| 1 | /* |
| 2 | * Copyright (C) 2010 Red Hat, Inc. |
| 3 | * |
| 4 | * written by Gerd Hoffmann <kraxel@redhat.com> |
| 5 | * |
| 6 | * This program is free software; you can redistribute it and/or |
| 7 | * modify it under the terms of the GNU General Public License as |
| 8 | * published by the Free Software Foundation; either version 2 or |
| 9 | * (at your option) version 3 of the License. |
| 10 | * |
| 11 | * This program is distributed in the hope that it will be useful, |
| 12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| 14 | * GNU General Public License for more details. |
| 15 | * |
| 16 | * You should have received a copy of the GNU General Public License |
| 17 | * along with this program; if not, see <http://www.gnu.org/licenses/>. |
| 18 | */ |
| 19 | |
| 20 | #include "qemu/osdep.h" |
| 21 | #include "hw/pci/pci.h" |
| 22 | #include "hw/core/qdev-properties.h" |
| 23 | #include "intel-hda.h" |
| 24 | #include "migration/vmstate.h" |
| 25 | #include "qemu/host-utils.h" |
| 26 | #include "qemu/module.h" |
| 27 | #include "intel-hda-defs.h" |
| 28 | #include "qemu/audio.h" |
| 29 | #include "trace.h" |
| 30 | #include "qom/object.h" |
| 31 | |
| 32 | /* -------------------------------------------------------------------------- */ |
| 33 | |
| 34 | typedef struct desc_param { |
| 35 | uint32_t id; |
| 36 | uint32_t val; |
| 37 | } desc_param; |
| 38 | |
| 39 | typedef struct desc_node { |
| 40 | uint32_t nid; |
| 41 | const char *name; |
| 42 | const desc_param *params; |
| 43 | uint32_t nparams; |
| 44 | uint32_t config; |
| 45 | uint32_t pinctl; |
| 46 | uint32_t *conn; |
| 47 | uint32_t stindex; |
| 48 | } desc_node; |
| 49 | |
| 50 | typedef struct desc_codec { |
| 51 | const char *name; |
| 52 | uint32_t iid; |
| 53 | const desc_node *nodes; |
| 54 | uint32_t nnodes; |
| 55 | } desc_codec; |
| 56 | |
| 57 | static const desc_param* hda_codec_find_param(const desc_node *node, uint32_t id) |
| 58 | { |
| 59 | int i; |
| 60 | |
| 61 | for (i = 0; i < node->nparams; i++) { |
| 62 | if (node->params[i].id == id) { |
| 63 | return &node->params[i]; |
| 64 | } |
| 65 | } |
| 66 | return NULL; |
| 67 | } |
| 68 | |
| 69 | static const desc_node* hda_codec_find_node(const desc_codec *codec, uint32_t nid) |
| 70 | { |
| 71 | int i; |
| 72 | |
| 73 | for (i = 0; i < codec->nnodes; i++) { |
| 74 | if (codec->nodes[i].nid == nid) { |
| 75 | return &codec->nodes[i]; |
| 76 | } |
| 77 | } |
| 78 | return NULL; |
| 79 | } |
| 80 | |
| 81 | static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as) |
| 82 | { |
| 83 | if (format & AC_FMT_TYPE_NON_PCM) { |
| 84 | return; |
| 85 | } |
| 86 | |
| 87 | as->freq = (format & AC_FMT_BASE_44K) ? 44100 : 48000; |
| 88 | |
| 89 | switch ((format & AC_FMT_MULT_MASK) >> AC_FMT_MULT_SHIFT) { |
| 90 | case 1: as->freq *= 2; break; |
| 91 | case 2: as->freq *= 3; break; |
| 92 | case 3: as->freq *= 4; break; |
| 93 | } |
| 94 | |
| 95 | switch ((format & AC_FMT_DIV_MASK) >> AC_FMT_DIV_SHIFT) { |
| 96 | case 1: as->freq /= 2; break; |
| 97 | case 2: as->freq /= 3; break; |
| 98 | case 3: as->freq /= 4; break; |
| 99 | case 4: as->freq /= 5; break; |
| 100 | case 5: as->freq /= 6; break; |
| 101 | case 6: as->freq /= 7; break; |
| 102 | case 7: as->freq /= 8; break; |
| 103 | } |
| 104 | |
| 105 | switch (format & AC_FMT_BITS_MASK) { |
| 106 | case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break; |
| 107 | case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break; |
| 108 | case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break; |
| 109 | } |
| 110 | |
| 111 | as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1; |
| 112 | } |
| 113 | |
| 114 | /* -------------------------------------------------------------------------- */ |
| 115 | /* |
