audio_hw.c revision 4b89e37ad290ef955abf8ac1d151728303311345
1/* 2 * Copyright (C) 2013-2014 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17#define LOG_TAG "audio_hw_primary" 18/*#define LOG_NDEBUG 0*/ 19/*#define VERY_VERY_VERBOSE_LOGGING*/ 20#ifdef VERY_VERY_VERBOSE_LOGGING 21#define ALOGVV ALOGV 22#else 23#define ALOGVV(a...) do { } while(0) 24#endif 25 26#include <errno.h> 27#include <pthread.h> 28#include <stdint.h> 29#include <sys/time.h> 30#include <stdlib.h> 31#include <math.h> 32#include <dlfcn.h> 33#include <sys/resource.h> 34#include <sys/prctl.h> 35 36#include <cutils/log.h> 37#include <cutils/str_parms.h> 38#include <cutils/properties.h> 39#include <cutils/atomic.h> 40#include <cutils/sched_policy.h> 41 42#include <hardware/audio_effect.h> 43#include <hardware/audio_alsaops.h> 44#include <system/thread_defs.h> 45#include <audio_effects/effect_aec.h> 46#include <audio_effects/effect_ns.h> 47#include "audio_hw.h" 48#include "audio_extn.h" 49#include "platform_api.h" 50#include <platform.h> 51#include "voice_extn.h" 52 53#include "sound/compress_params.h" 54 55#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024) 56#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4 57/* ToDo: Check and update a proper value in msec */ 58#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96 59#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000 60 61static unsigned int configured_low_latency_capture_period_size = 62 LOW_LATENCY_CAPTURE_PERIOD_SIZE; 63 64/* This constant enables extended precision handling. 65 * TODO The flag is off until more testing is done. 66 */ 67static const bool k_enable_extended_precision = false; 68 69struct pcm_config pcm_config_deep_buffer = { 70 .channels = 2, 71 .rate = DEFAULT_OUTPUT_SAMPLING_RATE, 72 .period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE, 73 .period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT, 74 .format = PCM_FORMAT_S16_LE, 75 .start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4, 76 .stop_threshold = INT_MAX, 77 .avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4, 78}; 79 80struct pcm_config pcm_config_low_latency = { 81 .channels = 2, 82 .rate = DEFAULT_OUTPUT_SAMPLING_RATE, 83 .period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE, 84 .period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT, 85 .format = PCM_FORMAT_S16_LE, 86 .start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4, 87 .stop_threshold = INT_MAX, 88 .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4, 89}; 90 91struct pcm_config pcm_config_hdmi_multi = { 92 .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */ 93 .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */ 94 .period_size = HDMI_MULTI_PERIOD_SIZE, 95 .period_count = HDMI_MULTI_PERIOD_COUNT, 96 .format = PCM_FORMAT_S16_LE, 97 .start_threshold = 0, 98 .stop_threshold = INT_MAX, 99 .avail_min = 0, 100}; 101 102struct pcm_config pcm_config_audio_capture = { 103 .channels = 2, 104 .period_count = AUDIO_CAPTURE_PERIOD_COUNT, 105 .format = PCM_FORMAT_S16_LE, 106}; 107 108const char * const use_case_table[AUDIO_USECASE_MAX] = { 109 [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback", 110 [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback", 111 [USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback", 112 [USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback", 113 114 [USECASE_AUDIO_RECORD] = "audio-record", 115 [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record", 116 117 [USECASE_AUDIO_HFP_SCO] = "hfp-sco", 118 [USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb", 119 120 [USECASE_VOICE_CALL] = "voice-call", 121 [USECASE_VOICE2_CALL] = "voice2-call", 122 [USECASE_VOLTE_CALL] = "volte-call", 123 [USECASE_QCHAT_CALL] = "qchat-call", 124 [USECASE_VOWLAN_CALL] = "vowlan-call", 125}; 126 127 128#define STRING_TO_ENUM(string) { #string, string } 129 130struct string_to_enum { 131 const char *name; 132 uint32_t value; 133}; 134 135static const struct string_to_enum out_channels_name_to_enum_table[] = { 136 STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), 137 STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), 138 STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), 139}; 140 141static int set_voice_volume_l(struct audio_device *adev, float volume); 142 143static bool is_supported_format(audio_format_t format) 144{ 145 if (format == AUDIO_FORMAT_MP3 || 146 format == AUDIO_FORMAT_AAC) 147 return true; 148 149 return false; 150} 151 152static int get_snd_codec_id(audio_format_t format) 153{ 154 int id = 0; 155 156 switch (format) { 157 case AUDIO_FORMAT_MP3: 158 id = SND_AUDIOCODEC_MP3; 159 break; 160 case AUDIO_FORMAT_AAC: 161 id = SND_AUDIOCODEC_AAC; 162 break; 163 default: 164 ALOGE("%s: Unsupported audio format", __func__); 165 } 166 167 return id; 168} 169 170int pcm_ioctl(void *pcm, int request, ...) 171{ 172 va_list ap; 173 void * arg; 174 int pcm_fd = *(int*)pcm; 175 176 va_start(ap, request); 177 arg = va_arg(ap, void *); 178 va_end(ap); 179 180 return ioctl(pcm_fd, request, arg); 181} 182 183int enable_audio_route(struct audio_device *adev, 184 struct audio_usecase *usecase) 185{ 186 snd_device_t snd_device; 187 char mixer_path[50]; 188 189 if (usecase == NULL) 190 return -EINVAL; 191 192 ALOGV("%s: enter: usecase(%d)", __func__, usecase->id); 193 194 if (usecase->type == PCM_CAPTURE) 195 snd_device = usecase->in_snd_device; 196 else 197 snd_device = usecase->out_snd_device; 198 199 strcpy(mixer_path, use_case_table[usecase->id]); 200 platform_add_backend_name(adev->platform, mixer_path, snd_device); 201 ALOGV("%s: apply and update mixer path: %s", __func__, mixer_path); 202 audio_route_apply_and_update_path(adev->audio_route, mixer_path); 203 204 ALOGV("%s: exit", __func__); 205 return 0; 206} 207 208int disable_audio_route(struct audio_device *adev, 209 struct audio_usecase *usecase) 210{ 211 snd_device_t snd_device; 212 char mixer_path[50]; 213 214 if (usecase == NULL) 215 return -EINVAL; 216 217 ALOGV("%s: enter: usecase(%d)", __func__, usecase->id); 218 if (usecase->type == PCM_CAPTURE) 219 snd_device = usecase->in_snd_device; 220 else 221 snd_device = usecase->out_snd_device; 222 strcpy(mixer_path, use_case_table[usecase->id]); 223 platform_add_backend_name(adev->platform, mixer_path, snd_device); 224 ALOGV("%s: reset and update mixer path: %s", __func__, mixer_path); 225 audio_route_reset_and_update_path(adev->audio_route, mixer_path); 226 227 ALOGV("%s: exit", __func__); 228 return 0; 229} 230 231int enable_snd_device(struct audio_device *adev, 232 snd_device_t snd_device) 233{ 234 if (snd_device < SND_DEVICE_MIN || 235 snd_device >= SND_DEVICE_MAX) { 236 ALOGE("%s: Invalid sound device %d", __func__, snd_device); 237 return -EINVAL; 238 } 239 240 adev->snd_dev_ref_cnt[snd_device]++; 241 if (adev->snd_dev_ref_cnt[snd_device] > 1) { 242 ALOGV("%s: snd_device(%d: %s) is already active", 243 __func__, snd_device, platform_get_snd_device_name(snd_device)); 244 return 0; 245 } 246 247 if (platform_send_audio_calibration(adev->platform, snd_device) < 0) { 248 adev->snd_dev_ref_cnt[snd_device]--; 249 return -EINVAL; 250 } 251 252 const char * dev_path = platform_get_snd_device_name(snd_device); 253 ALOGV("%s: snd_device(%d: %s)", __func__, snd_device, dev_path); 254 audio_route_apply_and_update_path(adev->audio_route, dev_path); 255 256 return 0; 257} 258 259int disable_snd_device(struct audio_device *adev, 260 snd_device_t snd_device) 261{ 262 if (snd_device < SND_DEVICE_MIN || 263 snd_device >= SND_DEVICE_MAX) { 264 ALOGE("%s: Invalid sound device %d", __func__, snd_device); 265 return -EINVAL; 266 } 267 if (adev->snd_dev_ref_cnt[snd_device] <= 0) { 268 ALOGE("%s: device ref cnt is already 0", __func__); 269 return -EINVAL; 270 } 271 adev->snd_dev_ref_cnt[snd_device]--; 272 if (adev->snd_dev_ref_cnt[snd_device] == 0) { 273 const char * dev_path = platform_get_snd_device_name(snd_device); 274 ALOGV("%s: snd_device(%d: %s)", __func__, 275 snd_device, dev_path); 276 audio_route_reset_and_update_path(adev->audio_route, dev_path); 277 } 278 return 0; 279} 280 281static void check_usecases_codec_backend(struct audio_device *adev, 282 struct audio_usecase *uc_info, 283 snd_device_t snd_device) 284{ 285 struct listnode *node; 286 struct audio_usecase *usecase; 287 bool switch_device[AUDIO_USECASE_MAX]; 288 int i, num_uc_to_switch = 0; 289 290 /* 291 * This function is to make sure that all the usecases that are active on 292 * the hardware codec backend are always routed to any one device that is 293 * handled by the hardware codec. 294 * For example, if low-latency and deep-buffer usecases are currently active 295 * on speaker and out_set_parameters(headset) is received on low-latency 296 * output, then we have to make sure deep-buffer is also switched to headset, 297 * because of the limitation that both the devices cannot be enabled 298 * at the same time as they share the same backend. 