diff options
Diffstat (limited to 'attic/muse/plugins/freeverb')
-rw-r--r-- | attic/muse/plugins/freeverb/Makefile.am | 17 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/allpass.h | 50 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/comb.h | 66 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/denormals.h | 28 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/freeverb.cpp | 166 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/readme.txt | 147 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/revmodel.cpp | 232 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/revmodel.h | 79 | ||||
-rw-r--r-- | attic/muse/plugins/freeverb/tuning.h | 60 |
9 files changed, 845 insertions, 0 deletions
diff --git a/attic/muse/plugins/freeverb/Makefile.am b/attic/muse/plugins/freeverb/Makefile.am new file mode 100644 index 00000000..81dbec2d --- /dev/null +++ b/attic/muse/plugins/freeverb/Makefile.am @@ -0,0 +1,17 @@ +include $(top_srcdir)/common.am +include $(top_srcdir)/plugins/plugins-install.am + +AM_CXXFLAGS += -O3 -fno-exceptions -fno-rtti -ffast-math + +plugins_LTLIBRARIES = freeverb.la + +plugins_DATA = + +freeverb_la_SOURCES = \ + freeverb.cpp \ + revmodel.cpp revmodel.h \ + allpass.h comb.h denormals.h tuning.h +freeverb_la_LDFLAGS = -module -avoid-version + +EXTRA_DIST = readme.txt + diff --git a/attic/muse/plugins/freeverb/allpass.h b/attic/muse/plugins/freeverb/allpass.h new file mode 100644 index 00000000..4eb1c1a0 --- /dev/null +++ b/attic/muse/plugins/freeverb/allpass.h @@ -0,0 +1,50 @@ +// 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; } + }; + + +// Big to inline - but crucial for speed + + +#endif//_allpass diff --git a/attic/muse/plugins/freeverb/comb.h b/attic/muse/plugins/freeverb/comb.h new file mode 100644 index 00000000..d2e0f871 --- /dev/null +++ b/attic/muse/plugins/freeverb/comb.h @@ -0,0 +1,66 @@ +// 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; } + }; + + +// Big to inline - but crucial for speed + + +#endif //_comb_ + +//ends diff --git a/attic/muse/plugins/freeverb/denormals.h b/attic/muse/plugins/freeverb/denormals.h new file mode 100644 index 00000000..d18412b4 --- /dev/null +++ b/attic/muse/plugins/freeverb/denormals.h @@ -0,0 +1,28 @@ +// 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) + +#define undenormalise(sample) \ + { \ + float anti_denormal = 1e-18; \ + sample += anti_denormal; \ + sample -= anti_denormal; \ + } + +#endif//_denormals_ + +//ends + diff --git a/attic/muse/plugins/freeverb/freeverb.cpp b/attic/muse/plugins/freeverb/freeverb.cpp new file mode 100644 index 00000000..0385e887 --- /dev/null +++ b/attic/muse/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/plugins/freeverb/readme.txt b/attic/muse/plugins/freeverb/readme.txt new file mode 100644 index 00000000..2c1349a3 --- /dev/null +++ b/attic/muse/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/plugins/freeverb/revmodel.cpp b/attic/muse/plugins/freeverb/revmodel.cpp new file mode 100644 index 00000000..c72ee22b --- /dev/null +++ b/attic/muse/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/plugins/freeverb/revmodel.h b/attic/muse/plugins/freeverb/revmodel.h new file mode 100644 index 00000000..bfa1f0b3 --- /dev/null +++ b/attic/muse/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/plugins/freeverb/tuning.h b/attic/muse/plugins/freeverb/tuning.h new file mode 100644 index 00000000..ced89252 --- /dev/null +++ b/attic/muse/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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