Use master state for trampolines
[lsnes.git] / src / core / audioapi.cpp
blobb771f0df9fd41d7fe7ca7f9cd1616aa14b6978ff
1 #include "core/advdumper.hpp"
2 #include "core/audioapi.hpp"
3 #include "core/dispatch.hpp"
4 #include "core/framerate.hpp"
5 #include "core/instance.hpp"
6 #include "library/minmax.hpp"
7 #include "library/threads.hpp"
9 #include <cstring>
10 #include <cmath>
11 #include <iostream>
12 #include <unistd.h>
13 #include <sys/time.h>
15 #define MUSIC_BUFFERS 8
16 #define MAX_VOICE_ADJUST 200
18 bool audioapi_instance::vu_disabled = false;
20 audioapi_instance::dummy_cb_proc::dummy_cb_proc(audioapi_instance& _parent)
21 : parent(_parent)
25 int audioapi_instance::dummy_cb_proc::operator()()
27 int16_t buf[16384];
28 uint64_t last_ts = framerate_regulator::get_utime();
29 while(!parent.dummy_cb_quit) {
30 uint64_t cur_ts = framerate_regulator::get_utime();
31 uint64_t dt = cur_ts - last_ts;
32 last_ts = cur_ts;
33 unsigned samples = dt / 25;
34 if(samples > 16384)
35 samples = 16384; //Don't get crazy.
36 if(parent.dummy_cb_active_play)
37 parent.get_mixed(buf, samples, false);
38 if(parent.dummy_cb_active_record)
39 parent.put_voice(NULL, samples);
40 usleep(10000);
42 return 0;
45 namespace
47 // | -1 1 -1 1 | 1 0 0 0 |
48 // | 0 0 0 1 | 0 1 0 0 |
49 // | 1 1 1 1 | 0 0 1 0 |
50 // | 8 4 2 1 | 0 0 0 1 |
53 // | 6 0 0 0 |-1 4 -3 1 | |-1 3 -3 1|
54 // | 0 6 0 0 | 3 -6 3 0 | 1/6 | 3 -6 3 0|
55 // | 0 0 6 0 |-2 -4 6 -1 | |-2 -3 6 -1|
56 // | 0 0 0 6 | 0 6 0 0 | | 0 6 0 0|
60 void cubicitr_solve(double v1, double v2, double v3, double v4, double& A, double& B, double& C, double& D)
62 A = (-v1 + 3 * v2 - 3 * v3 + v4) / 6;
63 B = (v1 - 2 * v2 + v3) / 2;
64 C = (-2 * v1 - 3 * v2 + 6 * v3 - v4) / 6;
65 D = v2;
69 audioapi_instance::resampler::resampler()
71 position = 0;
72 vAl = vBl = vCl = vDl = 0;
73 vAr = vBr = vCr = vDr = 0;
76 void audioapi_instance::resampler::resample(float*& in, size_t& insize, float*& out, size_t& outsize, double ratio,
77 bool stereo)
79 double iratio = 1 / ratio;
80 while(outsize) {
81 double newpos = position + iratio;
82 while(newpos >= 1) {
83 //Gotta load a new sample.
84 if(!insize)
85 goto exit;
86 vAl = vBl; vBl = vCl; vCl = vDl; vDl = in[0];
87 vAr = vBr; vBr = vCr; vCr = vDr; vDr = in[stereo ? 1 : 0];
88 --insize;
89 in += (stereo ? 2 : 1);
90 newpos = newpos - 1;
92 position = newpos;
93 double A, B, C, D;
94 cubicitr_solve(vAl, vBl, vCl, vDl, A, B, C, D);
95 *(out++) = ((A * position + B) * position + C) * position + D;
96 if(stereo) {
97 cubicitr_solve(vAr, vBr, vCr, vDr, A, B, C, D);
98 *(out++) = ((A * position + B) * position + C) * position + D;
100 --outsize;
102 exit:
106 audioapi_instance::audioapi_instance()
107 : dummyproc(*this)
109 music_ptr = 0;
110 last_complete_music_seen = MUSIC_BUFFERS + 1;
111 last_complete_music = MUSIC_BUFFERS;
112 for(unsigned i = 0; i < MUSIC_BUFFERS; i++) {
113 music_size[i] = 0;
114 music_rate[i] = 48000;
115 music_stereo[i] = false;
117 voicep_get = 0;
118 voicep_put = 0;
119 voicer_get = 0;
120 voicer_put = 0;
121 voice_rate_play = 40000;
122 orig_voice_rate_play = 40000;
123 voice_rate_rec = 40000;
124 dummy_cb_active_record = false;
125 dummy_cb_active_play = false;
126 dummy_cb_quit = false;
127 _music_volume = 1;
128 _voicep_volume = 32767.0;
129 _voicer_volume = 1.0/32768;
