detuned compared to the left channel
+in order to create a stereo effect.
+
+Hopefully, you should find the code in the components drawer a model of
+brevity and clarity. Notice that I don't use any "coding conventions".
+Personally, I think that coding conventions suck. They are meant to make
+the code "clearer", but they inevitably do the complete opposite, making
+the code completely unfathomable. Anyone whose done Windows programming
+with its - frankly stupid - "Hungarian notation" will know exactly what
+I mean. Coding conventions typically promote issues that are irrelevant
+up to the status of appearing supremely important. It may have helped back
+people in the days when compilers where somewhat feeble in their type-safety,
+but not in the new millenium with advanced C++ compilers.
+
+Imagine if we rewrote the English language to conform to coding conventions.
+After all, The arguments should be just as valid for the English language as
+they are for a computer language. For example, we could put a lower-case "n"
+in front of every noun, a lower-case "p" in front of a persons name,
+a lower-case "v" in front of every verb, and a lower-case "a" in front
+of every adjective. Can you imagine what the English language would look like?
+All in the name of "clarity". It's just as stupid to do this for computer
+code as it would be to do it for the English language. I hope that the
+code for Freeverb in the components drawer demonstrates this, and helps start
+a movement back towards sanity in coding practices.
+
+
+Background
+----------
+
+Why is the Freeverb code now public domain? Simple. I only intended to create
+Freeverb to provide me and my friends with studio-quality reverb for free.
+I never intended to make any money out of it. However, I simply do not have the
+time to develop it any further. I'm working on a "concept album" at the moment,
+and I'll never finish it if I spend any more time programming.
+
+In any case, I make more far money as a contract programmer - making Mobile
+Internet products - than I ever could writing plugins, so it simply doesn't
+make financial sense for me to spend any more time on it.
+
+Rather than give Freeverb to any particular individual or organisation
+to profit from it, I've decided to give it away to the internet community
+at large, so that quality, FREE (or at the very least, low-cost) reverbs can
+be developed for all platforms.
+
+Feel free to use the source code for Freeverb in any of your own products,
+whether they are also available for free, or even if they are commercial -
+I really don't mind. You may do with the code whatever you wish. If you use
+it in a product (whether commercial or not), it would be very nice of you,
+if you were to send me a copy of your product - although I appreciate that
+this isn't always possible in all circumstances.
+
+HOWEVER, please don't bug me with questions about how to use this code.
+I gave away Freeverb because I don't have time to maintain it. That means
+I *certainly* don't have time to answer questions about the source code, so
+please don't email questions to me. I *will* ignore them. If you can't figure
+the code for Freeverb out - then find somebody who can. I hope that either
+way, you enjoy experimenting with it.
+
+
+Disclaimer
+----------
+
+This software and source code is given away for free, without any warranties
+of any kind. It has been given away to the internet community as a free gift,
+so please treat it in the same spirit.
+
+
+I hope this code is useful and interesting to you all!
+I hope you have lots of fun experimenting with it and make good products!
+
+Very best regards,
+Jezar.
