mirror of
https://github.com/cookiengineer/audacity
synced 2025-10-25 15:53:52 +02:00
Update nyquist to SVN r331 (r3.16+)
------------------------------------------------------------------------ r331 | rbd | 2020-10-13 12:40:12 -0500 (Tue, 13 Oct 2020) | 2 lines Also forgot to install NyquistWords.txt ------------------------------------------------------------------------ r330 | rbd | 2020-10-13 12:34:06 -0500 (Tue, 13 Oct 2020) | 2 lines Forgot to move nyquistman.pdf from docsrc/s2h to release ------------------------------------------------------------------------ r329 | rbd | 2020-10-13 11:32:33 -0500 (Tue, 13 Oct 2020) | 2 lines Updated some version numbers for 3.16. ------------------------------------------------------------------------ r328 | rbd | 2020-10-13 11:20:52 -0500 (Tue, 13 Oct 2020) | 2 lines Fixed NyquistIDE antialiasing for plot text, fix format of message. ------------------------------------------------------------------------ r327 | rbd | 2020-10-12 21:01:53 -0500 (Mon, 12 Oct 2020) | 2 lines Fixed a couple of format problems in manual. This version of Nyquist has been tested wtih macOS, Linux, 32&64-bit Windows. ------------------------------------------------------------------------ r326 | rbd | 2020-10-12 20:21:38 -0500 (Mon, 12 Oct 2020) | 1 line Modified WIN32 32-bit XLisp to use 64-bit FIXNUMs. This allows XLisp and Nyquist to handle big sounds even on 32-bit machines. Probably at some cost, but inner loops are mostly float and int32, and the Nyquist release is 64-bit anyway. Maybe we'll have to run some benchmarks on Audacity, which is still 32-bit on Windows. ------------------------------------------------------------------------ r325 | rbd | 2020-10-12 13:16:57 -0500 (Mon, 12 Oct 2020) | 1 line Win64 passes bigfiletest.lsp now. This version should work on all 64-bit systems now. These changes untested on Linux and macOS. ------------------------------------------------------------------------ r324 | rbd | 2020-10-11 21:31:53 -0500 (Sun, 11 Oct 2020) | 2 lines I couldn't free enough space on my linux box, so I adjusted the bigfiletest to write 8-bit ulaw. It's still >4GB and >4G samples. Works on Linux. ------------------------------------------------------------------------ r323 | rbd | 2020-10-11 19:41:25 -0500 (Sun, 11 Oct 2020) | 2 lines Missing file from last commit. ------------------------------------------------------------------------ r322 | rbd | 2020-10-11 19:36:08 -0500 (Sun, 11 Oct 2020) | 1 line Found another case where WIN64 needs int64_t instead of long for sample count. ------------------------------------------------------------------------ r321 | rbd | 2020-10-11 19:33:25 -0500 (Sun, 11 Oct 2020) | 3 lines Fixed s-save to handle optional and keyword parameters (which should never have been mixed in the first place). Documentation cleanup - should be final for this version. ------------------------------------------------------------------------ r320 | rbd | 2020-10-11 14:44:37 -0500 (Sun, 11 Oct 2020) | 2 lines Fixes to handle IRCAM sound format and tests for big file io working on macOS. ------------------------------------------------------------------------ r319 | rbd | 2020-10-10 21:31:58 -0500 (Sat, 10 Oct 2020) | 2 lines Changes for linux and to avoid compiler warnings on linux. ------------------------------------------------------------------------ r318 | rbd | 2020-10-10 20:50:23 -0500 (Sat, 10 Oct 2020) | 1 line This is the test used for Win64 version. ------------------------------------------------------------------------ r317 | rbd | 2020-10-10 20:34:34 -0500 (Sat, 10 Oct 2020) | 1 line This version works on Win64. Need to test changes on macOS and linux. ------------------------------------------------------------------------ r316 | rbd | 2020-10-10 19:59:15 -0500 (Sat, 10 Oct 2020) | 2 lines PWL changes to avoid