1 /* 2 * QEMU ALSA audio driver 3 * 4 * Copyright (c) 2005 Vassili Karpov (malc) 5 * 6 * Permission is hereby granted, free of charge, to any person obtaining a copy 7 * of this software and associated documentation files (the "Software"), to deal 8 * in the Software without restriction, including without limitation the rights 9 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell 10 * copies of the Software, and to permit persons to whom the Software is 11 * furnished to do so, subject to the following conditions: 12 * 13 * The above copyright notice and this permission notice shall be included in 14 * all copies or substantial portions of the Software. 15 * 16 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR 17 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, 18 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL 19 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER 20 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, 21 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN 22 * THE SOFTWARE. 23 */ 24 25 #include "qemu/osdep.h" 26 #include <alsa/asoundlib.h> 27 #include "qemu/main-loop.h" 28 #include "qemu/module.h" 29 #include "audio.h" 30 #include "trace.h" 31 32 #pragma GCC diagnostic ignored "-Waddress" 33 34 #define AUDIO_CAP "alsa" 35 #include "audio_int.h" 36 37 struct pollhlp { 38 snd_pcm_t *handle; 39 struct pollfd *pfds; 40 int count; 41 int mask; 42 AudioState *s; 43 }; 44 45 typedef struct ALSAVoiceOut { 46 HWVoiceOut hw; 47 snd_pcm_t *handle; 48 struct pollhlp pollhlp; 49 Audiodev *dev; 50 } ALSAVoiceOut; 51 52 typedef struct ALSAVoiceIn { 53 HWVoiceIn hw; 54 snd_pcm_t *handle; 55 struct pollhlp pollhlp; 56 Audiodev *dev; 57 } ALSAVoiceIn; 58 59 struct alsa_params_req { 60 int freq; 61 snd_pcm_format_t fmt; 62 int nchannels; 63 }; 64 65 struct alsa_params_obt { 66 int freq; 67 AudioFormat fmt; 68 int endianness; 69 int nchannels; 70 snd_pcm_uframes_t samples; 71 }; 72 73 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) 74 { 75 va_list ap; 76 77 va_start (ap, fmt); 78 AUD_vlog (AUDIO_CAP, fmt, ap); 79 va_end (ap); 80 81 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); 82 } 83 84 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( 85 int err, 86 const char *typ, 87 const char *fmt, 88 ... 89 ) 90 { 91 va_list ap; 92 93 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ); 94 95 va_start (ap, fmt); 96 AUD_vlog (AUDIO_CAP, fmt, ap); 97 va_end (ap); 98 99 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); 100 } 101 102 static void alsa_fini_poll (struct pollhlp *hlp) 103 { 104 int i; 105 struct pollfd *pfds = hlp->pfds; 106 107 if (pfds) { 108 for (i = 0; i < hlp->count; ++i) { 109 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); 110 } 111 g_free (pfds); 112 } 113 hlp->pfds = NULL; 114 hlp->count = 0; 115 hlp->handle = NULL; 116 } 117 118 static void alsa_anal_close1 (snd_pcm_t **handlep) 119 { 120 int err = snd_pcm_close (*handlep); 121 if (err) { 122 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep); 123 } 124 *handlep = NULL; 125 } 126 127 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp) 128 { 129 alsa_fini_poll (hlp); 130 alsa_anal_close1 (handlep); 131 } 132 133 static int alsa_recover (snd_pcm_t *handle) 134 { 135 int err = snd_pcm_prepare (handle); 136 if (err < 0) { 137 alsa_logerr (err, "Failed to prepare handle %p\n", handle); 138 return -1; 139 } 140 return 0; 141 } 142 143 static int alsa_resume (snd_pcm_t *handle) 144 { 145 int err = snd_pcm_resume (handle); 146 if (err < 0) { 147 alsa_logerr (err, "Failed to resume handle %p\n", handle); 148 return -1; 149 } 150 return 0; 151 } 152 153 static void alsa_poll_handler (void *opaque) 154 { 155 int err, count; 156 snd_pcm_state_t state; 157 struct pollhlp *hlp = opaque; 158 unsigned short revents; 159 160 count = poll (hlp->pfds, hlp->count, 0); 161 if (count < 0) { 162 dolog ("alsa_poll_handler: poll %s\n", strerror (errno)); 163 return; 164 } 165 166 if (!count) { 167 return; 168 } 169 170 /* XXX: ALSA example uses initial count, not the one returned by 171 poll, correct? */ 172 err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds, 173 hlp->count, &revents); 174 if (err < 0) { 175 alsa_logerr (err, "snd_pcm_poll_descriptors_revents"); 176 return; 177 } 178 179 if (!