| 116 | * HDA codec descriptions |
| 117 | */ |
| 118 | |
| 119 | /* some defines */ |
| 120 | |
| 121 | #define QEMU_HDA_ID_VENDOR 0x1af4 |
| 122 | #define QEMU_HDA_PCM_FORMATS (AC_SUPPCM_BITS_16 | \ |
| 123 | 0x1fc /* 16 -> 96 kHz */) |
| 124 | #define QEMU_HDA_AMP_NONE (0) |
| 125 | #define QEMU_HDA_AMP_STEPS 0x4a |
| 126 | |
| 127 | #define PARAM mixemu |
| 128 | #define HDA_MIXER |
| 129 | #include "hda-codec-common.h" |
| 130 | |
| 131 | #define PARAM nomixemu |
| 132 | #include "hda-codec-common.h" |
| 133 | |
| 134 | #define HDA_TIMER_TICKS (SCALE_MS) |
| 135 | #define B_SIZE sizeof(st->buf) |
| 136 | #define B_MASK (sizeof(st->buf) - 1) |
| 137 | |
| 138 | /* -------------------------------------------------------------------------- */ |
| 139 | |
| 140 | static const char *fmt2name[] = { |
| 141 | [ AUDIO_FORMAT_U8 ] = "PCM-U8", |
| 142 | [ AUDIO_FORMAT_S8 ] = "PCM-S8", |
| 143 | [ AUDIO_FORMAT_U16 ] = "PCM-U16", |
| 144 | [ AUDIO_FORMAT_S16 ] = "PCM-S16", |
| 145 | [ AUDIO_FORMAT_U32 ] = "PCM-U32", |
| 146 | [ AUDIO_FORMAT_S32 ] = "PCM-S32", |
| 147 | }; |
| 148 | |
| 149 | #define TYPE_HDA_AUDIO "hda-audio" |
| 150 | OBJECT_DECLARE_SIMPLE_TYPE(HDAAudioState, HDA_AUDIO) |
| 151 | |
| 152 | typedef struct HDAAudioStream HDAAudioStream; |
| 153 | |
| 154 | struct HDAAudioStream { |
| 155 | HDAAudioState *state; |
| 156 | const desc_node *node; |
| 157 | bool output, running; |
| 158 | uint32_t stream; |
| 159 | uint32_t channel; |
| 160 | uint32_t format; |
| 161 | uint32_t gain_left, gain_right; |
| 162 | bool mute_left, mute_right; |
| 163 | struct audsettings as; |
| 164 | union { |
| 165 | SWVoiceIn *in; |
| 166 | SWVoiceOut *out; |
| 167 | } voice; |
| 168 | uint8_t compat_buf[HDA_BUFFER_SIZE]; |
| 169 | uint32_t compat_bpos; |
| 170 | uint8_t buf[8192]; /* size must be power of two */ |
| 171 | int64_t rpos; |
| 172 | int64_t wpos; |
| 173 | QEMUTimer *buft; |
| 174 | int64_t buft_start; |
| 175 | }; |
| 176 | |
| 177 | struct HDAAudioState { |
| 178 | HDACodecDevice hda; |
| 179 | const char *name; |
| 180 | |
| 181 | AudioBackend *audio_be; |
| 182 | const desc_codec *desc; |
| 183 | HDAAudioStream st[4]; |
| 184 | bool running_compat[16]; |
| 185 | bool running_real[2 * 16]; |
| 186 | |
| 187 | /* properties */ |
| 188 | uint32_t debug; |
| 189 | bool mixer; |
| 190 | }; |
| 191 | |
| 192 | static inline uint32_t hda_bytes_per_second(HDAAudioStream *st) |
| 193 | { |
| 194 | return 2 * (uint32_t)st->as.nchannels * (uint32_t)st->as.freq; |
| 195 | } |
| 196 | |
| 197 | static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos) |
| 198 | { |
| 199 | int64_t limit = B_SIZE / 8; |
| 200 | int64_t corr = 0; |
| 201 | |
| 202 | if (target_pos > limit) { |
| 203 | corr = HDA_TIMER_TICKS; |
| 204 | } |
| 205 | if (target_pos < -limit) { |
| 206 | corr = -HDA_TIMER_TICKS; |
| 207 | } |
| 208 | if (target_pos < -(2 * limit)) { |
| 209 | corr = -(4 * HDA_TIMER_TICKS); |
| 210 | } |
| 211 | if (corr == 0) { |
| 212 | return; |
| 213 | } |
| 214 | |
| 215 | trace_hda_audio_adjust(st->node->name, target_pos); |
| 216 | st->buft_start += corr; |
| 217 | } |
| 218 | |
| 219 | static void hda_audio_input_timer(void *opaque) |
| 220 | { |
| 221 | HDAAudioStream *st = opaque; |
| 222 | |
| 223 | int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); |
| 224 | |
| 225 | int64_t uptime = now - st->buft_start; |
| 226 | int64_t wpos = st->wpos; |
| 227 | int64_t rpos = st->rpos; |
| 228 | int64_t wanted_rpos; |
| 229 | |
| 230 | if (uptime <= 0) { |
| 231 | /* wanted_rpos <= 0 */ |
| 232 | goto out_timer; |
| 233 | } |
| 234 | |
| 235 | wanted_rpos = muldiv64(uptime, hda_bytes_per_second(st), |
| 236 | NANOSECONDS_PER_SECOND); |
| 237 | wanted_rpos &= -4; /* IMPORTANT! clip to frames */ |
| 238 | |
| 239 | if (wanted_rpos <= rpos) { |
