📄 siosc.alg
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(SIOSC-ALG(NAME "siosc"); wave tables are provided in the argument "lis" as follows:; (tab0 t1 tab1 t2 tab2 t3 tab3 ... tN tabN); where tab0 is the initial table, the table is linearly interpolated until; sample t1, at which point the table is tab1. From there, tab1 is interpolated; to tab2 at t2, etc. The sound stops at the terminate time of s_fm, so; if that comes before tN, tabN is used for the remainder of the sound.; t1, t2, ... tN are fixnums with sample counts(ARGUMENTS ("LVAL" "lis") ("rate_type" "sr") ("double" "hz") ("time_type" "t0") ("sound_type" "s_fm"))(SUPPORT-FUNCTIONS "/* sisosc_table_init -- set up first two tables for interpolation *//**/void siosc_table_init(siosc_susp_type susp){ sound_type snd; if (!susp->lis) xlfail(\"bad table list in SIOSC\"); snd = getsound(car(susp->lis)); susp->table_b_ptr_ptr = sound_to_table(snd); susp->table_b_samps = susp->table_b_ptr_ptr->samples; susp->lis = cdr(susp->lis); susp->table_sr = snd->sr;}/* siosc_table_update -- outer loop processing, get next table *//**/long siosc_table_update(siosc_susp_type susp, long cur){ long n; /* swap ampramps: */ susp->ampramp_a = 1.0; susp->ampramp_b = 0.0; /* swap tables: */ table_unref(susp->table_a_ptr); susp->table_a_ptr = susp->table_b_ptr_ptr; susp->table_a_samps = susp->table_b_samps; susp->table_b_ptr_ptr = NULL; /* so we do not try to unref it */ if (susp->lis) { sound_type snd; /* compute slope */ susp->next_breakpoint = getfixnum(car(susp->lis)); susp->lis = cdr(susp->lis); n = susp->next_breakpoint - cur; susp->ampslope = 1.0 / n; /* build new table: */ if (!susp->lis) xlfail(\"bad table list in SIOSC\"); snd = getsound(car(susp->lis)); susp->table_b_ptr_ptr = sound_to_table(snd); susp->table_b_samps = susp->table_b_ptr_ptr->samples; if (susp->table_b_ptr_ptr->length != susp->table_len || susp->table_sr != snd->sr) xlfail(\"mismatched tables passed to SIOSC\") ; susp->lis = cdr(susp->lis); } else { /* use only table a */ susp->ampslope = 0.0; susp->next_breakpoint = 0x7FFFFFFF; n = 0x7FFFFFFF; } return n;}")(STATE ("double" "table_len" "0.0") ("double" "ph_incr" "0.0") ("table_type" "table_a_ptr" "NULL") ("table_type" "table_b_ptr_ptr" "NULL") ("sample_type *" "table_a_samps" "NULL") ("sample_type *" "table_b_samps" "NULL") ("double" "table_sr" "0.0") ("double" "phase" "0.0") ("LVAL" "lis" "lis") ("long" "next_breakpoint" "0") ("double" "ampramp_a" "1.0") ("double" "ampramp_b" "0.0") ("double" "ampslope" "0.0; siosc_table_init(susp); susp->ph_incr = hz * susp->table_len / sr; s_fm->scale = (sample_type) (s_fm->scale * (susp->table_len / sr))") )(ALWAYS-SCALE s_fm)(INLINE-INTERPOLATION T) ; so that modulation can be low frequency(STEP-FUNCTION s_fm)(TERMINATE (MIN s_fm))(LOGICAL-STOP (MIN s_fm))(INNER-LOOP-LOCALS " long table_index; double xa, xb;"); Implementation notes:; "lis" points to the next time to be used, or NULL(OUTER-LOOP " { long cur = susp->susp.current + cnt; n = susp->next_breakpoint - cur; if (n == 0) n = siosc_table_update(susp, cur); } togo = MIN(n, togo);")(INNER-LOOP " table_index = (long) phase; xa = table_a_samps[table_index]; xb = table_b_samps[table_index]; output = (sample_type) (ampramp_a * (xa + (phase - table_index) * (table_a_samps[table_index + 1] - xa)) + ampramp_b * (xb + (phase - table_index) * (table_b_samps[table_index + 1] - xb))); ampramp_a -= ampslope; ampramp_b += ampslope; phase += ph_incr + s_fm; while (phase > table_len) phase -= table_len; /* watch out for negative frequencies! */ while (phase < 0) phase += table_len")(CONSTANT "ph_incr" "table_len" "table_ptr" "table_a_samps" "table_b_samps" "ampslope" "table_a_ptr" "table_b_ptr_ptr")(SAMPLE-RATE "sr")(FINALIZATION " table_unref(susp->table_a_ptr); table_unref(susp->table_b_ptr_ptr);"))
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