lxdream 0.9.1| filename | src/aica/audio.c |
| changeset | 1089:a3984d242909 |
| prev | 1024:c67f2d61ab97 |
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| author | nkeynes |
| date | Mon Dec 07 08:59:50 2009 +1000 (14 years ago) |
| permissions | -rw-r--r-- |
| last change | Mask audio RAM accesses from the mixer to ensure they're in bounds. Uncertain if this is correct (vs eg zero-fill), but it's at least consistent with the masking of the channel start position. |
| view | annotate | diff | log | raw |
1 /**
2 * $Id$
3 *
4 * Audio mixer core. Combines all the active streams into a single sound
5 * buffer for output.
6 *
7 * Copyright (c) 2005 Nathan Keynes.
8 *
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
13 *
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
18 */
20 #include "aica/aica.h"
21 #include "aica/audio.h"
22 #include <glib/gmem.h>
23 #include "dream.h"
24 #include <assert.h>
25 #include <string.h>
27 #define MAX_AUDIO_DRIVERS 16
28 extern struct audio_driver audio_null_driver;
29 extern struct audio_driver audio_osx_driver;
30 extern struct audio_driver audio_pulse_driver;
31 extern struct audio_driver audio_esd_driver;
32 extern struct audio_driver audio_alsa_driver;
33 extern struct audio_driver audio_sdl_driver;
35 static int audio_driver_count = 0;
36 static audio_driver_t audio_driver_list[MAX_AUDIO_DRIVERS] = {};
38 #define NUM_BUFFERS 3
39 #define MS_PER_BUFFER 100
41 #define BUFFER_EMPTY 0
42 #define BUFFER_WRITING 1
43 #define BUFFER_FULL 2
45 struct audio_state {
46 audio_buffer_t output_buffers[NUM_BUFFERS];
47 int write_buffer;
48 int read_buffer;
49 uint32_t output_format;
50 uint32_t output_rate;
51 uint32_t output_sample_size;
52 struct audio_channel channels[AUDIO_CHANNEL_COUNT];
53 } audio;
55 audio_driver_t audio_driver = NULL;
57 #define NEXT_BUFFER() ((audio.write_buffer == NUM_BUFFERS-1) ? 0 : audio.write_buffer+1)
59 /**
60 * Preserve audio channel state only - don't bother saving the buffers
61 */
62 void audio_save_state( FILE *f )
63 {
64 fwrite( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
65 }
67 int audio_load_state( FILE *f )
68 {
69 int read = fread( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
70 return (read == AUDIO_CHANNEL_COUNT ? 0 : -1 );
71 }
73 static int audio_driver_priority_compare(const void *a, const void *b)
74 {
75 audio_driver_t ada = *(audio_driver_t *)a;
76 audio_driver_t adb = *(audio_driver_t *)b;
77 return ada->priority - adb->priority;
78 }
80 static int audio_driver_name_compare(const void *a, const void *b)
81 {
82 audio_driver_t ada = *(audio_driver_t *)a;
83 audio_driver_t adb = *(audio_driver_t *)b;
84 return strcasecmp( ada->name, adb->name );
85 }
89 gboolean audio_register_driver( audio_driver_t driver )
90 {
91 if( audio_driver_count >= MAX_AUDIO_DRIVERS ) {
92 return FALSE;
93 }
94 audio_driver_list[audio_driver_count++] = driver;
95 qsort( audio_driver_list, audio_driver_count, sizeof( audio_driver_t ), audio_driver_priority_compare );
96 return TRUE;
97 }
99 audio_driver_t get_audio_driver_by_name( const char *name )
100 {
101 int i;
102 if( name == NULL ) {
103 return audio_driver_list[0];
104 }
105 for( i=0; i < audio_driver_count; i++ ) {
106 if( strcasecmp( audio_driver_list[i]->name, name ) == 0 ) {
107 return audio_driver_list[i];
108 }
109 }
111 return NULL;
112 }
114 void print_audio_drivers( FILE * out )
115 {
116 int i;
117 audio_driver_t temp_list[MAX_AUDIO_DRIVERS];
118 memcpy( temp_list, audio_driver_list, audio_driver_count*sizeof(audio_driver_t) );
119 qsort( temp_list, audio_driver_count, sizeof(audio_driver_t), audio_driver_name_compare );
120 fprintf( out, "Available audio drivers:\n" );
121 for( i=0; i < audio_driver_count; i++ ) {
