| 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 "qemu/error-report.h" |
| 30 | #include "qemu/audio.h" |
| 31 | #include "qom/object.h" |
| 32 | #include "trace.h" |
| 33 | |
| 34 | #pragma GCC diagnostic ignored "-Waddress" |
| 35 | |
| 36 | #include "audio_int.h" |
| 37 | |
| 38 | #define TYPE_AUDIO_ALSA "audio-alsa" |
| 39 | OBJECT_DECLARE_SIMPLE_TYPE(AudioALSA, AUDIO_ALSA) |
| 40 | |
| 41 | static AudioBackendClass *audio_alsa_parent_class; |
| 42 | |
| 43 | struct AudioALSA { |
| 44 | AudioMixengBackend parent_obj; |
| 45 | }; |
| 46 | |
| 47 | |
| 48 | struct pollhlp { |
| 49 | snd_pcm_t *handle; |
| 50 | struct pollfd *pfds; |
| 51 | int count; |
| 52 | int mask; |
| 53 | AudioMixengBackend *s; |
| 54 | }; |
| 55 | |
| 56 | typedef struct ALSAVoiceOut { |
| 57 | HWVoiceOut hw; |
| 58 | snd_pcm_t *handle; |
| 59 | struct pollhlp pollhlp; |
| 60 | } ALSAVoiceOut; |
| 61 | |
| 62 | typedef struct ALSAVoiceIn { |
| 63 | HWVoiceIn hw; |
| 64 | snd_pcm_t *handle; |
| 65 | struct pollhlp pollhlp; |
| 66 | } ALSAVoiceIn; |
| 67 | |
| 68 | struct alsa_params_req { |
| 69 | int freq; |
| 70 | snd_pcm_format_t fmt; |
| 71 | int nchannels; |
| 72 | }; |
| 73 | |
| 74 | struct alsa_params_obt { |
| 75 | int freq; |
| 76 | AudioFormat fmt; |
| 77 | int endianness; |
| 78 | int nchannels; |
| 79 | snd_pcm_uframes_t samples; |
| 80 | }; |
| 81 | |
| 82 | static void G_GNUC_PRINTF(2, 3) alsa_logerr(int err, const char *fmt, ...) |
| 83 | { |
| 84 | va_list ap; |
| 85 | |
| 86 | error_printf("alsa: "); |
| 87 | va_start(ap, fmt); |
| 88 | error_vprintf(fmt, ap); |
| 89 | va_end(ap); |
| 90 | error_printf(" Reason: %s", snd_strerror(err)); |
| 91 | error_printf("\n"); |
| 92 | } |
| 93 | |
| 94 | static void G_GNUC_PRINTF(3, 4) alsa_logerr2(int err, const char *typ, |
| 95 | const char *fmt, ...) |
| 96 | { |
| 97 | va_list ap; |
| 98 | |
| 99 | error_printf("alsa: Could not initialize %s:", typ); |
| 100 | va_start(ap, fmt); |
| 101 | error_vprintf(fmt, ap); |
| 102 | va_end(ap); |
| 103 | error_printf(" Reason: %s", snd_strerror(err)); |
| 104 | error_printf("\n"); |
| 105 | } |
| 106 | |
| 107 | static void alsa_fini_poll (struct pollhlp *hlp) |
| 108 | { |
| 109 | int i; |
| 110 | struct pollfd *pfds = hlp->pfds; |
| 111 | |
| 112 | if (pfds) { |
| 113 | for (i = 0; i < hlp->count; ++i) { |
| 114 | qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL); |
| 115 | } |
| 116 | g_free (pfds); |
| 117 | } |
| 118 | hlp->pfds = NULL; |
| 119 | hlp->count = 0; |
| 120 | hlp->handle = NULL; |
| 121 | } |
| 122 | |
| 123 | static void alsa_anal_close1 (snd_pcm_t **handlep) |
| 124 | { |
| 125 | int err = snd_pcm_close (*handlep); |
| 126 | if (err) { |
| 127 | alsa_logerr(err, "Failed to close PCM handle %p", *handlep); |
| 128 | } |
| 129 | *handlep = NULL; |
| 130 | } |
| 131 | |
| 132 | static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp) |
| 133 | { |
| 134 | alsa_fini_poll (hlp); |
| 135 | alsa_anal_close1 (handlep); |
| 136 | } |
| 137 | |
| 138 | static int alsa_recover (snd_pcm_t *handle) |
| 139 | { |
| 140 | int err = snd_pcm_prepare (handle); |
| 141 | if (err < 0) { |
| 142 | alsa_logerr(err, "Failed to prepare handle %p", handle); |
| 143 | return -1; |
| 144 | } |
| 145 | return 0; |
| 146 | } |
| 147 | |
| 148 | static int alsa_resume (snd_pcm_t *handle) |
| 149 | { |
| 150 | int err = snd_pcm_resume (handle); |
| 151 | if (err < 0) { |
| 152 | alsa_logerr(err, "Failed to resume handle %p", handle); |
