lxdream 0.9.1| filename | src/aica/audio.c |
| changeset | 700:4650d0c7f6f9 |
| prev | 697:479b8c213f61 |
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| author | nkeynes |
| date | Mon Jul 14 00:09:51 2008 +0000 (15 years ago) |
| permissions | -rw-r--r-- |
| last change | Fix version number in configure.in Remove obsolete x86_64 AM conditional Add --disable-translator option for testing purposes |
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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>
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;
34 audio_driver_t audio_driver_list[] = {
35 #ifdef HAVE_CORE_AUDIO
36 &audio_osx_driver,
37 #endif
38 #ifdef HAVE_PULSE
39 &audio_pulse_driver,
40 #endif
41 #ifdef HAVE_ESOUND
42 &audio_esd_driver,
43 #endif
44 #ifdef HAVE_ALSA
45 &audio_alsa_driver,
46 #endif
47 &audio_null_driver,
48 NULL };
50 #define NUM_BUFFERS 3
51 #define MS_PER_BUFFER 100
53 #define BUFFER_EMPTY 0
54 #define BUFFER_WRITING 1
55 #define BUFFER_FULL 2
57 struct audio_state {
58 audio_buffer_t output_buffers[NUM_BUFFERS];
59 int write_buffer;
60 int read_buffer;
61 uint32_t output_format;
62 uint32_t output_rate;
63 uint32_t output_sample_size;
64 struct audio_channel channels[AUDIO_CHANNEL_COUNT];
65 } audio;
67 audio_driver_t audio_driver = NULL;
69 #define NEXT_BUFFER() ((audio.write_buffer == NUM_BUFFERS-1) ? 0 : audio.write_buffer+1)
71 extern char *arm_mem;
73 /**
74 * Preserve audio channel state only - don't bother saving the buffers
75 */
76 void audio_save_state( FILE *f )
77 {
78 fwrite( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
79 }
81 int audio_load_state( FILE *f )
82 {
83 int read = fread( &audio.channels[0], sizeof(struct audio_channel), AUDIO_CHANNEL_COUNT, f );
84 return (read == AUDIO_CHANNEL_COUNT ? 0 : -1 );
85 }
87 audio_driver_t get_audio_driver_by_name( const char *name )
88 {
89 int i;
90 if( name == NULL ) {
91 return audio_driver_list[0];
92 }
93 for( i=0; audio_driver_list[i] != NULL; i++ ) {
94 if( strcasecmp( audio_driver_list[i]->name, name ) == 0 ) {
95 return audio_driver_list[i];
96 }
97 }
99 return NULL;
100 }
102 void print_audio_drivers( FILE * out )
103 {
104 int i;
105 fprintf( out, "Available audio drivers:\n" );
106 for( i=0; audio_driver_list[i] != NULL; i++ ) {
107 fprintf( out, " %-8s %s\n", audio_driver_list[i]->name,
108 gettext(audio_driver_list[i]->description) );
109 }
110 }
112 audio_driver_t audio_init_driver( const char *preferred_driver )
113 {
114 audio_driver_t audio_driver = get_audio_driver_by_name(preferred_driver);
115 if( audio_driver == NULL ) {
116 ERROR( "Audio driver '%s' not found, aborting.", preferred_driver );
117 exit(2);
118 } else if( audio_set_driver( audio_driver ) == FALSE ) {
119 ERROR( "Failed to initialize audio driver '%s', using null driver",
120 audio_driver->name );
121 audio_set_driver( &audio_null_driver );
122 }
123 }
125 /**
126 * Set the output driver, sample rate and format. Also initializes the
127 * output buffers, flushing any current data and reallocating as
128 * necessary.
