/* audio.c vi:se ts=3 sw=3: */ /* $Id: audio.c,v 4.2 1994/02/04 14:53:03 espie Exp espie $ * $Log: audio.c,v $ * Revision 4.2 1994/02/04 14:53:03 espie * *** empty log message *** * * Revision 4.1 1994/01/12 16:10:20 espie * Fixed up last minute problems. * * Revision 4.0 1994/01/11 17:39:36 espie * Lots of changes. * * Revision 1.5 1994/01/09 17:36:22 Espie * Generalized open.c. * * Revision 1.4 1994/01/08 03:55:43 Espie * Just new names, provide for better ! * * Revision 1.3 1994/01/05 14:54:09 Espie * *** empty log message *** * * Revision 1.2 1993/12/28 13:54:44 Espie * Finally corrected irksome probleme with MAX_PITCH. * * Revision 1.1 1993/12/26 00:55:53 Espie * Initial revision * * Revision 3.10 1993/12/04 16:12:50 espie * First changes to augment the number of channels. * * Revision 3.9 1993/11/17 15:30:16 espie * Added one level of abstraction. Look through player.c for the * old high-level functions. * * Revision 3.8 1993/08/04 11:34:33 espie * *** empty log message *** * * Revision 3.7 1993/07/18 10:39:44 espie * Cleaned up some code. * * Revision 3.6 1992/11/27 10:29:00 espie * General cleanup * * Revision 3.5 1992/11/24 10:51:19 espie * Optimized output and fixed up some details. * Unrolled code for oversample = 1 to be more efficient at * higher frequency (since a high frequency is better than * a higher oversample). * * Revision 3.4 1992/11/23 17:18:59 espie * *** empty log message *** * * Revision 3.3 1992/11/23 10:12:23 espie * Fixed up BIG bug. * * Revision 3.2 1992/11/20 14:53:32 espie * Added finetune. * * Revision 3.1 1992/11/19 20:44:47 espie * Protracker commands. * * Revision 3.0 1992/11/18 16:08:05 espie * New release. * * Revision 2.14 1992/11/06 19:31:53 espie * Fixed missing parameter type. * fix_xxx for better speed. * set_volume. * Added possibility to get back to MONO for the sgi. * Added stereo capabilities to the indigo version. * Minor bug: a SAMPLE_FAULT is a minor error, * we should first check that there was no other * error before setting it. * New resample function coming from the player. * Added more notes. * * Revision 2.1 1991/11/17 23:07:58 espie * Just computes some frequency-related parameters. * * */ #include #ifndef __386BSD__ #include #else #include #endif #include #include "defs.h" #include "song.h" #include "channel.h" #include "tags.h" #include "extern.h" ID("$Id: audio.c,v 4.2 1994/02/04 14:53:03 espie Exp espie $") /* DO_NOTHING is also used for the automaton */ #define DO_NOTHING 0 #define PLAY 1 #define REPLAY 2 #define MAX_CHANNELS 8 LOCAL struct audio_channel { struct sample_info *samp; int mode; unsigned long pointer; unsigned long step; int volume; int pitch; } chan[MAX_CHANNELS]; LOCAL struct sample_info dummy = { NULL, 0, 0, 0, 0, 0, 0, 0, NULL, NULL }; LOCAL int allocated = 0; struct audio_channel *new_channel_tag_list(prop) struct tag *prop; { struct audio_channel *new; new = &chan[allocated++]; new->mode = DO_NOTHING; new->pointer = 0; new->step = 0; new->pitch = 0; new->volume = 0; new->samp = &dummy; return new; } void release_audio_channels() { allocated = 0; } /* Have to get some leeway for vibrato (since we don't bound pitch with * vibrato). This is conservative. */ #define VIB_MAXDEPTH 150 #define C fix_to_int(ch->pointer) LOCAL unsigned long step_table[REAL_MAX_PITCH + VIB_MAXDEPTH]; /* holds the increment for finding the next sampled * byte at a given pitch (see resample() ). */ /* creates a table for converting ``amiga'' pitch * to a step rate at a given resampling frequency. * For accuracy, we don't use floating point, but * instead fixed point ( << ACCURACY). * IMPORTANT NOTE: we need to make it fit within 32 bits (long), which * must be enough for ACCURACY + log2(max sample length) */ LOCAL void create_step_table(oversample, output_fr) int oversample; /* we sample oversample i for each byte output */ int output_fr; /* output frequency */ { double note_fr; /* note frequency (in Hz) */ double step; int pitch; /* amiga pitch */ step_table[0] = 0; for (pitch = 1; pitch < REAL_MAX_PITCH + VIB_MAXDEPTH; pitch++) { note_fr = AMIGA_CLOCKFREQ / pitch; /* int_to_fix(1) is the normalizing factor */ step = note_fr / output_fr * int_to_fix(1) / oversample; step_table[pitch] = (long)step; } } LOCAL void readjust_pitch() { int i; for (i = 0; i < allocated; i++) chan[i].step = step_table[chan[i].pitch]; } void init_tables(oversample, frequency) int oversample, frequency; { create_step_table(oversample, frequency); readjust_pitch(); } /* The playing mechanism itself. * According to the current channel automaton, * we resample the instruments in real time to * generate output. */ void resample(oversample, number) int oversample; int number; { int i; /* sample counter */ int channel; /* channel counter */ int sampling; /* oversample counter */ int value[NUMBER_TRACKS]; /* recombinations of the various data */ struct audio_channel *ch; /* safety check: we can't have a segv there, provided * chan points to a valid sample. * For `empty' samples, what is needed is fix_length = 0 * and rp_start = NULL */ /* special case optimization */ if (oversample == 1) { /* do the resampling, i.e., actually play sounds */ for (i = 0; i < number; i++) { for (channel = 0; channel < allocated; channel++) { ch = chan + channel; switch(ch->mode) { case DO_NOTHING: value[channel] = 0; break; case PLAY: /* Since we now have fix_length, we can * do that check with improved performance */ if (ch->pointer < ch->samp->fix_length) { value[channel] = ch->samp->start[C] * ch->volume; ch->pointer += ch->step; break; } else { ch->mode = REPLAY; ch->pointer -= ch->samp->fix_length; /* FALLTHRU */ } case REPLAY: /* is there a replay ? */ if (!ch->samp->rp_start) { ch->mode = DO_NOTHING; break; } while (ch->pointer >= ch->samp->fix_rp_length) ch->pointer -= ch->samp->fix_rp_length; value[channel] = ch->samp->rp_start[C] * ch->volume; ch->pointer += ch->step; break; } } output_samples(value[0]+value[3], value[1]+value[2]); } } else { for (i = 0; i < number; i++) { for (channel = 0; channel < allocated; channel++) { value[channel] = 0; for (sampling = 0; sampling < oversample; sampling++) { ch = chan + channel; switch(ch->mode) { case DO_NOTHING: break; case PLAY: /* Since we now have fix_length, we can * do that check with improved performance */ if (ch->pointer < ch->samp->fix_length) { value[channel] += ch->samp->start[C] * ch->volume; ch->pointer += ch->step; break; } else { ch->mode = REPLAY; ch->pointer -= ch->samp->fix_length; /* FALLTHRU */ } case REPLAY: /* is there a replay ? */ if (!ch->samp->rp_start) { ch->mode = DO_NOTHING; break; } while (ch->pointer >= ch->samp->fix_rp_length) ch->pointer -= ch->samp->fix_rp_length; value[channel] += ch->samp->rp_start[C] * ch->volume; ch->pointer += ch->step; break; } } } output_samples((value[0]+value[3])/oversample, (value[1]+value[2])/oversample); } } flush_buffer(); } /* setting up a given note */ void play_note(au, samp, pitch) struct audio_channel *au; struct sample_info *samp; int pitch; { au->pointer = 0; au->pitch = pitch; au->step = step_table[pitch]; if (samp) { au->samp = samp; au->mode = PLAY; } else au->mode = DO_NOTHING; } /* changing the current pitch (value * may be temporary, and not stored * in channel pitch, for instance vibratos. */ void set_play_pitch(au, pitch) struct audio_channel *au; int pitch; { /* save current pitch in case we want to change * the step table on the run */ au->pitch = pitch; au->step = step_table[pitch]; } /* changing the current volume. You HAVE to get through * there so that it will work on EVERY machine. */ void set_play_volume(au, volume) struct audio_channel *au; int volume; { au->volume = volume; } void set_play_position(au, pos) struct audio_channel *au; int pos; { au->pointer = int_to_fix(pos); }