+Technology Consultant
+Dreampoint Design and Engineering
+http://www.dreampoint.co.uk
+
+
+//ends
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/revmodel.cpp b/attic/muse_qt4_evolution/plugins/freeverb/revmodel.cpp
new file mode 100644
index 00000000..c72ee22b
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/revmodel.cpp
@@ -0,0 +1,232 @@
+// Reverb model implementation
+//
+// Written by Jezar at Dreampoint, June 2000
+// http://www.dreampoint.co.uk
+// This code is public domain
+
+#include <stdio.h>
+#include "revmodel.h"
+
+//---------------------------------------------------------
+// Revmodel
+//---------------------------------------------------------
+
+Revmodel::Revmodel()
+ {
+ // Tie the components to their buffers
+ combL[0].setbuffer(bufcombL1,combtuningL1);
+ combR[0].setbuffer(bufcombR1,combtuningR1);
+ combL[1].setbuffer(bufcombL2,combtuningL2);
+ combR[1].setbuffer(bufcombR2,combtuningR2);
+ combL[2].setbuffer(bufcombL3,combtuningL3);
+ combR[2].setbuffer(bufcombR3,combtuningR3);
+ combL[3].setbuffer(bufcombL4,combtuningL4);
+ combR[3].setbuffer(bufcombR4,combtuningR4);
+ combL[4].setbuffer(bufcombL5,combtuningL5);
+ combR[4].setbuffer(bufcombR5,combtuningR5);
+ combL[5].setbuffer(bufcombL6,combtuningL6);
+ combR[5].setbuffer(bufcombR6,combtuningR6);
+ combL[6].setbuffer(bufcombL7,combtuningL7);
+ combR[6].setbuffer(bufcombR7,combtuningR7);
+ combL[7].setbuffer(bufcombL8,combtuningL8);
+ combR[7].setbuffer(bufcombR8,combtuningR8);
+ allpassL[0].setbuffer(bufallpassL1,allpasstuningL1);
+ allpassR[0].setbuffer(bufallpassR1,allpasstuningR1);
+ allpassL[1].setbuffer(bufallpassL2,allpasstuningL2);
+ allpassR[1].setbuffer(bufallpassR2,allpasstuningR2);
+ allpassL[2].setbuffer(bufallpassL3,allpasstuningL3);
+ allpassR[2].setbuffer(bufallpassR3,allpasstuningR3);
+ allpassL[3].setbuffer(bufallpassL4,allpasstuningL4);
+ allpassR[3].setbuffer(bufallpassR4,allpasstuningR4);
+
+ // Set default values
+ allpassL[0].setfeedback(0.5f);
+ allpassR[0].setfeedback(0.5f);
+ allpassL[1].setfeedback(0.5f);
+ allpassR[1].setfeedback(0.5f);
+ allpassL[2].setfeedback(0.5f);
+ allpassR[2].setfeedback(0.5f);
+ allpassL[3].setfeedback(0.5f);
+ allpassR[3].setfeedback(0.5f);
+
+ param[0] = initialroom;
+ param[1] = initialdamp;
+ param[2] = initialwet;
+
+ setroomsize(initialroom);
+ setdamp(initialdamp);
+ setwidth(initialwidth);
+ setmode(initialmode);
+
+ // Buffer will be full of rubbish - so we MUST mute them
+
+ for (int i = 0; i < numcombs; i++) {
+ combL[i].mute();
+ combR[i].mute();
+ }
+ for (int i=0;i<numallpasses;i++) {
+ allpassL[i].mute();
+ allpassR[i].mute();
+ }
+ }
+
+//---------------------------------------------------------
+// activate
+//---------------------------------------------------------
+
+void Revmodel::activate()
+ {
+ *port[4] = param[0];
+ *port[5] = param[1];
+ *port[6] = param[2];
+ }
+
+//---------------------------------------------------------
+// processreplace
+//---------------------------------------------------------
+
+void Revmodel::processreplace(long n)
+ {
+ if (param[0] != *port[4]) {
+ param[0] = *port[4];
+ setroomsize(param[0]);
+ }
+ if (param[1] != *port[5]) {
+ param[1] = *port[5];
+ setdamp(param[1]);
+ }
+
+ float wet = (1.0f - *port[6]) * scalewet;
+ float dry = *port[6] * scaledry;
+ float wet1 = wet * (width/2 + 0.5f);
+ float wet2 = wet * ((1-width)/2);
+
+ for (int i = 0; i < n; ++i) {
+ float outL = 0;
+ float outR = 0;
+ float input = (port[0][i] + port[1][i]) * gain;
+
+ // Accumulate comb filters in parallel
