ref: a41d051499b5be820f4eb2a96efb8019bd1fcd53
parent: 9c44e4133323d462ed11d537a09cb9efa3c77803
author: Iliyas Jorio <[email protected]>
date: Wed Jul 29 18:42:55 EDT 2020
mod/s3m playback with ibxm
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -13,6 +13,10 @@
add_executable(Candy_Crisis
src/support/cmixer.cpp
src/support/cmixer.h
+ src/support/ibxm.c
+ src/support/ibxm.h
+ src/support/ModStream.cpp
+ src/support/ModStream.h
src/blitter.cpp
src/blitter.h
src/CandyCrisis.cpp
--- a/src/music.cpp
+++ b/src/music.cpp
@@ -3,6 +3,8 @@
#include "stdafx.h"
#include <string.h>
+#include <vector>
+#include <fstream>
#include "main.h"
#include "music.h"
@@ -11,10 +13,7 @@
#include "soundfx.h"
#include "graphics.h"
-#if 0
-#include "fmod.hpp"
-#include "fmod_errors.h"
-#endif
+#include "support/ModStream.h"
const int k_noMusic = -1;
const int k_songs = 14;
@@ -24,112 +23,92 @@
static MBoolean s_musicFast = false;
int s_musicPaused = 0;
-#if 0
-static FMOD::Channel* s_musicChannel = NULL;
-static FMOD::Sound* s_musicModule = NULL;
-#endif
+static cmixer::ModStream* s_musicChannel = NULL;
+
void EnableMusic( MBoolean on )
{-#if 0
if (s_musicChannel)
{- FMOD_RESULT result = s_musicChannel->setVolume(on? 0.75f: 0.0f);
- FMOD_ERRCHECK(result);
+ s_musicChannel->SetGain(on? 0.75: 0.0);
}
-#endif
}
void FastMusic( void )
{-#if 0
- if (s_musicModule && !s_musicFast)
+ if (s_musicChannel && !s_musicFast)
{- FMOD_RESULT result = s_musicModule->setMusicSpeed(1.3f);
- FMOD_ERRCHECK(result);
-
+ printf("Implement Me! FastMusic (1.3x playback speed)\n");s_musicFast = true;
}
-#endif
}
void SlowMusic( void )
{-#if 0
- if (s_musicModule && s_musicFast)
+ if (s_musicChannel && s_musicFast)
{- FMOD_RESULT result = s_musicModule->setMusicSpeed(1.0f);
- FMOD_ERRCHECK(result);
-
+ printf("Implement Me! SlowMusic\n");s_musicFast = false;
}
-#endif
}
void PauseMusic( void )
{-#if 0
if (s_musicChannel)
{- FMOD_RESULT result = s_musicChannel->setPaused(true);
- FMOD_ERRCHECK(result);
-
+ s_musicChannel->Pause();
s_musicPaused++;
}
-#endif
}
void ResumeMusic( void )
{-#if 0
if (s_musicChannel)
{- FMOD_RESULT result = s_musicChannel->setPaused(false);
- FMOD_ERRCHECK(result);
-
+ s_musicChannel->Play();
s_musicPaused--;
}
-#endif
}
+static std::vector<char> LoadFile(char const* filename)
+{+ std::ifstream ifs(filename, std::ios::binary | std::ios::ate);
+ auto pos = ifs.tellg();
+ std::vector<char> bytes(pos);
+ ifs.seekg(0, std::ios::beg);
+ ifs.read(&bytes[0], pos);
+ return bytes;
+}
+
void ChooseMusic( short which )
{-#if 0
if (s_musicChannel != NULL)
{- s_musicChannel->stop();
+ delete s_musicChannel;
s_musicChannel = NULL;
}
-
- if (s_musicModule != NULL)
- {- s_musicModule->release();
- s_musicModule = NULL;
- }
-
+
musicSelection = -1;
if (which >= 0 && which <= k_songs)
{- FMOD_RESULT result = g_fmod->createSound(QuickResourceName("mod", which+128, ""), FMOD_DEFAULT, 0, &s_musicModule);- FMOD_ERRCHECK(result);
+ printf("Music: %d\n" , which + 128);- result = g_fmod->playSound(FMOD_CHANNEL_FREE, s_musicModule, true, &s_musicChannel);
- FMOD_ERRCHECK(result);
-
- result = s_musicChannel->setPriority(10); // prioritize music first--WAVs should never knock out a MOD
- FMOD_ERRCHECK(result);
-
+ auto qrn = QuickResourceName("mod", which+128, ".mod");+ if (!FileExists(qrn)) {+ qrn = QuickResourceName("mod", which+128, ".s3m");+ }
+ if (!FileExists(qrn)) {+ return;
+ }
+ auto rawFileData = LoadFile(qrn);
+
+ s_musicChannel = new cmixer::ModStream(LoadFile(qrn));
+
EnableMusic(musicOn);
-
- result = s_musicModule->setLoopCount(-1);
- FMOD_ERRCHECK(result);
-
- result = s_musicChannel->setPaused(false);
- FMOD_ERRCHECK(result);
-
+ s_musicChannel->Play();
+
musicSelection = which;
s_musicPaused = 0;
}
-#endif
}
--- /dev/null
+++ b/src/support/ModStream.cpp
@@ -1,0 +1,59 @@
+#include <climits>
+#include <cstdio>
+#include "ModStream.h"
+
+using namespace cmixer;
+
+ModStream::ModStream(std::vector<char> &&rawModuleData)
+ : Source(44100, INT_MAX)
+ , moduleFile(rawModuleData)
+ , replayBuffer(2048*8)
+ , rbOffset(0)
+ , rbLength(0)
+{+ ibxm::data d;
+ d.buffer = moduleFile.data();
+ d.length = moduleFile.size();
+ char errors[256];
+ errors[0] = '\0';
+ this->module = ibxm::module_load(&d, errors);
+ this->replay = ibxm::new_replay(this->module, 44100, 0);
+ //printf("%p IBXM Error: %s\n", this->module, errors);+}
+
+void ModStream::Rewind2()
+{+ printf("Rewind not supported\n");+}
+
+void ModStream::FillBuffer(int16_t *output, int length)
+{+ length /= 2;
+
+ while (length > 0) {+ // refill replay buffer if exhausted
+ if (rbLength == 0) {+ rbOffset = 0;
+ rbLength = ibxm::replay_get_audio(replay, replayBuffer.data(), 0);
+ }
+
+ // number of stereo samples to copy from replay buffer to output buffer
+ int nToCopy = std::min(rbLength, length);
+
+ int *input = &replayBuffer[rbOffset * 2];
+
+ // Copy samples
+ for (int i = 0; i < nToCopy * 2; i++) {+ int sample = *(input++);
+
+ if (sample < -32768) sample = -32768;
+ if (sample > 32767) sample = 32767;
+
+ *(output++) = sample;
+ }
+
+ rbOffset += nToCopy;
+ rbLength -= nToCopy;
+ length -= nToCopy;
+ }
+}
--- /dev/null
+++ b/src/support/ModStream.h
@@ -1,0 +1,26 @@
+#pragma once
+
+#include "cmixer.h"
+
+namespace ibxm {+extern "C" {+#include "ibxm.h"
+}
+}
+
+namespace cmixer {+class ModStream : public Source {+ ibxm::module* module;
+ ibxm::replay* replay;
+ std::vector<char> moduleFile;
+ std::vector<int> replayBuffer;
+ int rbOffset;
+ int rbLength;
+
+ void Rewind2();
+ void FillBuffer(int16_t* buffer, int length);
+
+public:
+ ModStream(std::vector<char>&& rawModule);
+};
+}
\ No newline at end of file
--- /dev/null
+++ b/src/support/ibxm.c
@@ -1,0 +1,1947 @@
+
+#include "stdlib.h"
+#include "string.h"
+
+#include "ibxm.h"
+
+const char *IBXM_VERSION = "ibxm/ac mod/xm/s3m replay 20191214 (c)[email protected]";
+
+static const int FP_SHIFT = 15, FP_ONE = 32768, FP_MASK = 32767;
+
+static const int exp2_table[] = {+ 32768, 32946, 33125, 33305, 33486, 33667, 33850, 34034,
+ 34219, 34405, 34591, 34779, 34968, 35158, 35349, 35541,
+ 35734, 35928, 36123, 36319, 36516, 36715, 36914, 37114,
+ 37316, 37518, 37722, 37927, 38133, 38340, 38548, 38757,
+ 38968, 39180, 39392, 39606, 39821, 40037, 40255, 40473,
+ 40693, 40914, 41136, 41360, 41584, 41810, 42037, 42265,
+ 42495, 42726, 42958, 43191, 43425, 43661, 43898, 44137,
+ 44376, 44617, 44859, 45103, 45348, 45594, 45842, 46091,
+ 46341, 46593, 46846, 47100, 47356, 47613, 47871, 48131,
+ 48393, 48655, 48920, 49185, 49452, 49721, 49991, 50262,
+ 50535, 50810, 51085, 51363, 51642, 51922, 52204, 52488,
+ 52773, 53059, 53347, 53637, 53928, 54221, 54515, 54811,
+ 55109, 55408, 55709, 56012, 56316, 56622, 56929, 57238,
+ 57549, 57861, 58176, 58491, 58809, 59128, 59449, 59772,
+ 60097, 60423, 60751, 61081, 61413, 61746, 62081, 62419,
+ 62757, 63098, 63441, 63785, 64132, 64480, 64830, 65182,
+ 65536
+};
+
+static const short sine_table[] = {+ 0, 24, 49, 74, 97, 120, 141, 161, 180, 197, 212, 224, 235, 244, 250, 253,
+ 255, 253, 250, 244, 235, 224, 212, 197, 180, 161, 141, 120, 97, 74, 49, 24
+};
+
+struct note {+ unsigned char key, instrument, volume, effect, param;
+};
+
+struct channel {+ struct replay *replay;
+ struct instrument *instrument;
+ struct sample *sample;
+ struct note note;
+ int id, key_on, random_seed, pl_row;
+ int sample_off, sample_idx, sample_fra, freq, ampl, pann;
+ int volume, panning, fadeout_vol, vol_env_tick, pan_env_tick;
+ int period, porta_period, retrig_count, fx_count, av_count;
+ int porta_up_param, porta_down_param, tone_porta_param, offset_param;
+ int fine_porta_up_param, fine_porta_down_param, xfine_porta_param;
+ int arpeggio_param, vol_slide_param, gvol_slide_param, pan_slide_param;
+ int fine_vslide_up_param, fine_vslide_down_param;
+ int retrig_volume, retrig_ticks, tremor_on_ticks, tremor_off_ticks;
+ int vibrato_type, vibrato_phase, vibrato_speed, vibrato_depth;
+ int tremolo_type, tremolo_phase, tremolo_speed, tremolo_depth;
+ int tremolo_add, vibrato_add, arpeggio_add;
+};
+
+struct replay {+ int sample_rate, interpolation, global_vol;
+ int seq_pos, break_pos, row, next_row, tick;
+ int speed, tempo, pl_count, pl_chan;
+ int *ramp_buf;
+ char **play_count;
+ struct channel *channels;
+ struct module *module;
+};
+
+static int exp_2( int x ) {+ int c, m, y;
+ int x0 = ( x & FP_MASK ) >> ( FP_SHIFT - 7 );
+ c = exp2_table[ x0 ];
+ m = exp2_table[ x0 + 1 ] - c;
+ y = ( m * ( x & ( FP_MASK >> 7 ) ) >> 8 ) + c;
+ return ( y << FP_SHIFT ) >> ( FP_SHIFT - ( x >> FP_SHIFT ) );
+}
+
+static int log_2( int x ) {+ int step;
+ int y = 16 << FP_SHIFT;
+ for( step = y; step > 0; step >>= 1 ) {+ if( exp_2( y - step ) >= x ) {+ y -= step;
+ }
+ }
+ return y;
+}
+
