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parameters. These seem to be used primarily to restrict percussion

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instruments to specific notes.
- Changed note velocity to not use the volume curve in recompute_amp(), since
  this sounds closer to TiMidity++, although I don't believe it's correct
  MIDI behavior. Also changed expression so that it scales the channel volume
  before going through the curve.
- Reworked load_instrument to be less opaque.
- Went through the TiMidity code and removed pretty much all of the SDL_mixer
  extensions. The only exception would be kill_others(), which I reworked
  into a kill_key_group() function, which should be useful for DLS
  instruments in the future.


SVN r918 (trunk)
This commit is contained in:
Randy Heit 2008-04-16 05:41:03 +00:00
parent 52eeff6db8
commit 9450db3cec
8 changed files with 789 additions and 1959 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

14
docs/rh-log.txt
View file

@ -1,3 +1,17 @@
April 15, 2008
- Added support for the GUS patch format's scale_frequency and scale_factor
parameters. These seem to be used primarily to restrict percussion
instruments to specific notes.
- Changed note velocity to not use the volume curve in recompute_amp(), since
this sounds closer to TiMidity++, although I don't believe it's correct
MIDI behavior. Also changed expression so that it scales the channel volume
before going through the curve.
- Reworked load_instrument to be less opaque.
- Went through the TiMidity code and removed pretty much all of the SDL_mixer
extensions. The only exception would be kill_others(), which I reworked
into a kill_key_group() function, which should be useful for DLS
instruments in the future.
April 15, 2008 (Changes by Graf Zahl)
- Added translation support to multipatch textures. Not tested yet!
- Added Martin Howe's morph weapon update.

917
src/timidity/instrum.cpp
View file

File diff suppressed because it is too large Load diff

66
src/timidity/instrum_dls.cpp
View file

@ -1121,29 +1121,29 @@ static void load_region_dls(Renderer *song, Sample *sample, DLS_Instrument *ins,
DLS_Region *rgn = &ins->regions[index];
DLS_Wave *wave = &song->patches->waveList[rgn->wlnk->ulTableIndex];
if (!(rgn->header->fusOptions & F_RGN_OPTION_SELFNONEXCLUSIVE))
{
sample->exclusiveClass = (SBYTE)rgn->header->usKeyGroup;
}
sample->self_nonexclusive = !!(rgn->header->fusOptions & F_RGN_OPTION_SELFNONEXCLUSIVE);
sample->key_group = (SBYTE)rgn->header->usKeyGroup;
sample->low_freq = SDWORD(note_to_freq(rgn->header->RangeKey.usLow));
sample->high_freq = SDWORD(note_to_freq(rgn->header->RangeKey.usHigh));
sample->root_freq = SDWORD(note_to_freq(rgn->wsmp->usUnityNote));
sample->low_vel = rgn->header->RangeVelocity.usLow;
sample->high_vel = rgn->header->RangeVelocity.usHigh;
sample->modes = wave->format->wBitsPerSample == 8 ? MODES_UNSIGNED : MODES_16BIT;
sample->modes = wave->format->wBitsPerSample == 8 ? PATCH_UNSIGNED : PATCH_16;
sample->sample_rate = wave->format->dwSamplesPerSec;
sample->data = NULL;
sample->data_length = wave->length;
convert_sample_data(sample, wave->data);
if (rgn->wsmp->cSampleLoops) {
sample->modes |= (MODES_LOOPING|MODES_SUSTAIN);
if (rgn->wsmp->cSampleLoops)
{
sample->modes |= (PATCH_LOOPEN | PATCH_SUSTAIN);
sample->loop_start = rgn->wsmp_loop->ulStart / 2;
sample->loop_end = sample->loop_start + (rgn->wsmp_loop->ulLength / 2);
}
sample->volume = 1.0f;
if (sample->modes & MODES_SUSTAIN) {
if (sample->modes & PATCH_SUSTAIN)
{
int value;
double attack, hold, decay, release; int sustain;
CONNECTIONLIST *art = NULL;
@ -1192,12 +1192,7 @@ static void load_region_dls(Renderer *song, Sample *sample, DLS_Instrument *ins,
sample->envelope_offset[RELEASEC] = to_offset(0);
sample->envelope_rate[RELEASEC] = to_offset(1);
sample->modes |= MODES_ENVELOPE;
}
for (int j = ATTACK; j < DELAY; j++)
{
sample->modulation_rate[j] = float(sample->envelope_rate[j]);
sample->modulation_offset[j] = float(sample->envelope_offset[j]);
sample->modes |= PATCH_NO_SRELEASE;
}
sample->data_length <<= FRACTION_BITS;
@ -1205,55 +1200,54 @@ static void load_region_dls(Renderer *song, Sample *sample, DLS_Instrument *ins,
sample->loop_end <<= FRACTION_BITS;
}
InstrumentLayer *load_instrument_dls(Renderer *song, int drum, int bank, int instrument)
Instrument *load_instrument_dls(Renderer *song, int drum, int bank, int instrument)
{
InstrumentLayer *layer;
Instrument *inst;
DWORD i;
DLS_Instrument *dls_ins = NULL;
if (song->patches == NULL)
{
return NULL;
}
drum = drum ? 0x80000000 : 0;
for (i = 0; i < song->patches->cInstruments; ++i) {
for (i = 0; i < song->patches->cInstruments; ++i)
{
dls_ins = &song->patches->instruments[i];
if ((dls_ins->header->Locale.ulBank & 0x80000000) == (ULONG)drum &&
((dls_ins->header->Locale.ulBank >> 8) & 0xFF) == (ULONG)bank &&
dls_ins->header->Locale.ulInstrument == (ULONG)instrument)
break;
}
if (i == song->patches->cInstruments && !bank) {
for (i = 0; i < song->patches->cInstruments; ++i) {
if (i == song->patches->cInstruments && bank == 0)
{
for (i = 0; i < song->patches->cInstruments; ++i)
{
dls_ins = &song->patches->instruments[i];
if ((dls_ins->header->Locale.ulBank & 0x80000000) == (ULONG)drum &&
dls_ins->header->Locale.ulInstrument == (ULONG)instrument)
break;
}
}
if (i == song->patches->cInstruments) {
if (i == song->patches->cInstruments)
{
// SNDDBG(("Couldn't find %s instrument %d in bank %d\n", drum ? "drum" : "melodic", instrument, bank));
return NULL;
}
layer = (InstrumentLayer *)safe_malloc(sizeof(InstrumentLayer));
layer->lo = 0;
layer->hi = 127;
layer->instrument = (Instrument *)safe_malloc(sizeof(Instrument));
layer->instrument->type = INST_DLS;
layer->instrument->samples = dls_ins->header->cRegions;
layer->instrument->sample = (Sample *)safe_malloc(layer->instrument->samples * sizeof(Sample));
layer->instrument->left_samples = layer->instrument->samples;
layer->instrument->left_sample = layer->instrument->sample;
layer->instrument->right_samples = 0;
layer->instrument->right_sample = NULL;
memset(layer->instrument->sample, 0, layer->instrument->samples * sizeof(Sample));
inst = (Instrument *)safe_malloc(sizeof(Instrument));
inst->type = INST_DLS;
inst->samples = dls_ins->header->cRegions;
inst->sample = (Sample *)safe_malloc(inst->samples * sizeof(Sample));
memset(inst->sample, 0, inst->samples * sizeof(Sample));
/*
printf("Found %s instrument %d in bank %d named %s with %d regions\n", drum ? "drum" : "melodic", instrument, bank, dls_ins->name, inst->samples);
*/
for (i = 0; i < dls_ins->header->cRegions; ++i) {
load_region_dls(song, &layer->instrument->sample[i], dls_ins, i);
for (i = 0; i < dls_ins->header->cRegions; ++i)
{
load_region_dls(song, &inst->sample[i], dls_ins, i);
}
return layer;
return inst;
}
#endif /* !TEST_MAIN_DLS */

