Mercurial > audlegacy-plugins
view src/timidity/libtimidity/resample.c @ 439:84029a5250c1 trunk
[svn] - status icon plugin: made a tooltip popup that displays metadata of the current song
author | giacomo |
---|---|
date | Tue, 16 Jan 2007 10:34:12 -0800 |
parents | 3da1b8942b8b |
children | f14d11bf9cbb |
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/* TiMidity -- Experimental MIDI to WAVE converter Copyright (C) 1995 Tuukka Toivonen <toivonen@clinet.fi> This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License along with this program; if not, write to the Free Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. resample.c */ #if HAVE_CONFIG_H # include <config.h> #endif #include "audacious/vfs.h" #include <math.h> #include <stdlib.h> #include "timidity.h" #include "timidity_internal.h" #include "options.h" #include "common.h" #include "instrum.h" #include "playmidi.h" #include "tables.h" #include "resample.h" /*************** resampling with fixed increment *****************/ static sample_t *rs_plain(MidSong *song, int v, sint32 *countptr) { /* Play sample until end, then free the voice. */ sample_t v1, v2; MidVoice *vp=&(song->voice[v]); sample_t *dest=song->resample_buffer, *src=vp->sample->data; sint32 ofs=vp->sample_offset, incr=vp->sample_increment, le=vp->sample->data_length, count=*countptr; sint32 i; if (incr<0) incr = -incr; /* In case we're coming out of a bidir loop */ /* Precalc how many times we should go through the loop. NOTE: Assumes that incr > 0 and that ofs <= le */ i = (le - ofs) / incr + 1; if (i > count) { i = count; count = 0; } else count -= i; while (i--) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; } if (ofs >= le) { if (ofs == le) *dest++ = src[ofs >> FRACTION_BITS]; vp->status=VOICE_FREE; *countptr-=count+1; } vp->sample_offset=ofs; /* Update offset */ return song->resample_buffer; } static sample_t *rs_loop(MidSong *song, MidVoice *vp, sint32 count) { /* Play sample until end-of-loop, skip back and continue. */ sample_t v1, v2; sint32 ofs=vp->sample_offset, incr=vp->sample_increment, le=vp->sample->loop_end, ll=le - vp->sample->loop_start; sample_t *dest=song->resample_buffer, *src=vp->sample->data; sint32 i; while (count) { if (ofs >= le) /* NOTE: Assumes that ll > incr and that incr > 0. */ ofs -= ll; /* Precalc how many times we should go through the loop */ i = (le - ofs) / incr + 1; if (i > count) { i = count; count = 0; } else count -= i; while (i--) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; } } vp->sample_offset=ofs; /* Update offset */ return song->resample_buffer; } static sample_t *rs_bidir(MidSong *song, MidVoice *vp, sint32 count) { sample_t v1, v2; sint32 ofs=vp->sample_offset, incr=vp->sample_increment, le=vp->sample->loop_end, ls=vp->sample->loop_start; sample_t *dest=song->resample_buffer, *src=vp->sample->data; sint32 le2 = le<<1, ls2 = ls<<1, i; /* Play normally until inside the loop region */ if (ofs <= ls) { /* NOTE: Assumes that incr > 0, which is NOT always the case when doing bidirectional looping. I have yet to see a case where both ofs <= ls AND incr < 0, however. */ i = (ls - ofs) / incr + 1; if (i > count) { i = count; count = 0; } else count -= i; while (i--) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; } } /* Then do the bidirectional looping */ while(count) { /* Precalc how many times we should go through the loop */ i = ((incr > 0 ? le : ls) - ofs) / incr + 1; if (i > count) { i = count; count = 0; } else count -= i; while (i--) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; } if (ofs>=le) { /* fold the overshoot back in */ ofs = le2 - ofs; incr *= -1; } else if (ofs <= ls) { ofs = ls2 - ofs; incr *= -1; } } vp->sample_increment=incr; vp->sample_offset=ofs; /* Update offset */ return song->resample_buffer; } /*********************** vibrato versions ***************************/ /* We only need to compute one half of the vibrato sine cycle */ static int vib_phase_to_inc_ptr(int phase) { if (phase < MID_VIBRATO_SAMPLE_INCREMENTS/2) return