| 153 | return -1; |
| 154 | } |
| 155 | return 0; |
| 156 | } |
| 157 | |
| 158 | static void alsa_poll_handler (void *opaque) |
| 159 | { |
| 160 | int err, count; |
| 161 | snd_pcm_state_t state; |
| 162 | struct pollhlp *hlp = opaque; |
| 163 | unsigned short revents; |
| 164 | |
| 165 | count = poll (hlp->pfds, hlp->count, 0); |
| 166 | if (count < 0) { |
| 167 | warn_report("alsa_poll_handler: poll %s", strerror(errno)); |
| 168 | return; |
| 169 | } |
| 170 | |
| 171 | if (!count) { |
| 172 | return; |
| 173 | } |
| 174 | |
| 175 | /* XXX: ALSA example uses initial count, not the one returned by |
| 176 | poll, correct? */ |
| 177 | err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds, |
| 178 | hlp->count, &revents); |
| 179 | if (err < 0) { |
| 180 | alsa_logerr(err, "snd_pcm_poll_descriptors_revents"); |
| 181 | return; |
| 182 | } |
| 183 | |
| 184 | if (!(revents & hlp->mask)) { |
| 185 | trace_alsa_revents(revents); |
| 186 | return; |
| 187 | } |
| 188 | |
| 189 | state = snd_pcm_state (hlp->handle); |
| 190 | switch (state) { |
| 191 | case SND_PCM_STATE_SETUP: |
| 192 | alsa_recover (hlp->handle); |
| 193 | break; |
| 194 | |
| 195 | case SND_PCM_STATE_XRUN: |
| 196 | alsa_recover (hlp->handle); |
| 197 | break; |
| 198 | |
| 199 | case SND_PCM_STATE_SUSPENDED: |
| 200 | alsa_resume (hlp->handle); |
| 201 | break; |
| 202 | |
| 203 | case SND_PCM_STATE_PREPARED: |
| 204 | audio_run(hlp->s, "alsa run (prepared)"); |
| 205 | break; |
| 206 | |
| 207 | case SND_PCM_STATE_RUNNING: |
| 208 | audio_run(hlp->s, "alsa run (running)"); |
| 209 | break; |
| 210 | |
| 211 | default: |
| 212 | warn_report("alsa: Unexpected state %d", state); |
| 213 | } |
| 214 | } |
| 215 | |
| 216 | static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) |
| 217 | { |
| 218 | int i, count, err; |
| 219 | struct pollfd *pfds; |
| 220 | |
| 221 | count = snd_pcm_poll_descriptors_count (handle); |
| 222 | if (count <= 0) { |
| 223 | warn_report("alsa: Could not initialize poll mode: " |
| 224 | "Invalid number of poll descriptors %d", count); |
| 225 | return -1; |
| 226 | } |
| 227 | |
| 228 | pfds = g_new0(struct pollfd, count); |
| 229 | |
| 230 | err = snd_pcm_poll_descriptors (handle, pfds, count); |
| 231 | if (err < 0) { |
| 232 | alsa_logerr(err, "Could not initialize poll mode: Could not obtain poll descriptors"); |
| 233 | g_free (pfds); |
| 234 | return -1; |
| 235 | } |
| 236 | |
| 237 | for (i = 0; i < count; ++i) { |
| 238 | if (pfds[i].events & POLLIN) { |
| 239 | qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp); |
| 240 | } |
| 241 | if (pfds[i].events & POLLOUT) { |
| 242 | trace_alsa_pollout(i, pfds[i].fd); |
| 243 | qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp); |
| 244 | } |
| 245 | trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err); |
| 246 | |
| 247 | } |
| 248 | hlp->pfds = pfds; |
| 249 | hlp->count = count; |
| 250 | hlp->handle = handle; |
| 251 | hlp->mask = mask; |
| 252 | return 0; |
| 253 | } |
| 254 | |
| 255 | static int alsa_poll_out (HWVoiceOut *hw) |
| 256 | { |
| 257 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 258 | |
| 259 | return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT); |
| 260 | } |
| 261 | |
| 262 | static int alsa_poll_in (HWVoiceIn *hw) |
| 263 | { |
| 264 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 265 | |
| 266 | return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN); |
| 267 | } |
