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 (nchannels != 1 && nchannels != 2) { 497 alsa_logerr2 (err, typ, 498 "Can not handle obtained number of channels %d\n", 499 nchannels); 500 goto err; 501 } 502 503 if (apdo->buffer_length) { 504 int dir = 0; 505 unsigned int btime = apdo->buffer_length; 506 507 err = snd_pcm_hw_params_set_buffer_time_near( 508 handle, hw_params, &btime, &dir); 509 510 if (err < 0) { 511 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n", 512 apdo->buffer_length); 513 goto err; 514 } 515 516 if (apdo->has_buffer_length && btime != apdo->buffer_length) { 517 dolog("Requested buffer time %" PRId32 518 " was rejected, using %u\n", apdo->buffer_length, btime); 519 } 520 } 521 522 if (apdo->period_length) { 523 int dir = 0; 524 unsigned int ptime = apdo->period_length; 525 526 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime, 527 &dir); 528 529 if (err < 0) { 530 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n", 531 apdo->period_length); 532 goto err; 533 } 534 535 if (apdo->has_period_length && ptime != apdo->period_length) { 536 dolog("Requested period time %" PRId32 " was rejected, using %d\n", 537 apdo->period_length, ptime); 538 } 539 } 540 541 err = snd_pcm_hw_params (handle, hw_params); 542 if (err < 0) { 543 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); 544 goto err; 545 } 546 547 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); 548 if (err < 0) { 549 alsa_logerr2 (err, typ, "Failed to get buffer size\n"); 550 goto err; 551 } 552 553 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); 554 if (err < 0) { 555 alsa_logerr2 (err, typ, "Failed to get format\n"); 556 goto err; 557 } 558 559 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { 560 dolog ("Invalid format was returned %d\n", obtfmt); 561 goto err; 562 } 563 564 err = snd_pcm_prepare (handle); 565 if (err < 0) { 566 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); 567 goto err; 568 } 569 570 if (!in && aopts->has_threshold && aopts->threshold) { 571 struct audsettings as = { .freq = freq }; 572 alsa_set_threshold( 573 handle, 574 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo), 575 &as, aopts->threshold)); 576 } 577 578 obt->nchannels = nchannels; 579 obt->freq = freq; 580 obt->samples = obt_buffer_size; 581 582 *handlep = handle; 583 584 if (obtfmt != req->fmt || 585 obt->nchannels != req->nchannels || 586 obt->freq != req->freq) { 587 dolog ("Audio parameters for %s\n", typ); 588 alsa_dump_info(req, obt, obtfmt, apdo); 589 } 590 591 #ifdef DEBUG 592 alsa_dump_info(req, obt, obtfmt, pdo); 593 #endif 594 return 0; 595 596 err: 597 alsa_anal_close1 (&handle); 598 return -1; 599 } 600 601 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) 602 { 603 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 604 size_t pos = 0; 605 size_t len_frames = len >> hw->info.shift; 606 607 while (len_frames) { 608 char *src = advance(buf, pos); 609 snd_pcm_sframes_t written; 610 611 written = snd_pcm_writei(alsa->handle, src, len_frames); 612 613 if (written <= 0) { 614 switch (written) { 615 case 0: 616 trace_alsa_wrote_zero(len_frames); 617 return pos; 618 619 case -EPIPE: 620 if (alsa_recover(alsa->handle)) { 621 alsa_logerr(written, "Failed to write %zu frames\n", 622 len_frames); 623 return pos; 624 } 625 trace_alsa_xrun_out(); 626 continue; 627 628 case -ESTRPIPE: 629 /* 630 * stream is suspended and waiting for an application 631 * recovery 632 */ 633 if (alsa_resume(alsa->handle)) { 634 alsa_logerr(written, "Failed to write %zu frames\n", 635 len_frames); 636 return pos; 637 } 638 trace_alsa_resume_out(); 639 continue; 640 641 