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 } else { 282 return SND_PCM_FORMAT_S16_LE; 283 } 284 285 case AUDIO_FORMAT_U16: 286 if (endianness) { 287 return SND_PCM_FORMAT_U16_BE; 288 } else { 289 return SND_PCM_FORMAT_U16_LE; 290 } 291 292 case AUDIO_FORMAT_S32: 293 if (endianness) { 294 return SND_PCM_FORMAT_S32_BE; 295 } else { 296 return SND_PCM_FORMAT_S32_LE; 297 } 298 299 case AUDIO_FORMAT_U32: 300 if (endianness) { 301 return SND_PCM_FORMAT_U32_BE; 302 } else { 303 return SND_PCM_FORMAT_U32_LE; 304 } 305 306 case AUDIO_FORMAT_F32: 307 if (endianness) { 308 return SND_PCM_FORMAT_FLOAT_BE; 309 } else { 310 return SND_PCM_FORMAT_FLOAT_LE; 311 } 312 313 default: 314 dolog ("Internal logic error: Bad audio format %d\n", fmt); 315 #ifdef DEBUG_AUDIO 316 abort (); 317 #endif 318 return SND_PCM_FORMAT_U8; 319 } 320 } 321 322 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt, 323 int *endianness) 324 { 325 switch (alsafmt) { 326 case SND_PCM_FORMAT_S8: 327 *endianness = 0; 328 *fmt = AUDIO_FORMAT_S8; 329 break; 330 331 case SND_PCM_FORMAT_U8: 332 *endianness = 0; 333 *fmt = AUDIO_FORMAT_U8; 334 break; 335 336 case SND_PCM_FORMAT_S16_LE: 337 *endianness = 0; 338 *fmt = AUDIO_FORMAT_S16; 339 break; 340 341 case SND_PCM_FORMAT_U16_LE: 342 *endianness = 0; 343 *fmt = AUDIO_FORMAT_U16; 344 break; 345 346 case SND_PCM_FORMAT_S16_BE: 347 *endianness = 1; 348 *fmt = AUDIO_FORMAT_S16; 349 break; 350 351 case SND_PCM_FORMAT_U16_BE: 352 *endianness = 1; 353 *fmt = AUDIO_FORMAT_U16; 354 break; 355 356 case SND_PCM_FORMAT_S32_LE: 357 *endianness = 0; 358 *fmt = AUDIO_FORMAT_S32; 359 break; 360 361 case SND_PCM_FORMAT_U32_LE: 362 *endianness = 0; 363 *fmt = AUDIO_FORMAT_U32; 364 break; 365 366 case SND_PCM_FORMAT_S32_BE: 367 *endianness = 1; 368 *fmt = AUDIO_FORMAT_S32; 369 break; 370 371 case SND_PCM_FORMAT_U32_BE: 372 *endianness = 1; 373 *fmt = AUDIO_FORMAT_U32; 374 break; 375 376 case SND_PCM_FORMAT_FLOAT_LE: 377 *endianness = 0; 378 *fmt = AUDIO_FORMAT_F32; 379 break; 380 381 case SND_PCM_FORMAT_FLOAT_BE: 382 *endianness = 1; 383 *fmt = AUDIO_FORMAT_F32; 384 break; 385 386 default: 387 dolog ("Unrecognized audio format %d\n", alsafmt); 388 return -1; 389 } 390 391 return 0; 392 } 393 394 static void alsa_dump_info (struct alsa_params_req *req, 395 struct alsa_params_obt *obt, 396 snd_pcm_format_t obtfmt, 397 AudiodevAlsaPerDirectionOptions *apdo) 398 { 399 dolog("parameter | requested value | obtained value\n"); 400 dolog("format | %10d | %10d\n", req->fmt, obtfmt); 401 dolog("channels | %10d | %10d\n", 402 req->nchannels, obt->nchannels); 403 dolog("frequency | %10d | %10d\n", req->freq, obt->freq); 404 dolog("============================================\n"); 405 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n", 406 apdo->buffer_length, apdo->period_length); 407 dolog("obtained: samples %ld\n", obt->samples); 408 } 409 410 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) 411 { 412 int err; 413 snd_pcm_sw_params_t *sw_params; 414 415 snd_pcm_sw_params_alloca (&sw_params); 416 417 err = snd_pcm_sw_params_current (handle, sw_params); 418 if (err < 0) { 419 dolog ("Could not fully initialize DAC\n"); 420 alsa_logerr (err, "Failed to get current software parameters\n"); 421 return; 422 } 423 424 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); 425 if (err < 0) { 426 dolog ("Could not fully initialize DAC\n"); 427 alsa_logerr (err, "Failed to set software threshold to %ld\n", 428 threshold); 429 return; 430 } 431 432 err = snd_pcm_sw_params (handle, sw_params); 433 if (err < 0) { 434 dolog ("Could not fully initialize DAC\n"); 435 alsa_logerr (err, "Failed to set software parameters\n"); 436 return; 437 } 438 } 439 440 static int alsa_open(bool in, struct alsa_params_req *req, 441 struct alsa_params_obt *obt, snd_pcm_t **handlep, 442 Audiodev *dev) 443 { 444 AudiodevAlsaOptions *aopts = &dev->u.alsa; 445 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out; 446 snd_pcm_t *handle; 447 snd_pcm_hw_params_t *hw_params; 448 int err; 449 unsigned int freq, nchannels; 450 const char *pcm_name = apdo->has_dev ? apdo->dev : "default"; 451 snd_pcm_uframes_t obt_buffer_size; 452 const char *typ = in ? "ADC" : "DAC"; 453 snd_pcm_format_t obtfmt; 454 455 freq = req->freq; 456 nchannels = req->nchannels; 457 458 snd_pcm_hw_params_alloca (&hw_params); 459 460 err = snd_pcm_open ( 461 &handle, 462 pcm_name, 463 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 464 SND_PCM_NONBLOCK 465 ); 466 if (err < 0) { 467 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); 468 return -1; 469 } 470 471 err = snd_pcm_hw_params_any (handle, hw_params); 472 if (err < 0) { 473 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); 474 goto err; 475 } 476 477 err = snd_pcm_hw_params_set_access ( 478 handle, 479 hw_params, 480 SND_PCM_ACCESS_RW_INTERLEAVED 481 ); 482 if (err < 0) { 483 alsa_logerr2 (err, typ, "Failed to set access type\n"); 484 goto err; 485 } 486 487 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); 488 if (err < 0) { 489 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); 490 } 491 492 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); 493 if (err < 0) { 494 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); 495 goto err; 496 } 497 498 err = snd_pcm_hw_params_set_channels_near ( 499 handle, 500 hw_params, 501 &nchannels 502 ); 503 if (err < 0) { 504 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", 505 req->nchannels); 506 goto err; 507 } 508 509 if (apdo->buffer_length) { 510 int dir = 0; 511 unsigned int btime = apdo->buffer_length; 512 513 err = snd_pcm_hw_params_set_buffer_time_near( 514 handle, hw_params, &btime, &dir); 515 516 if (err < 0) { 517 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n", 518 apdo->buffer_length); 519 goto err; 520 } 521 522 if (apdo->has_buffer_length && btime != apdo->buffer_length) { 523 dolog("Requested buffer time %" PRId32 524 " was rejected, using %u\n", apdo->buffer_length, btime); 525 } 526 } 527 528 if (apdo->period_length) { 529 int dir = 0; 530 unsigned int ptime = apdo->period_length; 531 532 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime, 533 &dir); 534 535 if (err < 0) { 536 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n", 537 apdo->period_length); 538 goto err; 539 } 540 541 if (apdo->has_period_length && ptime != apdo->period_length) { 542 dolog("Requested period time %" PRId32 " was rejected, using %d\n", 543 apdo->period_length, ptime); 544 } 545 } 546 547 err = snd_pcm_hw_params (handle, hw_params); 548 if (err < 0) { 549 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); 550 goto err; 551 } 552 553 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); 554 if (err < 0) { 555 alsa_logerr2 (err, typ, "Failed to get buffer size\n"); 556 goto err; 557 } 558 559 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); 560 if (err < 0) { 561 alsa_logerr2 (err, typ, "Failed to get format\n"); 562 goto err; 563 } 564 565 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { 566 dolog ("Invalid format was returned %d\n", obtfmt); 567 goto err; 568 } 569 570 err = snd_pcm_prepare (handle); 571 if (err < 0) { 572 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); 573 goto err; 574 } 575 576 if (!in && aopts->has_threshold && aopts->threshold) { 577 struct audsettings as = { .freq = freq }; 578 alsa_set_threshold( 579 handle, 580 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo), 581 &as, aopts->threshold)); 582 } 583 584 obt->nchannels = nchannels; 585 obt->freq = freq; 586 obt->samples = obt_buffer_size; 587 588 *handlep = handle; 589 590 if (obtfmt != req->fmt || 591 obt->nchannels != req->nchannels || 592 obt->freq != req->freq) { 593 dolog ("Audio parameters for %s\n", typ); 594 alsa_dump_info(req, obt, obtfmt, apdo); 595 } 596 597 #ifdef DEBUG 598 alsa_dump_info(req, obt, obtfmt, apdo); 599 #endif 600 return 0; 601 602 err: 603 alsa_anal_close1 (&handle); 604 return -1; 605 } 606 607 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) 608 { 609 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 610 size_t pos = 0; 611 size_t len_frames = len / hw->info.bytes_per_frame; 612 613 while (len_frames) { 614 char *src = advance(buf, pos); 615 snd_pcm_sframes_t written; 616 617 written = snd_pcm_writei(alsa->handle, src, len_frames); 618 619 if (written <= 0) { 620 switch (written) { 621 case 0: 622 trace_alsa_wrote_zero(len_frames); 623 return pos; 624 625 case -EPIPE: 626 if (alsa_recover(alsa->handle)) { 627 alsa_logerr(written, "Failed to write %zu frames\n", 628 len_frames); 629 return pos; 630 } 631 trace_alsa_xrun_out(); 632 continue; 633 634 case -ESTRPIPE: 635 /* 636 * stream is suspended and waiting for an application 637 * recovery 638 */ 639 if (alsa_resume(alsa->handle)) { 640 alsa_logerr(written, "Failed to write %zu frames\n", 641 len_frames); 642 return pos; 643 } 644 trace_alsa_resume_out(); 645 continue; 646 647 case -EAGAIN: 648 return pos; 649 650 default: 651 alsa_logerr(written, "Failed to write %zu frames from %p\n", 652 len, src); 653 return pos; 654 } 655 } 656 657 pos += written * hw->info.bytes_per_frame; 658 if (written < len_frames) { 659 break; 660 } 661 len_frames -= written; 662 } 663 664 return pos; 665 } 666 667 static void alsa_fini_out (HWVoiceOut *hw) 668 { 669 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 670 671 ldebug ("alsa_fini\n"); 672 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 673 } 674 675 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, 676 void *drv_opaque) 677 { 678 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 679 struct alsa_params_req req; 680 struct alsa_params_obt obt; 681 snd_pcm_t *handle; 682 struct audsettings obt_as; 683 Audiodev *dev = drv_opaque; 684 685 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 686 req.freq = as->freq; 687 req.nchannels = as->nchannels; 688 689 if (alsa_open(0, &req, &obt, &handle, dev)) { 690 return -1; 691 } 692 693 obt_as.freq = obt.freq; 694 obt_as.nchannels = obt.nchannels; 695 obt_as.fmt = obt.fmt; 696 obt_as.endianness = obt.endianness; 697 698 audio_pcm_init_info (&hw->info, &obt_as); 699 hw->samples = obt.samples; 700 701 alsa->pollhlp.s = hw->s; 702 alsa->handle = handle; 703 alsa->dev = dev; 704 return 0; 705 } 706 707 #define VOICE_CTL_PAUSE 0 708 #define VOICE_CTL_PREPARE 1 709 #define VOICE_CTL_START 2 710 711 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) 712 { 713 int err; 714 715 if (ctl == VOICE_CTL_PAUSE) { 716 err = snd_pcm_drop (handle); 717 if (err < 0) { 718 alsa_logerr (err, "Could not stop %s\n", typ); 719 return -1; 720 } 721 } else { 722 err = snd_pcm_prepare (handle); 723 if (err < 0) { 724 alsa_logerr (err, "Could not prepare handle for %s\n", typ); 725 return -1; 726 } 727 if (ctl == VOICE_CTL_START) { 728 err = snd_pcm_start(handle); 729 if (err < 0) { 730 alsa_logerr (err, "Could not start handle for %s\n", typ); 731 return -1; 732 } 733 } 734 } 735 736 return 0; 737 } 738 739 static void alsa_enable_out(HWVoiceOut *hw, bool enable) 740 { 741 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 742 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out; 743 744 if (enable) { 745 bool poll_mode = apdo->try_poll; 746 747 ldebug("enabling voice\n"); 748 if (poll_mode && alsa_poll_out(hw)) { 749 poll_mode = 0; 750 } 751 hw->poll_mode = poll_mode; 752 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE); 753 } else { 754 ldebug("disabling voice\n"); 755 if (hw->poll_mode) { 756 hw->poll_mode = 0; 757 alsa_fini_poll(&alsa->pollhlp); 758 } 759 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE); 760 } 761 } 762 763 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) 764 { 765 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 766 struct alsa_params_req req; 767 struct alsa_params_obt obt; 768 snd_pcm_t *handle; 769 struct audsettings obt_as; 770 Audiodev *dev = drv_opaque; 771 772 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 773 req.freq = as->freq; 774 req.nchannels = as->nchannels; 775 776 if (alsa_open(1, &req, &obt, &handle, dev)) { 777 return -1; 778 } 779 780 obt_as.freq = obt.freq; 781 obt_as.nchannels = obt.nchannels; 782 obt_as.fmt = obt.fmt; 783 obt_as.endianness = obt.endianness; 784 785 audio_pcm_init_info (&hw->info, &obt_as); 786 hw->samples = obt.samples; 787 788 alsa->pollhlp.s = hw->s; 789 alsa->handle = handle; 790 alsa->dev = dev; 791 return 0; 792 } 793 794 static void alsa_fini_in (HWVoiceIn *hw) 795 { 796 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 797 798 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 799 } 800 801 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) 802 { 803 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 804 size_t pos = 0; 805 806 while (len) { 807 void *dst = advance(buf, pos); 808 snd_pcm_sframes_t nread; 809 810 nread = snd_pcm_readi( 811 alsa->handle, dst, len / hw->info.bytes_per_frame); 812 813 if (nread <= 0) { 814 switch (nread) { 815 case 0: 816 trace_alsa_read_zero(len); 817 return pos; 818 819 case -EPIPE: 820 if (alsa_recover(alsa->handle)) { 821 alsa_logerr(nread, "Failed to read %zu frames\n", len); 822 return pos; 823 } 824 trace_alsa_xrun_in(); 825 continue; 826 827 case -EAGAIN: 828 return pos; 829 830 default: 831 alsa_logerr(nread, "Failed to read %zu frames to %p\n", 832 len, dst); 833 return pos; 834 } 835 } 836 837 pos += nread * hw->info.bytes_per_frame; 838 len -= nread * hw->info.bytes_per_frame; 839 } 840 841 return pos; 842 } 843 844 static void alsa_enable_in(HWVoiceIn *hw, bool enable) 845 { 846 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 847 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in; 848 849 if (enable) { 850 bool poll_mode = apdo->try_poll; 851 852 ldebug("enabling voice\n"); 853 if (poll_mode && alsa_poll_in(hw)) { 854 poll_mode = 0; 855 } 856 hw->poll_mode = poll_mode; 857 858 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START); 859 } else { 860 ldebug ("disabling voice\n"); 861 if (hw->poll_mode) { 862 hw->poll_mode = 0; 863 alsa_fini_poll(&alsa->pollhlp); 864 } 865 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE); 866 } 867 } 868 869 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo) 870 { 871 if (!apdo->has_try_poll) { 872 apdo->try_poll = true; 873 apdo->has_try_poll = true; 874 } 875 } 876 877 static void *alsa_audio_init(Audiodev *dev) 878 { 879 AudiodevAlsaOptions *aopts; 880 assert(dev->driver == AUDIODEV_DRIVER_ALSA); 881 882 aopts = &dev->u.alsa; 883 alsa_init_per_direction(aopts->in); 884 alsa_init_per_direction(aopts->out); 885 886 /* 887 * need to define them, as otherwise alsa produces no sound 888 * doesn't set has_* so alsa_open can identify it wasn't set by the user 889 */ 890 if (!dev->u.alsa.out->has_period_length) { 891 /* 1024 frames assuming 44100Hz */ 892 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100; 893 } 894 if (!dev->u.alsa.out->has_buffer_length) { 895 /* 4096 frames assuming 44100Hz */ 896 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100; 897 } 898 899 /* 900 * OptsVisitor sets unspecified optional fields to zero, but do not depend 901 * on it... 902 */ 903 if (!dev->u.alsa.in->has_period_length) { 904 dev->u.alsa.in->period_length = 0; 905 } 906 if (!dev->u.alsa.in->has_buffer_length) { 907 dev->u.alsa.in->buffer_length = 0; 908 } 909 910 return dev; 911 } 912 913 static void alsa_audio_fini (void *opaque) 914 { 915 } 916 917 static struct audio_pcm_ops alsa_pcm_ops = { 918 .init_out = alsa_init_out, 919 .fini_out = alsa_fini_out, 920 .write = alsa_write, 921 .run_buffer_out = audio_generic_run_buffer_out, 922 .enable_out = alsa_enable_out, 923 924 .init_in = alsa_init_in, 925 .fini_in = alsa_fini_in, 926 .read = alsa_read, 927 .run_buffer_in = audio_generic_run_buffer_in, 928 .enable_in = alsa_enable_in, 929 }; 930 931 static struct audio_driver alsa_audio_driver = { 932 .name = "alsa", 933 .descr = "ALSA http://www.alsa-project.org", 934 .init = alsa_audio_init, 935 .fini = alsa_audio_fini, 936 .pcm_ops = &alsa_pcm_ops, 937 .can_be_default = 1, 938 .max_voices_out = INT_MAX, 939 .max_voices_in = INT_MAX, 940 .voice_size_out = sizeof (ALSAVoiceOut), 941 .voice_size_in = sizeof (ALSAVoiceIn) 942 }; 943 944 static void register_audio_alsa(void) 945 { 946 audio_driver_register(&alsa_audio_driver); 947 } 948 type_init(register_audio_alsa); 949