| 268 | |
| 269 | static snd_pcm_format_t aud_to_alsafmt(AudioFormat fmt, bool big_endian) |
| 270 | { |
| 271 | switch (fmt) { |
| 272 | case AUDIO_FORMAT_S8: |
| 273 | return SND_PCM_FORMAT_S8; |
| 274 | |
| 275 | case AUDIO_FORMAT_U8: |
| 276 | return SND_PCM_FORMAT_U8; |
| 277 | |
| 278 | case AUDIO_FORMAT_S16: |
| 279 | return big_endian ? SND_PCM_FORMAT_S16_BE : SND_PCM_FORMAT_S16_LE; |
| 280 | |
| 281 | case AUDIO_FORMAT_U16: |
| 282 | return big_endian ? SND_PCM_FORMAT_U16_BE : SND_PCM_FORMAT_U16_LE; |
| 283 | |
| 284 | case AUDIO_FORMAT_S32: |
| 285 | return big_endian ? SND_PCM_FORMAT_S32_BE : SND_PCM_FORMAT_S32_LE; |
| 286 | |
| 287 | case AUDIO_FORMAT_U32: |
| 288 | return big_endian ? SND_PCM_FORMAT_U32_BE : SND_PCM_FORMAT_U32_LE; |
| 289 | |
| 290 | case AUDIO_FORMAT_F32: |
| 291 | return big_endian ? SND_PCM_FORMAT_FLOAT_BE : SND_PCM_FORMAT_FLOAT_LE; |
| 292 | |
| 293 | default: |
| 294 | warn_report("alsa: Internal logic error: Bad audio format %d", fmt); |
| 295 | return SND_PCM_FORMAT_U8; |
| 296 | } |
| 297 | } |
| 298 | |
| 299 | static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt, |
| 300 | int *endianness) |
| 301 | { |
| 302 | switch (alsafmt) { |
| 303 | case SND_PCM_FORMAT_S8: |
| 304 | *endianness = 0; |
| 305 | *fmt = AUDIO_FORMAT_S8; |
| 306 | break; |
| 307 | |
| 308 | case SND_PCM_FORMAT_U8: |
| 309 | *endianness = 0; |
| 310 | *fmt = AUDIO_FORMAT_U8; |
| 311 | break; |
| 312 | |
| 313 | case SND_PCM_FORMAT_S16_LE: |
| 314 | *endianness = 0; |
| 315 | *fmt = AUDIO_FORMAT_S16; |
| 316 | break; |
| 317 | |
| 318 | case SND_PCM_FORMAT_U16_LE: |
| 319 | *endianness = 0; |
| 320 | *fmt = AUDIO_FORMAT_U16; |
| 321 | break; |
| 322 | |
| 323 | case SND_PCM_FORMAT_S16_BE: |
| 324 | *endianness = 1; |
| 325 | *fmt = AUDIO_FORMAT_S16; |
| 326 | break; |
| 327 | |
| 328 | case SND_PCM_FORMAT_U16_BE: |
| 329 | *endianness = 1; |
| 330 | *fmt = AUDIO_FORMAT_U16; |
| 331 | break; |
| 332 | |
| 333 | case SND_PCM_FORMAT_S32_LE: |
| 334 | *endianness = 0; |
| 335 | *fmt = AUDIO_FORMAT_S32; |
| 336 | break; |
| 337 | |
| 338 | case SND_PCM_FORMAT_U32_LE: |
| 339 | *endianness = 0; |
| 340 | *fmt = AUDIO_FORMAT_U32; |
| 341 | break; |
| 342 | |
| 343 | case SND_PCM_FORMAT_S32_BE: |
| 344 | *endianness = 1; |
| 345 | *fmt = AUDIO_FORMAT_S32; |
| 346 | break; |
| 347 | |
| 348 | case SND_PCM_FORMAT_U32_BE: |
| 349 | *endianness = 1; |
| 350 | *fmt = AUDIO_FORMAT_U32; |
| 351 | break; |
| 352 | |
| 353 | case SND_PCM_FORMAT_FLOAT_LE: |
| 354 | *endianness = 0; |
| 355 | *fmt = AUDIO_FORMAT_F32; |
| 356 | break; |
| 357 | |
| 358 | case SND_PCM_FORMAT_FLOAT_BE: |
| 359 | *endianness = 1; |
| 360 | *fmt = AUDIO_FORMAT_F32; |
| 361 | break; |
| 362 | |
| 363 | default: |
| 364 | warn_report("alsa: Unrecognized audio format %d", alsafmt); |
| 365 | return -1; |
| 366 | } |
| 367 | |
| 368 | return 0; |
| 369 | } |
| 370 | |
| 371 | static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) |
| 372 | { |
| 373 | int err; |
| 374 | snd_pcm_sw_params_t *sw_params; |
| 375 | |
| 376 | snd_pcm_sw_params_alloca (&sw_params); |
| 377 | |
| 378 | err = snd_pcm_sw_params_current (handle, sw_params); |
| 379 | if (err < 0) { |
| 380 | error_report("alsa: Could not fully initialize DAC"); |
| 381 | alsa_logerr(err, "Failed to get current software parameters"); |
| 382 | return; |
| 383 | } |
| 384 | |
| 385 | err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold); |
| 386 | if (err < 0) { |
