siosc.c

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	if (n) do { /* the inner sample computation loop */
	    long table_index;
            double xa, xb;
	    if (s_fm_pHaSe_ReG >= 1.0) {
/* fixup-depends s_fm */
		/* pick up next sample as s_fm_x1_sample: */
		susp->s_fm_ptr++;
		susp_took(s_fm_cnt, 1);
		s_fm_pHaSe_ReG -= 1.0;
		susp_check_term_log_samples_break(s_fm, s_fm_ptr, s_fm_cnt, s_fm_x1_sample_reg);
		s_fm_x1_sample_reg = susp_current_sample(s_fm, s_fm_ptr);
	    }
	    table_index = (long) phase_reg;
            xa = table_a_samps_reg[table_index];
            xb = table_b_samps_reg[table_index];
            *out_ptr_reg++ = (sample_type) 
                      (ampramp_a_reg * (xa + (phase_reg - table_index) * 
                          (table_a_samps_reg[table_index + 1] - xa)) +
                       ampramp_b_reg * (xb + (phase_reg - table_index) * 
                          (table_b_samps_reg[table_index + 1] - xb)));
            ampramp_a_reg -= ampslope_reg;
            ampramp_b_reg += ampslope_reg;
            phase_reg += ph_incr_reg + s_fm_x1_sample_reg;
            while (phase_reg > table_len_reg) phase_reg -= table_len_reg;
            /* watch out for negative frequencies! */
            while (phase_reg < 0) phase_reg += table_len_reg;
	    s_fm_pHaSe_ReG += s_fm_pHaSe_iNcR_rEg;
	} while (--n); /* inner loop */

	togo -= n;
	susp->phase = phase_reg;
	susp->ampramp_a = ampramp_a_reg;
	susp->ampramp_b = ampramp_b_reg;
	susp->s_fm_pHaSe = s_fm_pHaSe_ReG;
	susp->s_fm_x1_sample = s_fm_x1_sample_reg;
	out_ptr += togo;
	cnt += togo;
    } /* outer loop */

    /* test for termination */
    if (togo == 0 && cnt == 0) {
	snd_list_terminate(snd_list);
    } else {
	snd_list->block_len = cnt;
	susp->susp.current += cnt;
    }
    /* test for logical stop */
    if (susp->logically_stopped) {
	snd_list->logically_stopped = true;
    } else if (susp->susp.log_stop_cnt == susp->susp.current) {
	susp->logically_stopped = true;
    }
} /* siosc_i_fetch */


void siosc_r_fetch(register siosc_susp_type susp, snd_list_type snd_list)
{
    int cnt = 0; /* how many samples computed */
    sample_type s_fm_val;
    int togo;
    int n;
    sample_block_type out;
    register sample_block_values_type out_ptr;

    register sample_block_values_type out_ptr_reg;

    register double table_len_reg;
    register double ph_incr_reg;
    register sample_type * table_a_samps_reg;
    register sample_type * table_b_samps_reg;
    register double phase_reg;
    register double ampramp_a_reg;
    register double ampramp_b_reg;
    register double ampslope_reg;
    falloc_sample_block(out, "siosc_r_fetch");
    out_ptr = out->samples;
    snd_list->block = out;

    /* make sure sounds are primed with first values */
    if (!susp->started) {
	susp->started = true;
	susp->s_fm_pHaSe = 1.0;
    }

    susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);

    while (cnt < max_sample_block_len) { /* outer loop */
	/* first compute how many samples to generate in inner loop: */
	/* don't overflow the output sample block: */
	togo = max_sample_block_len - cnt;

	/* grab next s_fm_x1_sample when phase goes past 1.0; */
	/* use s_fm_n (computed below) to avoid roundoff errors: */
	if (susp->s_fm_n <= 0) {
	    susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
	    susp->s_fm_x1_sample = susp_fetch_sample(s_fm, s_fm_ptr, s_fm_cnt);
	    susp->s_fm_pHaSe -= 1.0;
	    /* s_fm_n gets number of samples before phase exceeds 1.0: */
	    susp->s_fm_n = (long) ((1.0 - susp->s_fm_pHaSe) *
					susp->output_per_s_fm);
	}
	togo = MIN(togo, susp->s_fm_n);
	s_fm_val = susp->s_fm_x1_sample;
	/* don't run past terminate time */
	if (susp->terminate_cnt != UNKNOWN &&
	    susp->terminate_cnt <= susp->susp.current + cnt + togo) {
	    togo = susp->terminate_cnt - (susp->susp.current + cnt);
	    if (togo == 0) break;
	}


