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root/cebix/BasiliskII/src/Unix/audio_oss_esd.cpp
Revision: 1.12
Committed: 2002-01-15T14:58:37Z (22 years, 10 months ago) by cebix
Branch: MAIN
CVS Tags: snapshot-15012002
Changes since 1.11: +1 -1 lines
Log Message:
- documentation updates
- 2001 -> 2002
- version 0.9 -> 1.0

File Contents

# Content
1 /*
2 * audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
3 *
4 * Basilisk II (C) 1997-2002 Christian Bauer
5 *
6 * This program is free software; you can redistribute it and/or modify
7 * it under the terms of the GNU General Public License as published by
8 * the Free Software Foundation; either version 2 of the License, or
9 * (at your option) any later version.
10 *
11 * This program is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 * GNU General Public License for more details.
15 *
16 * You should have received a copy of the GNU General Public License
17 * along with this program; if not, write to the Free Software
18 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
19 */
20
21 #include "sysdeps.h"
22
23 #include <sys/ioctl.h>
24 #include <unistd.h>
25 #include <errno.h>
26 #include <pthread.h>
27 #include <semaphore.h>
28
29 #ifdef __linux__
30 #include <linux/soundcard.h>
31 #endif
32
33 #ifdef __FreeBSD__
34 #include <machine/soundcard.h>
35 #endif
36
37 #include "cpu_emulation.h"
38 #include "main.h"
39 #include "prefs.h"
40 #include "user_strings.h"
41 #include "audio.h"
42 #include "audio_defs.h"
43
44 #ifdef ENABLE_ESD
45 #include <esd.h>
46 #endif
47
48 #define DEBUG 0
49 #include "debug.h"
50
51
52 // The currently selected audio parameters (indices in audio_sample_rates[] etc. vectors)
53 static int audio_sample_rate_index = 0;
54 static int audio_sample_size_index = 0;
55 static int audio_channel_count_index = 0;
56
57 // Constants
58 #define DSP_NAME "/dev/dsp"
59
60 // Global variables
61 static int audio_fd = -1; // fd of /dev/dsp or ESD
62 static int mixer_fd = -1; // fd of /dev/mixer
63 static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
64 static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
65 static int sound_buffer_size; // Size of sound buffer in bytes
66 static bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
67 static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
68 static pthread_t stream_thread; // Audio streaming thread
69 static pthread_attr_t stream_thread_attr; // Streaming thread attributes
70 static bool stream_thread_active = false; // Flag: streaming thread installed
71 static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
72
73 // Prototypes
74 static void *stream_func(void *arg);
75
76
77 /*
78 * Initialization
79 */
80
81 // Set AudioStatus to reflect current audio stream format
82 static void set_audio_status_format(void)
83 {
84 AudioStatus.sample_rate = audio_sample_rates[audio_sample_rate_index];
85 AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
86 AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
87 }
88
89 // Init using /dev/dsp, returns false on error
90 static bool open_dsp(void)
91 {
92 // Open /dev/dsp
93 audio_fd = open(DSP_NAME, O_WRONLY);
94 if (audio_fd < 0) {
95 fprintf(stderr, "WARNING: Cannot open %s (%s)\n", DSP_NAME, strerror(errno));
96 return false;
97 }
98
99 printf("Using " DSP_NAME " audio output\n");
100
101 // Get supported sample formats
102 if (audio_sample_sizes.empty()) {
103 unsigned long format;
104 ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
105 if (format & AFMT_U8)
106 audio_sample_sizes.push_back(8);
107 if (format & (AFMT_S16_BE | AFMT_S16_LE))
108 audio_sample_sizes.push_back(16);
109
110 int stereo = 0;
111 if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 0)
112 audio_channel_counts.push_back(1);
113 stereo = 1;
114 if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 1)
115 audio_channel_counts.push_back(2);
116
117 if (audio_sample_sizes.empty() || audio_channel_counts.empty()) {
