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1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474 475 476 477 478 479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519 520 521 522 523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552 553 554 555 556 557 558 559 560 561 562 563 564 565 566 567 568 569 570 571 572 573 574 575 576 577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592 593 594 595 596 597 598 599 600 601 602 603 604 605 606 607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623 624 625 626 627 628 629 630 631 632 633 634 635 636 637 638 639 640 641 642 643 644 645 646 647 648 649 650 651 652 653 654 655 656 657 658 659 660 661 662 663 664 665 666 667 668 669 670 671 672 673 674 675 676 677 678 679 680 681 682 683 684 685 686 687 688 689 690 691 692 693 694 695 696 697 698 699 700 701 702 703 704 705 706 707 708 709 710 711 712 713 714 715 716 717 718 719 720 721 722 723 724 725 726 727 728 729 730 731 732 733 734 735 736 737 738 739 740 741 742 743 744 745 746 747 748 749 750 751 752 753 754 755 756 757 758 759 760 761 762 763 764 765 766 767 768 769 770 771 772 773 774 775 776 777 778 779 780 781 782 783 784 785 786 787 788 789 790 791 792 793 794 795 796 797 798 799 800 801 802 803 804 805 806 807 808 809 810 811 812 813 814 815 816 817 818 819 820 821 822 823 824 825 826 827 828 829 830 831 832 833 834 835 836 837 838 839 840 841 842 843 844 845 846 847 848 849 850 851 852 853 854 855 856 857 858 859 860 861 862 863 864 865 866 867 868 869 870 871 872 873 874 875 876 877 878 879 880 881 882 883 884 885 886 887 888 889 890 891 892 893 894 895 896 897 898 899 900 901 902 903 904 905 906 907 908 909 910 911 912 913 914 915 916 917 918 919 920 921 922 923 924 925 926 927 928 929 930 931 932 933 934 935 936 937 938 939 940 941 942 943 944 945 946 947 948 949 950 951 952 953 954 955 956 957 958 959 960 961 962 963 964 965 966 967 968 969 970 971 972 973 974 975 976 977 978 979 980 981 982 983 984 985 986 987 988 989 990 991 992 993 994 995 996 997 998 999 1000 1001 1002 1003 1004 1005 1006 1007 1008 1009 1010 1011 1012 1013 1014 1015 1016 1017 | // SPDX-License-Identifier: GPL-2.0-only /* * HD-audio stream operations */ #include <linux/kernel.h> #include <linux/delay.h> #include <linux/export.h> #include <linux/clocksource.h> #include <sound/compress_driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/hdaudio.h> #include <sound/hda_register.h> #include "trace.h" /* * the hdac_stream library is intended to be used with the following * transitions. The states are not formally defined in the code but loosely * inspired by boolean variables. Note that the 'prepared' field is not used * in this library but by the callers during the hw_params/prepare transitions * * | * stream_init() | * v * +--+-------+ * | unused | * +--+----+--+ * | ^ * stream_assign() | | stream_release() * v | * +--+----+--+ * | opened | * +--+----+--+ * | ^ * stream_reset() | | * stream_setup() | | stream_cleanup() * v | * +--+----+--+ * | prepared | * +--+----+--+ * | ^ * stream_start() | | stream_stop() * v | * +--+----+--+ * | running | * +----------+ */ /** * snd_hdac_get_stream_stripe_ctl - get stripe control value * @bus: HD-audio core bus * @substream: PCM substream */ int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; unsigned int channels = runtime->channels, rate = runtime->rate, bits_per_sample = runtime->sample_bits, max_sdo_lines, value, sdo_line; /* T_AZA_GCAP_NSDO is 1:2 bitfields in GCAP */ max_sdo_lines = snd_hdac_chip_readl(bus, GCAP) & AZX_GCAP_NSDO; /* following is from HD audio spec */ for (sdo_line = max_sdo_lines; sdo_line > 0; sdo_line >>= 1) { if (rate > 48000) value = (channels * bits_per_sample * (rate / 48000)) / sdo_line; else value = (channels * bits_per_sample) / sdo_line; if (value >= bus->sdo_limit) break; } /* stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines */ return sdo_line >> 1; } EXPORT_SYMBOL_GPL(snd_hdac_get_stream_stripe_ctl); /** * snd_hdac_stream_init - initialize each stream (aka device) * @bus: HD-audio core bus * @azx_dev: HD-audio core stream object to initialize * @idx: stream index number * @direction: stream direction (SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE) * @tag: the tag id to assign * * Assign the starting bdl address to each stream (device) and initialize. */ void snd_hdac_stream_init(struct hdac_bus *bus, struct hdac_stream *azx_dev, int idx, int direction, int tag) { azx_dev->bus = bus; /* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ azx_dev->sd_addr = bus->remap_addr + (0x20 * idx + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << idx; azx_dev->index = idx; azx_dev->direction = direction; azx_dev->stream_tag = tag; snd_hdac_dsp_lock_init(azx_dev); list_add_tail(&azx_dev->list, &bus->stream_list); if (bus->spbcap) { azx_dev->spib_addr = bus->spbcap + AZX_SPB_BASE + AZX_SPB_INTERVAL * idx + AZX_SPB_SPIB; azx_dev->fifo_addr = bus->spbcap + AZX_SPB_BASE + AZX_SPB_INTERVAL * idx + AZX_SPB_MAXFIFO; } if (bus->drsmcap) azx_dev->dpibr_addr = bus->drsmcap + AZX_DRSM_BASE + AZX_DRSM_INTERVAL * idx; } EXPORT_SYMBOL_GPL(snd_hdac_stream_init); /** * snd_hdac_stream_start - start a stream * @azx_dev: HD-audio core stream to start * * Start a stream, set start_wallclk and set the running flag. */ void snd_hdac_stream_start(struct hdac_stream *azx_dev) { struct hdac_bus *bus = azx_dev->bus; int stripe_ctl; trace_snd_hdac_stream_start(bus, azx_dev); azx_dev->start_wallclk = snd_hdac_chip_readl(bus, WALLCLK); /* enable SIE */ snd_hdac_chip_updatel(bus, INTCTL, 1 << azx_dev->index, 1 << azx_dev->index); /* set stripe control */ if (azx_dev->stripe) { if (azx_dev->substream) stripe_ctl = snd_hdac_get_stream_stripe_ctl(bus, azx_dev->substream); else stripe_ctl = 0; snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, stripe_ctl); } /* set DMA start and interrupt mask */ if (bus->access_sdnctl_in_dword) snd_hdac_stream_updatel(azx_dev, SD_CTL, 0, SD_CTL_DMA_START | SD_INT_MASK); else snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_DMA_START | SD_INT_MASK); azx_dev->running = true; } EXPORT_SYMBOL_GPL(snd_hdac_stream_start); /** * snd_hdac_stream_clear - helper to clear stream registers and stop DMA transfers * @azx_dev: HD-audio core stream to stop */ static void snd_hdac_stream_clear(struct hdac_stream *azx_dev) { snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ if (azx_dev->stripe) snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); azx_dev->running = false; } /** * snd_hdac_stream_stop - stop a stream * @azx_dev: HD-audio core stream to stop * * Stop a stream DMA and disable stream interrupt */ void snd_hdac_stream_stop(struct hdac_stream *azx_dev) { trace_snd_hdac_stream_stop(azx_dev->bus, azx_dev); snd_hdac_stream_clear(azx_dev); /* disable SIE */ snd_hdac_chip_updatel(azx_dev->bus, INTCTL, 1 << azx_dev->index, 0); } EXPORT_SYMBOL_GPL(snd_hdac_stream_stop); /** * snd_hdac_stop_streams - stop all streams * @bus: HD-audio