130 last_adjust = false;
133 audioapi_instance::~audioapi_instance()
135 quit();
138 std::pair<unsigned, unsigned> audioapi_instance::voice_rate()
140 return std::make_pair(voice_rate_rec, voice_rate_play);
143 unsigned audioapi_instance::orig_voice_rate()
145 return orig_voice_rate_play;
148 void audioapi_instance::voice_rate(unsigned rate_rec, unsigned rate_play)
150 if(rate_rec)
151 voice_rate_rec = rate_rec;
152 else
153 voice_rate_rec = 40000;
154 dummy_cb_active_record = !rate_rec;
155 if(rate_play)
156 orig_voice_rate_play = voice_rate_play = rate_play;
157 else
158 orig_voice_rate_play = voice_rate_play = 40000;
159 dummy_cb_active_play = !rate_play;
162 unsigned audioapi_instance::voice_p_status()
164 unsigned p = voicep_put;
165 unsigned g = voicep_get;
166 if(g > p)
167 return g - p - 1;
168 else
169 return voicep_bufsize - (p - g) - 1;
172 unsigned audioapi_instance::voice_p_status2()
174 unsigned p = voicep_put;
175 unsigned g = voicep_get;
176 if(g > p)
177 return voicep_bufsize - (g - p);
178 else
179 return (p - g);
182 unsigned audioapi_instance::voice_r_status()
184 unsigned p = voicer_put;
185 unsigned g = voicer_get;
186 if(g > p)
187 return voicer_bufsize - (g - p);
188 else
189 return (p - g);
192 void audioapi_instance::play_voice(float* samples, size_t count)
194 unsigned ptr = voicep_put;
195 for(size_t i = 0; i < count; i++) {
196 voicep_buffer[ptr++] = samples[i];
197 if(ptr == voicep_bufsize)
198 ptr = 0;
200 voicep_put = ptr;
203 void audioapi_instance::record_voice(float* samples, size_t count)
205 unsigned ptr = voicer_get;
206 for(size_t i = 0; i < count; i++) {
207 samples[i] = voicer_buffer[ptr++];
208 if(ptr == voicer_bufsize)
209 ptr = 0;
211 voicer_get = ptr;
214 void audioapi_instance::submit_buffer(int16_t* samples, size_t count, bool stereo, double rate)
216 if(stereo)
217 for(unsigned i = 0; i < count; i++)
218 CORE().mdumper->on_sample(samples[2 * i + 0], samples[2 * i + 1]);
219 else
220 for(unsigned i = 0; i < count; i++)
221 CORE().mdumper->on_sample(samples[i], samples[i]);
222 //Limit buffers to avoid overrunning.
223 if(count > music_bufsize / (stereo ? 2 : 1))
224 count = music_bufsize / (stereo ? 2 : 1);
225 unsigned bidx = last_complete_music;
226 bidx = (bidx > (MUSIC_BUFFERS - 2)) ? 0 : bidx + 1;
227 memcpy(music_buffer + bidx * music_bufsize, samples, count * (stereo ? 2 : 1) * sizeof(int16_t));
228 music_stereo[bidx] = stereo;
229 music_rate[bidx] = rate;
230 music_size[bidx] = count;
231 last_complete_music = bidx;
234 struct audioapi_instance::buffer audioapi_instance::get_music(size_t played)
236 unsigned midx = last_complete_music_seen;
237 unsigned midx2 = last_complete_music;
238 if(midx2 >= MUSIC_BUFFERS) {
239 //Special case: No buffer.
240 struct buffer out;
241 out.samples = NULL;
242 out.pointer = 0;
243 //The rest are arbitrary.
244 out.total = 64;
245 out.stereo = false;
246 out.rate = 48000;
247 return out;
249 //Handle ACK. If the current buffer is too old, we want to ignore the ACK.
250 if(midx >= MUSIC_BUFFERS) {
251 //Load initial buffer.
252 midx = last_complete_music_seen = 0;
253 music_ptr = 0;
254 } else {
255 music_ptr += played;
256 //Otherwise, check if current buffer is not next on the line to be overwritten.
257 if((midx2 + 1) % MUSIC_BUFFERS == midx) {
258 //It is, bump buffer by one.