36
src/timidity/mix.cpp
View file

@ -37,15 +37,14 @@ int recompute_envelope(Voice *v)
stage = v->envelope_stage;
if (stage >= DELAY)
if (stage > RELEASEC)
{
/* Envelope ran out. */
int tmp = (v->status == VOICE_DIE); /* Already displayed as dead */
v->status = VOICE_FREE;
return 1;
}
if (v->sample->modes & MODES_ENVELOPE)
if (v->sample->modes & PATCH_NO_SRELEASE)
{
if (v->status == VOICE_ON || v->status == VOICE_SUSTAINED)
{
@ -60,7 +59,9 @@ int recompute_envelope(Voice *v)
v->envelope_stage = stage + 1;
if (v->envelope_volume == v->sample->envelope_offset[stage])
{
return recompute_envelope(v);
}
v->envelope_target = v->sample->envelope_offset[stage];
v->envelope_increment = v->sample->envelope_rate[stage];
if (v->envelope_target < v->envelope_volume)
@ -75,28 +76,30 @@ void apply_envelope_to_amp(Voice *v)
{
ramp = v->right_amp;
if (v->tremolo_phase_increment)
if (v->tremolo_phase_increment != 0)
{
lamp *= v->tremolo_volume;
ramp *= v->tremolo_volume;
}
if (v->sample->modes & MODES_ENVELOPE)
if (v->sample->modes & PATCH_NO_SRELEASE)
{
double vol = calc_vol(v->envelope_volume / float(1 << 30));
lamp *= vol;
ramp *= vol;
}
v->left_mix = float(lamp);
v->right_mix = float(ramp);
}
else
{
if (v->tremolo_phase_increment)
if (v->tremolo_phase_increment != 0)
{
lamp *= v->tremolo_volume;
if (v->sample->modes & MODES_ENVELOPE)
}
if (v->sample->modes & PATCH_NO_SRELEASE)
{
lamp *= calc_vol(v->envelope_volume / float(1 << 30));
}
v->left_mix = float(lamp);
}
}
@ -104,13 +107,14 @@ void apply_envelope_to_amp(Voice *v)
static int update_envelope(Voice *v)
{
v->envelope_volume += v->envelope_increment;
/* Why is there no ^^ operator?? */
if (((v->envelope_increment < 0) && (v->envelope_volume <= v->envelope_target)) ||
((v->envelope_increment > 0) && (v->envelope_volume >= v->envelope_target)))
{
v->envelope_volume = v->envelope_target;
if (recompute_envelope(v))
{
return 1;
}
}
return 0;
}
@ -119,7 +123,7 @@ static void update_tremolo(Voice *v)
{
int depth = v->sample->tremolo_depth << 7;
if (v->tremolo_sweep)
if (v->tremolo_sweep != 0)
{
/* Update sweep position */
@ -151,12 +155,14 @@ static void update_tremolo(Voice *v)
/* Returns 1 if the note died */
static int update_signal(Voice *v)
{
if (v->envelope_increment && update_envelope(v))
if (v->envelope_increment != 0 && update_envelope(v))
{
return 1;
if (v->tremolo_phase_increment)
}
if (v->tremolo_phase_increment != 0)
{
update_tremolo(v);
}
apply_envelope_to_amp(v);
return 0;
}