core bus */ void snd_hdac_stop_streams(struct hdac_bus *bus) { struct hdac_stream *stream; list_for_each_entry(stream, &bus->stream_list, list) snd_hdac_stream_stop(stream); } EXPORT_SYMBOL_GPL(snd_hdac_stop_streams); /** * snd_hdac_stop_streams_and_chip - stop all streams and chip if running * @bus: HD-audio core bus */ void snd_hdac_stop_streams_and_chip(struct hdac_bus *bus) { if (bus->chip_init) { snd_hdac_stop_streams(bus); snd_hdac_bus_stop_chip(bus); } } EXPORT_SYMBOL_GPL(snd_hdac_stop_streams_and_chip); /** * snd_hdac_stream_reset - reset a stream * @azx_dev: HD-audio core stream to reset */ void snd_hdac_stream_reset(struct hdac_stream *azx_dev) { unsigned char val; int dma_run_state; snd_hdac_stream_clear(azx_dev); dma_run_state = snd_hdac_stream_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START; snd_hdac_stream_updateb(azx_dev, SD_CTL, 0, SD_CTL_STREAM_RESET); /* wait for hardware to report that the stream entered reset */ snd_hdac_stream_readb_poll(azx_dev, SD_CTL, val, (val & SD_CTL_STREAM_RESET), 3, 300); if (azx_dev->bus->dma_stop_delay && dma_run_state) udelay(azx_dev->bus->dma_stop_delay); snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_STREAM_RESET, 0); /* wait for hardware to report that the stream is out of reset */ snd_hdac_stream_readb_poll(azx_dev, SD_CTL, val, !(val & SD_CTL_STREAM_RESET), 3, 300); /* reset first position - may not be synced with hw at this time */ if (azx_dev->posbuf) *azx_dev->posbuf = 0; } EXPORT_SYMBOL_GPL(snd_hdac_stream_reset); /** * snd_hdac_stream_setup - set up the SD for streaming * @azx_dev: HD-audio core stream to set up * @code_loading: Whether the stream is for PCM or code-loading. */ int snd_hdac_stream_setup(struct hdac_stream *azx_dev, bool code_loading) { struct hdac_bus *bus = azx_dev->bus; struct snd_pcm_runtime *runtime; unsigned int val; u16 reg; int ret; if (azx_dev->substream) runtime = azx_dev->substream->runtime; else runtime = NULL; /* make sure the run bit is zero for SD */ snd_hdac_stream_clear(azx_dev); /* program the stream_tag */ val = snd_hdac_stream_readl(azx_dev, SD_CTL); val = (val & ~SD_CTL_STREAM_TAG_MASK) | (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT); if (!bus->snoop) val |= SD_CTL_TRAFFIC_PRIO; snd_hdac_stream_writel(azx_dev, SD_CTL, val); /* program the length of samples in cyclic buffer */ snd_hdac_stream_writel(azx_dev, SD_CBL, azx_dev->bufsize); /* program the stream format */ /* this value needs to be the same as the one programmed */ snd_hdac_stream_writew(azx_dev, SD_FORMAT, azx_dev->format_val); /* program the stream LVI (last valid index) of the BDL */ snd_hdac_stream_writew(azx_dev, SD_LVI, azx_dev->frags - 1); /* program the BDL address */ /* lower BDL address */ snd_hdac_stream_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl.addr); /* upper BDL address */ snd_hdac_stream_writel(azx_dev, SD_BDLPU, upper_32_bits(azx_dev->bdl.addr)); /* enable the position buffer */ if (bus->use_posbuf && bus->posbuf.addr) { if (!