+static char* data_ascii( struct data *data, int offset, int length, char *dest ) {+ int idx, chr;
+ memset( dest, 32, length );
+ if( offset > data->length ) {+ offset = data->length;
+ }
+ if( ( unsigned int ) offset + length > ( unsigned int ) data->length ) {+ length = data->length - offset;
+ }
+ for( idx = 0; idx < length; idx++ ) {+ chr = data->buffer[ offset + idx ] & 0xFF;
+ if( chr > 32 ) {+ dest[ idx ] = chr;
+ }
+ }
+ return dest;
+}
+
+static int data_s8( struct data *data, int offset ) {+ int value = 0;
+ if( offset < data->length ) {+ value = data->buffer[ offset ];
+ value = ( value & 0x7F ) - ( value & 0x80 );
+ }
+ return value;
+}
+
+static int data_u8( struct data *data, int offset ) {+ int value = 0;
+ if( offset < data->length ) {+ value = data->buffer[ offset ] & 0xFF;
+ }
+ return value;
+}
+
+static int data_u16be( struct data *data, int offset ) {+ int value = 0;
+ if( offset + 1 < data->length ) {+ value = ( ( data->buffer[ offset ] & 0xFF ) << 8 )
+ | ( data->buffer[ offset + 1 ] & 0xFF );
+ }
+ return value;
+}
+
+static int data_u16le( struct data *data, int offset ) {+ int value = 0;
+ if( offset + 1 < data->length ) {+ value = ( data->buffer[ offset ] & 0xFF )
+ | ( ( data->buffer[ offset + 1 ] & 0xFF ) << 8 );
+ }
+ return value;
+}
+
+static unsigned int data_u32le( struct data *data, int offset ) {+ unsigned int value = 0;
+ if( offset + 3 < data->length ) {+ value = ( data->buffer[ offset ] & 0xFF )
+ | ( ( data->buffer[ offset + 1 ] & 0xFF ) << 8 )
+ | ( ( data->buffer[ offset + 2 ] & 0xFF ) << 16 )
+ | ( ( data->buffer[ offset + 3 ] & 0xFF ) << 24 );
+ }
+ return value;
+}
+
+static void data_sam_s8( struct data *data, int offset, int count, short *dest ) {+ int idx, amp, length = data->length;
+ char *buffer = data->buffer;
+ if( offset > length ) {+ offset = length;
+ }
+ if( offset + count > length ) {+ count = length - offset;
+ }
+ for( idx = 0; idx < count; idx++ ) {+ amp = ( buffer[ offset + idx ] & 0xFF ) << 8;
+ dest[ idx ] = ( amp & 0x7FFF ) - ( amp & 0x8000 );
+ }
+}
+
+static void data_sam_s16le( struct data *data, int offset, int count, short *dest ) {+ int idx, amp, length = data->length;
+ char *buffer = data->buffer;
+ if( offset > length ) {+ offset = length;
+ }
+ if( offset + count * 2 > length ) {+ count = ( length - offset ) / 2;
+ }
+ for( idx = 0; idx < count; idx++ ) {+ amp = ( buffer[ offset + idx * 2 ] & 0xFF ) | ( buffer[ offset + idx * 2 + 1 ] << 8 );
+ dest[ idx ] = ( amp & 0x7FFF ) - ( amp & 0x8000 );
+ }
+}
+
+static int envelope_next_tick( struct envelope *envelope, int tick, int key_on ) {+ tick++;
+ if( envelope->looped && tick >= envelope->loop_end_tick ) {+ tick = envelope->loop_start_tick;
+ }
+ if( envelope->sustain && key_on && tick >= envelope->sustain_tick ) {+ tick = envelope->sustain_tick;
+ }
+ return tick;
+}
+
+static int envelope_calculate_ampl( struct envelope *envelope, int tick ) {+ int idx, point, dt, da;
+ int ampl = envelope->points_ampl[ envelope->num_points - 1 ];
+ if( tick < envelope->points_tick[ envelope->num_points - 1 ] ) {+ point = 0;
+ for( idx = 1; idx < envelope->num_points; idx++ ) {+ if( envelope->points_tick[ idx ] <= tick ) {+ point = idx;
+ }
+ }
+ dt = envelope->points_tick[ point + 1 ] - envelope->points_tick[ point ];
+ da = envelope->points_ampl[ point + 1 ] - envelope->points_ampl[ point ];
+ ampl = envelope->points_ampl[ point ];
+ ampl += ( ( da << 24 ) / dt ) * ( tick - envelope->points_tick[ point ] ) >> 24;
+ }
+ return ampl;
+}
+
+static void sample_ping_pong( struct sample *sample ) {+ int idx;
+ int loop_start = sample->loop_start;
+ int loop_length = sample->loop_length;
+ int loop_end = loop_start + loop_length;
+ short *sample_data = sample->data;
+ short *new_data = calloc( loop_end + loop_length + 1, sizeof( short ) );
+ if( new_data ) {+ memcpy( new_data, sample_data, loop_end * sizeof( short ) );
+ for( idx = 0; idx < loop_length; idx++ ) {+ new_data[ loop_end + idx ] = sample_data[ loop_end - idx - 1 ];
+ }
+ free( sample->data );
+ sample->data = new_data;
+ sample->loop_length *= 2;
+ sample->data[ loop_start + sample->loop_length ] = sample->data[ loop_start ];
+ }
+}
+
+/* Deallocate the specified module. */
+void dispose_module( struct module *module ) {+ int idx, sam;
+ struct instrument *instrument;
+ free( module->default_panning );
+ free( module->sequence );
+ if( module->patterns ) {+ for( idx = 0; idx < module->num_patterns; idx++ ) {+ free( module->patterns[ idx ].data );
+ }
+ free( module->patterns );
+ }
+ if( module->instruments ) {+ for( idx = 0; idx <= module->num_instruments; idx++ ) {+ instrument = &module->instruments[ idx ];
+ if( instrument->samples ) {+ for( sam = 0; sam < instrument->num_samples; sam++ ) {+ free( instrument->samples[ sam ].data );
+ }
+ free( instrument->samples );
+ }
+ }
+ free( module->instruments );
+ }
+ free( module );
+}
+
+static struct module* module_load_xm( struct data *data, char *message ) {+ int delta_env, offset, next_offset, idx, entry;
+ int num_rows, num_notes, pat_data_len, pat_data_offset;
+ int sam, sam_head_offset, sam_data_bytes, sam_data_samples;
+ int num_samples, sam_loop_start, sam_loop_length, amp;
+ int note, flags, key, ins, vol, fxc, fxp;
+ int point, point_tick, point_offset;
+ int looped, ping_pong, sixteen_bit;
+ char ascii[ 16 ], *pattern_data;
+ struct instrument *instrument;
+ struct sample *sample;
+ struct module *module = calloc( 1, sizeof( struct module ) );
+ if( module ) {+ if( data_u16le( data, 58 ) != 0x0104 ) {+ strcpy( message, "XM format version must be 0x0104!" );
+ dispose_module( module );
+ return NULL;
+ }
+ data_ascii( data, 17, 20, module->name );
+ delta_env = !memcmp( data_ascii( data, 38, 15, ascii ), "DigiBooster Pro", 15 );
+ offset = 60 + data_u32le( data, 60 );
+ module->sequence_len = data_u16le( data, 64 );
+ module->restart_pos = data_u16le( data, 66 );
+ module->num_channels = data_u16le( data, 68 );
+ module->num_patterns = data_u16le( data, 70 );
+ module->num_instruments = data_u16le( data, 72 );
+ module->linear_periods = data_u16le( data, 74 ) & 0x1;
+ module->default_gvol = 64;
+ module->default_speed = data_u16le( data, 76 );
+ module->default_tempo = data_u16le( data, 78 );
+ module->c2_rate = 8363;
+ module->gain = 64;
+ module->default_panning = calloc( module->num_channels, sizeof( unsigned char ) );
+ if( !module->default_panning ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( idx = 0; idx < module->num_channels; idx++ ) {+ module->default_panning[ idx ] = 128;
+ }
+ module->sequence = calloc( module->sequence_len, sizeof( unsigned char ) );
+ if( !module->sequence ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( idx = 0; idx < module->sequence_len; idx++ ) {+ entry = data_u8( data, 80 + idx );
+ module->sequence[ idx ] = entry < module->num_patterns ? entry : 0;
+ }
+ module->patterns = calloc( module->num_patterns, sizeof( struct pattern ) );
+ if( !module->patterns ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( idx = 0; idx < module->num_patterns; idx++ ) {+ if( data_u8( data, offset + 4 ) ) {+ strcpy( message, "Unknown pattern packing type!" );
+ dispose_module( module );
+ return NULL;
+ }
+ num_rows = data_u16le( data, offset + 5 );
+ if( num_rows < 1 ) {+ num_rows = 1;
+ }
+ pat_data_len = data_u16le( data, offset + 7 );
+ offset += data_u32le( data, offset );
+ next_offset = offset + pat_data_len;
+ num_notes = num_rows * module->num_channels;
+ pattern_data = calloc( num_notes, 5 );
+ if( !pattern_data ) {+ dispose_module( module );
+ return NULL;
+ }
+ module->patterns[ idx ].num_channels = module->num_channels;
+ module->patterns[ idx ].num_rows = num_rows;
+ module->patterns[ idx ].data = pattern_data;
+ if( pat_data_len > 0 ) {+ pat_data_offset = 0;
+ for( note = 0; note < num_notes; note++ ) {+ flags = data_u8( data, offset );
+ if( ( flags & 0x80 ) == 0 ) {+ flags = 0x1F;
+ } else {+ offset++;
+ }
+ key = ( flags & 0x01 ) > 0 ? data_u8( data, offset++ ) : 0;
+ pattern_data[ pat_data_offset++ ] = key;
+ ins = ( flags & 0x02 ) > 0 ? data_u8( data, offset++ ) : 0;
+ pattern_data[ pat_data_offset++ ] = ins;
+ vol = ( flags & 0x04 ) > 0 ? data_u8( data, offset++ ) : 0;
+ pattern_data[ pat_data_offset++ ] = vol;
+ fxc = ( flags & 0x08 ) > 0 ? data_u8( data, offset++ ) : 0;
+ fxp = ( flags & 0x10 ) > 0 ? data_u8( data, offset++ ) : 0;
+ if( fxc >= 0x40 ) {+ fxc = fxp = 0;
+ }
+ pattern_data[ pat_data_offset++ ] = fxc;
+ pattern_data[ pat_data_offset++ ] = fxp;
+ }
+ }
+ offset = next_offset;
+ }
+ module->instruments = calloc( module->num_instruments + 1, sizeof( struct instrument ) );
+ if( !module->instruments ) {+ dispose_module( module );
+ return NULL;
+ }
+ instrument = &module->instruments[ 0 ];
+ instrument->samples = calloc( 1, sizeof( struct sample ) );