(revents & hlp->mask)) { 180 trace_alsa_revents(revents); 181 return; 182 } 183 184 state = snd_pcm_state (hlp->handle); 185 switch (state) { 186 case SND_PCM_STATE_SETUP: 187 alsa_recover (hlp->handle); 188 break; 189 190 case SND_PCM_STATE_XRUN: 191 alsa_recover (hlp->handle); 192 break; 193 194 case SND_PCM_STATE_SUSPENDED: 195 alsa_resume (hlp->handle); 196 break; 197 198 case SND_PCM_STATE_PREPARED: 199 audio_run(hlp->s, "alsa run (prepared)"); 200 break; 201 202 case SND_PCM_STATE_RUNNING: 203 audio_run(hlp->s, "alsa run (running)"); 204 break; 205 206 default: 207 dolog ("Unexpected state %d\n", state); 208 } 209 } 210 211 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) 212 { 213 int i, count, err; 214 struct pollfd *pfds; 215 216 count = snd_pcm_poll_descriptors_count (handle); 217 if (count <= 0) { 218 dolog ("Could not initialize poll mode\n" 219 "Invalid number of poll descriptors %d\n", count); 220 return -1; 221 } 222 223 pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); 224 if (!pfds) { 225 dolog ("Could not initialize poll mode\n"); 226 return -1; 227 } 228 229 err = snd_pcm_poll_descriptors (handle, pfds, count); 230 if (err < 0) { 231 alsa_logerr (err, "Could not initialize poll mode\n" 232 "Could not obtain poll descriptors\n"); 233 g_free (pfds); 234 return -1; 235 } 236 237 for (i = 0; i < count; ++i) { 238 if (pfds[i].events & POLLIN) { 239 qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp); 240 } 241 if (pfds[i].events & POLLOUT) { 242 trace_alsa_pollout(i, pfds[i].fd); 243 qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp); 244 } 245 trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err); 246 247 } 248 hlp->pfds = pfds; 249 hlp->count = count; 250 hlp->handle = handle; 251 hlp->mask = mask; 252 return 0; 253 } 254 255 static int alsa_poll_out (HWVoiceOut *hw) 256 { 257 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 258 259 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT); 260 } 261 262 static int alsa_poll_in (HWVoiceIn *hw) 263 { 264 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 265 266 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); 267 } 268 269 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness) 270 { 271 switch (fmt) { 272 case AUDIO_FORMAT_S8: 273 return SND_PCM_FORMAT_S8; 274 275 case AUDIO_FORMAT_U8: 276 return SND_PCM_FORMAT_U8; 277 278 case AUDIO_FORMAT_S16: 279 if (endianness) { 280 return SND_PCM_FORMAT_S16_BE; 281 } 282 else { 283 return SND_PCM_FORMAT_S16_LE; 284 } 285 286 case AUDIO_FORMAT_U16: 287 if (endianness) { 288 return SND_PCM_FORMAT_U16_BE; 289 } 290 else { 291 return SND_PCM_FORMAT_U16_LE; 292 } 293 294 case AUDIO_FORMAT_S32: 295 if (endianness) { 296 return SND_PCM_FORMAT_S32_BE; 297 } 298 else { 299 return SND_PCM_FORMAT_S32_LE; 300 } 301 302 case AUDIO_FORMAT_U32: 303 if (endianness) { 304 return SND_PCM_FORMAT_U32_BE; 305 } 306 else { 307 return SND_PCM_FORMAT_U32_LE; 308 } 309 310 default: 311 dolog ("Internal logic error: Bad audio format %d\n", fmt); 312 #ifdef DEBUG_AUDIO 313 abort (); 314 #endif 315 return SND_PCM_FORMAT_U8; 316 } 317 } 318 319 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt, 320 int *endianness) 321 { 322 switch (alsafmt) { 323 case