299 */ 300 /* Disable all the usecases on the shared backend other than the 301 specified usecase */ 302 for (i = 0; i < AUDIO_USECASE_MAX; i++) 303 switch_device[i] = false; 304 305 list_for_each(node, &adev->usecase_list) { 306 usecase = node_to_item(node, struct audio_usecase, list); 307 if (usecase->type != PCM_CAPTURE && 308 usecase != uc_info && 309 usecase->out_snd_device != snd_device && 310 usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) { 311 ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", 312 __func__, use_case_table[usecase->id], 313 platform_get_snd_device_name(usecase->out_snd_device)); 314 disable_audio_route(adev, usecase); 315 switch_device[usecase->id] = true; 316 num_uc_to_switch++; 317 } 318 } 319 320 if (num_uc_to_switch) { 321 list_for_each(node, &adev->usecase_list) { 322 usecase = node_to_item(node, struct audio_usecase, list); 323 if (switch_device[usecase->id]) { 324 disable_snd_device(adev, usecase->out_snd_device); 325 } 326 } 327 328 list_for_each(node, &adev->usecase_list) { 329 usecase = node_to_item(node, struct audio_usecase, list); 330 if (switch_device[usecase->id]) { 331 enable_snd_device(adev, snd_device); 332 } 333 } 334 335 /* Re-route all the usecases on the shared backend other than the 336 specified usecase to new snd devices */ 337 list_for_each(node, &adev->usecase_list) { 338 usecase = node_to_item(node, struct audio_usecase, list); 339 /* Update the out_snd_device only before enabling the audio route */ 340 if (switch_device[usecase->id] ) { 341 usecase->out_snd_device = snd_device; 342 enable_audio_route(adev, usecase); 343 } 344 } 345 } 346} 347 348static void check_and_route_capture_usecases(struct audio_device *adev, 349 struct audio_usecase *uc_info, 350 snd_device_t snd_device) 351{ 352 struct listnode *node; 353 struct audio_usecase *usecase; 354 bool switch_device[AUDIO_USECASE_MAX]; 355 int i, num_uc_to_switch = 0; 356 357 /* 358 * This function is to make sure that all the active capture usecases 359 * are always routed to the same input sound device. 360 * For example, if audio-record and voice-call usecases are currently 361 * active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece) 362 * is received for voice call then we have to make sure that audio-record 363 * usecase is also switched to earpiece i.e. voice-dmic-ef, 364 * because of the limitation that two devices cannot be enabled 365 * at the same time if they share the same backend. 366 */ 367 for (i = 0; i < AUDIO_USECASE_MAX; i++) 368 switch_device[i] = false; 369 370 list_for_each(node, &adev->usecase_list) { 371 usecase = node_to_item(node, struct audio_usecase, list); 372 if (usecase->type != PCM_PLAYBACK && 373 usecase != uc_info && 374 usecase->in_snd_device != snd_device) { 375 ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", 376 __func__, use_case_table[usecase->id], 377 platform_get_snd_device_name(usecase->in_snd_device)); 378 disable_audio_route(adev, usecase); 379 switch_device[usecase->id] = true; 380 num_uc_to_switch++; 381 } 382 } 383 384 if (num_uc_to_switch) { 385 list_for_each(node, &adev->usecase_list) { 386 usecase = node_to_item(node, struct audio_usecase, list); 387 if (switch_device[usecase->id]) { 388 disable_snd_device(adev, usecase->in_snd_device); 389 } 390 } 391 392 list_for_each(node, &adev->usecase_list) { 393 usecase = node_to_item(node, struct audio_usecase, list); 394 if (switch_device[usecase->id]) { 395 enable_snd_device(adev, snd_device); 396 } 397 } 398 399 /* Re-route all the usecases on the shared backend other than the 400 specified usecase to new snd devices */ 401 list_for_each(node, &adev->usecase_list) { 402 usecase = node_to_item(node, struct audio_usecase, list); 403 /* Update the in_snd_device only before enabling the audio route */ 404 if (switch_device[usecase->id] ) { 405 usecase->in_snd_device = snd_device; 406 enable_audio_route(adev, usecase); 407 } 408 } 409 } 410} 411 412/* must be called with hw device mutex locked */ 413static int read_hdmi_channel_masks(struct stream_out *out) 414{ 415 int ret = 0; 416 int channels = platform_edid_get_max_channels(out->dev->platform); 417 418 switch (channels) { 419 /* 420 * Do not handle stereo output in Multi-channel cases 421 * Stereo case is handled in normal playback path 422 */ 423 case 6: 424 ALOGV("%s: HDMI supports 5.1", __func__); 425 out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1; 426 break; 427 case 8: 428 ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__); 429 out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1; 430 out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1; 431 break; 432 default: 433 ALOGE("HDMI does not support multi channel playback"); 434 ret = -ENOSYS; 435 break; 436 } 437 return ret; 438} 439 440struct audio_usecase *get_usecase_from_list(struct audio_device *adev, 441 audio_usecase_t uc_id) 442{ 443 struct audio_usecase *usecase; 444 struct listnode *node; 445 446 list_for_each(node, &adev->usecase_list) { 447 usecase = node_to_item(node, struct audio_usecase, list); 448 if (usecase->id == uc_id) 449 return usecase; 450 } 451 return NULL; 452} 453 454int select_devices(struct audio_device *adev, 455 audio_usecase_t uc_id) 456{ 457 snd_device_t out_snd_device = SND_DEVICE_NONE; 458 snd_device_t in_snd_device = SND_DEVICE_NONE; 459 struct audio_usecase *usecase = NULL; 460 struct audio_usecase *vc_usecase = NULL; 461 struct audio_usecase *hfp_usecase = NULL; 462 audio_usecase_t hfp_ucid; 463 struct listnode *node; 464 int status = 0; 465 466 usecase = get_usecase_from_list(adev, uc_id); 467 if (usecase == NULL) { 468 ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id); 469 return -EINVAL; 470 } 471 472 if ((usecase->type == VOICE_CALL) || 473 (usecase->type == PCM_HFP_CALL)) { 474 out_snd_device = platform_get_output_snd_device(adev->platform, 475 usecase->stream.out->devices); 476 in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices); 477 usecase->devices = usecase->stream.out->devices; 478 } else { 479 /* 480 * If the voice call is active, use the sound devices of voice call usecase 481 * so that it would not result any device switch. All the usecases will 482 * be switched to new device when select_devices() is called for voice call 483 * usecase. This is to avoid switching devices for voice call when 484 * check_usecases_codec_backend() is called below. 485 */ 486 if (voice_is_in_call(adev)) { 487 vc_usecase = get_usecase_from_list(adev, USECASE_VOICE_CALL); 488 if ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) || 489 (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL)) { 490 in_snd_device = vc_usecase->in_snd_device; 491 out_snd_device = vc_usecase->out_snd_device; 492 } 493 } else if (audio_extn_hfp_is_active(adev)) { 494 hfp_ucid = audio_extn_hfp_get_usecase(); 495 hfp_usecase = get_usecase_from_list(adev, hfp_ucid); 496 if (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) { 497 in_snd_device = hfp_usecase->in_snd_device; 498 out_snd_device = hfp_usecase->out_snd_device; 499 } 500 } 501 if (usecase->type == PCM_PLAYBACK) { 502 usecase->devices = usecase->stream.out->devices; 503 in_snd_device = SND_DEVICE_NONE; 504 if (out_snd_device == SND_DEVICE_NONE) { 505 out_snd_device = platform_get_output_snd_device(adev->platform, 506 usecase->stream.out->devices); 507 if (usecase->stream.out == adev->primary_output && 508 adev->active_input && 509 adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) { 510 select_devices(adev, adev->active_input->usecase); 511 } 512 } 513 } else if (usecase->type == PCM_CAPTURE) { 514 usecase->devices = usecase->stream.in->device; 515 out_snd_device = SND_DEVICE_NONE; 516 if (in_snd_device == SND_DEVICE_NONE) { 517 if (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION && 518 adev->primary_output && !adev->primary_output->standby) { 519 in_snd_device = platform_get_input_snd_device(adev->platform, 520 adev->primary_output->devices); 521 } else { 522 in_snd_device = platform_get_input_snd_device(adev->platform, 523 AUDIO_DEVICE_NONE); 524 } 525 } 526 } 527 } 528 529 if (out_snd_device == usecase->out_snd_device && 530 in_snd_device == usecase->in_snd_device) { 531 return 0; 532 } 533 534 ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__, 535 out_snd_device, platform_get_snd_device_name(out_snd_device), 536 in_snd_device, platform_get_snd_device_name(in_snd_device)); 537 538 /* 539 * Limitation: While in call, to do a device switch we need to disable 540 * and enable both RX and TX devices though one of them is same as current 541 * device. 542 */ 543 if (usecase->type == VOICE_CALL) { 544 status = platform_switch_voice_call_device_pre(adev->platform); 545 } 546 547 /* Disable current sound devices */ 548 if (usecase->out_snd_device != SND_DEVICE_NONE) { 549 disable_audio_route(adev, usecase); 550 disable_snd_device(adev, usecase->out_snd_device); 551 } 552 553 if (usecase->in_snd_device != SND_DEVICE_NONE) { 554 disable_audio_route(adev, usecase); 555 disable_snd_device(adev, usecase->in_snd_device); 556 } 557 558 /* Applicable only on the targets that has external modem. 