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