+ if( !instrument->samples ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( ins = 1; ins <= module->num_instruments; ins++ ) {+ instrument = &module->instruments[ ins ];
+ data_ascii( data, offset + 4, 22, instrument->name );
+ num_samples = data_u16le( data, offset + 27 );
+ instrument->num_samples = ( num_samples > 0 ) ? num_samples : 1;
+ instrument->samples = calloc( instrument->num_samples, sizeof( struct sample ) );
+ if( !instrument->samples ) {+ dispose_module( module );
+ return NULL;
+ }
+ if( num_samples > 0 ) {+ for( key = 0; key < 96; key++ ) {+ instrument->key_to_sample[ key + 1 ] = data_u8( data, offset + 33 + key );
+ }
+ point_tick = 0;
+ for( point = 0; point < 12; point++ ) {+ point_offset = offset + 129 + ( point * 4 );
+ point_tick = ( delta_env ? point_tick : 0 ) + data_u16le( data, point_offset );
+ instrument->vol_env.points_tick[ point ] = point_tick;
+ instrument->vol_env.points_ampl[ point ] = data_u16le( data, point_offset + 2 );
+ }
+ point_tick = 0;
+ for( point = 0; point < 12; point++ ) {+ point_offset = offset + 177 + ( point * 4 );
+ point_tick = ( delta_env ? point_tick : 0 ) + data_u16le( data, point_offset );
+ instrument->pan_env.points_tick[ point ] = point_tick;
+ instrument->pan_env.points_ampl[ point ] = data_u16le( data, point_offset + 2 );
+ }
+ instrument->vol_env.num_points = data_u8( data, offset + 225 );
+ if( instrument->vol_env.num_points > 12 ) {+ instrument->vol_env.num_points = 0;
+ }
+ instrument->pan_env.num_points = data_u8( data, offset + 226 );
+ if( instrument->pan_env.num_points > 12 ) {+ instrument->pan_env.num_points = 0;
+ }
+ instrument->vol_env.sustain_tick = instrument->vol_env.points_tick[ data_u8( data, offset + 227 ) & 0xF ];
+ instrument->vol_env.loop_start_tick = instrument->vol_env.points_tick[ data_u8( data, offset + 228 ) & 0xF ];
+ instrument->vol_env.loop_end_tick = instrument->vol_env.points_tick[ data_u8( data, offset + 229 ) & 0xF ];
+ instrument->pan_env.sustain_tick = instrument->pan_env.points_tick[ data_u8( data, offset + 230 ) & 0xF ];
+ instrument->pan_env.loop_start_tick = instrument->pan_env.points_tick[ data_u8( data, offset + 231 ) & 0xF ];
+ instrument->pan_env.loop_end_tick = instrument->pan_env.points_tick[ data_u8( data, offset + 232 ) & 0xF ];
+ instrument->vol_env.enabled = instrument->vol_env.num_points > 0 && ( data_u8( data, offset + 233 ) & 0x1 );
+ instrument->vol_env.sustain = ( data_u8( data, offset + 233 ) & 0x2 ) > 0;
+ instrument->vol_env.looped = ( data_u8( data, offset + 233 ) & 0x4 ) > 0;
+ instrument->pan_env.enabled = instrument->pan_env.num_points > 0 && ( data_u8( data, offset + 234 ) & 0x1 );
+ instrument->pan_env.sustain = ( data_u8( data, offset + 234 ) & 0x2 ) > 0;
+ instrument->pan_env.looped = ( data_u8( data, offset + 234 ) & 0x4 ) > 0;
+ instrument->vib_type = data_u8( data, offset + 235 );
+ instrument->vib_sweep = data_u8( data, offset + 236 );
+ instrument->vib_depth = data_u8( data, offset + 237 );
+ instrument->vib_rate = data_u8( data, offset + 238 );
+ instrument->vol_fadeout = data_u16le( data, offset + 239 );
+ }
+ offset += data_u32le( data, offset );
+ sam_head_offset = offset;
+ offset += num_samples * 40;
+ for( sam = 0; sam < num_samples; sam++ ) {+ sample = &instrument->samples[ sam ];
+ sam_data_bytes = data_u32le( data, sam_head_offset );
+ sam_loop_start = data_u32le( data, sam_head_offset + 4 );
+ sam_loop_length = data_u32le( data, sam_head_offset + 8 );
+ sample->volume = data_u8( data, sam_head_offset + 12 );
+ sample->fine_tune = data_s8( data, sam_head_offset + 13 );
+ looped = ( data_u8( data, sam_head_offset + 14 ) & 0x3 ) > 0;
+ ping_pong = ( data_u8( data, sam_head_offset + 14 ) & 0x2 ) > 0;
+ sixteen_bit = ( data_u8( data, sam_head_offset + 14 ) & 0x10 ) > 0;
+ sample->panning = data_u8( data, sam_head_offset + 15 ) + 1;
+ sample->rel_note = data_s8( data, sam_head_offset + 16 );
+ data_ascii( data, sam_head_offset + 18, 22, sample->name );
+ sam_head_offset += 40;
+ sam_data_samples = sam_data_bytes;
+ if( sixteen_bit ) {+ sam_data_samples = sam_data_samples >> 1;
+ sam_loop_start = sam_loop_start >> 1;
+ sam_loop_length = sam_loop_length >> 1;
+ }
+ if( !looped || ( sam_loop_start + sam_loop_length ) > sam_data_samples ) {+ sam_loop_start = sam_data_samples;
+ sam_loop_length = 0;
+ }
+ sample->loop_start = sam_loop_start;
+ sample->loop_length = sam_loop_length;
+ sample->data = calloc( sam_data_samples + 1, sizeof( short ) );
+ if( sample->data ) {+ if( sixteen_bit ) {+ data_sam_s16le( data, offset, sam_data_samples, sample->data );
+ } else {+ data_sam_s8( data, offset, sam_data_samples, sample->data );
+ }
+ amp = 0;
+ for( idx = 0; idx < sam_data_samples; idx++ ) {+ amp = amp + sample->data[ idx ];
+ amp = ( amp & 0x7FFF ) - ( amp & 0x8000 );
+ sample->data[ idx ] = amp;
+ }
+ sample->data[ sam_loop_start + sam_loop_length ] = sample->data[ sam_loop_start ];
+ if( ping_pong ) {+ sample_ping_pong( sample );
+ }
+ } else {+ dispose_module( module );
+ return NULL;
+ }
+ offset += sam_data_bytes;
+ }
+ }
+ }
+ return module;
+}
+
+static struct module* module_load_s3m( struct data *data, char *message ) {+ int idx, module_data_idx, inst_offset, flags;
+ int version, sixteen_bit, tune, signed_samples;
+ int stereo_mode, default_pan, channel_map[ 32 ];
+ int sample_offset, sample_length, loop_start, loop_length;
+ int pat_offset, note_offset, row, chan, token;
+ int key, ins, volume, effect, param, panning;
+ char *pattern_data;
+ struct instrument *instrument;
+ struct sample *sample;
+ struct module *module = calloc( 1, sizeof( struct module ) );
+ if( module ) {+ data_ascii( data, 0, 28, module->name );
+ module->sequence_len = data_u16le( data, 32 );
+ module->num_instruments = data_u16le( data, 34 );
+ module->num_patterns = data_u16le( data, 36 );
+ flags = data_u16le( data, 38 );
+ version = data_u16le( data, 40 );
+ module->fast_vol_slides = ( ( flags & 0x40 ) == 0x40 ) || version == 0x1300;
+ signed_samples = data_u16le( data, 42 ) == 1;
+ if( data_u32le( data, 44 ) != 0x4d524353 ) {+ strcpy( message, "Not an S3M file!" );
+ dispose_module( module );
+ return NULL;
+ }
+ module->default_gvol = data_u8( data, 48 );
+ module->default_speed = data_u8( data, 49 );
+ module->default_tempo = data_u8( data, 50 );
+ module->c2_rate = 8363;
+ module->gain = data_u8( data, 51 ) & 0x7F;
+ stereo_mode = ( data_u8( data, 51 ) & 0x80 ) == 0x80;
+ default_pan = data_u8( data, 53 ) == 0xFC;
+ for( idx = 0; idx < 32; idx++ ) {+ channel_map[ idx ] = -1;
+ if( data_u8( data, 64 + idx ) < 16 ) {+ channel_map[ idx ] = module->num_channels++;
+ }
+ }
+ module->sequence = calloc( module->sequence_len, sizeof( unsigned char ) );
+ if( !module->sequence ){+ dispose_module( module );
+ return NULL;
+ }
+ for( idx = 0; idx < module->sequence_len; idx++ ) {+ module->sequence[ idx ] = data_u8( data, 96 + idx );
+ }
+ module_data_idx = 96 + module->sequence_len;
+ module->instruments = calloc( module->num_instruments + 1, sizeof( struct instrument ) );
+ if( !module->instruments ) {+ dispose_module( module );
+ return NULL;
+ }
+ instrument = &module->instruments[ 0 ];
+ instrument->num_samples = 1;
+ instrument->samples = calloc( 1, sizeof( struct sample ) );
+ if( !instrument->samples ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( ins = 1; ins <= module->num_instruments; ins++ ) {+ instrument = &module->instruments[ ins ];
+ instrument->num_samples = 1;
+ instrument->samples = calloc( 1, sizeof( struct sample ) );
+ if( !instrument->samples ) {+ dispose_module( module );
+ return NULL;
+ }
+ sample = &instrument->samples[ 0 ];
+ inst_offset = data_u16le( data, module_data_idx ) << 4;
+ module_data_idx += 2;
+ data_ascii( data, inst_offset + 48, 28, instrument->name );
+ if( data_u8( data, inst_offset ) == 1 && data_u16le( data, inst_offset + 76 ) == 0x4353 ) {+ sample_offset = ( data_u8( data, inst_offset + 13 ) << 20 )
+ + ( data_u16le( data, inst_offset + 14 ) << 4 );
+ sample_length = data_u32le( data, inst_offset + 16 );
+ loop_start = data_u32le( data, inst_offset + 20 );
+ loop_length = data_u32le( data, inst_offset + 24 ) - loop_start;
+ sample->volume = data_u8( data, inst_offset + 28 );
+ if( data_u8( data, inst_offset + 30 ) != 0 ) {+ strcpy( message, "Packed samples not supported!" );
+ dispose_module( module );
+ return NULL;
+ }
+ if( loop_start + loop_length > sample_length ) {+ loop_length = sample_length - loop_start;
+ }
+ if( loop_length < 1 || !( data_u8( data, inst_offset + 31 ) & 0x1 ) ) {+ loop_start = sample_length;
+ loop_length = 0;
+ }
+ sample->loop_start = loop_start;
+ sample->loop_length = loop_length;
+ /* stereo = data_u8( data, inst_offset + 31 ) & 0x2; */
+ sixteen_bit = data_u8( data, inst_offset + 31 ) & 0x4;