559 * New device information should be sent to modem before enabling 560 * the devices to reduce in-call device switch time. 561 */ 562 if (usecase->type == VOICE_CALL) 563 status = platform_switch_voice_call_enable_device_config(adev->platform, 564 out_snd_device, 565 in_snd_device); 566 567 /* Enable new sound devices */ 568 if (out_snd_device != SND_DEVICE_NONE) { 569 if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) 570 check_usecases_codec_backend(adev, usecase, out_snd_device); 571 enable_snd_device(adev, out_snd_device); 572 } 573 574 if (in_snd_device != SND_DEVICE_NONE) { 575 check_and_route_capture_usecases(adev, usecase, in_snd_device); 576 enable_snd_device(adev, in_snd_device); 577 } 578 579 if (usecase->type == VOICE_CALL) 580 status = platform_switch_voice_call_device_post(adev->platform, 581 out_snd_device, 582 in_snd_device); 583 584 usecase->in_snd_device = in_snd_device; 585 usecase->out_snd_device = out_snd_device; 586 587 enable_audio_route(adev, usecase); 588 589 /* Applicable only on the targets that has external modem. 590 * Enable device command should be sent to modem only after 591 * enabling voice call mixer controls 592 */ 593 if (usecase->type == VOICE_CALL) 594 status = platform_switch_voice_call_usecase_route_post(adev->platform, 595 out_snd_device, 596 in_snd_device); 597 598 return status; 599} 600 601static int stop_input_stream(struct stream_in *in) 602{ 603 int i, ret = 0; 604 struct audio_usecase *uc_info; 605 struct audio_device *adev = in->dev; 606 607 adev->active_input = NULL; 608 609 ALOGV("%s: enter: usecase(%d: %s)", __func__, 610 in->usecase, use_case_table[in->usecase]); 611 uc_info = get_usecase_from_list(adev, in->usecase); 612 if (uc_info == NULL) { 613 ALOGE("%s: Could not find the usecase (%d) in the list", 614 __func__, in->usecase); 615 return -EINVAL; 616 } 617 618 /* 1. Disable stream specific mixer controls */ 619 disable_audio_route(adev, uc_info); 620 621 /* 2. Disable the tx device */ 622 disable_snd_device(adev, uc_info->in_snd_device); 623 624 list_remove(&uc_info->list); 625 free(uc_info); 626 627 ALOGV("%s: exit: status(%d)", __func__, ret); 628 return ret; 629} 630 631int start_input_stream(struct stream_in *in) 632{ 633 /* 1. Enable output device and stream routing controls */ 634 int ret = 0; 635 struct audio_usecase *uc_info; 636 struct audio_device *adev = in->dev; 637 638 ALOGV("%s: enter: usecase(%d)", __func__, in->usecase); 639 in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE); 640 if (in->pcm_device_id < 0) { 641 ALOGE("%s: Could not find PCM device id for the usecase(%d)", 642 __func__, in->usecase); 643 ret = -EINVAL; 644 goto error_config; 645 } 646 647 adev->active_input = in; 648 uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase)); 649 uc_info->id = in->usecase; 650 uc_info->type = PCM_CAPTURE; 651 uc_info->stream.in = in; 652 uc_info->devices = in->device; 653 uc_info->in_snd_device = SND_DEVICE_NONE; 654 uc_info->out_snd_device = SND_DEVICE_NONE; 655 656 list_add_tail(&adev->usecase_list, &uc_info->list); 657 select_devices(adev, in->usecase); 658 659 ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d", 660 __func__, adev->snd_card, in->pcm_device_id, in->config.channels); 661 in->pcm = pcm_open(adev->snd_card, in->pcm_device_id, 662 PCM_IN, &in->config); 663 if (in->pcm && !pcm_is_ready(in->pcm)) { 664 ALOGE("%s: %s", __func__, pcm_get_error(in->pcm)); 665 pcm_close(in->pcm); 666 in->pcm = NULL; 667 ret = -EIO; 668 goto error_open; 669 } 670 ALOGV("%s: exit", __func__); 671 return ret; 672 673error_open: 674 stop_input_stream(in); 675 676error_config: 677 adev->active_input = NULL; 678 ALOGD("%s: exit: status(%d)", __func__, ret); 679 680 return ret; 681} 682 683/* must be called with out->lock locked */ 684static int send_offload_cmd_l(struct stream_out* out, int command) 685{ 686 struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd)); 687 688 ALOGVV("%s %d", __func__, command); 689 690 cmd->cmd = command; 691 list_add_tail(&out->offload_cmd_list, &cmd->node); 692 pthread_cond_signal(&out->offload_cond); 693 return 0; 694} 695 696/* must be called iwth out->lock locked */ 697static void stop_compressed_output_l(struct stream_out *out) 698{ 699 out->offload_state = OFFLOAD_STATE_IDLE; 700 out->playback_started = 0; 701 out->send_new_metadata = 1; 702 if (out->compr != NULL) { 703 compress_stop(out->compr); 704 while (out->offload_thread_blocked) { 705 pthread_cond_wait(&out->cond, &out->lock); 706 } 707 } 708} 709 710static void *offload_thread_loop(void *context) 711{ 712 struct stream_out *out = (struct stream_out *) context; 713 struct listnode *item; 714 715 out->offload_state = OFFLOAD_STATE_IDLE; 716 out->playback_started = 0; 717 718 setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO); 719 set_sched_policy(0, SP_FOREGROUND); 720 prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0); 721 722 ALOGV("%s", __func__); 723 pthread_mutex_lock(&out->lock); 724 for (;;) { 725 struct offload_cmd *cmd = NULL; 726 stream_callback_event_t event; 727 bool send_callback = false; 728 729 ALOGVV("%s offload_cmd_list %d out->offload_state %d", 730 __func__, list_empty(&out->offload_cmd_list), 731 out->offload_state); 732 if (list_empty(&out->offload_cmd_list)) { 733 ALOGV("%s SLEEPING", __func__); 734 pthread_cond_wait(&out->offload_cond, &out->lock); 735 ALOGV("%s RUNNING", __func__); 736 continue; 737 } 738 739 item = list_head(&out->offload_cmd_list); 740 cmd = node_to_item(item, struct offload_cmd, node); 741 list_remove(item); 742 743 ALOGVV("%s STATE %d CMD %d out->compr %p", 744 __func__, out->offload_state, cmd->cmd, out->compr); 745 746 if (cmd->cmd == OFFLOAD_CMD_EXIT) { 747 free(cmd); 748 break; 749 } 750 751 if (out->compr == NULL) { 752 ALOGE("%s: Compress handle is NULL", __func__); 753 pthread_cond_signal(&out->cond); 754 continue; 755 } 756 out->offload_thread_blocked = true; 757 pthread_mutex_unlock(&out->lock); 758 send_callback = false; 759 switch(cmd->cmd) { 760 case OFFLOAD_CMD_WAIT_FOR_BUFFER: 761 compress_wait(out->compr, -1); 762 send_callback = true; 763 event = STREAM_CBK_EVENT_WRITE_READY; 764 break; 765 case OFFLOAD_CMD_PARTIAL_DRAIN: 766 compress_next_track(out->compr); 767 compress_partial_drain(out->compr); 768 send_callback = true; 769 event = STREAM_CBK_EVENT_DRAIN_READY; 770 break; 771 case OFFLOAD_CMD_DRAIN: 772 compress_drain(out->compr); 773 send_callback = true; 774 event = STREAM_CBK_EVENT_DRAIN_READY; 775 break; 776 default: 777 ALOGE("%s unknown command received: %d", __func__, cmd->cmd); 778 break; 779 } 780 pthread_mutex_lock(&out->lock); 781 out->offload_thread_blocked = false; 782 pthread_cond_signal(&out->cond); 783 if (send_callback) { 784 out->offload_callback(event, NULL, out->offload_cookie); 785 } 786 free(cmd); 787 } 788 789 pthread_cond_signal(&out->cond); 790 while (!list_empty(&out->offload_cmd_list)) { 791 item = list_head(&out->offload_cmd_list); 792 list_remove(item); 793 free(node_to_item(item, struct offload_cmd, node)); 794 } 795 pthread_mutex_unlock(&out->lock); 796 797 return NULL; 798} 799 800static int create_offload_callback_thread(struct stream_out *out) 801{ 802 pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL); 803 list_init(&out->offload_cmd_list); 804 pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL, 805 offload_thread_loop, out); 806 return 0; 807} 808 809static int destroy_offload_callback_thread(struct stream_out *out) 810{ 811 pthread_mutex_lock(&out->lock); 812 stop_compressed_output_l(out); 813 send_offload_cmd_l(out, OFFLOAD_CMD_EXIT); 814 815 pthread_mutex_unlock(&out->lock); 816 pthread_join(out->offload_thread, (void **) NULL); 817 pthread_cond_destroy(&out->offload_cond); 818 819 return 0; 820} 821 822static bool allow_hdmi_channel_config(struct audio_device *adev) 823{ 824 struct listnode *node; 825 struct audio_usecase *usecase; 826 bool ret = true; 827 828 list_for_each(node, &adev->usecase_list) { 829 usecase = node_to_item(node, struct audio_usecase, list); 830 if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { 831 /* 832 * If voice call is already existing, do not proceed further to avoid 833 * disabling/enabling both RX and TX devices, CSD calls, etc. 834 * Once the voice call done, the HDMI channels can be configured to 835 * max channels of remaining use cases. 