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Merge remote-tracking branch 'origin/stable-0.4' into sfxr

Browse Source
This commit is contained in:
Wong Cho Ching
2014-01-30 00:02:54 +08:00
parent 277ff5b99e daad00e822
commit fa1b337517
204 changed files with 3085 additions and 2253 deletions
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3
plugins/flp_import/FlpImport.cpp
View File

@@ -1744,8 +1744,7 @@ p->putValue( jt->pos, value, false );
{
continue;
}
trackContentObject * tco =
bb_tracks[it->pattern]->createTCO( midiTime() );
trackContentObject * tco = bb_tracks[it->pattern]->createTCO( MidiTime() );
tco->movePosition( it->position );
if( it->length != DefaultTicksPerTact )
{

10
plugins/ladspa_effect/cmt/src/init.cpp
View File

@@ -108,18 +108,26 @@ public:
finalise_modules();
}
} g_oStartupShutdownHandler;
} ;
/*****************************************************************************/
extern "C"
{
const LADSPA_Descriptor *
ladspa_descriptor(unsigned long Index) {
static StartupShutdownHandler handler;
if (Index < g_lPluginCount)
return g_ppsRegisteredDescriptors[Index];
else
return NULL;
}
};
/*****************************************************************************/
/* EOF */

20
plugins/ladspa_effect/swh/CMakeLists.txt
View File

@@ -29,8 +29,14 @@ FOREACH(_item ${PLUGIN_SOURCES})
ENDFOREACH(_item ${PLUGIN_SOURCES})
IF(LMMS_BUILD_WIN32)
SET(PIC_FLAGS "")
ELSE(LMMS_BUILD_WIN32)
SET(PIC_FLAGS "-fPIC")
ENDIF(LMMS_BUILD_WIN32)
ADD_LIBRARY(iir STATIC util/iir.c)
SET_TARGET_PROPERTIES(iir PROPERTIES COMPILE_FLAGS "-fPIC")
SET_TARGET_PROPERTIES(iir PROPERTIES COMPILE_FLAGS "${PIC_FLAGS}")
TARGET_LINK_LIBRARIES(bandpass_a_iir_1893 iir)
TARGET_LINK_LIBRARIES(bandpass_iir_1892 iir)
TARGET_LINK_LIBRARIES(butterworth_1902 iir)
@@ -40,21 +46,21 @@ TARGET_LINK_LIBRARIES(notch_iir_1894 iir)
FILE(GLOB GSM_SOURCES ${CMAKE_CURRENT_SOURCE_DIR}/gsm/*.c)
ADD_LIBRARY(gsm STATIC ${GSM_SOURCES})
SET_TARGET_PROPERTIES(gsm PROPERTIES COMPILE_FLAGS "-fPIC")
SET_TARGET_PROPERTIES(gsm PROPERTIES COMPILE_FLAGS "${PIC_FLAGS}")
TARGET_LINK_LIBRARIES(gsm_1215 gsm)
ADD_LIBRARY(gverb STATIC gverb/gverb.c gverb/gverbdsp.c)
SET_TARGET_PROPERTIES(gverb PROPERTIES COMPILE_FLAGS "-fPIC")
SET_TARGET_PROPERTIES(gverb PROPERTIES COMPILE_FLAGS "${PIC_FLAGS}")
TARGET_LINK_LIBRARIES(gverb_1216 gverb)
ADD_LIBRARY(blo STATIC util/blo.c)
SET_TARGET_PROPERTIES(blo PROPERTIES COMPILE_FLAGS "-fPIC")
SET_TARGET_PROPERTIES(blo PROPERTIES COMPILE_FLAGS "${PIC_FLAGS}")
TARGET_LINK_LIBRARIES(hermes_filter_1200 blo)
ADD_LIBRARY(rms STATIC util/rms.c)
ADD_LIBRARY(db STATIC util/db.c)
SET_TARGET_PROPERTIES(rms PROPERTIES COMPILE_FLAGS "-fPIC")
SET_TARGET_PROPERTIES(db PROPERTIES COMPILE_FLAGS "-fPIC")
SET_TARGET_PROPERTIES(rms PROPERTIES COMPILE_FLAGS "${PIC_FLAGS}")
SET_TARGET_PROPERTIES(db PROPERTIES COMPILE_FLAGS "${PIC_FLAGS}")
TARGET_LINK_LIBRARIES(sc1_1425 rms db)
TARGET_LINK_LIBRARIES(sc2_1426 rms db)
TARGET_LINK_LIBRARIES(sc3_1427 rms db)
@@ -63,6 +69,6 @@ TARGET_LINK_LIBRARIES(sc4m_1916 rms db)
TARGET_LINK_LIBRARIES(se4_1883 rms db)
ADD_LIBRARY(pitchscale STATIC util/pitchscale.c)
SET_TARGET_PROPERTIES(pitchscale PROPERTIES COMPILE_FLAGS "-fPIC")
SET_TARGET_PROPERTIES(pitchscale PROPERTIES COMPILE_FLAGS "${PIC_FLAGS}")
TARGET_LINK_LIBRARIES(pitch_scale_1193 pitchscale)
TARGET_LINK_LIBRARIES(pitch_scale_1194 pitchscale)

2
plugins/ladspa_effect/swh/util/blo.c
View File

@@ -48,7 +48,9 @@ blo_h_tables *blo_h_tables_new(int table_size)
unsigned int h;
size_t all_tables_size = sizeof(float) * (table_size + BLO_TABLE_WR)
* (BLO_N_HARMONICS - 1) * 2;
#ifndef WIN32
int shm_fd;
#endif
char shm_path[128];
this = malloc(sizeof(blo_h_tables));