1145
src/timidity/playmidi.cpp
View file

File diff suppressed because it is too large Load diff

41
src/timidity/resample.cpp
View file

@ -215,12 +215,12 @@ static sample_t *rs_bidir(sample_t *resample_buffer, Voice *vp, int count)
/* We only need to compute one half of the vibrato sine cycle */
static int vib_phase_to_inc_ptr(int phase)
{
if (phase < VIBRATO_SAMPLE_INCREMENTS/2)
return VIBRATO_SAMPLE_INCREMENTS/2-1-phase;
else if (phase >= 3*VIBRATO_SAMPLE_INCREMENTS/2)
return 5*VIBRATO_SAMPLE_INCREMENTS/2-1-phase;
if (phase < VIBRATO_SAMPLE_INCREMENTS / 2)
return VIBRATO_SAMPLE_INCREMENTS / 2 - 1 - phase;
else if (phase >= VIBRATO_SAMPLE_INCREMENTS * 3 / 2)
return VIBRATO_SAMPLE_INCREMENTS * 5 / 2 - 1 - phase;
else
return phase-VIBRATO_SAMPLE_INCREMENTS/2;
return phase - VIBRATO_SAMPLE_INCREMENTS / 2;
}
static int update_vibrato(float output_rate, Voice *vp, int sign)
@ -229,8 +229,8 @@ static int update_vibrato(float output_rate, Voice *vp, int sign)
int phase;
double a, pb;
if (vp->vibrato_phase++ >= 2*VIBRATO_SAMPLE_INCREMENTS-1)
vp->vibrato_phase=0;
if (vp->vibrato_phase++ >= 2 * VIBRATO_SAMPLE_INCREMENTS - 1)
vp->vibrato_phase = 0;
phase = vib_phase_to_inc_ptr(vp->vibrato_phase);
if (vp->vibrato_sample_increment[phase])
@ -244,7 +244,7 @@ static int update_vibrato(float output_rate, Voice *vp, int sign)
/* Need to compute this sample increment. */
depth = vp->sample->vibrato_depth << 7;
if (vp->vibrato_sweep)
if (vp->vibrato_sweep != 0)
{
/* Need to update sweep */
vp->vibrato_sweep_position += vp->vibrato_sweep;
@ -265,7 +265,7 @@ static int update_vibrato(float output_rate, Voice *vp, int sign)
pb = (sine(vp->vibrato_phase * (1.0/(2*VIBRATO_SAMPLE_INCREMENTS)))
* (double)(depth) * VIBRATO_AMPLITUDE_TUNING);
a *= pow(2.0, pb / (8191 * 12.f));
a *= pow(2.0, pb / (8192 * 12.f));
/* If the sweep's over, we can store the newly computed sample_increment */
if (!vp->vibrato_sweep)
@ -511,11 +511,11 @@ sample_t *resample_voice(Renderer *song, Voice *vp, int *countptr)
if (vp->vibrato_control_ratio)
{
if ((modes & MODES_LOOPING) &&
((modes & MODES_ENVELOPE) ||
if ((modes & PATCH_LOOPEN) &&
((modes & PATCH_NO_SRELEASE) ||
(vp->status == VOICE_ON || vp->status == VOICE_SUSTAINED)))
{
if (modes & MODES_PINGPONG)
if (modes & PATCH_BIDIR)
return rs_vib_bidir(song->resample_buffer, song->rate, vp, *countptr);
else
return rs_vib_loop(song->resample_buffer, song->rate, vp, *countptr);
@ -527,11 +527,11 @@ sample_t *resample_voice(Renderer *song, Voice *vp, int *countptr)
}
else
{
if ((modes & MODES_LOOPING) &&
((modes & MODES_ENVELOPE) ||
if ((modes & PATCH_LOOPEN) &&
((modes & PATCH_NO_SRELEASE) ||
(vp->status == VOICE_ON || vp->status == VOICE_SUSTAINED)))
{
if (modes & MODES_PINGPONG)
if (modes & PATCH_BIDIR)
return rs_bidir(song->resample_buffer, vp, *countptr);
else
return rs_loop(song->resample_buffer, vp, *countptr);
@ -554,11 +554,14 @@ void pre_resample(Renderer *song, Sample *sp)
"C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B"
};
cmsg(CMSG_INFO, VERB_NOISY, " * pre-resampling for note %d (%s%d)",
sp->note_to_use,
note_name[sp->note_to_use % 12], (sp->note_to_use & 0x7F) / 12);
if (sp->scale_factor != 0)
return;
a = (sp->sample_rate * note_to_freq(sp->note_to_use)) / (sp->root_freq * song->rate);
cmsg(CMSG_INFO, VERB_NOISY, " * pre-resampling for note %d (%s%d)\n",
sp->scale_note,
note_name[sp->scale_note % 12], (sp->scale_note & 0x7F) / 12);
a = (sp->sample_rate * note_to_freq(sp->scale_note)) / (sp->root_freq * song->rate);
if (a <= 0)
return;
newlen = (int)(sp->data_length / a);