compiler warning. ------------------------------------------------------------------------ r315 | rbd | 2020-10-10 19:34:04 -0500 (Sat, 10 Oct 2020) | 2 lines A few more changes for 64-bit sample counts on Win64 ------------------------------------------------------------------------ r314 | rbd | 2020-10-10 13:19:42 -0500 (Sat, 10 Oct 2020) | 2 lines Fixed int64_t declaration in gate.alg ------------------------------------------------------------------------ r313 | rbd | 2020-10-10 12:07:40 -0500 (Sat, 10 Oct 2020) | 2 lines Fixes to gate for long sounds ------------------------------------------------------------------------ r312 | rbd | 2020-10-10 11:47:29 -0500 (Sat, 10 Oct 2020) | 2 lines Fixed sound_save types for intgen ------------------------------------------------------------------------ r311 | rbd | 2020-10-10 11:09:01 -0500 (Sat, 10 Oct 2020) | 2 lines Fixed a 64-bit sample count problem in siosc.alg ------------------------------------------------------------------------ r310 | rbd | 2020-10-10 11:03:12 -0500 (Sat, 10 Oct 2020) | 2 lines Fixed sndmax to handle 64-bit sample counts. ------------------------------------------------------------------------ r309 | rbd | 2020-10-10 10:57:04 -0500 (Sat, 10 Oct 2020) | 2 lines Forgot to re-translate all tran/*.alg files with fix for int64 cast to int32. This version compiles on macOS and ready for test on Win64. ------------------------------------------------------------------------ r308 | rbd | 2020-10-10 10:16:05 -0500 (Sat, 10 Oct 2020) | 2 lines Everything seems to compile and run on macOS now. Moving changes to Windows for test. ------------------------------------------------------------------------ r307 | rbd | 2020-10-10 09:23:45 -0500 (Sat, 10 Oct 2020) | 1 line Added casts to avoid compiler warnings and to review changes to support 64-bit sample counts on Windows. Still not complete, and waiting to regenerate and compile tran directory code after updates to translation code that will insert more casts. ------------------------------------------------------------------------ r306 | rbd | 2020-10-09 21:55:15 -0500 (Fri, 09 Oct 2020) | 2 lines Rebuilt seqfnint.c from header files. ------------------------------------------------------------------------ r305 | rbd | 2020-10-09 21:53:33 -0500 (Fri, 09 Oct 2020) | 1 line Changed some FIXNUMS to LONG to avoid compiler warnings in seqfnint.c ------------------------------------------------------------------------ r304 | rbd | 2020-10-09 21:44:03 -0500 (Fri, 09 Oct 2020) | 2 lines I discovered forgotten regression-test.lsp and added test that requires 64-bit sample counts to pass. Fixed a few bugs revealed by running the type-checking regression tests. ------------------------------------------------------------------------ r303 | rbd | 2020-10-09 12:28:58 -0500 (Fri, 09 Oct 2020) | 2 lines Changes for 64-bit sample counts broke mult-channel s-save. Fixed in the commit for macOS. ------------------------------------------------------------------------ r302 | rbd | 2020-10-09 10:03:39 -0500 (Fri, 09 Oct 2020) | 2 lines Changed snd-play to return samples computed and used that to make a test for computing long sounds that would overflow 32-bit length counts. ------------------------------------------------------------------------ r301 | rbd | 2020-10-09 09:11:26 -0500 (Fri, 09 Oct 2020) | 2 lines corrected mistake in delaycv.alg and re-translated ------------------------------------------------------------------------ r300 | rbd | 2020-10-09 09:09:06 -0500 (Fri, 09 Oct 2020) | 2 lines Fix to delaycv.alg -- "s" changed to "input" to avoid matching "s" in "sample_type". ------------------------------------------------------------------------ r299 | rbd | 2020-10-09 09:03:33 -0500 (Fri, 09 Oct 2020) | 4 lines To avoid compiler warnings, XLisp interfaces to C int and