456
plugins/ladspa_effect/swh/vocoder_1337.c Normal file
View File

@@ -0,0 +1,456 @@
/* vocoder.c
Version 0.3
LADSPA Unique ID: 1337
Version 0.31
Added stereo output, renamed input/output ports, added,
added a control for stereo balance
Version 0.3
Added support for changing bands in real time 2003-12-09
Version 0.2
Adapted to LADSPA by Josh Green <jgreen@users.sourceforge.net>
15.6.2001 (for the LinuxTag 2001!)
Original program can be found at:
http://www.sirlab.de/linux/
Author: Achim Settelmeier <settel-linux@sirlab.de>
Adapted to LMMS by Hexasoft (hexasoft.corp@free.fr)
Licence: GPL
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
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, see <http://www.gnu.org/licenses/>.
*/
/* not familiar with WINDOWS stuff. Saw this in other sources, it should be needed */
#ifdef WIN32
#define _WINDOWS_DLL_EXPORT_ __declspec(dllexport)
int bIsFirstTime = 1;
void _init(); // forward declaration
#else
#define _WINDOWS_DLL_EXPORT_
#endif
/*****************************************************************************/
/* general includes */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
/*****************************************************************************/
/* LADSPA headers */
#include <ladspa.h>
/*****************************************************************************/
#define LADSPA_UNIQUE_ID 1337
#define MAX_BANDS 16 /* max 16 bandsn should be increased */
#define AMPLIFIER 16.0
struct bandpass
{
LADSPA_Data c, f, att;
LADSPA_Data freq;
LADSPA_Data low1, low2;
LADSPA_Data mid1, mid2;
LADSPA_Data high1, high2;
LADSPA_Data y;
};
struct bands_out{
LADSPA_Data decay;
LADSPA_Data oldval;
LADSPA_Data level; /* 0.0 - 1.0 level of this output band */
};
const LADSPA_Data decay_table[] =
{
1/100.0,
1/100.0, 1/100.0, 1/100.0,
1/125.0, 1/125.0, 1/125.0,
1/166.0, 1/166.0, 1/166.0,
1/200.0, 1/200.0, 1/200.0,
1/250.0, 1/250.0, 1/250.0
};
/* The port numbers for the plugin: */
#define PORT_FORMANT 0 /* the track to "vocodify */
#define PORT_CARRIER 1 /* the track to control 1st track */
#define PORT_OUTPUT 2 /* left output */
#define PORT_OUTPUT2 3 /* right output */
#define CTRL_BANDCOUNT 4 /* selected # of bands to use */
#define CTRL_PAN 5 /* stereo balance for outputs */
#define CTRL_BAND1LVL 6 /* start of bands level */
#define PORT_COUNT 6 + MAX_BANDS /* bands level */
/* useful macros */
#undef CLAMP
#define CLAMP(x, low, high) (((x) > (high)) ? (high) : (((x) < (low)) ? (low) : (x)))
/* Instance data for the vocoder plugin */
typedef struct {
LADSPA_Data SampleRate;
int num_bands; /* current number of bands */
float mainvol; /* main volume */
struct bandpass bands_formant[MAX_BANDS]; /* one structure per band */
struct bandpass bands_carrier[MAX_BANDS]; /* one structure per band */
struct bands_out bands_out[MAX_BANDS]; /* one structure per band */
/* Ports */
LADSPA_Data * portFormant; /* Formant signal port data location */
LADSPA_Data * portCarrier; /* Carrier signal port data location */
LADSPA_Data * portOutput; /* Output audio port data location */
LADSPA_Data * portOutput2; /* Output audio port data location (copy of previous one) */
LADSPA_Data * ctrlPan; /* PAN for output */
LADSPA_Data * ctrlBandCount; /* Band count control */
LADSPA_Data * ctrlBandLevels[MAX_BANDS]; /* level controls for each band */
} VocoderInstance;
/*****************************************************************************/
/* Construct a new plugin instance. */
LADSPA_Handle
instantiateVocoder(const LADSPA_Descriptor * Descriptor,
unsigned long SampleRate) {
VocoderInstance * vocoder;
vocoder = (VocoderInstance *)malloc(sizeof(VocoderInstance));
if (vocoder == NULL)
return NULL;
vocoder->SampleRate = (LADSPA_Data)SampleRate;
vocoder->num_bands = -1;
return vocoder;
}
/*****************************************************************************/
/* Initialise and activate a plugin instance. */
void
activateVocoder(LADSPA_Handle Instance) {
VocoderInstance *vocoder = (VocoderInstance *)Instance;
int i;
vocoder->mainvol = 1.0 * AMPLIFIER;
for (i = 0; i < MAX_BANDS; i++)
vocoder->bands_out[i].oldval = 0.0;
}
/*****************************************************************************/
/* Connect a port to a data location. */
void
connectPortToVocoder(LADSPA_Handle Instance,
unsigned long Port,
LADSPA_Data * DataLocation) {
VocoderInstance * vocoder;
vocoder = (VocoderInstance *)Instance;
switch (Port) {
case PORT_FORMANT: /* formant port? */
vocoder->portFormant = DataLocation;
break;
case PORT_CARRIER: /* carrier port? */
vocoder->portCarrier = DataLocation;
break;
case PORT_OUTPUT: /* output port? */
vocoder->portOutput = DataLocation;
break;
case PORT_OUTPUT2: /* output port? */
vocoder->portOutput2 = DataLocation;
break;
case CTRL_BANDCOUNT: /* band count control? */
vocoder->ctrlBandCount = DataLocation;
break;
case CTRL_PAN: /* Pan control? */
vocoder->ctrlPan = DataLocation;
break;
default: /* a band level control? */
if (Port >= CTRL_BAND1LVL && Port < CTRL_BAND1LVL + MAX_BANDS)
vocoder->ctrlBandLevels[Port - CTRL_BAND1LVL] = DataLocation;
break;
}
}
/*****************************************************************************/
// vocoder_do_bandpasses /*fold00*/
void vocoder_do_bandpasses(struct bandpass *bands, LADSPA_Data sample,
VocoderInstance *vocoder)
{
int i;
for (i=0; i < vocoder->num_bands; i++)
{
bands[i].high1 = sample - bands[i].f * bands[i].mid1 - bands[i].low1;
bands[i].mid1 += bands[i].high1 * bands[i].c;
bands[i].low1 += bands[i].mid1;
bands[i].high2 = bands[i].low1 - bands[i].f * bands[i].mid2
- bands[i].low2;
bands[i].mid2 += bands[i].high2 * bands[i].c;
bands[i].low2 += bands[i].mid2;
bands[i].y = bands[i].high2 * bands[i].att;
}
}
/* Run a vocoder instance for a block of SampleCount samples. */
void
runVocoder(LADSPA_Handle Instance,
unsigned long SampleCount)
{
VocoderInstance *vocoder = (VocoderInstance *)Instance;
int i, j, numbands, pan;
float a;
LADSPA_Data x, c;
float fl, fr;
numbands = (int)(*vocoder->ctrlBandCount);
if (numbands < 1 || numbands > MAX_BANDS) numbands = MAX_BANDS;
/* initialize bandpass information if num_bands control has changed,
or on first run */
if (vocoder->num_bands != numbands)
{
vocoder->num_bands = numbands;
for(i=0; i < numbands; i++)
{
memset(&vocoder->bands_formant[i], 0, sizeof(struct bandpass));
a = 16.0 * i/(double)numbands; // stretch existing bands
if (a < 4.0)
vocoder->bands_formant[i].freq = 150 + 420 * a / 4.0;
else
vocoder->bands_formant[i].freq = 600 * pow (1.23, a - 4.0);
c = vocoder->bands_formant[i].freq * 2 * M_PI / vocoder->SampleRate;
vocoder->bands_formant[i].c = c * c;
vocoder->bands_formant[i].f = 0.4/c;
vocoder->bands_formant[i].att =
1/(6.0 + ((exp (vocoder->bands_formant[i].freq
/ vocoder->SampleRate) - 1) * 10));
memcpy(&vocoder->bands_carrier[i],
&vocoder->bands_formant[i], sizeof(struct bandpass));
vocoder->bands_out[i].decay = decay_table[(int)a];
vocoder->bands_out[i].level =
CLAMP (*vocoder->ctrlBandLevels[i], 0.0, 1.0);
}
}
else /* get current values of band level controls */
{
for (i = 0; i < numbands; i++)
vocoder->bands_out[i].level = CLAMP (*vocoder->ctrlBandLevels[i],
0.0, 1.0);
}
for (i=0; i < SampleCount; i++)
{
vocoder_do_bandpasses (vocoder->bands_carrier,
vocoder->portCarrier[i], vocoder);
vocoder_do_bandpasses (vocoder->bands_formant,
vocoder->portFormant[i], vocoder);
vocoder->portOutput[i] = 0.0;
vocoder->portOutput2[i] = 0.0;
for (j=0; j < numbands; j++)
{
vocoder->bands_out[j].oldval = vocoder->bands_out[j].oldval
+ (fabs (vocoder->bands_formant[j].y)
- vocoder->bands_out[j].oldval)
* vocoder->bands_out[j].decay;
x = vocoder->bands_carrier[j].y * vocoder->bands_out[j].oldval;
vocoder->portOutput[i] += x * vocoder->bands_out[j].level;
vocoder->portOutput2[i] += x * vocoder->bands_out[j].level;
}
/* treat paning + main volume */
pan = (int)(*vocoder->ctrlPan);
fl = fr = 1.;
if (pan != 0) { /* no paning, don't compute useless values */
if (pan > 0) { /* reduce left */
fl = (100.-pan)/100.;
} else {
fr = (100.+pan)/100.;
}
}
/* apply volume and paning */
vocoder->portOutput[i] *= vocoder->mainvol * fl;
vocoder->portOutput2[i] *= vocoder->mainvol * fr;
}
}
/*****************************************************************************/
/* Throw away a vocoder instance. */
void
cleanupVocoder(LADSPA_Handle Instance)
{
VocoderInstance * Vocoder;
Vocoder = (VocoderInstance *)Instance;
free(Vocoder);
}
/*****************************************************************************/
LADSPA_Descriptor * g_psDescriptor = NULL;
/*****************************************************************************/
/* _init() is called automatically when the plugin library is first
loaded. */
void
_init() {
char ** pcPortNames;
LADSPA_PortDescriptor * piPortDescriptors;
LADSPA_PortRangeHint * psPortRangeHints;
int i;
g_psDescriptor = (LADSPA_Descriptor *)malloc(sizeof(LADSPA_Descriptor));
if (g_psDescriptor) {
g_psDescriptor->UniqueID = LADSPA_UNIQUE_ID;
g_psDescriptor->Label = strdup("vocoder-lmms");
g_psDescriptor->Properties = LADSPA_PROPERTY_HARD_RT_CAPABLE;
g_psDescriptor->Name = strdup("Vocoder for LMMS");
g_psDescriptor->Maker = strdup("Achim Settelmeier (adapted to LADSPA by Josh Green, adapted to LMMS by Hexasoft)");
g_psDescriptor->Copyright = strdup("GPL");
g_psDescriptor->PortCount = PORT_COUNT;
piPortDescriptors = (LADSPA_PortDescriptor *)calloc(PORT_COUNT,
sizeof(LADSPA_PortDescriptor));
g_psDescriptor->PortDescriptors
= (const LADSPA_PortDescriptor *)piPortDescriptors;
piPortDescriptors[PORT_FORMANT]
= LADSPA_PORT_INPUT | LADSPA_PORT_AUDIO;
piPortDescriptors[PORT_CARRIER]
= LADSPA_PORT_INPUT | LADSPA_PORT_AUDIO;
piPortDescriptors[PORT_OUTPUT]
= LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
piPortDescriptors[PORT_OUTPUT2]
= LADSPA_PORT_OUTPUT | LADSPA_PORT_AUDIO;
piPortDescriptors[CTRL_BANDCOUNT]
= LADSPA_PORT_INPUT | LADSPA_PORT_CONTROL;
piPortDescriptors[CTRL_PAN]
= LADSPA_PORT_INPUT | LADSPA_PORT_CONTROL;
pcPortNames = (char **)calloc(PORT_COUNT, sizeof(char *));
g_psDescriptor->PortNames = (const char **)pcPortNames;
pcPortNames[PORT_FORMANT] = strdup("Formant-in");
pcPortNames[PORT_CARRIER] = strdup("Carrier-in");
pcPortNames[PORT_OUTPUT] = strdup("Output-out");
pcPortNames[PORT_OUTPUT2] = strdup("Output2-out");
pcPortNames[CTRL_BANDCOUNT] = strdup("Number of bands");
pcPortNames[CTRL_PAN] = strdup("Left/Right");
psPortRangeHints = ((LADSPA_PortRangeHint *)
calloc(PORT_COUNT, sizeof(LADSPA_PortRangeHint)));
g_psDescriptor->PortRangeHints
= (const LADSPA_PortRangeHint *)psPortRangeHints;
psPortRangeHints[PORT_FORMANT].HintDescriptor = 0;
psPortRangeHints[PORT_CARRIER].HintDescriptor = 0;
psPortRangeHints[PORT_OUTPUT].HintDescriptor = 0;
psPortRangeHints[PORT_OUTPUT2].HintDescriptor = 0;
psPortRangeHints[CTRL_BANDCOUNT].HintDescriptor
= LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE
| LADSPA_HINT_INTEGER;
psPortRangeHints[CTRL_BANDCOUNT].LowerBound = 1;
psPortRangeHints[CTRL_BANDCOUNT].UpperBound = MAX_BANDS;
psPortRangeHints[CTRL_PAN].HintDescriptor
= LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE
| LADSPA_HINT_INTEGER;
psPortRangeHints[CTRL_PAN].LowerBound = -100;
psPortRangeHints[CTRL_PAN].UpperBound = +100;
for (i=CTRL_BAND1LVL; i < CTRL_BAND1LVL + MAX_BANDS; i++)
{
piPortDescriptors[i] = LADSPA_PORT_INPUT | LADSPA_PORT_CONTROL;
pcPortNames[i] = malloc (sizeof ("Band 99 Level"));
sprintf(pcPortNames[i], "Band %d Level", i - CTRL_BAND1LVL + 1);
psPortRangeHints[i].HintDescriptor
= LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE;
psPortRangeHints[i].LowerBound = 0;
psPortRangeHints[i].UpperBound = 1;
}
g_psDescriptor->instantiate = instantiateVocoder;
g_psDescriptor->connect_port = connectPortToVocoder;
g_psDescriptor->activate = activateVocoder;
g_psDescriptor->run = runVocoder;
g_psDescriptor->run_adding = NULL;
g_psDescriptor->set_run_adding_gain = NULL;
g_psDescriptor->deactivate = NULL;
g_psDescriptor->cleanup = cleanupVocoder;
}
}
/*****************************************************************************/
/* _fini() is called automatically when the library is unloaded. */
void
_fini() {
long lIndex;
if (g_psDescriptor) {
free((char *)g_psDescriptor->Label);
free((char *)g_psDescriptor->Name);
free((char *)g_psDescriptor->Maker);
free((char *)g_psDescriptor->Copyright);
free((LADSPA_PortDescriptor *)g_psDescriptor->PortDescriptors);
for (lIndex = 0; lIndex < g_psDescriptor->PortCount; lIndex++)
free((char *)(g_psDescriptor->PortNames[lIndex]));
free((char **)g_psDescriptor->PortNames);
free((LADSPA_PortRangeHint *)g_psDescriptor->PortRangeHints);
free(g_psDescriptor);
}
}
/*****************************************************************************/
/* Return a descriptor of the requested plugin type. Only one plugin
type is available in this library. */
const LADSPA_Descriptor *
ladspa_descriptor(unsigned long Index) {
if (Index == 0)
return g_psDescriptor;
else
return NULL;
}
/*****************************************************************************/
/* EOF */

