1 /* 2 * QEMU ALSA audio driver 3 * 4 * Copyright (c) 2008-2010 The Android Open Source Project 5 * Copyright (c) 2005 Vassili Karpov (malc) 6 * 7 * Permission is hereby granted, free of charge, to any person obtaining a copy 8 * of this software and associated documentation files (the "Software"), to deal 9 * in the Software without restriction, including without limitation the rights 10 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell 11 * copies of the Software, and to permit persons to whom the Software is 12 * furnished to do so, subject to the following conditions: 13 * 14 * The above copyright notice and this permission notice shall be included in 15 * all copies or substantial portions of the Software. 16 * 17 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR 18 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, 19 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL 20 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER 21 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, 22 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN 23 * THE SOFTWARE. 24 */ 25 #include <alsa/asoundlib.h> 26 #include "qemu-common.h" 27 #include "qemu-char.h" 28 #include "audio.h" 29 30 #if QEMU_GNUC_PREREQ(4, 3) 31 #pragma GCC diagnostic ignored "-Waddress" 32 #endif 33 34 #define AUDIO_CAP "alsa" 35 #include "audio_int.h" 36 #include <dlfcn.h> 37 #include <pthread.h> 38 #include "qemu_debug.h" 39 40 #define DEBUG 1 41 42 #if DEBUG 43 # include <stdio.h> 44 # define D(...) VERBOSE_PRINT(audio,__VA_ARGS__) 45 # define D_ACTIVE VERBOSE_CHECK(audio) 46 # define O(...) VERBOSE_PRINT(audioout,__VA_ARGS__) 47 # define I(...) VERBOSE_PRINT(audioin,__VA_ARGS__) 48 #else 49 # define D(...) ((void)0) 50 # define D_ACTIVE 0 51 # define O(...) ((void)0) 52 # define I(...) ((void)0) 53 #endif 54 55 #define DYNLINK_FUNCTIONS \ 56 DYNLINK_FUNC(size_t,snd_pcm_sw_params_sizeof,(void)) \ 57 DYNLINK_FUNC(int,snd_pcm_hw_params_current,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)) \ 58 DYNLINK_FUNC(int,snd_pcm_sw_params_set_start_threshold,(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val)) \ 59 DYNLINK_FUNC(int,snd_pcm_sw_params,(snd_pcm_t *pcm, snd_pcm_sw_params_t *params)) \ 60 DYNLINK_FUNC(int,snd_pcm_sw_params_current,(snd_pcm_t *pcm, snd_pcm_sw_params_t *params)) \ 61 DYNLINK_FUNC(size_t,snd_pcm_hw_params_sizeof,(void)) \ 62 DYNLINK_FUNC(int,snd_pcm_hw_params_any,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)) \ 63 DYNLINK_FUNC(int,snd_pcm_hw_params_set_access,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t _access)) \ 64 DYNLINK_FUNC(int,snd_pcm_hw_params_get_format,(const snd_pcm_hw_params_t *params, snd_pcm_format_t *val)) \ 65 DYNLINK_FUNC(int,snd_pcm_hw_params_set_format,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val)) \ 66 DYNLINK_FUNC(int,snd_pcm_hw_params_set_rate_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir)) \ 67 DYNLINK_FUNC(int,snd_pcm_hw_params_set_channels_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val)) \ 68 DYNLINK_FUNC(int,snd_pcm_hw_params_set_buffer_time_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir)) \ 69 DYNLINK_FUNC(int,snd_pcm_hw_params,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params)) \ 70 DYNLINK_FUNC(int,snd_pcm_hw_params_get_buffer_size,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val)) \ 71 DYNLINK_FUNC(int,snd_pcm_prepare,(snd_pcm_t *pcm)) \ 72 DYNLINK_FUNC(int,snd_pcm_hw_params_get_period_size,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir)) \ 73 DYNLINK_FUNC(int,snd_pcm_hw_params_get_period_size_min,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir)) \ 74 DYNLINK_FUNC(int,snd_pcm_hw_params_set_period_size,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int dir)) \ 75 DYNLINK_FUNC(int,snd_pcm_hw_params_get_buffer_size_min,(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val)) \ 76 DYNLINK_FUNC(int,snd_pcm_hw_params_set_buffer_size,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val)) \ 77 DYNLINK_FUNC(int,snd_pcm_hw_params_set_period_time_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir)) \ 78 DYNLINK_FUNC(snd_pcm_sframes_t,snd_pcm_avail_update,(snd_pcm_t *pcm)) \ 79 DYNLINK_FUNC(int,snd_pcm_drop,(snd_pcm_t *pcm)) \ 80 DYNLINK_FUNC(snd_pcm_sframes_t,snd_pcm_writei,(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size)) \ 81 