| 240 | /* we already transmitted the data */ |
| 241 | goto out_timer; |
| 242 | } |
| 243 | |
| 244 | int64_t to_transfer = MIN(wpos - rpos, wanted_rpos - rpos); |
| 245 | while (to_transfer) { |
| 246 | uint32_t start = (rpos & B_MASK); |
| 247 | uint32_t chunk = MIN(B_SIZE - start, to_transfer); |
| 248 | int rc = hda_codec_xfer( |
| 249 | &st->state->hda, st->stream, false, st->buf + start, chunk); |
| 250 | if (!rc) { |
| 251 | break; |
| 252 | } |
| 253 | rpos += chunk; |
| 254 | to_transfer -= chunk; |
| 255 | st->rpos += chunk; |
| 256 | } |
| 257 | |
| 258 | out_timer: |
| 259 | |
| 260 | if (st->running) { |
| 261 | timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS); |
| 262 | } |
| 263 | } |
| 264 | |
| 265 | static void hda_audio_input_cb(void *opaque, int avail) |
| 266 | { |
| 267 | HDAAudioStream *st = opaque; |
| 268 | |
| 269 | int64_t wpos = st->wpos; |
| 270 | int64_t rpos = st->rpos; |
| 271 | |
| 272 | int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail); |
| 273 | |
| 274 | while (to_transfer) { |
| 275 | uint32_t start = (uint32_t) (wpos & B_MASK); |
| 276 | uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer); |
| 277 | uint32_t read = audio_be_read(st->state->audio_be, st->voice.in, |
| 278 | st->buf + start, chunk); |
| 279 | wpos += read; |
| 280 | to_transfer -= read; |
| 281 | st->wpos += read; |
| 282 | if (chunk != read) { |
| 283 | break; |
| 284 | } |
| 285 | } |
| 286 | |
| 287 | hda_timer_sync_adjust(st, -((wpos - rpos) - (B_SIZE >> 1))); |
| 288 | } |
| 289 | |
| 290 | static void hda_audio_output_timer(void *opaque) |
| 291 | { |
| 292 | HDAAudioStream *st = opaque; |
| 293 | |
| 294 | int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); |
| 295 | |
| 296 | int64_t uptime = now - st->buft_start; |
| 297 | int64_t wpos = st->wpos; |
| 298 | int64_t rpos = st->rpos; |
| 299 | int64_t wanted_wpos; |
| 300 | |
| 301 | if (uptime <= 0) { |
| 302 | /* wanted_wpos <= 0 */ |
| 303 | goto out_timer; |
| 304 | } |
| 305 | |
| 306 | wanted_wpos = muldiv64(uptime, hda_bytes_per_second(st), |
| 307 | NANOSECONDS_PER_SECOND); |
| 308 | wanted_wpos &= -4; /* IMPORTANT! clip to frames */ |
| 309 | |
| 310 | if (wanted_wpos <= wpos) { |
| 311 | /* we already received the data */ |
| 312 | goto out_timer; |
| 313 | } |
| 314 | |
| 315 | int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos); |
| 316 | while (to_transfer) { |
| 317 | uint32_t start = (wpos & B_MASK); |
| 318 | uint32_t chunk = MIN(B_SIZE - start, to_transfer); |
| 319 | int rc = hda_codec_xfer( |
| 320 | &st->state->hda, st->stream, true, st->buf + start, chunk); |
| 321 | if (!rc) { |
| 322 | break; |
| 323 | } |
| 324 | wpos += chunk; |
| 325 | to_transfer -= chunk; |
| 326 | st->wpos += chunk; |
| 327 | } |
| 328 | |
| 329 | out_timer: |
| 330 | |
| 331 | if (st->running) { |
| 332 | timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS); |
| 333 | } |
| 334 | } |
| 335 | |
| 336 | static void hda_audio_output_cb(void *opaque, int avail) |
| 337 | { |
| 338 | HDAAudioStream *st = opaque; |
| 339 | |
| 340 | int64_t wpos = st->wpos; |
| 341 | int64_t rpos = st->rpos; |
| 342 | |
| 343 | int64_t to_transfer = MIN(wpos - rpos, avail); |
| 344 | |
| 345 | if (wpos - rpos == B_SIZE) { |
| 346 | /* drop buffer, reset timer adjust */ |
| 347 | st->rpos = 0; |
| 348 | st->wpos = 0; |
| 349 | st->buft_start = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); |
| 350 | trace_hda_audio_overrun(st->node->name); |
| 351 | return; |
| 352 | } |
| 353 | |
| 354 | while (to_transfer) { |
| 355 | uint32_t start = (uint32_t) (rpos & B_MASK); |
| 356 | uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer); |
| 357 | uint32_t written = audio_be_write(st->state->audio_be, st->voice.out, |