13
plugins/midi_import/MidiImport.cpp
View File

@@ -38,6 +38,7 @@
#include "pattern.h"
#include "Instrument.h"
#include "MainWindow.h"
#include "MidiTime.h"
#include "debug.h"
#include "embed.h"
#include "song.h"
@@ -152,7 +153,7 @@ public:
AutomationTrack * at;
AutomationPattern * ap;
midiTime lastPos;
MidiTime lastPos;
smfMidiCC & create( TrackContainer* tc )
{
@@ -172,11 +173,11 @@ public:
}
smfMidiCC & putValue( midiTime time, AutomatableModel * objModel, float value )
smfMidiCC & putValue( MidiTime time, AutomatableModel * objModel, float value )
{
if( !ap || time > lastPos + DefaultTicksPerTact )
{
midiTime pPos = midiTime( time.getTact(), 0 );
MidiTime pPos = MidiTime( time.getTact(), 0 );
ap = dynamic_cast<AutomationPattern*>(
at->createTCO(0) );
ap->movePosition( pPos );
@@ -186,7 +187,7 @@ public:
lastPos = time;
time = time - ap->startPosition();
ap->putValue( time, value, false );
ap->changeLength( midiTime( time.getTact() + 1, 0 ) );
ap->changeLength( MidiTime( time.getTact() + 1, 0 ) );
return *this;
}
@@ -212,7 +213,7 @@ public:
Instrument * it_inst;
bool isSF2;
bool hasNotes;
midiTime lastEnd;
MidiTime lastEnd;
smfMidiChannel * create( TrackContainer* tc )
{
@@ -247,7 +248,7 @@ public:
{
if( !p || n.pos() > lastEnd + DefaultTicksPerTact )
{
midiTime pPos = midiTime(n.pos().getTact(), 0 );
MidiTime pPos = MidiTime( n.pos().getTact(), 0 );
p = dynamic_cast<pattern *>( it->createTCO( 0 ) );
p->movePosition( pPos );
}

6
plugins/midi_import/MidiImport.h
View File

@@ -29,7 +29,7 @@
#include <QtCore/QPair>
#include <QtCore/QVector>
#include "midi.h"
#include "MidiEvent.h"
#include "ImportFilter.h"
@@ -117,8 +117,8 @@ private:
}
typedef QVector<QPair<int, midiEvent> > eventVector;
eventVector m_events;
typedef QVector<QPair<int, MidiEvent> > EventVector;
EventVector m_events;
int m_timingDivision;
} ;

2
plugins/opl2/CMakeLists.txt
View File

@@ -1,3 +1,3 @@
INCLUDE(BuildPlugin)
BUILD_PLUGIN(OPL2 opl2instrument.cpp opl2instrument.h opl.h kemuopl.h adlibemu.c adlibemu.h fmopl.c fmopl.h temuopl.cpp temuopl.h MOCFILES opl2instrument.h EMBEDDED_RESOURCES ${CMAKE_CURRENT_SOURCE_DIR}/*.png)
BUILD_PLUGIN(OPL2 opl2instrument.cpp opl2instrument.h opl.h fmopl.c fmopl.h temuopl.cpp temuopl.h MOCFILES opl2instrument.h EMBEDDED_RESOURCES ${CMAKE_CURRENT_SOURCE_DIR}/*.png)

502
plugins/opl2/COPYING.LESSER Normal file
View File

@@ -0,0 +1,502 @@
GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
[This is the first released version of the Lesser GPL. It also counts
as the successor of the GNU Library Public License, version 2, hence
the version number 2.1.]
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
Licenses are intended to guarantee your freedom to share and change
free software--to make sure the software is free for all its users.
This license, the Lesser General Public License, applies to some
specially designated software packages--typically libraries--of the
Free Software Foundation and other authors who decide to use it. You
can use it too, but we suggest you first think carefully about whether
this license or the ordinary General Public License is the better
strategy to use in any particular case, based on the explanations below.
When we speak of free software, we are referring to freedom of use,
not price. Our General Public Licenses are designed to make sure that
you have the freedom to distribute copies of free software (and charge
for this service if you wish); that you receive source code or can get
it if you want it; that you can change the software and use pieces of
it in new free programs; and that you are informed that you can do
these things.
To protect your rights, we need to make restrictions that forbid
distributors to deny you these rights or to ask you to surrender these
rights. These restrictions translate to certain responsibilities for
you if you distribute copies of the library or if you modify it.
For example, if you distribute copies of the library, whether gratis
or for a fee, you must give the recipients all the rights that we gave
you. You must make sure that they, too, receive or can get the source
code. If you link other code with the library, you must provide
complete object files to the recipients, so that they can relink them
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That's all there is to it!

