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 #pragma GCC diagnostic ignored "-Waddress" 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 Audiodev *dev; 51 } ALSAVoiceOut; 52 53 typedef struct ALSAVoiceIn { 54 HWVoiceIn hw; 55 snd_pcm_t *handle; 56 void *pcm_buf; 57 struct pollhlp pollhlp; 58 Audiodev *dev; 59 } ALSAVoiceIn; 60 61 struct alsa_params_req { 62 int freq; 63 snd_pcm_format_t fmt; 64 int nchannels; 65 }; 66 67 struct alsa_params_obt { 68 int freq; 69 AudioFormat fmt; 70 int endianness; 71 int nchannels; 72 snd_pcm_uframes_t samples; 73 }; 74 75 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...) 76 { 77 va_list ap; 78 79 va_start (ap, fmt); 80 AUD_vlog (AUDIO_CAP, fmt, ap); 81 va_end (ap); 82 83 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); 84 } 85 86 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 ( 87 int err, 88 const char *typ, 89 const char *fmt, 90 ... 91 ) 92 { 93 va_list ap; 94 95 AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ); 96 97 va_start (ap, fmt); 98 AUD_vlog (AUDIO_CAP, fmt, ap); 99 va_end (ap); 100 101 AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err)); 102 } 103 104 static void alsa_fini_poll (struct pollhlp *hlp) 105 { 106 int i; 107 struct pollfd *pfds = hlp->pfds; 108 109 if (pfds) { 110 for (i = 0; i < hlp->count; ++i) { 111 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); 112 } 113 g_free (pfds); 114 } 115 hlp->pfds = NULL; 116 hlp->count = 0; 117 hlp->handle = NULL; 118 } 119 120 static void alsa_anal_close1 (snd_pcm_t **handlep) 121 { 122 int err = snd_pcm_close (*handlep); 123 if (err) { 124 alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep); 125 } 126 *handlep = NULL; 127 } 128 129 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp) 130 { 131 alsa_fini_poll (hlp); 132 alsa_anal_close1 (handlep); 133 } 134 135 static int alsa_recover (snd_pcm_t *handle) 136 { 137 int err = snd_pcm_prepare (handle); 138 if (err < 0) { 139 alsa_logerr (err, "Failed to prepare handle %p\n", handle); 140 return -1; 141 } 142 return 0; 143 } 144 145 static int alsa_resume (snd_pcm_t *handle) 146 { 147 int err = snd_pcm_resume (handle); 148 if (err < 0) { 149 alsa_logerr (err, "Failed to resume handle %p\n", handle); 150 return -1; 151 } 152 return 0; 153 } 154 155 static void alsa_poll_handler (void *opaque) 156 { 157 int err, count; 158 snd_pcm_state_t state; 159 struct pollhlp *hlp = opaque; 160 unsigned short revents; 161 162 count = poll (hlp->pfds, hlp->count, 0); 163 if (count < 0) { 164 dolog ("alsa_poll_handler: poll %s\n", strerror (errno)); 165 return; 166 } 167 168 if (!count) { 169 return; 170 } 171 172 /* XXX: ALSA example uses initial count, not the one returned by 173 poll, correct? */ 174 err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds, 175 hlp->count, &revents); 176 if (err < 0) { 177 alsa_logerr (err, "snd_pcm_poll_descriptors_revents"); 178 return; 179 } 180 181 if (!