129 */
130 gboolean audio_set_driver( audio_driver_t driver )
131 {
132 uint32_t bytes_per_sample = 1;
133 uint32_t samples_per_buffer;
134 int i;
136 if( audio_driver == NULL || driver != NULL ) {
137 if( driver == NULL )
138 driver = &audio_null_driver;
139 if( driver != audio_driver ) {
140 if( !driver->init() )
141 return FALSE;
142 audio_driver = driver;
143 }
144 }
146 switch( driver->sample_format & AUDIO_FMT_SAMPLE_MASK ) {
147 case AUDIO_FMT_8BIT:
148 bytes_per_sample = 1;
149 break;
150 case AUDIO_FMT_16BIT:
151 bytes_per_sample = 2;
152 break;
153 case AUDIO_FMT_FLOAT:
154 bytes_per_sample = 4;
155 break;
156 }
158 if( driver->sample_format & AUDIO_FMT_STEREO )
159 bytes_per_sample <<= 1;
160 if( driver->sample_rate == audio.output_rate &&
161 bytes_per_sample == audio.output_sample_size )
162 return TRUE;
163 samples_per_buffer = (driver->sample_rate * MS_PER_BUFFER / 1000);
164 for( i=0; i<NUM_BUFFERS; i++ ) {
165 if( audio.output_buffers[i] != NULL )
166 free(audio.output_buffers[i]);
167 audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );
168 audio.output_buffers[i]->length = samples_per_buffer * bytes_per_sample;
169 audio.output_buffers[i]->posn = 0;
170 audio.output_buffers[i]->status = BUFFER_EMPTY;
171 }
172 audio.output_format = driver->sample_format;
173 audio.output_rate = driver->sample_rate;
174 audio.output_sample_size = bytes_per_sample;
175 audio.write_buffer = 0;
176 audio.read_buffer = 0;
178 return TRUE;
179 }
181 /**
182 * Mark the current write buffer as full and prepare the next buffer for
183 * writing. Returns the next buffer to write to.
184 * If all buffers are full, returns NULL.
185 */
186 audio_buffer_t audio_next_write_buffer( )
187 {
188 audio_buffer_t result = NULL;
189 audio_buffer_t current = audio.output_buffers[audio.write_buffer];
190 current->status = BUFFER_FULL;
191 if( audio.read_buffer == audio.write_buffer &&
192 audio_driver->process_buffer( current ) ) {
193 audio_next_read_buffer();
194 }
195 int next_buffer = NEXT_BUFFER();
196 result = audio.output_buffers[next_buffer];
197 if( result->status == BUFFER_FULL )
198 return NULL;
199 else {
200 audio.write_buffer = next_buffer;
201 result->status = BUFFER_WRITING;
202 return result;
203 }
204 }
206 /**
207 * Mark the current read buffer as empty and return the next buffer for
208 * reading. If there is no next buffer yet, returns NULL.
209 */
210 audio_buffer_t audio_next_read_buffer( )
211 {
212 audio_buffer_t current = audio.output_buffers[audio.read_buffer];
213 if( current->status == BUFFER_FULL ) {
214 // Current read buffer has data, which we've just emptied
215 current->status = BUFFER_EMPTY;
216 current->posn = 0;
217 audio.read_buffer++;
218 if( audio.read_buffer == NUM_BUFFERS )
219 audio.read_buffer = 0;
221 current = audio.output_buffers[audio.read_buffer];
222 if( current->status == BUFFER_FULL ) {
223 current->posn = 0;
224 return current;
225 }
226 else return NULL;
227 } else {
228 return NULL;
229 }
231 }
233 /*************************** ADPCM ***********************************/
235 /**
236 * The following section borrows heavily from ffmpeg, which is
237 * copyright (c) 2001-2003 by the fine folks at the ffmpeg project,
238 * distributed under the GPL version 2 or later.