836 */ 837 if (usecase->id == USECASE_VOICE_CALL) { 838 ALOGD("%s: voice call is active, no change in HDMI channels", 839 __func__); 840 ret = false; 841 break; 842 } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) { 843 ALOGD("%s: multi channel playback is active, " 844 "no change in HDMI channels", __func__); 845 ret = false; 846 break; 847 } 848 } 849 } 850 return ret; 851} 852 853static int check_and_set_hdmi_channels(struct audio_device *adev, 854 unsigned int channels) 855{ 856 struct listnode *node; 857 struct audio_usecase *usecase; 858 859 /* Check if change in HDMI channel config is allowed */ 860 if (!allow_hdmi_channel_config(adev)) 861 return 0; 862 863 if (channels == adev->cur_hdmi_channels) { 864 ALOGD("%s: Requested channels are same as current", __func__); 865 return 0; 866 } 867 868 platform_set_hdmi_channels(adev->platform, channels); 869 adev->cur_hdmi_channels = channels; 870 871 /* 872 * Deroute all the playback streams routed to HDMI so that 873 * the back end is deactivated. Note that backend will not 874 * be deactivated if any one stream is connected to it. 875 */ 876 list_for_each(node, &adev->usecase_list) { 877 usecase = node_to_item(node, struct audio_usecase, list); 878 if (usecase->type == PCM_PLAYBACK && 879 usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { 880 disable_audio_route(adev, usecase); 881 } 882 } 883 884 /* 885 * Enable all the streams disabled above. Now the HDMI backend 886 * will be activated with new channel configuration 887 */ 888 list_for_each(node, &adev->usecase_list) { 889 usecase = node_to_item(node, struct audio_usecase, list); 890 if (usecase->type == PCM_PLAYBACK && 891 usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { 892 enable_audio_route(adev, usecase); 893 } 894 } 895 896 return 0; 897} 898 899static int stop_output_stream(struct stream_out *out) 900{ 901 int i, ret = 0; 902 struct audio_usecase *uc_info; 903 struct audio_device *adev = out->dev; 904 905 ALOGV("%s: enter: usecase(%d: %s)", __func__, 906 out->usecase, use_case_table[out->usecase]); 907 uc_info = get_usecase_from_list(adev, out->usecase); 908 if (uc_info == NULL) { 909 ALOGE("%s: Could not find the usecase (%d) in the list", 910 __func__, out->usecase); 911 return -EINVAL; 912 } 913 914 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD && 915 adev->visualizer_stop_output != NULL) 916 adev->visualizer_stop_output(out->handle); 917 918 /* 1. Get and set stream specific mixer controls */ 919 disable_audio_route(adev, uc_info); 920 921 /* 2. Disable the rx device */ 922 disable_snd_device(adev, uc_info->out_snd_device); 923 924 list_remove(&uc_info->list); 925 free(uc_info); 926 927 /* Must be called after removing the usecase from list */ 928 if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) 929 check_and_set_hdmi_channels(adev, DEFAULT_HDMI_OUT_CHANNELS); 930 931 ALOGV("%s: exit: status(%d)", __func__, ret); 932 return ret; 933} 934 935int start_output_stream(struct stream_out *out) 936{ 937 int ret = 0; 938 struct audio_usecase *uc_info; 939 struct audio_device *adev = out->dev; 940 941 ALOGV("%s: enter: usecase(%d: %s) devices(%#x)", 942 __func__, out->usecase, use_case_table[out->usecase], out->devices); 943 out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK); 944 if (out->pcm_device_id < 0) { 945 ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)", 946 __func__, out->pcm_device_id, out->usecase); 947 ret = -EINVAL; 948 goto error_config; 949 } 950 951 uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase)); 952 uc_info->id = out->usecase; 953 uc_info->type = PCM_PLAYBACK; 954 uc_info->stream.out = out; 955 uc_info->devices = out->devices; 956 uc_info->in_snd_device = SND_DEVICE_NONE; 957 uc_info->out_snd_device = SND_DEVICE_NONE; 958 959 /* This must be called before adding this usecase to the list */ 960 if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) 961 check_and_set_hdmi_channels(adev, out->config.channels); 962 963 list_add_tail(&adev->usecase_list, &uc_info->list); 964 965 select_devices(adev, out->usecase); 966 967 ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)", 968 __func__, adev->snd_card, out->pcm_device_id, out->config.format); 969 if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) { 970 out->pcm = pcm_open(adev->snd_card, out->pcm_device_id, 971 PCM_OUT | PCM_MONOTONIC, &out->config); 972 if (out->pcm && !pcm_is_ready(out->pcm)) { 973 ALOGE("%s: %s", __func__, pcm_get_error(out->pcm)); 974 pcm_close(out->pcm); 975 out->pcm = NULL; 976 ret = -EIO; 977 goto error_open; 978 } 979 } else { 980 out->pcm = NULL; 981 out->compr = compress_open(adev->snd_card, out->pcm_device_id, 982 COMPRESS_IN, &out->compr_config); 983 if (out->compr && !is_compress_ready(out->compr)) { 984 ALOGE("%s: %s", __func__, compress_get_error(out->compr)); 985 compress_close(out->compr); 986 out->compr = NULL; 987 ret = -EIO; 988 goto error_open; 989 } 990 if (out->offload_callback) 991 compress_nonblock(out->compr, out->non_blocking); 992 993 if (adev->visualizer_start_output != NULL) 994 adev->visualizer_start_output(out->handle); 995 } 996 ALOGV("%s: exit", __func__); 997 return 0; 998error_open: 999 stop_output_stream(out); 1000error_config: 1001 return ret; 1002} 1003 1004static int check_input_parameters(uint32_t sample_rate, 1005 audio_format_t format, 1006 int channel_count) 1007{ 1008 if (format != AUDIO_FORMAT_PCM_16_BIT) return -EINVAL; 1009 1010 if ((channel_count < 1) || (channel_count > 2)) return -EINVAL; 1011 1012 switch (sample_rate) { 1013 case 8000: 1014 case 11025: 1015 case 12000: 1016 case 16000: 1017 case 22050: 1018 case 24000: 1019 case 32000: 1020 case 44100: 1021 case 48000: 1022 break; 1023 default: 1024 return -EINVAL; 1025 } 1026 1027 return 0; 1028} 1029 1030static size_t get_input_buffer_size(uint32_t sample_rate, 1031 audio_format_t format, 1032 int channel_count) 1033{ 1034 size_t size = 0; 1035 1036 if (check_input_parameters(sample_rate, format, channel_count) != 0) 1037 return 0; 1038 1039 size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000; 1040 if (sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE) 1041 size = configured_low_latency_capture_period_size; 1042 /* ToDo: should use frame_size computed based on the format and 1043 channel_count here. */ 1044 size *= sizeof(short) * channel_count; 1045 1046 /* make sure the size is multiple of 32 bytes 1047 * At 48 kHz mono 16-bit PCM: 1048 * 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15) 1049 * 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10) 1050 */ 1051 size += 0x1f; 1052 size &= ~0x1f; 1053 1054 return size; 1055} 1056 1057static uint32_t out_get_sample_rate(const struct audio_stream *stream) 1058{ 1059 struct stream_out *out = (struct stream_out *)stream; 1060 1061 return out->sample_rate; 1062} 1063 1064static int out_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused) 1065{ 1066 return -ENOSYS; 1067} 1068 1069static size_t out_get_buffer_size(const struct audio_stream *stream) 1070{ 1071 struct stream_out *out = (struct stream_out *)stream; 1072 1073 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1074 return out->compr_config.fragment_size; 1075 } 1076 1077 return out->config.period_size * audio_stream_frame_size(stream); 1078} 1079 1080static uint32_t out_get_channels(const struct audio_stream *stream) 1081{ 1082 struct stream_out *out = (struct stream_out *)stream; 1083 1084 return out->channel_mask; 1085} 1086 1087static audio_format_t out_get_format(const struct audio_stream *stream) 1088{ 1089 struct stream_out *out = (struct stream_out *)stream; 1090 1091 return out->format; 1092} 1093 1094static int out_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) 1095{ 1096 return -ENOSYS; 1097} 1098 1099static int out_standby(struct audio_stream *stream) 1100{ 1101 struct stream_out *out = (struct stream_out *)stream; 1102 struct audio_device *adev = out->dev; 1103 1104 ALOGV("%s: enter: usecase(%d: %s)", __func__, 1105 out->usecase, use_case_table[out->usecase]); 1106 1107 pthread_mutex_lock(&out->lock); 1108 if (!out->standby) { 1109 pthread_mutex_lock(&adev->lock); 1110 out->standby = true; 1111 if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1112 if (out->pcm) { 1113 pcm_close(out->pcm); 1114 out->pcm = NULL; 1115 } 1116 } else { 1117 stop_compressed_output_l(out); 1118 out->gapless_mdata.encoder_delay = 0; 1119 out->gapless_mdata.encoder_padding = 0; 1120 if (out->compr != NULL) { 1121 compress_close(out->compr); 1122 out->compr = NULL; 1123 } 1124 } 1125 stop_output_stream(out); 1126 pthread_mutex_unlock(&adev->lock); 1127 } 1128 pthread_mutex_unlock(&out->lock); 1129 ALOGV("%s: exit", __func__); 1130 return 0; 1131} 1132 1133static int out_dump(const struct audio_stream *stream __unused, int fd __unused) 1134{ 1135 return 0; 1136} 1137 1138static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms) 1139{ 1140 int ret = 0; 1141 char value[32]; 1142 struct compr_gapless_mdata tmp_mdata; 1143 1144 if (!out || !parms) { 1145 return -EINVAL; 1146 } 1147 1148 ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value)); 1149 if (ret >= 0) { 1150 tmp_mdata.encoder_delay = atoi(value); //whats a good limit check? 