608
plugins/opl2/adlibemu.c
View File

@@ -1,608 +0,0 @@
/*
* ADLIBEMU.C
* Copyright (C) 1998-2001 Ken Silverman
* Ken Silverman's official web site: "http://www.advsys.net/ken"
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/*
This file is a digital Adlib emulator for OPL2 and possibly OPL3
Features that could be added in a future version:
- Amplitude and Frequency Vibrato Bits (not hard, but a big speed hit)
- Global Keyboard Split Number Bit (need to research this one some more)
- 2nd Adlib chip for OPL3 (simply need to make my cell array bigger)
- Advanced connection modes of OPL3 (Just need to add more "docell" cases)
- L/R Stereo bits of OPL3 (Need adlibgetsample to return stereo)
Features that aren't worth supporting:
- Anything related to adlib timers&interrupts (Sorry - I always used IRQ0)
- Composite sine wave mode (CSM) (Supported only on ancient cards)
I'm not sure about a few things in my code:
- Attack curve. What function is this anyway? I chose to use an order-3
polynomial to approximate but this doesn't seem right.
- Attack/Decay/Release constants - my constants may not be exact
- What should ADJUSTSPEED be?
- Haven't verified that Global Keyboard Split Number Bit works yet
- Some of the drums don't always sound right. It's pretty hard to guess
the exact waveform of drums when you look at random data which is
slightly randomized due to digital ADC recording.
- Adlib seems to have a lot more treble than my emulator does. I'm not
sure if this is simply unfixable due to the sound blaster's different
filtering on FM and digital playback or if it's a serious bug in my
code.
*/
#include <math.h>
#include <string.h>
#if !defined(max) && !defined(__cplusplus)
#define max(a,b) (((a) > (b)) ? (a) : (b))
#endif
#if !defined(min) && !defined(__cplusplus)
#define min(a,b) (((a) < (b)) ? (a) : (b))
#endif
#define PI 3.141592653589793
#define MAXCELLS 18
#define WAVPREC 2048
static float AMPSCALE=(8192.0);
#define FRQSCALE (49716/512.0)
//Constants for Ken's Awe32, on a PII-266 (Ken says: Use these for KSM's!)
#define MODFACTOR 4.0 //How much of modulator cell goes into carrier
#define MFBFACTOR 1.0 //How much feedback goes back into modulator
#define ADJUSTSPEED 0.75 //0<=x<=1 Simulate finite rate of change of state
//Constants for Ken's Awe64G, on a P-133
//#define MODFACTOR 4.25 //How much of modulator cell goes into carrier
//#define MFBFACTOR 0.5 //How much feedback goes back into modulator
//#define ADJUSTSPEED 0.85 //0<=x<=1 Simulate finite rate of change of state
typedef struct
{
float val, t, tinc, vol, sustain, amp, mfb;
float a0, a1, a2, a3, decaymul, releasemul;
short *waveform;
long wavemask;
void (*cellfunc)(void *, float);
unsigned char flags, dum0, dum1, dum2;
} celltype;
static long numspeakers, bytespersample;
static float recipsamp;
static celltype cell[MAXCELLS];
static signed short wavtable[WAVPREC*3];
static float kslmul[4] = {0.0,0.5,0.25,1.0};
static float frqmul[16] = {.5,1,2,3,4,5,6,7,8,9,10,10,12,12,15,15}, nfrqmul[16];
static unsigned char adlibreg[256], ksl[8][16];
static unsigned char modulatorbase[9] = {0,1,2,8,9,10,16,17,18};
static unsigned char odrumstat = 0;
static unsigned char base2cell[22] = {0,1,2,0,1,2,0,0,3,4,5,3,4,5,0,0,6,7,8,6,7,8};
float lvol[9] = {1,1,1,1,1,1,1,1,1}; //Volume multiplier on left speaker
float rvol[9] = {1,1,1,1,1,1,1,1,1}; //Volume multiplier on right speaker
long lplc[9] = {0,0,0,0,0,0,0,0,0}; //Samples to delay on left speaker
long rplc[9] = {0,0,0,0,0,0,0,0,0}; //Samples to delay on right speaker
long nlvol[9], nrvol[9];
long nlplc[9], nrplc[9];
long rend = 0;
#define FIFOSIZ 256
static float *rptr[9], *nrptr[9];
static float rbuf[9][FIFOSIZ*2];
static float snd[FIFOSIZ*2];
#ifndef USING_ASM
#define _inline
#endif
#ifdef USING_ASM
static _inline void ftol (float f, long *a)
{
_asm
{
mov eax, a
fld f
fistp dword ptr [eax]
}
}
#else
static void ftol(float f, long *a) {
*a=f;
}
#endif
#define ctc ((celltype *)c) //A rare attempt to make code easier to read!
void docell4 (void *c, float modulator) { }
void docell3 (void *c, float modulator)
{
long i;
ftol(ctc->t+modulator,&i);
ctc->t += ctc->tinc;
ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED;
}
void docell2 (void *c, float modulator)
{
long i;
ftol(ctc->t+modulator,&i);
void *amp_void = &ctc->amp;
long *amp_long = (long *)amp_void;
if (*amp_long <= 0x37800000)
{
ctc->amp = 0;
ctc->cellfunc = docell4;
}
ctc->amp *= ctc->releasemul;
ctc->t += ctc->tinc;
ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED;
}
void docell1 (void *c, float modulator)
{
long i;
ftol(ctc->t+modulator,&i);
void *amp_void = &ctc->amp;
long *amp_long = (long *)amp_void;
void *sustain_void = &ctc->sustain;
long *sustain_long = (long *)sustain_void;
if (*amp_long <= *sustain_long)
{
if (ctc->flags&32)
{
ctc->amp = ctc->sustain;
ctc->cellfunc = docell3;
}
else
ctc->cellfunc = docell2;
}
else
ctc->amp *= ctc->decaymul;
ctc->t += ctc->tinc;
ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED;
}
void docell0 (void *c, float modulator)
{
long i;
ftol(ctc->t+modulator,&i);
ctc->amp = ((ctc->a3*ctc->amp + ctc->a2)*ctc->amp + ctc->a1)*ctc->amp + ctc->a0;
void *amp_void = &ctc->amp;
long *amp_long = (long *)amp_void;
if (*amp_long > 0x3f800000)
{
ctc->amp = 1;
ctc->cellfunc = docell1;
}
ctc->t += ctc->tinc;
ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED;
}
static long waveform[8] = {WAVPREC,WAVPREC>>1,WAVPREC,(WAVPREC*3)>>2,0,0,(WAVPREC*5)>>2,WAVPREC<<1};
static long wavemask[8] = {WAVPREC-1,WAVPREC-1,(WAVPREC>>1)-1,(WAVPREC>>1)-1,WAVPREC-1,((WAVPREC*3)>>2)-1,WAVPREC>>1,WAVPREC-1};
static long wavestart[8] = {0,WAVPREC>>1,0,WAVPREC>>2,0,0,0,WAVPREC>>3};
static float attackconst[4] = {1/2.82624,1/2.25280,1/1.88416,1/1.59744};
static float decrelconst[4] = {1/39.28064,1/31.41608,1/26.17344,1/22.44608};
void cellon (long i, long j, celltype *c, unsigned char iscarrier)
{
long frn, oct, toff;
float f;
frn = ((((long)adlibreg[i+0xb0])&3)<<8) + (long)adlibreg[i+0xa0];
oct = ((((long)adlibreg[i+0xb0])>>2)&7);
toff = (oct<<1) + ((frn>>9)&((frn>>8)|(((adlibreg[8]>>6)&1)^1)));
if (!(adlibreg[j+0x20]&16)) toff >>= 2;
f = pow(2.0,(adlibreg[j+0x60]>>4)+(toff>>2)-1)*attackconst[toff&3]*recipsamp;
c->a0 = .0377*f; c->a1 = 10.73*f+1; c->a2 = -17.57*f; c->a3 = 7.42*f;
f = -7.4493*decrelconst[toff&3]*recipsamp;
c->decaymul = pow(2.0,f*pow(2.0,(adlibreg[j+0x60]&15)+(toff>>2)));
c->releasemul = pow(2.0,f*pow(2.0,(adlibreg[j+0x80]&15)+(toff>>2)));
c->wavemask = wavemask[adlibreg[j+0xe0]&7];
c->waveform = &wavtable[waveform[adlibreg[j+0xe0]&7]];
if (!(adlibreg[1]&0x20)) c->waveform = &wavtable[WAVPREC];
c->t = wavestart[adlibreg[j+0xe0]&7];
c->flags = adlibreg[j+0x20];
c->cellfunc = docell0;
c->tinc = (float)(frn<<oct)*nfrqmul[adlibreg[j+0x20]&15];
c->vol = pow(2.0,((float)(adlibreg[j+0x40]&63) +
(float)kslmul[adlibreg[j+0x40]>>6]*ksl[oct][frn>>6]) * -.125 - 14);
c->sustain = pow(2.0,(float)(adlibreg[j+0x80]>>4) * -.5);
if (!iscarrier) c->amp = 0;
c->mfb = pow(2.0,((adlibreg[i+0xc0]>>1)&7)+5)*(WAVPREC/2048.0)*MFBFACTOR;
if (!(adlibreg[i+0xc0]&14)) c->mfb = 0;
c->val = 0;
}
//This function (and bug fix) written by Chris Moeller
void cellfreq (signed long i, signed long j, celltype *c)
{
long frn, oct;
frn = ((((long)adlibreg[i+0xb0])&3)<<8) + (long)adlibreg[i+0xa0];
oct = ((((long)adlibreg[i+0xb0])>>2)&7);
c->tinc = (float)(frn<<oct)*nfrqmul[adlibreg[j+0x20]&15];
c->vol = pow(2.0,((float)(adlibreg[j+0x40]&63) +
(float)kslmul[adlibreg[j+0x40]>>6]*ksl[oct][frn>>6]) * -.125 - 14);
}
static long initfirstime = 0;
void adlibinit (long dasamplerate, long danumspeakers, long dabytespersample)
{
long i, j, frn, oct;
memset((void *)adlibreg,0,sizeof(adlibreg));
memset((void *)cell,0,sizeof(celltype)*MAXCELLS);
memset((void *)rbuf,0,sizeof(rbuf));
rend = 0; odrumstat = 0;
for(i=0;i<MAXCELLS;i++)
{
cell[i].cellfunc = docell4;
cell[i].amp = 0;
cell[i].vol = 0;
cell[i].t = 0;
cell[i].tinc = 0;
cell[i].wavemask = 0;
cell[i].waveform = &wavtable[WAVPREC];
}
numspeakers = danumspeakers;
bytespersample = dabytespersample;
recipsamp = 1.0 / (float)dasamplerate;
for(i=15;i>=0;i--) nfrqmul[i] = frqmul[i]*recipsamp*FRQSCALE*(WAVPREC/2048.0);
if (!initfirstime)
{
initfirstime = 1;
for(i=0;i<(WAVPREC>>1);i++)
{
wavtable[i] =
wavtable[(i<<1) +WAVPREC] = (signed short)(16384*sin((float)((i<<1) )*PI*2/WAVPREC));
wavtable[(i<<1)+1+WAVPREC] = (signed short)(16384*sin((float)((i<<1)+1)*PI*2/WAVPREC));
}
for(i=0;i<(WAVPREC>>3);i++)
{
wavtable[i+(WAVPREC<<1)] = wavtable[i+(WAVPREC>>3)]-16384;
wavtable[i+((WAVPREC*17)>>3)] = wavtable[i+(WAVPREC>>2)]+16384;
}
//[table in book]*8/3
ksl[7][0] = 0; ksl[7][1] = 24; ksl[7][2] = 32; ksl[7][3] = 37;
ksl[7][4] = 40; ksl[7][5] = 43; ksl[7][6] = 45; ksl[7][7] = 47;
ksl[7][8] = 48; for(i=9;i<16;i++) ksl[7][i] = i+41;
for(j=6;j>=0;j--)
for(i=0;i<16;i++)
{
oct = (long)ksl[j+1][i]-8; if (oct < 0) oct = 0;
ksl[j][i] = (unsigned char)oct;
}
}
else
{
for(i=0;i<9;i++)
{
frn = ((((long)adlibreg[i+0xb0])&3)<<8) + (long)adlibreg[i+0xa0];
oct = ((((long)adlibreg[i+0xb0])>>2)&7);
cell[i].tinc = (float)(frn<<oct)*nfrqmul[adlibreg[modulatorbase[i]+0x20]&15];
}
}
}
void adlib0 (long i, long v)
{
unsigned char tmp = adlibreg[i];
adlibreg[i] = v;
if (i == 0xbd)
{
if ((v&16) > (odrumstat&16)) //BassDrum
{
cellon(6,16,&cell[6],0);
cellon(6,19,&cell[15],1);
cell[15].vol *= 2;
}
if ((v&8) > (odrumstat&8)) //Snare
{
cellon(16,20,&cell[16],0);
cell[16].tinc *= 2*(nfrqmul[adlibreg[17+0x20]&15] / nfrqmul[adlibreg[20+0x20]&15]);
if (((adlibreg[20+0xe0]&7) >= 3) && ((adlibreg[20+0xe0]&7) <= 5)) cell[16].vol = 0;
cell[16].vol *= 2;
}
if ((v&4) > (odrumstat&4)) //TomTom
{
cellon(8,18,&cell[8],0);
cell[8].vol *= 2;
}
if ((v&2) > (odrumstat&2)) //Cymbal
{
cellon(17,21,&cell[17],0);
cell[17].wavemask = wavemask[5];
cell[17].waveform = &wavtable[waveform[5]];
cell[17].tinc *= 16; cell[17].vol *= 2;
//cell[17].waveform = &wavtable[WAVPREC]; cell[17].wavemask = 0;
//if (((adlibreg[21+0xe0]&7) == 0) || ((adlibreg[21+0xe0]&7) == 6))
// cell[17].waveform = &wavtable[(WAVPREC*7)>>2];
//if (((adlibreg[21+0xe0]&7) == 2) || ((adlibreg[21+0xe0]&7) == 3))
// cell[17].waveform = &wavtable[(WAVPREC*5)>>2];
}
if ((v&1) > (odrumstat&1)) //Hihat
{
cellon(7,17,&cell[7],0);
if (((adlibreg[17+0xe0]&7) == 1) || ((adlibreg[17+0xe0]&7) == 4) ||
((adlibreg[17+0xe0]&7) == 5) || ((adlibreg[17+0xe0]&7) == 7)) cell[7].vol = 0;
if ((adlibreg[17+0xe0]&7) == 6) { cell[7].wavemask = 0; cell[7].waveform = &wavtable[(WAVPREC*7)>>2]; }
}
odrumstat = v;
}
else if (((unsigned)(i-0x40) < (unsigned)22) && ((i&7) < 6))
{
if ((i&7) < 3) // Modulator
cellfreq(base2cell[i-0x40],i-0x40,&cell[base2cell[i-0x40]]);
else // Carrier
cellfreq(base2cell[i-0x40],i-0x40,&cell[base2cell[i-0x40]+9]);
}
else if ((unsigned)(i-0xa0) < (unsigned)9)
{
cellfreq(i-0xa0,modulatorbase[i-0xa0],&cell[i-0xa0]);
cellfreq(i-0xa0,modulatorbase[i-0xa0]+3,&cell[i-0xa0+9]);
}
else if ((unsigned)(i-0xb0) < (unsigned)9)
{
if ((v&32) > (tmp&32))
{
cellon(i-0xb0,modulatorbase[i-0xb0],&cell[i-0xb0],0);
cellon(i-0xb0,modulatorbase[i-0xb0]+3,&cell[i-0xb0+9],1);
}
else if ((v&32) < (tmp&32))
cell[i-0xb0].cellfunc = cell[i-0xb0+9].cellfunc = docell2;
cellfreq(i-0xb0,modulatorbase[i-0xb0],&cell[i-0xb0]);
cellfreq(i-0xb0,modulatorbase[i-0xb0]+3,&cell[i-0xb0+9]);
}
//outdata(i,v);
}
#ifdef USING_ASM
static long fpuasm;
static float fakeadd = 8388608.0+128.0;
static _inline void clipit8 (float f, long a)
{
_asm
{
mov edi, a
fld dword ptr f
fadd dword ptr fakeadd
fstp dword ptr fpuasm
mov eax, fpuasm
test eax, 0x007fff00
jz short skipit
shr eax, 16
xor eax, -1
skipit: mov byte ptr [edi], al
}
}
static _inline void clipit16 (float f, long a)
{
_asm
{
mov eax, a
fld dword ptr f
fist word ptr [eax]
cmp word ptr [eax], 0x8000
jne short skipit2
fst dword ptr [fpuasm]
cmp fpuasm, 0x80000000
sbb word ptr [eax], 0
skipit2: fstp st
}
}
#else
static void clipit8(float f,unsigned char *a) {
f/=256.0;
f+=128.0;
if (f>254.5) *a=255;
else if (f<0.5) *a=0;
else *a=f;
}
static void clipit16(float f,short *a) {
if (f>32766.5) *a=32767;
else if (f<-32767.5) *a=-32768;
else *a=f;
}
#endif
void adlibsetvolume(int i) {
AMPSCALE=i;
}
void adlibgetsample (unsigned char *sndptr, long numbytes)
{
long i, j, k=0, ns, endsamples, rptrs, numsamples;
celltype *cptr;
float f;
short *sndptr2=(short *)sndptr;
numsamples = (numbytes>>(numspeakers+bytespersample-2));
if (bytespersample == 1) f = AMPSCALE/256.0; else f = AMPSCALE;
if (numspeakers == 1)
{
nlvol[0] = lvol[0]*f;
for(i=0;i<9;i++) rptr[i] = &rbuf[0][0];
rptrs = 1;
}
else
{
rptrs = 0;
for(i=0;i<9;i++)
{
if ((!i) || (lvol[i] != lvol[i-1]) || (rvol[i] != rvol[i-1]) ||
(lplc[i] != lplc[i-1]) || (rplc[i] != rplc[i-1]))
{
nlvol[rptrs] = lvol[i]*f;
nrvol[rptrs] = rvol[i]*f;
nlplc[rptrs] = rend-min(max(lplc[i],0),FIFOSIZ);
nrplc[rptrs] = rend-min(max(rplc[i],0),FIFOSIZ);
rptrs++;
}
rptr[i] = &rbuf[rptrs-1][0];
}
}
//CPU time used to be somewhat less when emulator was only mono!
// Because of no delay fifos!
for(ns=0;ns<numsamples;ns+=endsamples)
{
endsamples = min(FIFOSIZ*2-rend,FIFOSIZ);
endsamples = min(endsamples,numsamples-ns);
for(i=0;i<9;i++)
nrptr[i] = &rptr[i][rend];
for(i=0;i<rptrs;i++)
memset((void *)&rbuf[i][rend],0,endsamples*sizeof(float));
if (adlibreg[0xbd]&0x20)
{
//BassDrum (j=6)
if (cell[15].cellfunc != docell4)
{
if (adlibreg[0xc6]&1)
{
for(i=0;i<endsamples;i++)
{
(cell[15].cellfunc)((void *)&cell[15],0.0);
nrptr[6][i] += cell[15].val;
}
}
else
{
for(i=0;i<endsamples;i++)
{
(cell[6].cellfunc)((void *)&cell[6],cell[6].val*cell[6].mfb);
(cell[15].cellfunc)((void *)&cell[15],cell[6].val*WAVPREC*MODFACTOR);
nrptr[6][i] += cell[15].val;
}
}
}
//Snare/Hihat (j=7), Cymbal/TomTom (j=8)
if ((cell[7].cellfunc != docell4) || (cell[8].cellfunc != docell4) || (cell[16].cellfunc != docell4) || (cell[17].cellfunc != docell4))
{
for(i=0;i<endsamples;i++)
{
k = k*1664525+1013904223;
(cell[16].cellfunc)((void *)&cell[16],k&((WAVPREC>>1)-1)); //Snare
(cell[7].cellfunc)((void *)&cell[7],k&(WAVPREC-1)); //Hihat
(cell[17].cellfunc)((void *)&cell[17],k&((WAVPREC>>3)-1)); //Cymbal
(cell[8].cellfunc)((void *)&cell[8],0.0); //TomTom
nrptr[7][i] += cell[7].val + cell[16].val;
nrptr[8][i] += cell[8].val + cell[17].val;
}
}
}
for(j=9-1;j>=0;j--)
{
if ((adlibreg[0xbd]&0x20) && (j >= 6) && (j < 9)) continue;
cptr = &cell[j]; k = j;
if (adlibreg[0xc0+k]&1)
{
if ((cptr[9].cellfunc == docell4) && (cptr->cellfunc == docell4)) continue;
for(i=0;i<endsamples;i++)
{
(cptr->cellfunc)((void *)cptr,cptr->val*cptr->mfb);
(cptr->cellfunc)((void *)&cptr[9],0);
nrptr[j][i] += cptr[9].val + cptr->val;
}
}
else
{
if (cptr[9].cellfunc == docell4) continue;
for(i=0;i<endsamples;i++)
{
(cptr->cellfunc)((void *)cptr,cptr->val*cptr->mfb);
(cptr[9].cellfunc)((void *)&cptr[9],cptr->val*WAVPREC*MODFACTOR);
nrptr[j][i] += cptr[9].val;
}
}
}
if (numspeakers == 1)
{
if (bytespersample == 1)
{
for(i=endsamples-1;i>=0;i--)
clipit8(nrptr[0][i]*nlvol[0],sndptr+1);
}
else
{
for(i=endsamples-1;i>=0;i--)
clipit16(nrptr[0][i]*nlvol[0],sndptr2+i);
}
}
else
{
memset((void *)snd,0,endsamples*sizeof(float)*2);
for(j=0;j<rptrs;j++)
{
for(i=0;i<endsamples;i++)
{
snd[(i<<1) ] += rbuf[j][(nlplc[j]+i)&(FIFOSIZ*2-1)]*nlvol[j];
snd[(i<<1)+1] += rbuf[j][(nrplc[j]+i)&(FIFOSIZ*2-1)]*nrvol[j];
}
nlplc[j] += endsamples;
nrplc[j] += endsamples;
}
if (bytespersample == 1)
{
for(i=(endsamples<<1)-1;i>=0;i--)
clipit8(snd[i],sndptr+i);
}
else
{
for(i=(endsamples<<1)-1;i>=0;i--)
clipit16(snd[i],sndptr2+i);
}
}
sndptr = sndptr+(numspeakers*endsamples);
sndptr2 = sndptr2+(numspeakers*endsamples);
rend = ((rend+endsamples)&(FIFOSIZ*2-1));
}
}