+ tune = ( log_2( data_u32le( data, inst_offset + 32 ) ) - log_2( module->c2_rate ) ) * 12;
+ sample->rel_note = tune >> FP_SHIFT;
+ sample->fine_tune = ( tune & FP_MASK ) >> ( FP_SHIFT - 7 );
+ sample->data = calloc( sample_length + 1, sizeof( short ) );
+ if( sample->data ) {+ if( sixteen_bit ) {+ data_sam_s16le( data, sample_offset, sample_length, sample->data );
+ } else {+ data_sam_s8( data, sample_offset, sample_length, sample->data );
+ }
+ if( !signed_samples ) {+ for( idx = 0; idx < sample_length; idx++ ) {+ sample->data[ idx ] = ( sample->data[ idx ] & 0xFFFF ) - 32768;
+ }
+ }
+ sample->data[ loop_start + loop_length ] = sample->data[ loop_start ];
+ } else {+ dispose_module( module );
+ return NULL;
+ }
+ }
+ }
+ module->patterns = calloc( module->num_patterns, sizeof( struct pattern ) );
+ if( !module->patterns ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( idx = 0; idx < module->num_patterns; idx++ ) {+ module->patterns[ idx ].num_channels = module->num_channels;
+ module->patterns[ idx ].num_rows = 64;
+ pattern_data = calloc( module->num_channels * 64, 5 );
+ if( !pattern_data ) {+ dispose_module( module );
+ return NULL;
+ }
+ module->patterns[ idx ].data = pattern_data;
+ pat_offset = ( data_u16le( data, module_data_idx ) << 4 ) + 2;
+ row = 0;
+ while( row < 64 ) {+ token = data_u8( data, pat_offset++ );
+ if( token ) {+ key = ins = 0;
+ if( ( token & 0x20 ) == 0x20 ) {+ /* Key + Instrument.*/
+ key = data_u8( data, pat_offset++ );
+ ins = data_u8( data, pat_offset++ );
+ if( key < 0xFE ) {+ key = ( key >> 4 ) * 12 + ( key & 0xF ) + 1;
+ } else if( key == 0xFF ) {+ key = 0;
+ }
+ }
+ volume = 0;
+ if( ( token & 0x40 ) == 0x40 ) {+ /* Volume Column.*/
+ volume = ( data_u8( data, pat_offset++ ) & 0x7F ) + 0x10;
+ if( volume > 0x50 ) {+ volume = 0;
+ }
+ }
+ effect = param = 0;
+ if( ( token & 0x80 ) == 0x80 ) {+ /* Effect + Param.*/
+ effect = data_u8( data, pat_offset++ );
+ param = data_u8( data, pat_offset++ );
+ if( effect < 1 || effect >= 0x40 ) {+ effect = param = 0;
+ } else if( effect > 0 ) {+ effect += 0x80;
+ }
+ }
+ chan = channel_map[ token & 0x1F ];
+ if( chan >= 0 ) {+ note_offset = ( row * module->num_channels + chan ) * 5;
+ pattern_data[ note_offset ] = key;
+ pattern_data[ note_offset + 1 ] = ins;
+ pattern_data[ note_offset + 2 ] = volume;
+ pattern_data[ note_offset + 3 ] = effect;
+ pattern_data[ note_offset + 4 ] = param;
+ }
+ } else {+ row++;
+ }
+ }
+ module_data_idx += 2;
+ }
+ module->default_panning = calloc( module->num_channels, sizeof( unsigned char ) );
+ if( module->default_panning ) {+ for( chan = 0; chan < 32; chan++ ) {+ if( channel_map[ chan ] >= 0 ) {+ panning = 7;
+ if( stereo_mode ) {+ panning = 12;
+ if( data_u8( data, 64 + chan ) < 8 ) {+ panning = 3;
+ }
+ }
+ if( default_pan ) {+ flags = data_u8( data, module_data_idx + chan );
+ if( ( flags & 0x20 ) == 0x20 ) {+ panning = flags & 0xF;
+ }
+ }
+ module->default_panning[ channel_map[ chan ] ] = panning * 17;
+ }
+ }
+ } else {+ dispose_module( module );
+ return NULL;
+ }
+ }
+ return module;
+}
+
+static struct module* module_load_mod( struct data *data, char *message ) {+ int idx, pat, module_data_idx, pat_data_len, pat_data_idx;
+ int period, key, ins, effect, param, fine_tune;
+ int sample_length, loop_start, loop_length;
+ char *pattern_data;
+ struct instrument *instrument;
+ struct sample *sample;
+ struct module *module = calloc( 1, sizeof( struct module ) );
+ if( module ) {+ data_ascii( data, 0, 20, module->name );
+ module->sequence_len = data_u8( data, 950 ) & 0x7F;
+ module->restart_pos = data_u8( data, 951 ) & 0x7F;
+ if( module->restart_pos >= module->sequence_len ) {+ module->restart_pos = 0;
+ }
+ module->sequence = calloc( 128, sizeof( unsigned char ) );
+ if( !module->sequence ){+ dispose_module( module );
+ return NULL;
+ }
+ for( idx = 0; idx < 128; idx++ ) {+ pat = data_u8( data, 952 + idx ) & 0x7F;
+ module->sequence[ idx ] = pat;
+ if( pat >= module->num_patterns ) {+ module->num_patterns = pat + 1;
+ }
+ }
+ switch( data_u16be( data, 1082 ) ) {+ case 0x4b2e: /* M.K. */
+ case 0x4b21: /* M!K! */
+ case 0x5434: /* FLT4 */
+ module->num_channels = 4;
+ module->c2_rate = 8287;
+ module->gain = 64;
+ break;
+ case 0x484e: /* xCHN */
+ module->num_channels = data_u8( data, 1080 ) - 48;
+ module->c2_rate = 8363;
+ module->gain = 32;
+ break;
+ case 0x4348: /* xxCH */
+ module->num_channels = ( data_u8( data, 1080 ) - 48 ) * 10;
+ module->num_channels += data_u8( data, 1081 ) - 48;
+ module->c2_rate = 8363;
+ module->gain = 32;
+ break;
+ default:
+ strcpy( message, "MOD Format not recognised!" );
+ dispose_module( module );
+ return NULL;
+ }
+ module->default_gvol = 64;
+ module->default_speed = 6;
+ module->default_tempo = 125;
+ module->default_panning = calloc( module->num_channels, sizeof( unsigned char ) );
+ if( !module->default_panning ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( idx = 0; idx < module->num_channels; idx++ ) {+ module->default_panning[ idx ] = 51;
+ if( ( idx & 3 ) == 1 || ( idx & 3 ) == 2 ) {+ module->default_panning[ idx ] = 204;
+ }
+ }
+ module_data_idx = 1084;
+ module->patterns = calloc( module->num_patterns, sizeof( struct pattern ) );
+ if( !module->patterns ) {+ dispose_module( module );
+ return NULL;
+ }
+ pat_data_len = module->num_channels * 64 * 5;
+ for( pat = 0; pat < module->num_patterns; pat++ ) {+ module->patterns[ pat ].num_channels = module->num_channels;
+ module->patterns[ pat ].num_rows = 64;
+ pattern_data = calloc( 1, pat_data_len );
+ if( !pattern_data ) {+ dispose_module( module );
+ return NULL;
+ }
+ module->patterns[ pat ].data = pattern_data;
+ for( pat_data_idx = 0; pat_data_idx < pat_data_len; pat_data_idx += 5 ) {+ period = ( data_u8( data, module_data_idx ) & 0xF ) << 8;
+ period = ( period | data_u8( data, module_data_idx + 1 ) ) * 4;
+ if( period >= 112 && period <= 6848 ) {+ key = -12 * log_2( ( period << FP_SHIFT ) / 29021 );
+ key = ( key + ( key & ( FP_ONE >> 1 ) ) ) >> FP_SHIFT;
+ pattern_data[ pat_data_idx ] = key;
+ }
+ ins = ( data_u8( data, module_data_idx + 2 ) & 0xF0 ) >> 4;
+ ins = ins | ( data_u8( data, module_data_idx ) & 0x10 );
+ pattern_data[ pat_data_idx + 1 ] = ins;
+ effect = data_u8( data, module_data_idx + 2 ) & 0x0F;
+ param = data_u8( data, module_data_idx + 3 );
+ if( param == 0 && ( effect < 3 || effect == 0xA ) ) {+ effect = 0;
+ }
+ if( param == 0 && ( effect == 5 || effect == 6 ) ) {+ effect -= 2;
+ }
+ if( effect == 8 ) {+ if( module->num_channels == 4 ) {+ effect = param = 0;
+ } else if( param > 128 ) {+ param = 128;
+ } else {+ param = ( param * 255 ) >> 7;
+ }
+ }
+ pattern_data[ pat_data_idx + 3 ] = effect;
+ pattern_data[ pat_data_idx + 4 ] = param;
+ module_data_idx += 4;
+ }
+ }
+ module->num_instruments = 31;
+ module->instruments = calloc( module->num_instruments + 1, sizeof( struct instrument ) );
+ if( !module->instruments ) {+ dispose_module( module );
+ return NULL;
+ }
+ instrument = &module->instruments[ 0 ];
+ instrument->num_samples = 1;
+ instrument->samples = calloc( 1, sizeof( struct sample ) );
+ if( !instrument->samples ) {+ dispose_module( module );
+ return NULL;
+ }
+ for( ins = 1; ins <= module->num_instruments; ins++ ) {+ instrument = &module->instruments[ ins ];
+ instrument->num_samples = 1;
+ instrument->samples = calloc( 1, sizeof( struct sample ) );
+ if( !instrument->samples ) {+ dispose_module( module );
+ return NULL;
+ }
+ sample = &instrument->samples[ 0 ];
+ data_ascii( data, ins * 30 - 10, 22, instrument->name );
+ sample_length = data_u16be( data, ins * 30 + 12 ) * 2;
+ fine_tune = ( data_u8( data, ins * 30 + 14 ) & 0xF ) << 4;
+ sample->fine_tune = ( fine_tune & 0x7F ) - ( fine_tune & 0x80 );
+ sample->volume = data_u8( data, ins * 30 + 15 ) & 0x7F;
+ if( sample->volume > 64 ) {+ sample->volume = 64;
+ }
+ loop_start = data_u16be( data, ins * 30 + 16 ) * 2;
+ loop_length = data_u16be( data, ins * 30 + 18 ) * 2;
+ if( loop_start + loop_length > sample_length ) {+ if( loop_start / 2 + loop_length <= sample_length ) {+ /* Some old modules have loop start in bytes. */
+ loop_start = loop_start / 2;
+ } else {+ loop_length = sample_length - loop_start;
+ }
+ }
+ if( loop_length < 4 ) {+ loop_start = sample_length;
+ loop_length = 0;
+ }
+ sample->loop_start = loop_start;
+ sample->loop_length = loop_length;
+ sample->data = calloc( sample_length + 1, sizeof( short ) );
+ if( sample->data ) {+ data_sam_s8( data, module_data_idx, sample_length, sample->data );
+ sample->data[ loop_start + loop_length ] = sample->data[ loop_start ];
+ } else {+ dispose_module( module );
+ return NULL;
+ }
+ module_data_idx += sample_length;
+ }
+ }
+ return module;
+}
+
+/* Allocate and initialize a module from the specified data, returns NULL on error.