case -EAGAIN: 642 return pos; 643 644 default: 645 alsa_logerr(written, "Failed to write %zu frames from %p\n", 646 len, src); 647 return pos; 648 } 649 } 650 651 pos += written << hw->info.shift; 652 if (written < len_frames) { 653 break; 654 } 655 len_frames -= written; 656 } 657 658 return pos; 659 } 660 661 static void alsa_fini_out (HWVoiceOut *hw) 662 { 663 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 664 665 ldebug ("alsa_fini\n"); 666 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 667 } 668 669 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, 670 void *drv_opaque) 671 { 672 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 673 struct alsa_params_req req; 674 struct alsa_params_obt obt; 675 snd_pcm_t *handle; 676 struct audsettings obt_as; 677 Audiodev *dev = drv_opaque; 678 679 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 680 req.freq = as->freq; 681 req.nchannels = as->nchannels; 682 683 if (alsa_open(0, &req, &obt, &handle, dev)) { 684 return -1; 685 } 686 687 obt_as.freq = obt.freq; 688 obt_as.nchannels = obt.nchannels; 689 obt_as.fmt = obt.fmt; 690 obt_as.endianness = obt.endianness; 691 692 audio_pcm_init_info (&hw->info, &obt_as); 693 hw->samples = obt.samples; 694 695 alsa->pollhlp.s = hw->s; 696 alsa->handle = handle; 697 alsa->dev = dev; 698 return 0; 699 } 700 701 #define VOICE_CTL_PAUSE 0 702 #define VOICE_CTL_PREPARE 1 703 #define VOICE_CTL_START 2 704 705 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) 706 { 707 int err; 708 709 if (ctl == VOICE_CTL_PAUSE) { 710 err = snd_pcm_drop (handle); 711 if (err < 0) { 712 alsa_logerr (err, "Could not stop %s\n", typ); 713 return -1; 714 } 715 } 716 else { 717 err = snd_pcm_prepare (handle); 718 if (err < 0) { 719 alsa_logerr (err, "Could not prepare handle for %s\n", typ); 720 return -1; 721 } 722 if (ctl == VOICE_CTL_START) { 723 err = snd_pcm_start(handle); 724 if (err < 0) { 725 alsa_logerr (err, "Could not start handle for %s\n", typ); 726 return -1; 727 } 728 } 729 } 730 731 return 0; 732 } 733 734 static void alsa_enable_out(HWVoiceOut *hw, bool enable) 735 { 736 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 737 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out; 738 739 if (enable) { 740 bool poll_mode = apdo->try_poll; 741 742 ldebug("enabling voice\n"); 743 if (poll_mode && alsa_poll_out(hw)) { 744 poll_mode = 0; 745 } 746 hw->poll_mode = poll_mode; 747 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE); 748 } else { 749 ldebug("disabling voice\n"); 750 if (hw->poll_mode) { 751 hw->poll_mode = 0; 752 alsa_fini_poll(&alsa->pollhlp); 753 } 754 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE); 755 } 756 } 757 758 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) 759 { 760 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 761 struct alsa_params_req req; 762 struct alsa_params_obt obt; 763 snd_pcm_t *handle; 764 struct audsettings obt_as; 765 Audiodev *dev = drv_opaque; 766 767 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 768 req.freq = as->freq; 769 req.nchannels = as->nchannels; 770 771 if (alsa_open(1, &req, &obt, &handle, dev)) { 772 return -1; 773 } 774 775 obt_as.freq = obt.freq; 776 obt_as.nchannels = obt.nchannels; 777 obt_as.fmt = obt.fmt; 778 obt_as.endianness = obt.endianness; 779 780 audio_pcm_init_info (&hw->info, &obt_as); 781 hw->samples = obt.samples; 782 783 alsa->pollhlp.s = hw->s; 784 alsa->handle = handle; 785 alsa->dev = dev; 786 return 0; 787 } 788 789 static void alsa_fini_in (HWVoiceIn *hw) 790 { 791 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 792 793 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 