(snd_hdac_chip_readl(bus, DPLBASE) & AZX_DPLBASE_ENABLE)) snd_hdac_chip_writel(bus, DPLBASE, (u32)bus->posbuf.addr | AZX_DPLBASE_ENABLE); } /* set the interrupt enable bits in the descriptor control register */ snd_hdac_stream_updatel(azx_dev, SD_CTL, 0, SD_INT_MASK); if (!code_loading) { /* Once SDxFMT is set, the controller programs SDxFIFOS to non-zero value. */ ret = snd_hdac_stream_readw_poll(azx_dev, SD_FIFOSIZE, reg, reg & AZX_SD_FIFOSIZE_MASK, 3, 300); if (ret) dev_dbg(bus->dev, "polling SD_FIFOSIZE 0x%04x failed: %d\n", AZX_REG_SD_FIFOSIZE, ret); azx_dev->fifo_size = reg; } /* when LPIB delay correction gives a small negative value, * we ignore it; currently set the threshold statically to * 64 frames */ if (runtime && runtime->period_size > 64) azx_dev->delay_negative_threshold = -frames_to_bytes(runtime, 64); else azx_dev->delay_negative_threshold = 0; /* wallclk has 24Mhz clock source */ if (runtime) azx_dev->period_wallclk = (((runtime->period_size * 24000) / runtime->rate) * 1000); return 0; } EXPORT_SYMBOL_GPL(snd_hdac_stream_setup); /** * snd_hdac_stream_cleanup - cleanup a stream * @azx_dev: HD-audio core stream to clean up */ void snd_hdac_stream_cleanup(struct hdac_stream *azx_dev) { snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0); snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0); snd_hdac_stream_writel(azx_dev, SD_CTL, 0); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; azx_dev->format_val = 0; } EXPORT_SYMBOL_GPL(snd_hdac_stream_cleanup); /** * snd_hdac_stream_assign - assign a stream for the PCM * @bus: HD-audio core bus * @substream: PCM substream to assign * * Look for an unused stream for the given PCM substream, assign it * and return the stream object. If no stream is free, returns NULL. * The function tries to keep using the same stream object when it's used * beforehand. Also, when bus->reverse_assign flag is set, the last free * or matching entry is returned. This is needed for some strange codecs. */ struct hdac_stream *snd_hdac_stream_assign(struct hdac_bus *bus, struct snd_pcm_substream *substream) { struct hdac_stream *azx_dev; struct hdac_stream *res = NULL; /* make a non-zero unique key for the substream */ int key = (substream->number << 2) | (substream->stream + 1); if (substream->pcm) key |= (substream->pcm->device << 16); spin_lock_irq(&bus->reg_lock); list_for_each_entry(azx_dev, &bus->stream_list, list) { if (azx_dev->direction != substream->stream) continue; if (azx_dev->opened) continue; if (azx_dev->assigned_key == key) { res = azx_dev; break; } if (!res || bus->reverse_assign) res = azx_dev; } if (res) { res->opened = 1; res->running = 0; res->assigned_key = key; res->substream = substream; } spin_unlock_irq(&bus->reg_lock); return res; } EXPORT_SYMBOL_GPL(snd_hdac_stream_assign); /** * snd_hdac_stream_release_locked - release the assigned stream * @azx_dev: HD-audio core stream to release * * Release the stream that has been assigned by snd_hdac_stream_assign(). * The bus->reg_lock needs to be taken at a higher level */ void snd_hdac_stream_release_locked(struct hdac_stream *azx_dev) { azx_dev->opened = 0; azx_dev->running = 0; azx_dev->substream = NULL; } EXPORT_SYMBOL_GPL(snd_hdac_stream_release_locked); /** * snd_hdac_stream_release - release the assigned stream * @azx_dev: HD-audio core stream to release * * Release the stream that has been assigned by snd_hdac_stream_assign(). */ void snd_hdac_stream_release(struct hdac_stream *azx_dev) { struct hdac_bus *bus = azx_dev->bus; spin_lock_irq(&bus->reg_lock); snd_hdac_stream_release_locked(azx_dev); spin_unlock_irq(&bus->reg_lock); } EXPORT_SYMBOL_GPL(snd_hdac_stream_release); /** * snd_hdac_get_stream - return hdac_stream based on stream_tag and * direction * * @bus: HD-audio core bus * @dir: direction for the stream to be found * @stream_tag: stream tag for stream to be found */ struct hdac_stream *snd_hdac_get_stream(struct hdac_bus *bus, int dir, int stream_tag) { struct hdac_stream *s; list_for_each_entry(s, &bus->stream_list, list) { if (s->direction == dir && s->stream_tag == stream_tag) return s; } return NULL; } EXPORT_SYMBOL_GPL(snd_hdac_get_stream); /* * set up a BDL entry */ static int setup_bdle(struct hdac_bus *bus, struct snd_dma_buffer *dmab, struct hdac_stream *azx_dev, __le32 **bdlp, int ofs, int size, int with_ioc) { __le32 *bdl = *bdlp; while (size > 0) { dma_addr_t addr; int chunk; if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES) return -EINVAL; addr = snd_sgbuf_get_addr(dmab, ofs); /* program the address field of the BDL entry */ bdl[0] = cpu_to_le32((u32)addr); bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ chunk = snd_sgbuf_get_chunk_size(dmab, ofs, size); /* one BDLE cannot cross 4K boundary on CTHDA chips */ if (bus->align_bdle_4k) { u32 remain = 0x1000 - (ofs & 0xfff); if (chunk > remain) chunk = remain; } bdl[2] = cpu_to_le32(chunk); /* program the IOC to enable interrupt * only when the whole fragment is processed */ size -= chunk; bdl[3] = (size || !with_ioc) ? 0 : cpu_to_le32(0x01); bdl += 4; azx_dev->frags++; ofs += chunk; } *bdlp = bdl; return ofs; } /** * snd_hdac_stream_setup_periods - set up BDL entries * @azx_dev: HD-audio core stream to set up * * Set up the buffer descriptor table of the given stream based on the * period and buffer sizes of the assigned PCM substream. */ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev) { struct hdac_bus *bus = azx_dev->bus; struct snd_pcm_substream *substream = azx_dev->substream; struct snd_compr_stream *cstream = azx_dev->cstream; struct snd_pcm_runtime *runtime = NULL; struct snd_dma_buffer *dmab; __le32 *bdl; int i, ofs, periods, period_bytes; int pos_adj, pos_align; if (substream) { runtime = substream->runtime; dmab = snd_pcm_get_dma_buf(substream); } else if (cstream) { dmab = snd_pcm_get_dma_buf(cstream); } else { WARN(1, "No substream or cstream assigned\n"); return -EINVAL; } /* reset BDL address */ snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0); snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0); period_bytes = azx_dev->period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ bdl = (__le32 *)azx_dev->bdl.area; ofs = 0; azx_dev->frags = 0; pos_adj = bus->bdl_pos_adj; if (runtime && !azx_dev->no_period_wakeup && pos_adj > 0) { pos_align = pos_adj; pos_adj = DIV_ROUND_UP(pos_adj * runtime->rate, 48000); if (!pos_adj) pos_adj = pos_align; else pos_adj = roundup(pos_adj, pos_align); pos_adj = frames_to_bytes(runtime, pos_adj); if (pos_adj >= period_bytes) { dev_warn(bus->dev, "Too big adjustment %d\n", pos_adj); pos_adj = 0; } else { ofs = setup_bdle(bus, dmab, azx_dev, &bdl, ofs, pos_adj, true); if (ofs < 0) goto error; } } else pos_adj = 0; for (i = 0; i < periods; i++) { if (i == periods - 1 && pos_adj) ofs = setup_bdle(bus, dmab, azx_dev, &bdl, ofs, period_bytes - pos_adj, 0); else ofs = setup_bdle(bus, dmab, azx_dev, &bdl, ofs, period_bytes, !azx_dev->no_period_wakeup); if (ofs < 0) goto error; } return 0; error: dev_dbg(bus->dev, "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); return -EINVAL; } EXPORT_SYMBOL_GPL(snd_hdac_stream_setup_periods); /** * snd_hdac_stream_set_params - set stream parameters * @azx_dev: HD-audio core stream for which parameters are to be set * @format_val: format value parameter * * Setup the HD-audio core stream parameters from substream of the stream * and passed format value */ int snd_hdac_stream_set_params(struct hdac_stream *azx_dev, unsigned int