259 if(!last_adjust && voice_rate_play > orig_voice_rate_play - MAX_VOICE_ADJUST)
260 voice_rate_play--;
261 last_adjust = true;
262 midx = last_complete_music_seen = (midx + 1) % MUSIC_BUFFERS;
263 music_ptr = 0;
264 } else if(music_ptr >= music_size[midx] && midx != midx2) {
265 //It isn't, but current buffer is finished.
266 midx = last_complete_music_seen = (midx + 1) % MUSIC_BUFFERS;
267 music_ptr = 0;
268 last_adjust = false;
269 } else if(music_ptr >= music_size[midx] && midx == midx2) {
270 if(!last_adjust && voice_rate_play < orig_voice_rate_play + MAX_VOICE_ADJUST)
271 voice_rate_play++;
272 last_adjust = true;
273 //Current buffer is finished, but there is no new buffer.
274 //Send silence.
275 } else {
276 last_adjust = false;
277 //Can continue.
280 //Fill the structure.
281 struct buffer out;
282 if(music_ptr < music_size[midx]) {
283 out.samples = music_buffer + midx * music_bufsize;
284 out.pointer = music_ptr;
285 out.total = music_size[midx];
286 out.stereo = music_stereo[midx];
287 out.rate = music_rate[midx];
288 } else {
289 //Run out of buffers to play.
290 out.samples = NULL;
291 out.pointer = 0;
292 out.total = 64; //Arbitrary.
293 out.stereo = music_stereo[midx];
294 out.rate = music_rate[midx];
295 if(out.rate < 100)
296 out.rate = 48000; //Apparently there are buffers with zero rate.
298 return out;
301 void audioapi_instance::get_voice(float* samples, size_t count)
303 unsigned g = voicep_get;
304 unsigned p = voicep_put;
305 if(samples) {
306 for(size_t i = 0; i < count; i++) {
307 if(g != p)
308 samples[i] = _voicep_volume * voicep_buffer[g++];
309 else
310 samples[i] = 0.0;
311 if(g == voicep_bufsize)
312 g = 0;
314 } else {
315 for(size_t i = 0; i < count; i++) {
316 if(g != p)
317 g++;
318 if(g == voicep_bufsize)
319 g = 0;
322 voicep_get = g;
325 void audioapi_instance::put_voice(float* samples, size_t count)
327 unsigned ptr = voicer_put;
328 vu_vin(samples, count, false, voice_rate_rec, _voicer_volume);
329 for(size_t i = 0; i < count; i++) {
330 voicer_buffer[ptr++] = samples ? _voicer_volume * samples[i] : 0.0;
331 if(ptr == voicer_bufsize)
332 ptr = 0;
334 voicer_put = ptr;
337 void audioapi_instance::init()
339 voicep_get = 0;
340 voicep_put = 0;
341 voicer_get = 0;
342 voicer_put = 0;
343 last_complete_music = 3;
344 last_complete_music_seen = 4;
345 dummy_cb_active_play = true;
346 dummy_cb_active_record = true;
347 dummy_cb_quit = false;
348 dummythread = new threads::thread(dummyproc);
351 void audioapi_instance::quit()
353 dummy_cb_quit = true;
354 if(dummythread) {
355 dummythread->join();
356 dummythread = NULL;
360 void audioapi_instance::music_volume(float volume)
362 _music_volume = volume;
365 float audioapi_instance::music_volume()
367 return _music_volume;
370 void audioapi_instance::voicep_volume(float volume)
372 _voicep_volume = volume * 32767;
375 float audioapi_instance::voicep_volume()
377 return _voicep_volume / 32767;
380 void audioapi_instance::voicer_volume(float volume)
382 _voicer_volume = volume / 32768;
385 float audioapi_instance::voicer_volume()
387 return _voicer_volume * 32768;
390 void audioapi_instance::get_mixed(int16_t* samples, size_t count, bool stereo)
392 const size_t intbuf_size = 256;
393 float intbuf[intbuf_size];
394 float intbuf2[intbuf_size];
395 while(count > 0) {
396 buffer b = get_music(0);
397 float* in = intbuf;
398 float* out = intbuf2;
399 size_t outdata_used;
400 if(b.stereo) {
401 size_t indata = min(b.total - b.pointer, intbuf_size / 2);
402 size_t outdata = min(intbuf_size / 2, count);
403 size_t indata_used = indata;
