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 #define DEBUG_ALSA 0 38 39 struct pollhlp { 40 snd_pcm_t *handle; 41 struct pollfd *pfds; 42 int count; 43 int mask; 44 AudioState *s; 45 }; 46 47 typedef struct ALSAVoiceOut { 48 HWVoiceOut hw; 49 snd_pcm_t *handle; 50 struct pollhlp pollhlp; 51 Audiodev *dev; 52 } ALSAVoiceOut; 53 54 typedef struct ALSAVoiceIn { 55 HWVoiceIn hw; 56 snd_pcm_t *handle; 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 G_GNUC_PRINTF (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 G_GNUC_PRINTF (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(hlp->s, "alsa run (prepared)"); 202 break; 203 204 case SND_PCM_STATE_RUNNING: 205 audio_run(hlp->s, "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 = g_new0(struct pollfd, count); 226 227 err = snd_pcm_poll_descriptors (handle, pfds, count); 228 if (err < 0) { 229 alsa_logerr (err, "Could not initialize poll mode\n" 230 "Could not obtain poll descriptors\n"); 231 g_free (pfds); 232 return -1; 233 } 234 235 for (i = 0; i < count; ++i) { 236 if (pfds[i].events & POLLIN) { 237 qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp); 238 } 239 if (pfds[i].events & POLLOUT) { 240 trace_alsa_pollout(i, pfds[i].fd); 241 qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp); 242 } 243 trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err); 244 245 } 246 hlp->pfds = pfds; 247 hlp->count = count; 248 hlp->handle = handle; 249 hlp->mask = mask; 250 return 0; 251 } 252 253 static int alsa_poll_out (HWVoiceOut *hw) 254 { 255 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 256 257 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT); 258 } 259 260 static int alsa_poll_in (HWVoiceIn *hw) 261 { 262 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 263 264 return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); 265 } 266 267 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness) 268 { 269 switch (fmt) { 270 case AUDIO_FORMAT_S8: 271 return SND_PCM_FORMAT_S8; 272 273 case AUDIO_FORMAT_U8: 274 return SND_PCM_FORMAT_U8; 275 276 case AUDIO_FORMAT_S16: 277 if (endianness) { 278 return SND_PCM_FORMAT_S16_BE; 279 } else { 280 return SND_PCM_FORMAT_S16_LE; 281 } 282 283 case AUDIO_FORMAT_U16: 284 if (endianness) { 285 return SND_PCM_FORMAT_U16_BE; 286 } else { 287 return SND_PCM_FORMAT_U16_LE; 288 } 289 290 case AUDIO_FORMAT_S32: 291 if (endianness) { 292 return SND_PCM_FORMAT_S32_BE; 293 } else { 294 return SND_PCM_FORMAT_S32_LE; 295 } 296 297 case AUDIO_FORMAT_U32: 298 if (endianness) { 299 return SND_PCM_FORMAT_U32_BE; 300 } else { 301 return SND_PCM_FORMAT_U32_LE; 302 } 303 304 case AUDIO_FORMAT_F32: 305 if (endianness) { 306 return SND_PCM_FORMAT_FLOAT_BE; 307 } else { 308 return SND_PCM_FORMAT_FLOAT_LE; 309 } 310 311 default: 312 dolog ("Internal logic error: Bad audio format %d\n", fmt); 313 #ifdef DEBUG_AUDIO 314 abort (); 315 #endif 316 return SND_PCM_FORMAT_U8; 317 } 318 } 319 320 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt, 321 int *endianness) 322 { 323 switch (alsafmt) { 324 case SND_PCM_FORMAT_S8: 325 *endianness = 0; 326 *fmt = AUDIO_FORMAT_S8; 327 break; 328 329 case SND_PCM_FORMAT_U8: 330 *endianness = 0; 331 *fmt = AUDIO_FORMAT_U8; 332 break; 333 334 case SND_PCM_FORMAT_S16_LE: 335 *endianness = 0; 336 *fmt = AUDIO_FORMAT_S16; 337 break; 338 339 case SND_PCM_FORMAT_U16_LE: 340 *endianness = 0; 341 *fmt = AUDIO_FORMAT_U16; 342 break; 343 344 case SND_PCM_FORMAT_S16_BE: 345 *endianness = 1; 346 *fmt = AUDIO_FORMAT_S16; 347 break; 348 349 case SND_PCM_FORMAT_U16_BE: 350 *endianness = 1; 351 *fmt = AUDIO_FORMAT_U16; 352 break; 353 354 case SND_PCM_FORMAT_S32_LE: 355 *endianness = 0; 356 *fmt = AUDIO_FORMAT_S32; 357 break; 358 359 case SND_PCM_FORMAT_U32_LE: 360 *endianness = 0; 361 *fmt = AUDIO_FORMAT_U32; 362 break; 363 364 case SND_PCM_FORMAT_S32_BE: 365 *endianness = 1; 366 *fmt = AUDIO_FORMAT_S32; 367 break; 368 369 case SND_PCM_FORMAT_U32_BE: 370 *endianness = 1; 371 *fmt = AUDIO_FORMAT_U32; 372 break; 373 374 case SND_PCM_FORMAT_FLOAT_LE: 375 *endianness = 0; 376 *fmt = AUDIO_FORMAT_F32; 377 break; 378 379 case SND_PCM_FORMAT_FLOAT_BE: 380 *endianness = 1; 381 *fmt = AUDIO_FORMAT_F32; 382 break; 383 384 default: 385 dolog ("Unrecognized audio format %d\n", alsafmt); 386 return -1; 387 } 388 389 return 0; 390 } 391 392 static void alsa_dump_info (struct alsa_params_req *req, 393 struct alsa_params_obt *obt, 394 snd_pcm_format_t obtfmt, 395 AudiodevAlsaPerDirectionOptions *apdo) 396 { 397 dolog("parameter | requested value | obtained value\n"); 398 dolog("format | %10d | %10d\n", req->fmt, obtfmt); 399 dolog("channels | %10d | %10d\n", 400 req->nchannels, obt->nchannels); 401 dolog("frequency | %10d | %10d\n", req->freq, obt->freq); 402 dolog("============================================\n"); 403 dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n", 404 apdo->buffer_length, apdo->period_length); 405 dolog("obtained: samples %ld\n", obt->samples); 406 } 407 408 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) 409 { 410 int err; 411 snd_pcm_sw_params_t *sw_params; 412 413 snd_pcm_sw_params_alloca (&sw_params); 414 415 err = snd_pcm_sw_params_current (handle, sw_params); 416 if (err < 0) { 417 dolog ("Could not fully initialize DAC\n"); 418 alsa_logerr (err, "Failed to get current software parameters\n"); 419 return; 420 } 421 422 err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); 423 if (err < 0) { 424 dolog ("Could not fully initialize DAC\n"); 425 alsa_logerr (err, "Failed to set software threshold to %ld\n", 426 threshold); 427 return; 428 } 429 430 err = snd_pcm_sw_params (handle, sw_params); 431 if (err < 0) { 432 dolog ("Could not fully initialize DAC\n"); 433 alsa_logerr (err, "Failed to set software parameters\n"); 434 return; 435 } 436 } 437 438 static int alsa_open(bool in, struct alsa_params_req *req, 439 struct alsa_params_obt *obt, snd_pcm_t **handlep, 440 Audiodev *dev) 441 { 442 AudiodevAlsaOptions *aopts = &dev->u.alsa; 443 AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out; 444 snd_pcm_t *handle; 445 snd_pcm_hw_params_t *hw_params; 446 int err; 447 unsigned int freq, nchannels; 448 const char *pcm_name = apdo->dev ?: "default"; 449 snd_pcm_uframes_t obt_buffer_size; 450 const char *typ = in ? "ADC" : "DAC"; 451 snd_pcm_format_t obtfmt; 452 453 freq = req->freq; 454 nchannels = req->nchannels; 455 456 snd_pcm_hw_params_alloca (&hw_params); 457 458 err = snd_pcm_open ( 459 &handle, 460 pcm_name, 461 in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, 462 SND_PCM_NONBLOCK 463 ); 464 if (err < 0) { 465 alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name); 466 return -1; 467 } 468 469 err = snd_pcm_hw_params_any (handle, hw_params); 470 if (err < 0) { 471 alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n"); 472 goto err; 473 } 474 475 err = snd_pcm_hw_params_set_access ( 476 handle, 477 hw_params, 478 SND_PCM_ACCESS_RW_INTERLEAVED 479 ); 480 if (err < 0) { 481 alsa_logerr2 (err, typ, "Failed to set access type\n"); 482 goto err; 483 } 484 485 err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); 486 if (err < 0) { 487 alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt); 488 } 489 490 err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); 491 if (err < 0) { 492 alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq); 493 goto err; 494 } 495 496 err = snd_pcm_hw_params_set_channels_near ( 497 handle, 498 hw_params, 499 &nchannels 500 ); 501 if (err < 0) { 502 alsa_logerr2 (err, typ, "Failed to set number of channels %d\n", 503 req->nchannels); 504 goto err; 505 } 506 507 if (apdo->buffer_length) { 508 int dir = 0; 509 unsigned int btime = apdo->buffer_length; 510 511 err = snd_pcm_hw_params_set_buffer_time_near( 512 