122 fprintf( out, " %-8s %s\n", temp_list[i]->name,
123 gettext(temp_list[i]->description) );
124 }
125 }
127 audio_driver_t audio_init_driver( const char *preferred_driver )
128 {
129 audio_driver_t audio_driver = get_audio_driver_by_name(preferred_driver);
130 if( audio_driver == NULL ) {
131 ERROR( "Audio driver '%s' not found, aborting.", preferred_driver );
132 exit(2);
133 } else if( audio_set_driver( audio_driver ) == FALSE ) {
134 int i;
135 for( i=0; i < audio_driver_count; i++ ) {
136 if( audio_driver_list[i] != audio_driver &&
137 audio_set_driver( audio_driver_list[i] ) ) {
138 ERROR( "Failed to initialize audio driver %s, falling back to %s",
139 audio_driver->name, audio_driver_list[i]->name );
140 return audio_driver_list[i];
141 }
142 }
143 ERROR( "Unable to intialize any audio driver, aborting." );
144 exit(2);
145 }
146 return audio_driver;
147 }
149 void audio_start_driver(void)
150 {
151 if( audio_driver != NULL && audio_driver->start != NULL ) {
152 audio_driver->start();
153 }
154 }
156 void audio_stop_driver(void)
157 {
158 if( audio_driver != NULL && audio_driver->stop != NULL ) {
159 audio_driver->stop();
160 }
161 }
163 /**
164 * Set the output driver, sample rate and format. Also initializes the
165 * output buffers, flushing any current data and reallocating as
166 * necessary.
167 */
168 gboolean audio_set_driver( audio_driver_t driver )
169 {
170 uint32_t bytes_per_sample = 1;
171 uint32_t samples_per_buffer;
172 int i;
174 if( audio_driver == NULL || driver != NULL ) {
175 if( driver == NULL )
176 driver = &audio_null_driver;
177 if( driver != audio_driver ) {
178 if( !driver->init() )
179 return FALSE;
180 audio_driver = driver;
181 }
182 }
184 switch( driver->sample_format & AUDIO_FMT_SAMPLE_MASK ) {
185 case AUDIO_FMT_8BIT:
186 bytes_per_sample = 1;
187 break;
188 case AUDIO_FMT_16BIT:
189 bytes_per_sample = 2;
190 break;
191 case AUDIO_FMT_FLOAT:
192 bytes_per_sample = 4;
193 break;
194 }
196 if( driver->sample_format & AUDIO_FMT_STEREO )
197 bytes_per_sample <<= 1;
198 if( driver->sample_rate == audio.output_rate &&
199 bytes_per_sample == audio.output_sample_size )
200 return TRUE;
201 samples_per_buffer = (driver->sample_rate * MS_PER_BUFFER / 1000);
202 for( i=0; i<NUM_BUFFERS; i++ ) {
203 if( audio.output_buffers[i] != NULL )
204 free(audio.output_buffers[i]);
205 audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );
206 audio.output_buffers[i]->length = samples_per_buffer * bytes_per_sample;
207 audio.output_buffers[i]->posn = 0;
208 audio.output_buffers[i]->status = BUFFER_EMPTY;
209 }
210 audio.output_format = driver->sample_format;
211 audio.output_rate = driver->sample_rate;
212 audio.output_sample_size = bytes_per_sample;
213 audio.write_buffer = 0;
214 audio.read_buffer = 0;
216 return TRUE;
217 }
219 /**
220 * Mark the current write buffer as full and prepare the next buffer for
221 * writing. Returns the next buffer to write to.
222 * If all buffers are full, returns NULL.
223 */
224 audio_buffer_t audio_next_write_buffer( )
225 {
226 audio_buffer_t result = NULL;
227 audio_buffer_t current = audio.output_buffers[audio.write_buffer];
228 current->status = BUFFER_FULL;
229 if( audio.read_buffer == audio.write_buffer &&
230 audio_driver->process_buffer( current ) ) {
231 audio_next_read_buffer();
232 }
233 int next_buffer = NEXT_BUFFER();
234 result = audio.output_buffers[next_buffer];
235 if( result->status == BUFFER_FULL )
236 return NULL;
237 else {
238 audio.write_buffer = next_buffer;
239 result->status = BUFFER_WRITING;
240 return result;
241 }
242 }
244 /**
245 * Mark the current read buffer as empty and return the next buffer for
246 * reading. If there is no next buffer yet, returns NULL.