| 387 | error_report("alsa: Could not fully initialize DAC"); |
| 388 | alsa_logerr(err, "Failed to set software threshold to %ld", threshold); |
| 389 | return; |
| 390 | } |
| 391 | |
| 392 | err = snd_pcm_sw_params (handle, sw_params); |
| 393 | if (err < 0) { |
| 394 | error_report("alsa: Could not fully initialize DAC"); |
| 395 | alsa_logerr(err, "Failed to set software parameters"); |
| 396 | return; |
| 397 | } |
| 398 | } |
| 399 | |
| 400 | static int alsa_open(bool in, struct alsa_params_req *req, |
| 401 | struct alsa_params_obt *obt, snd_pcm_t **handlep, |
| 402 | Audiodev *dev) |
| 403 | { |
| 404 | AudiodevAlsaOptions *aopts = &dev->u.alsa; |
| 405 | AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out; |
| 406 | snd_pcm_t *handle; |
| 407 | snd_pcm_hw_params_t *hw_params; |
| 408 | int err; |
| 409 | unsigned int freq, nchannels; |
| 410 | const char *pcm_name = apdo->dev ?: "default"; |
| 411 | snd_pcm_uframes_t obt_buffer_size; |
| 412 | const char *typ = in ? "ADC" : "DAC"; |
| 413 | snd_pcm_format_t obtfmt; |
| 414 | |
| 415 | freq = req->freq; |
| 416 | nchannels = req->nchannels; |
| 417 | |
| 418 | snd_pcm_hw_params_alloca (&hw_params); |
| 419 | |
| 420 | err = snd_pcm_open ( |
| 421 | &handle, |
| 422 | pcm_name, |
| 423 | in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK, |
| 424 | SND_PCM_NONBLOCK |
| 425 | ); |
| 426 | if (err < 0) { |
| 427 | alsa_logerr2(err, typ, "Failed to open `%s'", pcm_name); |
| 428 | return -1; |
| 429 | } |
| 430 | |
| 431 | err = snd_pcm_hw_params_any (handle, hw_params); |
| 432 | if (err < 0) { |
| 433 | alsa_logerr2(err, typ, "Failed to initialize hardware parameters"); |
| 434 | goto err; |
| 435 | } |
| 436 | |
| 437 | err = snd_pcm_hw_params_set_access ( |
| 438 | handle, |
| 439 | hw_params, |
| 440 | SND_PCM_ACCESS_RW_INTERLEAVED |
| 441 | ); |
| 442 | if (err < 0) { |
| 443 | alsa_logerr2(err, typ, "Failed to set access type"); |
| 444 | goto err; |
| 445 | } |
| 446 | |
| 447 | err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); |
| 448 | if (err < 0) { |
| 449 | alsa_logerr2(err, typ, "Failed to set format %d", req->fmt); |
| 450 | } |
| 451 | |
| 452 | err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0); |
| 453 | if (err < 0) { |
| 454 | alsa_logerr2(err, typ, "Failed to set frequency %d", req->freq); |
| 455 | goto err; |
| 456 | } |
| 457 | |
| 458 | err = snd_pcm_hw_params_set_channels_near ( |
| 459 | handle, |
| 460 | hw_params, |
| 461 | &nchannels |
| 462 | ); |
| 463 | if (err < 0) { |
| 464 | alsa_logerr2(err, typ, "Failed to set number of channels %d", req->nchannels); |
| 465 | goto err; |
| 466 | } |
| 467 | |
| 468 | if (apdo->buffer_length) { |
| 469 | int dir = 0; |
| 470 | unsigned int btime = apdo->buffer_length; |
| 471 | |
| 472 | err = snd_pcm_hw_params_set_buffer_time_near( |
| 473 | handle, hw_params, &btime, &dir); |
| 474 | |
| 475 | if (err < 0) { |
| 476 | alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32, |
| 477 | apdo->buffer_length); |
| 478 | goto err; |
| 479 | } |
| 480 | |
| 481 | if (apdo->has_buffer_length && btime != apdo->buffer_length) { |
| 482 | warn_report("alsa: Requested buffer time %" PRId32 " was rejected, using %u", |
| 483 | apdo->buffer_length, btime); |
| 484 | } |
| 485 | } |
| 486 | |
| 487 | if (apdo->period_length) { |
| 488 | int dir = 0; |
| 489 | unsigned int ptime = apdo->period_length; |
| 490 | |
| 491 | err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime, |
