Import nyquist_3.05.orig.tar.gz

[dgit import orig nyquist_3.05.orig.tar.gz]

1 files changed, 374 insertions, 0 deletions

diff --git a/tran/alpassvc.c b/tran/alpassvc.c
new file mode 100644
index 0000000..e5bdf4b
--- /dev/null
+++ b/tran/alpassvc.c
@@ -0,0 +1,374 @@
+#include "stdio.h"
+#ifndef mips
+#include "stdlib.h"
+#endif
+#include "xlisp.h"
+#include "sound.h"
+
+#include "falloc.h"
+#include "cext.h"
+#include "alpassvc.h"
+
+void alpassvc_free();
+
+
+typedef struct alpassvc_susp_struct {
+    snd_susp_node susp;
+    long terminate_cnt;
+    sound_type input;
+    long input_cnt;
+    sample_block_values_type input_ptr;
+    sound_type delaysnd;
+    long delaysnd_cnt;
+    sample_block_values_type delaysnd_ptr;
+
+    float delay_scale_factor;
+    double feedback;
+    long buflen;
+    sample_type *delaybuf;
+    sample_type *delayptr;
+    sample_type *endptr;
+} alpassvc_susp_node, *alpassvc_susp_type;
+
+
+void alpassvc_nn_fetch(register alpassvc_susp_type susp, snd_list_type snd_list)
+{
+    int cnt = 0; /* how many samples computed */
+    int togo;
+    int n;
+    sample_block_type out;
+    register sample_block_values_type out_ptr;
+
+    register sample_block_values_type out_ptr_reg;
+
+    register float delay_scale_factor_reg;
+    register double feedback_reg;
+    register long buflen_reg;
+    register sample_type * delayptr_reg;
+    register sample_type * endptr_reg;
+    register sample_block_values_type delaysnd_ptr_reg;
+    register sample_block_values_type input_ptr_reg;
+    falloc_sample_block(out, "alpassvc_nn_fetch");
+    out_ptr = out->samples;
+    snd_list->block = out;
+
+    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;
+
+	/* don't run past the input input sample block: */
+	susp_check_term_samples(input, input_ptr, input_cnt);
+	togo = min(togo, susp->input_cnt);
+
+	/* don't run past the delaysnd input sample block: */
+	susp_check_samples(delaysnd, delaysnd_ptr, delaysnd_cnt);
+	togo = min(togo, susp->delaysnd_cnt);
+
+	/* 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;
+	}
+
+	n = togo;
+	delay_scale_factor_reg = susp->delay_scale_factor;
+	feedback_reg = susp->feedback;
+	buflen_reg = susp->buflen;
+	delayptr_reg = susp->delayptr;
+	endptr_reg = susp->endptr;
+	delaysnd_ptr_reg = susp->delaysnd_ptr;
+	input_ptr_reg = susp->input_ptr;
+	out_ptr_reg = out_ptr;
+	if (n) do { /* the inner sample computation loop */
+        register sample_type y, z, delaysamp;
+        register int delayi;
+        register sample_type *yptr;
+
+        /* compute where to read y, we want y to be delay_snd samples
+         * after delay_ptr, where we write the new sample. First, 
+         * conver from seconds to samples. Note: don't use actual sound_type
+         * names in comments! The translator isn't smart enough.
+         */
+        delaysamp = *delaysnd_ptr_reg++ * delay_scale_factor_reg;
+        delayi = (int) delaysamp; /* get integer part */
+        delaysamp = delaysamp - delayi; /* get phase */
+        yptr = delayptr_reg + buflen_reg - (delayi + 1);
+        if (yptr >= endptr_reg) yptr -= buflen_reg;
+        /* now get y, the out-put of the delay, using interpolation */
+        /* note that as phase increases, we use more of yptr[0] because
+           positive phase means longer buffer means read earlier sample */
+        y = (float) ((yptr[0] * delaysamp) + (yptr[1] * (1.0 - delaysamp)));
+        /* WARNING: no check to keep delaysamp in range, so do this in LISP */
+
+        *delayptr_reg++ = z = (sample_type) (feedback_reg * y + *input_ptr_reg++);
+        /* Time out to update the buffer:
+         * this is a tricky buffer: buffer[0] == buffer[bufflen]
+         * the logical length is bufflen, but the actual length
+         * is bufflen + 1 to allow for a repeated sample at the
+         * end. This allows for efficient interpolation.