794 } 795 796 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) 797 { 798 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 799 size_t pos = 0; 800 801 while (len) { 802 void *dst = advance(buf, pos); 803 snd_pcm_sframes_t nread; 804 805 nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift); 806 807 if (nread <= 0) { 808 switch (nread) { 809 case 0: 810 trace_alsa_read_zero(len); 811 return pos;; 812 813 case -EPIPE: 814 if (alsa_recover(alsa->handle)) { 815 alsa_logerr(nread, "Failed to read %zu frames\n", len); 816 return pos; 817 } 818 trace_alsa_xrun_in(); 819 continue; 820 821 case -EAGAIN: 822 return pos; 823 824 default: 825 alsa_logerr(nread, "Failed to read %zu frames to %p\n", 826 len, dst); 827 return pos;; 828 } 829 } 830 831 pos += nread << hw->info.shift; 832 len -= nread << hw->info.shift; 833 } 834 835 return pos; 836 } 837 838 static void alsa_enable_in(HWVoiceIn *hw, bool enable) 839 { 840 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 841 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in; 842 843 if (enable) { 844 bool poll_mode = apdo->try_poll; 845 846 ldebug("enabling voice\n"); 847 if (poll_mode && alsa_poll_in(hw)) { 848 poll_mode = 0; 849 } 850 hw->poll_mode = poll_mode; 851 852 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START); 853 } else { 854 ldebug ("disabling voice\n"); 855 if (hw->poll_mode) { 856 hw->poll_mode = 0; 857 alsa_fini_poll(&alsa->pollhlp); 858 } 859 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE); 860 } 861 } 862 863 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo) 864 { 865 if (!apdo->has_try_poll) { 866 apdo->try_poll = true; 867 apdo->has_try_poll = true; 868 } 869 } 870 871 static void *alsa_audio_init(Audiodev *dev) 872 { 873 AudiodevAlsaOptions *aopts; 874 assert(dev->driver == AUDIODEV_DRIVER_ALSA); 875 876 aopts = &dev->u.alsa; 877 alsa_init_per_direction(aopts->in); 878 alsa_init_per_direction(aopts->out); 879 880 /* 881 * need to define them, as otherwise alsa produces no sound 882 * doesn't set has_* so alsa_open can identify it wasn't set by the user 883 */ 884 if (!dev->u.alsa.out->has_period_length) { 885 /* 1024 frames assuming 44100Hz */ 886 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100; 887 } 888 if (!dev->u.alsa.out->has_buffer_length) { 889 /* 4096 frames assuming 44100Hz */ 890 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100; 891 } 892 893 /* 894 * OptsVisitor sets unspecified optional fields to zero, but do not depend 895 * on it... 896 */ 897 if (!dev->u.alsa.in->has_period_length) { 898 dev->u.alsa.in->period_length = 0; 899 } 900 if (!dev->u.alsa.in->has_buffer_length) { 901 dev->u.alsa.in->buffer_length = 0; 902 } 903 904 return dev; 905 } 906 907 static void alsa_audio_fini (void *opaque) 908 { 909 } 910 911 static struct audio_pcm_ops alsa_pcm_ops = { 912 .init_out = alsa_init_out, 913 .fini_out = alsa_fini_out, 914 .write = alsa_write, 915 .enable_out = alsa_enable_out, 916 917 .init_in = alsa_init_in, 918 .fini_in = alsa_fini_in, 919 .read = alsa_read, 920 .enable_in = alsa_enable_in, 921 }; 922 923 static struct audio_driver alsa_audio_driver = { 924 .name = "alsa", 925 .descr = "ALSA http://www.alsa-project.org", 926 .init = alsa_audio_init, 927 .fini = alsa_audio_fini, 928 .pcm_ops = &alsa_pcm_ops, 929 .can_be_default = 1, 930 .max_voices_out = INT_MAX, 931 .max_voices_in = INT_MAX, 932 .voice_size_out = sizeof (ALSAVoiceOut), 933 .voice_size_in = sizeof (ALSAVoiceIn) 934 }; 935 936 static void register_audio_alsa(void) 937 { 938 audio_driver_register(&alsa_audio_driver); 939 } 940 type_init(register_audio_alsa); 941