+ Message must point to a 64-character buffer to receive error messages. */
+struct module* module_load( struct data *data, char *message ) {+ char ascii[ 16 ];
+ struct module* module;
+ if( !memcmp( data_ascii( data, 0, 16, ascii ), "Extended Module:", 16 ) ) {+ module = module_load_xm( data, message );
+ } else if( !memcmp( data_ascii( data, 44, 4, ascii ), "SCRM", 4 ) ) {+ module = module_load_s3m( data, message );
+ } else {+ module = module_load_mod( data, message );
+ }
+ return module;
+}
+
+static void pattern_get_note( struct pattern *pattern, int row, int chan, struct note *dest ) {+ int offset = ( row * pattern->num_channels + chan ) * 5;
+ if( offset >= 0 && row < pattern->num_rows && chan < pattern->num_channels ) {+ dest->key = pattern->data[ offset ];
+ dest->instrument = pattern->data[ offset + 1 ];
+ dest->volume = pattern->data[ offset + 2 ];
+ dest->effect = pattern->data[ offset + 3 ];
+ dest->param = pattern->data[ offset + 4 ];
+ } else {+ memset( dest, 0, sizeof( struct note ) );
+ }
+}
+
+static void channel_init( struct channel *channel, struct replay *replay, int idx ) {+ memset( channel, 0, sizeof( struct channel ) );
+ channel->replay = replay;
+ channel->id = idx;
+ channel->panning = replay->module->default_panning[ idx ];
+ channel->instrument = &replay->module->instruments[ 0 ];
+ channel->sample = &channel->instrument->samples[ 0 ];
+ channel->random_seed = ( idx + 1 ) * 0xABCDEF;
+}
+
+static void channel_volume_slide( struct channel *channel ) {+ int up = channel->vol_slide_param >> 4;
+ int down = channel->vol_slide_param & 0xF;
+ if( down == 0xF && up > 0 ) {+ /* Fine slide up.*/
+ if( channel->fx_count == 0 ) {+ channel->volume += up;
+ }
+ } else if( up == 0xF && down > 0 ) {+ /* Fine slide down.*/
+ if( channel->fx_count == 0 ) {+ channel->volume -= down;
+ }
+ } else if( channel->fx_count > 0 || channel->replay->module->fast_vol_slides ) {+ /* Normal.*/
+ channel->volume += up - down;
+ }
+ if( channel->volume > 64 ) {+ channel->volume = 64;
+ }
+ if( channel->volume < 0 ) {+ channel->volume = 0;
+ }
+}
+
+static void channel_porta_up( struct channel *channel, int param ) {+ switch( param & 0xF0 ) {+ case 0xE0: /* Extra-fine porta.*/
+ if( channel->fx_count == 0 ) {+ channel->period -= param & 0xF;
+ }
+ break;
+ case 0xF0: /* Fine porta.*/
+ if( channel->fx_count == 0 ) {+ channel->period -= ( param & 0xF ) << 2;
+ }
+ break;
+ default:/* Normal porta.*/
+ if( channel->fx_count > 0 ) {+ channel->period -= param << 2;
+ }
+ break;
+ }
+ if( channel->period < 0 ) {+ channel->period = 0;
+ }
+}
+
+static void channel_porta_down( struct channel *channel, int param ) {+ if( channel->period > 0 ) {+ switch( param & 0xF0 ) {+ case 0xE0: /* Extra-fine porta.*/
+ if( channel->fx_count == 0 ) {+ channel->period += param & 0xF;
+ }
+ break;
+ case 0xF0: /* Fine porta.*/
+ if( channel->fx_count == 0 ) {+ channel->period += ( param & 0xF ) << 2;
+ }
+ break;
+ default:/* Normal porta.*/
+ if( channel->fx_count > 0 ) {+ channel->period += param << 2;
+ }
+ break;
+ }
+ if( channel->period > 65535 ) {+ channel->period = 65535;
+ }
+ }
+}
+
+static void channel_tone_porta( struct channel *channel ) {+ if( channel->period > 0 ) {+ if( channel->period < channel->porta_period ) {+ channel->period += channel->tone_porta_param << 2;
+ if( channel->period > channel->porta_period ) {+ channel->period = channel->porta_period;
+ }
+ } else {+ channel->period -= channel->tone_porta_param << 2;
+ if( channel->period < channel->porta_period ) {+ channel->period = channel->porta_period;
+ }
+ }
+ }
+}
+
+static int channel_waveform( struct channel *channel, int phase, int type ) {+ int amplitude = 0;
+ switch( type ) {+ default: /* Sine. */
+ amplitude = sine_table[ phase & 0x1F ];
+ if( ( phase & 0x20 ) > 0 ) {+ amplitude = -amplitude;
+ }
+ break;
+ case 6: /* Saw Up.*/
+ amplitude = ( ( ( phase + 0x20 ) & 0x3F ) << 3 ) - 255;
+ break;
+ case 1: case 7: /* Saw Down. */
+ amplitude = 255 - ( ( ( phase + 0x20 ) & 0x3F ) << 3 );
+ break;
+ case 2: case 5: /* Square. */
+ amplitude = ( phase & 0x20 ) > 0 ? 255 : -255;
+ break;
+ case 3: case 8: /* Random. */
+ amplitude = ( channel->random_seed >> 20 ) - 255;
+ channel->random_seed = ( channel->random_seed * 65 + 17 ) & 0x1FFFFFFF;
+ break;
+ }
+ return amplitude;
+}
+
+static void channel_vibrato( struct channel *channel, int fine ) {+ int wave = channel_waveform( channel, channel->vibrato_phase, channel->vibrato_type & 0x3 );
+ channel->vibrato_add = wave * channel->vibrato_depth >> ( fine ? 7 : 5 );
+}
+
+static void channel_tremolo( struct channel *channel ) {+ int wave = channel_waveform( channel, channel->tremolo_phase, channel->tremolo_type & 0x3 );
+ channel->tremolo_add = wave * channel->tremolo_depth >> 6;
+}
+
+static void channel_tremor( struct channel *channel ) {+ if( channel->retrig_count >= channel->tremor_on_ticks ) {+ channel->tremolo_add = -64;
+ }
+ if( channel->retrig_count >= ( channel->tremor_on_ticks + channel->tremor_off_ticks ) ) {+ channel->tremolo_add = channel->retrig_count = 0;
+ }
+}
+
+static void channel_retrig_vol_slide( struct channel *channel ) {+ if( channel->retrig_count >= channel->retrig_ticks ) {+ channel->retrig_count = channel->sample_idx = channel->sample_fra = 0;
+ switch( channel->retrig_volume ) {+ case 0x1: channel->volume = channel->volume - 1; break;
+ case 0x2: channel->volume = channel->volume - 2; break;
+ case 0x3: channel->volume = channel->volume - 4; break;
+ case 0x4: channel->volume = channel->volume - 8; break;
+ case 0x5: channel->volume = channel->volume - 16; break;
+ case 0x6: channel->volume = channel->volume * 2 / 3; break;
+ case 0x7: channel->volume = channel->volume >> 1; break;
+ case 0x8: /* ? */ break;
+ case 0x9: channel->volume = channel->volume + 1; break;
+ case 0xA: channel->volume = channel->volume + 2; break;
+ case 0xB: channel->volume = channel->volume + 4; break;
+ case 0xC: channel->volume = channel->volume + 8; break;
+ case 0xD: channel->volume = channel->volume + 16; break;
+ case 0xE: channel->volume = channel->volume * 3 / 2; break;
+ case 0xF: channel->volume = channel->volume << 1; break;
+ }
+ if( channel->volume < 0 ) {+ channel->volume = 0;
+ }
+ if( channel->volume > 64 ) {+ channel->volume = 64;
+ }
+ }
+}
+
+static void channel_trigger( struct channel *channel ) {+ int key, sam, porta, period, fine_tune, ins = channel->note.instrument;
+ struct sample *sample;
+ if( ins > 0 && ins <= channel->replay->module->num_instruments ) {+ channel->instrument = &channel->replay->module->instruments[ ins ];
+ key = channel->note.key < 97 ? channel->note.key : 0;
+ sam = channel->instrument->key_to_sample[ key ];
+ sample = &channel->instrument->samples[ sam ];
+ channel->volume = sample->volume >= 64 ? 64 : sample->volume & 0x3F;
+ if( sample->panning > 0 ) {+ channel->panning = ( sample->panning - 1 ) & 0xFF;
+ }
+ if( channel->period > 0 && sample->loop_length > 1 ) {+ /* Amiga trigger.*/
+ channel->sample = sample;
+ }
+ channel->sample_off = 0;
+ channel->vol_env_tick = channel->pan_env_tick = 0;
+ channel->fadeout_vol = 32768;
+ channel->key_on = 1;
+ }
+ if( channel->note.effect == 0x09 || channel->note.effect == 0x8F ) {+ /* Set Sample Offset. */
+ if( channel->note.param > 0 ) {+ channel->offset_param = channel->note.param;
+ }
+ channel->sample_off = channel->offset_param << 8;
+ }
+ if( channel->note.volume >= 0x10 && channel->note.volume < 0x60 ) {+ channel->volume = channel->note.volume < 0x50 ? channel->note.volume - 0x10 : 64;
+ }
+ switch( channel->note.volume & 0xF0 ) {+ case 0x80: /* Fine Vol Down.*/
+ channel->volume -= channel->note.volume & 0xF;
+ if( channel->volume < 0 ) {+ channel->volume = 0;
+ }
+ break;
+ case 0x90: /* Fine Vol Up.*/
+ channel->volume += channel->note.volume & 0xF;
+ if( channel->volume > 64 ) {+ channel->volume = 64;
+ }
+ break;
+ case 0xA0: /* Set Vibrato Speed.*/
+ if( ( channel->note.volume & 0xF ) > 0 ) {+ channel->vibrato_speed = channel->note.volume & 0xF;
+ }
+ break;
+ case 0xB0: /* Vibrato.*/
+ if( ( channel->note.volume & 0xF ) > 0 ) {+ channel->vibrato_depth = channel->note.volume & 0xF;
+ }
+ channel_vibrato( channel, 0 );
+ break;
+ case 0xC0: /* Set Panning.*/
+ channel->panning = ( channel->note.volume & 0xF ) * 17;
+ break;
+ case 0xF0: /* Tone Porta.*/
+ if( ( channel->note.volume & 0xF ) > 0 ) {+ channel->tone_porta_param = channel->note.volume & 0xF;
+ }
+ break;
+ }
+ if( channel->note.key > 0 ) {+ if( channel->note.key > 96 ) {+ channel->key_on = 0;
+ } else {+ porta = ( channel->note.volume & 0xF0 ) == 0xF0 ||
+ channel->note.effect == 0x03 || channel->note.effect == 0x05 ||
+ channel->note.effect == 0x87 || channel->note.effect == 0x8C;
+ if( !porta ) {+ ins = channel->instrument->key_to_sample[ channel->note.key ];
+ channel->sample = &channel->instrument->samples[ ins ];
+ }
+ fine_tune = channel->sample->fine_tune;
+ if( channel->note.effect == 0x75 || channel->note.effect == 0xF2 ) {+ /* Set Fine Tune. */
+ fine_tune = ( ( channel->note.param & 0xF ) << 4 ) - 128;
+ }
+ key = channel->note.key + channel->sample->rel_note;
+ if( key < 1 ) {+ key = 1;
+ }
+ if( key > 120 ) {+ key = 120;
+ }