118 WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
119 close(audio_fd);
120 audio_fd = -1;
121 return false;
122 }
123
124 audio_sample_rates.push_back(11025 << 16);
125 audio_sample_rates.push_back(22050 << 16);
126 int rate = 44100;
127 ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
128 if (rate > 22050)
129 audio_sample_rates.push_back(rate << 16);
130
131 // Default to highest supported values
132 audio_sample_rate_index = audio_sample_rates.size() - 1;
133 audio_sample_size_index = audio_sample_sizes.size() - 1;
134 audio_channel_count_index = audio_channel_counts.size() - 1;
135 }
136
137 // Set DSP parameters
138 unsigned long format;
139 if (audio_sample_sizes[audio_sample_size_index] == 8) {
140 format = AFMT_U8;
141 little_endian = false;
142 silence_byte = 0x80;
143 } else {
144 unsigned long sup_format;
145 ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &sup_format);
146 if (sup_format & AFMT_S16_BE) {
147 little_endian = false;
148 format = AFMT_S16_BE;
149 } else {
150 little_endian = true;
151 format = AFMT_S16_LE;
152 }
153 silence_byte = 0;
154 }
155 ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
156 int frag = 0x0004000c; // Block size: 4096 frames
157 ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
158 int stereo = (audio_channel_counts[audio_channel_count_index] == 2);
159 ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
160 int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
161 ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
162
163 // Get sound buffer size
164 ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
165 D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
166 return true;
167 }
168
169 // Init using ESD, returns false on error
170 static bool open_esd(void)
171 {
172 #ifdef ENABLE_ESD
173 int rate;
174 esd_format_t format = ESD_STREAM | ESD_PLAY;
175
176 if (audio_sample_sizes.empty()) {
177
178 // Default values
179 rate = 44100;
180 format |= (ESD_BITS16 | ESD_STEREO);
181
182 } else {
183
184 rate = audio_sample_rates[audio_sample_rate_index] >> 16;
185 if (audio_sample_sizes[audio_sample_size_index] == 8)
186 format |= ESD_BITS8;
187 else
188 format |= ESD_BITS16;
189 if (audio_channel_counts[audio_channel_count_index] == 1)
190 format |= ESD_MONO;
191 else
192 format |= ESD_STEREO;
193 }
194
195 #if WORDS_BIGENDIAN
196 little_endian = false;
197 #else
198 little_endian = true;
199 #endif
200 silence_byte = 0; // Is this correct for 8-bit mode?
201
202 // Open connection to ESD server
203 audio_fd = esd_play_stream(format, rate, NULL, NULL);
204 if (audio_fd < 0) {
205 fprintf(stderr, "WARNING: Cannot open ESD connection\n");
206 return false;
207 }
208
209 printf("Using ESD audio output\n");
210
211 // ESD supports a variety of twisted little audio formats, all different
212 if (audio_sample_sizes.empty()) {
213
214 // The reason we do this here is that we don't want to add sample
215 // rates etc. unless the ESD server connection could be opened
216 // (if ESD fails, /dev/dsp might be tried next)
217 audio_sample_rates.push_back(11025 << 16);
218 audio_sample_rates.push_back(22050 << 16);
219 audio_sample_rates.push_back(44100 << 16);
220 audio_sample_sizes.push_back(8);
221 audio_sample_sizes.push_back(16);
222 audio_channel_counts.push_back(1);
223 audio_channel_counts.push_back(2);
224
225 // Default to highest supported values
226 audio_sample_rate_index = audio_sample_rates.size() - 1;
227 audio_sample_size_index = audio_sample_sizes.size() - 1;
228 audio_channel_count_index = audio_channel_counts.size() - 1;
229 }
230
231 // Sound buffer size = 4096 frames
232 audio_frames_per_block = 4096;
233 return true;
234 #else
235 // ESD is not enabled, shut up the compiler
236 return false;
237 #endif
238 }
239
240 static bool open_audio(void)
241 {
242 #ifdef ENABLE_ESD
243 // If ESPEAKER is set, the user probably wants to use ESD, so try that first