1151 } else { 1152 return -EINVAL; 1153 } 1154 1155 ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value)); 1156 if (ret >= 0) { 1157 tmp_mdata.encoder_padding = atoi(value); 1158 } else { 1159 return -EINVAL; 1160 } 1161 1162 out->gapless_mdata = tmp_mdata; 1163 out->send_new_metadata = 1; 1164 ALOGV("%s new encoder delay %u and padding %u", __func__, 1165 out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding); 1166 1167 return 0; 1168} 1169 1170 1171static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 1172{ 1173 struct stream_out *out = (struct stream_out *)stream; 1174 struct audio_device *adev = out->dev; 1175 struct audio_usecase *usecase; 1176 struct listnode *node; 1177 struct str_parms *parms; 1178 char value[32]; 1179 int ret, val = 0; 1180 bool select_new_device = false; 1181 int status = 0; 1182 1183 ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s", 1184 __func__, out->usecase, use_case_table[out->usecase], kvpairs); 1185 parms = str_parms_create_str(kvpairs); 1186 ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); 1187 if (ret >= 0) { 1188 val = atoi(value); 1189 pthread_mutex_lock(&out->lock); 1190 pthread_mutex_lock(&adev->lock); 1191 1192 /* 1193 * When HDMI cable is unplugged the music playback is paused and 1194 * the policy manager sends routing=0. But the audioflinger 1195 * continues to write data until standby time (3sec). 1196 * As the HDMI core is turned off, the write gets blocked. 1197 * Avoid this by routing audio to speaker until standby. 1198 */ 1199 if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL && 1200 val == AUDIO_DEVICE_NONE) { 1201 val = AUDIO_DEVICE_OUT_SPEAKER; 1202 } 1203 1204 /* 1205 * select_devices() call below switches all the usecases on the same 1206 * backend to the new device. Refer to check_usecases_codec_backend() in 1207 * the select_devices(). But how do we undo this? 1208 * 1209 * For example, music playback is active on headset (deep-buffer usecase) 1210 * and if we go to ringtones and select a ringtone, low-latency usecase 1211 * will be started on headset+speaker. As we can't enable headset+speaker 1212 * and headset devices at the same time, select_devices() switches the music 1213 * playback to headset+speaker while starting low-lateny usecase for ringtone. 1214 * So when the ringtone playback is completed, how do we undo the same? 1215 * 1216 * We are relying on the out_set_parameters() call on deep-buffer output, 1217 * once the ringtone playback is ended. 1218 * NOTE: We should not check if the current devices are same as new devices. 1219 * Because select_devices() must be called to switch back the music 1220 * playback to headset. 1221 */ 1222 if (val != 0) { 1223 out->devices = val; 1224 1225 if (!out->standby) 1226 select_devices(adev, out->usecase); 1227 1228 if ((adev->mode == AUDIO_MODE_IN_CALL) && 1229 !voice_is_in_call(adev) && 1230 (out == adev->primary_output)) { 1231 ret = voice_start_call(adev); 1232 } else if ((adev->mode == AUDIO_MODE_IN_CALL) && 1233 voice_is_in_call(adev) && 1234 (out == adev->primary_output)) { 1235 voice_update_devices_for_all_voice_usecases(adev); 1236 } 1237 } 1238 1239 if ((adev->mode == AUDIO_MODE_NORMAL) && 1240 voice_is_in_call(adev) && 1241 (out == adev->primary_output)) { 1242 ret = voice_stop_call(adev); 1243 } 1244 1245 pthread_mutex_unlock(&adev->lock); 1246 pthread_mutex_unlock(&out->lock); 1247 } 1248 1249 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1250 parse_compress_metadata(out, parms); 1251 } 1252 1253 str_parms_destroy(parms); 1254 ALOGV("%s: exit: code(%d)", __func__, status); 1255 return status; 1256} 1257 1258static char* out_get_parameters(const struct audio_stream *stream, const char *keys) 1259{ 1260 struct stream_out *out = (struct stream_out *)stream; 1261 struct str_parms *query = str_parms_create_str(keys); 1262 char *str; 1263 char value[256]; 1264 struct str_parms *reply = str_parms_create(); 1265 size_t i, j; 1266 int ret; 1267 bool first = true; 1268 ALOGV("%s: enter: keys - %s", __func__, keys); 1269 ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value)); 1270 if (ret >= 0) { 1271 value[0] = '\0'; 1272 i = 0; 1273 while (out->supported_channel_masks[i] != 0) { 1274 for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) { 1275 if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) { 1276 if (!first) { 1277 strcat(value, "|"); 1278 } 1279 strcat(value, out_channels_name_to_enum_table[j].name); 1280 first = false; 1281 break; 1282 } 1283 } 1284 i++; 1285 } 1286 str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value); 1287 str = str_parms_to_str(reply); 1288 } else { 1289 str = strdup(keys); 1290 } 1291 str_parms_destroy(query); 1292 str_parms_destroy(reply); 1293 ALOGV("%s: exit: returns - %s", __func__, str); 1294 return str; 1295} 1296 1297static uint32_t out_get_latency(const struct audio_stream_out *stream) 1298{ 1299 struct stream_out *out = (struct stream_out *)stream; 1300 1301 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) 1302 return COMPRESS_OFFLOAD_PLAYBACK_LATENCY; 1303 1304 return (out->config.period_count * out->config.period_size * 1000) / 1305 (out->config.rate); 1306} 1307 1308static int out_set_volume(struct audio_stream_out *stream, float left, 1309 float right) 1310{ 1311 struct stream_out *out = (struct stream_out *)stream; 1312 int volume[2]; 1313 1314 if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) { 1315 /* only take left channel into account: the API is for stereo anyway */ 1316 out->muted = (left == 0.0f); 1317 return 0; 1318 } else if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1319 const char *mixer_ctl_name = "Compress Playback Volume"; 1320 struct audio_device *adev = out->dev; 1321 struct mixer_ctl *ctl; 1322 1323 ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); 1324 if (!ctl) { 1325 ALOGE("%s: Could not get ctl for mixer cmd - %s", 1326 __func__, mixer_ctl_name); 1327 return -EINVAL; 1328 } 1329 volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX); 1330 volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX); 1331 mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0])); 1332 return 0; 1333 } 1334 1335 return -ENOSYS; 1336} 1337 1338static ssize_t out_write(struct audio_stream_out *stream, const void *buffer, 1339 size_t bytes) 1340{ 1341 struct stream_out *out = (struct stream_out *)stream; 1342 struct audio_device *adev = out->dev; 1343 ssize_t ret = 0; 1344 1345 pthread_mutex_lock(&out->lock); 1346 if (out->standby) { 1347 out->standby = false; 1348 pthread_mutex_lock(&adev->lock); 1349 ret = start_output_stream(out); 1350 pthread_mutex_unlock(&adev->lock); 1351 /* ToDo: If use case is compress offload should return 0 */ 1352 if (ret != 0) { 1353 out->standby = true; 1354 goto exit; 1355 } 1356 } 1357 1358 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1359 ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes); 1360 if (out->send_new_metadata) { 1361 ALOGVV("send new gapless metadata"); 1362 compress_set_gapless_metadata(out->compr, &out->gapless_mdata); 1363 out->send_new_metadata = 0; 1364 } 1365 1366 ret = compress_write(out->compr, buffer, bytes); 1367 ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret); 1368 if (ret >= 0 && ret < (ssize_t)bytes) { 1369 send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); 1370 } 1371 if (!out->playback_started) { 1372 compress_start(out->compr); 1373 out->playback_started = 1; 1374 out->offload_state = OFFLOAD_STATE_PLAYING; 1375 } 1376 pthread_mutex_unlock(&out->lock); 1377 return ret; 1378 } else { 1379 if (out->pcm) { 1380 if (out->muted) 1381 memset((void *)buffer, 0, bytes); 1382 ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes); 1383 ret = pcm_write(out->pcm, (void *)buffer, bytes); 1384 if (ret == 0) 1385 out->written += bytes / (out->config.channels * sizeof(short)); 1386 } 1387 } 1388 1389exit: 1390 pthread_mutex_unlock(&out->lock); 1391 1392 if (ret != 0) { 1393 if (out->pcm) 1394 ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(out->pcm)); 1395 out_standby(&out->stream.common); 1396 usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) / 1397 out_get_sample_rate(&out->stream.common)); 1398 } 1399 return bytes; 1400} 1401 1402static int out_get_render_position(const struct audio_stream_out *stream, 1403 uint32_t *dsp_frames) 1404{ 1405 struct stream_out *out = (struct stream_out *)stream; 1406 *dsp_frames = 0; 1407 if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) { 1408 pthread_mutex_lock(&out->lock); 1409 if (out->compr != NULL) { 1410 compress_get_tstamp(out->compr, (unsigned long *)dsp_frames, 1411 &out->sample_rate); 1412 ALOGVV("%s rendered frames %d sample_rate %d", 1413 __func__, *dsp_frames, out->sample_rate); 1414 } 1415 pthread_mutex_unlock(&out->lock); 1416 return 0; 1417 } else 1418 return -EINVAL; 1419} 1420 1421static int out_add_audio_effect(const struct audio_stream *stream __unused, 1422 effect_handle_t effect __unused) 1423{ 1424 