long are now specified as LONG rather than FIXNUM, and the stubs that call the C functions cast FIXNUMs from XLisp into longs before calling C functions. ------------------------------------------------------------------------ r298 | rbd | 2020-10-08 22:20:26 -0500 (Thu, 08 Oct 2020) | 2 lines This commit has many more fixes to handle long (64-bit) sounds, including a lot of fixes for warnings by Visual Studio assigning int64_t to long (works on macOS, doesn't work on VS). This was compiled and tested on macOS, and even computed a 27.1-hour sound using OSC, LP, SUM and MULT (haven't tested I/O yet). ------------------------------------------------------------------------ r297 | rbd | 2020-10-07 13:04:02 -0500 (Wed, 07 Oct 2020) | 2 lines This is a major cleanup. It started with the goal of changing long to int64_t for sample counts so that on 64-bit windows, where long is only 32-bits, the sample counts would nevertheless be 64-bit allowing long sounds, which was a limitation for long recordings in Audacity. Since I was using compiler warnings to track possible loss-of-precision conversions from 64-bit sample counts, and there were *many* warnings, I started cleaning up *all* the warnings and ended up with a very large set of changes, including "modernizing" C declarations that date back to XLisp and CMU MIDI Toolkit code and were never changed. This version runs all the examples.sal code on macOS, but will surely have problems on Windows and Linux given the number of changes. ------------------------------------------------------------------------ r296 | rbd | 2020-10-06 13:34:20 -0500 (Tue, 06 Oct 2020) | 2 lines More changes from long to int64_t for sample counts. ------------------------------------------------------------------------ r295 | rbd | 2020-10-06 11:53:49 -0500 (Tue, 06 Oct 2020) | 2 lines More work on using 64-bit sample counts. Changed MAX_STOP from 32-bit to 64-bit limit. ------------------------------------------------------------------------ r294 | rbd | 2020-10-06 11:48:05 -0500 (Tue, 06 Oct 2020) | 2 lines Made some changes so that sample counts are int64_t (for windows) instead of long to support sample counts above 31 bits. ------------------------------------------------------------------------ r293 | rbd | 2020-10-04 21:30:55 -0500 (Sun, 04 Oct 2020) | 2 lines Fixed a few minor things for Linux and tested on Linux. ------------------------------------------------------------------------ r292 | rbd | 2020-10-04 21:00:28 -0500 (Sun, 04 Oct 2020) | 2 lines Update extensions: all are minor changes. ------------------------------------------------------------------------ r291 | rbd | 2020-09-24 13:59:31 -0500 (Thu, 24 Sep 2020) | 2 lines New implementation of seq and seqrep, added get-real-time, documented get-real-time, fixed examples.sal and examples.lsp which are now in lib rather than extensions (so they are now back in the basic installation), other cleanup. ------------------------------------------------------------------------ r290 | rbd | 2020-08-16 16:24:52 -0500 (Sun, 16 Aug 2020) | 2 lines Fixed bug in snd-gate, revised GATE and NOISE-GATE to handle multi-channel sound. RMS now handles multi-channel input. S-AVG added to take multichannel input (but not used, because RMS could not be written without making SND-SRATE convert multichannel sound to vector of floats. That seems to be going toward a fully vectorized model. Not going there for now. ------------------------------------------------------------------------ r289 | rbd | 2020-07-09 16:27:45 -0500 (Thu, 09 Jul 2020) | 2 lines Added GET-REAL-TIME function to XLISP. May not work yet on Windows. Various fixes for compiler warnings. I noticed FLAC doesn't work (I guess it never did) and I cannot figure out how this even links because flac_min seems to be undefined. Something to look at later.
