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 case AUDIO_FORMAT_F32: 311 if (endianness) { 312 return SND_PCM_FORMAT_FLOAT_BE; 313 } else { 314 return SND_PCM_FORMAT_FLOAT_LE; 315 } 316 317 default: 318 dolog ("Internal logic error: Bad audio format %d\n", fmt); 319 #ifdef DEBUG_AUDIO 320 abort (); 321 #endif 322 return SND_PCM_FORMAT_U8; 323 } 324 } 325 326 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt, 327 int *endianness) 328 { 329 switch (alsafmt) { 330 case SND_PCM_FORMAT_S8: 331 *endianness = 0; 332 *fmt = AUDIO_FORMAT_S8; 333 break; 334 335 case SND_PCM_FORMAT_U8: 336 *endianness = 0; 337 *fmt = AUDIO_FORMAT_U8; 338 break; 339 340 case SND_PCM_FORMAT_S16_LE: 341 *endianness = 0; 342 *fmt = AUDIO_FORMAT_S16; 343 break; 344 345 case SND_PCM_FORMAT_U16_LE: 346 *endianness = 0; 347 *fmt = AUDIO_FORMAT_U16; 348 break; 349 350 case SND_PCM_FORMAT_S16_BE: 351 *endianness = 1; 352 *fmt = AUDIO_FORMAT_S16; 353 break; 354 355 case SND_PCM_FORMAT_U16_BE: 356 *endianness = 1; 357 *fmt = AUDIO_FORMAT_U16; 358 break; 359 360 case SND_PCM_FORMAT_S32_LE: 361 *endianness = 0; 362 *fmt = AUDIO_FORMAT_S32; 363 break; 364 365 case SND_PCM_FORMAT_U32_LE: 366 *endianness = 0; 367 *fmt = AUDIO_FORMAT_U32; 368 break; 369 370 case SND_PCM_FORMAT_S32_BE: 371 *endianness = 1; 372 *fmt = AUDIO_FORMAT_S32; 373 break; 374 375 case SND_PCM_FORMAT_U32_BE: 376 *endianness = 1; 377 *fmt = AUDIO_FORMAT_U32; 378 break; 379 380 case SND_PCM_FORMAT_FLOAT_LE: 381 *endianness = 0; 382 *fmt = AUDIO_FORMAT_F32; 383 break; 384 385 case SND_PCM_FORMAT_FLOAT_BE: 386 *endianness = 1; 387 *fmt = AUDIO_FORMAT_F32; 388 break; 389 390 default: 391 dolog ("Unrecognized audio format %d\n", alsafmt); 392 return -1; 393 } 394 395 return 0; 396 } 397 398 static void alsa_dump_info (struct alsa_params_req *req, 399 struct alsa_params_obt *obt, 400 snd_pcm_format_t obtfmt, 401 AudiodevAlsaPerDirectionOptions *apdo) 402 { 403 dolog("parameter | requested value | obtained value\n"); 404 dolog("format | %10d | %10d\n", req->fmt, obtfmt); 405 dolog("channels | %10d | %10d\n", 406 req->nchannels, obt->nchannels); 407 dolog("frequency | %10d | %10d\n", req->freq, obt->freq); 408 dolog("============================================\n"); 409 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n", 410 apdo->buffer_length, apdo->period_length); 411 dolog("obtained: samples %ld\n", obt->samples); 412 } 413 414 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) 415 { 416 int err; 417 snd_pcm_sw_params_t *sw_params; 418 419 snd_pcm_sw_params_alloca (&sw_params); 420 421 err = snd_pcm_sw_params_current (handle, sw_params); 422 if (err < 0) { 423 dolog ("Could not fully initialize DAC\n"); 424 alsa_logerr (err, "Failed to get current software parameters\n"); 425 return; 426 } 427 428 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); 429 if (err < 0) { 430 dolog ("Could not fully initialize DAC\n"); 431 alsa_logerr (err, "Failed to set software threshold to %ld\n", 432 threshold); 433 return; 434 } 435 436 err = snd_pcm_sw_params (handle, sw_params); 437 if (err < 0) { 438 dolog ("Could not fully initialize DAC\n"); 439 alsa_logerr (err, "Failed to set software parameters\n"); 440 return; 441 } 442 } 443 444 static int alsa_open(bool in, struct alsa_params_req *req, 445 struct alsa_params_obt *obt, snd_pcm_t **handlep, 446 Audiodev *dev) 447 { 448 AudiodevAlsaOptions *aopts = &dev->u.alsa; 449 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out; 450 snd_pcm_t *handle; 451 snd_pcm_hw_params_t *hw_params; 452 int err; 453 unsigned int freq, nchannels; 454 const char *pcm_name = apdo->has_dev ? apdo->dev : "default"; 455 snd_pcm_uframes_t obt_buffer_size; 456 const char *typ = in ? "ADC" : "DAC"; 457 snd_pcm_format_t obtfmt; 458 459 freq = req->freq; 460 nchannels = req->nchannels; 461 462 snd_pcm_hw_params_alloca (&hw_params); 463 464 err = snd_pcm_open ( 465 &handle, 466 pcm_name, 467 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 