| 358 | st->buf + start, chunk); |
| 359 | rpos += written; |
| 360 | to_transfer -= written; |
| 361 | st->rpos += written; |
| 362 | if (chunk != written) { |
| 363 | break; |
| 364 | } |
| 365 | } |
| 366 | |
| 367 | hda_timer_sync_adjust(st, (wpos - rpos) - (B_SIZE >> 1)); |
| 368 | } |
| 369 | |
| 370 | static void hda_audio_set_running(HDAAudioStream *st, bool running) |
| 371 | { |
| 372 | if (st->node == NULL) { |
| 373 | return; |
| 374 | } |
| 375 | if (st->running == running) { |
| 376 | return; |
| 377 | } |
| 378 | st->running = running; |
| 379 | trace_hda_audio_running(st->node->name, st->stream, st->running); |
| 380 | if (running) { |
| 381 | int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL); |
| 382 | st->rpos = 0; |
| 383 | st->wpos = 0; |
| 384 | st->buft_start = now; |
| 385 | timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS); |
| 386 | } else { |
| 387 | timer_del(st->buft); |
| 388 | } |
| 389 | if (st->output) { |
| 390 | audio_be_set_active_out(st->state->audio_be, st->voice.out, st->running); |
| 391 | } else { |
| 392 | audio_be_set_active_in(st->state->audio_be, st->voice.in, st->running); |
| 393 | } |
| 394 | } |
| 395 | |
| 396 | static void hda_audio_set_amp(HDAAudioStream *st) |
| 397 | { |
| 398 | bool muted; |
| 399 | uint32_t left, right; |
| 400 | |
| 401 | if (st->node == NULL) { |
| 402 | return; |
| 403 | } |
| 404 | |
| 405 | muted = st->mute_left && st->mute_right; |
| 406 | left = st->mute_left ? 0 : st->gain_left; |
| 407 | right = st->mute_right ? 0 : st->gain_right; |
| 408 | |
| 409 | left = left * 255 / QEMU_HDA_AMP_STEPS; |
| 410 | right = right * 255 / QEMU_HDA_AMP_STEPS; |
| 411 | |
| 412 | if (!st->state->mixer) { |
| 413 | return; |
| 414 | } |
| 415 | if (st->output) { |
| 416 | audio_be_set_volume_out_lr(st->state->audio_be, st->voice.out, |
| 417 | muted, left, right); |
| 418 | } else { |
| 419 | audio_be_set_volume_in_lr(st->state->audio_be, st->voice.in, |
| 420 | muted, left, right); |
| 421 | } |
| 422 | } |
| 423 | |
| 424 | static void hda_audio_setup(HDAAudioStream *st) |
| 425 | { |
| 426 | audio_callback_fn cb; |
| 427 | |
| 428 | if (st->node == NULL) { |
| 429 | return; |
| 430 | } |
| 431 | |
| 432 | trace_hda_audio_format(st->node->name, st->as.nchannels, |
| 433 | fmt2name[st->as.fmt], st->as.freq); |
| 434 | |
| 435 | if (st->output) { |
| 436 | cb = hda_audio_output_cb; |
| 437 | timer_del(st->buft); |
| 438 | st->voice.out = audio_be_open_out(st->state->audio_be, st->voice.out, |
| 439 | st->node->name, st, cb, &st->as); |
| 440 | } else { |
| 441 | cb = hda_audio_input_cb; |
| 442 | timer_del(st->buft); |
| 443 | st->voice.in = audio_be_open_in(st->state->audio_be, st->voice.in, |
| 444 | st->node->name, st, cb, &st->as); |
| 445 | } |
| 446 | } |
| 447 | |
| 448 | static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data) |
| 449 | { |
| 450 | HDAAudioState *a = HDA_AUDIO(hda); |
| 451 | HDAAudioStream *st; |
| 452 | const desc_node *node = NULL; |
| 453 | const desc_param *param; |
| 454 | uint32_t verb, payload, response, count, shift; |
| 455 | |
| 456 | if ((data & 0x70000) == 0x70000) { |
| 457 | /* 12/8 id/payload */ |
| 458 | verb = (data >> 8) & 0xfff; |
| 459 | payload = data & 0x00ff; |
| 460 | } else { |
| 461 | /* 4/16 id/payload */ |
| 462 | verb = (data >> 8) & 0xf00; |
| 463 | payload = data & 0xffff; |
| 464 | } |
| 465 | |
| 466 | node = hda_codec_find_node(a->desc, nid); |
| 467 | if (node == NULL) { |
| 468 | goto fail; |
| 469 | } |
| 470 | dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n", |
| 471 | __func__, nid, node->name, verb, payload); |
| 472 | |
| 473 | switch (verb) { |
| 474 | /* all nodes */ |
| 475 | case AC_VERB_PARAMETERS: |
| 476 | param = hda_codec_find_param(node, payload); |
| 477 | if (param == NULL) { |