format_val) { struct snd_pcm_substream *substream = azx_dev->substream; struct snd_compr_stream *cstream = azx_dev->cstream; unsigned int bufsize, period_bytes; unsigned int no_period_wakeup; int err; if (substream) { bufsize = snd_pcm_lib_buffer_bytes(substream); period_bytes = snd_pcm_lib_period_bytes(substream); no_period_wakeup = substream->runtime->no_period_wakeup; } else if (cstream) { bufsize = cstream->runtime->buffer_size; period_bytes = cstream->runtime->fragment_size; no_period_wakeup = 0; } else { return -EINVAL; } if (bufsize != azx_dev->bufsize || period_bytes != azx_dev->period_bytes || format_val != azx_dev->format_val || no_period_wakeup != azx_dev->no_period_wakeup) { azx_dev->bufsize = bufsize; azx_dev->period_bytes = period_bytes; azx_dev->format_val = format_val; azx_dev->no_period_wakeup = no_period_wakeup; err = snd_hdac_stream_setup_periods(azx_dev); if (err < 0) return err; } return 0; } EXPORT_SYMBOL_GPL(snd_hdac_stream_set_params); static u64 azx_cc_read(const struct cyclecounter *cc) { struct hdac_stream *azx_dev = container_of(cc, struct hdac_stream, cc); return snd_hdac_chip_readl(azx_dev->bus, WALLCLK); } static void azx_timecounter_init(struct hdac_stream *azx_dev, bool force, u64 last) { struct timecounter *tc = &azx_dev->tc; struct cyclecounter *cc = &azx_dev->cc; u64 nsec; cc->read = azx_cc_read; cc->mask = CLOCKSOURCE_MASK(32); /* * Calculate the optimal mult/shift values. The counter wraps * around after ~178.9 seconds. */ clocks_calc_mult_shift(&cc->mult, &cc->shift, 24000000, NSEC_PER_SEC, 178); nsec = 0; /* audio time is elapsed time since trigger */ timecounter_init(tc, cc, nsec); if (force) { /* * force timecounter to use predefined value, * used for synchronized starts */ tc->cycle_last = last; } } /** * snd_hdac_stream_timecounter_init - initialize time counter * @azx_dev: HD-audio core stream (master stream) * @streams: bit flags of streams to set up * * Initializes the time counter of streams marked by the bit flags (each * bit corresponds to the stream index). * The trigger timestamp of PCM substream assigned to the given stream is * updated accordingly, too. */ void snd_hdac_stream_timecounter_init(struct hdac_stream *azx_dev, unsigned int streams) { struct hdac_bus *bus = azx_dev->bus; struct snd_pcm_runtime *runtime = azx_dev->substream->runtime; struct hdac_stream *s; bool inited = false; u64 cycle_last = 0; list_for_each_entry(s, &bus->stream_list, list) { if ((streams & (1 << s->index))) { azx_timecounter_init(s, inited, cycle_last); if (!inited) { inited = true; cycle_last = s->tc.cycle_last; } } } snd_pcm_gettime(runtime, &runtime->trigger_tstamp); runtime->trigger_tstamp_latched = true; } EXPORT_SYMBOL_GPL(snd_hdac_stream_timecounter_init); /** * snd_hdac_stream_sync_trigger - turn on/off stream sync register * @azx_dev: HD-audio core stream (master stream) * @set: true = set, false = clear * @streams: bit flags of streams to sync * @reg: the stream sync register address */ void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, unsigned int streams, unsigned int reg) { struct hdac_bus *bus = azx_dev->bus; unsigned int val; if (!reg) reg = AZX_REG_SSYNC; val = _snd_hdac_chip_readl(bus, reg); if (set) val |= streams; else val &= ~streams; _snd_hdac_chip_writel(bus, reg, val); } EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger); /** * snd_hdac_stream_sync - sync with start/stop trigger operation * @azx_dev: HD-audio core stream (master stream) * @start: true = start, false = stop * @streams: bit flags of streams to sync * * For @start = true, wait until all FIFOs get ready. * For @start = false, wait until all RUN bits are cleared. */ void snd_hdac_stream_sync(struct hdac_stream *azx_dev, bool start, unsigned int streams) { struct hdac_bus *bus = azx_dev->bus; int nwait, timeout; struct hdac_stream *s; for (timeout = 5000; timeout; timeout--) { nwait = 0; list_for_each_entry(s, &bus->stream_list, list) { if (!