| 492 | &dir); |
| 493 | |
| 494 | if (err < 0) { |
| 495 | alsa_logerr2(err, typ, "Failed to set period time to %" PRId32, |
| 496 | apdo->period_length); |
| 497 | goto err; |
| 498 | } |
| 499 | |
| 500 | if (apdo->has_period_length && ptime != apdo->period_length) { |
| 501 | warn_report("alsa: Requested period time %" PRId32 " was rejected, using %d", |
| 502 | apdo->period_length, ptime); |
| 503 | } |
| 504 | } |
| 505 | |
| 506 | err = snd_pcm_hw_params (handle, hw_params); |
| 507 | if (err < 0) { |
| 508 | alsa_logerr2(err, typ, "Failed to apply audio parameters"); |
| 509 | goto err; |
| 510 | } |
| 511 | |
| 512 | err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size); |
| 513 | if (err < 0) { |
| 514 | alsa_logerr2(err, typ, "Failed to get buffer size"); |
| 515 | goto err; |
| 516 | } |
| 517 | |
| 518 | err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); |
| 519 | if (err < 0) { |
| 520 | alsa_logerr2(err, typ, "Failed to get format"); |
| 521 | goto err; |
| 522 | } |
| 523 | |
| 524 | if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { |
| 525 | error_report("alsa: Invalid format was returned %d", obtfmt); |
| 526 | goto err; |
| 527 | } |
| 528 | |
| 529 | err = snd_pcm_prepare (handle); |
| 530 | if (err < 0) { |
| 531 | alsa_logerr2(err, typ, "Could not prepare handle %p", handle); |
| 532 | goto err; |
| 533 | } |
| 534 | |
| 535 | if (!in && aopts->has_threshold && aopts->threshold) { |
| 536 | struct audsettings as = { .freq = freq }; |
| 537 | alsa_set_threshold( |
| 538 | handle, |
| 539 | audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo), |
| 540 | &as, aopts->threshold)); |
| 541 | } |
| 542 | |
| 543 | obt->nchannels = nchannels; |
| 544 | obt->freq = freq; |
| 545 | obt->samples = obt_buffer_size; |
| 546 | |
| 547 | *handlep = handle; |
| 548 | |
| 549 | trace_alsa_info_params(req->fmt, obtfmt, req->nchannels, obt->nchannels, |
| 550 | req->freq, obt->freq); |
| 551 | trace_alsa_info_samples(apdo->buffer_length, apdo->period_length, obt->samples); |
| 552 | |
| 553 | return 0; |
| 554 | |
| 555 | err: |
| 556 | alsa_anal_close1 (&handle); |
| 557 | return -1; |
| 558 | } |
| 559 | |
| 560 | static size_t alsa_buffer_get_free(HWVoiceOut *hw) |
| 561 | { |
| 562 | ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw; |
| 563 | snd_pcm_sframes_t avail; |
| 564 | size_t alsa_free, generic_free, generic_in_use; |
| 565 | |
| 566 | avail = snd_pcm_avail_update(alsa->handle); |
| 567 | if (avail < 0) { |
| 568 | if (avail == -EPIPE) { |
| 569 | if (!alsa_recover(alsa->handle)) { |
| 570 | avail = snd_pcm_avail_update(alsa->handle); |
| 571 | } |
| 572 | } |
| 573 | if (avail < 0) { |
| 574 | alsa_logerr(avail, "Could not obtain number of available frames"); |
| 575 | avail = 0; |
| 576 | } |
| 577 | } |
| 578 | |
| 579 | alsa_free = avail * hw->info.bytes_per_frame; |
| 580 | generic_free = audio_generic_buffer_get_free(hw); |
| 581 | generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free; |
| 582 | if (generic_in_use) { |
| 583 | /* |
| 584 | * This code can only be reached in the unlikely case that |
| 585 | * snd_pcm_avail_update() returned a larger number of frames |
| 586 | * than snd_pcm_writei() could write. Make sure that all |
| 587 | * remaining bytes in the generic buffer can be written. |
| 588 | */ |
| 589 | alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0; |
| 590 | } |
| 591 | |
| 592 | return alsa_free; |
| 593 | } |