| 478 | goto fail; |
| 479 | } |
| 480 | hda_codec_response(hda, true, param->val); |
| 481 | break; |
| 482 | case AC_VERB_GET_SUBSYSTEM_ID: |
| 483 | hda_codec_response(hda, true, a->desc->iid); |
| 484 | break; |
| 485 | |
| 486 | /* all functions */ |
| 487 | case AC_VERB_GET_CONNECT_LIST: |
| 488 | param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN); |
| 489 | count = param ? param->val : 0; |
| 490 | response = 0; |
| 491 | shift = 0; |
| 492 | while (payload < count && shift < 32) { |
| 493 | response |= node->conn[payload] << shift; |
| 494 | payload++; |
| 495 | shift += 8; |
| 496 | } |
| 497 | hda_codec_response(hda, true, response); |
| 498 | break; |
| 499 | |
| 500 | /* pin widget */ |
| 501 | case AC_VERB_GET_CONFIG_DEFAULT: |
| 502 | hda_codec_response(hda, true, node->config); |
| 503 | break; |
| 504 | case AC_VERB_GET_PIN_WIDGET_CONTROL: |
| 505 | hda_codec_response(hda, true, node->pinctl); |
| 506 | break; |
| 507 | case AC_VERB_SET_PIN_WIDGET_CONTROL: |
| 508 | if (node->pinctl != payload) { |
| 509 | dprint(a, 1, "unhandled pin control bit\n"); |
| 510 | } |
| 511 | hda_codec_response(hda, true, 0); |
| 512 | break; |
| 513 | |
| 514 | /* audio in/out widget */ |
| 515 | case AC_VERB_SET_CHANNEL_STREAMID: |
| 516 | st = a->st + node->stindex; |
| 517 | if (st->node == NULL) { |
| 518 | goto fail; |
| 519 | } |
| 520 | hda_audio_set_running(st, false); |
| 521 | st->stream = (payload >> 4) & 0x0f; |
| 522 | st->channel = payload & 0x0f; |
| 523 | dprint(a, 2, "%s: stream %d, channel %d\n", |
| 524 | st->node->name, st->stream, st->channel); |
| 525 | hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]); |
| 526 | hda_codec_response(hda, true, 0); |
| 527 | break; |
| 528 | case AC_VERB_GET_CONV: |
| 529 | st = a->st + node->stindex; |
| 530 | if (st->node == NULL) { |
| 531 | goto fail; |
| 532 | } |
| 533 | response = st->stream << 4 | st->channel; |
| 534 | hda_codec_response(hda, true, response); |
| 535 | break; |
| 536 | case AC_VERB_SET_STREAM_FORMAT: |
| 537 | st = a->st + node->stindex; |
| 538 | if (st->node == NULL) { |
| 539 | goto fail; |
| 540 | } |
| 541 | st->format = payload; |
| 542 | hda_codec_parse_fmt(st->format, &st->as); |
| 543 | hda_audio_setup(st); |
| 544 | hda_codec_response(hda, true, 0); |
| 545 | break; |
| 546 | case AC_VERB_GET_STREAM_FORMAT: |
| 547 | st = a->st + node->stindex; |
| 548 | if (st->node == NULL) { |
| 549 | goto fail; |
| 550 | } |
| 551 | hda_codec_response(hda, true, st->format); |
| 552 | break; |
| 553 | case AC_VERB_GET_AMP_GAIN_MUTE: |
| 554 | st = a->st + node->stindex; |
| 555 | if (st->node == NULL) { |
| 556 | goto fail; |
| 557 | } |
| 558 | if (payload & AC_AMP_GET_LEFT) { |
| 559 | response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0); |
| 560 | } else { |
| 561 | response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0); |
| 562 | } |
| 563 | hda_codec_response(hda, true, response); |
| 564 | break; |
| 565 | case AC_VERB_SET_AMP_GAIN_MUTE: |
| 566 | st = a->st + node->stindex; |
| 567 | if (st->node == NULL) { |
| 568 | goto fail; |
| 569 | } |
| 570 | dprint(a, 1, "amp (%s): %s%s%s%s index %d gain %3d %s\n", |
| 571 | st->node->name, |
| 572 | (payload & AC_AMP_SET_OUTPUT) ? "o" : "-", |
| 573 | (payload & AC_AMP_SET_INPUT) ? "i" : "-", |
| 574 | (payload & AC_AMP_SET_LEFT) ? "l" : "-", |
| 575 | (payload & AC_AMP_SET_RIGHT) ? "r" : "-", |
| 576 | (payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT, |
| 577 | (payload & AC_AMP_GAIN), |
| 578 | (payload & AC_AMP_MUTE) ? "muted" : ""); |
| 579 | if (payload & AC_AMP_SET_LEFT) { |
| 580 | st->gain_left = payload & AC_AMP_GAIN; |
| 581 | st->mute_left = payload & AC_AMP_MUTE; |
| 582 | } |
| 583 | if (payload & AC_AMP_SET_RIGHT) { |