54
src/timidity/timidity.cpp
View file

@ -413,6 +413,13 @@ void FreeAll()
int LoadConfig(const char *filename)
{
/* !!! FIXME: This may be ugly, but slightly less so than requiring the
* default search path to have only one element. I think.
*
* We only need to include the likely locations for the config
* file itself since that file should contain any other directory
* that needs to be added to the search path.
*/
clear_pathlist();
#ifdef _WIN32
add_to_pathlist("C:\\TIMIDITY");
@ -443,25 +450,21 @@ Renderer::Renderer(float sample_rate)
{
rate = sample_rate;
patches = NULL;
default_instrument = NULL;
#ifdef FAST_DECAY
fast_decay = true;
#else
fast_decay = false;
#endif
resample_buffer_size = 0;
resample_buffer = NULL;
adjust_panning_immediately = false;
control_ratio = clamp(int(rate / CONTROLS_PER_SECOND), 1, MAX_CONTROL_RATIO);
lost_notes = 0;
cut_notes = 0;
default_instrument = NULL;
default_program = DEFAULT_PROGRAM;
if (def_instr_name.IsNotEmpty())
set_default_instrument(def_instr_name);
voices = DEFAULT_VOICES;
memset(voice, 0, sizeof(voice));
memset(drumvolume, 0, sizeof(drumvolume));
memset(drumpanpot, 0, sizeof(drumpanpot));
memset(drumreverberation, 0, sizeof(drumreverberation));
memset(drumchorusdepth, 0, sizeof(drumchorusdepth));
drumchannels = DEFAULT_DRUMCHANNELS;
}
@ -492,22 +495,7 @@ void Renderer::ComputeOutput(float *buffer, int count)
{
if (v->status != VOICE_FREE)
{
if (v->sample_offset == 0 && v->echo_delay_count)
{
if (v->echo_delay_count >= count)
{
v->echo_delay_count -= count;
}
else
{
mix_voice(this, buffer + v->echo_delay_count, v, count - v->echo_delay_count);
v->echo_delay_count = 0;
}
}
else
{
mix_voice(this, buffer, v, count);
}
mix_voice(this, buffer, v, count);
}
}
}
@ -520,6 +508,11 @@ void Renderer::MarkInstrument(int banknum, int percussion, int instr)
{
return;
}
if (banknum != 0)
{
/* Mark the standard bank in case it's not defined by this one. */
MarkInstrument(0, percussion, instr);
}
if (percussion)
{
bank = drumset[banknum];
@ -532,9 +525,9 @@ void Renderer::MarkInstrument(int banknum, int percussion, int instr)
{
return;
}
if (bank->tone[instr].layer == NULL)
if (bank->instrument[instr] == NULL)
{
bank->tone[instr].layer = MAGIC_LOAD_INSTRUMENT;
bank->instrument[instr] = MAGIC_LOAD_INSTRUMENT;
}
}
@ -557,9 +550,6 @@ void cmsg(int type, int verbosity_level, const char *fmt, ...)
va_start(args, fmt);
vsprintf(buf, fmt, args);
va_end(args);
size_t l = strlen(buf);
buf[l] = '\n';
buf[l+1] = '\0';
OutputDebugString(buf);
#endif
}