239 */
241 #define CLAMP_TO_SHORT(value) \
242 if (value > 32767) \
243 value = 32767; \
244 else if (value < -32768) \
245 value = -32768; \
247 static const int yamaha_indexscale[] = {
248 230, 230, 230, 230, 307, 409, 512, 614,
249 230, 230, 230, 230, 307, 409, 512, 614
250 };
252 static const int yamaha_difflookup[] = {
253 1, 3, 5, 7, 9, 11, 13, 15,
254 -1, -3, -5, -7, -9, -11, -13, -15
255 };
257 static inline short adpcm_yamaha_decode_nibble( audio_channel_t c,
258 unsigned char nibble )
259 {
260 if( c->adpcm_step == 0 ) {
261 c->adpcm_predict = 0;
262 c->adpcm_step = 127;
263 }
265 c->adpcm_predict += (c->adpcm_step * yamaha_difflookup[nibble]) >> 3;
266 CLAMP_TO_SHORT(c->adpcm_predict);
267 c->adpcm_step = (c->adpcm_step * yamaha_indexscale[nibble]) >> 8;
268 c->adpcm_step = CLAMP(c->adpcm_step, 127, 24567);
269 return c->adpcm_predict;
270 }
272 /*************************** Sample mixer *****************************/
274 /**
275 * Mix a single output sample.
276 */
277 void audio_mix_samples( int num_samples )
278 {
279 int i, j;
280 int32_t result_buf[num_samples][2];
282 memset( &result_buf, 0, sizeof(result_buf) );
284 for( i=0; i < AUDIO_CHANNEL_COUNT; i++ ) {
285 audio_channel_t channel = &audio.channels[i];
286 if( channel->active ) {
287 int32_t sample;
288 int vol_left = (channel->vol * (32 - channel->pan)) >> 5;
289 int vol_right = (channel->vol * (channel->pan + 1)) >> 5;
290 switch( channel->sample_format ) {
291 case AUDIO_FMT_16BIT:
292 for( j=0; j<num_samples; j++ ) {
293 sample = ((int16_t *)(arm_mem + channel->start))[channel->posn];
294 result_buf[j][0] += sample * vol_left;
295 result_buf[j][1] += sample * vol_right;
297 channel->posn_left += channel->sample_rate;
298 while( channel->posn_left > audio.output_rate ) {
299 channel->posn_left -= audio.output_rate;
300 channel->posn++;
302 if( channel->posn == channel->end ) {
303 if( channel->loop ) {
304 channel->posn = channel->loop_start;
305 channel->loop = LOOP_LOOPED;
306 } else {
307 audio_stop_channel(i);
308 j = num_samples;
309 break;
310 }
311 }
312 }
313 }
314 break;
315 case AUDIO_FMT_8BIT:
316 for( j=0; j<num_samples; j++ ) {
317 sample = ((int8_t *)(arm_mem + channel->start))[channel->posn] << 8;
318 result_buf[j][0] += sample * vol_left;
319 result_buf[j][1] += sample * vol_right;
321 channel->posn_left += channel->sample_rate;
322 while( channel->posn_left > audio.output_rate ) {
323 channel->posn_left -= audio.output_rate;
324 channel->posn++;
326 if( channel->posn == channel->end ) {
327 if( channel->loop ) {
328 channel->posn = channel->loop_start;
329 channel->loop = LOOP_LOOPED;
330 } else {
331 audio_stop_channel(i);
332 j = num_samples;
333 break;
334 }
335 }
336 }
337 }
338 break;
339 case AUDIO_FMT_ADPCM:
340 for( j=0; j<num_samples; j++ ) {
341 sample = (int16_t)channel->adpcm_predict;
342 result_buf[j][0] += sample * vol_left;
343 result_buf[j][1] += sample * vol_right;
344 channel->posn_left += channel->sample_rate;
345 while( channel->posn_left > audio.output_rate ) {
346 channel->posn_left -= audio.output_rate;
347 channel->posn++;
348 if( channel->posn == channel->end ) {
349 if( channel->loop ) {
350 channel->posn = channel->loop_start;
