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