return 0; 1425} 1426 1427static int out_remove_audio_effect(const struct audio_stream *stream __unused, 1428 effect_handle_t effect __unused) 1429{ 1430 return 0; 1431} 1432 1433static int out_get_next_write_timestamp(const struct audio_stream_out *stream __unused, 1434 int64_t *timestamp __unused) 1435{ 1436 return -EINVAL; 1437} 1438 1439static int out_get_presentation_position(const struct audio_stream_out *stream, 1440 uint64_t *frames, struct timespec *timestamp) 1441{ 1442 struct stream_out *out = (struct stream_out *)stream; 1443 int ret = -1; 1444 unsigned long dsp_frames; 1445 1446 pthread_mutex_lock(&out->lock); 1447 1448 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1449 if (out->compr != NULL) { 1450 compress_get_tstamp(out->compr, &dsp_frames, 1451 &out->sample_rate); 1452 ALOGVV("%s rendered frames %ld sample_rate %d", 1453 __func__, dsp_frames, out->sample_rate); 1454 *frames = dsp_frames; 1455 ret = 0; 1456 /* this is the best we can do */ 1457 clock_gettime(CLOCK_MONOTONIC, timestamp); 1458 } 1459 } else { 1460 if (out->pcm) { 1461 size_t avail; 1462 if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) { 1463 size_t kernel_buffer_size = out->config.period_size * out->config.period_count; 1464 int64_t signed_frames = out->written - kernel_buffer_size + avail; 1465 // This adjustment accounts for buffering after app processor. 1466 // It is based on estimated DSP latency per use case, rather than exact. 1467 signed_frames -= 1468 (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL); 1469 1470 // It would be unusual for this value to be negative, but check just in case ... 1471 if (signed_frames >= 0) { 1472 *frames = signed_frames; 1473 ret = 0; 1474 } 1475 } 1476 } 1477 } 1478 1479 pthread_mutex_unlock(&out->lock); 1480 1481 return ret; 1482} 1483 1484static int out_set_callback(struct audio_stream_out *stream, 1485 stream_callback_t callback, void *cookie) 1486{ 1487 struct stream_out *out = (struct stream_out *)stream; 1488 1489 ALOGV("%s", __func__); 1490 pthread_mutex_lock(&out->lock); 1491 out->offload_callback = callback; 1492 out->offload_cookie = cookie; 1493 pthread_mutex_unlock(&out->lock); 1494 return 0; 1495} 1496 1497static int out_pause(struct audio_stream_out* stream) 1498{ 1499 struct stream_out *out = (struct stream_out *)stream; 1500 int status = -ENOSYS; 1501 ALOGV("%s", __func__); 1502 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1503 pthread_mutex_lock(&out->lock); 1504 if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) { 1505 status = compress_pause(out->compr); 1506 out->offload_state = OFFLOAD_STATE_PAUSED; 1507 } 1508 pthread_mutex_unlock(&out->lock); 1509 } 1510 return status; 1511} 1512 1513static int out_resume(struct audio_stream_out* stream) 1514{ 1515 struct stream_out *out = (struct stream_out *)stream; 1516 int status = -ENOSYS; 1517 ALOGV("%s", __func__); 1518 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1519 status = 0; 1520 pthread_mutex_lock(&out->lock); 1521 if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) { 1522 status = compress_resume(out->compr); 1523 out->offload_state = OFFLOAD_STATE_PLAYING; 1524 } 1525 pthread_mutex_unlock(&out->lock); 1526 } 1527 return status; 1528} 1529 1530static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type ) 1531{ 1532 struct stream_out *out = (struct stream_out *)stream; 1533 int status = -ENOSYS; 1534 ALOGV("%s", __func__); 1535 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1536 pthread_mutex_lock(&out->lock); 1537 if (type == AUDIO_DRAIN_EARLY_NOTIFY) 1538 status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN); 1539 else 1540 status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN); 1541 pthread_mutex_unlock(&out->lock); 1542 } 1543 return status; 1544} 1545 1546static int out_flush(struct audio_stream_out* stream) 1547{ 1548 struct stream_out *out = (struct stream_out *)stream; 1549 ALOGV("%s", __func__); 1550 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1551 pthread_mutex_lock(&out->lock); 1552 stop_compressed_output_l(out); 1553 pthread_mutex_unlock(&out->lock); 1554 return 0; 1555 } 1556 return -ENOSYS; 1557} 1558 1559/** audio_stream_in implementation **/ 1560static uint32_t in_get_sample_rate(const struct audio_stream *stream) 1561{ 1562 struct stream_in *in = (struct stream_in *)stream; 1563 1564 return in->config.rate; 1565} 1566 1567static int in_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused) 1568{ 1569 return -ENOSYS; 1570} 1571 1572static size_t in_get_buffer_size(const struct audio_stream *stream) 1573{ 1574 struct stream_in *in = (struct stream_in *)stream; 1575 1576 return in->config.period_size * audio_stream_frame_size(stream); 1577} 1578 1579static uint32_t in_get_channels(const struct audio_stream *stream) 1580{ 1581 struct stream_in *in = (struct stream_in *)stream; 1582 1583 return in->channel_mask; 1584} 1585 1586static audio_format_t in_get_format(const struct audio_stream *stream __unused) 1587{ 1588 return AUDIO_FORMAT_PCM_16_BIT; 1589} 1590 1591static int in_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) 1592{ 1593 return -ENOSYS; 1594} 1595 1596static int in_standby(struct audio_stream *stream) 1597{ 1598 struct stream_in *in = (struct stream_in *)stream; 1599 struct audio_device *adev = in->dev; 1600 int status = 0; 1601 ALOGV("%s: enter", __func__); 1602 pthread_mutex_lock(&in->lock); 1603 if (!in->standby) { 1604 pthread_mutex_lock(&adev->lock); 1605 in->standby = true; 1606 if (in->pcm) { 1607 pcm_close(in->pcm); 1608 in->pcm = NULL; 1609 } 1610 status = stop_input_stream(in); 1611 pthread_mutex_unlock(&adev->lock); 1612 } 1613 pthread_mutex_unlock(&in->lock); 1614 ALOGV("%s: exit: status(%d)", __func__, status); 1615 return status; 1616} 1617 1618static int in_dump(const struct audio_stream *stream __unused, int fd __unused) 1619{ 1620 return 0; 1621} 1622 1623static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 1624{ 1625 struct stream_in *in = (struct stream_in *)stream; 1626 struct audio_device *adev = in->dev; 1627 struct str_parms *parms; 1628 char *str; 1629 char value[32]; 1630 int ret, val = 0; 1631 int status = 0; 1632 1633 ALOGV("%s: enter: kvpairs=%s", __func__, kvpairs); 1634 parms = str_parms_create_str(kvpairs); 1635 1636 ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value)); 1637 1638 pthread_mutex_lock(&in->lock); 1639 pthread_mutex_lock(&adev->lock); 1640 if (ret >= 0) { 1641 val = atoi(value); 1642 /* no audio source uses val == 0 */ 1643 if ((in->source != val) && (val != 0)) { 1644 in->source = val; 1645 } 1646 } 1647 1648 ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); 1649 1650 if (ret >= 0) { 1651 val = atoi(value); 1652 if ((in->device != val) && (val != 0)) { 1653 in->device = val; 1654 /* If recording is in progress, change the tx device to new device */ 1655 if (!in->standby) 1656 status = select_devices(adev, in->usecase); 1657 } 1658 } 1659 1660 pthread_mutex_unlock(&adev->lock); 1661 pthread_mutex_unlock(&in->lock); 1662 1663 str_parms_destroy(parms); 1664 ALOGV("%s: exit: status(%d)", __func__, status); 1665 return status; 1666} 1667 1668static char* in_get_parameters(const struct audio_stream *stream __unused, 1669 const char *keys __unused) 1670{ 1671 return strdup(""); 1672} 1673 1674static int in_set_gain(struct audio_stream_in *stream __unused, float gain __unused) 1675{ 1676 return 0; 1677} 1678 1679static ssize_t in_read(struct audio_stream_in *stream, void *buffer, 1680 size_t bytes) 1681{ 1682 struct stream_in *in = (struct stream_in *)stream; 1683 struct audio_device *adev = in->dev; 1684 int i, ret = -1; 1685 1686 pthread_mutex_lock(&in->lock); 1687 if (in->standby) { 1688 pthread_mutex_lock(&adev->lock); 1689 ret = start_input_stream(in); 1690 pthread_mutex_unlock(&adev->lock); 1691 if (ret != 0) { 1692 goto exit; 1693 } 1694 in->standby = 0; 1695 } 1696 1697 if (in->pcm) { 1698 ret = pcm_read(in->pcm, buffer, bytes); 1699 } 1700 1701 /* 1702 * Instead of writing zeroes here, we could trust the hardware 1703 * to always provide zeroes when muted. 