475
src/timidity/timidity.h
View file

@ -33,33 +33,27 @@ config.h
*/
/* Acoustic Grand Piano seems to be the usual default instrument. */
#define DEFAULT_PROGRAM 0
#define DEFAULT_PROGRAM 0
/* 9 here is MIDI channel 10, which is the standard percussion channel.
Some files (notably C:\WINDOWS\CANYON.MID) think that 16 is one too.
On the other hand, some files know that 16 is not a drum channel and
try to play music on it. This is now a runtime option, so this isn't
a critical choice anymore. */
#define DEFAULT_DRUMCHANNELS (1<<9)
/*#define DEFAULT_DRUMCHANNELS ((1<<9) | (1<<15))*/
#define DEFAULT_DRUMCHANNELS (1<<9)
/*#define DEFAULT_DRUMCHANNELS ((1<<9) | (1<<15))*/
/* Default sampling rate, default polyphony, and maximum polyphony.
All but the last can be overridden from the command line. */
#define DEFAULT_RATE 32000
#define DEFAULT_VOICES 32
#define MAX_VOICES 256
#define MAXCHAN 16
#define MAXNOTE 128
/* Default polyphony, and maximum polyphony. */
#define DEFAULT_VOICES 32
#define MAX_VOICES 256
#define MAXCHAN 16
#define MAXNOTE 128
/* 1000 here will give a control ratio of 22:1 with 22 kHz output.
Higher CONTROLS_PER_SECOND values allow more accurate rendering
of envelopes and tremolo. The cost is CPU time. */
#define CONTROLS_PER_SECOND 1000
/* Make envelopes twice as fast. Saves ~20% CPU time (notes decay
faster) and sounds more like a GUS. There is now a command line
option to toggle this as well. */
//#define FAST_DECAY
#define CONTROLS_PER_SECOND 1000
/* How many bits to use for the fractional part of sample positions.
This affects tonal accuracy. The entire position counter must fit
@ -67,7 +61,7 @@ config.h
a sample is 1048576 samples (2 megabytes in memory). The GUS gets
by with just 9 bits and a little help from its friends...
"The GUS does not SUCK!!!" -- a happy user :) */
#define FRACTION_BITS 12
#define FRACTION_BITS 12
/* For some reason the sample volume is always set to maximum in all
patch files. Define this for a crude adjustment that may help
@ -76,7 +70,7 @@ config.h
/* The number of samples to use for ramping out a dying note. Affects
click removal. */
#define MAX_DIE_TIME 20
#define MAX_DIE_TIME 20
/**************************************************************************/
/* Anything below this shouldn't need to be changed unless you're porting
@ -84,37 +78,37 @@ config.h
/**************************************************************************/
/* change FRACTION_BITS above, not these */
#define INTEGER_BITS (32 - FRACTION_BITS)
#define INTEGER_MASK (0xFFFFFFFF << FRACTION_BITS)
#define FRACTION_MASK (~ INTEGER_MASK)
#define MAX_SAMPLE_SIZE (1 << INTEGER_BITS)
#define INTEGER_BITS (32 - FRACTION_BITS)
#define INTEGER_MASK (0xFFFFFFFF << FRACTION_BITS)
#define FRACTION_MASK (~ INTEGER_MASK)
#define MAX_SAMPLE_SIZE (1 << INTEGER_BITS)
/* This is enforced by some computations that must fit in an int */
#define MAX_CONTROL_RATIO 255
#define MAX_CONTROL_RATIO 255
#define MAX_AMPLIFICATION 800
#define MAX_AMPLIFICATION 800
/* The TiMiditiy configuration file */
#define CONFIG_FILE "timidity.cfg"
typedef float sample_t;
typedef float final_volume_t;
#define FINAL_VOLUME(v) (v)
#define FINAL_VOLUME(v) (v)
#define FSCALE(a,b) ((a) * (float)(1<<(b)))
#define FSCALENEG(a,b) ((a) * (1.0L / (float)(1<<(b))))
#define FSCALE(a,b) ((a) * (float)(1<<(b)))
#define FSCALENEG(a,b) ((a) * (1.0L / (float)(1<<(b))))
/* Vibrato and tremolo Choices of the Day */
#define SWEEP_TUNING 38
#define VIBRATO_AMPLITUDE_TUNING 1.0L
#define VIBRATO_RATE_TUNING 38
#define TREMOLO_AMPLITUDE_TUNING 1.0L
#define TREMOLO_RATE_TUNING 38
#define SWEEP_TUNING 38
#define VIBRATO_AMPLITUDE_TUNING 1.0
#define VIBRATO_RATE_TUNING 38
#define TREMOLO_AMPLITUDE_TUNING 1.0
#define TREMOLO_RATE_TUNING 38
#define SWEEP_SHIFT 16
#define RATE_SHIFT 5
#define SWEEP_SHIFT 16