(revents & hlp->mask)) { 182 trace_alsa_revents(revents); 183 return; 184 } 185 186 state = snd_pcm_state (hlp->handle); 187 switch (state) { 188 case SND_PCM_STATE_SETUP: 189 alsa_recover (hlp->handle); 190 break; 191 192 case SND_PCM_STATE_XRUN: 193 alsa_recover (hlp->handle); 194 break; 195 196 case SND_PCM_STATE_SUSPENDED: 197 alsa_resume (hlp->handle); 198 break; 199 200 case SND_PCM_STATE_PREPARED: 201 audio_run ("alsa run (prepared)"); 202 break; 203 204 case SND_PCM_STATE_RUNNING: 205 audio_run ("alsa run (running)"); 206 break; 207 208 default: 209 dolog ("Unexpected state %d\n", state); 210 } 211 } 212 213 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) 214 { 215 int i, count, err; 216 struct pollfd *pfds; 217 218 count = snd_pcm_poll_descriptors_count (handle); 219 if (count <= 0) { 220 dolog ("Could not initialize poll mode\n" 221 "Invalid number of poll descriptors %d\n", count); 222 return -1; 223 } 224 225 pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); 226 if (!pfds) { 227 dolog ("Could not initialize poll mode\n"); 228 return -1; 229 } 230 231 err = snd_pcm_poll_descriptors (handle, pfds, count); 232 if (err < 0) { 233 alsa_logerr (err, "Could not initialize poll mode\n" 234 "Could not obtain poll descriptors\n"); 235 g_free (pfds); 236 return -1; 237 } 238 239 for (i = 0; i < count; ++i) { 240 if (pfds[i].events & POLLIN) { 241 qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp); 242 } 243 if (pfds[i].events & POLLOUT) { 244 trace_alsa_pollout(i, pfds[i].fd); 245 qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp); 246 } 247 trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err); 248 249 } 250 hlp->pfds = pfds; 251 hlp->count = count; 252 hlp->handle = handle; 253 hlp->mask = mask; 254 return 0; 255 } 256 257 static int alsa_poll_out (HWVoiceOut *hw) 258 { 259 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 260 261 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT); 262 } 263 264 static int alsa_poll_in (HWVoiceIn *hw) 265 { 266 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 267 268 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); 269 } 270 271 static int alsa_write (SWVoiceOut *sw, void *buf, int len) 272 { 273 return audio_pcm_sw_write (sw, buf, len); 274 } 275 276 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness) 277 { 278 switch (fmt) { 279 case AUDIO_FORMAT_S8: 280 return SND_PCM_FORMAT_S8; 281 282 case AUDIO_FORMAT_U8: 283 return SND_PCM_FORMAT_U8; 284 285 case AUDIO_FORMAT_S16: 286 if (endianness) { 287 return SND_PCM_FORMAT_S16_BE; 288 } 289 else { 290 return SND_PCM_FORMAT_S16_LE; 291 } 292 293 case AUDIO_FORMAT_U16: 294 if (endianness) { 295 return SND_PCM_FORMAT_U16_BE; 296 } 297 else { 298 return SND_PCM_FORMAT_U16_LE; 299 } 300 301 case AUDIO_FORMAT_S32: 302 if (endianness) { 303 return SND_PCM_FORMAT_S32_BE; 304 } 305 else { 306 return SND_PCM_FORMAT_S32_LE; 307 } 308 309 case AUDIO_FORMAT_U32: 310 if (endianness) { 311 return SND_PCM_FORMAT_U32_BE; 312 } 313 else { 314 return SND_PCM_FORMAT_U32_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 default: 381 dolog ("Unrecognized audio format %d\n", alsafmt); 382 return -1; 383 } 384 385 return 0; 386 } 387 388 static void alsa_dump_info (struct alsa_params_req *req, 389 struct alsa_params_obt *obt, 390 snd_pcm_format_t obtfmt, 391 AudiodevAlsaPerDirectionOptions *apdo) 392 { 393 dolog("parameter | requested value | obtained value\n"); 394 dolog("format | %10d | %10d\n", req->fmt, obtfmt); 395 dolog("channels | %10d | %10d\n", 396 req->nchannels, obt->nchannels); 397 dolog("frequency | %10d | %10d\n", req->freq, obt->freq); 398 dolog("============================================\n"); 399 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n", 400 apdo->buffer_length, apdo->period_length); 401 dolog("obtained: samples %ld\n", obt->samples); 402 } 403 404 