DYNLINK_FUNC(snd_pcm_sframes_t,snd_pcm_readi,(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size)) \ 82 DYNLINK_FUNC(snd_pcm_state_t,snd_pcm_state,(snd_pcm_t *pcm)) \ 83 DYNLINK_FUNC(const char*,snd_strerror,(int errnum)) \ 84 DYNLINK_FUNC(int,snd_pcm_open,(snd_pcm_t **pcm, const char *name,snd_pcm_stream_t stream, int mode)) \ 85 DYNLINK_FUNC(int,snd_pcm_close,(snd_pcm_t *pcm)) \ 86 DYNLINK_FUNC(int,snd_pcm_hw_params_set_buffer_size_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val)) \ 87 DYNLINK_FUNC(int,snd_pcm_hw_params_set_period_size_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val, int *dir)) \ 88 DYNLINK_FUNC(int,snd_pcm_hw_params_get_format,(const snd_pcm_hw_params_t *params, snd_pcm_format_t *val)) \ 89 DYNLINK_FUNC(int,snd_pcm_resume,(snd_pcm_t *pcm)) \ 90 DYNLINK_FUNC(int,snd_pcm_poll_descriptors_revents,(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents)) \ 91 DYNLINK_FUNC(int,snd_pcm_poll_descriptors_count,(snd_pcm_t *pcm)) \ 92 DYNLINK_FUNC(int,snd_pcm_poll_descriptors,(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space)) \ 93 94 #define DYNLINK_FUNCTIONS_INIT \ 95 alsa_dynlink_init 96 97 #include "dynlink.h" 98 99 /* these are inlined functions in the original headers */ 100 #define FF_snd_pcm_hw_params_alloca(ptr) \ 101 do { *ptr = (snd_pcm_hw_params_t *) alloca(FF(snd_pcm_hw_params_sizeof)()); memset(*ptr, 0, FF(snd_pcm_hw_params_sizeof)()); } while (0) 102 103 #define FF_snd_pcm_sw_params_alloca(ptr) \ 104 do { *ptr = (snd_pcm_sw_params_t *) alloca(FF(snd_pcm_sw_params_sizeof)()); memset(*ptr, 0, FF(snd_pcm_sw_params_sizeof)()); } while (0) 105 106 static void* alsa_lib; 107 108 struct pollhlp { 109 snd_pcm_t *handle; 110 struct pollfd *pfds; 111 int count; 112 int mask; 113 }; 114 115 typedef struct ALSAVoiceOut { 116 HWVoiceOut hw; 117 int wpos; 118 int pending; 119 void *pcm_buf; 120 snd_pcm_t *handle; 121 struct pollhlp pollhlp; 122 } ALSAVoiceOut; 123 124 typedef struct ALSAVoiceIn { 125 HWVoiceIn hw; 126 snd_pcm_t *handle; 127 void *pcm_buf; 128 struct pollhlp pollhlp; 129 } ALSAVoiceIn; 130 131 static struct { 132 int size_in_usec_in; 133 int size_in_usec_out; 134 const char *pcm_name_in; 135 const char *pcm_name_out; 136 unsigned int buffer_size_in; 137 unsigned int period_size_in; 138 unsigned int buffer_size_out; 139 unsigned int period_size_out; 140 unsigned int threshold; 141 142 int buffer_size_in_overridden; 143 int period_size_in_overridden; 144 145 int buffer_size_out_overridden; 146 int period_size_out_overridden; 147 int verbose; 148 } conf = { 149 .buffer_size_out = 4096, 150 .period_size_out = 1024, 151 .pcm_name_out = "default", 152 .pcm_name_in = "default", 153 }; 154 155 struct alsa_params_req { 156 int freq; 157 snd_pcm_format_t fmt; 158 int nchannels; 159 int size_in_usec; 160 int override_mask; 161 unsigned int buffer_size; 162 unsigned int period_size; 163 }; 164 165 struct alsa_params_obt { 166 int freq; 167 audfmt_e fmt; 168 int endianness; 169 int nchannels; 170 snd_pcm_uframes_t samples; 171 }; 172 173 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) 174 { 175 va_list ap; 176 177 va_start (ap, fmt); 178 AUD_vlog (AUDIO_CAP, fmt, ap); 179 va_end (ap); 180 181 AUD_log (AUDIO_CAP, "Reason: %s\n", FF(snd_strerror) (err)); 182 } 183 184 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( 185 int err, 186 const char *typ, 187 const char *fmt, 188 ... 189 ) 190 { 191 va_list ap; 192 193 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ); 194 195 va_start (ap, fmt); 196 AUD_vlog (AUDIO_CAP, fmt, ap); 197 va_end (ap); 198 199 AUD_log (AUDIO_CAP, "Reason: %s\n", FF(snd_strerror) (err)); 200 } 201 202 static void alsa_fini_poll (struct pollhlp *hlp) 203 { 204 int i; 205 struct pollfd *pfds = hlp->pfds; 206 207 if (pfds) { 208 for (i = 0; i < hlp->count; ++i) { 209 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); 210 } 211 qemu_free (pfds); 212 } 213 hlp->pfds = NULL; 214 hlp->count = 0; 215 hlp->handle = NULL; 216 } 217 218 static void alsa_anal_close1 (snd_pcm_t **handlep) 219 { 220 int err = FF(snd_pcm_close) (*handlep); 221 if (err) { 222 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep); 223 } 224 *handlep = NULL; 225 } 226 227 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp) 228 { 229 alsa_fini_poll (hlp); 230 alsa_anal_close1 (handlep); 231 } 232 233 static int alsa_recover (snd_pcm_t *handle) 234 { 235 int err = FF(snd_pcm_prepare) (handle); 