SND_PCM_FORMAT_S8: 324 *endianness = 0; 325 *fmt = AUDIO_FORMAT_S8; 326 break; 327 328 case SND_PCM_FORMAT_U8: 329 *endianness = 0; 330 *fmt = AUDIO_FORMAT_U8; 331 break; 332 333 case SND_PCM_FORMAT_S16_LE: 334 *endianness = 0; 335 *fmt = AUDIO_FORMAT_S16; 336 break; 337 338 case SND_PCM_FORMAT_U16_LE: 339 *endianness = 0; 340 *fmt = AUDIO_FORMAT_U16; 341 break; 342 343 case SND_PCM_FORMAT_S16_BE: 344 *endianness = 1; 345 *fmt = AUDIO_FORMAT_S16; 346 break; 347 348 case SND_PCM_FORMAT_U16_BE: 349 *endianness = 1; 350 *fmt = AUDIO_FORMAT_U16; 351 break; 352 353 case SND_PCM_FORMAT_S32_LE: 354 *endianness = 0; 355 *fmt = AUDIO_FORMAT_S32; 356 break; 357 358 case SND_PCM_FORMAT_U32_LE: 359 *endianness = 0; 360 *fmt = AUDIO_FORMAT_U32; 361 break; 362 363 case SND_PCM_FORMAT_S32_BE: 364 *endianness = 1; 365 *fmt = AUDIO_FORMAT_S32; 366 break; 367 368 case SND_PCM_FORMAT_U32_BE: 369 *endianness = 1; 370 *fmt = AUDIO_FORMAT_U32; 371 break; 372 373 default: 374 dolog ("Unrecognized audio format %d\n", alsafmt); 375 return -1; 376 } 377 378 return 0; 379 } 380 381 static void alsa_dump_info (struct alsa_params_req *req, 382 struct alsa_params_obt *obt, 383 snd_pcm_format_t obtfmt, 384 AudiodevAlsaPerDirectionOptions *apdo) 385 { 386 dolog("parameter | requested value | obtained value\n"); 387 dolog("format | %10d | %10d\n", req->fmt, obtfmt); 388 dolog("channels | %10d | %10d\n", 389 req->nchannels, obt->nchannels); 390 dolog("frequency | %10d | %10d\n", req->freq, obt->freq); 391 dolog("============================================\n"); 392 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n", 393 apdo->buffer_length, apdo->period_length); 394 dolog("obtained: samples %ld\n", obt->samples); 395 } 396 397 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) 398 { 399 int err; 400 snd_pcm_sw_params_t *sw_params; 401 402 snd_pcm_sw_params_alloca (&sw_params); 403 404 err = snd_pcm_sw_params_current (handle, sw_params); 405 if (err < 0) { 406 dolog ("Could not fully initialize DAC\n"); 407 alsa_logerr (err, "Failed to get current software parameters\n"); 408 return; 409 } 410 411 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); 412 if (err < 0) { 413 dolog ("Could not fully initialize DAC\n"); 414 alsa_logerr (err, "Failed to set software threshold to %ld\n", 415 threshold); 416 return; 417 } 418 419 err = snd_pcm_sw_params (handle, sw_params); 420 if (err < 0) { 421 dolog ("Could not fully initialize DAC\n"); 422 alsa_logerr (err, "Failed to set software parameters\n"); 423 return; 424 } 425 } 426 427 static int alsa_open(bool in, struct alsa_params_req *req, 428 struct alsa_params_obt *obt, snd_pcm_t **handlep, 429 Audiodev *dev) 430 { 431 AudiodevAlsaOptions *aopts = &dev->u.alsa; 432 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out; 433 snd_pcm_t *handle; 434 snd_pcm_hw_params_t *hw_params; 435 int err; 436 unsigned int freq, nchannels; 437 const char *pcm_name = apdo->has_dev ? apdo->dev : "default"; 438 snd_pcm_uframes_t obt_buffer_size; 439 const char *typ = in ? "ADC" : "DAC"; 440 snd_pcm_format_t obtfmt; 441 442 freq = req->freq; 443 nchannels = req->nchannels; 444 445 snd_pcm_hw_params_alloca (&hw_params); 446 447 err = snd_pcm_open ( 448 &handle, 449 pcm_name, 450 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 451 SND_PCM_NONBLOCK 452 ); 453 if (err < 0) { 454 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); 455 return -1; 456 } 457 458 err = snd_pcm_hw_params_any (handle, hw_params); 459 if (err < 0) { 460 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); 461 goto err; 462 } 463 464 err = snd_pcm_hw_params_set_access ( 465 handle, 466 hw_params, 467 SND_PCM_ACCESS_RW_INTERLEAVED 468 ); 469 if (err < 0) { 470 alsa_logerr2 (err, typ, "Failed to set access type\n"); 471 goto