	/* don't run past logical stop time */
	if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
	    int to_stop = susp->susp.log_stop_cnt - (susp->susp.current + cnt);
	    /* break if to_stop == 0 (we're at the logical stop)
	     * AND cnt > 0 (we're not at the beginning of the
	     * output block).
	     */
	    if (to_stop < togo) {
		if (to_stop == 0) {
		    if (cnt) {
			togo = 0;
			break;
		    } else /* keep togo as is: since cnt == 0, we
		            * can set the logical stop flag on this
		            * output block
		            */
			susp->logically_stopped = true;
		} else /* limit togo so we can start a new
		        * block at the LST
		        */
		    togo = to_stop;
	    }
	}


        { long cur = susp->susp.current + cnt;
            n = susp->next_breakpoint - cur;
            if (n == 0) n = siosc_table_update(susp, cur);
        }
        togo = MIN(n, togo);

	n = togo;
	table_len_reg = susp->table_len;
	ph_incr_reg = susp->ph_incr;
	table_a_samps_reg = susp->table_a_samps;
	table_b_samps_reg = susp->table_b_samps;
	phase_reg = susp->phase;
	ampramp_a_reg = susp->ampramp_a;
	ampramp_b_reg = susp->ampramp_b;
	ampslope_reg = susp->ampslope;
	out_ptr_reg = out_ptr;
	if (n) do { /* the inner sample computation loop */
	    long table_index;
            double xa, xb;
	    table_index = (long) phase_reg;
            xa = table_a_samps_reg[table_index];
            xb = table_b_samps_reg[table_index];
            *out_ptr_reg++ = (sample_type) 
                      (ampramp_a_reg * (xa + (phase_reg - table_index) * 
                          (table_a_samps_reg[table_index + 1] - xa)) +
                       ampramp_b_reg * (xb + (phase_reg - table_index) * 
                          (table_b_samps_reg[table_index + 1] - xb)));
            ampramp_a_reg -= ampslope_reg;
            ampramp_b_reg += ampslope_reg;
            phase_reg += ph_incr_reg + s_fm_val;
            while (phase_reg > table_len_reg) phase_reg -= table_len_reg;
            /* watch out for negative frequencies! */
            while (phase_reg < 0) phase_reg += table_len_reg;
	} while (--n); /* inner loop */

	susp->phase = phase_reg;
	susp->ampramp_a = ampramp_a_reg;
	susp->ampramp_b = ampramp_b_reg;
	out_ptr += togo;
	susp->s_fm_pHaSe += togo * susp->s_fm_pHaSe_iNcR;
	susp->s_fm_n -= togo;
	cnt += togo;
    } /* outer loop */

    /* test for termination */
    if (togo == 0 && cnt == 0) {
	snd_list_terminate(snd_list);
    } else {
	snd_list->block_len = cnt;
	susp->susp.current += cnt;
    }
    /* test for logical stop */
    if (susp->logically_stopped) {
	snd_list->logically_stopped = true;
    } else if (susp->susp.log_stop_cnt == susp->susp.current) {
	susp->logically_stopped = true;
    }
} /* siosc_r_fetch */


void siosc_toss_fetch(susp, snd_list)
  register siosc_susp_type susp;
  snd_list_type snd_list;
{
    long final_count = susp->susp.toss_cnt;
    time_type final_time = susp->susp.t0;
    long n;