This commit is contained in:
Leland Lucius
@@ -15,10 +15,10 @@ void siosc_free(snd_susp_type a_susp);
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typedef struct siosc_susp_struct {
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snd_susp_node susp;
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boolean started;
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long terminate_cnt;
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int64_t terminate_cnt;
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boolean logically_stopped;
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sound_type s_fm;
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long s_fm_cnt;
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int s_fm_cnt;
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sample_block_values_type s_fm_ptr;
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/* support for interpolation of s_fm */
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@@ -28,7 +28,7 @@ typedef struct siosc_susp_struct {
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/* support for ramp between samples of s_fm */
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double output_per_s_fm;
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long s_fm_n;
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int64_t s_fm_n;
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double table_len;
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double ph_incr;
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@@ -39,7 +39,7 @@ typedef struct siosc_susp_struct {
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double table_sr;
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double phase;
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LVAL lis;
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long next_breakpoint;
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int64_t next_breakpoint;
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double ampramp_a;
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double ampramp_b;
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double ampslope;
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@@ -65,9 +65,9 @@ void siosc_table_init(siosc_susp_type susp)
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/* siosc_table_update -- outer loop processing, get next table */
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/**/
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long siosc_table_update(siosc_susp_type susp, long cur)
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int64_t siosc_table_update(siosc_susp_type susp, int64_t cur)
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{
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long n;
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int64_t n;
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/* swap ampramps: */
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susp->ampramp_a = 1.0;
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@@ -101,8 +101,7 @@ long siosc_table_update(siosc_susp_type susp, long cur)
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susp->lis = cdr(susp->lis);
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} else { /* use only table a */
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susp->ampslope = 0.0;
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susp->next_breakpoint = 0x7FFFFFFF;
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n = 0x7FFFFFFF;
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susp->next_breakpoint = n = 0x7FFFFFFFFFFFFFFF;
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}
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return n;
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}
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@@ -134,67 +133,67 @@ void siosc_s_fetch(snd_susp_type a_susp, snd_list_type snd_list)
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snd_list->block = out;
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while (cnt < max_sample_block_len) { /* outer loop */
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/* first compute how many samples to generate in inner loop: */
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/* don't overflow the output sample block: */
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togo = max_sample_block_len - cnt;
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/* first compute how many samples to generate in inner loop: */
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/* don't overflow the output sample block: */
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togo = max_sample_block_len - cnt;
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/* don't run past the s_fm input sample block: */
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susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
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togo = min(togo, susp->s_fm_cnt);
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/* don't run past the s_fm input sample block: */
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susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
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togo = min(togo, susp->s_fm_cnt);
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/* don't run past terminate time */
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if (susp->terminate_cnt != UNKNOWN &&
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susp->terminate_cnt <= susp->susp.current + cnt + togo) {
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togo = susp->terminate_cnt - (susp->susp.current + cnt);
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if (togo < 0) togo = 0; /* avoids rounding errros */
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if (togo == 0) break;
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}
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/* don't run past logical stop time */
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if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
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int to_stop = susp->susp.log_stop_cnt - (susp->susp.current + cnt);
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/* break if to_stop == 0 (we're at the logical stop)
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* AND cnt > 0 (we're not at the beginning of the
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* output block).
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*/
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if (to_stop < 0) to_stop = 0; /* avoids rounding errors */
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if (to_stop < togo) {
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if (to_stop == 0) {
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if (cnt) {
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togo = 0;
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break;
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} else /* keep togo as is: since cnt == 0, we
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* can set the logical stop flag on this
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* output block
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*/
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susp->logically_stopped = true;
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} else /* limit togo so we can start a new
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* block at the LST
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*/
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togo = to_stop;
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}
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}
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{ long cur = susp->susp.current + cnt;
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n = susp->next_breakpoint - cur;
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if (n == 0) n = siosc_table_update(susp, cur);
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/* don't run past terminate time */
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if (susp->terminate_cnt != UNKNOWN &&
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susp->terminate_cnt <= susp->susp.current + cnt + togo) {
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togo = (int) (susp->terminate_cnt - (susp->susp.current + cnt));
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if (togo < 0) togo = 0; /* avoids rounding errros */
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if (togo == 0) break;
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}
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togo = min(n, togo);
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n = togo;
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table_len_reg = susp->table_len;
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ph_incr_reg = susp->ph_incr;
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table_a_samps_reg = susp->table_a_samps;
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table_b_samps_reg = susp->table_b_samps;
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phase_reg = susp->phase;
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ampramp_a_reg = susp->ampramp_a;
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ampramp_b_reg = susp->ampramp_b;
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ampslope_reg = susp->ampslope;
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s_fm_ptr_reg = susp->s_fm_ptr;
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out_ptr_reg = out_ptr;
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if (n) do { /* the inner sample computation loop */
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/* don't run past logical stop time */
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if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
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int64_t to_stop = susp->susp.log_stop_cnt - (susp->susp.current + cnt);
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/* break if to_stop == 0 (we're at the logical stop)
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* AND cnt > 0 (we're not at the beginning of the
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* output block).