244 if (getenv("ESPEAKER"))
245 if (open_esd())
246 goto dev_opened;
247 #endif
248
249 // Try to open /dev/dsp
250 if (open_dsp())
251 goto dev_opened;
252
253 #ifdef ENABLE_ESD
254 // Hm, /dev/dsp failed so we try ESD again if ESPEAKER wasn't set
255 if (!getenv("ESPEAKER"))
256 if (open_esd())
257 goto dev_opened;
258 #endif
259
260 // No audio device succeeded
261 WarningAlert(GetString(STR_NO_AUDIO_WARN));
262 return false;
263
264 // Device opened, set AudioStatus
265 dev_opened:
266 sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) * audio_channel_counts[audio_channel_count_index] * audio_frames_per_block;
267 set_audio_status_format();
268
269 // Start streaming thread
270 pthread_attr_init(&stream_thread_attr);
271 #if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
272 if (geteuid() == 0) {
273 pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
274 pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
275 struct sched_param fifo_param;
276 fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
277 pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
278 }
279 #endif
280 stream_thread_cancel = false;
281 stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
282
283 // Everything went fine
284 audio_open = true;
285 return true;
286 }
287
288 void AudioInit(void)
289 {
290 // Init audio status (reasonable defaults) and feature flags
291 AudioStatus.sample_rate = 44100 << 16;
292 AudioStatus.sample_size = 16;
293 AudioStatus.channels = 2;
294 AudioStatus.mixer = 0;
295 AudioStatus.num_sources = 0;
296 audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;
297
298 // Sound disabled in prefs? Then do nothing
299 if (PrefsFindBool("nosound"))
300 return;
301
302 // Init semaphore
303 if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
304 return;
305 sem_inited = true;
306
307 // Try to open /dev/mixer
308 mixer_fd = open("/dev/mixer", O_RDWR);
309 if (mixer_fd < 0)
310 printf("WARNING: Cannot open /dev/mixer (%s)", strerror(errno));
311
312 // Open and initialize audio device
313 open_audio();
314 }
315
316
317 /*
318 * Deinitialization
319 */
320
321 static void close_audio(void)
322 {
323 // Stop stream and delete semaphore
324 if (stream_thread_active) {
325 stream_thread_cancel = true;
326 #ifdef HAVE_PTHREAD_CANCEL
327 pthread_cancel(stream_thread);
328 #endif
329 pthread_join(stream_thread, NULL);
330 stream_thread_active = false;
331 }
332
333 // Close /dev/dsp or ESD socket
334 if (audio_fd >= 0) {
335 close(audio_fd);
336 audio_fd = -1;
337 }
338
339 audio_open = false;
340 }
341
342 void AudioExit(void)
343 {
344 // Close audio device
345 close_audio();
346
347 // Delete semaphore
348 if (sem_inited) {
349 sem_destroy(&audio_irq_done_sem);
350 sem_inited = false;
351 }
352
353 // Close /dev/mixer
354 if (mixer_fd >= 0) {
355 close(mixer_fd);
356 mixer_fd = -1;
357 }
358 }
359
360
361 /*
362 * First source added, start audio stream
363 */
364
365 void audio_enter_stream()
366 {
367 // Streaming thread is always running to avoid clicking noises
368 }
369
370
371 /*
372 * Last source removed, stop audio stream
373 */
374
375 void audio_exit_stream()
376 {
377 // Streaming thread is always running to avoid clicking noises
378 }
379
380
381 /*
382 * Streaming function
383 */
384
385 static void *stream_func(void *arg)
386 {
387 int16 *silent_buffer = new int16[sound_buffer_size / 2];
388 int16 *last_buffer = new int16[sound_buffer_size / 2];
389 memset(silent_buffer, silence_byte, sound_buffer_size);
390
391 while (!stream_thread_cancel) {
392 if (AudioStatus.num_sources) {
393
394 // Trigger audio interrupt to get new buffer
395 D(bug("stream: triggering irq\n"));
396 SetInterruptFlag(INTFLAG_AUDIO);
397 TriggerInterrupt();
398 D(bug("stream: waiting for ack\n"));
399 sem_wait(&audio_irq_done_sem);
400 D(bug("stream: ack received\n"));
401
402 // Get size of audio data
403 uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