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) 405 { 406 int err; 407 snd_pcm_sw_params_t *sw_params; 408 409 snd_pcm_sw_params_alloca (&sw_params); 410 411 err = snd_pcm_sw_params_current (handle, sw_params); 412 if (err < 0) { 413 dolog ("Could not fully initialize DAC\n"); 414 alsa_logerr (err, "Failed to get current software parameters\n"); 415 return; 416 } 417 418 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); 419 if (err < 0) { 420 dolog ("Could not fully initialize DAC\n"); 421 alsa_logerr (err, "Failed to set software threshold to %ld\n", 422 threshold); 423 return; 424 } 425 426 err = snd_pcm_sw_params (handle, sw_params); 427 if (err < 0) { 428 dolog ("Could not fully initialize DAC\n"); 429 alsa_logerr (err, "Failed to set software parameters\n"); 430 return; 431 } 432 } 433 434 static int alsa_open(bool in, struct alsa_params_req *req, 435 struct alsa_params_obt *obt, snd_pcm_t **handlep, 436 Audiodev *dev) 437 { 438 AudiodevAlsaOptions *aopts = &dev->u.alsa; 439 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out; 440 snd_pcm_t *handle; 441 snd_pcm_hw_params_t *hw_params; 442 int err; 443 unsigned int freq, nchannels; 444 const char *pcm_name = apdo->has_dev ? apdo->dev : "default"; 445 snd_pcm_uframes_t obt_buffer_size; 446 const char *typ = in ? "ADC" : "DAC"; 447 snd_pcm_format_t obtfmt; 448 449 freq = req->freq; 450 nchannels = req->nchannels; 451 452 snd_pcm_hw_params_alloca (&hw_params); 453 454 err = snd_pcm_open ( 455 &handle, 456 pcm_name, 457 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 458 SND_PCM_NONBLOCK 459 ); 460 if (err < 0) { 461 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); 462 return -1; 463 } 464 465 err = snd_pcm_hw_params_any (handle, hw_params); 466 if (err < 0) { 467 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); 468 goto err; 469 } 470 471 err = snd_pcm_hw_params_set_access ( 472 handle, 473 hw_params, 474 SND_PCM_ACCESS_RW_INTERLEAVED 475 ); 476 if (err < 0) { 477 alsa_logerr2 (err, typ, "Failed to set access type\n"); 478 goto err; 479 } 480 481 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); 482 if (err < 0) { 483 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); 484 } 485 486 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); 487 if (err < 0) { 488 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); 489 goto err; 490 } 491 492 err = snd_pcm_hw_params_set_channels_near ( 493 handle, 494 hw_params, 495 &nchannels 496 ); 497 if (err < 0) { 498 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", 499 req->nchannels); 500 goto err; 501 } 502 503 if (nchannels != 1 && nchannels != 2) { 504 alsa_logerr2 (err, typ, 505 "Can not handle obtained number of channels %d\n", 506 nchannels); 507 goto err; 508 } 509 510 if (apdo->buffer_length) { 511 int dir = 0; 512 unsigned int btime = apdo->buffer_length; 513 514 err = snd_pcm_hw_params_set_buffer_time_near( 515 handle, hw_params, &btime, &dir); 516 517 if (err < 0) { 518 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n", 519 apdo->buffer_length); 520 goto err; 521 } 522 523 if (apdo->has_buffer_length && btime != apdo->buffer_length) { 524 dolog("Requested buffer time %" PRId32 525 " was rejected, using %u\n", apdo->buffer_length, btime); 526 } 527 } 528 529 if (apdo->period_length) { 530 int dir = 0; 531 unsigned int ptime = apdo->period_length; 532 533 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime, 