+ for (int k = 0; k < numcombs; k++) {
+ outL += combL[k].process(input);
+ outR += combR[k].process(input);
+ }
+
+ // Feed through allpasses in series
+ for (int k=0; k < numallpasses; k++) {
+ outL = allpassL[k].process(outL);
+ outR = allpassR[k].process(outR);
+ }
+
+ // Calculate output REPLACING anything already there
+ port[2][i] = outL*wet1 + outR*wet2 + port[0][i]*dry;
+ port[3][i] = outR*wet1 + outL*wet2 + port[1][i]*dry;
+ }
+ }
+
+void Revmodel::processmix(long n)
+ {
+ if (param[0] != *port[4]) {
+ param[0] = *port[4];
+ setroomsize(param[0]);
+ }
+ if (param[1] != *port[5]) {
+ param[1] = *port[5];
+ setdamp(param[1]);
+ }
+
+ float wet = (1.0f - *port[6]) * scalewet;
+ float dry = *port[6] * scaledry;
+ float wet1 = wet * (width/2 + 0.5f);
+ float wet2 = wet * ((1-width)/2);
+
+ for (int i = 0; i < n; ++i) {
+ float outL = 0;
+ float outR = 0;
+ float input = (port[0][i] + port[1][i]) * gain;
+
+ // Accumulate comb filters in parallel
+ for (int k = 0; k < numcombs; k++) {
+ outL += combL[k].process(input);
+ outR += combR[k].process(input);
+ }
+
+ // Feed through allpasses in series
+ for (int k=0; k < numallpasses; k++) {
+ outL = allpassL[k].process(outL);
+ outR = allpassR[k].process(outR);
+ }
+
+ // Calculate output REPLACING anything already there
+ port[2][i] += outL*wet1 + outR*wet2 + port[0][i]*dry;
+ port[3][i] += outR*wet1 + outL*wet2 + port[1][i]*dry;
+ }
+ }
+
+//---------------------------------------------------------
+// update
+// Recalculate internal values after parameter change
+//---------------------------------------------------------
+
+void Revmodel::update()
+ {
+ if (mode >= freezemode) {
+ roomsize1 = 1;
+ damp1 = 0;
+ gain = muted;
+ }
+ else {
+ roomsize1 = roomsize;
+ damp1 = damp;
+ gain = fixedgain;
+ }
+
+ for (int i = 0; i < numcombs; i++) {
+ combL[i].setfeedback(roomsize1);
+ combR[i].setfeedback(roomsize1);
+ }
+
+ for (int i = 0; i < numcombs; i++) {
+ combL[i].setdamp(damp1);
+ combR[i].setdamp(damp1);
+ }
+ }
+
+// The following get/set functions are not inlined, because
+// speed is never an issue when calling them, and also
+// because as you develop the reverb model, you may
+// wish to take dynamic action when they are called.
+
+void Revmodel::setroomsize(float value)
+ {
+ roomsize = (value*scaleroom) + offsetroom;
+ update();
+ }
+
+float Revmodel::getroomsize()
+ {
+ return (roomsize-offsetroom)/scaleroom;
+ }
+
+void Revmodel::setdamp(float value)
+ {
+ damp = value*scaledamp;
+ update();
+ }
+
+void Revmodel::setwidth(float value)
+ {
+ width = value;
+ update();
+ }
+
+void Revmodel::setmode(float value)
+ {
+ mode = value;
+ update();
+ }
+
+float Revmodel::getmode()
+ {
+ return (mode >= freezemode) ? 1 : 0;
+ }
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/revmodel.h b/attic/muse_qt4_evolution/plugins/freeverb/revmodel.h
new file mode 100644
index 00000000..bfa1f0b3
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/revmodel.h
@@ -0,0 +1,79 @@
+// Reverb model declaration
+//
+// Written by Jezar at Dreampoint, June 2000
+// http://www.dreampoint.co.uk
+// This code is public domain
+
+#ifndef _revmodel_
+#define _revmodel_
+
+#include "comb.h"
+#include "allpass.h"
+#include "tuning.h"
+#include "../../muse/ladspa.h"
+
+//---------------------------------------------------------
+// Revmodel
+//---------------------------------------------------------
+
+class Revmodel {
+ float gain;
+ float roomsize,roomsize1;
+ float damp,damp1;
+ float width;
+ float mode;
+
+ // Comb filters
+ comb combL[numcombs];