| 584 | st->gain_right = payload & AC_AMP_GAIN; |
| 585 | st->mute_right = payload & AC_AMP_MUTE; |
| 586 | } |
| 587 | hda_audio_set_amp(st); |
| 588 | hda_codec_response(hda, true, 0); |
| 589 | break; |
| 590 | |
| 591 | /* not supported */ |
| 592 | case AC_VERB_SET_POWER_STATE: |
| 593 | case AC_VERB_GET_POWER_STATE: |
| 594 | case AC_VERB_GET_SDI_SELECT: |
| 595 | hda_codec_response(hda, true, 0); |
| 596 | break; |
| 597 | default: |
| 598 | goto fail; |
| 599 | } |
| 600 | return; |
| 601 | |
| 602 | fail: |
| 603 | dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n", |
| 604 | __func__, nid, node ? node->name : "?", verb, payload); |
| 605 | hda_codec_response(hda, true, 0); |
| 606 | } |
| 607 | |
| 608 | static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output) |
| 609 | { |
| 610 | HDAAudioState *a = HDA_AUDIO(hda); |
| 611 | int s; |
| 612 | |
| 613 | a->running_compat[stnr] = running; |
| 614 | a->running_real[output * 16 + stnr] = running; |
| 615 | for (s = 0; s < ARRAY_SIZE(a->st); s++) { |
| 616 | if (a->st[s].node == NULL) { |
| 617 | continue; |
| 618 | } |
| 619 | if (a->st[s].output != output) { |
| 620 | continue; |
| 621 | } |
| 622 | if (a->st[s].stream != stnr) { |
| 623 | continue; |
| 624 | } |
| 625 | hda_audio_set_running(&a->st[s], running); |
| 626 | } |
| 627 | } |
| 628 | |
| 629 | static void hda_audio_init(HDACodecDevice *hda, |
| 630 | const struct desc_codec *desc, |
| 631 | Error **errp) |
| 632 | { |
| 633 | HDAAudioState *a = HDA_AUDIO(hda); |
| 634 | HDAAudioStream *st; |
| 635 | const desc_node *node; |
| 636 | const desc_param *param; |
| 637 | uint32_t i, type; |
| 638 | |
| 639 | if (!audio_be_check(&a->audio_be, errp)) { |
| 640 | return; |
| 641 | } |
| 642 | |
| 643 | a->desc = desc; |
| 644 | a->name = object_get_typename(OBJECT(a)); |
| 645 | dprint(a, 1, "%s: cad %d\n", __func__, a->hda.cad); |
| 646 | |
| 647 | for (i = 0; i < a->desc->nnodes; i++) { |
| 648 | node = a->desc->nodes + i; |
| 649 | param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP); |
| 650 | if (param == NULL) { |
| 651 | continue; |
| 652 | } |
| 653 | type = (param->val & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; |
| 654 | switch (type) { |
| 655 | case AC_WID_AUD_OUT: |
| 656 | case AC_WID_AUD_IN: |
| 657 | assert(node->stindex < ARRAY_SIZE(a->st)); |
| 658 | st = a->st + node->stindex; |
| 659 | st->state = a; |
| 660 | st->node = node; |
| 661 | if (type == AC_WID_AUD_OUT) { |
| 662 | /* unmute output by default */ |
| 663 | st->gain_left = QEMU_HDA_AMP_STEPS; |
| 664 | st->gain_right = QEMU_HDA_AMP_STEPS; |
| 665 | st->compat_bpos = sizeof(st->compat_buf); |
| 666 | st->output = true; |
| 667 | st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL, |
| 668 | hda_audio_output_timer, st); |
| 669 | } else { |
| 670 | st->output = false; |
| 671 | st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL, |
| 672 | hda_audio_input_timer, st); |
| 673 | } |
| 674 | st->format = AC_FMT_TYPE_PCM | AC_FMT_BITS_16 | |
| 675 | (1 << AC_FMT_CHAN_SHIFT); |
| 676 | hda_codec_parse_fmt(st->format, &st->as); |
| 677 | hda_audio_setup(st); |
| 678 | break; |
| 679 | } |
| 680 | } |
| 681 | } |
| 682 | |
| 683 | static void hda_audio_exit(HDACodecDevice *hda) |
| 684 | { |
| 685 | HDAAudioState *a = HDA_AUDIO(hda); |
| 686 | HDAAudioStream *st; |
| 687 | int i; |
| 688 | |
| 689 | dprint(a, 1, "%s\n", __func__); |
| 690 | for (i = 0; i < ARRAY_SIZE(a->st); i++) { |
| 691 | st = a->st + i; |
| 692 | if (st->node == NULL) { |
| 693 | continue; |
| 694 | } |
| 695 | timer_free(st->buft); |
| 696 | if (st->output) { |
| 697 | audio_be_close_out(a->audio_be, st->voice.out); |
| 698 | } else { |
| 699 | audio_be_close_in(a->audio_be, st->voice.in); |
| 700 | } |
| 701 | } |
| 702 | } |
| 703 | |