404 outdata_used = outdata;
405 if(b.samples)
406 for(size_t i = 0; i < 2 * indata; i++)
407 intbuf[i] = _music_volume * b.samples[i + 2 * b.pointer];
408 else
409 for(size_t i = 0; i < 2 * indata; i++)
410 intbuf[i] = 0;
411 music_resampler.resample(in, indata, out, outdata, (double)voice_rate_play / b.rate, true);
412 indata_used -= indata;
413 outdata_used -= outdata;
414 get_music(indata_used);
415 get_voice(intbuf, outdata_used);
417 vu_mleft(intbuf2, outdata_used, true, voice_rate_play, 1 / 32768.0);
418 vu_mright(intbuf2 + 1, outdata_used, true, voice_rate_play, 1 / 32768.0);
419 vu_vout(intbuf, outdata_used, false, voice_rate_play, 1 / 32768.0);
421 for(size_t i = 0; i < outdata_used * (stereo ? 2 : 1); i++)
422 intbuf2[i] = max(min(intbuf2[i] + intbuf[i / 2], 32766.0f), -32767.0f);
423 if(stereo)
424 for(size_t i = 0; i < outdata_used * 2; i++)
425 samples[i] = intbuf2[i];
426 else
427 for(size_t i = 0; i < outdata_used; i++)
428 samples[i] = (intbuf2[2 * i + 0] + intbuf2[2 * i + 1]) / 2;
429 } else {
430 size_t indata = min(b.total - b.pointer, intbuf_size);
431 size_t outdata = min(intbuf_size, count);
432 size_t indata_used = indata;
433 outdata_used = outdata;
434 if(b.samples)
435 for(size_t i = 0; i < indata; i++)
436 intbuf[i] = _music_volume * b.samples[i + b.pointer];
437 else
438 for(size_t i = 0; i < indata; i++)
439 intbuf[i] = 0;
440 music_resampler.resample(in, indata, out, outdata, (double)voice_rate_play / b.rate, false);
441 indata_used -= indata;
442 outdata_used -= outdata;
443 get_music(indata_used);
444 get_voice(intbuf, outdata_used);
446 vu_mleft(intbuf2, outdata_used, false, voice_rate_play, 1 / 32768.0);
447 vu_mright(intbuf2, outdata_used, false, voice_rate_play, 1 / 32768.0);
448 vu_vout(intbuf, outdata_used, false, voice_rate_play, 1 / 32768.0);
450 for(size_t i = 0; i < outdata_used; i++)
451 intbuf2[i] = max(min(intbuf2[i] + intbuf[i], 32766.0f), -32767.0f);
452 if(stereo)
453 for(size_t i = 0; i < outdata_used; i++) {
454 samples[2 * i + 0] = intbuf2[i];
455 samples[2 * i + 1] = intbuf2[i];
457 else
458 for(size_t i = 0; i < outdata_used; i++)
459 samples[i] = intbuf2[i];
461 samples += (stereo ? 2 : 1) * outdata_used;
462 count -= outdata_used;
466 audioapi_instance::vumeter::vumeter()
468 accumulator = 0;
469 samples = 0;
470 vu = -999.0;
473 void audioapi_instance::vumeter::operator()(float* asamples, size_t count, bool stereo, double rate, double scale)
475 size_t limit = rate / 25;
476 //If we already at or exceed limit, cut immediately.
477 if(samples >= limit)
478 update_vu();
479 if(asamples) {
480 double sscale = scale * scale;
481 size_t j = 0;
482 if(stereo)
483 for(size_t i = 0; i < count; i++) {
484 accumulator += sscale * asamples[j] * asamples[j];
485 j += 2;
486 samples++;
487 if(samples >= limit)
488 update_vu();
490 else
491 for(size_t i = 0; i < count; i++) {
492 accumulator += sscale * asamples[i] * asamples[i];
493 samples++;
494 if(samples >= limit)
495 update_vu();
497 } else
498 for(size_t i = 0; i < count; i++) {
499 samples++;
500 if(samples >= limit)
501 update_vu();
505 void audioapi_instance::vumeter::update_vu()
507 if(vu_disabled)
508 return;
509 if(!samples) {
510 vu = -999.0;
511 accumulator = 0;
512 } else {
513 double a = accumulator;
514 if(a < 1e-120)
515 a = 1e-120; //Don't take log of zero.
516 vu = 10 / log(10) * (log(a) - log(samples));
517 if(vu < -999.0)
518 vu = -999.0;
519 accumulator = 0;
520 samples = 0;
522 CORE().dispatch->vu_change();
525 void audioapi_instance::disable_vu_updates()
527 vu_disabled = true;