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*/
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if (to_stop < 0) to_stop = 0; /* avoids rounding errors */
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if (to_stop < togo) {
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if (to_stop == 0) {
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if (cnt) {
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togo = 0;
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break;
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} else /* keep togo as is: since cnt == 0, we
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* can set the logical stop flag on this
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* output block
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*/
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susp->logically_stopped = true;
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} else /* limit togo so we can start a new
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* block at the LST
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*/
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togo = (int) to_stop;
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}
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}
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{ int64_t cur = susp->susp.current + cnt;
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int64_t n64 = susp->next_breakpoint - cur;
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if (n64 == 0) n64 = siosc_table_update(susp, cur);
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if (n64 < togo) togo = (int) n64;
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}
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n = togo;
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table_len_reg = susp->table_len;
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ph_incr_reg = susp->ph_incr;
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table_a_samps_reg = susp->table_a_samps;
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table_b_samps_reg = susp->table_b_samps;
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phase_reg = susp->phase;
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ampramp_a_reg = susp->ampramp_a;
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ampramp_b_reg = susp->ampramp_b;
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ampslope_reg = susp->ampslope;
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s_fm_ptr_reg = susp->s_fm_ptr;
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out_ptr_reg = out_ptr;
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if (n) do { /* the inner sample computation loop */
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long table_index;
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double xa, xb;
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table_index = (long) phase_reg;
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@@ -211,30 +210,30 @@ void siosc_s_fetch(snd_susp_type a_susp, snd_list_type snd_list)
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while (phase_reg > table_len_reg) phase_reg -= table_len_reg;
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/* watch out for negative frequencies! */
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while (phase_reg < 0) phase_reg += table_len_reg;
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} while (--n); /* inner loop */
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} while (--n); /* inner loop */
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susp->phase = phase_reg;
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susp->ampramp_a = ampramp_a_reg;
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susp->ampramp_b = ampramp_b_reg;
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/* using s_fm_ptr_reg is a bad idea on RS/6000: */
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susp->s_fm_ptr += togo;
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out_ptr += togo;
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susp_took(s_fm_cnt, togo);
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cnt += togo;
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susp->phase = phase_reg;
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susp->ampramp_a = ampramp_a_reg;
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susp->ampramp_b = ampramp_b_reg;
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/* using s_fm_ptr_reg is a bad idea on RS/6000: */
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susp->s_fm_ptr += togo;
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out_ptr += togo;
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susp_took(s_fm_cnt, togo);
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cnt += togo;
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} /* outer loop */
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/* test for termination */
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if (togo == 0 && cnt == 0) {
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snd_list_terminate(snd_list);
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snd_list_terminate(snd_list);
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} else {
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snd_list->block_len = cnt;
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susp->susp.current += cnt;
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snd_list->block_len = cnt;