err; 472 } 473 474 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); 475 if (err < 0) { 476 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); 477 } 478 479 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); 480 if (err < 0) { 481 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); 482 goto err; 483 } 484 485 err = snd_pcm_hw_params_set_channels_near ( 486 handle, 487 hw_params, 488 &nchannels 489 ); 490 if (err < 0) { 491 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", 492 req->nchannels); 493 goto err; 494 } 495 496 if (apdo->buffer_length) { 497 int dir = 0; 498 unsigned int btime = apdo->buffer_length; 499 500 err = snd_pcm_hw_params_set_buffer_time_near( 501 handle, hw_params, &btime, &dir); 502 503 if (err < 0) { 504 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n", 505 apdo->buffer_length); 506 goto err; 507 } 508 509 if (apdo->has_buffer_length && btime != apdo->buffer_length) { 510 dolog("Requested buffer time %" PRId32 511 " was rejected, using %u\n", apdo->buffer_length, btime); 512 } 513 } 514 515 if (apdo->period_length) { 516 int dir = 0; 517 unsigned int ptime = apdo->period_length; 518 519 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime, 520 &dir); 521 522 if (err < 0) { 523 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n", 524 apdo->period_length); 525 goto err; 526 } 527 528 if (apdo->has_period_length && ptime != apdo->period_length) { 529 dolog("Requested period time %" PRId32 " was rejected, using %d\n", 530 apdo->period_length, ptime); 531 } 532 } 533 534 err = snd_pcm_hw_params (handle, hw_params); 535 if (err < 0) { 536 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); 537 goto err; 538 } 539 540 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); 541 if (err < 0) { 542 alsa_logerr2 (err, typ, "Failed to get buffer size\n"); 543 goto err; 544 } 545 546 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); 547 if (err < 0) { 548 alsa_logerr2 (err, typ, "Failed to get format\n"); 549 goto err; 550 } 551 552 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { 553 dolog ("Invalid format was returned %d\n", obtfmt); 554 goto err; 555 } 556 557 err = snd_pcm_prepare (handle); 558 if (err < 0) { 559 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); 560 goto err; 561 } 562 563 if (!in && aopts->has_threshold && aopts->threshold) { 564 struct audsettings as = { .freq = freq }; 565 alsa_set_threshold( 566 handle, 567 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo), 568 &as, aopts->threshold)); 569 } 570 571 obt->nchannels = nchannels; 572 obt->freq = freq; 573 obt->samples = obt_buffer_size; 574 575 *handlep = handle; 576 577 if (obtfmt != req->fmt || 578 obt->nchannels != req->nchannels || 579 obt->freq != req->freq) { 580 dolog ("Audio parameters for %s\n", typ); 581 alsa_dump_info(req, obt, obtfmt, apdo); 582 } 583 584 #ifdef DEBUG 585 alsa_dump_info(req, obt, obtfmt, pdo); 586 #endif 587 return 0; 588 589 err: 590 alsa_anal_close1 (&handle); 591 return -1; 592 } 593 594 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) 595 { 596 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 597 size_t pos = 0; 598 size_t len_frames = len / hw->info.bytes_per_frame; 599 600 while (len_frames) { 601 char *src = advance(buf, pos); 602 snd_pcm_sframes_t written; 603 604 written = snd_pcm_writei(alsa->handle, src, len_frames); 605 606 if (written <= 0) { 607 switch (written) { 608 case 0: 609 trace_alsa_wrote_zero(len_frames); 610 return pos; 611 612 case -EPIPE: 613 if (alsa_recover(alsa->handle)) { 614 alsa_logerr(written, "Failed to write %zu frames\n", 615 len_frames); 616 return pos; 617 } 618 trace_alsa_xrun_out(); 619 continue; 620 621 case -ESTRPIPE: 622 /* 623 * stream is suspended and waiting for an