+ period = ( key << 6 ) + ( fine_tune >> 1 );
+ if( channel->replay->module->linear_periods ) {+ channel->porta_period = 7744 - period;
+ } else {+ channel->porta_period = 29021 * exp_2( ( period << FP_SHIFT ) / -768 ) >> FP_SHIFT;
+ }
+ if( !porta ) {+ channel->period = channel->porta_period;
+ channel->sample_idx = channel->sample_off;
+ channel->sample_fra = 0;
+ if( channel->vibrato_type < 4 ) {+ channel->vibrato_phase = 0;
+ }
+ if( channel->tremolo_type < 4 ) {+ channel->tremolo_phase = 0;
+ }
+ channel->retrig_count = channel->av_count = 0;
+ }
+ }
+ }
+}
+
+static void channel_update_envelopes( struct channel *channel ) {+ if( channel->instrument->vol_env.enabled ) {+ if( !channel->key_on ) {+ channel->fadeout_vol -= channel->instrument->vol_fadeout;
+ if( channel->fadeout_vol < 0 ) {+ channel->fadeout_vol = 0;
+ }
+ }
+ channel->vol_env_tick = envelope_next_tick( &channel->instrument->vol_env,
+ channel->vol_env_tick, channel->key_on );
+ }
+ if( channel->instrument->pan_env.enabled ) {+ channel->pan_env_tick = envelope_next_tick( &channel->instrument->pan_env,
+ channel->pan_env_tick, channel->key_on );
+ }
+}
+
+static void channel_auto_vibrato( struct channel *channel ) {+ int sweep, rate, type, wave;
+ int depth = channel->instrument->vib_depth & 0x7F;
+ if( depth > 0 ) {+ sweep = channel->instrument->vib_sweep & 0x7F;
+ rate = channel->instrument->vib_rate & 0x7F;
+ type = channel->instrument->vib_type;
+ if( channel->av_count < sweep ) {+ depth = depth * channel->av_count / sweep;
+ }
+ wave = channel_waveform( channel, channel->av_count * rate >> 2, type + 4 );
+ channel->vibrato_add += wave * depth >> 8;
+ channel->av_count++;
+ }
+}
+
+static void channel_calculate_freq( struct channel *channel ) {+ int per = channel->period + channel->vibrato_add;
+ if( channel->replay->module->linear_periods ) {+ per = per - ( channel->arpeggio_add << 6 );
+ if( per < 28 || per > 7680 ) {+ per = 7680;
+ }
+ channel->freq = ( ( channel->replay->module->c2_rate >> 4 )
+ * exp_2( ( ( 4608 - per ) << FP_SHIFT ) / 768 ) ) >> ( FP_SHIFT - 4 );
+ } else {+ if( per > 29021 ) {+ per = 29021;
+ }
+ per = ( per << FP_SHIFT ) / exp_2( ( channel->arpeggio_add << FP_SHIFT ) / 12 );
+ if( per < 28 ) {+ per = 29021;
+ }
+ channel->freq = channel->replay->module->c2_rate * 1712 / per;
+ }
+}
+
+static void channel_calculate_ampl( struct channel *channel ) {+ int vol, range, env_pan = 32, env_vol = channel->key_on ? 64 : 0;
+ if( channel->instrument->vol_env.enabled ) {+ env_vol = envelope_calculate_ampl( &channel->instrument->vol_env, channel->vol_env_tick );
+ }
+ vol = channel->volume + channel->tremolo_add;
+ if( vol > 64 ) {+ vol = 64;
+ }
+ if( vol < 0 ) {+ vol = 0;
+ }
+ vol = ( vol * channel->replay->module->gain * FP_ONE ) >> 13;
+ vol = ( vol * channel->fadeout_vol ) >> 15;
+ channel->ampl = ( vol * channel->replay->global_vol * env_vol ) >> 12;
+ if( channel->instrument->pan_env.enabled ) {+ env_pan = envelope_calculate_ampl( &channel->instrument->pan_env, channel->pan_env_tick );
+ }
+ range = ( channel->panning < 128 ) ? channel->panning : ( 255 - channel->panning );
+ channel->pann = channel->panning + ( range * ( env_pan - 32 ) >> 5 );
+}
+
+static void channel_tick( struct channel *channel ) {+ channel->vibrato_add = 0;
+ channel->fx_count++;
+ channel->retrig_count++;
+ if( !( channel->note.effect == 0x7D && channel->fx_count <= channel->note.param ) ) {+ switch( channel->note.volume & 0xF0 ) {+ case 0x60: /* Vol Slide Down.*/
+ channel->volume -= channel->note.volume & 0xF;
+ if( channel->volume < 0 ) {+ channel->volume = 0;
+ }
+ break;
+ case 0x70: /* Vol Slide Up.*/
+ channel->volume += channel->note.volume & 0xF;
+ if( channel->volume > 64 ) {+ channel->volume = 64;
+ }
+ break;
+ case 0xB0: /* Vibrato.*/
+ channel->vibrato_phase += channel->vibrato_speed;
+ channel_vibrato( channel, 0 );
+ break;
+ case 0xD0: /* Pan Slide Left.*/
+ channel->panning -= channel->note.volume & 0xF;
+ if( channel->panning < 0 ) {+ channel->panning = 0;
+ }
+ break;
+ case 0xE0: /* Pan Slide Right.*/
+ channel->panning += channel->note.volume & 0xF;
+ if( channel->panning > 255 ) {+ channel->panning = 255;
+ }
+ break;
+ case 0xF0: /* Tone Porta.*/
+ channel_tone_porta( channel );
+ break;
+ }
+ }
+ switch( channel->note.effect ) {+ case 0x01: case 0x86: /* Porta Up. */
+ channel_porta_up( channel, channel->porta_up_param );
+ break;
+ case 0x02: case 0x85: /* Porta Down. */
+ channel_porta_down( channel, channel->porta_down_param );
+ break;
+ case 0x03: case 0x87: /* Tone Porta. */
+ channel_tone_porta( channel );
+ break;
+ case 0x04: case 0x88: /* Vibrato. */
+ channel->vibrato_phase += channel->vibrato_speed;
+ channel_vibrato( channel, 0 );
+ break;
+ case 0x05: case 0x8C: /* Tone Porta + Vol Slide. */
+ channel_tone_porta( channel );
+ channel_volume_slide( channel );
+ break;
+ case 0x06: case 0x8B: /* Vibrato + Vol Slide. */
+ channel->vibrato_phase += channel->vibrato_speed;
+ channel_vibrato( channel, 0 );
+ channel_volume_slide( channel );
+ break;
+ case 0x07: case 0x92: /* Tremolo. */
+ channel->tremolo_phase += channel->tremolo_speed;
+ channel_tremolo( channel );
+ break;
+ case 0x0A: case 0x84: /* Vol Slide. */
+ channel_volume_slide( channel );
+ break;
+ case 0x11: /* Global Volume Slide. */
+ channel->replay->global_vol = channel->replay->global_vol
+ + ( channel->gvol_slide_param >> 4 )
+ - ( channel->gvol_slide_param & 0xF );
+ if( channel->replay->global_vol < 0 ) {+ channel->replay->global_vol = 0;
+ }
+ if( channel->replay->global_vol > 64 ) {+ channel->replay->global_vol = 64;
+ }
+ break;
+ case 0x19: /* Panning Slide. */
+ channel->panning = channel->panning
+ + ( channel->pan_slide_param >> 4 )
+ - ( channel->pan_slide_param & 0xF );
+ if( channel->panning < 0 ) {+ channel->panning = 0;
+ }
+ if( channel->panning > 255 ) {+ channel->panning = 255;
+ }
+ break;
+ case 0x1B: case 0x91: /* Retrig + Vol Slide. */
+ channel_retrig_vol_slide( channel );
+ break;
+ case 0x1D: case 0x89: /* Tremor. */
+ channel_tremor( channel );
+ break;
+ case 0x79: /* Retrig. */
+ if( channel->fx_count >= channel->note.param ) {+ channel->fx_count = 0;
+ channel->sample_idx = channel->sample_fra = 0;
+ }
+ break;
+ case 0x7C: case 0xFC: /* Note Cut. */
+ if( channel->note.param == channel->fx_count ) {+ channel->volume = 0;
+ }
+ break;
+ case 0x7D: case 0xFD: /* Note Delay. */
+ if( channel->note.param == channel->fx_count ) {+ channel_trigger( channel );
+ }
+ break;
+ case 0x8A: /* Arpeggio. */
+ if( channel->fx_count == 1 ) {+ channel->arpeggio_add = channel->arpeggio_param >> 4;
+ } else if( channel->fx_count == 2 ) {+ channel->arpeggio_add = channel->arpeggio_param & 0xF;
+ } else {+ channel->arpeggio_add = channel->fx_count = 0;
+ }
+ break;
+ case 0x95: /* Fine Vibrato. */
+ channel->vibrato_phase += channel->vibrato_speed;
+ channel_vibrato( channel, 1 );
+ break;
+ }
+ channel_auto_vibrato( channel );
+ channel_calculate_freq( channel );
+ channel_calculate_ampl( channel );
+ channel_update_envelopes( channel );
+}
+
+static void channel_row( struct channel *channel, struct note *note ) {+ channel->note = *note;
+ channel->retrig_count++;
+ channel->vibrato_add = channel->tremolo_add = channel->arpeggio_add = channel->fx_count = 0;
+ if( !( ( note->effect == 0x7D || note->effect == 0xFD ) && note->param > 0 ) ) {+ /* Not note delay.*/
+ channel_trigger( channel );
+ }
+ switch( channel->note.effect ) {+ case 0x01: case 0x86: /* Porta Up. */
+ if( channel->note.param > 0 ) {+ channel->porta_up_param = channel->note.param;
+ }
+ channel_porta_up( channel, channel->porta_up_param );
+ break;
+ case 0x02: case 0x85: /* Porta Down. */
+ if( channel->note.param > 0 ) {+ channel->porta_down_param = channel->note.param;
+ }
+ channel_porta_down( channel, channel->porta_down_param );
+ break;
+ case 0x03: case 0x87: /* Tone Porta. */
+ if( channel->note.param > 0 ) {+ channel->tone_porta_param = channel->note.param;
+ }
+ break;
+ case 0x04: case 0x88: /* Vibrato. */
+ if( ( channel->note.param >> 4 ) > 0 ) {+ channel->vibrato_speed = channel->note.param >> 4;
+ }
+ if( ( channel->note.param & 0xF ) > 0 ) {+ channel->vibrato_depth = channel->note.param & 0xF;
+ }
+ channel_vibrato( channel, 0 );
+ break;
+ case 0x05: case 0x8C: /* Tone Porta + Vol Slide. */
+ if( channel->note.param > 0 ) {+ channel->vol_slide_param = channel->note.param;
+ }
+ channel_volume_slide( channel );
+ break;
+ case 0x06: case 0x8B: /* Vibrato + Vol Slide. */
+ if( channel->note.param > 0 ) {+ channel->vol_slide_param = channel->note.param;
+ }
+ channel_vibrato( channel, 0 );
+ channel_volume_slide( channel );
+ break;
+ case 0x07: case 0x92: /* Tremolo. */
+ if( ( channel->note.param >> 4 ) > 0 ) {+ channel->tremolo_speed = channel->note.param >> 4;
+ }
+ if( ( channel->note.param & 0xF ) > 0 ) {+ channel->tremolo_depth = channel->note.param & 0xF;
+ }
+ channel_tremolo( channel );
+ break;
+ case 0x08: /* Set Panning.*/
+ channel->panning = channel->note.param & 0xFF;
+ break;
+ case 0x0A: case 0x84: /* Vol Slide. */
+ if( channel->note.param > 0 ) {+ channel->vol_slide_param = channel->note.param;
+ }
+ channel_volume_slide( channel );