#define RATE_SHIFT 5
#define VIBRATO_SAMPLE_INCREMENTS 32
#define VIBRATO_SAMPLE_INCREMENTS 32
#ifndef PI
#define PI 3.14159265358979323846
@ -188,21 +182,20 @@ FileReader *open_filereader(const char *name, int open, int *plumpnum);
controls.h
*/
#define CMSG_INFO 0
#define CMSG_WARNING 1
#define CMSG_ERROR 2
#define CMSG_FATAL 3
#define CMSG_TRACE 4
#define CMSG_TIME 5
#define CMSG_TOTAL 6
#define CMSG_FILE 7
#define CMSG_TEXT 8
enum
{
CMSG_INFO,
CMSG_WARNING,
CMSG_ERROR
};
#define VERB_NORMAL 0
#define VERB_VERBOSE 1
#define VERB_NOISY 2
#define VERB_DEBUG 3
#define VERB_DEBUG_SILLY 4
enum
{
VERB_NORMAL,
VERB_VERBOSE,
VERB_NOISY,
VERB_DEBUG
};
void cmsg(int type, int verbosity_level, const char *fmt, ...);
@ -217,114 +210,174 @@ struct Sample
loop_start, loop_end, data_length,
sample_rate, low_vel, high_vel, low_freq, high_freq, root_freq;
SDWORD
envelope_rate[7], envelope_offset[7];
envelope_rate[6], envelope_offset[6];
float
modulation_rate[7], modulation_offset[7];
float
volume, resonance,
modEnvToFilterFc, modEnvToPitch, modLfoToFilterFc;
volume;
sample_t *data;
SDWORD
tremolo_sweep_increment, tremolo_phase_increment,
lfo_sweep_increment, lfo_phase_increment,
vibrato_sweep_increment, vibrato_control_ratio,
cutoff_freq;
vibrato_sweep_increment, vibrato_control_ratio;
BYTE
reverberation, chorusdepth,
tremolo_depth, vibrato_depth,
modes,
attenuation;
modes;
WORD
freq_center, panning;
SBYTE
note_to_use, exclusiveClass;
panning, scale_factor;
SWORD
keyToModEnvHold, keyToModEnvDecay,
keyToVolEnvHold, keyToVolEnvDecay;
SDWORD
freq_scale;
scale_note;
bool
self_nonexclusive;
BYTE
key_group;
};
void convert_sample_data(Sample *sample, const void *data);
void free_instruments();
/* Bits in modes: */
#define MODES_16BIT (1<<0)
#define MODES_UNSIGNED (1<<1)
#define MODES_LOOPING (1<<2)
#define MODES_PINGPONG (1<<3)
#define MODES_REVERSE (1<<4)
#define MODES_SUSTAIN (1<<5)
#define MODES_ENVELOPE (1<<6)
#define MODES_FAST_RELEASE (1<<7)
/* Patch definition: */
enum
{
HEADER_SIZE = 12,
ID_SIZE = 10,
DESC_SIZE = 60,
RESERVED_SIZE = 40,
PATCH_HEADER_RESERVED_SIZE = 36,
LAYER_RESERVED_SIZE = 40,
PATCH_DATA_RESERVED_SIZE = 36,
INST_NAME_SIZE = 16,
ENVELOPES = 6,
MAX_LAYERS = 4
};
#define GF1_HEADER_TEXT "GF1PATCH110"
#define INST_GUS 0
#define INST_SF2 1
#define INST_DLS 2
enum
{
PATCH_16 = (1<<0),
PATCH_UNSIGNED = (1<<1),
PATCH_LOOPEN = (1<<2),
PATCH_BIDIR = (1<<3),
PATCH_BACKWARD = (1<<4),
PATCH_SUSTAIN = (1<<5),
PATCH_NO_SRELEASE = (1<<6),
PATCH_FAST_REL = (1<<7)
};
#ifdef _MSC_VER
#pragma pack(push, 1)
#define GCC_PACKED
#else
#define GCC_PACKED __attribute__((__packed__))
#endif
struct GF1PatchHeader
{
char Header[HEADER_SIZE];
char GravisID[ID_SIZE]; /* Id = "ID#000002" */
char Description[DESC_SIZE];
BYTE Instruments;
BYTE Voices;
BYTE Channels;
WORD WaveForms;
WORD MasterVolume;
DWORD DataSize;
BYTE Reserved[PATCH_HEADER_RESERVED_SIZE];
} GCC_PACKED;
struct GF1InstrumentData
{
WORD Instrument;
char InstrumentName[INST_NAME_SIZE];
int InstrumentSize;
BYTE Layers;
BYTE Reserved[RESERVED_SIZE];
} GCC_PACKED;
struct GF1LayerData
{
BYTE LayerDuplicate;
BYTE Layer;
int LayerSize;
BYTE Samples;
BYTE Reserved[LAYER_RESERVED_SIZE];
} GCC_PACKED;
struct GF1PatchData
{
char WaveName[7];
BYTE Fractions;
int WaveSize;
int StartLoop;
int EndLoop;
WORD SampleRate;
int LowFrequency;
int HighFrequency;
int RootFrequency;
SWORD Tune;
BYTE Balance;
BYTE EnvelopeRate[6];
BYTE EnvelopeOffset[6];
BYTE TremoloSweep;
BYTE TremoloRate;
BYTE TremoloDepth;
BYTE VibratoSweep;
BYTE VibratoRate;
BYTE VibratoDepth;
BYTE Modes;
SWORD ScaleFrequency;
WORD ScaleFactor; /* From 0 to 2048 or 0 to 2 */
BYTE Reserved[PATCH_DATA_RESERVED_SIZE];
} GCC_PACKED;
#ifdef _MSC_VER
#pragma pack(pop)
#endif
#undef GCC_PACKED
enum
{
INST_GUS,
INST_DLS
};
struct Instrument
{
Instrument();
~Instrument();
int type;