handle, hw_params, &btime, &dir); 513 514 if (err < 0) { 515 alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n", 516 apdo->buffer_length); 517 goto err; 518 } 519 520 if (apdo->has_buffer_length && btime != apdo->buffer_length) { 521 dolog("Requested buffer time %" PRId32 522 " was rejected, using %u\n", apdo->buffer_length, btime); 523 } 524 } 525 526 if (apdo->period_length) { 527 int dir = 0; 528 unsigned int ptime = apdo->period_length; 529 530 err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime, 531 &dir); 532 533 if (err < 0) { 534 alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n", 535 apdo->period_length); 536 goto err; 537 } 538 539 if (apdo->has_period_length && ptime != apdo->period_length) { 540 dolog("Requested period time %" PRId32 " was rejected, using %d\n", 541 apdo->period_length, ptime); 542 } 543 } 544 545 err = snd_pcm_hw_params (handle, hw_params); 546 if (err < 0) { 547 alsa_logerr2 (err, typ, "Failed to apply audio parameters\n"); 548 goto err; 549 } 550 551 err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); 552 if (err < 0) { 553 alsa_logerr2 (err, typ, "Failed to get buffer size\n"); 554 goto err; 555 } 556 557 err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); 558 if (err < 0) { 559 alsa_logerr2 (err, typ, "Failed to get format\n"); 560 goto err; 561 } 562 563 if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { 564 dolog ("Invalid format was returned %d\n", obtfmt); 565 goto err; 566 } 567 568 err = snd_pcm_prepare (handle); 569 if (err < 0) { 570 alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); 571 goto err; 572 } 573 574 if (!in && aopts->has_threshold && aopts->threshold) { 575 struct audsettings as = { .freq = freq }; 576 alsa_set_threshold( 577 handle, 578 audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo), 579 &as, aopts->threshold)); 580 } 581 582 obt->nchannels = nchannels; 583 obt->freq = freq; 584 obt->samples = obt_buffer_size; 585 586 *handlep = handle; 587 588 if (DEBUG_ALSA || obtfmt != req->fmt || 589 obt->nchannels != req->nchannels || obt->freq != req->freq) { 590 dolog ("Audio parameters for %s\n", typ); 591 alsa_dump_info(req, obt, obtfmt, apdo); 592 } 593 594 return 0; 595 596 err: 597 alsa_anal_close1 (&handle); 598 return -1; 599 } 600 601 static size_t alsa_buffer_get_free(HWVoiceOut *hw) 602 { 603 ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw; 604 snd_pcm_sframes_t avail; 605 size_t alsa_free, generic_free, generic_in_use; 606 607 avail = snd_pcm_avail_update(alsa->handle); 608 if (avail < 0) { 609 if (avail == -EPIPE) { 610 if (!alsa_recover(alsa->handle)) { 611 avail = snd_pcm_avail_update(alsa->handle); 612 } 613 } 614 if (avail < 0) { 615 alsa_logerr(avail, 616 "Could not obtain number of available frames\n"); 617 avail = 0; 618 } 619 } 620 621 alsa_free = avail * hw->info.bytes_per_frame; 622 generic_free = audio_generic_buffer_get_free(hw); 623 generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free; 624 if (generic_in_use) { 625 /* 626 * This code can only be reached in the unlikely case that 627 * snd_pcm_avail_update() returned a larger number of frames 628 * than snd_pcm_writei() could write. Make sure that all 629 * remaining bytes in the generic buffer can be written. 630 */ 631 alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0; 632 } 633 634 return alsa_free; 635 } 636 637 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) 638 { 639 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 640 size_t pos = 0; 641 size_t len_frames = len / hw->info.bytes_per_frame; 642 643 while (len_frames) { 644 char *src = advance(buf, pos); 645 snd_pcm_sframes_t written; 646 647 written = snd_pcm_writei(alsa->handle, src, len_frames); 648 649 if (written <= 0) { 650 switch (written) { 651 case 0: 652 trace_alsa_wrote_zero(len_frames); 653 return pos; 654 655 case -EPIPE: 656 if (alsa_recover(alsa->handle)) { 657 alsa_logerr(written, "Failed to write %zu frames\n", 658 len_frames); 659 return pos; 660 } 661 trace_alsa_xrun_out(); 662 continue; 663 664 case -ESTRPIPE: 665 /* 666 * stream is suspended and waiting for an application 667 * recovery 668 */ 669 if (alsa_resume(alsa->handle)) { 670 alsa_logerr(written, "Failed to write %zu frames\n", 671 len_frames); 672 return pos; 673 } 674 trace_alsa_resume_out(); 675 continue; 676 677 case -EAGAIN: 678 return pos; 679 680 default: 681 alsa_logerr(written, "Failed to write %zu frames from %p\n", 682 len, src); 683 return pos; 684 } 685 } 686 687 pos += written * hw->info.bytes_per_frame; 688 if (written < len_frames) { 689 break; 690 } 691 len_frames -= written; 692 } 693 694 return pos; 695 } 696 697 static void alsa_fini_out (HWVoiceOut *hw) 698 { 699 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 700 701 ldebug ("alsa_fini\n"); 702 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 703 } 704 705 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as, 706 void *drv_opaque) 707 { 708 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 709 struct alsa_params_req req; 710 struct alsa_params_obt obt; 711 snd_pcm_t *handle; 712 struct audsettings obt_as; 713 Audiodev *dev = drv_opaque; 714 715 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 716 req.freq = as->freq; 717 req.nchannels = as->nchannels; 718 719 if (alsa_open(0, &req, &obt, &handle, dev)) { 720 return -1; 721 } 722 723 obt_as.freq = obt.freq; 724 obt_as.nchannels = obt.nchannels; 725 obt_as.fmt = obt.fmt; 726 obt_as.endianness = obt.endianness; 727 728 audio_pcm_init_info (&hw->info, &obt_as); 729 hw->samples = obt.samples; 730 731 alsa->pollhlp.s = hw->s; 732 alsa->handle = handle; 733 alsa->dev = dev; 734 return 0; 735 } 736 737 #define VOICE_CTL_PAUSE 0 738 #define VOICE_CTL_PREPARE 1 739 #define VOICE_CTL_START 2 740 741 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) 742 { 743 int err; 744 745 if (ctl == VOICE_CTL_PAUSE) { 746 err = snd_pcm_drop (handle); 747 if (err < 0) { 748 alsa_logerr (err, "Could not stop %s\n", typ); 749 return -1; 750 } 751 } else { 752 err = snd_pcm_prepare (handle); 753 if (err < 0) { 754 alsa_logerr (err, "Could not prepare handle for %s\n", typ); 755 return -1; 756 } 757 if (ctl == VOICE_CTL_START) { 758 err = snd_pcm_start(handle); 759 if (err < 0) { 760 alsa_logerr (err, "Could not start handle for %s\n", typ); 761 return -1; 762 } 763 } 764 } 765 766 return 0; 767 } 768 769 static void alsa_enable_out(HWVoiceOut *hw, bool enable) 770 { 771 ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; 772 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out; 773 774 if (enable) { 775 bool poll_mode = apdo->try_poll; 776 777 ldebug("enabling voice\n"); 778 if (poll_mode && alsa_poll_out(hw)) { 779 poll_mode = 0; 780 } 781 hw->poll_mode = poll_mode; 782 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE); 783 } else { 784 ldebug("disabling voice\n"); 785 if (hw->poll_mode) { 786 hw->poll_mode = 0; 787 alsa_fini_poll(&alsa->pollhlp); 788 } 789 alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE); 790 } 791 } 792 793 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque) 794 { 795 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 796 struct alsa_params_req req; 797 struct alsa_params_obt obt; 798 snd_pcm_t *handle; 799 struct audsettings obt_as; 800 Audiodev *dev = drv_opaque; 801 802 req.fmt = aud_to_alsafmt (as->fmt, as->endianness); 803 req.freq = as->freq; 804 req.nchannels = as->nchannels; 805 806 if (alsa_open(1, &req, &obt, &handle, dev)) { 807 return -1; 808 } 809 810 obt_as.freq = obt.freq; 811 obt_as.nchannels = obt.nchannels; 812 obt_as.fmt = obt.fmt; 813 obt_as.endianness = obt.endianness; 814 815 audio_pcm_init_info (&hw->info, &obt_as); 816 hw->samples = obt.samples; 817 818 alsa->pollhlp.s = hw->s; 819 alsa->handle = handle; 820 alsa->dev = dev; 821 return 0; 822 } 823 824 static void alsa_fini_in (HWVoiceIn *hw) 825 { 826 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 827 828 alsa_anal_close (&alsa->handle, &alsa->pollhlp); 829 } 830 831 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) 832 { 833 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 834 size_t pos = 0; 835 836 while (len) { 837 void *dst = advance(buf, pos); 838 snd_pcm_sframes_t nread; 839 840 nread = snd_pcm_readi( 841 alsa->handle, dst, len / hw->info.bytes_per_frame); 842 843 if (nread <= 0) { 844 switch (nread) { 845 case 0: 846 trace_alsa_read_zero(len); 847 return pos; 848 849 case -EPIPE: 850 if (alsa_recover(alsa->handle)) { 851 alsa_logerr(nread, "Failed to read %zu frames\n", len); 852 return pos; 853 } 854 trace_alsa_xrun_in(); 855 continue; 856 857 case -EAGAIN: 858 return pos; 859 860 default: 861 alsa_logerr(nread, "Failed to read %zu frames to %p\n", 862 len, dst); 863 return pos; 864 } 865 } 866 867 pos += nread * hw->info.bytes_per_frame; 868 len -= nread * hw->info.bytes_per_frame; 869 } 870 871 return pos; 872 } 873 874 static void alsa_enable_in(HWVoiceIn *hw, bool enable) 875 { 876 ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; 877 AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in; 878 879 if (enable) { 880 bool poll_mode = apdo->try_poll; 881 882 ldebug("enabling voice\n"); 883 if (poll_mode && alsa_poll_in(hw)) { 884 poll_mode = 0; 885 } 886 hw->poll_mode = poll_mode; 887 888 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START); 889 } else { 890 ldebug ("disabling voice\n"); 891 if (hw->poll_mode) { 892 hw->poll_mode = 0; 893 alsa_fini_poll(&alsa->pollhlp); 894 } 895 alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE); 896 } 897 } 898 899 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo) 900 { 901 if (!apdo->has_try_poll) { 902 apdo->try_poll = true; 903 apdo->has_try_poll = true; 904 } 905 } 906 907 static void *alsa_audio_init(Audiodev *dev, Error **errp) 908 { 909 AudiodevAlsaOptions *aopts; 910 assert(dev->driver == AUDIODEV_DRIVER_ALSA); 911 912 aopts = &dev->u.alsa; 913 alsa_init_per_direction(aopts->in); 914 alsa_init_per_direction(aopts->out); 915 916 /* don't set has_* so alsa_open can identify it wasn't set by the user */ 917 if (!dev->u.alsa.out->has_period_length) { 918 /* 256 frames assuming 44100Hz */ 919 dev->u.alsa.out->period_length = 5805; 920 } 921 if (!dev->u.alsa.out->has_buffer_length) { 922 /* 4096 frames assuming 44100Hz */ 923 dev->u.alsa.out->buffer_length = 92880; 924 } 925 926 if (!dev->u.alsa.in->has_period_length) { 927 /* 256 frames assuming 44100Hz */ 928 dev->u.alsa.in->period_length = 5805; 929 } 930 if (!dev->u.alsa.in->has_buffer_length) { 931 /* 4096 frames assuming 44100Hz */ 932 dev->u.alsa.in->buffer_length = 92880; 933 } 934 935 return dev; 936 } 937 938 static void alsa_audio_fini (void *opaque) 939 { 940 } 941 942 static struct audio_pcm_ops alsa_pcm_ops = { 943 .init_out = alsa_init_out, 944 .fini_out = alsa_fini_out, 945 .write = alsa_write, 946 .buffer_get_free = alsa_buffer_get_free, 947 .run_buffer_out = audio_generic_run_buffer_out, 948 .enable_out = alsa_enable_out, 949 950 .init_in = alsa_init_in, 951 .fini_in = alsa_fini_in, 952 .read = alsa_read, 953 .run_buffer_in = audio_generic_run_buffer_in, 954 .enable_in = alsa_enable_in, 955 }; 956 957 static struct audio_driver alsa_audio_driver = { 958 .name = "alsa", 959 .descr = "ALSA http://www.alsa-project.org", 960 .init = alsa_audio_init, 961 .fini = alsa_audio_fini, 962 .pcm_ops = &alsa_pcm_ops, 963 .max_voices_out = INT_MAX, 964 .max_voices_in = INT_MAX, 965 .voice_size_out = sizeof (ALSAVoiceOut), 966 .voice_size_in = sizeof (ALSAVoiceIn) 967 }; 968 969 static void register_audio_alsa(void) 970 { 971 audio_driver_register(&alsa_audio_driver); 972 } 973 type_init(register_audio_alsa); 974