+ break;
+ case 0x0C: /* Set Volume. */
+ channel->volume = channel->note.param >= 64 ? 64 : channel->note.param & 0x3F;
+ break;
+ case 0x10: case 0x96: /* Set Global Volume. */
+ channel->replay->global_vol = channel->note.param >= 64 ? 64 : channel->note.param & 0x3F;
+ break;
+ case 0x11: /* Global Volume Slide. */
+ if( channel->note.param > 0 ) {+ channel->gvol_slide_param = channel->note.param;
+ }
+ break;
+ case 0x14: /* Key Off. */
+ channel->key_on = 0;
+ break;
+ case 0x15: /* Set Envelope Tick. */
+ channel->vol_env_tick = channel->pan_env_tick = channel->note.param & 0xFF;
+ break;
+ case 0x19: /* Panning Slide. */
+ if( channel->note.param > 0 ) {+ channel->pan_slide_param = channel->note.param;
+ }
+ break;
+ case 0x1B: case 0x91: /* Retrig + Vol Slide. */
+ if( ( channel->note.param >> 4 ) > 0 ) {+ channel->retrig_volume = channel->note.param >> 4;
+ }
+ if( ( channel->note.param & 0xF ) > 0 ) {+ channel->retrig_ticks = channel->note.param & 0xF;
+ }
+ channel_retrig_vol_slide( channel );
+ break;
+ case 0x1D: case 0x89: /* Tremor. */
+ if( ( channel->note.param >> 4 ) > 0 ) {+ channel->tremor_on_ticks = channel->note.param >> 4;
+ }
+ if( ( channel->note.param & 0xF ) > 0 ) {+ channel->tremor_off_ticks = channel->note.param & 0xF;
+ }
+ channel_tremor( channel );
+ break;
+ case 0x21: /* Extra Fine Porta. */
+ if( channel->note.param > 0 ) {+ channel->xfine_porta_param = channel->note.param;
+ }
+ switch( channel->xfine_porta_param & 0xF0 ) {+ case 0x10:
+ channel_porta_up( channel, 0xE0 | ( channel->xfine_porta_param & 0xF ) );
+ break;
+ case 0x20:
+ channel_porta_down( channel, 0xE0 | ( channel->xfine_porta_param & 0xF ) );
+ break;
+ }
+ break;
+ case 0x71: /* Fine Porta Up. */
+ if( channel->note.param > 0 ) {+ channel->fine_porta_up_param = channel->note.param;
+ }
+ channel_porta_up( channel, 0xF0 | ( channel->fine_porta_up_param & 0xF ) );
+ break;
+ case 0x72: /* Fine Porta Down. */
+ if( channel->note.param > 0 ) {+ channel->fine_porta_down_param = channel->note.param;
+ }
+ channel_porta_down( channel, 0xF0 | ( channel->fine_porta_down_param & 0xF ) );
+ break;
+ case 0x74: case 0xF3: /* Set Vibrato Waveform. */
+ if( channel->note.param < 8 ) {+ channel->vibrato_type = channel->note.param;
+ }
+ break;
+ case 0x77: case 0xF4: /* Set Tremolo Waveform. */
+ if( channel->note.param < 8 ) {+ channel->tremolo_type = channel->note.param;
+ }
+ break;
+ case 0x7A: /* Fine Vol Slide Up. */
+ if( channel->note.param > 0 ) {+ channel->fine_vslide_up_param = channel->note.param;
+ }
+ channel->volume += channel->fine_vslide_up_param;
+ if( channel->volume > 64 ) {+ channel->volume = 64;
+ }
+ break;
+ case 0x7B: /* Fine Vol Slide Down. */
+ if( channel->note.param > 0 ) {+ channel->fine_vslide_down_param = channel->note.param;
+ }
+ channel->volume -= channel->fine_vslide_down_param;
+ if( channel->volume < 0 ) {+ channel->volume = 0;
+ }
+ break;
+ case 0x7C: case 0xFC: /* Note Cut. */
+ if( channel->note.param <= 0 ) {+ channel->volume = 0;
+ }
+ break;
+ case 0x8A: /* Arpeggio. */
+ if( channel->note.param > 0 ) {+ channel->arpeggio_param = channel->note.param;
+ }
+ break;
+ case 0x95: /* Fine Vibrato.*/
+ if( ( channel->note.param >> 4 ) > 0 ) {+ channel->vibrato_speed = channel->note.param >> 4;
+ }
+ if( ( channel->note.param & 0xF ) > 0 ) {+ channel->vibrato_depth = channel->note.param & 0xF;
+ }
+ channel_vibrato( channel, 1 );
+ break;
+ case 0xF8: /* Set Panning. */
+ channel->panning = channel->note.param * 17;
+ break;
+ }
+ channel_auto_vibrato( channel );
+ channel_calculate_freq( channel );
+ channel_calculate_ampl( channel );
+ channel_update_envelopes( channel );
+}
+
+static void channel_resample( struct channel *channel, int *mix_buf,
+ int offset, int count, int sample_rate, int interpolate ) {+ struct sample *sample = channel->sample;
+ int l_gain, r_gain, sam_idx, sam_fra, step;
+ int loop_len, loop_end, out_idx, out_end, y, m, c;
+ short *sample_data = channel->sample->data;
+ if( channel->ampl > 0 ) {+ l_gain = channel->ampl * ( 255 - channel->pann ) >> 8;
+ r_gain = channel->ampl * channel->pann >> 8;
+ sam_idx = channel->sample_idx;
+ sam_fra = channel->sample_fra;
+ step = ( channel->freq << ( FP_SHIFT - 3 ) ) / ( sample_rate >> 3 );
+ loop_len = sample->loop_length;
+ loop_end = sample->loop_start + loop_len;
+ out_idx = offset * 2;
+ out_end = ( offset + count ) * 2;
+ if( interpolate ) {+ while( out_idx < out_end ) {+ if( sam_idx >= loop_end ) {+ if( loop_len > 1 ) {+ while( sam_idx >= loop_end ) {+ sam_idx -= loop_len;
+ }
+ } else {+ break;
+ }
+ }
+ c = sample_data[ sam_idx ];
+ m = sample_data[ sam_idx + 1 ] - c;
+ y = ( ( m * sam_fra ) >> FP_SHIFT ) + c;
+ mix_buf[ out_idx++ ] += ( y * l_gain ) >> FP_SHIFT;
+ mix_buf[ out_idx++ ] += ( y * r_gain ) >> FP_SHIFT;
+ sam_fra += step;
+ sam_idx += sam_fra >> FP_SHIFT;
+ sam_fra &= FP_MASK;
+ }
+ } else {+ while( out_idx < out_end ) {+ if( sam_idx >= loop_end ) {+ if( loop_len > 1 ) {+ while( sam_idx >= loop_end ) {+ sam_idx -= loop_len;
+ }
+ } else {+ break;
+ }
+ }
+ y = sample_data[ sam_idx ];
+ mix_buf[ out_idx++ ] += ( y * l_gain ) >> FP_SHIFT;
+ mix_buf[ out_idx++ ] += ( y * r_gain ) >> FP_SHIFT;
+ sam_fra += step;
+ sam_idx += sam_fra >> FP_SHIFT;
+ sam_fra &= FP_MASK;
+ }
+ }
+ }
+}
+
+static void channel_update_sample_idx( struct channel *channel, int count, int sample_rate ) {+ struct sample *sample = channel->sample;
+ int step = ( channel->freq << ( FP_SHIFT - 3 ) ) / ( sample_rate >> 3 );
+ channel->sample_fra += step * count;
+ channel->sample_idx += channel->sample_fra >> FP_SHIFT;
+ if( channel->sample_idx > sample->loop_start ) {+ if( sample->loop_length > 1 ) {+ channel->sample_idx = sample->loop_start
+ + ( channel->sample_idx - sample->loop_start ) % sample->loop_length;
+ } else {+ channel->sample_idx = sample->loop_start;
+ }
+ }
+ channel->sample_fra &= FP_MASK;
+}
+
+static void replay_row( struct replay *replay ) {+ int idx, count;
+ struct note note;
+ struct pattern *pattern;
+ struct channel *channel;
+ struct module *module = replay->module;
+ if( replay->next_row < 0 ) {+ replay->break_pos = replay->seq_pos + 1;
+ replay->next_row = 0;
+ }
+ if( replay->break_pos >= 0 ) {+ if( replay->break_pos >= module->sequence_len ) {+ replay->break_pos = replay->next_row = 0;
+ }
+ while( module->sequence[ replay->break_pos ] >= module->num_patterns ) {+ replay->break_pos++;
+ if( replay->break_pos >= module->sequence_len ) {+ replay->break_pos = replay->next_row = 0;
+ }
+ }
+ replay->seq_pos = replay->break_pos;
+ for( idx = 0; idx < module->num_channels; idx++ ) {+ replay->channels[ idx ].pl_row = 0;
+ }
+ replay->break_pos = -1;
+ }
+ pattern = &module->patterns[ module->sequence[ replay->seq_pos ] ];
+ replay->row = replay->next_row;
+ if( replay->row >= pattern->num_rows ) {+ replay->row = 0;
+ }
+ if( replay->play_count && replay->play_count[ 0 ] ) {+ count = replay->play_count[ replay->seq_pos ][ replay->row ];
+ if( replay->pl_count < 0 && count < 127 ) {+ replay->play_count[ replay->seq_pos ][ replay->row ] = count + 1;
+ }
+ }
+ replay->next_row = replay->row + 1;
+ if( replay->next_row >= pattern->num_rows ) {+ replay->next_row = -1;
+ }
+ for( idx = 0; idx < module->num_channels; idx++ ) {+ channel = &replay->channels[ idx ];
+ pattern_get_note( pattern, replay->row, idx, ¬e );
+ if( note.effect == 0xE ) {+ note.effect = 0x70 | ( note.param >> 4 );
+ note.param &= 0xF;
+ }
+ if( note.effect == 0x93 ) {+ note.effect = 0xF0 | ( note.param >> 4 );
+ note.param &= 0xF;
+ }
+ if( note.effect == 0 && note.param > 0 ) {+ note.effect = 0x8A;
+ }
+ channel_row( channel, ¬e );
+ switch( note.effect ) {+ case 0x81: /* Set Speed. */
+ if( note.param > 0 ) {+ replay->tick = replay->speed = note.param;
+ }
+ break;
+ case 0xB: case 0x82: /* Pattern Jump.*/
+ if( replay->pl_count < 0 ) {+ replay->break_pos = note.param;
+ replay->next_row = 0;
+ }
+ break;
+ case 0xD: case 0x83: /* Pattern Break.*/
+ if( replay->pl_count < 0 ) {+ if( replay->break_pos < 0 ) {+ replay->break_pos = replay->seq_pos + 1;
+ }
+ replay->next_row = ( note.param >> 4 ) * 10 + ( note.param & 0xF );
+ }
+ break;
+ case 0xF: /* Set Speed/Tempo.*/
+ if( note.param > 0 ) {+ if( note.param < 32 ) {+ replay->tick = replay->speed = note.param;
+ } else {+ replay->tempo = note.param;
+ }
+ }
+ break;
+ case 0x94: /* Set Tempo.*/
+ if( note.param > 32 ) {+ replay->tempo = note.param;
+ }
+ break;
+ case 0x76: case 0xFB : /* Pattern Loop.*/
+ if( note.param == 0 ) {+ /* Set loop marker on this channel. */
+ channel->pl_row = replay->row;
+ }
+ if( channel->pl_row < replay->row && replay->break_pos < 0 ) {+ /* Marker valid. */
+ if( replay->pl_count < 0 ) {+ /* Not already looping, begin. */
+ replay->pl_count = note.param;
+ replay->pl_chan = idx;
+ }
+ if( replay->pl_chan == idx ) {+ /* Next Loop.*/
+ if( replay->pl_count == 0 ) {+ /* Loop finished. Invalidate current marker. */
+ channel->pl_row = replay->row + 1;
+ } else {+ /* Loop. */
+ replay->next_row = channel->pl_row;
+ }
+ replay->pl_count--;
+ }
+ }
+ break;
+ case 0x7E: case 0xFE: /* Pattern Delay.*/
+ replay->tick = replay->speed + replay->speed * note.param;
+ break;
+ }