26
plugins/opl2/adlibemu.h
View File

@@ -1,26 +0,0 @@
/*
* ADLIBEMU.H
* Copyright (C) 1998-2001 Ken Silverman
* Ken Silverman's official web site: "http://www.advsys.net/ken"
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
void adlibinit(long dasamplerate,long danumspeakers,long dabytespersample);
void adlib0(long i,long v);
void adlibgetsample(void *sndptr,long numbytes);
void adlibsetvolume(int i);
void randoinsts();
extern float lvol[9],rvol[9],lplc[9],rplc[9];

2
plugins/opl2/fmopl.c
View File

@@ -596,7 +596,7 @@ static void init_timetables( FM_OPL *OPL , int ARRATE , int DRRATE )
OPL->AR_TABLE[i] = rate / ARRATE;
OPL->DR_TABLE[i] = rate / DRRATE;
}
for (i = 60;i < 76;i++)
for (i = 60;i < 75;i++)
{
OPL->AR_TABLE[i] = EG_AED-1;
OPL->DR_TABLE[i] = OPL->DR_TABLE[60];

61
plugins/opl2/kemuopl.h
View File

@@ -1,61 +0,0 @@
/*
* Adplug - Replayer for many OPL2/OPL3 audio file formats.
* Copyright (C) 1999 - 2005 Simon Peter, <dn.tlp@gmx.net>, et al.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* kemuopl.h - Emulated OPL using Ken Silverman's emulator, by Simon Peter
* <dn.tlp@gmx.net>
*/
#ifndef H_ADPLUG_KEMUOPL
#define H_ADPLUG_KEMUOPL
#include "opl.h"
extern "C" {
#include "adlibemu.h"
}
class CKemuopl: public Copl
{
public:
CKemuopl(int rate, bool bit16, bool usestereo)
: use16bit(bit16), stereo(usestereo)
{
adlibinit(rate, usestereo ? 2 : 1, bit16 ? 2 : 1);
currType = TYPE_OPL2;
};
void update(short *buf, int samples)
{
if(use16bit) samples *= 2;
if(stereo) samples *= 2;
adlibgetsample(buf, samples);
}
// template methods
void write(int reg, int val)
{
if(currChip == 0)
adlib0(reg, val);
};
void init() {};
private:
bool use16bit,stereo;
};
#endif

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plugins/opl2/logo.png
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235
plugins/opl2/opl2instrument.cpp
View File