468 SND_PCM_NONBLOCK 469 ); 470 if (err < 0) { 471 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); 472 return -1; 473 } 474 475 err = snd_pcm_hw_params_any (handle, hw_params); 476 if (err < 0) { 477 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); 478 goto err; 479 } 480 481 err = snd_pcm_hw_params_set_access ( 482 handle, 483 hw_params, 484 SND_PCM_ACCESS_RW_INTERLEAVED 485 ); 486 if (err < 0) { 487 alsa_logerr2 (err, typ, "Failed to set access type\n"); 488 goto err; 489 } 490 491 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); 492 if (err < 0) { 493 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); 494 } 495 496 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); 497 if (err < 0) { 498 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); 499 goto err; 500 } 501 502 err = snd_pcm_hw_params_set_channels_near ( 503 handle, 504 hw_params, 505 &nchannels 506 ); 507 if (err < 0) { 508 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", 509 req->nchannels); 510 goto err; 511 } 512 513 if (apdo->buffer_length) { 514 int dir = 0; 515 unsigned int btime = apdo->buffer_length; 516 517 err = snd_pcm_hw_params_set_buffer_time_near( 518 handle, hw_params, &btime, &dir); 519 520 if (err < 0) { 521 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n", 522 apdo->buffer_length); 523 goto err; 524 } 525 526 if (apdo->has_buffer_length && btime != apdo->buffer_length) { 527 dolog("Requested buffer time %" PRId32 528 " was rejected, using %u\n", apdo->buffer_length, btime); 529 } 530 } 531 532 if (apdo->period_length) { 533 int dir = 0; 534 unsigned int ptime = apdo->period_length; 535 536 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime, 537 &dir); 538 539 if (err < 0) { 540 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n", 541 apdo->period_length); 542 goto err; 543 } 544 545 if (apdo->has_period_length && ptime != apdo->period_length) { 546 dolog("Requested period time %" PRId32 " was rejected, using %d\n", 547 apdo->period_length, ptime); 548 } 549 } 550 551 err = snd_pcm_hw_params (handle, hw_params); 552 if (err < 0) { 553 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); 554 goto err; 555 } 556 557 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); 558 if (err < 0) { 559 alsa_logerr2 (err, typ, "Failed to get buffer size\n"); 560 goto err; 561 } 562 563 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); 564 if (err < 0) { 565 alsa_logerr2 (err, typ, "Failed to get format\n"); 566 goto err; 567 } 568 569 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { 570 dolog ("Invalid format was returned %d\n", obtfmt); 571 goto err; 572 } 573 574 err = snd_pcm_prepare (handle); 575 if (err < 0) { 576 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); 577 goto err; 578 } 579 580 if (!in && aopts->has_threshold && aopts->threshold) { 581 struct audsettings as = { .freq = freq }; 582 alsa_set_threshold( 583 handle, 584 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo), 585 &as, aopts->threshold)); 586 } 587 588 obt->nchannels = nchannels; 589 obt->freq = freq; 590 obt->samples = obt_buffer_size; 591 592 *handlep = handle; 593 594 if (obtfmt != req->fmt || 595 obt->nchannels != req->nchannels || 596 obt->freq != req->freq) { 597 dolog ("Audio parameters for %s\n", typ); 598 alsa_dump_info(req, obt, obtfmt, apdo); 599 } 600 601 #ifdef DEBUG 602 alsa_dump_info(req, obt, obtfmt, pdo); 603 #endif 604 return 0; 605 606 err: 607 alsa_anal_close1 (&handle); 608 return -1; 609 } 610 611 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) 612 { 613 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 614 size_t pos = 0; 615 size_t len_frames = len / hw->info.bytes_per_frame; 616 617 while (len_frames) { 618 char *src = advance(buf, pos); 619 snd_pcm_sframes_t written; 620 621 written = snd_pcm_writei(alsa->handle, src, len_frames); 622 623 if (written <= 0) { 624 switch (written) { 625 case 0: 626 