| 704 | static int hda_audio_post_load(void *opaque, int version) |
| 705 | { |
| 706 | HDAAudioState *a = opaque; |
| 707 | HDAAudioStream *st; |
| 708 | int i; |
| 709 | |
| 710 | dprint(a, 1, "%s\n", __func__); |
| 711 | if (version == 1) { |
| 712 | /* assume running_compat[] is for output streams */ |
| 713 | for (i = 0; i < ARRAY_SIZE(a->running_compat); i++) |
| 714 | a->running_real[16 + i] = a->running_compat[i]; |
| 715 | } |
| 716 | |
| 717 | for (i = 0; i < ARRAY_SIZE(a->st); i++) { |
| 718 | st = a->st + i; |
| 719 | if (st->node == NULL) |
| 720 | continue; |
| 721 | hda_codec_parse_fmt(st->format, &st->as); |
| 722 | hda_audio_setup(st); |
| 723 | hda_audio_set_amp(st); |
| 724 | hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]); |
| 725 | } |
| 726 | return 0; |
| 727 | } |
| 728 | |
| 729 | static void hda_audio_reset(DeviceState *dev) |
| 730 | { |
| 731 | HDAAudioState *a = HDA_AUDIO(dev); |
| 732 | HDAAudioStream *st; |
| 733 | int i; |
| 734 | |
| 735 | dprint(a, 1, "%s\n", __func__); |
| 736 | for (i = 0; i < ARRAY_SIZE(a->st); i++) { |
| 737 | st = a->st + i; |
| 738 | if (st->node != NULL) { |
| 739 | hda_audio_set_running(st, false); |
| 740 | } |
| 741 | } |
| 742 | } |
| 743 | |
| 744 | static bool vmstate_hda_audio_stream_buf_needed(void *opaque) |
| 745 | { |
| 746 | HDAAudioStream *st = opaque; |
| 747 | return st->state; |
| 748 | } |
| 749 | |
| 750 | static const VMStateDescription vmstate_hda_audio_stream_buf = { |
| 751 | .name = "hda-audio-stream/buffer", |
| 752 | .version_id = 1, |
| 753 | .needed = vmstate_hda_audio_stream_buf_needed, |
| 754 | .fields = (const VMStateField[]) { |
| 755 | VMSTATE_BUFFER(buf, HDAAudioStream), |
| 756 | VMSTATE_INT64(rpos, HDAAudioStream), |
| 757 | VMSTATE_INT64(wpos, HDAAudioStream), |
| 758 | VMSTATE_TIMER_PTR(buft, HDAAudioStream), |
| 759 | VMSTATE_INT64(buft_start, HDAAudioStream), |
| 760 | VMSTATE_END_OF_LIST() |
| 761 | } |
| 762 | }; |
| 763 | |
| 764 | static const VMStateDescription vmstate_hda_audio_stream = { |
| 765 | .name = "hda-audio-stream", |
| 766 | .version_id = 1, |
| 767 | .fields = (const VMStateField[]) { |
| 768 | VMSTATE_UINT32(stream, HDAAudioStream), |
| 769 | VMSTATE_UINT32(channel, HDAAudioStream), |
| 770 | VMSTATE_UINT32(format, HDAAudioStream), |
| 771 | VMSTATE_UINT32(gain_left, HDAAudioStream), |
| 772 | VMSTATE_UINT32(gain_right, HDAAudioStream), |
| 773 | VMSTATE_BOOL(mute_left, HDAAudioStream), |
| 774 | VMSTATE_BOOL(mute_right, HDAAudioStream), |
| 775 | VMSTATE_UINT32(compat_bpos, HDAAudioStream), |
| 776 | VMSTATE_BUFFER(compat_buf, HDAAudioStream), |
| 777 | VMSTATE_END_OF_LIST() |
| 778 | }, |
| 779 | .subsections = (const VMStateDescription * const []) { |
| 780 | &vmstate_hda_audio_stream_buf, |
| 781 | NULL |
| 782 | } |
| 783 | }; |
| 784 | |
| 785 | static const VMStateDescription vmstate_hda_audio = { |
| 786 | .name = "hda-audio", |
| 787 | .version_id = 2, |
| 788 | .post_load = hda_audio_post_load, |
| 789 | .fields = (const VMStateField[]) { |
| 790 | VMSTATE_STRUCT_ARRAY(st, HDAAudioState, 4, 0, |
| 791 | vmstate_hda_audio_stream, |
| 792 | HDAAudioStream), |
| 793 | VMSTATE_BOOL_ARRAY(running_compat, HDAAudioState, 16), |
| 794 | VMSTATE_BOOL_ARRAY_V(running_real, HDAAudioState, 2 * 16, 2), |
| 795 | VMSTATE_END_OF_LIST() |
| 796 | } |
| 797 | }; |
| 798 | |
| 799 | static const Property hda_audio_properties[] = { |
| 800 | DEFINE_AUDIO_PROPERTIES(HDAAudioState, audio_be), |
| 801 | DEFINE_PROP_UINT32("debug", HDAAudioState, debug, 0), |
| 802 | DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer, true), |
| 803 | }; |
| 804 | |
| 805 | static void hda_audio_init_output(HDACodecDevice *hda, Error **errp) |
| 806 | { |
| 807 | HDAAudioState *a = HDA_AUDIO(hda); |