@@ -1,7 +1,7 @@
/*
* OPL2 FM synth
*
* Copyright (c) 2013 Raine M. Ekman <raine/at/iki/fi>
* Copyright (c) 2014 Raine M. Ekman <raine/at/iki/fi>
*
* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
*
@@ -23,10 +23,10 @@
*/
// TODO:
// - Pitch bend
// - Velocity (and aftertouch) sensitivity
// - in FM mode: OP2 level, add mode: OP1 and OP2 levels
// - Better voice allocation: long releases get cut short :(
// - .sbi (or similar) file loading into models
// - RT safety = get rid of mutex = make emulator code thread-safe
// - Extras:
// - double release: first release is in effect until noteoff (heard if percussive sound),
// second is switched in just before key bit cleared (is this useful???)
@@ -53,7 +53,6 @@
#include "opl.h"
#include "temuopl.h"
#include "kemuopl.h"
#include "embed.cpp"
#include "math.h"
@@ -92,6 +91,9 @@ Plugin * PLUGIN_EXPORT lmms_plugin_main( Model *, void * _data )
// the emulator code isn't really ready for threads
QMutex opl2instrument::emulatorMutex;
// Weird ordering of voice parameters
const unsigned int adlib_opadd[9] = {0x00, 0x01, 0x02, 0x08, 0x09, 0x0A, 0x10, 0x11, 0x12};
opl2instrument::opl2instrument( InstrumentTrack * _instrument_track ) :
Instrument( _instrument_track, &OPL2_plugin_descriptor ),
m_patchModel( 0, 0, 127, this, tr( "Patch" ) ),
@@ -139,28 +141,38 @@ opl2instrument::opl2instrument( InstrumentTrack * _instrument_track ) :
InstrumentPlayHandle * iph = new InstrumentPlayHandle( this );
engine::mixer()->addPlayHandle( iph );
// Voices are laid out in a funny way...
// adlib_opadd = {0x00, 0x01, 0x02, 0x08, 0x09, 0x0A, 0x10, 0x11, 0x12};
// Create an emulator - samplerate, 16 bit, mono
// CTemuopl is the better one, CKemuopl kinda sucks (some sounds silent, pitch goes flat after a while)
emulatorMutex.lock();
// theEmulator = new CKemuopl(engine::mixer()->processingSampleRate(), true, false);
theEmulator = new CTemuopl(engine::mixer()->processingSampleRate(), true, false);
theEmulator->init();
// Enable waveform selection
theEmulator->write(0x01,0x20);
emulatorMutex.unlock();
//Initialize voice values
voiceNote[0] = 0;
voiceLRU[0] = 0;
for(int i=1; i<9; ++i) {
voiceNote[i] = OPL2_VOICE_FREE;
voiceLRU[i] = i;
}
updatePatch();
// Can the buffer size change suddenly? I bet that would break lots of stuff
frameCount = engine::mixer()->framesPerPeriod();
renderbuffer = new short[frameCount];
// Some kind of sane default
// Some kind of sane defaults
pitchbend = 0;
pitchBendRange = 100;
RPNcoarse = RPNfine = 255;
tuneEqual(69, 440);
for(int i=1; i<9; ++i) {
voiceNote[i] = OPL2_VOICE_FREE;
}
connect( engine::mixer(), SIGNAL( sampleRateChanged() ),
this, SLOT( reloadEmulator() ) );
// Connect knobs
@@ -207,8 +219,15 @@ opl2instrument::opl2instrument( InstrumentTrack * _instrument_track ) :
MOD_CON( trem_depth_mdl );
}
opl2instrument::~opl2instrument() {
delete theEmulator;
engine::mixer()->removePlayHandles( instrumentTrack() );
delete [] renderbuffer;
}
// Samplerate changes when choosing oversampling, so this is more or less mandatory
void opl2instrument::reloadEmulator() {
delete theEmulator;
emulatorMutex.lock();
theEmulator = new CTemuopl(engine::mixer()->processingSampleRate(), true, false);
theEmulator->init();
@@ -216,50 +235,141 @@ void opl2instrument::reloadEmulator() {
emulatorMutex.unlock();
for(int i=1; i<9; ++i) {
voiceNote[i] = OPL2_VOICE_FREE;
voiceLRU[i] = i;
}
updatePatch();
}
bool opl2instrument::handleMidiEvent( const midiEvent & _me,
const midiTime & _time )
// This shall only be called from code protected by the holy Mutex!
void opl2instrument::setVoiceVelocity(int voice, int vel) {
int vel_adjusted;
// Velocity calculation, some kind of approximation
// Only calculate for operator 1 if in adding mode, don't want to change timbre
if( fm_mdl.value() == false ) {
vel_adjusted = 63 - ( op1_lvl_mdl.value() * vel/127.0) ;
} else {
vel_adjusted = 63 - op1_lvl_mdl.value();
}
theEmulator->write(0x40+adlib_opadd[voice],
( (int)op1_scale_mdl.value() & 0x03 << 6) +
( vel_adjusted & 0x3f ) );
vel_adjusted = 63 - ( op2_lvl_mdl.value() * vel/127.0 );
// vel_adjusted = 63 - op2_lvl_mdl.value();
theEmulator->write(0x43+adlib_opadd[voice],
( (int)op2_scale_mdl.value() & 0x03 << 6) +
( vel_adjusted & 0x3f ) );
}
// Pop least recently used voice - why does it sometimes lose a voice (mostly 0)?
int opl2instrument::popVoice() {
int tmp = voiceLRU[0];
for( int i=0; i<8; ++i) {
voiceLRU[i] = voiceLRU[i+1];
}
voiceLRU[8] = OPL2_NO_VOICE;
return tmp;
}
int opl2instrument::pushVoice(int v) {
int i;
for(i=8; i>0; --i) {
if( voiceLRU[i-1] != OPL2_NO_VOICE ) {
break;
}
}
voiceLRU[i] = v;
return i;
}
bool opl2instrument::handleMidiEvent( const MidiEvent& event, const MidiTime& time )
{
emulatorMutex.lock();
// Real dummy version... Should at least add:
// - smarter voice allocation:
// - reuse same note, now we have round robin-ish
// - what to do when voices run out and so on...
// - mono mode
//
int key;
static int lastvoice=0;
if( _me.m_type == MidiNoteOn ) {
// to get us in line with MIDI
key = _me.key() +12;
for(int i=lastvoice+1; i!=lastvoice; ++i,i%=9) {
if( voiceNote[i] == OPL2_VOICE_FREE ) {
theEmulator->write(0xA0+i, fnums[key] & 0xff);
theEmulator->write(0xB0+i, 32 + ((fnums[key] & 0x1f00) >> 8) );
// printf("%d: %d %d\n", key, (fnums[key] & 0x1c00) >> 10, fnums[key] & 0x3ff);
voiceNote[i] = key;
// printf("Voice %d on\n",i);
lastvoice=i;
break;
int key, vel, voice, tmp_pb;
switch(event.type()) {
case MidiNoteOn:
// to get us in line with MIDI(?)
key = event.key() +12;
vel = event.velocity();
voice = popVoice();
if( voice != OPL2_NO_VOICE ) {
// Turn voice on, NB! the frequencies are straight by voice number,
// not by the adlib_opadd table!
theEmulator->write(0xA0+voice, fnums[key] & 0xff);
theEmulator->write(0xB0+voice, 32 + ((fnums[key] & 0x1f00) >> 8) );
setVoiceVelocity(voice, vel);
voiceNote[voice] = key;
velocities[key] = vel;
}
break;
case MidiNoteOff:
key = event.key() +12;
for(voice=0; voice<9; ++voice) {
if( voiceNote[voice] == key ) {
theEmulator->write(0xA0+voice, fnums[key] & 0xff);
theEmulator->write(0xB0+voice, (fnums[key] & 0x1f00) >> 8 );
voiceNote[voice] = OPL2_VOICE_FREE;
pushVoice(voice);
}
}
velocities[key] = 0;
break;
case MidiKeyPressure:
key = event.key() +12;
vel = event.velocity();
if( velocities[key] != 0) {
velocities[key] = vel;
}
for(voice=0; voice<9; ++voice) {
if(voiceNote[voice] == key) {
setVoiceVelocity(voice, vel);
}
}
} else if( _me.m_type == MidiNoteOff ) {
key = _me.key() +12;
for(int i=0; i<9; ++i) {
if( voiceNote[i] == key ) {
theEmulator->write(0xA0+i, fnums[key] & 0xff);
theEmulator->write(0xB0+i, (fnums[key] & 0x1f00) >> 8 );
voiceNote[i] = OPL2_VOICE_FREE;
}
break;
case MidiPitchBend:
// Update fnumber table
// Pitchbend should be in the range 0...16383 but the new range knob gets it wrong.
// tmp_pb = (2*BEND_CENTS)*((float)event.m_data.m_param[0]/16383)-BEND_CENTS;
// Something like 100 cents = 8192, but offset by 8192 so the +/-100 cents range goes from 0...16383?
tmp_pb = ( event.pitchBend()-8192 ) * pitchBendRange / 8192;
if( tmp_pb != pitchbend ) {
pitchbend = tmp_pb;
tuneEqual(69, 440.0);
}
} else {
printf("Midi event type %d\n",_me.m_type);
// 224 - pitch wheel
// 160 - aftertouch?
}
// Update pitch of sounding notes
for( int v=0; v<9; ++v ) {
if( voiceNote[v] != OPL2_VOICE_FREE ) {
theEmulator->write(0xA0+v, fnums[voiceNote[v] ] & 0xff);
theEmulator->write(0xB0+v, 32 + ((fnums[voiceNote[v]] & 0x1f00) >> 8) );
}
}
break;
case MidiControlChange:
switch (event.controllerNumber()) {
case MidiControllerRegisteredParameterNumberLSB:
RPNfine = event.controllerValue();
break;
case MidiControllerRegisteredParameterNumberMSB:
RPNcoarse = event.controllerValue();
break;
case MidiControllerDataEntry:
if( (RPNcoarse << 8) + RPNfine == MidiPitchBendSensitivityRPN) {
pitchBendRange = event.controllerValue() * 100;
}
break;
default:
printf("Midi CC %02x %02x\n", event.controllerNumber(), event.controllerValue() );
break;
}
break;
default:
printf("Midi event type %d\n",event.type());
}
emulatorMutex.unlock();
return true;
}
@@ -332,7 +442,6 @@ void opl2instrument::saveSettings( QDomDocument & _doc, QDomElement & _this )
void opl2instrument::loadSettings( const QDomElement & _this )
{
printf("loadSettings!\n");
op1_a_mdl.loadSettings( _this, "op1_a" );
op1_d_mdl.loadSettings( _this, "op1_d" );
op1_s_mdl.loadSettings( _this, "op1_s" );
@@ -366,19 +475,14 @@ void opl2instrument::loadSettings( const QDomElement & _this )
}
// Load a preset in binary form
// Load a patch into the emulator
void opl2instrument::loadPatch(unsigned char inst[14]) {
const unsigned int adlib_opadd[] = {0x00, 0x01, 0x02, 0x08, 0x09, 0x0A, 0x10, 0x11, 0x12};
// Set all voices
printf("%02x %02x %02x %02x %02x ",inst[0],inst[1],inst[2],inst[3],inst[4]);
printf("%02x %02x %02x %02x %02x %02x\n",inst[5],inst[6],inst[7],inst[8],inst[9],inst[10]);
emulatorMutex.lock();
for(int v=0; v<9; ++v) {
theEmulator->write(0x20+adlib_opadd[v],inst[0]); // op1 AM/VIB/EG/KSR/Multiplier
theEmulator->write(0x23+adlib_opadd[v],inst[1]); // op2
theEmulator->write(0x40+adlib_opadd[v],inst[2]); // op1 KSL/Output Level
theEmulator->write(0x43+adlib_opadd[v],inst[3]); // op2
// theEmulator->write(0x40+adlib_opadd[v],inst[2]); // op1 KSL/Output Level - these are handled by noteon/aftertouch code
// theEmulator->write(0x43+adlib_opadd[v],inst[3]); // op2
theEmulator->write(0x60+adlib_opadd[v],inst[4]); // op1 A/D
theEmulator->write(0x63+adlib_opadd[v],inst[5]); // op2
theEmulator->write(0x80+adlib_opadd[v],inst[6]); // op1 S/R
@@ -391,10 +495,10 @@ void opl2instrument::loadPatch(unsigned char inst[14]) {
}
void opl2instrument::tuneEqual(int center, float Hz) {
float tmp;
for(int n=0; n<128; ++n) {
float tmp = Hz*pow(2, (n-center)/12.0);
tmp = Hz*pow( 2, ( n - center ) / 12.0 + pitchbend / 1200.0 );
fnums[n] = Hz2fnum( tmp );
//printf("%d: %d %d %f\n", n, (fnums[n] & 0x1c00) >> 10, fnums[n] & 0x3ff,tmp);
}
}
@@ -412,7 +516,6 @@ int opl2instrument::Hz2fnum(float Hz) {
// Load one of the default patches
void opl2instrument::loadGMPatch() {
unsigned char *inst = midi_fm_instruments[m_patchModel.value()];
// printf("loadGMPatch: %d ", m_patchModel.value());
loadPatch(inst);
}
@@ -423,7 +526,6 @@ void opl2instrument::loadGMPatch() {
// Update patch from the models to the chip emulation
void opl2instrument::updatePatch() {
printf("updatePatch()\n");
unsigned char *inst = midi_fm_instruments[0];
inst[0] = ( op1_trem_mdl.value() ? 128 : 0 ) +
( op1_vib_mdl.value() ? 64 : 0 ) +
@@ -456,10 +558,16 @@ void opl2instrument::updatePatch() {
inst[12] = 0;
inst[13] = 0;
// Not part of the patch per se
// Not part of the per-voice patch info
theEmulator->write(0xBD, (trem_depth_mdl.value() ? 128 : 0 ) +
(vib_depth_mdl.value() ? 64 : 0 ));
// have to do this, as the level knobs might've changed
for( int voice = 0; voice < 9 ; ++voice) {
if(voiceNote[voice]!=OPL2_VOICE_FREE) {
setVoiceVelocity(voice, velocities[voiceNote[voice]] );
}
}
loadPatch(inst);
}
@@ -552,9 +660,9 @@ opl2instrumentView::opl2instrumentView( Instrument * _instrument,
pal.setBrush( backgroundRole(), PLUGIN_NAME::getIconPixmap(
"artwork" ) );
setPalette( pal );
}
opl2instrumentView::~opl2instrumentView() {
// Nobody else seems to delete their knobs and buttons?
}
void opl2instrumentView::modelChanged()
@@ -604,4 +712,5 @@ void opl2instrumentView::modelChanged()
}
#include "moc_opl2instrument.cxx"