236 if (err < 0) { 237 alsa_logerr (err, "Failed to prepare handle %p\n", handle); 238 return -1; 239 } 240 return 0; 241 } 242 243 static int alsa_resume (snd_pcm_t *handle) 244 { 245 int err = FF(snd_pcm_resume) (handle); 246 if (err < 0) { 247 alsa_logerr (err, "Failed to resume handle %p\n", handle); 248 return -1; 249 } 250 return 0; 251 } 252 253 static void alsa_poll_handler (void *opaque) 254 { 255 int err, count; 256 snd_pcm_state_t state; 257 struct pollhlp *hlp = opaque; 258 unsigned short revents; 259 260 count = poll (hlp->pfds, hlp->count, 0); 261 if (count < 0) { 262 dolog ("alsa_poll_handler: poll %s\n", strerror (errno)); 263 return; 264 } 265 266 if (!count) { 267 return; 268 } 269 270 /* XXX: ALSA example uses initial count, not the one returned by 271 poll, correct? */ 272 err = FF(snd_pcm_poll_descriptors_revents) (hlp->handle, hlp->pfds, 273 hlp->count, &revents); 274 if (err < 0) { 275 alsa_logerr (err, "snd_pcm_poll_descriptors_revents"); 276 return; 277 } 278 279 if (!(revents & hlp->mask)) { 280 if (conf.verbose) { 281 dolog ("revents = %d\n", revents); 282 } 283 return; 284 } 285 286 state = FF(snd_pcm_state) (hlp->handle); 287 switch (state) { 288 case SND_PCM_STATE_SETUP: 289 alsa_recover (hlp->handle); 290 break; 291 292 case SND_PCM_STATE_XRUN: 293 alsa_recover (hlp->handle); 294 break; 295 296 case SND_PCM_STATE_SUSPENDED: 297 alsa_resume (hlp->handle); 298 break; 299 300 case SND_PCM_STATE_PREPARED: 301 audio_run ("alsa run (prepared)"); 302 break; 303 304 case SND_PCM_STATE_RUNNING: 305 audio_run ("alsa run (running)"); 306 break; 307 308 default: 309 dolog ("Unexpected state %d\n", state); 310 } 311 } 312 313 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) 314 { 315 int i, count, err; 316 struct pollfd *pfds; 317 318 count = FF(snd_pcm_poll_descriptors_count) (handle); 319 if (count <= 0) { 320 dolog ("Could not initialize poll mode\n" 321 "Invalid number of poll descriptors %d\n", count); 322 return -1; 323 } 324 325 pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); 326 if (!pfds) { 327 dolog ("Could not initialize poll mode\n"); 328 return -1; 329 } 330 331 err = FF(snd_pcm_poll_descriptors) (handle, pfds, count); 332 if (err < 0) { 333 alsa_logerr (err, "Could not initialize poll mode\n" 334 "Could not obtain poll descriptors\n"); 335 qemu_free (pfds); 336 return -1; 337 } 338 339 for (i = 0; i < count; ++i) { 340 if (pfds[i].events & POLLIN) { 341 err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, 342 NULL, hlp); 343 } 344 if (pfds[i].events & POLLOUT) { 345 if (conf.verbose) { 346 dolog ("POLLOUT %d %d\n", i, pfds[i].fd); 347 } 348 err = qemu_set_fd_handler (pfds[i].fd, NULL, 349 alsa_poll_handler, hlp); 350 } 351 if (conf.verbose) { 352 dolog ("Set handler events=%#x index=%d fd=%d err=%d\n", 353 pfds[i].events, i, pfds[i].fd, err); 354 } 355 356 if (err) { 357 dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n", 358 pfds[i].events, i, pfds[i].fd, err); 359 360 while (i--) { 361 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); 362 } 363 qemu_free (pfds); 364 return -1; 365 } 366 } 367 hlp->pfds = pfds; 368 hlp->count = count; 369 hlp->handle = handle; 370 hlp->mask = mask; 371 return 0; 372 } 373 374 static int alsa_poll_out (HWVoiceOut *hw) 375 { 376 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 377 378 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT); 379 } 380 381 static int alsa_poll_in (HWVoiceIn *hw) 382 { 383 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 384 385 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); 386 } 387 388 static int alsa_write (SWVoiceOut *sw, void *buf, int len) 389 { 390 return audio_pcm_sw_write (sw, buf, len); 391 } 392 393 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt) 394 { 395 switch (fmt) { 396 case AUD_FMT_S8: 397 return SND_PCM_FORMAT_S8; 398 399 case AUD_FMT_U8: 400 return SND_PCM_FORMAT_U8; 401 402 case AUD_FMT_S16: 403 return SND_PCM_FORMAT_S16_LE; 404 405 case AUD_FMT_U16: 406 return SND_PCM_FORMAT_U16_LE; 407 408 case AUD_FMT_S32: 409 return SND_PCM_FORMAT_S32_LE; 410 411 case AUD_FMT_U32: 412 return SND_PCM_FORMAT_U32_LE; 413 414 default: 415 dolog ("Internal logic error: Bad audio format %d\n", fmt); 416 #ifdef DEBUG_AUDIO 417 abort (); 418 #endif 419 return SND_PCM_FORMAT_U8; 420 } 421 } 422 423 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, 