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susp->susp.current += cnt;
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}
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/* test for logical stop */
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if (susp->logically_stopped) {
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snd_list->logically_stopped = true;
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snd_list->logically_stopped = true;
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} else if (susp->susp.log_stop_cnt == susp->susp.current) {
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susp->logically_stopped = true;
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susp->logically_stopped = true;
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}
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} /* siosc_s_fetch */
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@@ -267,81 +266,81 @@ void siosc_i_fetch(snd_susp_type a_susp, snd_list_type snd_list)
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/* make sure sounds are primed with first values */
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if (!susp->started) {
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susp->started = true;
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susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
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susp->s_fm_x1_sample = susp_fetch_sample(s_fm, s_fm_ptr, s_fm_cnt);
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susp->started = true;
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susp_check_term_log_samples(s_fm, s_fm_ptr, s_fm_cnt);
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susp->s_fm_x1_sample = susp_fetch_sample(s_fm, s_fm_ptr, s_fm_cnt);
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}
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while (cnt < max_sample_block_len) { /* outer loop */
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/* first compute how many samples to generate in inner loop: */
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/* don't overflow the output sample block: */
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togo = max_sample_block_len - cnt;
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/* first compute how many samples to generate in inner loop: */
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/* don't overflow the output sample block: */
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togo = max_sample_block_len - cnt;
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/* don't run past terminate time */
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if (susp->terminate_cnt != UNKNOWN &&
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susp->terminate_cnt <= susp->susp.current + cnt + togo) {
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togo = susp->terminate_cnt - (susp->susp.current + cnt);
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if (togo < 0) togo = 0; /* avoids rounding errros */
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if (togo == 0) break;
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}
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/* don't run past logical stop time */
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if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
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int to_stop = susp->susp.log_stop_cnt - (susp->susp.current + cnt);
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/* break if to_stop == 0 (we're at the logical stop)
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* AND cnt > 0 (we're not at the beginning of the
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* output block).
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*/
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if (to_stop < 0) to_stop = 0; /* avoids rounding errors */
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if (to_stop < togo) {
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if (to_stop == 0) {
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if (cnt) {
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togo = 0;
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break;
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} else /* keep togo as is: since cnt == 0, we
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* can set the logical stop flag on this
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* output block
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*/
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susp->logically_stopped = true;
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} else /* limit togo so we can start a new
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* block at the LST
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*/
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togo = to_stop;
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}
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}
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{ long cur = susp->susp.current + cnt;
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n = susp->next_breakpoint - cur;
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if (n == 0) n = siosc_table_update(susp, cur);
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/* don't run past terminate time */
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if (susp->terminate_cnt != UNKNOWN &&