247 */
248 audio_buffer_t audio_next_read_buffer( )
249 {
250 audio_buffer_t current = audio.output_buffers[audio.read_buffer];
251 if( current->status == BUFFER_FULL ) {
252 // Current read buffer has data, which we've just emptied
253 current->status = BUFFER_EMPTY;
254 current->posn = 0;
255 audio.read_buffer++;
256 if( audio.read_buffer == NUM_BUFFERS )
257 audio.read_buffer = 0;
259 current = audio.output_buffers[audio.read_buffer];
260 if( current->status == BUFFER_FULL ) {
261 current->posn = 0;
262 return current;
263 }
264 else return NULL;
265 } else {
266 return NULL;
267 }
269 }
271 /*************************** ADPCM ***********************************/
273 /**
274 * The following section borrows heavily from ffmpeg, which is
275 * copyright (c) 2001-2003 by the fine folks at the ffmpeg project,
276 * distributed under the GPL version 2 or later.
277 */
279 #define CLAMP_TO_SHORT(value) \
280 if (value > 32767) \
281 value = 32767; \
282 else if (value < -32768) \
283 value = -32768; \
285 static const int yamaha_indexscale[] = {
286 230, 230, 230, 230, 307, 409, 512, 614,
287 230, 230, 230, 230, 307, 409, 512, 614
288 };
290 static const int yamaha_difflookup[] = {
291 1, 3, 5, 7, 9, 11, 13, 15,
292 -1, -3, -5, -7, -9, -11, -13, -15
293 };
295 static inline short adpcm_yamaha_decode_nibble( audio_channel_t c,
296 unsigned char nibble )
297 {
298 if( c->adpcm_step == 0 ) {
299 c->adpcm_predict = 0;
300 c->adpcm_step = 127;
301 }
303 c->adpcm_predict += (c->adpcm_step * yamaha_difflookup[nibble]) >> 3;
304 CLAMP_TO_SHORT(c->adpcm_predict);
305 c->adpcm_step = (c->adpcm_step * yamaha_indexscale[nibble]) >> 8;
306 c->adpcm_step = CLAMP(c->adpcm_step, 127, 24567);
307 return c->adpcm_predict;
308 }
310 /*************************** Sample mixer *****************************/
312 /**
313 * Mix a single output sample.
314 */
315 void audio_mix_samples( int num_samples )
316 {
317 int i, j;
318 int32_t result_buf[num_samples][2];
320 memset( &result_buf, 0, sizeof(result_buf) );
322 for( i=0; i < AUDIO_CHANNEL_COUNT; i++ ) {
323 audio_channel_t channel = &audio.channels[i];
324 if( channel->active ) {
325 int32_t sample;
326 int vol_left = (channel->vol * (32 - channel->pan)) >> 5;
327 int vol_right = (channel->vol * (channel->pan + 1)) >> 5;
328 switch( channel->sample_format ) {
329 case AUDIO_FMT_16BIT:
330 for( j=0; j<num_samples; j++ ) {
331 sample = *(int16_t *)(aica_main_ram + ((channel->start + channel->posn*2)&AUDIO_MEM_MASK));
332 result_buf[j][0] += sample * vol_left;
333 result_buf[j][1] += sample * vol_right;
335 channel->posn_left += channel->sample_rate;
336 while( channel->posn_left > audio.output_rate ) {
337 channel->posn_left -= audio.output_rate;
338 channel->posn++;
340 if( channel->posn == channel->end ) {
341 if( channel->loop ) {
342 channel->posn = channel->loop_start;
343 channel->loop = LOOP_LOOPED;
344 } else {
345 audio_stop_channel(i);
346 j = num_samples;
347 break;
348 }
349 }
350 }
351 }
352 break;
353 case AUDIO_FMT_8BIT:
354 for( j=0; j<num_samples; j++ ) {
355 sample = (*(int8_t *)(aica_main_ram + ((channel->start + channel->posn)&AUDIO_MEM_MASK))) << 8;
356 result_buf[j][0] += sample * vol_left;
357 result_buf[j][1] += sample * vol_right;
359 channel->posn_left += channel->sample_rate;
360 while( channel->posn_left > audio.output_rate ) {
361 channel->posn_left -= audio.output_rate;
362 channel->posn++;
364 if( channel->posn == channel->end ) {
365 if( channel->loop ) {
366 channel->posn = channel->loop_start;
367 channel->loop = LOOP_LOOPED;
368 } else {
369 audio_stop_channel(i);
370 j = num_samples;
371 break;
372 }
373 }
374 }
375 }
376 break;
377 case AUDIO_FMT_ADPCM:
378 for( j=0; j<num_samples; j++ ) {
379 sample = (int16_t)channel->adpcm_predict;
380 result_buf[j][0] += sample * vol_left;
381 result_buf[j][1] += sample * vol_right;
382 channel->posn_left += channel->sample_rate;
383 while( channel->posn_left > audio.output_rate ) {
384 channel->posn_left -= audio.output_rate;
385 channel->posn++;
386 if( channel->posn == channel->end ) {
387 if( channel->loop ) {
388 channel->posn = channel->loop_start;
389 channel->loop = LOOP_LOOPED;
390 channel->adpcm_predict = 0;
391 channel->adpcm_step = 0;
392 } else {
393 audio_stop_channel(i);
394 j = num_samples;
395 break;
396 }
397 }
398 uint8_t data = *(uint8_t *)(aica_main_ram + ((channel->start + (channel->posn>>1))&AUDIO_MEM_MASK));
399 if( channel->posn&1 ) {
400 adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
401 } else {
402 adpcm_yamaha_decode_nibble( channel, data & 0x0F );
403 }
404 }
405 }
406 break;
407 default:
408 break;
409 }
410 }
411 }
413 /* Down-render to the final output format */
414 audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
415 if( buf->status == BUFFER_FULL ) {
416 buf = audio_next_write_buffer();
417 if( buf == NULL ) { // no available space
418 return;
419 }
420 }
422 switch( audio.output_format & AUDIO_FMT_SAMPLE_MASK ) {
423 case AUDIO_FMT_FLOAT: {
424 float scale = 1.0/SHRT_MAX;
425 float *data = (float *)&buf->data[buf->posn];
426 for( j=0; j<num_samples; j++ ) {
427 *data++ = scale * (result_buf[j][0] >> 6);
428 *data++ = scale * (result_buf[j][1] >> 6);
429 buf->posn += 8;
430 if( buf->posn == buf->length ) {
431 buf = audio_next_write_buffer();
432 if( buf == NULL ) {
433 break;
434 }
435 data = (float *)&buf->data[0];
436 }
437 }
438 break;
439 }
440 case AUDIO_FMT_16BIT: {
441 int16_t *data = (int16_t *)&buf->data[buf->posn];
442 for( j=0; j < num_samples; j++ ) {
443 *data++ = (int16_t)(result_buf[j][0] >> 6);
444 *data++ = (int16_t)(result_buf[j][1] >> 6);
445 buf->posn += 4;
446 if( buf->posn == buf->length ) {
447 buf = audio_next_write_buffer();
448 if( buf == NULL ) {
449 // All buffers are full
450 break;
451 }
452 data = (int16_t *)&buf->data[0];
453 }
454 }
455 break;
456 }
457 case AUDIO_FMT_8BIT: {
458 int8_t *data = (int8_t *)&buf->data[buf->posn];
459 for( j=0; j < num_samples; j++ ) {
460 *data++ = (int8_t)(result_buf[j][0] >> 16);
461 *data++ = (int8_t)(result_buf[j][1] >> 16);
462 buf->posn += 2;
463 if( buf->posn == buf->length ) {
464 buf = audio_next_write_buffer();
465 if( buf == NULL ) {
466 // All buffers are full
467 break;
468 }
469 buf = audio.output_buffers[audio.write_buffer];
470 data = (int8_t *)&buf->data[0];
471 }
472 }
473 break;
474 }
475 }
476 }
478 /********************** Internal AICA calls ***************************/
480 audio_channel_t audio_get_channel( int channel )
481 {
482 return &audio.channels[channel];
483 }
485 void audio_start_stop_channel( int channel, gboolean start )
486 {
487 if( audio.channels[channel].active ) {
488 if( !start ) {
489 audio_stop_channel(channel);
490 }
491 } else if( start ) {
492 audio_start_channel(channel);
493 }
494 }
496 void audio_stop_channel( int channel )
497 {
498 audio.channels[channel].active = FALSE;
499 }
502 void audio_start_channel( int channel )
503 {
504 audio.channels[channel].posn = 0;
505 audio.channels[channel].posn_left = 0;
506 audio.channels[channel].active = TRUE;
507 if( audio.channels[channel].sample_format == AUDIO_FMT_ADPCM ) {
508 audio.channels[channel].adpcm_step = 0;
509 audio.channels[channel].adpcm_predict = 0;
510 uint8_t data = ((uint8_t *)(aica_main_ram + audio.channels[channel].start))[0];
511 adpcm_yamaha_decode_nibble( &audio.channels[channel], data & 0x0F );
512 }
513 }
.