351 channel->loop = LOOP_LOOPED;
352 channel->adpcm_predict = 0;
353 channel->adpcm_step = 0;
354 } else {
355 audio_stop_channel(i);
356 j = num_samples;
357 break;
358 }
359 }
360 uint8_t data = ((uint8_t *)(arm_mem + channel->start))[channel->posn>>1];
361 if( channel->posn&1 ) {
362 adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
363 } else {
364 adpcm_yamaha_decode_nibble( channel, data & 0x0F );
365 }
366 }
367 }
368 break;
369 default:
370 break;
371 }
372 }
373 }
375 /* Down-render to the final output format */
376 audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
377 if( buf->status == BUFFER_FULL ) {
378 buf = audio_next_write_buffer();
379 if( buf == NULL ) { // no available space
380 return;
381 }
382 }
384 switch( audio.output_format & AUDIO_FMT_SAMPLE_MASK ) {
385 case AUDIO_FMT_FLOAT: {
386 float scale = 1.0/SHRT_MAX;
387 float *data = (float *)&buf->data[buf->posn];
388 for( j=0; j<num_samples; j++ ) {
389 *data++ = scale * (result_buf[j][0] >> 6);
390 *data++ = scale * (result_buf[j][1] >> 6);
391 buf->posn += 8;
392 if( buf->posn == buf->length ) {
393 buf = audio_next_write_buffer();
394 if( buf == NULL ) {
395 break;
396 }
397 data = (float *)&buf->data[0];
398 }
399 }
400 break;
401 }
402 case AUDIO_FMT_16BIT: {
403 int16_t *data = (int16_t *)&buf->data[buf->posn];
404 for( j=0; j < num_samples; j++ ) {
405 *data++ = (int16_t)(result_buf[j][0] >> 6);
406 *data++ = (int16_t)(result_buf[j][1] >> 6);
407 buf->posn += 4;
408 if( buf->posn == buf->length ) {
409 buf = audio_next_write_buffer();
410 if( buf == NULL ) {
411 // All buffers are full
412 break;
413 }
414 data = (int16_t *)&buf->data[0];
415 }
416 }
417 break;
418 }
419 case AUDIO_FMT_8BIT: {
420 int8_t *data = (int8_t *)&buf->data[buf->posn];
421 for( j=0; j < num_samples; j++ ) {
422 *data++ = (int8_t)(result_buf[j][0] >> 16);
423 *data++ = (int8_t)(result_buf[j][1] >> 16);
424 buf->posn += 2;
425 if( buf->posn == buf->length ) {
426 buf = audio_next_write_buffer();
427 if( buf == NULL ) {
428 // All buffers are full
429 break;
430 }
431 buf = audio.output_buffers[audio.write_buffer];
432 data = (int8_t *)&buf->data[0];
433 }
434 }
435 break;
436 }
437 }
438 }
440 /********************** Internal AICA calls ***************************/
442 audio_channel_t audio_get_channel( int channel )
443 {
444 return &audio.channels[channel];
445 }
447 void audio_start_stop_channel( int channel, gboolean start )
448 {
449 if( audio.channels[channel].active ) {
450 if( !start ) {
451 audio_stop_channel(channel);
452 }
453 } else if( start ) {
454 audio_start_channel(channel);
455 }
456 }
458 void audio_stop_channel( int channel )
459 {
460 audio.channels[channel].active = FALSE;
461 }
464 void audio_start_channel( int channel )
465 {
466 audio.channels[channel].posn = 0;
467 audio.channels[channel].posn_left = 0;
468 audio.channels[channel].active = TRUE;
469 if( audio.channels[channel].sample_format == AUDIO_FMT_ADPCM ) {
470 audio.channels[channel].adpcm_step = 0;
471 audio.channels[channel].adpcm_predict = 0;
472 uint8_t data = ((uint8_t *)(arm_mem + audio.channels[channel].start))[0];
473 adpcm_yamaha_decode_nibble( &audio.channels[channel], data & 0x0F );
474 }
475 }
.