1704 */ 1705 if (ret == 0 && voice_get_mic_mute(adev) && !voice_is_in_call(adev)) 1706 memset(buffer, 0, bytes); 1707 1708exit: 1709 pthread_mutex_unlock(&in->lock); 1710 1711 if (ret != 0) { 1712 in_standby(&in->stream.common); 1713 ALOGV("%s: read failed - sleeping for buffer duration", __func__); 1714 usleep(bytes * 1000000 / audio_stream_frame_size(&in->stream.common) / 1715 in_get_sample_rate(&in->stream.common)); 1716 } 1717 return bytes; 1718} 1719 1720static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused) 1721{ 1722 return 0; 1723} 1724 1725static int add_remove_audio_effect(const struct audio_stream *stream, 1726 effect_handle_t effect, 1727 bool enable) 1728{ 1729 struct stream_in *in = (struct stream_in *)stream; 1730 int status = 0; 1731 effect_descriptor_t desc; 1732 1733 status = (*effect)->get_descriptor(effect, &desc); 1734 if (status != 0) 1735 return status; 1736 1737 pthread_mutex_lock(&in->lock); 1738 pthread_mutex_lock(&in->dev->lock); 1739 if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) && 1740 in->enable_aec != enable && 1741 (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) { 1742 in->enable_aec = enable; 1743 if (!in->standby) 1744 select_devices(in->dev, in->usecase); 1745 } 1746 pthread_mutex_unlock(&in->dev->lock); 1747 pthread_mutex_unlock(&in->lock); 1748 1749 return 0; 1750} 1751 1752static int in_add_audio_effect(const struct audio_stream *stream, 1753 effect_handle_t effect) 1754{ 1755 ALOGV("%s: effect %p", __func__, effect); 1756 return add_remove_audio_effect(stream, effect, true); 1757} 1758 1759static int in_remove_audio_effect(const struct audio_stream *stream, 1760 effect_handle_t effect) 1761{ 1762 ALOGV("%s: effect %p", __func__, effect); 1763 return add_remove_audio_effect(stream, effect, false); 1764} 1765 1766static int adev_open_output_stream(struct audio_hw_device *dev, 1767 audio_io_handle_t handle, 1768 audio_devices_t devices, 1769 audio_output_flags_t flags, 1770 struct audio_config *config, 1771 struct audio_stream_out **stream_out) 1772{ 1773 struct audio_device *adev = (struct audio_device *)dev; 1774 struct stream_out *out; 1775 int i, ret; 1776 1777 ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)", 1778 __func__, config->sample_rate, config->channel_mask, devices, flags); 1779 *stream_out = NULL; 1780 out = (struct stream_out *)calloc(1, sizeof(struct stream_out)); 1781 1782 if (devices == AUDIO_DEVICE_NONE) 1783 devices = AUDIO_DEVICE_OUT_SPEAKER; 1784 1785 out->flags = flags; 1786 out->devices = devices; 1787 out->dev = adev; 1788 out->format = config->format; 1789 out->sample_rate = config->sample_rate; 1790 out->channel_mask = AUDIO_CHANNEL_OUT_STEREO; 1791 out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO; 1792 out->handle = handle; 1793 1794 /* Init use case and pcm_config */ 1795 if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT && 1796 !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && 1797 out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { 1798 pthread_mutex_lock(&adev->lock); 1799 ret = read_hdmi_channel_masks(out); 1800 pthread_mutex_unlock(&adev->lock); 1801 if (ret != 0) 1802 goto error_open; 1803 1804 if (config->sample_rate == 0) 1805 config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE; 1806 if (config->channel_mask == 0) 1807 config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1; 1808 1809 out->channel_mask = config->channel_mask; 1810 out->sample_rate = config->sample_rate; 1811 out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH; 1812 out->config = pcm_config_hdmi_multi; 1813 out->config.rate = config->sample_rate; 1814 out->config.channels = popcount(out->channel_mask); 1815 out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2); 1816 } else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1817 if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version || 1818 config->offload_info.size != AUDIO_INFO_INITIALIZER.size) { 1819 ALOGE("%s: Unsupported Offload information", __func__); 1820 ret = -EINVAL; 1821 goto error_open; 1822 } 1823 if (!is_supported_format(config->offload_info.format)) { 1824 ALOGE("%s: Unsupported audio format", __func__); 1825 ret = -EINVAL; 1826 goto error_open; 1827 } 1828 1829 out->compr_config.codec = (struct snd_codec *) 1830 calloc(1, sizeof(struct snd_codec)); 1831 1832 out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD; 1833 if (config->offload_info.channel_mask) 1834 out->channel_mask = config->offload_info.channel_mask; 1835 else if (config->channel_mask) 1836 out->channel_mask = config->channel_mask; 1837 out->format = config->offload_info.format; 1838 out->sample_rate = config->offload_info.sample_rate; 1839 1840 out->stream.set_callback = out_set_callback; 1841 out->stream.pause = out_pause; 1842 out->stream.resume = out_resume; 1843 out->stream.drain = out_drain; 1844 out->stream.flush = out_flush; 1845 1846 out->compr_config.codec->id = 1847 get_snd_codec_id(config->offload_info.format); 1848 out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; 1849 out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS; 1850 out->compr_config.codec->sample_rate = 1851 compress_get_alsa_rate(config->offload_info.sample_rate); 1852 out->compr_config.codec->bit_rate = 1853 config->offload_info.bit_rate; 1854 out->compr_config.codec->ch_in = 1855 popcount(config->channel_mask); 1856 out->compr_config.codec->ch_out = out->compr_config.codec->ch_in; 1857 1858 if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) 1859 out->non_blocking = 1; 1860 1861 out->send_new_metadata = 1; 1862 create_offload_callback_thread(out); 1863 ALOGV("%s: offloaded output offload_info version %04x bit rate %d", 1864 __func__, config->offload_info.version, 1865 config->offload_info.bit_rate); 1866 } else { 1867 if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) { 1868 out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER; 1869 out->config = pcm_config_deep_buffer; 1870 } else { 1871 out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY; 1872 out->config = pcm_config_low_latency; 1873 } 1874 if (config->format != audio_format_from_pcm_format(out->config.format)) { 1875 if (k_enable_extended_precision 1876 && pcm_params_format_test(adev->use_case_table[out->usecase], 1877 pcm_format_from_audio_format(config->format))) { 1878 out->config.format = pcm_format_from_audio_format(config->format); 1879 /* out->format already set to config->format */ 1880 } else { 1881 /* deny the externally proposed config format 1882 * and use the one specified in audio_hw layer configuration. 1883 * Note: out->format is returned by out->stream.common.get_format() 1884 * and is used to set config->format in the code several lines below. 1885 */ 1886 out->format = audio_format_from_pcm_format(out->config.format); 1887 } 1888 } 1889 out->sample_rate = out->config.rate; 1890 } 1891 ALOGV("%s: Usecase(%s) config->format %#x out->config.format %#x\n", 1892 __func__, use_case_table[out->usecase], config->format, out->config.format); 1893 1894 if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) { 1895 if(adev->primary_output == NULL) 1896 adev->primary_output = out; 1897 else { 1898 ALOGE("%s: Primary output is already opened", __func__); 1899 ret = -EEXIST; 1900 goto error_open; 1901 } 1902 } 1903 1904 /* Check if this usecase is already existing */ 1905 pthread_mutex_lock(&adev->lock); 1906 if (get_usecase_from_list(adev, out->usecase) != NULL) { 1907 ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase); 1908 pthread_mutex_unlock(&adev->lock); 1909 ret = -EEXIST; 1910 goto error_open; 1911 } 1912 pthread_mutex_unlock(&adev->lock); 1913 1914 out->stream.common.get_sample_rate = out_get_sample_rate; 1915 out->stream.common.set_sample_rate = out_set_sample_rate; 1916 out->stream.common.get_buffer_size = out_get_buffer_size; 1917 out->stream.common.get_channels = out_get_channels; 1918 out->stream.common.get_format = out_get_format; 1919 out->stream.common.set_format = out_set_format; 1920 out->stream.common.standby = out_standby; 1921 out->stream.common.dump = out_dump; 1922 out->stream.common.set_parameters = out_set_parameters; 1923 out->stream.common.get_parameters = out_get_parameters; 1924 out->stream.common.add_audio_effect = out_add_audio_effect; 1925 out->stream.common.remove_audio_effect = out_remove_audio_effect; 1926 out->stream.get_latency = out_get_latency; 1927 out->stream.set_volume = out_set_volume; 1928 out->stream.write = out_write; 1929 out->stream.get_render_position = out_get_render_position; 1930 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 1931 out->stream.get_presentation_position = out_get_presentation_position; 1932 1933 out->standby = 1; 1934 /* out->muted = false; by calloc() */ 1935 /* out->written = 0; by calloc() */ 1936 1937 pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL); 1938 pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL); 1939 1940 config->format = out->stream.common.get_format(&out->stream.common); 1941 config->channel_mask = out->stream.common.get_channels(&out->stream.common); 1942 config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common); 1943 1944 *stream_out = &out->stream; 1945 ALOGV("%s: exit", __func__); 1946 return 0; 1947 1948error_open: 1949 free(out); 1950 *stream_out = NULL; 1951 ALOGD("%s: exit: ret %d", __func__, ret); 1952 return ret; 1953} 1954 1955static void adev_close_output_stream(struct audio_hw_device *dev __unused, 1956 struct audio_stream_out *stream) 1957{ 1958 struct stream_out *out = (struct stream_out *)stream; 1959 struct audio_device *adev = out->dev; 1960 1961 ALOGV("%s: enter", __func__); 1962 out_standby(&stream->common); 1963 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1964 destroy_offload_callback_thread(out); 1965 1966 if (out->compr_config.codec != NULL) 1967 free(out->compr_config.codec); 1968 } 1969 pthread_cond_destroy(&out->cond); 1970 pthread_mutex_destroy(&out->lock); 1971 free(stream); 1972 ALOGV("%s: exit", __func__); 1973} 1974 1975static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 1976{ 1977 struct audio_device *adev = (struct audio_device *)dev; 1978 struct str_parms *parms; 1979 char *str; 1980 char value[32]; 1981 int val; 1982 int ret; 1983 int status = 0; 1984 1985 ALOGD("%s: enter: %s", __func__, kvpairs); 1986 1987 pthread_mutex_lock(&adev->lock); 1988 1989 parms = str_parms_create_str(kvpairs); 1990 status = voice_set_parameters(adev, parms); 1991 if (status != 0) { 1992 goto done; 1993 } 1994 1995 ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value)); 1996 if (ret >= 0) { 1997 /* When set to false, HAL should disable EC and NS 1998 * But it is currently not supported. 