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susp->terminate_cnt <= susp->susp.current + cnt + togo) {
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togo = (int) (susp->terminate_cnt - (susp->susp.current + cnt));
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if (togo < 0) togo = 0; /* avoids rounding errros */
|
||||
if (togo == 0) break;
|
||||
}
|
||||
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;
|
||||
s_fm_pHaSe_ReG = susp->s_fm_pHaSe;
|
||||
s_fm_x1_sample_reg = susp->s_fm_x1_sample;
|
||||
out_ptr_reg = out_ptr;
|
||||
if (n) do { /* the inner sample computation loop */
|
||||
|
||||
/* don't run past logical stop time */
|
||||
if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
|
||||
int64_t 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 < 0) to_stop = 0; /* avoids rounding errors */
|
||||
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 = (int) to_stop;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
{ int64_t cur = susp->susp.current + cnt;
|
||||
int64_t n64 = susp->next_breakpoint - cur;
|
||||
if (n64 == 0) n64 = siosc_table_update(susp, cur);
|
||||
if (n64 < togo) togo = (int) n64;
|
||||
}
|
||||
|
||||
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;
|
||||
s_fm_pHaSe_ReG = susp->s_fm_pHaSe;
|
||||
s_fm_x1_sample_reg = susp->s_fm_x1_sample;
|
||||
out_ptr_reg = out_ptr;
|
||||
if (n) do { /* the inner sample computation loop */
|
||||
long table_index;
|
||||
double xa, xb;
|
||||
if (s_fm_pHaSe_ReG >= 1.0) {
|
||||
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);
|
||||
}
|
||||
/* 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];
|
||||
@@ -356,31 +355,31 @@ void siosc_i_fetch(snd_susp_type a_susp, snd_list_type snd_list)
|
||||
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 */
|
||||
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;
|
||||
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);
|
||||
snd_list_terminate(snd_list);
|
||||
} else {
|
||||
snd_list->block_len = cnt;
|
||||
susp->susp.current += cnt;
|
||||
snd_list->block_len = cnt;
|
||||
susp->susp.current += cnt;
|
||||
}
|
||||
/* test for logical stop */
|
||||
if (susp->logically_stopped) {
|
||||
snd_list->logically_stopped = true;
|
||||
snd_list->logically_stopped = true;
|
||||
} else if (susp->susp.log_stop_cnt == susp->susp.current) {
|
||||
susp->logically_stopped = true;
|
||||
susp->logically_stopped = true;
|
||||
}
|
||||
} /* siosc_i_fetch */
|
||||
|
||||
@@ -411,81 +410,81 @@ void siosc_r_fetch(snd_susp_type a_susp, snd_list_type snd_list)
|
||||
|
||||
/* make sure sounds are primed with first values */
|
||||
if (!susp->started) {
|
||||
susp->started = true;
|
||||
susp->s_fm_pHaSe = 1.0;
|
||||
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;
|
||||
/* 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) togo = 0; /* avoids rounding errros */
|
||||
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 < 0) to_stop = 0; /* avoids rounding errors */
|
||||
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);
|
||||
/* 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 = (int64_t) ((1.0 - susp->s_fm_pHaSe) *
|
||||
susp->output_per_s_fm);
|
||||
}
|
||||
togo = (int) 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 = (int) (susp->terminate_cnt - (susp->susp.current + cnt));
|
||||
if (togo < 0) togo = 0; /* avoids rounding errros */
|
||||
if (togo == 0) break;
|
||||
}
|
||||
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 */
|
||||
|
||||
/* don't run past logical stop time */
|
||||
if (!susp->logically_stopped && susp->susp.log_stop_cnt != UNKNOWN) {
|
||||
int64_t 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 < 0) to_stop = 0; /* avoids rounding errors */
|
||||
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 = (int) to_stop;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
{ int64_t cur = susp->susp.current + cnt;
|
||||
int64_t n64 = susp->next_breakpoint - cur;
|
||||
if (n64 == 0) n64 = siosc_table_update(susp, cur);
|
||||
if (n64 < togo) togo = (int) n64;
|
||||
}
|
||||
|
||||
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;
|
||||
@@ -502,46 +501,46 @@ void siosc_r_fetch(snd_susp_type a_susp, snd_list_type snd_list)
|
||||
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 */
|
||||
} 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;
|
||||
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);
|
||||
snd_list_terminate(snd_list);
|
||||
} else {
|
||||
snd_list->block_len = cnt;
|
||||
susp->susp.current += cnt;
|
||||
snd_list->block_len = cnt;
|
||||
susp->susp.current += cnt;
|
||||
}
|
||||
/* test for logical stop */
|
||||
if (susp->logically_stopped) {
|
||||
snd_list->logically_stopped = true;
|
||||
snd_list->logically_stopped = true;
|
||||
} else if (susp->susp.log_stop_cnt == susp->susp.current) {
|
||||
susp->logically_stopped = true;
|
||||
susp->logically_stopped = true;
|
||||
}
|
||||
} /* siosc_r_fetch */
|
||||
|
||||
|
||||
void siosc_toss_fetch(snd_susp_type a_susp, snd_list_type snd_list)
|
||||
{
|
||||
{
|
||||
siosc_susp_type susp = (siosc_susp_type) a_susp;
|
||||
time_type final_time = susp->susp.t0;
|
||||
long n;
|
||||
int n;
|
||||
|
||||
/* fetch samples from s_fm up to final_time for this block of zeros */
|
||||
while ((ROUNDBIG((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);
|
||||
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 = ROUNDBIG((final_time - susp->s_fm->t0) * susp->s_fm->sr -
|
||||
n = (int) ROUNDBIG((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);
|
||||
|
||||
Reference in New Issue
Block a user