/* Copyright (C) 1994-1995 Apogee Software, Ltd. 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 2 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, write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. */ /********************************************************************** module: AL_MIDI.C author: James R. Dose date: April 1, 1994 Low level routines to support General MIDI music on AdLib compatible cards. (c) Copyright 1994 James R. Dose. All Rights Reserved. **********************************************************************/ #include "driver_adlib.h" #include "_al_midi.h" #include "_multivc.h" #include "compat.h" #include "midi.h" #include "midifuncs.h" #include "opl3.h" enum { AdLibErr_Warning = -2, AdLibErr_Error = -1, AdLibErr_Ok = 0, }; static void AL_Shutdown(void); static int ErrorCode; int AdLibDrv_GetError(void) { return ErrorCode; } const char *AdLibDrv_ErrorString( int ErrorNumber ) { const char *ErrorString; switch( ErrorNumber ) { case AdLibErr_Warning : case AdLibErr_Error : ErrorString = AdLibDrv_ErrorString( ErrorCode ); break; case AdLibErr_Ok : ErrorString = "AdLib ok."; break; default: ErrorString = "Unknown AdLib error."; break; } return ErrorString; } int AdLibDrv_MIDI_Init(midifuncs *funcs) { AdLibDrv_MIDI_Shutdown(); Bmemset(funcs, 0, sizeof(midifuncs)); funcs->NoteOff = AL_NoteOff; funcs->NoteOn = AL_NoteOn; funcs->PolyAftertouch = nullptr; funcs->ControlChange = AL_ControlChange; funcs->ProgramChange = AL_ProgramChange; funcs->ChannelAftertouch = nullptr; funcs->PitchBend = AL_SetPitchBend; return AdLibErr_Ok; } void AdLibDrv_MIDI_HaltPlayback(void) { MV_UnhookMusicRoutine(); } void AdLibDrv_MIDI_Shutdown(void) { AdLibDrv_MIDI_HaltPlayback(); AL_Shutdown(); } int AdLibDrv_MIDI_StartPlayback(void (*service)(void)) { AdLibDrv_MIDI_HaltPlayback(); AL_Init(MV_MixRate); MV_HookMusicRoutine(service); return MIDI_Ok; } void AdLibDrv_MIDI_SetTempo(int tempo, int division) { MV_MIDIRenderTempo = tempo * division / 60; MV_MIDIRenderTimer = 0; } static opl3_chip chip; opl3_chip *AL_GetChip() { return &chip; } static constexpr unsigned int OctavePitch[MAX_OCTAVE+1] = { OCTAVE_0, OCTAVE_1, OCTAVE_2, OCTAVE_3, OCTAVE_4, OCTAVE_5, OCTAVE_6, OCTAVE_7, }; static unsigned int NoteMod12[MAX_NOTE+1]; static unsigned int NoteDiv12[MAX_NOTE+1]; // Pitch table //static unsigned NotePitch[ FINETUNE_MAX+1 ][ 12 ] = // { // { C, C_SHARP, D, D_SHARP, E, F, F_SHARP, G, G_SHARP, A, A_SHARP, B }, // }; static constexpr unsigned int NotePitch[FINETUNE_MAX+1][12] = { { 0x157, 0x16b, 0x181, 0x198, 0x1b0, 0x1ca, 0x1e5, 0x202, 0x220, 0x241, 0x263, 0x287 }, { 0x157, 0x16b, 0x181, 0x198, 0x1b0, 0x1ca, 0x1e5, 0x202, 0x220, 0x242, 0x264, 0x288 }, { 0x158, 0x16c, 0x182, 0x199, 0x1b1, 0x1cb, 0x1e6, 0x203, 0x221, 0x243, 0x265, 0x289 }, { 0x158, 0x16c, 0x183, 0x19a, 0x1b2, 0x1cc, 0x1e7, 0x204, 0x222, 0x244, 0x266, 0x28a }, { 0x159, 0x16d, 0x183, 0x19a, 0x1b3, 0x1cd, 0x1e8, 0x205, 0x223, 0x245, 0x267, 0x28b }, { 0x15a, 0x16e, 0x184, 0x19b, 0x1b3, 0x1ce, 0x1e9, 0x206, 0x224, 0x246, 0x268, 0x28c }, { 0x15a, 0x16e, 0x185, 0x19c, 0x1b4, 0x1ce, 0x1ea, 0x207, 0x225, 0x247, 0x269, 0x28e }, { 0x15b, 0x16f, 0x185, 0x19d, 0x1b5, 0x1cf, 0x1eb, 0x208, 0x226, 0x248, 0x26a, 0x28f }, { 0x15b, 0x170, 0x186, 0x19d, 0x1b6, 0x1d0, 0x1ec, 0x209, 0x227, 0x249, 0x26b, 0x290 }, { 0x15c, 0x170, 0x187, 0x19e, 0x1b7, 0x1d1, 0x1ec, 0x20a, 0x228, 0x24a, 0x26d, 0x291 }, { 0x15d, 0x171, 0x188, 0x19f, 0x1b7, 0x1d2, 0x1ed, 0x20b, 0x229, 0x24b, 0x26e, 0x292 }, { 0x15d, 0x172, 0x188, 0x1a0, 0x1b8, 0x1d3, 0x1ee, 0x20c, 0x22a, 0x24c, 0x26f, 0x293 }, { 0x15e, 0x172, 0x189, 0x1a0, 0x1b9, 0x1d4, 0x1ef, 0x20d, 0x22b, 0x24d, 0x270, 0x295 }, { 0x15f, 0x173, 0x18a, 0x1a1, 0x1ba, 0x1d4, 0x1f0, 0x20e, 0x22c, 0x24e, 0x271, 0x296 }, { 0x15f, 0x174, 0x18a, 0x1a2, 0x1bb, 0x1d5, 0x1f1, 0x20f, 0x22d, 0x24f, 0x272, 0x297 }, { 0x160, 0x174, 0x18b, 0x1a3, 0x1bb, 0x1d6, 0x1f2, 0x210, 0x22e, 0x250, 0x273, 0x298 }, { 0x161, 0x175, 0x18c, 0x1a3, 0x1bc, 0x1d7, 0x1f3, 0x211, 0x22f, 0x251, 0x274, 0x299 }, { 0x161, 0x176, 0x18c, 0x1a4, 0x1bd, 0x1d8, 0x1f4, 0x212, 0x230, 0x252, 0x276, 0x29b }, { 0x162, 0x176, 0x18d, 0x1a5, 0x1be, 0x1d9, 0x1f5, 0x212, 0x231, 0x254, 0x277, 0x29c }, { 0x162, 0x177, 0x18e, 0x1a6, 0x1bf, 0x1d9, 0x1f5, 0x213, 0x232, 0x255, 0x278, 0x29d }, { 0x163, 0x178, 0x18f, 0x1a6, 0x1bf, 0x1da, 0x1f6, 0x214, 0x233, 0x256, 0x279, 0x29e }, { 0x164, 0x179, 0x18f, 0x1a7, 0x1c0, 0x1db, 0x1f7, 0x215, 0x235, 0x257, 0x27a, 0x29f }, { 0x164, 0x179, 0x190, 0x1a8, 0x1c1, 0x1dc, 0x1f8, 0x216, 0x236, 0x258, 0x27b, 0x2a1 }, { 0x165, 0x17a, 0x191, 0x1a9, 0x1c2, 0x1dd, 0x1f9, 0x217, 0x237, 0x259, 0x27c, 0x2a2 }, { 0x166, 0x17b, 0x192, 0x1aa, 0x1c3, 0x1de, 0x1fa, 0x218, 0x238, 0x25a, 0x27e, 0x2a3 }, { 0x166, 0x17b, 0x192, 0x1aa, 0x1c3, 0x1df, 0x1fb, 0x219, 0x239, 0x25b, 0x27f, 0x2a4 }, { 0x167, 0x17c, 0x193, 0x1ab, 0x1c4, 0x1e0, 0x1fc, 0x21a, 0x23a, 0x25c, 0x280, 0x2a6 }, { 0x168, 0x17d, 0x194, 0x1ac, 0x1c5, 0x1e0, 0x1fd, 0x21b, 0x23b, 0x25d, 0x281, 0x2a7 }, { 0x168, 0x17d, 0x194, 0x1ad, 0x1c6, 0x1e1, 0x1fe, 0x21c, 0x23c, 0x25e, 0x282, 0x2a8 }, { 0x169, 0x17e, 0x195, 0x1ad, 0x1c7, 0x1e2, 0x1ff, 0x21d, 0x23d, 0x260, 0x283, 0x2a9 }, { 0x16a, 0x17f, 0x196, 0x1ae, 0x1c8, 0x1e3, 0x1ff, 0x21e, 0x23e, 0x261, 0x284, 0x2ab }, { 0x16a, 0x17f, 0x197, 0x1af, 0x1c8, 0x1e4, 0x200, 0x21f, 0x23f, 0x262, 0x286, 0x2ac } }; // Slot numbers as a function of the voice and the operator. // ( melodic only) static constexpr int slotVoice[NUMADLIBVOICES][2] = { { 0, 3 }, // voice 0 { 1, 4 }, // 1 { 2, 5 }, // 2 { 6, 9 }, // 3 { 7, 10 }, // 4 { 8, 11 }, // 5 { 12, 15 }, // 6 { 13, 16 }, // 7 { 14, 17 }, // 8 }; static int VoiceLevel[AL_NumChipSlots][2]; static int VoiceKsl[AL_NumChipSlots][2]; // This table gives the offset of each slot within the chip. // offset = fn( slot) static constexpr int8_t offsetSlot[AL_NumChipSlots] = { 0, 1, 2, 3, 4, 5, 8, 9, 10, 11, 12, 13, 16, 17, 18, 19, 20, 21 }; static int VoiceReserved[NUMADLIBVOICES * 2]; static AdLibVoice Voice[NUMADLIBVOICES * 2]; static AdLibVoiceList Voice_Pool; static AdLibChannel Channel[NUMADLIBCHANNELS]; static int