Mercurial > SDL_sound_CoreAudio
view decoders/timidity/resample.c @ 474:c66080364dff
Most decoders now report total sample play time, now. Technically, this
breaks binary compatibility with the 1.0 branch, since it extends the
Sound_Sample struct, but most (all?) programs are just passing pointers
allocated by SDL_sound around, and might be okay.
Source-level compatibility is not broken...yet! :)
--ryan.
-------- Original Message --------
Subject: SDL_sound patch: Finding total length of time of sound file.
Date: Sun, 26 Jan 2003 09:31:17 -0800 (PST)
Hi Ryan,
I am working with Eric Wing and helping him modify
SDL_sound. AS part of our efforts in improving and
enhancing SDL_sound, we like to submit this patch. We
modified the codecs to find the total time of a sound
file. Below is the explanation of the patch. The
patch is appended as an attachment to this email.
* MOTIVATION:
We needed the ability to get the total play time of a
sample (And we noticed that we're not the only ones).
Since SDL_sound blocks direct access to the specific
decoders, there is no way for a user to know this
information short of decoding the whole thing.
Because of this, we believe this will be a useful
addition, even though the accuracy may not be perfect
(subject to each decoder) or the information may not
always be available.
* CONTRIBUTORS:
Wesley Leong (modified the majority of the codecs and
verified the results)
Eric Wing (showed everyone how to do modify codec,
modified mikmod)
Wang Lam (modified a handful of codecs, researched
into specs and int overflow)
Ahilan Anantha (modified a few codecs and helped with
integer math)
* GENERAL ISSUES:
We chose the value to be milliseconds as an Sint32.
Milliseconds because that's what Sound_Seek takes as a
parameter and -1 to allow for instances/codecs where
the value could not be determined. We are
not sure if this is the final convention you want, so
we are willing to work with you on this.
We also expect the total_time field to be set on open
and never again modified by SDL_sound. Users may
access it directly much like the sample buffer and
buffer_size. We thought about recomputing the time
on DecodeAll, but since users may seek or decode small
chunks first, not all the data may be there. So this
is better done by the user. This may be good
information to document.
Currently, all the main codecs are implemented except
for QuickTime.
author | Ryan C. Gordon <icculus@icculus.org> |
---|---|
date | Sat, 08 May 2004 08:19:50 +0000 |
parents | a73c51c12452 |
children | f33471c47efe |
line wrap: on
line source
/* 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 <math.h> #include <stdio.h> #include <stdlib.h> #include "SDL_sound.h" #define __SDL_SOUND_INTERNAL__ #include "SDL_sound_internal.h" #include "timidity.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(MidiSong *song, int v, Sint32 *countptr) { /* Play sample until end, then free the voice. */ sample_t v1, v2; Voice *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(MidiSong *song, Voice *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(MidiSong *song, Voice *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 < 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; else return phase-VIBRATO_SAMPLE_INCREMENTS/2; } static Sint32 update_vibrato(MidiSong *song, Voice *vp, int sign) { Sint32 depth; int phase, pb; double a; 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]) { 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*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(MidiSong *song, int v, Sint32 *countptr) { /* Play sample until end, then free the voice. */ sample_t v1, v2; Voice *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(MidiSong *song, Voice *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(MidiSong *song, Voice *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(MidiSong *song, int v, Sint32 *countptr) { Sint32 ofs; Uint8 modes; Voice *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(MidiSong *song, Sample *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 SNDDBG((" * 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; }