application 624 * recovery 625 */ 626 if (alsa_resume(alsa->handle)) { 627 alsa_logerr(written, "Failed to write %zu frames\n", 628 len_frames); 629 return pos; 630 } 631 trace_alsa_resume_out(); 632 continue; 633 634 case -EAGAIN: 635 return pos; 636 637 default: 638 alsa_logerr(written, "Failed to write %zu frames from %p\n", 639 len, src); 640 return pos; 641 } 642 } 643 644 pos += written * hw->info.bytes_per_frame; 645 if (written < len_frames) { 646 break; 647 } 648 len_frames -= written; 649 } 650 651 return pos; 652 } 653 654 static void alsa_fini_out (HWVoiceOut *hw) 655 { 656 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 657 658 ldebug ("alsa_fini\n"); 659 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 660 } 661 662 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, 663 void *drv_opaque) 664 { 665 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 666 struct alsa_params_req req; 667 struct alsa_params_obt obt; 668 snd_pcm_t *handle; 669 struct audsettings obt_as; 670 Audiodev *dev = drv_opaque; 671 672 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 673 req.freq = as->freq; 674 req.nchannels = as->nchannels; 675 676 if (alsa_open(0, &req, &obt, &handle, dev)) { 677 return -1; 678 } 679 680 obt_as.freq = obt.freq; 681 obt_as.nchannels = obt.nchannels; 682 obt_as.fmt = obt.fmt; 683 obt_as.endianness = obt.endianness; 684 685 audio_pcm_init_info (&hw->info, &obt_as); 686 hw->samples = obt.samples; 687 688 alsa->pollhlp.s = hw->s; 689 alsa->handle = handle; 690 alsa->dev = dev; 691 return 0; 692 } 693 694 #define VOICE_CTL_PAUSE 0 695 #define VOICE_CTL_PREPARE 1 696 #define VOICE_CTL_START 2 697 698 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) 699 { 700 int err; 701 702 if (ctl == VOICE_CTL_PAUSE) { 703 err = snd_pcm_drop (handle); 704 if (err < 0) { 705 alsa_logerr (err, "Could not stop %s\n", typ); 706 return -1; 707 } 708 } 709 else { 710 err = snd_pcm_prepare (handle); 711 if (err < 0) { 712 alsa_logerr (err, "Could not prepare handle for %s\n", typ); 713 return -1; 714 } 715 if (ctl == VOICE_CTL_START) { 716 err = snd_pcm_start(handle); 717 if (err < 0) { 718 alsa_logerr (err, "Could not start handle for %s\n", typ); 719 return -1; 720 } 721 } 722 } 723 724 return 0; 725 } 726 727 static void alsa_enable_out(HWVoiceOut *hw, bool enable) 728 { 729 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 730 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out; 731 732 if (enable) { 733 bool poll_mode = apdo->try_poll; 734 735 ldebug("enabling voice\n"); 736 if (poll_mode && alsa_poll_out(hw)) { 737 poll_mode = 0; 738 } 739 hw->poll_mode = poll_mode; 740 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE); 741 } else { 742 ldebug("disabling voice\n"); 743 if (hw->poll_mode) { 744 hw->poll_mode = 0; 745 alsa_fini_poll(&alsa->pollhlp); 746 } 747 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE); 748 } 749 } 750 751 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) 752 { 753 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 754 struct alsa_params_req req; 755 struct alsa_params_obt obt; 756 snd_pcm_t *handle; 757 struct audsettings obt_as; 758 Audiodev *dev = drv_opaque; 759 760 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 761 req.freq = as->freq; 762 req.nchannels = as->nchannels; 763 764 if (alsa_open(1, &req, &obt, &handle, dev)) { 765 return -1; 766 } 767 768 obt_as.freq = obt.freq; 769 obt_as.nchannels = obt.nchannels; 770 obt_as.fmt = obt.fmt; 771 obt_as.endianness = obt.endianness; 772 773 audio_pcm_init_info (&hw->info, &obt_as); 774 hw->samples = obt.samples; 775 776 alsa->pollhlp.s = hw->s; 777 alsa->handle = handle; 778 alsa->dev = dev; 779 return 0; 780 } 781 782 static void alsa_fini_in (HWVoiceIn *hw) 783 { 784 