(streams & (1 << s->index))) continue; if (start) { /* check FIFO gets ready */ if (!(snd_hdac_stream_readb(s, SD_STS) & SD_STS_FIFO_READY)) nwait++; } else { /* check RUN bit is cleared */ if (snd_hdac_stream_readb(s, SD_CTL) & SD_CTL_DMA_START) { nwait++; /* * Perform stream reset if DMA RUN * bit not cleared within given timeout */ if (timeout == 1) snd_hdac_stream_reset(s); } } } if (!nwait) break; cpu_relax(); } } EXPORT_SYMBOL_GPL(snd_hdac_stream_sync); /** * snd_hdac_stream_spbcap_enable - enable SPIB for a stream * @bus: HD-audio core bus * @enable: flag to enable/disable SPIB * @index: stream index for which SPIB need to be enabled */ void snd_hdac_stream_spbcap_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); return; } mask |= (1 << index); if (enable) snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, mask); else snd_hdac_updatel(bus->spbcap, AZX_REG_SPB_SPBFCCTL, mask, 0); } EXPORT_SYMBOL_GPL(snd_hdac_stream_spbcap_enable); /** * snd_hdac_stream_set_spib - sets the spib value of a stream * @bus: HD-audio core bus * @azx_dev: hdac_stream * @value: spib value to set */ int snd_hdac_stream_set_spib(struct hdac_bus *bus, struct hdac_stream *azx_dev, u32 value) { if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); return -EINVAL; } writel(value, azx_dev->spib_addr); return 0; } EXPORT_SYMBOL_GPL(snd_hdac_stream_set_spib); /** * snd_hdac_stream_get_spbmaxfifo - gets the spib value of a stream * @bus: HD-audio core bus * @azx_dev: hdac_stream * * Return maxfifo for the stream */ int snd_hdac_stream_get_spbmaxfifo(struct hdac_bus *bus, struct hdac_stream *azx_dev) { if (!bus->spbcap) { dev_err(bus->dev, "Address of SPB capability is NULL\n"); return -EINVAL; } return readl(azx_dev->fifo_addr); } EXPORT_SYMBOL_GPL(snd_hdac_stream_get_spbmaxfifo); /** * snd_hdac_stream_drsm_enable - enable DMA resume for a stream * @bus: HD-audio core bus * @enable: flag to enable/disable DRSM * @index: stream index for which DRSM need to be enabled */ void snd_hdac_stream_drsm_enable(struct hdac_bus *bus, bool enable, int index) { u32 mask = 0; if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); return; } mask |= (1 << index); if (enable) snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, mask); else snd_hdac_updatel(bus->drsmcap, AZX_REG_DRSM_CTL, mask, 0); } EXPORT_SYMBOL_GPL(snd_hdac_stream_drsm_enable); /* * snd_hdac_stream_wait_drsm - wait for HW to clear RSM for a stream * @azx_dev: HD-audio core stream to await RSM for * * Returns 0 on success and -ETIMEDOUT upon a timeout. */ int snd_hdac_stream_wait_drsm(struct hdac_stream *azx_dev) { struct hdac_bus *bus = azx_dev->bus; u32 mask, reg; int ret; mask = 1 << azx_dev->index; ret = read_poll_timeout(snd_hdac_reg_readl, reg, !