lxdream 0.9.1| filename | src/aica/audio.c |
| changeset | 697:479b8c213f61 |
| prev | 657:c4143facbfcb |
| next | 700:4650d0c7f6f9 |
| author | nkeynes |
| date | Sun Jun 22 04:01:27 2008 +0000 (14 years ago) |
| permissions | -rw-r--r-- |
| last change | Commit work-in-progress CoreAudio driver (along with various changes to the audio subsystem) |
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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 1000
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 audio_driver_t audio_init_driver( const char *preferred_driver )
103 {
104 audio_driver_t audio_driver = get_audio_driver_by_name(preferred_driver);
105 if( audio_driver == NULL ) {
106 ERROR( "Audio driver '%s' not found, aborting.", preferred_driver );
107 exit(2);
108 } else if( audio_set_driver( audio_driver ) == FALSE ) {
109 ERROR( "Failed to initialize audio driver '%s', using null driver",
110 audio_driver->name );
111 audio_set_driver( &audio_null_driver );
112 }
113 }
115 /**
116 * Set the output driver, sample rate and format. Also initializes the
117 * output buffers, flushing any current data and reallocating as
118 * necessary.
119 */
120 gboolean audio_set_driver( audio_driver_t driver )
121 {
122 uint32_t bytes_per_sample = 1;
123 uint32_t samples_per_buffer;
124 int i;
126 if( audio_driver == NULL || driver != NULL ) {
127 if( driver == NULL )
128 driver = &audio_null_driver;
129 if( driver != audio_driver ) {
130 if( !driver->init() )
131 return FALSE;
132 audio_driver = driver;
133 }
134 }
136 switch( driver->sample_format & AUDIO_FMT_SAMPLE_MASK ) {
137 case AUDIO_FMT_8BIT:
138 bytes_per_sample = 1;
139 break;
140 case AUDIO_FMT_16BIT:
141 bytes_per_sample = 2;
142 break;
143 case AUDIO_FMT_FLOAT:
144 bytes_per_sample = 4;
145 break;
146 }
148 if( driver->sample_format & AUDIO_FMT_STEREO )
149 bytes_per_sample <<= 1;
150 if( driver->sample_rate == audio.output_rate &&
151 bytes_per_sample == audio.output_sample_size )
152 return TRUE;
153 samples_per_buffer = (driver->sample_rate * MS_PER_BUFFER / 1000);
154 for( i=0; i<NUM_BUFFERS; i++ ) {
155 if( audio.output_buffers[i] != NULL )
156 free(audio.output_buffers[i]);
157 audio.output_buffers[i] = g_malloc0( sizeof(struct audio_buffer) + samples_per_buffer * bytes_per_sample );
158 audio.output_buffers[i]->length = samples_per_buffer * bytes_per_sample;
159 audio.output_buffers[i]->posn = 0;
160 audio.output_buffers[i]->status = BUFFER_EMPTY;
161 }
162 audio.output_format = driver->sample_format;
163 audio.output_rate = driver->sample_rate;
164 audio.output_sample_size = bytes_per_sample;
165 audio.write_buffer = 0;
166 audio.read_buffer = 0;
168 return TRUE;
169 }
171 /**
172 * Mark the current write buffer as full and prepare the next buffer for
173 * writing. Returns the next buffer to write to.
174 * If all buffers are full, returns NULL.
175 */
176 audio_buffer_t audio_next_write_buffer( )
177 {
178 audio_buffer_t result = NULL;
179 audio_buffer_t current = audio.output_buffers[audio.write_buffer];
180 current->status = BUFFER_FULL;
181 if( audio.read_buffer == audio.write_buffer &&
182 audio_driver->process_buffer( current ) ) {
183 audio_next_read_buffer();
184 }
185 int next_buffer = NEXT_BUFFER();
186 result = audio.output_buffers[next_buffer];
187 if( result->status == BUFFER_FULL )
188 return NULL;
189 else {
190 audio.write_buffer = next_buffer;
191 result->status = BUFFER_WRITING;
192 return result;
193 }
194 }
196 /**
197 * Mark the current read buffer as empty and return the next buffer for
198 * reading. If there is no next buffer yet, returns NULL.