| 808 | const struct desc_codec *desc = &output_mixemu; |
| 809 | |
| 810 | if (!a->mixer) { |
| 811 | desc = &output_nomixemu; |
| 812 | } |
| 813 | |
| 814 | hda_audio_init(hda, desc, errp); |
| 815 | } |
| 816 | |
| 817 | static void hda_audio_init_duplex(HDACodecDevice *hda, Error **errp) |
| 818 | { |
| 819 | HDAAudioState *a = HDA_AUDIO(hda); |
| 820 | const struct desc_codec *desc = &duplex_mixemu; |
| 821 | |
| 822 | if (!a->mixer) { |
| 823 | desc = &duplex_nomixemu; |
| 824 | } |
| 825 | |
| 826 | hda_audio_init(hda, desc, errp); |
| 827 | } |
| 828 | |
| 829 | static void hda_audio_init_micro(HDACodecDevice *hda, Error **errp) |
| 830 | { |
| 831 | HDAAudioState *a = HDA_AUDIO(hda); |
| 832 | const struct desc_codec *desc = µ_mixemu; |
| 833 | |
| 834 | if (!a->mixer) { |
| 835 | desc = µ_nomixemu; |
| 836 | } |
| 837 | |
| 838 | hda_audio_init(hda, desc, errp); |
| 839 | } |
| 840 | |
| 841 | static void hda_audio_base_class_init(ObjectClass *klass, const void *data) |
| 842 | { |
| 843 | DeviceClass *dc = DEVICE_CLASS(klass); |
| 844 | HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass); |
| 845 | |
| 846 | k->exit = hda_audio_exit; |
| 847 | k->command = hda_audio_command; |
| 848 | k->stream = hda_audio_stream; |
| 849 | set_bit(DEVICE_CATEGORY_SOUND, dc->categories); |
| 850 | device_class_set_legacy_reset(dc, hda_audio_reset); |
| 851 | dc->vmsd = &vmstate_hda_audio; |
| 852 | device_class_set_props(dc, hda_audio_properties); |
| 853 | } |
| 854 | |
| 855 | static const TypeInfo hda_audio_info = { |
| 856 | .name = TYPE_HDA_AUDIO, |
| 857 | .parent = TYPE_HDA_CODEC_DEVICE, |
| 858 | .instance_size = sizeof(HDAAudioState), |
| 859 | .class_init = hda_audio_base_class_init, |
| 860 | .abstract = true, |
| 861 | }; |
| 862 | |
| 863 | static void hda_audio_output_class_init(ObjectClass *klass, const void *data) |
| 864 | { |
| 865 | DeviceClass *dc = DEVICE_CLASS(klass); |
| 866 | HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass); |
| 867 | |
| 868 | k->init = hda_audio_init_output; |
| 869 | dc->desc = "HDA Audio Codec, output-only (line-out)"; |
| 870 | } |
| 871 | |
| 872 | static const TypeInfo hda_audio_output_info = { |
| 873 | .name = "hda-output", |
| 874 | .parent = TYPE_HDA_AUDIO, |
| 875 | .class_init = hda_audio_output_class_init, |
| 876 | }; |
| 877 | |
| 878 | static void hda_audio_duplex_class_init(ObjectClass *klass, const void *data) |
| 879 | { |
| 880 | DeviceClass *dc = DEVICE_CLASS(klass); |
| 881 | HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass); |
| 882 | |
| 883 | k->init = hda_audio_init_duplex; |
| 884 | dc->desc = "HDA Audio Codec, duplex (line-out, line-in)"; |
| 885 | } |
| 886 | |
| 887 | static const TypeInfo hda_audio_duplex_info = { |
| 888 | .name = "hda-duplex", |
| 889 | .parent = TYPE_HDA_AUDIO, |
| 890 | .class_init = hda_audio_duplex_class_init, |
| 891 | }; |
| 892 | |
| 893 | static void hda_audio_micro_class_init(ObjectClass *klass, const void *data) |
| 894 | { |
| 895 | DeviceClass *dc = DEVICE_CLASS(klass); |
| 896 | HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass); |
| 897 | |
| 898 | k->init = hda_audio_init_micro; |
| 899 | dc->desc = "HDA Audio Codec, duplex (speaker, microphone)"; |
| 900 | } |
| 901 | |
| 902 | static const TypeInfo hda_audio_micro_info = { |
| 903 | .name = "hda-micro", |
| 904 | .parent = TYPE_HDA_AUDIO, |
| 905 | .class_init = hda_audio_micro_class_init, |
| 906 | }; |
| 907 | |
| 908 | static void hda_audio_register_types(void) |
| 909 | { |
| 910 | type_register_static(&hda_audio_info); |
| 911 | type_register_static(&hda_audio_output_info); |
| 912 | type_register_static(&hda_audio_duplex_info); |
| 913 | type_register_static(&hda_audio_micro_info); |
| 914 | } |
| 915 | |
| 916 | type_init(hda_audio_register_types) |