| 594 | |
| 595 | static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) |
| 596 | { |
| 597 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 598 | size_t pos = 0; |
| 599 | size_t len_frames = len / hw->info.bytes_per_frame; |
| 600 | |
| 601 | while (len_frames) { |
| 602 | char *src = advance(buf, pos); |
| 603 | snd_pcm_sframes_t written; |
| 604 | |
| 605 | written = snd_pcm_writei(alsa->handle, src, len_frames); |
| 606 | |
| 607 | if (written <= 0) { |
| 608 | switch (written) { |
| 609 | case 0: |
| 610 | trace_alsa_wrote_zero(len_frames); |
| 611 | return pos; |
| 612 | |
| 613 | case -EPIPE: |
| 614 | if (alsa_recover(alsa->handle)) { |
| 615 | alsa_logerr(written, "Failed to write %zu frames", len_frames); |
| 616 | return pos; |
| 617 | } |
| 618 | trace_alsa_xrun_out(); |
| 619 | continue; |
| 620 | |
| 621 | case -ESTRPIPE: |
| 622 | /* |
| 623 | * stream is suspended and waiting for an application |
| 624 | * recovery |
| 625 | */ |
| 626 | if (alsa_resume(alsa->handle)) { |
| 627 | alsa_logerr(written, "Failed to write %zu frames", len_frames); |
| 628 | return pos; |
| 629 | } |
| 630 | trace_alsa_resume_out(); |
| 631 | continue; |
| 632 | |
| 633 | case -EAGAIN: |
| 634 | return pos; |
| 635 | |
| 636 | default: |
| 637 | alsa_logerr(written, "Failed to write %zu frames from %p", len_frames, src); |
| 638 | return pos; |
| 639 | } |
| 640 | } |
| 641 | |
| 642 | pos += written * hw->info.bytes_per_frame; |
| 643 | if (written < len_frames) { |
| 644 | break; |
| 645 | } |
| 646 | len_frames -= written; |
| 647 | } |
| 648 | |
| 649 | return pos; |
| 650 | } |
| 651 | |
| 652 | static void alsa_fini_out (HWVoiceOut *hw) |
| 653 | { |
| 654 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 655 | |
| 656 | trace_alsa_fini_out(); |
| 657 | alsa_anal_close (&alsa->handle, &alsa->pollhlp); |
| 658 | } |
| 659 | |
| 660 | static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as) |
| 661 | { |
| 662 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 663 | struct alsa_params_req req; |
| 664 | struct alsa_params_obt obt; |
| 665 | snd_pcm_t *handle; |
| 666 | struct audsettings obt_as; |
| 667 | Audiodev *dev = hw->s->dev; |
| 668 | |
| 669 | req.fmt = aud_to_alsafmt (as->fmt, as->big_endian); |
| 670 | req.freq = as->freq; |
| 671 | req.nchannels = as->nchannels; |
| 672 | |
| 673 | if (alsa_open(0, &req, &obt, &handle, dev)) { |
| 674 | return -1; |
| 675 | } |
| 676 | |
| 677 | obt_as.freq = obt.freq; |
| 678 | obt_as.nchannels = obt.nchannels; |
| 679 | obt_as.fmt = obt.fmt; |
| 680 | obt_as.big_endian = obt.endianness; |
| 681 | |
| 682 | audio_pcm_init_info (&hw->info, &obt_as); |
| 683 | hw->samples = obt.samples; |
| 684 | |
| 685 | alsa->pollhlp.s = hw->s; |
| 686 | alsa->handle = handle; |
| 687 | return 0; |
| 688 | } |
| 689 | |
| 690 | #define VOICE_CTL_PAUSE 0 |
| 691 | #define VOICE_CTL_PREPARE 1 |
| 692 | #define VOICE_CTL_START 2 |
| 693 | |
| 694 | static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl) |
| 695 | { |
| 696 | int err; |
| 697 | |
| 698 | if (ctl == VOICE_CTL_PAUSE) { |
| 699 | err = snd_pcm_drop (handle); |
| 700 | if (err < 0) { |
| 701 | alsa_logerr(err, "Could not stop %s", typ); |
| 702 | return -1; |
| 703 | } |
| 704 | } else { |
| 705 | err = snd_pcm_prepare (handle); |
| 706 | if (err < 0) { |
| 707 | alsa_logerr(err, "Could not prepare handle for %s", typ); |
| 708 | return -1; |
| 709 | } |
| 710 | if (ctl == VOICE_CTL_START) { |