534 &dir); 535 536 if (err < 0) { 537 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n", 538 apdo->period_length); 539 goto err; 540 } 541 542 if (apdo->has_period_length && ptime != apdo->period_length) { 543 dolog("Requested period time %" PRId32 " was rejected, using %d\n", 544 apdo->period_length, ptime); 545 } 546 } 547 548 err = snd_pcm_hw_params (handle, hw_params); 549 if (err < 0) { 550 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); 551 goto err; 552 } 553 554 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); 555 if (err < 0) { 556 alsa_logerr2 (err, typ, "Failed to get buffer size\n"); 557 goto err; 558 } 559 560 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); 561 if (err < 0) { 562 alsa_logerr2 (err, typ, "Failed to get format\n"); 563 goto err; 564 } 565 566 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { 567 dolog ("Invalid format was returned %d\n", obtfmt); 568 goto err; 569 } 570 571 err = snd_pcm_prepare (handle); 572 if (err < 0) { 573 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); 574 goto err; 575 } 576 577 if (!in && aopts->has_threshold && aopts->threshold) { 578 struct audsettings as = { .freq = freq }; 579 alsa_set_threshold( 580 handle, 581 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo), 582 &as, aopts->threshold)); 583 } 584 585 obt->nchannels = nchannels; 586 obt->freq = freq; 587 obt->samples = obt_buffer_size; 588 589 *handlep = handle; 590 591 if (obtfmt != req->fmt || 592 obt->nchannels != req->nchannels || 593 obt->freq != req->freq) { 594 dolog ("Audio parameters for %s\n", typ); 595 alsa_dump_info(req, obt, obtfmt, apdo); 596 } 597 598 #ifdef DEBUG 599 alsa_dump_info(req, obt, obtfmt, pdo); 600 #endif 601 return 0; 602 603 err: 604 alsa_anal_close1 (&handle); 605 return -1; 606 } 607 608 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle) 609 { 610 snd_pcm_sframes_t avail; 611 612 avail = snd_pcm_avail_update (handle); 613 if (avail < 0) { 614 if (avail == -EPIPE) { 615 if (!alsa_recover (handle)) { 616 avail = snd_pcm_avail_update (handle); 617 } 618 } 619 620 if (avail < 0) { 621 alsa_logerr (avail, 622 "Could not obtain number of available frames\n"); 623 return -1; 624 } 625 } 626 627 return avail; 628 } 629 630 static void alsa_write_pending (ALSAVoiceOut *alsa) 631 { 632 HWVoiceOut *hw = &alsa->hw; 633 634 while (alsa->pending) { 635 int left_till_end_samples = hw->samples - alsa->wpos; 636 int len = audio_MIN (alsa->pending, left_till_end_samples); 637 char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift); 638 639 while (len) { 640 snd_pcm_sframes_t written; 641 642 written = snd_pcm_writei (alsa->handle, src, len); 643 644 if (written <= 0) { 645 switch (written) { 646 case 0: 647 trace_alsa_wrote_zero(len); 648 return; 649 650 case -EPIPE: 651 if (alsa_recover (alsa->handle)) { 652 alsa_logerr (written, "Failed to write %d frames\n", 653 len); 654 return; 655 } 656 trace_alsa_xrun_out(); 657 continue; 658 659 case -ESTRPIPE: 660 /* stream is suspended and waiting for an 661 application recovery */ 662 if (alsa_resume (alsa->handle)) { 663 alsa_logerr (written, "Failed to write %d frames\n", 664 len); 665 return; 666 } 667 trace_alsa_resume_out(); 668 continue; 669 670 case -EAGAIN: 671 return; 672 673 default: 674 alsa_logerr (written, "Failed to write %d frames from %p\n", 675 len, src); 676 return; 677 } 678 } 679 680 