199 */
200 audio_buffer_t audio_next_read_buffer( )
201 {
202 audio_buffer_t current = audio.output_buffers[audio.read_buffer];
203 if( current->status == BUFFER_FULL ) {
204 // Current read buffer has data, which we've just emptied
205 current->status = BUFFER_EMPTY;
206 current->posn = 0;
207 audio.read_buffer++;
208 if( audio.read_buffer == NUM_BUFFERS )
209 audio.read_buffer = 0;
211 current = audio.output_buffers[audio.read_buffer];
212 if( current->status == BUFFER_FULL ) {
213 current->posn = 0;
214 return current;
215 }
216 else return NULL;
217 } else {
218 return NULL;
219 }
221 }
223 /*************************** ADPCM ***********************************/
225 /**
226 * The following section borrows heavily from ffmpeg, which is
227 * copyright (c) 2001-2003 by the fine folks at the ffmpeg project,
228 * distributed under the GPL version 2 or later.
229 */
231 #define CLAMP_TO_SHORT(value) \
232 if (value > 32767) \
233 value = 32767; \
234 else if (value < -32768) \
235 value = -32768; \
237 static const int yamaha_indexscale[] = {
238 230, 230, 230, 230, 307, 409, 512, 614,
239 230, 230, 230, 230, 307, 409, 512, 614
240 };
242 static const int yamaha_difflookup[] = {
243 1, 3, 5, 7, 9, 11, 13, 15,
244 -1, -3, -5, -7, -9, -11, -13, -15
245 };
247 static inline short adpcm_yamaha_decode_nibble( audio_channel_t c,
248 unsigned char nibble )
249 {
250 if( c->adpcm_step == 0 ) {
251 c->adpcm_predict = 0;
252 c->adpcm_step = 127;
253 }
255 c->adpcm_predict += (c->adpcm_step * yamaha_difflookup[nibble]) >> 3;
256 CLAMP_TO_SHORT(c->adpcm_predict);
257 c->adpcm_step = (c->adpcm_step * yamaha_indexscale[nibble]) >> 8;
258 c->adpcm_step = CLAMP(c->adpcm_step, 127, 24567);
259 return c->adpcm_predict;
260 }
262 /*************************** Sample mixer *****************************/
264 /**
265 * Mix a single output sample.
266 */
267 void audio_mix_samples( int num_samples )
268 {
269 int i, j;
270 int32_t result_buf[num_samples][2];
272 memset( &result_buf, 0, sizeof(result_buf) );
274 for( i=0; i < AUDIO_CHANNEL_COUNT; i++ ) {
275 audio_channel_t channel = &audio.channels[i];
276 if( channel->active ) {
277 int32_t sample;
278 int vol_left = (channel->vol * (32 - channel->pan)) >> 5;
279 int vol_right = (channel->vol * (channel->pan + 1)) >> 5;
280 switch( channel->sample_format ) {
281 case AUDIO_FMT_16BIT:
282 for( j=0; j<num_samples; j++ ) {
283 sample = ((int16_t *)(arm_mem + channel->start))[channel->posn];
284 result_buf[j][0] += sample * vol_left;
285 result_buf[j][1] += sample * vol_right;
287 channel->posn_left += channel->sample_rate;
288 while( channel->posn_left > audio.output_rate ) {
289 channel->posn_left -= audio.output_rate;
290 channel->posn++;
292 if( channel->posn == channel->end ) {
293 if( channel->loop ) {
294 channel->posn = channel->loop_start;
295 channel->loop = LOOP_LOOPED;
296 } else {
297 audio_stop_channel(i);
298 j = num_samples;
299 break;
300 }
301 }
302 }
303 }
304 break;
305 case AUDIO_FMT_8BIT:
306 for( j=0; j<num_samples; j++ ) {
307 sample = ((int8_t *)(arm_mem + channel->start))[channel->posn] << 8;
308 result_buf[j][0] += sample * vol_left;
309 result_buf[j][1] += sample * vol_right;
311 channel->posn_left += channel->sample_rate;
312 while( channel->posn_left > audio.output_rate ) {
313 channel->posn_left -= audio.output_rate;
314 channel->posn++;
316 if( channel->posn == channel->end ) {
317 if( channel->loop ) {
318 channel->posn = channel->loop_start;
319 channel->loop = LOOP_LOOPED;
320 } else {
321 audio_stop_channel(i);
322 j = num_samples;
323 break;
324 }
325 }
326 }
327 }
328 break;
329 case AUDIO_FMT_ADPCM:
330 for( j=0; j<num_samples; j++ ) {