424 int *endianness) 425 { 426 switch (alsafmt) { 427 case SND_PCM_FORMAT_S8: 428 *endianness = 0; 429 *fmt = AUD_FMT_S8; 430 break; 431 432 case SND_PCM_FORMAT_U8: 433 *endianness = 0; 434 *fmt = AUD_FMT_U8; 435 break; 436 437 case SND_PCM_FORMAT_S16_LE: 438 *endianness = 0; 439 *fmt = AUD_FMT_S16; 440 break; 441 442 case SND_PCM_FORMAT_U16_LE: 443 *endianness = 0; 444 *fmt = AUD_FMT_U16; 445 break; 446 447 case SND_PCM_FORMAT_S16_BE: 448 *endianness = 1; 449 *fmt = AUD_FMT_S16; 450 break; 451 452 case SND_PCM_FORMAT_U16_BE: 453 *endianness = 1; 454 *fmt = AUD_FMT_U16; 455 break; 456 457 case SND_PCM_FORMAT_S32_LE: 458 *endianness = 0; 459 *fmt = AUD_FMT_S32; 460 break; 461 462 case SND_PCM_FORMAT_U32_LE: 463 *endianness = 0; 464 *fmt = AUD_FMT_U32; 465 break; 466 467 case SND_PCM_FORMAT_S32_BE: 468 *endianness = 1; 469 *fmt = AUD_FMT_S32; 470 break; 471 472 case SND_PCM_FORMAT_U32_BE: 473 *endianness = 1; 474 *fmt = AUD_FMT_U32; 475 break; 476 477 default: 478 dolog ("Unrecognized audio format %d\n", alsafmt); 479 return -1; 480 } 481 482 return 0; 483 } 484 485 static void alsa_dump_info (struct alsa_params_req *req, 486 struct alsa_params_obt *obt, 487 snd_pcm_format_t obtfmt) 488 { 489 dolog ("parameter | requested value | obtained value\n"); 490 dolog ("format | %10d | %10d\n", req->fmt, obtfmt); 491 dolog ("channels | %10d | %10d\n", 492 req->nchannels, obt->nchannels); 493 dolog ("frequency | %10d | %10d\n", req->freq, obt->freq); 494 dolog ("============================================\n"); 495 dolog ("requested: buffer size %d period size %d\n", 496 req->buffer_size, req->period_size); 497 dolog ("obtained: samples %ld\n", obt->samples); 498 } 499 500 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) 501 { 502 int err; 503 snd_pcm_sw_params_t *sw_params; 504 505 FF_snd_pcm_sw_params_alloca (&sw_params); 506 507 err = FF(snd_pcm_sw_params_current) (handle, sw_params); 508 if (err < 0) { 509 dolog ("Could not fully initialize DAC\n"); 510 alsa_logerr (err, "Failed to get current software parameters\n"); 511 return; 512 } 513 514 err = FF(snd_pcm_sw_params_set_start_threshold) (handle, sw_params, threshold); 515 if (err < 0) { 516 dolog ("Could not fully initialize DAC\n"); 517 alsa_logerr (err, "Failed to set software threshold to %ld\n", 518 threshold); 519 return; 520 } 521 522 err = FF(snd_pcm_sw_params) (handle, sw_params); 523 if (err < 0) { 524 dolog ("Could not fully initialize DAC\n"); 525 alsa_logerr (err, "Failed to set software parameters\n"); 526 return; 527 } 528 } 529 530 static int alsa_open (int in, struct alsa_params_req *req, 531 struct alsa_params_obt *obt, snd_pcm_t **handlep) 532 { 533 snd_pcm_t *handle; 534 snd_pcm_hw_params_t *hw_params; 535 int err; 536 int size_in_usec; 537 unsigned int freq, nchannels; 538 const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out; 539 snd_pcm_uframes_t obt_buffer_size; 540 const char *typ = in ? "ADC" : "DAC"; 541 snd_pcm_format_t obtfmt; 542 543 freq = req->freq; 544 nchannels = req->nchannels; 545 size_in_usec = req->size_in_usec; 546 547 FF_snd_pcm_hw_params_alloca (&hw_params); 548 549 err = FF(snd_pcm_open) ( 550 &handle, 551 pcm_name, 552 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 553 SND_PCM_NONBLOCK 554 ); 555 if (err < 0) { 556 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); 557 return -1; 558 } 559 560 err = FF(snd_pcm_hw_params_any) (handle, hw_params); 561 if (err < 0) { 562 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); 563 goto err; 564 } 565 566 err = FF(snd_pcm_hw_params_set_access) ( 567 handle, 568 hw_params, 569 SND_PCM_ACCESS_RW_INTERLEAVED 570 ); 571 if (err < 0) { 572 alsa_logerr2 (err, typ, "Failed to set access type\n"); 573 goto err; 574 } 575 576 err = FF(snd_pcm_hw_params_set_format) (handle, hw_params, req->fmt); 577 if (err < 0 && conf.verbose) { 578 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); 579 goto err; 580 } 581 582 err = FF(snd_pcm_hw_params_set_rate_near) (handle, hw_params, &freq, 0); 583 if (err < 0) { 584 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); 585 goto err; 586 } 587 588 err = FF(snd_pcm_hw_params_set_channels_near) ( 589 handle, 590 hw_params, 591 &nchannels 592 ); 593 if (err < 0) { 594 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", 595 req->nchannels); 596 goto err; 597 } 598 599 if (nchannels != 1 && nchannels != 2) { 600 alsa_logerr2 (err, typ, 601 "Can not handle obtained number of channels %d\n", 602 nchannels); 603 goto err; 604 } 605 606 if (req->buffer_size) { 607 unsigned long obt; 608 609 if (size_in_usec) { 610 int dir = 0; 611 unsigned int btime = req->buffer_size; 612 613 err = FF(snd_pcm_hw_params_set_buffer_time_near) ( 614 handle, 615 hw_params, 616 &btime, 617 &dir 618 ); 619 obt = btime; 620 } 621 else { 622 snd_pcm_uframes_t bsize = req->buffer_size; 623 624 err = FF(snd_pcm_hw_params_set_buffer_size_near) ( 625 handle, 626 hw_params, 627 &bsize 628 ); 629 obt = bsize; 630 } 631 if (err < 0) { 632 alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n", 633 size_in_usec ? "time" : "size", req->buffer_size); 634 goto err; 635 } 636 637 if ((req->override_mask & 2) && (obt - req->buffer_size)) 638 dolog ("Requested buffer %s %u was rejected, using %lu\n", 639 size_in_usec ? "time" : "size", req->buffer_size, obt); 640 } 641 642 if (req->period_size) { 643 unsigned long obt; 644 645 if (size_in_usec) { 646 int dir = 0; 647 unsigned int ptime = req->period_size; 648 649 err = FF(snd_pcm_hw_params_set_period_time_near) ( 650 handle, 651 hw_params, 652 &ptime, 653 &dir 654 ); 655 obt = ptime; 656 } 657 else { 658 int dir = 0; 659 snd_pcm_uframes_t psize = req->period_size; 660 661 err = FF(snd_pcm_hw_params_set_period_size_near) ( 662 handle, 663 hw_params, 664 &psize, 665 &dir 666 ); 667 obt = psize; 668 } 669 670 if (err < 0) { 671 alsa_logerr2 (err, typ, "Failed to set period %s to %d\n", 672 size_in_usec ? "time" : "size", req->period_size); 673 goto err; 674 } 675 676 if (((req->override_mask & 1) && (obt - req->period_size))) 677 dolog ("Requested period %s %u was rejected, using %lu\n", 678 size_in_usec ? "time" : "size", req->period_size, obt); 679 } 680 681 err = FF(snd_pcm_hw_params) (handle, hw_params); 682 if (err < 0) { 683 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); 684 goto err; 685 } 686 687 err = FF(snd_pcm_hw_params_get_buffer_size) (hw_params, &obt_buffer_size); 688 if (err < 0) { 689 alsa_logerr2 (err, typ, "Failed to get buffer size\n"); 690 goto err; 691 } 692 693 err = FF(snd_pcm_hw_params_get_format) (hw_params, &obtfmt); 694 if (err < 0) { 695 alsa_logerr2 (err, typ, "Failed to get format\n"); 696 goto err; 697 } 698 699 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { 700 dolog ("Invalid format was returned %d\n", obtfmt); 701 goto err; 702 } 703 704 err = FF(snd_pcm_prepare) (handle); 705 if (err < 0) { 706 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); 707 goto err; 708 } 709 710 if (!in && conf.threshold) { 711 snd_pcm_uframes_t threshold; 712 int bytes_per_sec; 713 714 bytes_per_sec = freq << (nchannels == 2); 715 716 switch (obt->fmt) { 717 case AUD_FMT_S8: 718 case AUD_FMT_U8: 719 break; 720 721 case AUD_FMT_S16: 722 case AUD_FMT_U16: 723 bytes_per_sec <<= 1; 724 break; 725 726 case AUD_FMT_S32: 727 case AUD_FMT_U32: 728 bytes_per_sec <<= 2; 729 break; 730 } 731 732 threshold = (conf.threshold * bytes_per_sec) / 1000; 733 alsa_set_threshold (handle, threshold); 734 } 735 736 obt->nchannels = nchannels; 737 obt->freq = freq; 738 obt->samples = obt_buffer_size; 739 740 *handlep = handle; 741 742 if (conf.verbose && 743 (obtfmt != req->fmt || 744 obt->nchannels != req->nchannels || 745 obt->freq != req->freq)) { 746 dolog ("Audio parameters for %s\n", typ); 747 alsa_dump_info (req, obt, obtfmt); 748 } 749 750 #ifdef DEBUG 751 alsa_dump_info (req, obt, obtfmt); 752 #endif 753 return 0; 754 755 err: 756 alsa_anal_close1 (&handle); 757 return -1; 758 } 759 760 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle) 761 { 762 snd_pcm_sframes_t avail; 763 764 avail = FF(snd_pcm_avail_update) (handle); 765 if (avail < 0) { 766 if (avail == -EPIPE) { 767 if (!alsa_recover (handle)) { 768 avail = FF(snd_pcm_avail_update) (handle); 769 } 770 } 771 772 if (avail < 0) { 773 alsa_logerr (avail, 774 "Could not obtain number of available frames\n"); 775 return -1; 776 } 777 } 778 779 return avail; 780 } 781 782 static void alsa_write_pending (ALSAVoiceOut *alsa) 783 { 784 HWVoiceOut *hw = &alsa->hw; 785 786 while (alsa->pending) { 787 int left_till_end_samples = hw->samples - alsa->wpos; 788 int len = audio_MIN (alsa->pending, left_till_end_samples); 789 char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift); 790 791 while (len) { 792 snd_pcm_sframes_t written; 793 794 written = FF(snd_pcm_writei) (alsa->handle, src, len); 795 796 if (written <= 0) { 797 switch (written) { 798 case 0: 799 if (conf.verbose) { 800 dolog ("Failed to write %d frames (wrote zero)\n", len); 801 } 802 return; 803 804 case -EPIPE: 805 if (alsa_recover (alsa->handle)) { 806 alsa_logerr (written, "Failed to write %d frames\n", 807 len); 808 return; 809 } 810 if (conf.verbose) { 811 dolog ("Recovering from playback xrun\n"); 812 } 813 continue; 814 815 case -ESTRPIPE: 816 /* stream is suspended and waiting for an 817 application recovery */ 818 if (alsa_resume (alsa->handle)) { 819 alsa_logerr (written, "Failed to write %d frames\n", 820 len); 821 return; 822 } 823 if (conf.verbose) { 824 dolog ("Resuming suspended output stream\n"); 825 } 826 continue; 827 828 case -EAGAIN: 829 return; 830 831 default: 832 alsa_logerr (written, "Failed to write %d frames from %p\n", 833 len, src); 834 return; 835 } 836 } 837 838 alsa->wpos = (alsa->wpos + written) % hw->samples; 839 alsa->pending -= written; 840 len -= written; 841 } 842 } 843 } 844 845 static int alsa_run_out (HWVoiceOut *hw, int live) 846 { 847 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 848 int decr; 849 snd_pcm_sframes_t avail; 850 851 avail = alsa_get_avail (alsa->handle); 852 if (avail < 0) { 853 dolog ("Could not get number of available playback frames\n"); 854 return 0; 855 } 856 857 decr = audio_MIN (live, avail); 858 decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending); 859 alsa->pending += decr; 860 alsa_write_pending (alsa); 861 return decr; 862 } 863 864 static void alsa_fini_out (HWVoiceOut *hw) 865 { 866 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 867 868 ldebug ("alsa_fini\n"); 869 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 870 871 if (alsa->pcm_buf) { 872 qemu_free (alsa->pcm_buf); 873 alsa->pcm_buf = NULL; 874 } 875 } 876 877 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as) 878 { 879 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 880 struct alsa_params_req req; 881 struct alsa_params_obt obt; 882 snd_pcm_t *handle; 883 struct audsettings obt_as; 884 int result = -1; 885 886 /* shut alsa debug spew */ 887 if (!D_ACTIVE) 888 stdio_disable(); 889 890 req.fmt = aud_to_alsafmt (as->fmt); 891 req.freq = as->freq; 892 req.nchannels = as->nchannels; 893 req.period_size = conf.period_size_out; 894 req.buffer_size = conf.buffer_size_out; 895 req.size_in_usec = conf.size_in_usec_out; 896 req.override_mask = 897 (conf.period_size_out_overridden ? 1 : 0) | 898 (conf.buffer_size_out_overridden ? 2 : 0); 899 900 if (alsa_open (0, &req, &obt, &handle)) { 901 goto Exit; 902 } 903 904 obt_as.freq = obt.freq; 905 obt_as.nchannels = obt.nchannels; 906 obt_as.fmt = obt.fmt; 907 obt_as.endianness = obt.endianness; 908 909 audio_pcm_init_info (&hw->info, &obt_as); 910 hw->samples = obt.samples; 911 912 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift); 913 if (!alsa->pcm_buf) { 914 dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n", 915 hw->samples, 1 << hw->info.shift); 916 alsa_anal_close1 (&handle); 917 goto Exit; 918 } 919 920 alsa->handle = handle; 921 result = 0; /* success */ 922 923 Exit: 924 if (!D_ACTIVE) 925 stdio_enable(); 926 927 return result; 928 } 929 930 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause) 931 { 932 int err; 933 934 if (pause) { 935 err = FF(snd_pcm_drop) (handle); 936 if (err < 0) { 937 alsa_logerr (err, "Could not stop %s\n", typ); 938 return -1; 939 } 940 } 941 else { 942 err = FF(snd_pcm_prepare) (handle); 943 if (err < 0) { 944 alsa_logerr (err, "Could not prepare handle for %s\n", typ); 945 return -1; 946 } 947 } 948 949 return 0; 950 } 951 952 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) 953 { 954 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 955 956 switch (cmd) { 957 case VOICE_ENABLE: 958 { 959 va_list ap; 960 int poll_mode; 961 962 va_start (ap, cmd); 963 poll_mode = va_arg (ap, int); 964 va_end (ap); 965 966 ldebug ("enabling voice\n"); 967 if (poll_mode && alsa_poll_out (hw)) { 968 poll_mode = 0; 969 } 970 hw->poll_mode = poll_mode; 971 return alsa_voice_ctl (alsa->handle, "playback", 0); 972 } 973 974 case VOICE_DISABLE: 975 ldebug ("disabling voice\n"); 976 return alsa_voice_ctl (alsa->handle, "playback", 1); 977 } 978 979 return -1; 980 } 981 982 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as) 983 { 984 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 985 struct alsa_params_req req; 986 struct alsa_params_obt obt; 987 snd_pcm_t *handle; 988 struct audsettings obt_as; 989 int result = -1; 990 991 /* shut alsa debug spew */ 992 if (!D_ACTIVE) 993 stdio_disable(); 994 995 req.fmt = aud_to_alsafmt (as->fmt); 996 req.freq = as->freq; 997 req.nchannels = as->nchannels; 998 req.period_size = conf.period_size_in; 999 req.buffer_size = conf.buffer_size_in; 1000 req.size_in_usec = conf.size_in_usec_in; 1001 req.override_mask = 1002 (conf.period_size_in_overridden ? 