    /* fetch samples from s_fm up to final_time for this block of zeros */
    while ((round((final_time - susp->s_fm->t0) * susp->s_fm->sr)) >=
	   susp->s_fm->current)
	susp_get_samples(s_fm, s_fm_ptr, s_fm_cnt);
    /* convert to normal processing when we hit final_count */
    /* we want each signal positioned at final_time */
    n = round((final_time - susp->s_fm->t0) * susp->s_fm->sr -
         (susp->s_fm->current - susp->s_fm_cnt));
    susp->s_fm_ptr += n;
    susp_took(s_fm_cnt, n);
    susp->susp.fetch = susp->susp.keep_fetch;
    (*(susp->susp.fetch))(susp, snd_list);
}


void siosc_mark(siosc_susp_type susp)
{
    if (susp->lis) mark(susp->lis);
    sound_xlmark(susp->s_fm);
}


void siosc_free(siosc_susp_type susp)
{
    table_unref(susp->table_a_ptr);
    table_unref(susp->table_b_ptr_ptr);
    sound_unref(susp->s_fm);
    ffree_generic(susp, sizeof(siosc_susp_node), "siosc_free");
}


void siosc_print_tree(siosc_susp_type susp, int n)
{
    indent(n);
    stdputstr("s_fm:");
    sound_print_tree_1(susp->s_fm, n);
}


sound_type snd_make_siosc(LVAL lis, rate_type sr, double hz, time_type t0, sound_type s_fm)
{
    register siosc_susp_type susp;
    /* sr specified as input parameter */
    /* t0 specified as input parameter */
    int interp_desc = 0;
    sample_type scale_factor = 1.0F;
    time_type t0_min = t0;
    falloc_generic(susp, siosc_susp_node, "snd_make_siosc");
    susp->table_len = 0.0;
    susp->ph_incr = 0.0;
    susp->table_a_ptr = NULL;
    susp->table_b_ptr_ptr = NULL;
    susp->table_a_samps = NULL;
    susp->table_b_samps = NULL;
    susp->table_sr = 0.0;
    susp->phase = 0.0;
    susp->lis = lis;
    susp->next_breakpoint = 0;
    susp->ampramp_a = 1.0;
    susp->ampramp_b = 0.0;
    susp->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));

    /* select a susp fn based on sample rates */
    interp_desc = (interp_desc << 2) + interp_style(s_fm, sr);
    switch (interp_desc) {
      case INTERP_n: /* handled below */
      case INTERP_s: susp->susp.fetch = siosc_s_fetch; break;
      case INTERP_i: susp->susp.fetch = siosc_i_fetch; break;
      case INTERP_r: susp->susp.fetch = siosc_r_fetch; break;
      default: snd_badsr(); break;
    }

    susp->terminate_cnt = UNKNOWN;
    /* handle unequal start times, if any */
    if (t0 < s_fm->t0) sound_prepend_zeros(s_fm, t0);
    /* minimum start time over all inputs: */
    t0_min = MIN(s_fm->t0, t0);
    /* how many samples to toss before t0: */
    susp->susp.toss_cnt = (long) ((t0 - t0_min) * sr + 0.5);
    if (susp->susp.toss_cnt > 0) {
	susp->susp.keep_fetch = susp->susp.fetch;
	susp->susp.fetch = siosc_toss_fetch;
    }

    /* initialize susp state */
    susp->susp.free = siosc_free;
    susp->susp.sr = sr;
    susp->susp.t0 = t0;
    susp->susp.mark = siosc_mark;
    susp->susp.print_tree = siosc_print_tree;
    susp->susp.name = "siosc";
    susp->logically_stopped = false;
    susp->susp.log_stop_cnt = logical_stop_cnt_cvt(s_fm);
    susp->started = false;
    susp->susp.current = 0;
    susp->s_fm = s_fm;
    susp->s_fm_cnt = 0;
    susp->s_fm_pHaSe = 0.0;
    susp->s_fm_pHaSe_iNcR = s_fm->sr / sr;
    susp->s_fm_n = 0;
    susp->output_per_s_fm = sr / s_fm->sr;
    return sound_create((snd_susp_type)susp, t0, sr, scale_factor);
}


sound_type snd_siosc(LVAL lis, rate_type sr, double hz, time_type t0, sound_type s_fm)
{
    sound_type s_fm_copy = sound_copy(s_fm);
    return snd_make_siosc(lis, sr, hz, t0, s_fm_copy);
}