331 sample = (int16_t)channel->adpcm_predict;
332 result_buf[j][0] += sample * vol_left;
333 result_buf[j][1] += sample * vol_right;
334 channel->posn_left += channel->sample_rate;
335 while( channel->posn_left > audio.output_rate ) {
336 channel->posn_left -= audio.output_rate;
337 channel->posn++;
338 if( channel->posn == channel->end ) {
339 if( channel->loop ) {
340 channel->posn = channel->loop_start;
341 channel->loop = LOOP_LOOPED;
342 channel->adpcm_predict = 0;
343 channel->adpcm_step = 0;
344 } else {
345 audio_stop_channel(i);
346 j = num_samples;
347 break;
348 }
349 }
350 uint8_t data = ((uint8_t *)(arm_mem + channel->start))[channel->posn>>1];
351 if( channel->posn&1 ) {
352 adpcm_yamaha_decode_nibble( channel, (data >> 4) & 0x0F );
353 } else {
354 adpcm_yamaha_decode_nibble( channel, data & 0x0F );
355 }
356 }
357 }
358 break;
359 default:
360 break;
361 }
362 }
363 }
365 /* Down-render to the final output format */
366 audio_buffer_t buf = audio.output_buffers[audio.write_buffer];
367 if( buf->status == BUFFER_FULL ) {
368 buf = audio_next_write_buffer();
369 if( buf == NULL ) { // no available space
370 return;
371 }
372 }
374 switch( audio.output_format & AUDIO_FMT_SAMPLE_MASK ) {
375 case AUDIO_FMT_FLOAT: {
376 float scale = 1.0/SHRT_MAX;
377 float *data = (float *)&buf->data[buf->posn];
378 for( j=0; j<num_samples; j++ ) {
379 *data++ = scale * (result_buf[j][0] >> 6);
380 *data++ = scale * (result_buf[j][1] >> 6);
381 buf->posn += 8;
382 if( buf->posn == buf->length ) {
383 buf = audio_next_write_buffer();
384 if( buf == NULL ) {
385 break;
386 }
387 data = (float *)&buf->data[0];
388 }
389 }
390 break;
391 }
392 case AUDIO_FMT_16BIT: {
393 int16_t *data = (int16_t *)&buf->data[buf->posn];
394 for( j=0; j < num_samples; j++ ) {
395 *data++ = (int16_t)(result_buf[j][0] >> 6);
396 *data++ = (int16_t)(result_buf[j][1] >> 6);
397 buf->posn += 4;
398 if( buf->posn == buf->length ) {
399 buf = audio_next_write_buffer();
400 if( buf == NULL ) {
401 // All buffers are full
402 break;
403 }
404 data = (int16_t *)&buf->data[0];
405 }
406 }
407 break;
408 }
409 case AUDIO_FMT_8BIT: {
410 int8_t *data = (uint8_t *)&buf->data[buf->posn];
411 for( j=0; j < num_samples; j++ ) {
412 *data++ = (int8_t)(result_buf[j][0] >> 16);
413 *data++ = (int8_t)(result_buf[j][1] >> 16);
414 buf->posn += 2;
415 if( buf->posn == buf->length ) {
416 buf = audio_next_write_buffer();
417 if( buf == NULL ) {
418 // All buffers are full
419 break;
420 }
421 buf = audio.output_buffers[audio.write_buffer];
422 data = (uint8_t *)&buf->data[0];
423 }
424 }
425 break;
426 }
427 }
428 }
430 /********************** Internal AICA calls ***************************/
432 audio_channel_t audio_get_channel( int channel )
433 {
434 return &audio.channels[channel];
435 }
437 void audio_start_stop_channel( int channel, gboolean start )
438 {
439 if( audio.channels[channel].active ) {
440 if( !start ) {
441 audio_stop_channel(channel);
442 }
443 } else if( start ) {
444 audio_start_channel(channel);
445 }
446 }
448 void audio_stop_channel( int channel )
449 {
450 audio.channels[channel].active = FALSE;
451 }
454 void audio_start_channel( int channel )
455 {
456 audio.channels[channel].posn = 0;
457 audio.channels[channel].posn_left = 0;
458 audio.channels[channel].active = TRUE;
459 if( audio.channels[channel].sample_format == AUDIO_FMT_ADPCM ) {
460 audio.channels[channel].adpcm_step = 0;
461 audio.channels[channel].adpcm_predict = 0;
462 uint8_t data = ((uint8_t *)(arm_mem + audio.channels[channel].start))[0];
463 adpcm_yamaha_decode_nibble( &audio.channels[channel], data & 0x0F );
464 }
465 }
.