1 : 0) | 1003 (conf.buffer_size_in_overridden ? 2 : 0); 1004 1005 if (alsa_open (1, &req, &obt, &handle)) { 1006 goto Exit; 1007 } 1008 1009 obt_as.freq = obt.freq; 1010 obt_as.nchannels = obt.nchannels; 1011 obt_as.fmt = obt.fmt; 1012 obt_as.endianness = obt.endianness; 1013 1014 audio_pcm_init_info (&hw->info, &obt_as); 1015 hw->samples = obt.samples; 1016 1017 alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); 1018 if (!alsa->pcm_buf) { 1019 dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n", 1020 hw->samples, 1 << hw->info.shift); 1021 alsa_anal_close1 (&handle); 1022 goto Exit; 1023 } 1024 1025 alsa->handle = handle; 1026 result = 0; /* success */ 1027 1028 Exit: 1029 if (!D_ACTIVE) 1030 stdio_enable(); 1031 1032 return result; 1033 } 1034 1035 static void alsa_fini_in (HWVoiceIn *hw) 1036 { 1037 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 1038 1039 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 1040 1041 if (alsa->pcm_buf) { 1042 qemu_free (alsa->pcm_buf); 1043 alsa->pcm_buf = NULL; 1044 } 1045 } 1046 1047 static int alsa_run_in (HWVoiceIn *hw) 1048 { 1049 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 1050 int hwshift = hw->info.shift; 1051 int i; 1052 int live = audio_pcm_hw_get_live_in (hw); 1053 int dead = hw->samples - live; 1054 int decr; 1055 struct { 1056 int add; 1057 int len; 1058 } bufs[2] = { 1059 { .add = hw->wpos, .len = 0 }, 1060 { .add = 0, .len = 0 } 1061 }; 1062 snd_pcm_sframes_t avail; 1063 snd_pcm_uframes_t read_samples = 0; 1064 1065 if (!dead) { 1066 return 0; 1067 } 1068 1069 avail = alsa_get_avail (alsa->handle); 1070 if (avail < 0) { 1071 dolog ("Could not get number of captured frames\n"); 1072 return 0; 1073 } 1074 1075 if (!avail) { 1076 snd_pcm_state_t state; 1077 1078 state = FF(snd_pcm_state) (alsa->handle); 1079 switch (state) { 1080 case SND_PCM_STATE_PREPARED: 1081 avail = hw->samples; 1082 break; 1083 case SND_PCM_STATE_SUSPENDED: 1084 /* stream is suspended and waiting for an application recovery */ 1085 if (alsa_resume (alsa->handle)) { 1086 dolog ("Failed to resume suspended input stream\n"); 1087 return 0; 1088 } 1089 if (conf.verbose) { 1090 dolog ("Resuming suspended input stream\n"); 1091 } 1092 break; 1093 default: 1094 if (conf.verbose) { 1095 dolog ("No frames available and ALSA state is %d\n", state); 1096 } 1097 return 0; 1098 } 1099 } 1100 1101 decr = audio_MIN (dead, avail); 1102 if (!decr) { 1103 return 0; 1104 } 1105 1106 if (hw->wpos + decr > hw->samples) { 1107 bufs[0].len = (hw->samples - hw->wpos); 1108 bufs[1].len = (decr - (hw->samples - hw->wpos)); 1109 } 1110 else { 1111 bufs[0].len = decr; 1112 } 1113 1114 for (i = 0; i < 2; ++i) { 1115 void *src; 1116 struct st_sample *dst; 1117 snd_pcm_sframes_t nread; 1118 snd_pcm_uframes_t len; 1119 1120 len = bufs[i].len; 1121 1122 src = advance (alsa->pcm_buf, bufs[i].add << hwshift); 1123 dst = hw->conv_buf + bufs[i].add; 1124 1125 while (len) { 1126 nread = FF(snd_pcm_readi) (alsa->handle, src, len); 1127 1128 if (nread <= 0) { 1129 switch (nread) { 1130 case 0: 1131 if (conf.verbose) { 1132 dolog ("Failed to read %ld frames (read zero)\n", len); 1133 } 1134 goto exit; 1135 1136 case -EPIPE: 1137 if (alsa_recover (alsa->handle)) { 1138 alsa_logerr (nread, "Failed to read %ld frames\n", len); 1139 goto exit; 1140 } 1141 if (conf.verbose) { 1142 dolog ("Recovering from capture xrun\n"); 1143 } 1144 continue; 1145 1146 case -EAGAIN: 1147 goto exit; 1148 1149 default: 1150 alsa_logerr ( 1151 nread, 1152 "Failed to read %ld frames from %p\n", 1153 len, 1154 src 1155 ); 1156 goto exit; 1157 } 1158 } 1159 1160 hw->conv (dst, src, nread, &nominal_volume); 1161 1162 src = advance (src, nread << hwshift); 1163 dst += nread; 1164 1165 read_samples += nread; 1166 len -= nread; 1167 } 1168 } 1169 1170 exit: 1171 hw->wpos = (hw->wpos + read_samples) % hw->samples; 1172 return read_samples; 1173 } 1174 1175 static int alsa_read (SWVoiceIn *sw, void *buf, int size) 1176 { 1177 return audio_pcm_sw_read (sw, buf, size); 1178 } 1179 1180 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) 1181 { 1182 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 1183 1184 switch (cmd) { 1185 case VOICE_ENABLE: 1186 { 1187 va_list ap; 1188 int poll_mode; 1189 1190 va_start (ap, cmd); 1191 poll_mode = va_arg (ap, int); 1192 va_end (ap); 1193 1194 ldebug ("enabling voice\n"); 1195 if (poll_mode && alsa_poll_in (hw)) { 1196 poll_mode = 0; 1197 } 1198 hw->poll_mode = poll_mode; 1199 1200 return alsa_voice_ctl (alsa->handle, "capture", 0); 1201 } 1202 1203 case VOICE_DISABLE: 1204 ldebug ("disabling voice\n"); 1205 if (hw->poll_mode) { 1206 hw->poll_mode = 0; 1207 alsa_fini_poll (&alsa->pollhlp); 1208 } 1209 return alsa_voice_ctl (alsa->handle, "capture", 1); 1210 } 1211 1212 return -1; 1213 } 1214 1215 static void *alsa_audio_init (void) 1216 { 1217 void* result = NULL; 1218 1219 alsa_lib = dlopen( "libasound.so", RTLD_NOW ); 1220 if (alsa_lib == NULL) 1221 alsa_lib = dlopen( "libasound.so.2", RTLD_NOW ); 1222 1223 if (alsa_lib == NULL) { 1224 ldebug("could not find libasound on this system\n"); 1225 goto Exit; 1226 } 1227 1228 if (alsa_dynlink_init(alsa_lib) < 0) 1229 goto Fail; 1230 1231 result = &conf; 1232 goto Exit; 1233 1234 Fail: 1235 ldebug("%s: failed to open library\n", __FUNCTION__); 1236 dlclose(alsa_lib); 1237 1238 Exit: 1239 return result; 1240 } 1241 1242 static void alsa_audio_fini (void *opaque) 1243 { 1244 if (alsa_lib != NULL) { 1245 dlclose(alsa_lib); 1246 alsa_lib = NULL; 1247 } 1248 (void) opaque; 1249 } 1250 1251 static struct audio_option alsa_options[] = { 1252 { 1253 .name = "DAC_SIZE_IN_USEC", 1254 .tag = AUD_OPT_BOOL, 1255 .valp = &conf.size_in_usec_out, 1256 .descr = "DAC period/buffer size in microseconds (otherwise in frames)" 1257 }, 1258 { 1259 .name = "DAC_PERIOD_SIZE", 1260 .tag = AUD_OPT_INT, 1261 .valp = &conf.period_size_out, 1262 .descr = "DAC period size (0 to go with system default)", 1263 .overriddenp = &conf.period_size_out_overridden 1264 }, 1265 { 1266 .name = "DAC_BUFFER_SIZE", 1267 .tag = AUD_OPT_INT, 1268 .valp = &conf.buffer_size_out, 1269 .descr = "DAC buffer size (0 to go with system default)", 1270 .overriddenp = &conf.buffer_size_out_overridden 1271 }, 1272 { 1273 .name = "ADC_SIZE_IN_USEC", 1274 .tag = AUD_OPT_BOOL, 1275 .valp = &conf.size_in_usec_in, 1276 .descr = 1277 "ADC period/buffer size in microseconds (otherwise in frames)" 1278 }, 1279 { 1280 .name = "ADC_PERIOD_SIZE", 1281 .tag = AUD_OPT_INT, 1282 .valp = &conf.period_size_in, 1283 .descr = "ADC period size (0 to go with system default)", 1284 .overriddenp = &conf.period_size_in_overridden 1285 }, 1286 { 1287 .name = "ADC_BUFFER_SIZE", 1288 .tag = AUD_OPT_INT, 1289 .valp = &conf.buffer_size_in, 1290 .descr = "ADC buffer size (0 to go with system default)", 1291 .overriddenp = &conf.buffer_size_in_overridden 1292 }, 1293 { 1294 .name = "THRESHOLD", 1295 .tag = AUD_OPT_INT, 1296 .valp = &conf.threshold, 1297 .descr = "(undocumented)" 1298 }, 1299 { 1300 .name = "DAC_DEV", 1301 .tag = AUD_OPT_STR, 1302 .valp = &conf.pcm_name_out, 1303 .descr = "DAC device name (for instance dmix)" 1304 }, 1305 { 1306 .name = "ADC_DEV", 1307 .tag = AUD_OPT_STR, 1308 .valp = &conf.pcm_name_in, 1309 .descr = "ADC device name" 1310 }, 1311 { 1312 .name = "VERBOSE", 1313 .tag = AUD_OPT_BOOL, 1314 .valp = &conf.verbose, 1315 .descr = "Behave in a more verbose way" 1316 }, 1317 { /* End of list */ } 1318 }; 1319 1320 static struct audio_pcm_ops alsa_pcm_ops = { 1321 .init_out = alsa_init_out, 1322 .fini_out = alsa_fini_out, 1323 .run_out = alsa_run_out, 1324 .write = alsa_write, 1325 .ctl_out = alsa_ctl_out, 1326 1327 .init_in = alsa_init_in, 1328 .fini_in = alsa_fini_in, 1329 .run_in = alsa_run_in, 1330 .read = alsa_read, 1331 .ctl_in = alsa_ctl_in, 1332 }; 1333 1334 struct audio_driver alsa_audio_driver = { 1335 .name = "alsa", 1336 .descr = "ALSA http://www.alsa-project.org", 1337 .options = alsa_options, 1338 .init = alsa_audio_init, 1339 .fini = alsa_audio_fini, 1340 .pcm_ops = &alsa_pcm_ops, 1341 .can_be_default = 1, 1342 .max_voices_out = INT_MAX, 1343 .max_voices_in = INT_MAX, 1344 .voice_size_out = sizeof (ALSAVoiceOut), 1345 .voice_size_in = sizeof (ALSAVoiceIn) 1346 }; 1347