AL_LeftPort = ADLIB_PORT; static int AL_RightPort = ADLIB_PORT; int AL_Stereo; static int AL_SendStereo; static int AL_MaxMidiChannel = 16; #define OFFSET(structure, offset) (*((char **)&(structure)[offset])) #define LL_AddToTail(type, listhead, node) \ LL_AddNode((char *)(node), (char **)&((listhead)->end), (char **)&((listhead)->start), (intptr_t) & ((type *)0)->prev, \ (intptr_t) & ((type *)0)->next) #define LL_Remove(type, listhead, node) \ LL_RemoveNode((char *)(node), (char **)&((listhead)->start), (char **)&((listhead)->end), (intptr_t) & ((type *)0)->next, \ (intptr_t) & ((type *)0)->prev) static void LL_RemoveNode(char *item, char **head, char **tail, intptr_t next, intptr_t prev) { if (OFFSET(item, prev) == nullptr) *head = OFFSET(item, next); else OFFSET(OFFSET(item, prev), next) = OFFSET(item, next); if (OFFSET(item, next) == nullptr) *tail = OFFSET(item, prev); else OFFSET(OFFSET(item, next), prev) = OFFSET(item, prev); OFFSET(item, next) = nullptr; OFFSET(item, prev) = nullptr; } static void LL_AddNode(char *item, char **head, char **tail, intptr_t next, intptr_t prev) { OFFSET(item, prev) = nullptr; OFFSET(item, next) = *head; if (*head) OFFSET(*head, prev) = item; else *tail = item; *head = item; } static void AL_SendOutputToPort(int port, int reg, int data) { OPL3_WriteRegBuffered(&chip, (Bit16u)(reg + ((port & 2) << 7)), (Bit8u)data); } static void AL_SendOutput(int voice, int reg, int data) { if (AL_SendStereo) { AL_SendOutputToPort(AL_LeftPort, reg, data); AL_SendOutputToPort(AL_RightPort, reg, data); } else { int port = (voice == 0) ? AL_RightPort : AL_LeftPort; AL_SendOutputToPort(port, reg, data); } } static void AL_SetVoiceTimbre(int voice) { int channel = Voice[voice].channel; int patch = (channel == 9) ? Voice[voice].key + 128 : Channel[channel].Timbre; if (Voice[voice].timbre == patch) return; Voice[voice].timbre = patch; auto timbre = &ADLIB_TimbreBank[patch]; int port = Voice[voice].port; int voc = (voice >= NUMADLIBVOICES) ? voice - NUMADLIBVOICES : voice; int slot = slotVoice[voc][0]; int off = offsetSlot[slot]; VoiceLevel[slot][port] = 63 - (timbre->Level[0] & 0x3f); VoiceKsl[slot][port] = timbre->Level[0] & 0xc0; AL_SendOutput(port, 0xA0 + voc, 0); AL_SendOutput(port, 0xB0 + voc, 0); // Let voice clear the release AL_SendOutput(port, 0x80 + off, 0xff); AL_SendOutput(port, 0x60 + off, timbre->Env1[0]); AL_SendOutput(port, 0x80 + off, timbre->Env2[0]); AL_SendOutput(port, 0x20 + off, timbre->SAVEK[0]); AL_SendOutput(port, 0xE0 + off, timbre->Wave[0]); AL_SendOutput(port, 0x40 + off, timbre->Level[0]); slot = slotVoice[voc][1]; if (AL_SendStereo) { AL_SendOutputToPort(AL_LeftPort, 0xC0 + voice, (timbre->Feedback & 0x0f) | 0x20); AL_SendOutputToPort(AL_RightPort, 0xC0 + voice, (timbre->Feedback & 0x0f) | 0x10); } else AL_SendOutput(port, 0xC0 + voc, (timbre->Feedback & 0x0f) | 0x30); off = offsetSlot[slot]; VoiceLevel[slot][port] = 63 - (timbre->Level[1] & 0x3f); VoiceKsl[slot][port] = timbre->Level[1] & 0xc0; AL_SendOutput(port, 0x40 + off, 63); // Let voice clear the release