25
plugins/opl2/opl2instrument.h
View File

@@ -34,19 +34,23 @@
#include "pixmap_button.h"
#define OPL2_VOICE_FREE 255
#define OPL2_NO_VOICE 255
// The "normal" range for LMMS pitchbends
#define DEFAULT_BEND_CENTS 100
class opl2instrument : public Instrument
{
Q_OBJECT
public:
opl2instrument( InstrumentTrack * _instrument_track );
virtual ~opl2instrument();
virtual QString nodeName() const;
virtual PluginView * instantiateView( QWidget * _parent );
inline virtual bool isMidiBased() const { return true; }
virtual bool handleMidiEvent( const midiEvent & _me,
const midiTime & _time );
virtual bool handleMidiEvent( const MidiEvent& event, const MidiTime& time );
virtual void play( sampleFrame * _working_buffer );
void saveSettings( QDomDocument & _doc, QDomElement & _this );
@@ -107,20 +111,35 @@ private:
fpp_t frameCount;
short *renderbuffer;
int voiceNote[9];
int heldNotes[128];
// Least recently used voices
int voiceLRU[9];
// 0 - no note, >0 - note on velocity
int velocities[128];
// These include both octave and Fnumber
int fnums[128];
// in cents, range defaults to +/-100 cents (should this be changeable?)
int pitchbend;
int pitchBendRange;
int popVoice();
int pushVoice(int v);
int Hz2fnum(float Hz);
static QMutex emulatorMutex;
void setVoiceVelocity(int voice, int vel);
// Pitch bend range comes through RPNs.
int RPNcoarse, RPNfine;
};
class opl2instrumentView : public InstrumentView
{
Q_OBJECT
public:
opl2instrumentView( Instrument * _instrument, QWidget * _parent );
virtual ~opl2instrumentView();
lcdSpinBox *m_patch;
void modelChanged();

4
plugins/papu/gb_apu/Gb_Apu.h
View File

@@ -6,8 +6,8 @@
#ifndef GB_APU_H
#define GB_APU_H
typedef long gb_time_t; // clock cycle count
typedef unsigned gb_addr_t; // 16-bit address
typedef int gb_time_t; // clock cycle count
typedef int gb_addr_t; // 16-bit address
#include "Gb_Oscs.h"

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1
plugins/peak_controller_effect/peak_controller_effect.cpp
View File

@@ -81,6 +81,7 @@ PeakControllerEffect::~PeakControllerEffect()
if( idx >= 0 )
{
PeakController::s_effects.remove( idx );
engine::getSong()->removeController( m_autoController );
}
}

3
plugins/peak_controller_effect/peak_controller_effect_controls.cpp
View File

@@ -29,6 +29,7 @@
#include "PeakController.h"
#include "peak_controller_effect_controls.h"
#include "peak_controller_effect.h"
#include "preset_preview_play_handle.h"
PeakControllerEffectControls::
@@ -66,7 +67,7 @@ void PeakControllerEffectControls::loadSettings( const QDomElement & _this )
}
m_effect->m_effectId = effectId;
if( m_effect->m_autoController )
if( m_effect->m_autoController && presetPreviewPlayHandle::isPreviewing() == false )
{
delete m_effect->m_autoController;
m_effect->m_autoController = 0;

16
plugins/sf2_player/sf2_player.cpp
View File

@@ -87,7 +87,8 @@ sf2Instrument::sf2Instrument( InstrumentTrack * _instrument_track ) :
m_font( NULL ),
m_fontId( 0 ),
m_filename( "" ),
m_lastMidiPitch( 8192 ),
m_lastMidiPitch( -1 ),
m_lastMidiPitchRange( -1 ),
m_channel( 1 ),
m_bankNum( 0, 0, 999, this, tr("Bank") ),
m_patchNum( 0, 0, 127, this, tr("Patch") ),
@@ -679,12 +680,21 @@ void sf2Instrument::play( sampleFrame * _working_buffer )
const fpp_t frames = engine::mixer()->framesPerPeriod();
m_synthMutex.lock();
if( m_lastMidiPitch != instrumentTrack()->midiPitch() )
const int currentMidiPitch = instrumentTrack()->midiPitch();
if( m_lastMidiPitch != currentMidiPitch )
{
m_lastMidiPitch = instrumentTrack()->midiPitch();
m_lastMidiPitch = currentMidiPitch;
fluid_synth_pitch_bend( m_synth, m_channel, m_lastMidiPitch );
}
const int currentMidiPitchRange = instrumentTrack()->midiPitchRange();
if( m_lastMidiPitchRange != currentMidiPitchRange )
{
m_lastMidiPitchRange = currentMidiPitchRange;
fluid_synth_pitch_wheel_sens( m_synth, m_channel, m_lastMidiPitchRange );
}
if( m_internalSampleRate < engine::mixer()->processingSampleRate() &&
m_srcState != NULL )
{

1
plugins/sf2_player/sf2_player.h
View File

@@ -129,6 +129,7 @@ private:
int m_notesRunning[128];
sample_rate_t m_internalSampleRate;
int m_lastMidiPitch;
int m_lastMidiPitchRange;
int m_channel;
lcdSpinBoxModel m_bankNum;

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8
plugins/vestige/vestige.cpp
View File

@@ -310,13 +310,12 @@ void vestigeInstrument::play( sampleFrame * _buf )
bool vestigeInstrument::handleMidiEvent( const midiEvent & _me,
const midiTime & _time )
bool vestigeInstrument::handleMidiEvent( const MidiEvent& event, const MidiTime& time )
{
m_pluginMutex.lock();
if( m_plugin != NULL )
{
m_plugin->processMidiEvent( _me, _time );
m_plugin->processMidiEvent( event, time );
}
m_pluginMutex.unlock();
@@ -779,8 +778,7 @@ void VestigeInstrumentView::noteOffAll( void )
{
for( int key = 0; key <= MidiMaxNote; ++key )
{
m_vi->m_plugin->processMidiEvent(
midiEvent( MidiNoteOff, 0, key, 0 ), 0 );
m_vi->m_plugin->processMidiEvent( MidiEvent( MidiNoteOff, 0, key, 0 ), 0 );
}
}
m_vi->m_pluginMutex.unlock();

6
plugins/vestige/vestige.h
View File

@@ -1,7 +1,7 @@
/*
* vestige.h - instrument VeSTige for hosting VST-plugins
*
* Copyright (c) 2005-2012 Tobias Doerffel <tobydox/at/users.sourceforge.net>
* Copyright (c) 2005-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
*
@@ -34,7 +34,6 @@
#include "Instrument.h"
#include "InstrumentView.h"
#include "midi.h"
#include "note.h"
#include "knob.h"
@@ -69,8 +68,7 @@ public:
return true;
}
virtual bool handleMidiEvent( const midiEvent & _me,
const midiTime & _time );
virtual bool handleMidiEvent( const MidiEvent& event, const MidiTime& time );
virtual PluginView * instantiateView( QWidget * _parent );