+ }
+}
+
+static int replay_tick( struct replay *replay ) {+ int idx, num_channels, count = 1;
+ if( --replay->tick <= 0 ) {+ replay->tick = replay->speed;
+ replay_row( replay );
+ } else {+ num_channels = replay->module->num_channels;
+ for( idx = 0; idx < num_channels; idx++ ) {+ channel_tick( &replay->channels[ idx ] );
+ }
+ }
+ if( replay->play_count && replay->play_count[ 0 ] ) {+ count = replay->play_count[ replay->seq_pos ][ replay->row ] - 1;
+ }
+ return count;
+}
+
+static int module_init_play_count( struct module *module, char **play_count ) {+ int idx, pat, rows, len = 0;
+ for( idx = 0; idx < module->sequence_len; idx++ ) {+ pat = module->sequence[ idx ];
+ rows = ( pat < module->num_patterns ) ? module->patterns[ pat ].num_rows : 0;
+ if( play_count ) {+ play_count[ idx ] = play_count[ 0 ] ? &play_count[ 0 ][ len ] : NULL;
+ }
+ len += rows;
+ }
+ return len;
+}
+
+/* Set the pattern in the sequence to play. The tempo is reset to the default. */
+void replay_set_sequence_pos( struct replay *replay, int pos ) {+ int idx;
+ struct module *module = replay->module;
+ if( pos >= module->sequence_len ) {+ pos = 0;
+ }
+ replay->break_pos = pos;
+ replay->next_row = 0;
+ replay->tick = 1;
+ replay->global_vol = module->default_gvol;
+ replay->speed = module->default_speed > 0 ? module->default_speed : 6;
+ replay->tempo = module->default_tempo > 0 ? module->default_tempo : 125;
+ replay->pl_count = replay->pl_chan = -1;
+ if( replay->play_count ) {+ free( replay->play_count[ 0 ] );
+ free( replay->play_count );
+ }
+ replay->play_count = calloc( module->sequence_len, sizeof( char * ) );
+ if( replay->play_count ) {+ replay->play_count[ 0 ] = calloc( module_init_play_count( module, NULL ), sizeof( char ) );
+ module_init_play_count( module, replay->play_count );
+ }
+ for( idx = 0; idx < module->num_channels; idx++ ) {+ channel_init( &replay->channels[ idx ], replay, idx );
+ }
+ memset( replay->ramp_buf, 0, 128 * sizeof( int ) );
+ replay_tick( replay );
+}
+
+/* Deallocate the specified replay. */
+void dispose_replay( struct replay *replay ) {+ if( replay->play_count ) {+ free( replay->play_count[ 0 ] );
+ free( replay->play_count );
+ }
+ free( replay->ramp_buf );
+ free( replay->channels );
+ free( replay );
+}
+
+/* Allocate and initialize a replay with the specified sampling rate and interpolation. */
+struct replay* new_replay( struct module *module, int sample_rate, int interpolation ) {+ struct replay *replay = calloc( 1, sizeof( struct replay ) );
+ if( replay ) {+ replay->module = module;
+ replay->sample_rate = sample_rate;
+ replay->interpolation = interpolation;
+ replay->ramp_buf = calloc( 128, sizeof( int ) );
+ replay->channels = calloc( module->num_channels, sizeof( struct channel ) );
+ if( replay->ramp_buf && replay->channels ) {+ replay_set_sequence_pos( replay, 0 );
+ } else {+ dispose_replay( replay );
+ replay = NULL;
+ }
+ }
+ return replay;
+}
+
+static int calculate_tick_len( int tempo, int sample_rate ) {+ return ( sample_rate * 5 ) / ( tempo * 2 );
+}
+
+/* Returns the length of the output buffer required by replay_get_audio(). */
+int calculate_mix_buf_len( int sample_rate ) {+ return ( calculate_tick_len( 32, sample_rate ) + 65 ) * 4;
+}
+
+/* Returns the song duration in samples at the current sampling rate. */
+int replay_calculate_duration( struct replay *replay ) {+ int count = 0, duration = 0;
+ replay_set_sequence_pos( replay, 0 );
+ while( count < 1 ) {+ duration += calculate_tick_len( replay->tempo, replay->sample_rate );
+ count = replay_tick( replay );
+ }
+ replay_set_sequence_pos( replay, 0 );
+ return duration;
+}
+
+/* Seek to approximately the specified sample position.
+ The actual sample position reached is returned. */
+int replay_seek( struct replay *replay, int sample_pos ) {+ int idx, tick_len, current_pos = 0;
+ replay_set_sequence_pos( replay, 0 );
+ tick_len = calculate_tick_len( replay->tempo, replay->sample_rate );
+ while( ( sample_pos - current_pos ) >= tick_len ) {+ for( idx = 0; idx < replay->module->num_channels; idx++ ) {+ channel_update_sample_idx( &replay->channels[ idx ],
+ tick_len * 2, replay->sample_rate * 2 );
+ }
+ current_pos += tick_len;
+ replay_tick( replay );
+ tick_len = calculate_tick_len( replay->tempo, replay->sample_rate );
+ }
+ return current_pos;
+}
+
+static void replay_volume_ramp( struct replay *replay, int *mix_buf, int tick_len ) {+ int idx, a1, a2, ramp_rate = 256 * 2048 / replay->sample_rate;
+ for( idx = 0, a1 = 0; a1 < 256; idx += 2, a1 += ramp_rate ) {+ a2 = 256 - a1;
+ mix_buf[ idx ] = ( mix_buf[ idx ] * a1 + replay->ramp_buf[ idx ] * a2 ) >> 8;
+ mix_buf[ idx + 1 ] = ( mix_buf[ idx + 1 ] * a1 + replay->ramp_buf[ idx + 1 ] * a2 ) >> 8;
+ }
+ memcpy( replay->ramp_buf, &mix_buf[ tick_len * 2 ], 128 * sizeof( int ) );
+}
+
+/* 2:1 downsampling with simple but effective anti-aliasing. Buf must contain count * 2 + 1 stereo samples. */
+static void downsample( int *buf, int count ) {+ int idx, out_idx, out_len = count * 2;
+ for( idx = 0, out_idx = 0; out_idx < out_len; idx += 4, out_idx += 2 ) {+ buf[ out_idx ] = ( buf[ idx ] >> 2 ) + ( buf[ idx + 2 ] >> 1 ) + ( buf[ idx + 4 ] >> 2 );
+ buf[ out_idx + 1 ] = ( buf[ idx + 1 ] >> 2 ) + ( buf[ idx + 3 ] >> 1 ) + ( buf[ idx + 5 ] >> 2 );
+ }
+}
+
+/* Generates audio and returns the number of stereo samples written into mix_buf.
+ Individual channels may be excluded using the mute bitmask. */
+int replay_get_audio( struct replay *replay, int *mix_buf, int mute ) {+ struct channel *channel;
+ int idx, num_channels, tick_len = calculate_tick_len( replay->tempo, replay->sample_rate );
+ /* Clear output buffer. */
+ memset( mix_buf, 0, ( tick_len + 65 ) * 4 * sizeof( int ) );
+ /* Resample. */
+ num_channels = replay->module->num_channels;
+ for( idx = 0; idx < num_channels; idx++ ) {+ channel = &replay->channels[ idx ];
+ if( !( mute & 1 ) ) {+ channel_resample( channel, mix_buf, 0, ( tick_len + 65 ) * 2,
+ replay->sample_rate * 2, replay->interpolation );
+ }
+ channel_update_sample_idx( channel, tick_len * 2, replay->sample_rate * 2 );
+ mute >>= 1;
+ }
+ downsample( mix_buf, tick_len + 64 );
+ replay_volume_ramp( replay, mix_buf, tick_len );
+ replay_tick( replay );
+ return tick_len;
+}
+
+/* Returns the currently playing pattern in the sequence.*/
+int replay_get_sequence_pos( struct replay *replay ) {+ return replay->seq_pos;
+}
+
+/* Returns the currently playing row in the pattern. */
+int replay_get_row( struct replay *replay ) {+ return replay->row;
+}
--- /dev/null
+++ b/src/support/ibxm.h
@@ -1,0 +1,72 @@
+
+/* ibxm/ac mod/xm/s3m replay (c)[email protected] */
+
+extern const char *IBXM_VERSION;
+
+struct data {+ char *buffer;
+ int length;
+};
+
+struct sample {+ char name[32];
+ int loop_start, loop_length;
+ short volume, panning, rel_note, fine_tune, *data;
+};
+
+struct envelope {+ char enabled, sustain, looped, num_points;
+ short sustain_tick, loop_start_tick, loop_end_tick;
+ short points_tick[16], points_ampl[16];
+};
+
+struct instrument {+ int num_samples, vol_fadeout;
+ char name[32], key_to_sample[97];
+ char vib_type, vib_sweep, vib_depth, vib_rate;
+ struct envelope vol_env, pan_env;
+ struct sample *samples;
+};
+
+struct pattern {+ int num_channels, num_rows;
+ char *data;
+};
+
+struct module {+ char name[32];
+ int num_channels, num_instruments;
+ int num_patterns, sequence_len, restart_pos;
+ int default_gvol, default_speed, default_tempo, c2_rate, gain;
+ int linear_periods, fast_vol_slides;
+ unsigned char *default_panning, *sequence;
+ struct pattern *patterns;
+ struct instrument *instruments;
+};
+
+/* Allocate and initialize a module from the specified data, returns NULL on error.
+ Message must point to a 64-character buffer to receive error messages. */
+struct module *module_load(struct data *data, char *message);
+/* Deallocate the specified module. */
+void dispose_module(struct module *module);
+/* Allocate and initialize a replay with the specified module and sampling rate. */
+struct replay *new_replay(struct module *module, int sample_rate, int interpolation);
+/* Deallocate the specified replay. */
+void dispose_replay(struct replay *replay);
+/* Returns the song duration in samples at the current sampling rate. */
+int replay_calculate_duration(struct replay *replay);
+/* Seek to approximately the specified sample position.
+ The actual sample position reached is returned. */
+int replay_seek(struct replay *replay, int sample_pos);
+/* Set the pattern in the sequence to play. The tempo is reset to the default. */
+void replay_set_sequence_pos(struct replay *replay, int pos);
+/* Generates audio and returns the number of stereo samples written into mix_buf.
+ Individual channels may be excluded using the mute bitmask. */
+int replay_get_audio(struct replay *replay, int *mix_buf, int mute);
+/* Returns the currently playing pattern in the sequence.*/
+int replay_get_sequence_pos(struct replay *replay);
+/* Returns the currently playing row in the pattern. */
+int replay_get_row(struct replay *replay);
+/* Returns the length of the output buffer required by replay_get_audio(). */
+int calculate_mix_buf_len(int sample_rate);
+