alsa->wpos = (alsa->wpos + written) % hw->samples; 681 alsa->pending -= written; 682 len -= written; 683 } 684 } 685 } 686 687 static int alsa_run_out (HWVoiceOut *hw, int live) 688 { 689 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 690 int decr; 691 snd_pcm_sframes_t avail; 692 693 avail = alsa_get_avail (alsa->handle); 694 if (avail < 0) { 695 dolog ("Could not get number of available playback frames\n"); 696 return 0; 697 } 698 699 decr = audio_MIN (live, avail); 700 decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending); 701 alsa->pending += decr; 702 alsa_write_pending (alsa); 703 return decr; 704 } 705 706 static void alsa_fini_out (HWVoiceOut *hw) 707 { 708 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 709 710 ldebug ("alsa_fini\n"); 711 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 712 713 g_free(alsa->pcm_buf); 714 alsa->pcm_buf = NULL; 715 } 716 717 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, 718 void *drv_opaque) 719 { 720 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 721 struct alsa_params_req req; 722 struct alsa_params_obt obt; 723 snd_pcm_t *handle; 724 struct audsettings obt_as; 725 Audiodev *dev = drv_opaque; 726 727 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 728 req.freq = as->freq; 729 req.nchannels = as->nchannels; 730 731 if (alsa_open(0, &req, &obt, &handle, dev)) { 732 return -1; 733 } 734 735 obt_as.freq = obt.freq; 736 obt_as.nchannels = obt.nchannels; 737 obt_as.fmt = obt.fmt; 738 obt_as.endianness = obt.endianness; 739 740 audio_pcm_init_info (&hw->info, &obt_as); 741 hw->samples = obt.samples; 742 743 alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift); 744 if (!alsa->pcm_buf) { 745 dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n", 746 hw->samples, 1 << hw->info.shift); 747 alsa_anal_close1 (&handle); 748 return -1; 749 } 750 751 alsa->handle = handle; 752 alsa->dev = dev; 753 return 0; 754 } 755 756 #define VOICE_CTL_PAUSE 0 757 #define VOICE_CTL_PREPARE 1 758 #define VOICE_CTL_START 2 759 760 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) 761 { 762 int err; 763 764 if (ctl == VOICE_CTL_PAUSE) { 765 err = snd_pcm_drop (handle); 766 if (err < 0) { 767 alsa_logerr (err, "Could not stop %s\n", typ); 768 return -1; 769 } 770 } 771 else { 772 err = snd_pcm_prepare (handle); 773 if (err < 0) { 774 alsa_logerr (err, "Could not prepare handle for %s\n", typ); 775 return -1; 776 } 777 if (ctl == VOICE_CTL_START) { 778 err = snd_pcm_start(handle); 779 if (err < 0) { 780 alsa_logerr (err, "Could not start handle for %s\n", typ); 781 return -1; 782 } 783 } 784 } 785 786 return 0; 787 } 788 789 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...) 790 { 791 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 792 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out; 793 794 switch (cmd) { 795 case VOICE_ENABLE: 796 { 797 bool poll_mode = apdo->try_poll; 798 799 ldebug ("enabling voice\n"); 800 if (poll_mode && alsa_poll_out (hw)) { 801 poll_mode = 0; 802 } 803 hw->poll_mode = poll_mode; 804 return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE); 805 } 806 807 case VOICE_DISABLE: 808 ldebug ("disabling voice\n"); 809 if (hw->poll_mode) { 810 hw->poll_mode = 0; 811 alsa_fini_poll (&alsa->pollhlp); 812 } 813 return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE); 814 } 815 816 return -1; 817 } 818 819 