AL_SendOutput(port, 0x80 + off, 0xff); AL_SendOutput(port, 0x60 + off, timbre->Env1[1]); AL_SendOutput(port, 0x80 + off, timbre->Env2[1]); AL_SendOutput(port, 0x20 + off, timbre->SAVEK[1]); AL_SendOutput(port, 0xE0 + off, timbre->Wave[1]); } static void AL_SetVoiceVolume(int voice) { int channel = Voice[voice].channel; auto timbre = &ADLIB_TimbreBank[Voice[voice].timbre]; int velocity = Voice[voice].velocity + timbre->Velocity; velocity = min(velocity, MAX_VELOCITY); int const voc = (voice >= NUMADLIBVOICES) ? voice - NUMADLIBVOICES : voice; int const slot = slotVoice[voc][1]; int const port = Voice[voice].port; // amplitude uint32_t t1 = (unsigned int)VoiceLevel[slot][port] * (velocity + 0x80); t1 = (Channel[channel].Volume * t1) >> 15; if (!AL_SendStereo) { uint32_t volume = t1 ^ 63; volume |= (unsigned int)VoiceKsl[slot][port]; AL_SendOutput(port, 0x40 + offsetSlot[slot], volume); // Check if this timbre is Additive if (timbre->Feedback & 0x01) { int const slot = slotVoice[voc][0]; // amplitude uint32_t t2 = (unsigned int)VoiceLevel[slot][port] * (velocity + 0x80); t2 = (Channel[channel].Volume * t2) >> 15; volume = t2 ^ 63; volume |= (unsigned int)VoiceKsl[slot][port]; AL_SendOutput(port, 0x40 + offsetSlot[slot], volume); } return; } // Set left channel volume uint32_t volume = t1; if (Channel[channel].Pan < 64) { volume *= Channel[channel].Pan; volume >>= 6; } volume ^= 63; volume |= (unsigned int)VoiceKsl[slot][port]; AL_SendOutputToPort(AL_LeftPort, 0x40 + offsetSlot[slot], volume); // Set right channel volume volume = t1; if (Channel[channel].Pan > 64) { volume *= 127 - Channel[channel].Pan; volume >>= 6; } volume ^= 63; volume |= (unsigned int)VoiceKsl[slot][port]; AL_SendOutputToPort(AL_RightPort, 0x40 + offsetSlot[slot], volume); // Check if this timbre is Additive if (timbre->Feedback & 0x01) { // amplitude uint32_t t2 = (unsigned int)VoiceLevel[slot][port] * (velocity + 0x80); t2 = (Channel[channel].Volume * t2) >> 15; int const slot = slotVoice[voc][0]; // Set left channel volume volume = t2; if (Channel[channel].Pan < 64) { volume *= Channel[channel].Pan; volume >>= 6; } volume ^= 63; volume |= (unsigned int)VoiceKsl[slot][port]; AL_SendOutputToPort(AL_LeftPort, 0x40 + offsetSlot[slot], volume); // Set right channel volume volume = t2; if (Channel[channel].Pan > 64) { volume *= 127 - Channel[channel].Pan; volume >>= 6; } volume ^= 63; volume |= (unsigned int)VoiceKsl[slot][port]; AL_SendOutputToPort(AL_RightPort, 0x40 + offsetSlot[slot], volume); } } static int AL_AllocVoice(void) { if (Voice_Pool.start) { int voice = Voice_Pool.start->num; LL_Remove(AdLibVoice, &Voice_Pool, &Voice[voice]); return voice; } return AL_VoiceNotFound; } static int AL_GetVoice(int channel, int key) { auto voice = Channel[channel].Voices.start; while (voice != nullptr) { if (voice->key == (unsigned int)key) return voice->num; voice = voice->next; } return AL_VoiceNotFound; } static void AL_SetVoicePitch(int voice) { int port = Voice[voice].port; int voc = (voice >= NUMADLIBVOICES) ? voice - NUMADLIBVOICES : voice; int channel = Voice[voice].channel; int patch, note; if (channel == 9) { patch = Voice[voice].key + 128; note = ADLIB_TimbreBank[patch].Transpose; } else { patch = Channel[channel].Timbre; note = Voice[voice].key + ADLIB_TimbreBank[patch].Transpose; } note += Channel[channel].KeyOffset - 12; note = clamp(note, 0, MAX_NOTE); int detune = Channel[channel].KeyDetune; int ScaleNote = NoteMod12[note]; int Octave = NoteDiv12[note]; int pitch = OctavePitch[Octave] | NotePitch[detune][ScaleNote]; Voice[voice].pitchleft = pitch; pitch |= Voice[voice].status; if (!AL_SendStereo) { AL_SendOutput(port, 0xA0 + voc, pitch); AL_SendOutput(port, 0xB0 + voc, pitch >> 8); } else { AL_SendOutputToPort(AL_LeftPort, 0xA0 + voice, pitch); AL_SendOutputToPort(AL_LeftPort, 0xB0 + voice, pitch >> 8); if (channel != 9) detune += STEREO_DETUNE; if (detune > FINETUNE_MAX) { detune -= FINETUNE_RANGE; if (note < MAX_NOTE) { note++; ScaleNote = NoteMod12[note]; Octave = NoteDiv12[note]; } } pitch = OctavePitch[Octave] | NotePitch[detune][ScaleNote]; Voice[voice].pitchright = pitch; pitch |= Voice[voice].status; AL_SendOutputToPort(AL_RightPort, 0xA0 + voice, pitch); AL_SendOutputToPort(AL_RightPort, 0xB0 + voice, pitch >> 8); } } static void AL_SetChannelVolume(int channel, int volume) { volume = max(0, volume); volume = min(volume, AL_MaxVolume); Channel[channel].Volume = volume; auto voice = Channel[channel].Voices.start; while (voice != nullptr) { AL_SetVoiceVolume(voice->num); voice = voice->next; } } static void AL_SetChannelPan(int channel, int pan) { // Don't pan drum sounds if (channel != 9) Channel[channel].Pan = pan; } static void AL_SetChannelDetune(int channel, int detune) { Channel[channel].Detune = detune; } static void AL_ResetVoices(void) { Voice_Pool.start = nullptr; Voice_Pool.end = nullptr; int numvoices = NUMADLIBVOICES; if (!AL_Stereo) numvoices = NUMADLIBVOICES * 2; for (int index = 0; index < numvoices; index++) { if (VoiceReserved[index] == FALSE) { Voice[index].num = index; Voice[index].key = 0; Voice[index].velocity = 0; Voice[index].channel = -1; Voice[index].timbre = -1; Voice[index].port = (index < NUMADLIBVOICES) ? 0 : 1; Voice[index].status = NOTE_OFF; LL_AddToTail(AdLibVoice, &Voice_Pool, &Voice[index]); } } for (int index = 0; index < NUMADLIBCHANNELS; index++) { Channel[index].Voices.start = nullptr; Channel[index].Voices.end = nullptr; Channel[index].Timbre = 0; Channel[index].Pitchbend = 0; Channel[index].KeyOffset = 0; Channel[index].KeyDetune = 0; Channel[index].Volume = AL_DefaultChannelVolume; Channel[index].Pan = 64; Channel[index].RPN = 0; Channel[index].PitchBendRange = AL_DefaultPitchBendRange; Channel[index].PitchBendSemiTones = AL_DefaultPitchBendRange / 100; Channel[index].PitchBendHundreds = AL_DefaultPitchBendRange % 100; } } static void AL_CalcPitchInfo(void) { // int finetune; // double detune; for (int note = 0; note <= MAX_NOTE; note++) { NoteMod12[note] = note % 12; NoteDiv12[note] = note / 12; } // for( finetune = 1; finetune <= FINETUNE_MAX; finetune++ ) // { // detune = pow( 2, ( double )finetune / ( 12.0 * FINETUNE_RANGE ) ); // for( note = 0; note < 12; note++ ) // { // NotePitch[ finetune ][ note ] = ( ( double )NotePitch[ 0 ][ note ] * detune ); // } // } } static void AL_FlushCard(int port) { for (int i = 0; i < NUMADLIBVOICES; i++) { if (VoiceReserved[i] == FALSE) { unsigned slot1 = offsetSlot[slotVoice[i][0]]; unsigned slot2 = offsetSlot[slotVoice[i][1]]; AL_SendOutputToPort(port, 0xA0 + i, 0); AL_SendOutputToPort(port, 0xB0 + i, 0); AL_SendOutputToPort(port, 0xE0 + slot1, 0); AL_SendOutputToPort(port, 0xE0 + slot2, 0); // Set the envelope to be fast and quiet AL_SendOutputToPort(port, 0x60 + slot1, 0xff); AL_SendOutputToPort(port, 0x60 + slot2, 0xff); AL_SendOutputToPort(port, 0x80 + slot1, 0xff); AL_SendOutputToPort(port, 0x80 + slot2, 0xff); // Maximum attenuation AL_SendOutputToPort(port, 0x40 + slot1, 0xff); AL_SendOutputToPort(port, 0x40 + slot2, 0xff); } } } static void AL_StereoOn(void) { if (AL_Stereo && !AL_SendStereo) AL_SendStereo = TRUE; AL_SendOutputToPort(AL_RightPort, 0x5, 1); } static void AL_StereoOff(void) { if (AL_Stereo && AL_SendStereo) AL_SendStereo = FALSE; AL_SendOutputToPort(AL_RightPort, 0x5, 0); } static void AL_Reset(void) { AL_SendOutputToPort(ADLIB_PORT, 1, 0x20); AL_SendOutputToPort(ADLIB_PORT, 0x08, 0); // Set the values: AM Depth, VIB depth & Rhythm AL_SendOutputToPort(ADLIB_PORT, 0xBD, 0); AL_StereoOn(); AL_FlushCard(AL_LeftPort); AL_FlushCard(AL_RightPort); } static void AL_NoteOff(int channel, int key, int velocity) { UNREFERENCED_PARAMETER(velocity); // We only play channels 1 through 10 if (channel > AL_MaxMidiChannel) return; int voice = AL_GetVoice(channel, key); if (voice == AL_VoiceNotFound) return; Voice[voice].status = NOTE_OFF; int port = Voice[voice].port; int voc = (voice >= NUMADLIBVOICES) ? voice - NUMADLIBVOICES : voice; if (AL_SendStereo) { AL_SendOutputToPort(AL_LeftPort, 0xB0 + voice, hibyte(Voice[voice].pitchleft)); AL_SendOutputToPort(AL_RightPort, 0xB0 + voice, hibyte(Voice[voice].pitchright)); } else AL_SendOutput(port, 0xB0 + voc, hibyte(Voice[voice].pitchleft)); LL_Remove(AdLibVoice, &Channel[channel].Voices, &Voice[voice]); LL_AddToTail(AdLibVoice, &Voice_Pool, &Voice[voice]); } static void AL_NoteOn(int channel, int key, int velocity) { // We only play channels 1 through 10 if (channel > AL_MaxMidiChannel) return; if (velocity == 0) { AL_NoteOff(channel, key, velocity); return; } int voice = AL_AllocVoice(); if (voice == AL_VoiceNotFound) { if (Channel[9].Voices.start) { AL_NoteOff(9, Channel[9].Voices.start->key, 0); voice = AL_AllocVoice(); } if (voice == AL_VoiceNotFound) return; } Voice[voice].key = key; Voice[voice].channel = channel; Voice[voice].velocity = velocity; Voice[voice].status = NOTE_ON; LL_AddToTail(AdLibVoice, &Channel[channel].Voices, &Voice[voice]); AL_SetVoiceTimbre(voice); AL_SetVoiceVolume(voice); AL_SetVoicePitch(voice); } static inline void AL_AllNotesOff(int channel) { while (Channel[channel].Voices.start != nullptr) AL_NoteOff(channel, Channel[channel].Voices.start->key, 