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 785 786 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 787 } 788 789 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) 790 { 791 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 792 size_t pos = 0; 793 794 while (len) { 795 void *dst = advance(buf, pos); 796 snd_pcm_sframes_t nread; 797 798 nread = snd_pcm_readi( 799 alsa->handle, dst, len / hw->info.bytes_per_frame); 800 801 if (nread <= 0) { 802 switch (nread) { 803 case 0: 804 trace_alsa_read_zero(len); 805 return pos;; 806 807 case -EPIPE: 808 if (alsa_recover(alsa->handle)) { 809 alsa_logerr(nread, "Failed to read %zu frames\n", len); 810 return pos; 811 } 812 trace_alsa_xrun_in(); 813 continue; 814 815 case -EAGAIN: 816 return pos; 817 818 default: 819 alsa_logerr(nread, "Failed to read %zu frames to %p\n", 820 len, dst); 821 return pos;; 822 } 823 } 824 825 pos += nread * hw->info.bytes_per_frame; 826 len -= nread * hw->info.bytes_per_frame; 827 } 828 829 return pos; 830 } 831 832 static void alsa_enable_in(HWVoiceIn *hw, bool enable) 833 { 834 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 835 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in; 836 837 if (enable) { 838 bool poll_mode = apdo->try_poll; 839 840 ldebug("enabling voice\n"); 841 if (poll_mode && alsa_poll_in(hw)) { 842 poll_mode = 0; 843 } 844 hw->poll_mode = poll_mode; 845 846 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START); 847 } else { 848 ldebug ("disabling voice\n"); 849 if (hw->poll_mode) { 850 hw->poll_mode = 0; 851 alsa_fini_poll(&alsa->pollhlp); 852 } 853 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE); 854 } 855 } 856 857 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo) 858 { 859 if (!apdo->has_try_poll) { 860 apdo->try_poll = true; 861 apdo->has_try_poll = true; 862 } 863 } 864 865 static void *alsa_audio_init(Audiodev *dev) 866 { 867 AudiodevAlsaOptions *aopts; 868 assert(dev->driver == AUDIODEV_DRIVER_ALSA); 869 870 aopts = &dev->u.alsa; 871 alsa_init_per_direction(aopts->in); 872 alsa_init_per_direction(aopts->out); 873 874 /* 875 * need to define them, as otherwise alsa produces no sound 876 * doesn't set has_* so alsa_open can identify it wasn't set by the user 877 */ 878 if (!dev->u.alsa.out->has_period_length) { 879 /* 1024 frames assuming 44100Hz */ 880 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100; 881 } 882 if (!dev->u.alsa.out->has_buffer_length) { 883 /* 4096 frames assuming 44100Hz */ 884 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100; 885 } 886 887 /* 888 * OptsVisitor sets unspecified optional fields to zero, but do not depend 889 * on it... 890 */ 891 if (!dev->u.alsa.in->has_period_length) { 892 dev->u.alsa.in->period_length = 0; 893 } 894 if (!dev->u.alsa.in->has_buffer_length) { 895 dev->u.alsa.in->buffer_length = 0; 896 } 897 898 return dev; 899 } 900 901 static void alsa_audio_fini (void *opaque) 902 { 903 } 904 905 static struct audio_pcm_ops alsa_pcm_ops = { 906 .init_out = alsa_init_out, 907 .fini_out = alsa_fini_out, 908 .write = alsa_write, 909 .enable_out = alsa_enable_out, 910 911 .init_in = alsa_init_in, 912 .fini_in = alsa_fini_in, 913 .read = alsa_read, 914 .enable_in = alsa_enable_in, 915 }; 916 917 static struct audio_driver alsa_audio_driver = { 918 .name = "alsa", 919 .descr = "ALSA http://www.alsa-project.org", 920 .init = alsa_audio_init, 921 .fini = alsa_audio_fini, 922 .pcm_ops = &alsa_pcm_ops, 923 .can_be_default = 1, 924 .max_voices_out = INT_MAX, 925 .max_voices_in = INT_MAX, 926 .voice_size_out = sizeof (ALSAVoiceOut), 927 .voice_size_in = sizeof (ALSAVoiceIn) 928 }; 929 930 static void register_audio_alsa(void) 931 { 932 audio_driver_register(&alsa_audio_driver); 933 } 934 type_init(register_audio_alsa); 935