(reg & mask), 250, 2000, false, bus, bus->drsmcap + AZX_REG_DRSM_CTL); if (ret) dev_dbg(bus->dev, "polling RSM 0x%08x failed: %d\n", mask, ret); return ret; } EXPORT_SYMBOL_GPL(snd_hdac_stream_wait_drsm); /** * snd_hdac_stream_set_dpibr - sets the dpibr value of a stream * @bus: HD-audio core bus * @azx_dev: hdac_stream * @value: dpib value to set */ int snd_hdac_stream_set_dpibr(struct hdac_bus *bus, struct hdac_stream *azx_dev, u32 value) { if (!bus->drsmcap) { dev_err(bus->dev, "Address of DRSM capability is NULL\n"); return -EINVAL; } writel(value, azx_dev->dpibr_addr); return 0; } EXPORT_SYMBOL_GPL(snd_hdac_stream_set_dpibr); /** * snd_hdac_stream_set_lpib - sets the lpib value of a stream * @azx_dev: hdac_stream * @value: lpib value to set */ int snd_hdac_stream_set_lpib(struct hdac_stream *azx_dev, u32 value) { snd_hdac_stream_writel(azx_dev, SD_LPIB, value); return 0; } EXPORT_SYMBOL_GPL(snd_hdac_stream_set_lpib); #ifdef CONFIG_SND_HDA_DSP_LOADER /** * snd_hdac_dsp_prepare - prepare for DSP loading * @azx_dev: HD-audio core stream used for DSP loading * @format: HD-audio stream format * @byte_size: data chunk byte size * @bufp: allocated buffer * * Allocate the buffer for the given size and set up the given stream for * DSP loading. Returns the stream tag (>= 0), or a negative error code. */ int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format, unsigned int byte_size, struct snd_dma_buffer *bufp) { struct hdac_bus *bus = azx_dev->bus; __le32 *bdl; int err; snd_hdac_dsp_lock(azx_dev); spin_lock_irq(&bus->reg_lock); if (azx_dev->running || azx_dev->locked) { spin_unlock_irq(&bus->reg_lock); err = -EBUSY; goto unlock; } azx_dev->locked = true; spin_unlock_irq(&bus->reg_lock); err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, bus->dev, byte_size, bufp); if (err < 0) goto err_alloc; azx_dev->substream = NULL; azx_dev->bufsize = byte_size; azx_dev->period_bytes = byte_size; azx_dev->format_val = format; snd_hdac_stream_reset(azx_dev); /* reset BDL address */ snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0); snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0); azx_dev->frags = 0; bdl = (__le32 *)azx_dev->bdl.area; err = setup_bdle(bus, bufp, azx_dev, &bdl, 0, byte_size, 0); if (err < 0) goto error; snd_hdac_stream_setup(azx_dev, true); snd_hdac_dsp_unlock(azx_dev); return azx_dev->stream_tag; error: snd_dma_free_pages(bufp); err_alloc: spin_lock_irq(&bus->reg_lock); azx_dev->locked = false; spin_unlock_irq(&bus->reg_lock); unlock: snd_hdac_dsp_unlock(azx_dev); return err; } EXPORT_SYMBOL_GPL(snd_hdac_dsp_prepare); /** * snd_hdac_dsp_trigger - start / stop DSP loading * @azx_dev: HD-audio core stream used for DSP loading * @start: trigger start or stop */ void snd_hdac_dsp_trigger(struct hdac_stream *azx_dev, bool start) { if (start) snd_hdac_stream_start(azx_dev); else snd_hdac_stream_stop(azx_dev); } EXPORT_SYMBOL_GPL(snd_hdac_dsp_trigger); /** * snd_hdac_dsp_cleanup - clean up the stream from DSP loading to normal * @azx_dev: HD-audio core stream used for DSP loading * @dmab: buffer used by DSP loading */ void snd_hdac_dsp_cleanup(struct hdac_stream *azx_dev, struct snd_dma_buffer *dmab) { struct hdac_bus *bus = azx_dev->bus; if (!dmab->area || !azx_dev->locked) return; snd_hdac_dsp_lock(azx_dev); /* reset BDL address */ snd_hdac_stream_writel(azx_dev, SD_BDLPL, 0); snd_hdac_stream_writel(azx_dev, SD_BDLPU, 0); snd_hdac_stream_writel(azx_dev, SD_CTL, 0); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; azx_dev->format_val = 0; snd_dma_free_pages(dmab); dmab->area = NULL; spin_lock_irq(&bus->reg_lock); azx_dev->locked = false; spin_unlock_irq(&bus->reg_lock); snd_hdac_dsp_unlock(azx_dev); } EXPORT_SYMBOL_GPL(snd_hdac_dsp_cleanup); #endif /* CONFIG_SND_HDA_DSP_LOADER */ |