59
plugins/vst_base/RemoteVstPlugin.cpp
View File

@@ -89,7 +89,7 @@ struct ERect
#include "lmms_basics.h"
#include "midi.h"
#include "Midi.h"
#include "communication.h"
#include "VST_sync_shm.h"
@@ -132,8 +132,7 @@ public:
virtual void process( const sampleFrame * _in, sampleFrame * _out );
virtual void processMidiEvent( const midiEvent & _event,
const f_cnt_t _offset );
virtual void processMidiEvent( const MidiEvent& event, const f_cnt_t offset );
// set given sample-rate for plugin
virtual void updateSampleRate()
@@ -774,16 +773,15 @@ void RemoteVstPlugin::process( const sampleFrame * _in, sampleFrame * _out )
// dispatcher-call, so we create static copies of the
// data and post them
#define MIDI_EVENT_BUFFER_COUNT 1024
static char event_buf[sizeof( VstMidiEvent * ) *
MIDI_EVENT_BUFFER_COUNT +
sizeof( VstEvents )];
static char eventsBuffer[sizeof( VstEvents ) + sizeof( VstMidiEvent * ) * MIDI_EVENT_BUFFER_COUNT];
static VstMidiEvent vme[MIDI_EVENT_BUFFER_COUNT];
VstEvents * events = (VstEvents *) event_buf;
VstEvents* events = (VstEvents *) eventsBuffer;
events->reserved = 0;
events->numEvents = m_midiEvents.size();
int idx = 0;
for( VstMidiEventList::iterator it = m_midiEvents.begin();
it != m_midiEvents.end(); ++it, ++idx )
for( VstMidiEventList::iterator it = m_midiEvents.begin(); it != m_midiEvents.end(); ++it, ++idx )
{
memcpy( &vme[idx], &*it, sizeof( VstMidiEvent ) );
events->events[idx] = (VstEvent *) &vme[idx];
@@ -831,36 +829,37 @@ void RemoteVstPlugin::process( const sampleFrame * _in, sampleFrame * _out )
void RemoteVstPlugin::processMidiEvent( const midiEvent & _event,
const f_cnt_t _offset )
void RemoteVstPlugin::processMidiEvent( const MidiEvent& event, const f_cnt_t offset )
{
VstMidiEvent event;
VstMidiEvent vme;
event.type = kVstMidiType;
event.byteSize = 24;
event.deltaFrames = _offset;
event.flags = 0;
event.detune = 0;
event.noteLength = 0;
event.noteOffset = 0;
event.noteOffVelocity = 0;
event.reserved1 = 0;
event.reserved2 = 0;
event.midiData[0] = _event.m_type + _event.m_channel;
switch( _event.m_type )
vme.type = kVstMidiType;
vme.byteSize = 24;
vme.deltaFrames = offset;
vme.flags = 0;
vme.detune = 0;
vme.noteLength = 0;
vme.noteOffset = 0;
vme.noteOffVelocity = 0;
vme.reserved1 = 0;
vme.reserved2 = 0;
vme.midiData[0] = event.type() + event.channel();
switch( event.type() )
{
case MidiPitchBend:
event.midiData[1] = _event.m_data.m_param[0] & 0x7f;
event.midiData[2] = _event.m_data.m_param[0] >> 7;
vme.midiData[1] = event.param( 0 ) & 0x7f;
vme.midiData[2] = event.param( 1 ) >> 7;
break;
// TODO: handle more special cases
default:
event.midiData[1] = _event.key();
event.midiData[2] = _event.velocity();
vme.midiData[1] = event.key();
vme.midiData[2] = event.velocity();
break;
}
event.midiData[3] = 0;
m_midiEvents.push_back( event );
vme.midiData[3] = 0;
m_midiEvents.push_back( vme );
}

32
plugins/zynaddsubfx/LocalZynAddSubFx.cpp
View File

@@ -1,7 +1,7 @@
/*
* LocalZynAddSubFx.cpp - local implementation of ZynAddSubFx plugin
*
* Copyright (c) 2009-2013 Tobias Doerffel <tobydox/at/users.sourceforge.net>
* Copyright (c) 2009-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
*
@@ -190,47 +190,43 @@ void LocalZynAddSubFx::setLmmsWorkingDir( const std::string & _dir )
void LocalZynAddSubFx::processMidiEvent( const midiEvent & _e )
void LocalZynAddSubFx::processMidiEvent( const MidiEvent& event )
{
// all functions are called while m_master->mutex is held
static NULLMidiIn midiIn;
switch( _e.m_type )
switch( event.type() )
{
case MidiNoteOn:
if( _e.velocity() > 0 )
if( event.velocity() > 0 )
{
if( _e.key() <= 0 || _e.key() >= 128 )
if( event.key() <= 0 || event.key() >= 128 )
{
break;
}
if( m_runningNotes[_e.key()] > 0 )
if( m_runningNotes[event.key()] > 0 )
{
m_master->NoteOff( _e.channel(), _e.key() );
m_master->NoteOff( event.channel(), event.key() );
}
++m_runningNotes[_e.key()];
m_master->NoteOn( _e.channel(), _e.key(), _e.velocity() );
++m_runningNotes[event.key()];
m_master->NoteOn( event.channel(), event.key(), event.velocity() );
break;
}
case MidiNoteOff:
if( _e.key() <= 0 || _e.key() >= 128 )
if( event.key() <= 0 || event.key() >= 128 )
{
break;
}
if( --m_runningNotes[_e.key()] <= 0 )
if( --m_runningNotes[event.key()] <= 0 )
{
m_master->NoteOff( _e.channel(), _e.key() );
m_master->NoteOff( event.channel(), event.key() );
}
break;
case MidiPitchBend:
m_master->SetController( _e.channel(), C_pitchwheel,
_e.m_data.m_param[0] +
_e.m_data.m_param[1]*128-8192 );
m_master->SetController( event.channel(), C_pitchwheel, event.pitchBend()-8192 );
break;
case MidiControlChange:
m_master->SetController( _e.channel(),
midiIn.getcontroller( _e.m_data.m_param[0] ),
_e.m_data.m_param[1] );
m_master->SetController( event.channel(), midiIn.getcontroller( event.controllerNumber() ), event.controllerValue() );
break;
default:
break;

5
plugins/zynaddsubfx/LocalZynAddSubFx.h
View File

@@ -1,7 +1,7 @@
/*
* LocalZynAddSubFx.h - local implementation of ZynAddSubFx plugin
*
* Copyright (c) 2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
* Copyright (c) 2009-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
*
@@ -25,6 +25,7 @@
#ifndef _LOCAL_ZYNADDSUBFX_H
#define _LOCAL_ZYNADDSUBFX_H
#include "MidiEvent.h"
#include "note.h"
class Master;
@@ -48,7 +49,7 @@ public:
void setPresetDir( const std::string & _dir );
void setLmmsWorkingDir( const std::string & _dir );
void processMidiEvent( const midiEvent & _e );
void processMidiEvent( const MidiEvent& event );
void processAudio( sampleFrame * _out );

7
plugins/zynaddsubfx/RemoteZynAddSubFx.cpp
View File

@@ -1,7 +1,7 @@
/*
* RemoteZynAddSubFx.cpp - ZynAddSubFx-embedding plugin
*
* Copyright (c) 2008-2012 Tobias Doerffel <tobydox/at/users.sourceforge.net>
* Copyright (c) 2008-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
*
@@ -126,10 +126,9 @@ public:
}
// all functions are called while m_master->mutex is held
virtual void processMidiEvent( const midiEvent & _e,
const f_cnt_t /* _offset */ )
virtual void processMidiEvent( const MidiEvent& event, const f_cnt_t /* _offset */ )
{
LocalZynAddSubFx::processMidiEvent( _e );
LocalZynAddSubFx::processMidiEvent( event );
}

16
plugins/zynaddsubfx/ZynAddSubFx.cpp
View File

@@ -340,14 +340,13 @@ void ZynAddSubFxInstrument::play( sampleFrame * _buf )
bool ZynAddSubFxInstrument::handleMidiEvent( const midiEvent & _me,
const midiTime & _time )
bool ZynAddSubFxInstrument::handleMidiEvent( const MidiEvent& event, const MidiTime& time )
{
// do not forward external MIDI Control Change events if the according
// LED is not checked
if( _me.type() == MidiControlChange &&
_me.sourcePort() != this &&
m_forwardMidiCcModel.value() == false )
if( event.type() == MidiControlChange &&
event.sourcePort() != this &&
m_forwardMidiCcModel.value() == false )
{
return true;
}
@@ -355,11 +354,11 @@ bool ZynAddSubFxInstrument::handleMidiEvent( const midiEvent & _me,
m_pluginMutex.lock();
if( m_remotePlugin )
{
m_remotePlugin->processMidiEvent( _me, 0 );
m_remotePlugin->processMidiEvent( event, 0 );
}
else
{
m_plugin->processMidiEvent( _me );
m_plugin->processMidiEvent( event );
}
m_pluginMutex.unlock();
@@ -446,8 +445,7 @@ void ZynAddSubFxInstrument::initPlugin()
void ZynAddSubFxInstrument::sendControlChange( MidiControllers midiCtl, float value )
{
handleMidiEvent( midiEvent( MidiControlChange, instrumentTrack()->midiPort()->realOutputChannel(), midiCtl, (int) value, this ),
midiTime() );
handleMidiEvent( MidiEvent( MidiControlChange, instrumentTrack()->midiPort()->realOutputChannel(), midiCtl, (int) value, this ) );
}

3
plugins/zynaddsubfx/ZynAddSubFx.h
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@@ -70,8 +70,7 @@ public:
virtual void play( sampleFrame * _working_buffer );
virtual bool handleMidiEvent( const midiEvent & _me,
const midiTime & _time );
virtual bool handleMidiEvent( const MidiEvent& event, const MidiTime& time = MidiTime() );
virtual void saveSettings( QDomDocument & _doc, QDomElement & _parent );
virtual void loadSettings( const QDomElement & _this );

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plugins/zynaddsubfx/artwork.png
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2
plugins/zynaddsubfx/src/Params/Controller.cpp
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@@ -35,7 +35,7 @@ Controller::~Controller()
void Controller::defaults()
{
setpitchwheelbendrange(200); //2 halftones
setpitchwheelbendrange(100); //2 halftones
expression.receive = 1;
panning.depth = 64;
filtercutoff.depth = 64;

5
plugins/zynaddsubfx/src/Seq/Sequencer.cpp
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@@ -157,10 +157,9 @@ void Sequencer::resettime(timestruct *t)
t->abs = 0.0;
t->rel = 0.0;
timeval tval;
t->last = 0.0;
#ifndef OS_WINDOWS
timeval tval;
if(gettimeofday(&tval, NULL) == 0)
t->last = tval.tv_sec + tval.tv_usec * 0.000001;
#endif
@@ -168,9 +167,9 @@ void Sequencer::resettime(timestruct *t)
void Sequencer::updatecounter(timestruct *t)
{
timeval tval;
double current = 0.0;
#ifndef OS_WINDOWS
timeval tval;
if(gettimeofday(&tval, NULL) == 0)
current = tval.tv_sec + tval.tv_usec * 0.000001;
#endif