int samples;
Sample *sample;
int left_samples;
Sample *left_sample;
int right_samples;
Sample *right_sample;
};
struct InstrumentLayer
{
BYTE lo, hi;
Instrument *instrument;
InstrumentLayer *next;
};
struct cfg_type
{
int font_code;
int num;
const char *name;
};
#define FONT_NORMAL 0
#define FONT_FFF 1
#define FONT_SBK 2
#define FONT_TONESET 3
#define FONT_DRUMSET 4
#define FONT_PRESET 5
struct ToneBankElement
{
ToneBankElement() : layer(NULL), font_type(0), sf_ix(0), tuning(0),
ToneBankElement() :
note(0), amp(0), pan(0), strip_loop(0), strip_envelope(0), strip_tail(0)
{}
FString name;
InstrumentLayer *layer;
int font_type, sf_ix, tuning;
int note, amp, pan, strip_loop, strip_envelope, strip_tail;
};
/* A hack to delay instrument loading until after reading the
entire MIDI file. */
#define MAGIC_LOAD_INSTRUMENT ((InstrumentLayer *)(-1))
#define MAGIC_LOAD_INSTRUMENT ((Instrument *)(-1))
#define MAXPROG 128
#define MAXBANK 130
#define SFXBANK (MAXBANK-1)
#define SFXDRUM1 (MAXBANK-2)
#define SFXDRUM2 (MAXBANK-1)
#define XGDRUM 1
enum
{
MAXPROG = 128,
MAXBANK = 128
};
struct ToneBank
{
FString name;
ToneBankElement tone[MAXPROG];
ToneBank();
~ToneBank();
ToneBankElement *tone;
Instrument *instrument[MAXPROG];
};
#define SPECIAL_PROGRAM -1
#define SPECIAL_PROGRAM -1
extern void pcmap(int *b, int *v, int *p, int *drums);
@ -341,60 +394,55 @@ playmidi.h
*/
/* Midi events */
#define ME_NOTEOFF 0x80
#define ME_NOTEON 0x90
#define ME_KEYPRESSURE 0xA0
#define ME_CONTROLCHANGE 0xB0
#define ME_PROGRAM 0xC0
#define ME_CHANNELPRESSURE 0xD0
#define ME_PITCHWHEEL 0xE0
enum
{
ME_NOTEOFF = 0x80,
ME_NOTEON = 0x90,
ME_KEYPRESSURE = 0xA0,
ME_CONTROLCHANGE = 0xB0,
ME_PROGRAM = 0xC0,
ME_CHANNELPRESSURE = 0xD0,
ME_PITCHWHEEL = 0xE0
};
/* Controllers */
#define CTRL_BANK_SELECT 0
#define CTRL_DATA_ENTRY 6
#define CTRL_VOLUME 7
#define CTRL_PAN 10
#define CTRL_EXPRESSION 11
#define CTRL_SUSTAIN 64
#define CTRL_HARMONICCONTENT 71
#define CTRL_RELEASETIME 72
#define CTRL_ATTACKTIME 73
#define CTRL_BRIGHTNESS 74
#define CTRL_REVERBERATION 91
#define CTRL_CHORUSDEPTH 93
#define CTRL_NRPN_LSB 98
#define CTRL_NRPN_MSB 99
#define CTRL_RPN_LSB 100
#define CTRL_RPN_MSB 101
#define CTRL_ALL_SOUNDS_OFF 120
#define CTRL_RESET_CONTROLLERS 121
#define CTRL_ALL_NOTES_OFF 123
/* NRPNs */
#define NRPN_BRIGHTNESS 0x00A0
#define NRPN_HARMONICCONTENT 0x00A1
#define NRPN_DRUMVOLUME (26<<7) // drum number in low 7 bits
#define NRPN_DRUMPANPOT (28<<7) // "
#define NRPN_DRUMREVERBERATION (29<<7) // "
#define NRPN_DRUMCHORUSDEPTH (30<<7) // "
enum
{
CTRL_BANK_SELECT = 0,
CTRL_DATA_ENTRY = 6,
CTRL_VOLUME = 7,
CTRL_PAN = 10,
CTRL_EXPRESSION = 11,
CTRL_SUSTAIN = 64,
CTRL_HARMONICCONTENT = 71,
CTRL_RELEASETIME = 72,
CTRL_ATTACKTIME = 73,
CTRL_BRIGHTNESS = 74,
CTRL_REVERBERATION = 91,
CTRL_CHORUSDEPTH = 93,
CTRL_NRPN_LSB = 98,
CTRL_NRPN_MSB = 99,
CTRL_RPN_LSB = 100,
CTRL_RPN_MSB = 101,
CTRL_ALL_SOUNDS_OFF = 120,
CTRL_RESET_CONTROLLERS = 121,
CTRL_ALL_NOTES_OFF = 123
};
/* RPNs */
#define RPN_PITCH_SENS 0x0000
#define RPN_FINE_TUNING 0x0001
#define RPN_COARSE_TUNING 0x0002
#define RPN_RESET 0x3fff
#define SFX_BANKTYPE 64
enum
{
RPN_PITCH_SENS = 0x0000,
RPN_FINE_TUNING = 0x0001,
RPN_COARSE_TUNING = 0x0002,
RPN_RESET = 0x3fff
};
struct Channel
{
int
bank, program, sustain, pitchbend,
mono, /* one note only on this channel -- not implemented yet */
/* new stuff */
variationbank, reverberation, chorusdepth, harmoniccontent,
releasetime, attacktime, brightness, kit, sfx,
/* end new */
pitchsens;
WORD
volume, expression;
@ -404,84 +452,71 @@ struct Channel
rpn, nrpn;
bool
nrpn_mode;
char
transpose;
float
pitchfactor; /* precomputed pitch bend factor to save some fdiv's */
};
/* Causes the instrument's default panning to be used. */
#define NO_PANNING -1
#define NO_PANNING -1
/* envelope points */
#define MAXPOINT 7