404 if (apple_stream_info) {
405 int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
406 D(bug("stream: work_size %d\n", work_size));
407 if (work_size > sound_buffer_size)
408 work_size = sound_buffer_size;
409 if (work_size == 0)
410 goto silence;
411
412 // Send data to DSP
413 if (work_size == sound_buffer_size && !little_endian)
414 write(audio_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
415 else {
416 // Last buffer or little-endian DSP
417 if (little_endian) {
418 int16 *p = (int16 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
419 for (int i=0; i<work_size/2; i++)
420 last_buffer[i] = ntohs(p[i]);
421 } else
422 Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
423 memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
424 write(audio_fd, last_buffer, sound_buffer_size);
425 }
426 D(bug("stream: data written\n"));
427 } else
428 goto silence;
429
430 } else {
431
432 // Audio not active, play silence
433 silence: write(audio_fd, silent_buffer, sound_buffer_size);
434 }
435 }
436 delete[] silent_buffer;
437 delete[] last_buffer;
438 return NULL;
439 }
440
441
442 /*
443 * MacOS audio interrupt, read next data block
444 */
445
446 void AudioInterrupt(void)
447 {
448 D(bug("AudioInterrupt\n"));
449
450 // Get data from apple mixer
451 if (AudioStatus.mixer) {
452 M68kRegisters r;
453 r.a[0] = audio_data + adatStreamInfo;
454 r.a[1] = AudioStatus.mixer;
455 Execute68k(audio_data + adatGetSourceData, &r);
456 D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
457 } else
458 WriteMacInt32(audio_data + adatStreamInfo, 0);
459
460 // Signal stream function
461 sem_post(&audio_irq_done_sem);
462 D(bug("AudioInterrupt done\n"));
463 }
464
465
466 /*
467 * Set sampling parameters
468 * "index" is an index into the audio_sample_rates[] etc. vectors
469 * It is guaranteed that AudioStatus.num_sources == 0
470 */
471
472 bool audio_set_sample_rate(int index)
473 {
474 close_audio();
475 audio_sample_rate_index = index;
476 return open_audio();
477 }
478
479 bool audio_set_sample_size(int index)
480 {
481 close_audio();
482 audio_sample_size_index = index;
483 return open_audio();
484 }
485
486 bool audio_set_channels(int index)
487 {
488 close_audio();
489 audio_channel_count_index = index;
490 return open_audio();
491 }
492
493
494 /*
495 * Get/set volume controls (volume values received/returned have the left channel
496 * volume in the upper 16 bits and the right channel volume in the lower 16 bits;
497 * both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
498 */
499
500 bool audio_get_main_mute(void)
501 {
502 return false;
503 }
504
505 uint32 audio_get_main_volume(void)
506 {
507 if (mixer_fd >= 0) {
508 int vol;
509 if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
510 int left = vol >> 8;
511 int right = vol & 0xff;
512 return ((left * 256 / 100) << 16) | (right * 256 / 100);
513 }
514 }
515 return 0x01000100;
516 }
517
518 bool audio_get_speaker_mute(void)
519 {
520 return false;
521 }
522
523 uint32 audio_get_speaker_volume(void)
524 {
525 if (mixer_fd >= 0) {
526 int vol;
527 if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
528 int left = vol >> 8;
529 int right = vol & 0xff;
530 return ((left * 256 / 100) << 16) | (right * 256 / 100);
531 }
532 }
533 return 0x01000100;
534 }
535
536 void audio_set_main_mute(bool mute)
537 {
538 }
539
540 void audio_set_main_volume(uint32 vol)
541 {
542 if (mixer_fd >= 0) {
543 int left = vol >> 16;
544 int right = vol & 0xffff;
545 int p = ((left * 100 / 256) << 8) | (right * 100 / 256);
546 ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
547 }
548 }
549
550 void audio_set_speaker_mute(bool mute)
551 {
552 }
553
554 void audio_set_speaker_volume(uint32 vol)
555 {
556 if (mixer_fd >= 0) {
557 int left = vol >> 16;
558 int right = vol & 0xffff;
559 int p = ((left * 100 / 256) << 8) | (right * 100 / 256);
560 ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
561 }
562 }