+         */
+        if (delayptr_reg > endptr_reg) {
+            delayptr_reg = susp->delaybuf;
+            *delayptr_reg++ = *endptr_reg;
+        }
+        *out_ptr_reg++ = (sample_type) (y - feedback_reg * z);;
+	} while (--n); /* inner loop */
+
+	susp->buflen = buflen_reg;
+	susp->delayptr = delayptr_reg;
+	/* using delaysnd_ptr_reg is a bad idea on RS/6000: */
+	susp->delaysnd_ptr += togo;
+	/* using input_ptr_reg is a bad idea on RS/6000: */
+	susp->input_ptr += togo;
+	out_ptr += togo;
+	susp_took(input_cnt, togo);
+	susp_took(delaysnd_cnt, 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;
+    }
+} /* alpassvc_nn_fetch */
+
+
+void alpassvc_ns_fetch(register alpassvc_susp_type susp, snd_list_type snd_list)
+{
+    int cnt = 0; /* how many samples computed */
+    int togo;
+    int n;
+    sample_block_type out;
+    register sample_block_values_type out_ptr;
+
+    register sample_block_values_type out_ptr_reg;
+
+    register float delay_scale_factor_reg;
+    register double feedback_reg;
+    register long buflen_reg;
+    register sample_type * delayptr_reg;
+    register sample_type * endptr_reg;
+    register sample_type delaysnd_scale_reg = susp->delaysnd->scale;
+    register sample_block_values_type delaysnd_ptr_reg;
+    register sample_block_values_type input_ptr_reg;
+    falloc_sample_block(out, "alpassvc_ns_fetch");
+    out_ptr = out->samples;
+    snd_list->block = out;
+
+    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;
+
+	/* don't run past the input input sample block: */
+	susp_check_term_samples(input, input_ptr, input_cnt);
+	togo = min(togo, susp->input_cnt);
+
+	/* don't run past the delaysnd input sample block: */
+	susp_check_samples(delaysnd, delaysnd_ptr, delaysnd_cnt);
+	togo = min(togo, susp->delaysnd_cnt);
+
+	/* 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;
+	}
+
+	n = togo;
+	delay_scale_factor_reg = susp->delay_scale_factor;
+	feedback_reg = susp->feedback;
+	buflen_reg = susp->buflen;
+	delayptr_reg = susp->delayptr;
+	endptr_reg = susp->endptr;
+	delaysnd_ptr_reg = susp->delaysnd_ptr;
+	input_ptr_reg = susp->input_ptr;
+	out_ptr_reg = out_ptr;
+	if (n) do { /* the inner sample computation loop */
+        register sample_type y, z, delaysamp;
+        register int delayi;
+        register sample_type *yptr;
+
+        /* compute where to read y, we want y to be delay_snd samples
+         * after delay_ptr, where we write the new sample. First, 
+         * conver from seconds to samples. Note: don't use actual sound_type
+         * names in comments! The translator isn't smart enough.
+         */
+        delaysamp = (delaysnd_scale_reg * *delaysnd_ptr_reg++) * delay_scale_factor_reg;
+        delayi = (int) delaysamp; /* get integer part */
+        delaysamp = delaysamp - delayi; /* get phase */
+        yptr = delayptr_reg + buflen_reg - (delayi + 1);
+        if (yptr >= endptr_reg) yptr -= buflen_reg;
+        /* now get y, the out-put of the delay, using interpolation */
+        /* note that as phase increases, we use more of yptr[0] because
+           positive phase means longer buffer means read earlier sample */
+        y = (float) ((yptr[0] * delaysamp) + (yptr[1] * (1.0 - delaysamp)));
+        /* WARNING: no check to keep delaysamp in range, so do this in LISP */
+
+        *delayptr_reg++ = z = (sample_type) (feedback_reg * y + *input_ptr_reg++);
+        /* Time out to update the buffer:
+         * this is a tricky buffer: buffer[0] == buffer[bufflen]
+         * the logical length is bufflen, but the actual length
+         * is bufflen + 1 to allow for a repeated sample at the
+         * end. This allows for efficient interpolation.