trace_alsa_wrote_zero(len_frames); 627 return pos; 628 629 case -EPIPE: 630 if (alsa_recover(alsa->handle)) { 631 alsa_logerr(written, "Failed to write %zu frames\n", 632 len_frames); 633 return pos; 634 } 635 trace_alsa_xrun_out(); 636 continue; 637 638 case -ESTRPIPE: 639 /* 640 * stream is suspended and waiting for an application 641 * recovery 642 */ 643 if (alsa_resume(alsa->handle)) { 644 alsa_logerr(written, "Failed to write %zu frames\n", 645 len_frames); 646 return pos; 647 } 648 trace_alsa_resume_out(); 649 continue; 650 651 case -EAGAIN: 652 return pos; 653 654 default: 655 alsa_logerr(written, "Failed to write %zu frames from %p\n", 656 len, src); 657 return pos; 658 } 659 } 660 661 pos += written * hw->info.bytes_per_frame; 662 if (written < len_frames) { 663 break; 664 } 665 len_frames -= written; 666 } 667 668 return pos; 669 } 670 671 static void alsa_fini_out (HWVoiceOut *hw) 672 { 673 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 674 675 ldebug ("alsa_fini\n"); 676 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 677 } 678 679 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, 680 void *drv_opaque) 681 { 682 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 683 struct alsa_params_req req; 684 struct alsa_params_obt obt; 685 snd_pcm_t *handle; 686 struct audsettings obt_as; 687 Audiodev *dev = drv_opaque; 688 689 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 690 req.freq = as->freq; 691 req.nchannels = as->nchannels; 692 693 if (alsa_open(0, &req, &obt, &handle, dev)) { 694 return -1; 695 } 696 697 obt_as.freq = obt.freq; 698 obt_as.nchannels = obt.nchannels; 699 obt_as.fmt = obt.fmt; 700 obt_as.endianness = obt.endianness; 701 702 audio_pcm_init_info (&hw->info, &obt_as); 703 hw->samples = obt.samples; 704 705 alsa->pollhlp.s = hw->s; 706 alsa->handle = handle; 707 alsa->dev = dev; 708 return 0; 709 } 710 711 #define VOICE_CTL_PAUSE 0 712 #define VOICE_CTL_PREPARE 1 713 #define VOICE_CTL_START 2 714 715 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) 716 { 717 int err; 718 719 if (ctl == VOICE_CTL_PAUSE) { 720 err = snd_pcm_drop (handle); 721 if (err < 0) { 722 alsa_logerr (err, "Could not stop %s\n", typ); 723 return -1; 724 } 725 } 726 else { 727 err = snd_pcm_prepare (handle); 728 if (err < 0) { 729 alsa_logerr (err, "Could not prepare handle for %s\n", typ); 730 return -1; 731 } 732 if (ctl == VOICE_CTL_START) { 733 err = snd_pcm_start(handle); 734 if (err < 0) { 735 alsa_logerr (err, "Could not start handle for %s\n", typ); 736 return -1; 737 } 738 } 739 } 740 741 return 0; 742 } 743 744 static void alsa_enable_out(HWVoiceOut *hw, bool enable) 745 { 746 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 747 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out; 748 749 if (enable) { 750 bool poll_mode = apdo->try_poll; 751 752 ldebug("enabling voice\n"); 753 if (poll_mode && alsa_poll_out(hw)) { 754 poll_mode = 0; 755 } 756 hw->poll_mode = poll_mode; 757 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE); 758 } else { 759 ldebug("disabling voice\n"); 760 if (hw->poll_mode) { 761 hw->poll_mode = 0; 762 alsa_fini_poll(&alsa->pollhlp); 763 } 764 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE); 765 } 766 } 767 768 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) 769 { 770 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 771 struct alsa_params_req req; 772 struct alsa_params_obt obt; 773 snd_pcm_t *handle; 774 struct audsettings obt_as; 775 Audiodev *dev = drv_opaque; 776 777 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 778 req.freq = as->freq; 779 req.nchannels = as->nchannels; 780 781 if (alsa_open(1, &req, &obt, &handle, dev)) { 782 return -1; 783 } 784 785 obt_as.freq = obt.freq; 786 obt_as.nchannels = obt.nchannels; 787 obt_as.fmt = obt.fmt; 788 obt_as.endianness = obt.endianness; 789 790 audio_pcm_init_info (&hw->info, &obt_as); 791 hw->samples = obt.samples; 792 793 alsa->pollhlp.s = hw->s; 794 