+ comb combR[numcombs];
+
+ // Allpass filters
+ allpass allpassL[numallpasses];
+ allpass allpassR[numallpasses];
+
+ // Buffers for the combs
+ float bufcombL1[combtuningL1];
+ float bufcombR1[combtuningR1];
+ float bufcombL2[combtuningL2];
+ float bufcombR2[combtuningR2];
+ float bufcombL3[combtuningL3];
+ float bufcombR3[combtuningR3];
+ float bufcombL4[combtuningL4];
+ float bufcombR4[combtuningR4];
+ float bufcombL5[combtuningL5];
+ float bufcombR5[combtuningR5];
+ float bufcombL6[combtuningL6];
+ float bufcombR6[combtuningR6];
+ float bufcombL7[combtuningL7];
+ float bufcombR7[combtuningR7];
+ float bufcombL8[combtuningL8];
+ float bufcombR8[combtuningR8];
+
+ // Buffers for the allpasses
+ float bufallpassL1[allpasstuningL1];
+ float bufallpassR1[allpasstuningR1];
+ float bufallpassL2[allpasstuningL2];
+ float bufallpassR2[allpasstuningR2];
+ float bufallpassL3[allpasstuningL3];
+ float bufallpassR3[allpasstuningR3];
+ float bufallpassL4[allpasstuningL4];
+ float bufallpassR4[allpasstuningR4];
+ void update();
+
+ public:
+ LADSPA_Data* port[7];
+ float param[3];
+
+ Revmodel();
+ void processmix(long numsamples);
+ void processreplace(long numsamples);
+ void setroomsize(float value);
+ float getroomsize();
+ void setdamp(float value);
+ void setwidth(float value);
+ void setmode(float value);
+ float getmode();
+ void activate();
+ };
+
+#endif
diff --git a/attic/muse_qt4_evolution/plugins/freeverb/tuning.h b/attic/muse_qt4_evolution/plugins/freeverb/tuning.h
new file mode 100644
index 00000000..ced89252
--- /dev/null
+++ b/attic/muse_qt4_evolution/plugins/freeverb/tuning.h
@@ -0,0 +1,60 @@
+// Reverb model tuning values
+//
+// Written by Jezar at Dreampoint, June 2000
+// http://www.dreampoint.co.uk
+// This code is public domain
+
+#ifndef _tuning_
+#define _tuning_
+
+const int numcombs = 8;
+const int numallpasses = 4;
+const float muted = 0;
+const float fixedgain = 0.015f;
+const float scalewet = 3;
+const float scaledry = 2;
+const float scaledamp = 0.4f;
+const float scaleroom = 0.28f;
+const float offsetroom = 0.7f;
+const float initialroom = 0.5f;
+const float initialdamp = 0.5f;
+const float initialwet = 1/scalewet;
+const float initialdry = 0;
+const float initialwidth = 1;
+const float initialmode = 0;
+const float freezemode = 0.5f;
+const int stereospread = 23;
+
+// These values assume 44.1KHz sample rate
+// they will probably be OK for 48KHz sample rate
+// but would need scaling for 96KHz (or other) sample rates.
+// The values were obtained by listening tests.
+const int combtuningL1 = 1116;
+const int combtuningR1 = 1116+stereospread;
+const int combtuningL2 = 1188;
+const int combtuningR2 = 1188+stereospread;
+const int combtuningL3 = 1277;
+const int combtuningR3 = 1277+stereospread;
+const int combtuningL4 = 1356;
+const int combtuningR4 = 1356+stereospread;
+const int combtuningL5 = 1422;
+const int combtuningR5 = 1422+stereospread;
+const int combtuningL6 = 1491;
+const int combtuningR6 = 1491+stereospread;
+const int combtuningL7 = 1557;
+const int combtuningR7 = 1557+stereospread;
+const int combtuningL8 = 1617;
+const int combtuningR8 = 1617+stereospread;
+const int allpasstuningL1 = 556;
+const int allpasstuningR1 = 556+stereospread;
+const int allpasstuningL2 = 441;
+const int allpasstuningR2 = 441+stereospread;
+const int allpasstuningL3 = 341;
+const int allpasstuningR3 = 341+stereospread;
+const int allpasstuningL4 = 225;
+const int allpasstuningR4 = 225+stereospread;
+
+#endif//_tuning_
+
+//ends
+
--
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