+         */
+        if (delayptr_reg > endptr_reg) {
+            delayptr_reg = susp->delaybuf;
+            *delayptr_reg++ = *endptr_reg;
+        }
+        *out_ptr_reg++ = (sample_type) (y - feedback_reg * z);;
+	} while (--n); /* inner loop */
+
+	susp->buflen = buflen_reg;
+	susp->delayptr = delayptr_reg;
+	/* using delaysnd_ptr_reg is a bad idea on RS/6000: */
+	susp->delaysnd_ptr += togo;
+	/* using input_ptr_reg is a bad idea on RS/6000: */
+	susp->input_ptr += togo;
+	out_ptr += togo;
+	susp_took(input_cnt, togo);
+	susp_took(delaysnd_cnt, 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;
+    }
+} /* alpassvc_ns_fetch */
+
+
+void alpassvc_toss_fetch(susp, snd_list)
+  register alpassvc_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 input up to final_time for this block of zeros */
+    while ((round((final_time - susp->input->t0) * susp->input->sr)) >=
+	   susp->input->current)
+	susp_get_samples(input, input_ptr, input_cnt);
+    /* fetch samples from delaysnd up to final_time for this block of zeros */
+    while ((round((final_time - susp->delaysnd->t0) * susp->delaysnd->sr)) >=
+	   susp->delaysnd->current)
+	susp_get_samples(delaysnd, delaysnd_ptr, delaysnd_cnt);
+    /* convert to normal processing when we hit final_count */
+    /* we want each signal positioned at final_time */
+    n = round((final_time - susp->input->t0) * susp->input->sr -
+         (susp->input->current - susp->input_cnt));
+    susp->input_ptr += n;
+    susp_took(input_cnt, n);
+    n = round((final_time - susp->delaysnd->t0) * susp->delaysnd->sr -
+         (susp->delaysnd->current - susp->delaysnd_cnt));
+    susp->delaysnd_ptr += n;
+    susp_took(delaysnd_cnt, n);
+    susp->susp.fetch = susp->susp.keep_fetch;
+    (*(susp->susp.fetch))(susp, snd_list);
+}
+
+
+void alpassvc_mark(alpassvc_susp_type susp)
+{
+    sound_xlmark(susp->input);
+    sound_xlmark(susp->delaysnd);
+}
+
+
+void alpassvc_free(alpassvc_susp_type susp)
+{
+free(susp->delaybuf);    sound_unref(susp->input);
+    sound_unref(susp->delaysnd);
+    ffree_generic(susp, sizeof(alpassvc_susp_node), "alpassvc_free");
+}
+
+
+void alpassvc_print_tree(alpassvc_susp_type susp, int n)
+{
+    indent(n);
+    stdputstr("input:");
+    sound_print_tree_1(susp->input, n);
+
+    indent(n);
+    stdputstr("delaysnd:");
+    sound_print_tree_1(susp->delaysnd, n);
+}
+
+
+sound_type snd_make_alpassvc(sound_type input, sound_type delaysnd, double feedback, double maxdelay)
+{
+    register alpassvc_susp_type susp;
+    rate_type sr = max(input->sr, delaysnd->sr);
+    time_type t0 = max(input->t0, delaysnd->t0);
+    int interp_desc = 0;
+    sample_type scale_factor = 1.0F;
+    time_type t0_min = t0;
+    /* combine scale factors of linear inputs (INPUT) */
+    scale_factor *= input->scale;
+    input->scale = 1.0F;
+
+    /* try to push scale_factor back to a low sr input */
+    if (input->sr < sr) { input->scale = scale_factor; scale_factor = 1.0F; }
+
+    falloc_generic(susp, alpassvc_susp_node, "snd_make_alpassvc");
+    susp->delay_scale_factor = (float) (input->sr * delaysnd->scale);
+    susp->feedback = feedback;
+    susp->buflen = max(2, (long) (input->sr * maxdelay + 2.5));
+    susp->delaybuf = (sample_type *) calloc (susp->buflen + 1, sizeof(sample_type));
+    susp->delayptr = susp->delaybuf;
+    susp->endptr = susp->delaybuf + susp->buflen;
+
+    /* select a susp fn based on sample rates */
+    interp_desc = (interp_desc << 2) + interp_style(input, sr);
+    interp_desc = (interp_desc << 2) + interp_style(delaysnd, sr);
+    switch (interp_desc) {
+      case INTERP_nn: susp->susp.fetch = alpassvc_nn_fetch; break;
+      case INTERP_ns: susp->susp.fetch = alpassvc_ns_fetch; break;
+      default: snd_badsr(); break;
+    }
+
+    susp->terminate_cnt = UNKNOWN;
+    /* handle unequal start times, if any */
+    if (t0 < input->t0) sound_prepend_zeros(input, t0);
+    if (t0 < delaysnd->t0) sound_prepend_zeros(delaysnd, t0);
+    /* minimum start time over all inputs: */
+    t0_min = min(input->t0, min(delaysnd->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 = alpassvc_toss_fetch;
+    }
+
+    /* initialize susp state */
+    susp->susp.free = alpassvc_free;
+    susp->susp.sr = sr;
+    susp->susp.t0 = t0;
+    susp->susp.mark = alpassvc_mark;
+    susp->susp.print_tree = alpassvc_print_tree;
+    susp->susp.name = "alpassvc";
+    susp->susp.log_stop_cnt = UNKNOWN;
+    susp->susp.current = 0;
+    susp->input = input;
+    susp->input_cnt = 0;
+    susp->delaysnd = delaysnd;
+    susp->delaysnd_cnt = 0;
+    return sound_create((snd_susp_type)susp, t0, sr, scale_factor);
+}
+
+
+sound_type snd_alpassvc(sound_type input, sound_type delaysnd, double feedback, double maxdelay)
+{
+    sound_type input_copy = sound_copy(input);
+    sound_type delaysnd_copy = sound_copy(delaysnd);
+    return snd_make_alpassvc(input_copy, delaysnd_copy, feedback, maxdelay);
+}