| 711 | err = snd_pcm_start(handle); |
| 712 | if (err < 0) { |
| 713 | alsa_logerr(err, "Could not start handle for %s", typ); |
| 714 | return -1; |
| 715 | } |
| 716 | } |
| 717 | } |
| 718 | |
| 719 | return 0; |
| 720 | } |
| 721 | |
| 722 | static void alsa_enable_out(HWVoiceOut *hw, bool enable) |
| 723 | { |
| 724 | ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw; |
| 725 | AudiodevAlsaPerDirectionOptions *apdo = hw->s->dev->u.alsa.out; |
| 726 | |
| 727 | trace_alsa_enable_out(enable); |
| 728 | |
| 729 | if (enable) { |
| 730 | bool poll_mode = apdo->try_poll; |
| 731 | |
| 732 | if (poll_mode && alsa_poll_out(hw)) { |
| 733 | poll_mode = 0; |
| 734 | } |
| 735 | hw->poll_mode = poll_mode; |
| 736 | alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE); |
| 737 | } else { |
| 738 | if (hw->poll_mode) { |
| 739 | hw->poll_mode = 0; |
| 740 | alsa_fini_poll(&alsa->pollhlp); |
| 741 | } |
| 742 | alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE); |
| 743 | } |
| 744 | } |
| 745 | |
| 746 | static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as) |
| 747 | { |
| 748 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 749 | struct alsa_params_req req; |
| 750 | struct alsa_params_obt obt; |
| 751 | snd_pcm_t *handle; |
| 752 | struct audsettings obt_as; |
| 753 | Audiodev *dev = hw->s->dev; |
| 754 | |
| 755 | req.fmt = aud_to_alsafmt (as->fmt, as->big_endian); |
| 756 | req.freq = as->freq; |
| 757 | req.nchannels = as->nchannels; |
| 758 | |
| 759 | if (alsa_open(1, &req, &obt, &handle, dev)) { |
| 760 | return -1; |
| 761 | } |
| 762 | |
| 763 | obt_as.freq = obt.freq; |
| 764 | obt_as.nchannels = obt.nchannels; |
| 765 | obt_as.fmt = obt.fmt; |
| 766 | obt_as.big_endian = obt.endianness; |
| 767 | |
| 768 | audio_pcm_init_info (&hw->info, &obt_as); |
| 769 | hw->samples = obt.samples; |
| 770 | |
| 771 | alsa->pollhlp.s = hw->s; |
| 772 | alsa->handle = handle; |
| 773 | return 0; |
| 774 | } |
| 775 | |
| 776 | static void alsa_fini_in (HWVoiceIn *hw) |
| 777 | { |
| 778 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 779 | |
| 780 | alsa_anal_close (&alsa->handle, &alsa->pollhlp); |
| 781 | } |
| 782 | |
| 783 | static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) |
| 784 | { |
| 785 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 786 | size_t pos = 0; |
| 787 | |
| 788 | while (len) { |
| 789 | void *dst = advance(buf, pos); |
| 790 | snd_pcm_sframes_t nread; |
| 791 | |
| 792 | nread = snd_pcm_readi( |
| 793 | alsa->handle, dst, len / hw->info.bytes_per_frame); |
| 794 | |
| 795 | if (nread <= 0) { |
| 796 | switch (nread) { |
| 797 | case 0: |
| 798 | trace_alsa_read_zero(len); |
| 799 | return pos; |
| 800 | |
| 801 | case -EPIPE: |
| 802 | if (alsa_recover(alsa->handle)) { |
| 803 | alsa_logerr(nread, "Failed to read %zu frames", len); |
| 804 | return pos; |
| 805 | } |
| 806 | trace_alsa_xrun_in(); |
| 807 | continue; |
| 808 | |
| 809 | case -EAGAIN: |
| 810 | return pos; |
| 811 | |
| 812 | default: |
| 813 | alsa_logerr(nread, "Failed to read %zu frames to %p", len, dst); |
| 814 | return pos; |
| 815 | } |
| 816 | } |
| 817 | |
| 818 | pos += nread * hw->info.bytes_per_frame; |
| 819 | len -= nread * hw->info.bytes_per_frame; |
| 820 | } |
| 821 | |
| 822 | return pos; |
| 823 | } |
| 824 | |
| 825 | static void alsa_enable_in(HWVoiceIn *hw, bool enable) |
| 826 | { |
| 827 | ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw; |