MID_VIBRATO_SAMPLE_INCREMENTS/2-1-phase; else if (phase >= 3*MID_VIBRATO_SAMPLE_INCREMENTS/2) return 5*MID_VIBRATO_SAMPLE_INCREMENTS/2-1-phase; else return phase-MID_VIBRATO_SAMPLE_INCREMENTS/2; } static sint32 update_vibrato(MidSong *song, MidVoice *vp, int sign) { sint32 depth; int phase, pb; double a; if (vp->vibrato_phase++ >= 2*MID_VIBRATO_SAMPLE_INCREMENTS-1) vp->vibrato_phase=0; phase=vib_phase_to_inc_ptr(vp->vibrato_phase); if (vp->vibrato_sample_increment[phase]) { if (sign) return -vp->vibrato_sample_increment[phase]; else return vp->vibrato_sample_increment[phase]; } /* Need to compute this sample increment. */ depth=vp->sample->vibrato_depth<<7; if (vp->vibrato_sweep) { /* Need to update sweep */ vp->vibrato_sweep_position += vp->vibrato_sweep; if (vp->vibrato_sweep_position >= (1<<SWEEP_SHIFT)) vp->vibrato_sweep=0; else { /* Adjust depth */ depth *= vp->vibrato_sweep_position; depth >>= SWEEP_SHIFT; } } a = FSCALE(((double)(vp->sample->sample_rate) * (double)(vp->frequency)) / ((double)(vp->sample->root_freq) * (double)(song->rate)), FRACTION_BITS); pb=(int)((sine(vp->vibrato_phase * (SINE_CYCLE_LENGTH/(2*MID_VIBRATO_SAMPLE_INCREMENTS))) * (double)(depth) * VIBRATO_AMPLITUDE_TUNING)); if (pb<0) { pb=-pb; a /= bend_fine[(pb>>5) & 0xFF] * bend_coarse[pb>>13]; } else a *= bend_fine[(pb>>5) & 0xFF] * bend_coarse[pb>>13]; /* If the sweep's over, we can store the newly computed sample_increment */ if (!vp->vibrato_sweep) vp->vibrato_sample_increment[phase]=(sint32) a; if (sign) a = -a; /* need to preserve the loop direction */ return (sint32) a; } static sample_t *rs_vib_plain(MidSong *song, int v, sint32 *countptr) { /* Play sample until end, then free the voice. */ sample_t v1, v2; MidVoice *vp=&(song->voice[v]); sample_t *dest=song->resample_buffer, *src=vp->sample->data; sint32 le=vp->sample->data_length, ofs=vp->sample_offset, incr=vp->sample_increment, count=*countptr; int cc=vp->vibrato_control_counter; /* This has never been tested */ if (incr<0) incr = -incr; /* In case we're coming out of a bidir loop */ while (count--) { if (!cc--) { cc=vp->vibrato_control_ratio; incr=update_vibrato(song, vp, 0); } v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; if (ofs >= le) { if (ofs == le) *dest++ = src[ofs >> FRACTION_BITS]; vp->status=VOICE_FREE; *countptr-=count+1; break; } } vp->vibrato_control_counter=cc; vp->sample_increment=incr; vp->sample_offset=ofs; /* Update offset */ return song->resample_buffer; } static sample_t *rs_vib_loop(MidSong *song, MidVoice *vp, sint32 count) { /* Play sample until end-of-loop, skip back and continue. */ sample_t v1, v2; sint32 ofs=vp->sample_offset, incr=vp->sample_increment, le=vp->sample->loop_end, ll=le - vp->sample->loop_start; sample_t *dest=song->resample_buffer, *src=vp->sample->data; int cc=vp->vibrato_control_counter; sint32 i; int vibflag=0; while (count) { /* Hopefully the loop is longer than an increment */ if(ofs >= le) ofs -= ll; /* Precalc how many times to go through the loop, taking the vibrato control ratio into account this time. */ i = (le - ofs) / incr + 1; if(i > count) i = count; if(i > cc) { i = cc; vibflag = 1; } else cc -= i; count -= i; while(i--) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; } if(vibflag) { cc = vp->vibrato_control_ratio; incr = update_vibrato(song, vp, 0); vibflag = 0; } } vp->vibrato_control_counter=cc; vp->sample_increment=incr; vp->sample_offset=ofs; /* Update offset */ return song->resample_buffer; } static sample_t *rs_vib_bidir(MidSong *song, MidVoice *vp, sint32 count) { sample_t v1, v2; sint32 ofs=vp->sample_offset, incr=vp->sample_increment, le=vp->sample->loop_end, ls=vp->sample->loop_start; sample_t *dest=song->resample_buffer, *src=vp->sample->data; int cc=vp->vibrato_control_counter; sint32 le2=le<<1, ls2=ls<<1, i; int vibflag = 0; /* Play normally until inside the loop region */ while (count && (ofs <= ls)) { i = (ls - ofs) / incr + 1; if (i > count) i = count; if (i > cc) { i = cc; vibflag = 1; } else cc -= i; count -= i; while (i--) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; } if (vibflag) { cc = vp->vibrato_control_ratio; incr = update_vibrato(song, vp, 0); vibflag = 0; } } /* Then do the bidirectional looping */ while (count) { /* Precalc how many times we should go through the loop */ i = ((incr > 0 ? le : ls) - ofs) / incr + 1; if(i > count) i = count; if(i > cc) { i = cc; vibflag = 1; } else cc -= i; count -= i; while (i--) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS)+1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); ofs += incr; } if (vibflag) { cc = vp->vibrato_control_ratio; incr = update_vibrato(song, vp, (incr < 0)); vibflag = 0; } if (ofs >= le) { /* fold the overshoot back in */ ofs = le2 - ofs; incr *= -1; } else if (ofs <= ls) { ofs = ls2 - ofs; incr *= -1; } } vp->vibrato_control_counter=cc; vp->sample_increment=incr; vp->sample_offset=ofs; /* Update offset */ return song->resample_buffer; } sample_t *resample_voice(MidSong *song, int v, sint32 *countptr) { sint32 ofs; uint8 modes; MidVoice *vp=&(song->voice[v]); if (!(vp->sample->sample_rate)) { /* Pre-resampled data -- just update the offset and check if we're out of data. */ ofs=vp->sample_offset >> FRACTION_BITS; /* Kind of silly to use FRACTION_BITS here... */ if (*countptr >= (vp->sample->data_length>>FRACTION_BITS) - ofs) { /* Note finished. Free the voice. */ vp->status = VOICE_FREE; /* Let the caller know how much data we had left */ *countptr = (vp->sample->data_length>>FRACTION_BITS) - ofs; } else vp->sample_offset += *countptr << FRACTION_BITS; return vp->sample->data+ofs; } /* Need to resample. Use the proper function. */ modes=vp->sample->modes; if (vp->vibrato_control_ratio) { if ((modes & MODES_LOOPING) && ((modes & MODES_ENVELOPE) || (vp->status==VOICE_ON || vp->status==VOICE_SUSTAINED))) { if (modes & MODES_PINGPONG) return rs_vib_bidir(song, vp, *countptr); else return rs_vib_loop(song, vp, *countptr); } else return rs_vib_plain(song, v, countptr); } else { if ((modes & MODES_LOOPING) && ((modes & MODES_ENVELOPE) || (vp->status==VOICE_ON || vp->status==VOICE_SUSTAINED))) { if (modes & MODES_PINGPONG) return rs_bidir(song, vp, *countptr); else return rs_loop(song, vp, *countptr); } else return rs_plain(song, v, countptr); } } void pre_resample(MidSong *song, MidSample *sp) { double a, xdiff; sint32 incr, ofs, newlen, count; sint16 *newdata, *dest, *src = (sint16 *) sp->data; sint16 v1, v2, v3, v4, *vptr; #ifdef DEBUG_CHATTER static const char note_name[12][3] = { "C", "C#", "D", "D#", "E", "F", "F#", "G", "G#", "A", "A#", "B" }; #endif DEBUG_MSG(" * pre-resampling for note %d (%s%d)\n", sp->note_to_use, note_name[sp->note_to_use % 12], (sp->note_to_use & 0x7F) / 12); a = ((double) (sp->sample_rate) * freq_table[(int) (sp->note_to_use)]) / ((double) (sp->root_freq) * song->rate); newlen = (sint32)(sp->data_length / a); dest = newdata = safe_malloc(newlen >> (FRACTION_BITS - 1)); count = (newlen >> FRACTION_BITS) - 1; ofs = incr = (sp->data_length - (1 << FRACTION_BITS)) / count; if (--count) *dest++ = src[0]; /* Since we're pre-processing and this doesn't have to be done in real-time, we go ahead and do the full sliding cubic interpolation. */ while (--count) { vptr = src + (ofs >> FRACTION_BITS); /* * Electric Fence to the rescue: Accessing *(vptr - 1) is not a * good thing to do when vptr <= src. (TiMidity++ has a similar * safe-guard here.) */ v1 = (vptr > src) ? *(vptr - 1) : 0; v2 = *vptr; v3 = *(vptr + 1); v4 = *(vptr + 2); xdiff = FSCALENEG(ofs & FRACTION_MASK, FRACTION_BITS); *dest++ = (sint16)(v2 + (xdiff / 6.0) * (-2 * v1 - 3 * v2 + 6 * v3 - v4 + xdiff * (3 * (v1 - 2 * v2 + v3) + xdiff * (-v1 + 3 * (v2 - v3) + v4)))); ofs += incr; } if (ofs & FRACTION_MASK) { v1 = src[ofs >> FRACTION_BITS]; v2 = src[(ofs >> FRACTION_BITS) + 1]; *dest++ = v1 + (((v2 - v1) * (ofs & FRACTION_MASK)) >> FRACTION_BITS); } else *dest++ = src[ofs >> FRACTION_BITS]; sp->data_length = newlen; sp->loop_start = (sint32)(sp->loop_start / a); sp->loop_end = (sint32)(sp->loop_end / a); free(sp->data); sp->data = (sample_t *) newdata; sp->sample_rate = 0; }