#define MAXPOINT 6
struct Voice
{
BYTE
status, channel, note, velocity, clone_type;
status, channel, note, velocity;
Sample *sample;
Sample *left_sample;
Sample *right_sample;
int clone_voice;
float
orig_frequency, frequency;
int
sample_offset, loop_start, loop_end;
int
envelope_volume, modulation_volume;
int
envelope_target, modulation_target;
int
tremolo_sweep, tremolo_sweep_position, tremolo_phase,
lfo_sweep, lfo_sweep_position, lfo_phase,
vibrato_sweep, vibrato_sweep_position, vibrato_depth,
echo_delay_count;
int
echo_delay,
sample_increment,
envelope_increment,
modulation_increment,
tremolo_phase_increment,
lfo_phase_increment;
sample_offset, sample_increment,
envelope_volume, envelope_target, envelope_increment,
tremolo_sweep, tremolo_sweep_position,
tremolo_phase, tremolo_phase_increment,
vibrato_sweep, vibrato_sweep_position;
final_volume_t left_mix, right_mix;
float
left_amp, right_amp,
volume, tremolo_volume, lfo_volume;
left_amp, right_amp, tremolo_volume;
int
vibrato_sample_increment[VIBRATO_SAMPLE_INCREMENTS];
int
envelope_rate[MAXPOINT], envelope_offset[MAXPOINT];
int
vibrato_phase, vibrato_control_ratio, vibrato_control_counter,
envelope_stage, modulation_stage, control_counter,
modulation_delay, modulation_counter, panning, panned;
envelope_stage, control_counter, panning, panned;
};
/* Voice status options: */
#define VOICE_FREE 0
#define VOICE_ON 1
#define VOICE_SUSTAINED 2
#define VOICE_OFF 3
#define VOICE_DIE 4
enum
{
VOICE_FREE,
VOICE_ON,
VOICE_SUSTAINED,
VOICE_OFF,
VOICE_DIE
};
/* Voice panned options: */
#define PANNED_MYSTERY 0
#define PANNED_LEFT 1
#define PANNED_RIGHT 2
#define PANNED_CENTER 3
enum
{
PANNED_MYSTERY,
PANNED_LEFT,
PANNED_RIGHT,
PANNED_CENTER
};
/* Anything but PANNED_MYSTERY only uses the left volume */
/* Envelope stages: */
#define ATTACK 0
#define HOLD 1
#define DECAY 2
#define RELEASE 3
#define RELEASEB 4
#define RELEASEC 5
#define DELAY 6
enum
{
ATTACK,
HOLD,
DECAY,
RELEASE,
RELEASEB,
RELEASEC
};
#define ISDRUMCHANNEL(c) ((drumchannels & (1<<(c))))
@ -498,9 +533,7 @@ tables.h
#define sine(x) (sin((2*PI/1024.0) * (x)))
#define note_to_freq(x) (float(8175.7989473096690661233836992789 * pow(2.0, (x) / 12.0)))
#define calc_vol(x) (pow(2.0,((x)*6.0 - 6.0)))
#define XMAPMAX 800
#define calc_vol(x) (pow(2.0,((x)*6.0 - 6.0)))
/*
timidity.h
@ -517,27 +550,16 @@ struct Renderer
{
float rate;
DLS_Data *patches;
InstrumentLayer *default_instrument;
Instrument *default_instrument;
int default_program;
bool fast_decay;
int resample_buffer_size;
sample_t *resample_buffer;
Channel channel[16];
Voice voice[MAX_VOICES];
signed char drumvolume[MAXCHAN][MAXNOTE];
signed char drumpanpot[MAXCHAN][MAXNOTE];
signed char drumreverberation[MAXCHAN][MAXNOTE];
signed char drumchorusdepth[MAXCHAN][MAXNOTE];
int control_ratio, amp_with_poly;
int drumchannels;
int adjust_panning_immediately;
int voices;
int GM_System_On;
int XG_System_On;
int GS_System_On;
int XG_System_reverb_type;
int XG_System_chorus_type;
int XG_System_variation_type;
int lost_notes, cut_notes;
Renderer(float sample_rate);
@ -558,10 +580,8 @@ struct Renderer
int convert_vibrato_rate(BYTE rate);
void recompute_amp(Voice *v);
int vc_alloc(int not_this_voice);
void kill_others(int voice);
void clone_voice(Instrument *ip, int v, int note, int vel, int clone_type, int variationbank);
void xremap(int *banknumpt, int *this_notept, int this_kit);
void kill_key_group(int voice);
float calculate_scaled_frequency(Sample *sample, int note);
void start_note(int chan, int note, int vel, int voice);
void note_on(int chan, int note, int vel);
@ -579,7 +599,6 @@ struct Renderer
void reset_midi();
void select_sample(int voice, Instrument *instr, int vel);
void select_stereo_samples(int voice, InstrumentLayer *layer, int vel);
void recompute_freq(int voice);
void kill_note(int voice);