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) 820 { 821 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 822 struct alsa_params_req req; 823 struct alsa_params_obt obt; 824 snd_pcm_t *handle; 825 struct audsettings obt_as; 826 Audiodev *dev = drv_opaque; 827 828 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 829 req.freq = as->freq; 830 req.nchannels = as->nchannels; 831 832 if (alsa_open(1, &req, &obt, &handle, dev)) { 833 return -1; 834 } 835 836 obt_as.freq = obt.freq; 837 obt_as.nchannels = obt.nchannels; 838 obt_as.fmt = obt.fmt; 839 obt_as.endianness = obt.endianness; 840 841 audio_pcm_init_info (&hw->info, &obt_as); 842 hw->samples = obt.samples; 843 844 alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift); 845 if (!alsa->pcm_buf) { 846 dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n", 847 hw->samples, 1 << hw->info.shift); 848 alsa_anal_close1 (&handle); 849 return -1; 850 } 851 852 alsa->handle = handle; 853 alsa->dev = dev; 854 return 0; 855 } 856 857 static void alsa_fini_in (HWVoiceIn *hw) 858 { 859 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 860 861 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 862 863 g_free(alsa->pcm_buf); 864 alsa->pcm_buf = NULL; 865 } 866 867 static int alsa_run_in (HWVoiceIn *hw) 868 { 869 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 870 int hwshift = hw->info.shift; 871 int i; 872 int live = audio_pcm_hw_get_live_in (hw); 873 int dead = hw->samples - live; 874 int decr; 875 struct { 876 int add; 877 int len; 878 } bufs[2] = { 879 { .add = hw->wpos, .len = 0 }, 880 { .add = 0, .len = 0 } 881 }; 882 snd_pcm_sframes_t avail; 883 snd_pcm_uframes_t read_samples = 0; 884 885 if (!dead) { 886 return 0; 887 } 888 889 avail = alsa_get_avail (alsa->handle); 890 if (avail < 0) { 891 dolog ("Could not get number of captured frames\n"); 892 return 0; 893 } 894 895 if (!avail) { 896 snd_pcm_state_t state; 897 898 state = snd_pcm_state (alsa->handle); 899 switch (state) { 900 case SND_PCM_STATE_PREPARED: 901 avail = hw->samples; 902 break; 903 case SND_PCM_STATE_SUSPENDED: 904 /* stream is suspended and waiting for an application recovery */ 905 if (alsa_resume (alsa->handle)) { 906 dolog ("Failed to resume suspended input stream\n"); 907 return 0; 908 } 909 trace_alsa_resume_in(); 910 break; 911 default: 912 trace_alsa_no_frames(state); 913 return 0; 914 } 915 } 916 917 decr = audio_MIN (dead, avail); 918 if (!decr) { 919 return 0; 920 } 921 922 if (hw->wpos + decr > hw->samples) { 923 bufs[0].len = (hw->samples - hw->wpos); 924 bufs[1].len = (decr - (hw->samples - hw->wpos)); 925 } 926 else { 927 bufs[0].len = decr; 928 } 929 930 for (i = 0; i < 2; ++i) { 931 void *src; 932 struct st_sample *dst; 933 snd_pcm_sframes_t nread; 934 snd_pcm_uframes_t len; 935 936 len = bufs[i].len; 937 938 src = advance (alsa->pcm_buf, bufs[i].add << hwshift); 939 dst = hw->conv_buf + bufs[i].add; 940 941 while (len) { 942 nread = snd_pcm_readi (alsa->handle, src, len); 943 944 if (nread <= 0) { 945 switch (nread) { 946 case 0: 947 trace_alsa_read_zero(len); 948 goto exit; 949 950 case -EPIPE: 951 if (alsa_recover (alsa->handle)) { 952 alsa_logerr (nread, "Failed to read %ld frames\n", len); 953 goto exit; 954 } 955 trace_alsa_xrun_in(); 956 continue; 957 958 case -EAGAIN: 959 goto exit; 960 961 default: 962 alsa_logerr ( 963 nread, 964 "Failed to read %ld frames from %p\n", 965 len, 966 src 967 ); 968 