0); } static void AL_ControlChange(int channel, int type, int data) { // We only play channels 1 through 10 if (channel > AL_MaxMidiChannel) return; switch (type) { case MIDI_VOLUME: AL_SetChannelVolume(channel, data); break; case MIDI_PAN: AL_SetChannelPan(channel, data); break; case MIDI_DETUNE: AL_SetChannelDetune(channel, data); break; case MIDI_ALL_NOTES_OFF: AL_AllNotesOff(channel); break; case MIDI_RESET_ALL_CONTROLLERS: AL_ResetVoices(); AL_SetChannelVolume(channel, AL_DefaultChannelVolume); AL_SetChannelPan(channel, 64); AL_SetChannelDetune(channel, 0); break; case MIDI_RPN_MSB: Channel[channel].RPN &= 0x00FF; Channel[channel].RPN |= (data & 0xFF) << 8; break; case MIDI_RPN_LSB: Channel[channel].RPN &= 0xFF00; Channel[channel].RPN |= data & 0xFF; break; case MIDI_DATAENTRY_MSB: if (Channel[channel].RPN == MIDI_PITCHBEND_RPN) { Channel[channel].PitchBendSemiTones = data; Channel[channel].PitchBendRange = Channel[channel].PitchBendSemiTones * 100 + Channel[channel].PitchBendHundreds; } break; case MIDI_DATAENTRY_LSB: if (Channel[channel].RPN == MIDI_PITCHBEND_RPN) { Channel[channel].PitchBendHundreds = data; Channel[channel].PitchBendRange = Channel[channel].PitchBendSemiTones * 100 + Channel[channel].PitchBendHundreds; } break; } } static void AL_ProgramChange(int channel, int patch) { // We only play channels 1 through 10 if (channel > AL_MaxMidiChannel) return; Channel[channel].Timbre = patch; } static void AL_SetPitchBend(int channel, int lsb, int msb) { // We only play channels 1 through 10 if (channel > AL_MaxMidiChannel) return; int const pitchbend = lsb + (msb << 8); Channel[channel].Pitchbend = pitchbend; int TotalBend = pitchbend * Channel[channel].PitchBendRange; TotalBend /= (PITCHBEND_CENTER / FINETUNE_RANGE); Channel[channel].KeyOffset = (int)(TotalBend / FINETUNE_RANGE); Channel[channel].KeyOffset -= Channel[channel].PitchBendSemiTones; Channel[channel].KeyDetune = (unsigned int)(TotalBend % FINETUNE_RANGE); auto voice = Channel[channel].Voices.start; while (voice != nullptr) { AL_SetVoicePitch(voice->num); voice = voice->next; } } static void AL_Shutdown(void) { AL_StereoOff(); AL_ResetVoices(); AL_Reset(); } static int AL_Init(int rate) { OPL3_Reset(&chip, rate); AL_Stereo = TRUE; AL_LeftPort = ADLIB_PORT; AL_RightPort = ADLIB_PORT + 2; AL_CalcPitchInfo(); AL_Reset(); AL_ResetVoices(); return AdLibErr_Ok; } void AL_RegisterTimbreBank(uint8_t *timbres) { for (int i = 0; i < 256; i++) { ADLIB_TimbreBank[i].SAVEK[0] = *(timbres++); ADLIB_TimbreBank[i].SAVEK[1] = *(timbres++); ADLIB_TimbreBank[i].Level[0] = *(timbres++); ADLIB_TimbreBank[i].Level[1] = *(timbres++); ADLIB_TimbreBank[i].Env1[0] = *(timbres++); ADLIB_TimbreBank[i].Env1[1] = *(timbres++); ADLIB_TimbreBank[i].Env2[0] = *(timbres++); ADLIB_TimbreBank[i].Env2[1] = *(timbres++); ADLIB_TimbreBank[i].Wave[0] = *(timbres++); ADLIB_TimbreBank[i].Wave[1] = *(timbres++); ADLIB_TimbreBank[i].Feedback = *(timbres++); ADLIB_TimbreBank[i].Transpose = *(int8_t *)(timbres++); ADLIB_TimbreBank[i].Velocity = *(int8_t *)(timbres++); } }