| 828 | AudiodevAlsaPerDirectionOptions *apdo = hw->s->dev->u.alsa.in; |
| 829 | |
| 830 | trace_alsa_enable_in(enable); |
| 831 | |
| 832 | if (enable) { |
| 833 | bool poll_mode = apdo->try_poll; |
| 834 | |
| 835 | if (poll_mode && alsa_poll_in(hw)) { |
| 836 | poll_mode = 0; |
| 837 | } |
| 838 | hw->poll_mode = poll_mode; |
| 839 | |
| 840 | alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START); |
| 841 | } else { |
| 842 | if (hw->poll_mode) { |
| 843 | hw->poll_mode = 0; |
| 844 | alsa_fini_poll(&alsa->pollhlp); |
| 845 | } |
| 846 | alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE); |
| 847 | } |
| 848 | } |
| 849 | |
| 850 | static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo) |
| 851 | { |
| 852 | if (!apdo->has_try_poll) { |
| 853 | apdo->try_poll = false; |
| 854 | apdo->has_try_poll = true; |
| 855 | } |
| 856 | } |
| 857 | |
| 858 | static bool |
| 859 | audio_alsa_realize(AudioBackend *abe, Audiodev *dev, Error **errp) |
| 860 | { |
| 861 | AudiodevAlsaOptions *aopts; |
| 862 | assert(dev->driver == AUDIODEV_DRIVER_ALSA); |
| 863 | |
| 864 | aopts = &dev->u.alsa; |
| 865 | alsa_init_per_direction(aopts->in); |
| 866 | alsa_init_per_direction(aopts->out); |
| 867 | |
| 868 | /* don't set has_* so alsa_open can identify it wasn't set by the user */ |
| 869 | if (!dev->u.alsa.out->has_period_length) { |
| 870 | /* 256 frames assuming 44100Hz */ |
| 871 | dev->u.alsa.out->period_length = 5805; |
| 872 | } |
| 873 | if (!dev->u.alsa.out->has_buffer_length) { |
| 874 | /* 4096 frames assuming 44100Hz */ |
| 875 | dev->u.alsa.out->buffer_length = 92880; |
| 876 | } |
| 877 | |
| 878 | if (!dev->u.alsa.in->has_period_length) { |
| 879 | /* 256 frames assuming 44100Hz */ |
| 880 | dev->u.alsa.in->period_length = 5805; |
| 881 | } |
| 882 | if (!dev->u.alsa.in->has_buffer_length) { |
| 883 | /* 4096 frames assuming 44100Hz */ |
| 884 | dev->u.alsa.in->buffer_length = 92880; |
| 885 | } |
| 886 | |
| 887 | return audio_alsa_parent_class->realize(abe, dev, errp); |
| 888 | } |
| 889 | |
| 890 | static void audio_alsa_class_init(ObjectClass *klass, const void *data) |
| 891 | { |
| 892 | AudioBackendClass *b = AUDIO_BACKEND_CLASS(klass); |
| 893 | AudioMixengBackendClass *k = AUDIO_MIXENG_BACKEND_CLASS(klass); |
| 894 | |
| 895 | audio_alsa_parent_class = AUDIO_BACKEND_CLASS(object_class_get_parent(klass)); |
| 896 | |
| 897 | b->realize = audio_alsa_realize; |
| 898 | k->max_voices_out = INT_MAX; |
| 899 | k->max_voices_in = INT_MAX; |
| 900 | k->voice_size_out = sizeof(ALSAVoiceOut); |
| 901 | k->voice_size_in = sizeof(ALSAVoiceIn); |
| 902 | |
| 903 | k->init_out = alsa_init_out; |
| 904 | k->fini_out = alsa_fini_out; |
| 905 | k->write = alsa_write; |
| 906 | k->buffer_get_free = alsa_buffer_get_free; |
| 907 | k->run_buffer_out = audio_generic_run_buffer_out; |
| 908 | k->enable_out = alsa_enable_out; |
| 909 | |
| 910 | k->init_in = alsa_init_in; |
| 911 | k->fini_in = alsa_fini_in; |
| 912 | k->read = alsa_read; |
| 913 | k->run_buffer_in = audio_generic_run_buffer_in; |
| 914 | k->enable_in = alsa_enable_in; |
| 915 | } |
| 916 | |
| 917 | static const TypeInfo audio_types[] = { |
| 918 | { |
| 919 | .name = TYPE_AUDIO_ALSA, |
| 920 | .parent = TYPE_AUDIO_MIXENG_BACKEND, |
| 921 | .instance_size = sizeof(AudioALSA), |
| 922 | .class_init = audio_alsa_class_init, |
| 923 | }, |
| 924 | }; |
| 925 | |
| 926 | DEFINE_TYPES(audio_types) |
| 927 | module_obj(TYPE_AUDIO_ALSA); |