goto exit; 969 } 970 } 971 972 hw->conv (dst, src, nread); 973 974 src = advance (src, nread << hwshift); 975 dst += nread; 976 977 read_samples += nread; 978 len -= nread; 979 } 980 } 981 982 exit: 983 hw->wpos = (hw->wpos + read_samples) % hw->samples; 984 return read_samples; 985 } 986 987 static int alsa_read (SWVoiceIn *sw, void *buf, int size) 988 { 989 return audio_pcm_sw_read (sw, buf, size); 990 } 991 992 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) 993 { 994 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 995 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in; 996 997 switch (cmd) { 998 case VOICE_ENABLE: 999 { 1000 bool poll_mode = apdo->try_poll; 1001 1002 ldebug ("enabling voice\n"); 1003 if (poll_mode && alsa_poll_in (hw)) { 1004 poll_mode = 0; 1005 } 1006 hw->poll_mode = poll_mode; 1007 1008 return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START); 1009 } 1010 1011 case VOICE_DISABLE: 1012 ldebug ("disabling voice\n"); 1013 if (hw->poll_mode) { 1014 hw->poll_mode = 0; 1015 alsa_fini_poll (&alsa->pollhlp); 1016 } 1017 return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE); 1018 } 1019 1020 return -1; 1021 } 1022 1023 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo) 1024 { 1025 if (!apdo->has_try_poll) { 1026 apdo->try_poll = true; 1027 apdo->has_try_poll = true; 1028 } 1029 } 1030 1031 static void *alsa_audio_init(Audiodev *dev) 1032 { 1033 AudiodevAlsaOptions *aopts; 1034 assert(dev->driver == AUDIODEV_DRIVER_ALSA); 1035 1036 aopts = &dev->u.alsa; 1037 alsa_init_per_direction(aopts->in); 1038 alsa_init_per_direction(aopts->out); 1039 1040 /* 1041 * need to define them, as otherwise alsa produces no sound 1042 * doesn't set has_* so alsa_open can identify it wasn't set by the user 1043 */ 1044 if (!dev->u.alsa.out->has_period_length) { 1045 /* 1024 frames assuming 44100Hz */ 1046 dev->u.alsa.out->period_length = 1024 * 1000000 / 44100; 1047 } 1048 if (!dev->u.alsa.out->has_buffer_length) { 1049 /* 4096 frames assuming 44100Hz */ 1050 dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100; 1051 } 1052 1053 /* 1054 * OptsVisitor sets unspecified optional fields to zero, but do not depend 1055 * on it... 1056 */ 1057 if (!dev->u.alsa.in->has_period_length) { 1058 dev->u.alsa.in->period_length = 0; 1059 } 1060 if (!dev->u.alsa.in->has_buffer_length) { 1061 dev->u.alsa.in->buffer_length = 0; 1062 } 1063 1064 return dev; 1065 } 1066 1067 static void alsa_audio_fini (void *opaque) 1068 { 1069 } 1070 1071 static struct audio_pcm_ops alsa_pcm_ops = { 1072 .init_out = alsa_init_out, 1073 .fini_out = alsa_fini_out, 1074 .run_out = alsa_run_out, 1075 .write = alsa_write, 1076 .ctl_out = alsa_ctl_out, 1077 1078 .init_in = alsa_init_in, 1079 .fini_in = alsa_fini_in, 1080 .run_in = alsa_run_in, 1081 .read = alsa_read, 1082 .ctl_in = alsa_ctl_in, 1083 }; 1084 1085 static struct audio_driver alsa_audio_driver = { 1086 .name = "alsa", 1087 .descr = "ALSA http://www.alsa-project.org", 1088 .init = alsa_audio_init, 1089 .fini = alsa_audio_fini, 1090 .pcm_ops = &alsa_pcm_ops, 1091 .can_be_default = 1, 1092 .max_voices_out = INT_MAX, 1093 .max_voices_in = INT_MAX, 1094 .voice_size_out = sizeof (ALSAVoiceOut), 1095 .voice_size_in = sizeof (ALSAVoiceIn) 1096 }; 1097 1098 static void register_audio_alsa(void) 1099 { 1100 audio_driver_register(&alsa_audio_driver); 1101 } 1102 type_init(register_audio_alsa); 1103