alsa->handle = handle; 795 alsa->dev = dev; 796 return 0; 797 } 798 799 static void alsa_fini_in (HWVoiceIn *hw) 800 { 801 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 802 803 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 804 } 805 806 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) 807 { 808 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 809 size_t pos = 0; 810 811 while (len) { 812 void *dst = advance(buf, pos); 813 snd_pcm_sframes_t nread; 814 815 nread = snd_pcm_readi( 816 alsa->handle, dst, len / hw->info.bytes_per_frame); 817 818 if (nread <= 0) { 819 switch (nread) { 820 case 0: 821 trace_alsa_read_zero(len); 822 return pos;; 823 824 case -EPIPE: 825 if (alsa_recover(alsa->handle)) { 826 alsa_logerr(nread, "Failed to read %zu frames\n", len); 827 return pos; 828 } 829 trace_alsa_xrun_in(); 830 continue; 831 832 case -EAGAIN: 833 return pos; 834 835 default: 836 alsa_logerr(nread, "Failed to read %zu frames to %p\n", 837 len, dst); 838 return pos;; 839 } 840 } 841 842 pos += nread * hw->info.bytes_per_frame; 843 len -= nread * hw->info.bytes_per_frame; 844 } 845 846 return pos; 847 } 848 849 static void alsa_enable_in(HWVoiceIn *hw, bool enable) 850 { 851 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 852 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in; 853 854 if (enable) { 855 bool poll_mode = apdo->try_poll; 856 857 ldebug("enabling voice\n"); 858 if (poll_mode && alsa_poll_in(hw)) { 859 poll_mode = 0; 860 } 861 hw->poll_mode = poll_mode; 862 863 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START); 864 } else { 865 ldebug ("disabling voice\n"); 866 if (hw->poll_mode) { 867 hw->poll_mode = 0; 868 alsa_fini_poll(&alsa->pollhlp); 869 } 870 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE); 871 } 872 } 873 874 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo) 875 { 876 if (!apdo->has_try_poll) { 877 apdo->try_poll = true; 878 apdo->has_try_poll = true; 879 } 880 } 881 882 static void *alsa_audio_init(Audiodev *dev) 883 { 884 AudiodevAlsaOptions *aopts; 885 assert(dev->driver == AUDIODEV_DRIVER_ALSA); 886 887 aopts = &dev->u.alsa; 888 alsa_init_per_direction(aopts->in); 889 alsa_init_per_direction(aopts->out); 890 891 /* 892 * need to define them, as otherwise alsa produces no sound 893 * doesn't set has_* so alsa_open can identify it wasn't set by the user 894 */ 895 if (!dev->u.alsa.out->has_period_length) { 896 /* 1024 frames assuming 44100Hz */ 897 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100; 898 } 899 if (!dev->u.alsa.out->has_buffer_length) { 900 /* 4096 frames assuming 44100Hz */ 901 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100; 902 } 903 904 /* 905 * OptsVisitor sets unspecified optional fields to zero, but do not depend 906 * on it... 907 */ 908 if (!dev->u.alsa.in->has_period_length) { 909 dev->u.alsa.in->period_length = 0; 910 } 911 if (!dev->u.alsa.in->has_buffer_length) { 912 dev->u.alsa.in->buffer_length = 0; 913 } 914 915 return dev; 916 } 917 918 static void alsa_audio_fini (void *opaque) 919 { 920 } 921 922 static struct audio_pcm_ops alsa_pcm_ops = { 923 .init_out = alsa_init_out, 924 .fini_out = alsa_fini_out, 925 .write = alsa_write, 926 .run_buffer_out = audio_generic_run_buffer_out, 927 .enable_out = alsa_enable_out, 928 929 .init_in = alsa_init_in, 930 .fini_in = alsa_fini_in, 931 .read = alsa_read, 932 .enable_in = alsa_enable_in, 933 }; 934 935 static struct audio_driver alsa_audio_driver = { 936 .name = "alsa", 937 .descr = "ALSA http://www.alsa-project.org", 938 .init = alsa_audio_init, 939 .fini = alsa_audio_fini, 940 .pcm_ops = &alsa_pcm_ops, 941 .can_be_default = 1, 942 .max_voices_out = INT_MAX, 943 .max_voices_in = INT_MAX, 944 .voice_size_out = sizeof (ALSAVoiceOut), 945 .voice_size_in = sizeof (ALSAVoiceIn) 946 }; 947 948 static void register_audio_alsa(void) 949 { 950 audio_driver_register(&alsa_audio_driver); 951 } 952 type_init(register_audio_alsa); 953