1999 */ 2000 if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) 2001 adev->bluetooth_nrec = true; 2002 else 2003 adev->bluetooth_nrec = false; 2004 } 2005 2006 ret = str_parms_get_str(parms, "screen_state", value, sizeof(value)); 2007 if (ret >= 0) { 2008 if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) 2009 adev->screen_off = false; 2010 else 2011 adev->screen_off = true; 2012 } 2013 2014 ret = str_parms_get_int(parms, "rotation", &val); 2015 if (ret >= 0) { 2016 bool reverse_speakers = false; 2017 switch(val) { 2018 // FIXME: note that the code below assumes that the speakers are in the correct placement 2019 // relative to the user when the device is rotated 90deg from its default rotation. This 2020 // assumption is device-specific, not platform-specific like this code. 2021 case 270: 2022 reverse_speakers = true; 2023 break; 2024 case 0: 2025 case 90: 2026 case 180: 2027 break; 2028 default: 2029 ALOGE("%s: unexpected rotation of %d", __func__, val); 2030 status = -EINVAL; 2031 } 2032 if (status == 0) { 2033 if (adev->speaker_lr_swap != reverse_speakers) { 2034 adev->speaker_lr_swap = reverse_speakers; 2035 // only update the selected device if there is active pcm playback 2036 struct audio_usecase *usecase; 2037 struct listnode *node; 2038 list_for_each(node, &adev->usecase_list) { 2039 usecase = node_to_item(node, struct audio_usecase, list); 2040 if (usecase->type == PCM_PLAYBACK) { 2041 select_devices(adev, usecase->id); 2042 break; 2043 } 2044 } 2045 } 2046 } 2047 } 2048 2049 ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value)); 2050 if (ret >= 0) { 2051 adev->bt_wb_speech_enabled = !strcmp(value, AUDIO_PARAMETER_VALUE_ON); 2052 } 2053 2054 audio_extn_hfp_set_parameters(adev, parms); 2055done: 2056 str_parms_destroy(parms); 2057 pthread_mutex_unlock(&adev->lock); 2058 ALOGV("%s: exit with code(%d)", __func__, status); 2059 return status; 2060} 2061 2062static char* adev_get_parameters(const struct audio_hw_device *dev, 2063 const char *keys) 2064{ 2065 struct audio_device *adev = (struct audio_device *)dev; 2066 struct str_parms *reply = str_parms_create(); 2067 struct str_parms *query = str_parms_create_str(keys); 2068 char *str; 2069 2070 pthread_mutex_lock(&adev->lock); 2071 2072 voice_get_parameters(adev, query, reply); 2073 str = str_parms_to_str(reply); 2074 str_parms_destroy(query); 2075 str_parms_destroy(reply); 2076 2077 pthread_mutex_unlock(&adev->lock); 2078 ALOGV("%s: exit: returns - %s", __func__, str); 2079 return str; 2080} 2081 2082static int adev_init_check(const struct audio_hw_device *dev __unused) 2083{ 2084 return 0; 2085} 2086 2087static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 2088{ 2089 int ret; 2090 struct audio_device *adev = (struct audio_device *)dev; 2091 2092 pthread_mutex_lock(&adev->lock); 2093 ret = voice_set_volume(adev, volume); 2094 pthread_mutex_unlock(&adev->lock); 2095 2096 return ret; 2097} 2098 2099static int adev_set_master_volume(struct audio_hw_device *dev __unused, float volume __unused) 2100{ 2101 return -ENOSYS; 2102} 2103 2104static int adev_get_master_volume(struct audio_hw_device *dev __unused, 2105 float *volume __unused) 2106{ 2107 return -ENOSYS; 2108} 2109 2110static int adev_set_master_mute(struct audio_hw_device *dev __unused, bool muted __unused) 2111{ 2112 return -ENOSYS; 2113} 2114 2115static int adev_get_master_mute(struct audio_hw_device *dev __unused, bool *muted __unused) 2116{ 2117 return -ENOSYS; 2118} 2119 2120static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 2121{ 2122 struct audio_device *adev = (struct audio_device *)dev; 2123 2124 pthread_mutex_lock(&adev->lock); 2125 if (adev->mode != mode) { 2126 ALOGD("%s: mode %d\n", __func__, mode); 2127 adev->mode = mode; 2128 } 2129 pthread_mutex_unlock(&adev->lock); 2130 return 0; 2131} 2132 2133static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 2134{ 2135 int ret; 2136 struct audio_device *adev = (struct audio_device *)dev; 2137 2138 ALOGD("%s: state %d\n", __func__, state); 2139 pthread_mutex_lock(&adev->lock); 2140 ret = voice_set_mic_mute(adev, state); 2141 pthread_mutex_unlock(&adev->lock); 2142 2143 return ret; 2144} 2145 2146static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 2147{ 2148 *state = voice_get_mic_mute((struct audio_device *)dev); 2149 return 0; 2150} 2151 2152static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unused, 2153 const struct audio_config *config) 2154{ 2155 int channel_count = popcount(config->channel_mask); 2156 2157 return get_input_buffer_size(config->sample_rate, config->format, channel_count); 2158} 2159 2160static int adev_open_input_stream(struct audio_hw_device *dev, 2161 audio_io_handle_t handle __unused, 2162 audio_devices_t devices, 2163 struct audio_config *config, 2164 struct audio_stream_in **stream_in) 2165{ 2166 struct audio_device *adev = (struct audio_device *)dev; 2167 struct stream_in *in; 2168 int ret = 0, buffer_size, frame_size; 2169 int channel_count = popcount(config->channel_mask); 2170 2171 ALOGV("%s: enter", __func__); 2172 *stream_in = NULL; 2173 if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) 2174 return -EINVAL; 2175 2176 in = (struct stream_in *)calloc(1, sizeof(struct stream_in)); 2177 2178 pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL); 2179 2180 in->stream.common.get_sample_rate = in_get_sample_rate; 2181 in->stream.common.set_sample_rate = in_set_sample_rate; 2182 in->stream.common.get_buffer_size = in_get_buffer_size; 2183 in->stream.common.get_channels = in_get_channels; 2184 in->stream.common.get_format = in_get_format; 2185 in->stream.common.set_format = in_set_format; 2186 in->stream.common.standby = in_standby; 2187 in->stream.common.dump = in_dump; 2188 in->stream.common.set_parameters = in_set_parameters; 2189 in->stream.common.get_parameters = in_get_parameters; 2190 in->stream.common.add_audio_effect = in_add_audio_effect; 2191 in->stream.common.remove_audio_effect = in_remove_audio_effect; 2192 in->stream.set_gain = in_set_gain; 2193 in->stream.read = in_read; 2194 in->stream.get_input_frames_lost = in_get_input_frames_lost; 2195 2196 in->device = devices; 2197 in->source = AUDIO_SOURCE_DEFAULT; 2198 in->dev = adev; 2199 in->standby = 1; 2200 in->channel_mask = config->channel_mask; 2201 2202 /* Update config params with the requested sample rate and channels */ 2203 in->usecase = USECASE_AUDIO_RECORD; 2204#if LOW_LATENCY_CAPTURE_USE_CASE 2205 if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE) 2206 in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY; 2207#endif 2208 in->config = pcm_config_audio_capture; 2209 in->config.channels = channel_count; 2210 in->config.rate = config->sample_rate; 2211 2212 frame_size = audio_stream_frame_size((struct audio_stream *)in); 2213 buffer_size = get_input_buffer_size(config->sample_rate, 2214 config->format, 2215 channel_count); 2216 in->config.period_size = buffer_size / frame_size; 2217 2218 *stream_in = &in->stream; 2219 ALOGV("%s: exit", __func__); 2220 return 0; 2221 2222err_open: 2223 free(in); 2224 *stream_in = NULL; 2225 return ret; 2226} 2227 2228static void adev_close_input_stream(struct audio_hw_device *dev __unused, 2229 struct audio_stream_in *stream) 2230{ 2231 ALOGV("%s", __func__); 2232 2233 in_standby(&stream->common); 2234 free(stream); 2235 2236 return; 2237} 2238 2239static int adev_dump(const audio_hw_device_t *device __unused, int fd __unused) 2240{ 2241 return 0; 2242} 2243 2244/* verifies input and output devices and their capabilities. 2245 * 2246 * This verification is required when enabling extended bit-depth or 2247 * sampling rates, as not all qcom products support it. 2248 * 2249 * Suitable for calling only on initialization such as adev_open(). 2250 * It fills the audio_device use_case_table[] array. 2251 * 2252 * Has a side-effect that it needs to configure audio routing / devices 2253 * in order to power up the devices and read the device parameters. 2254 * It does not acquire any hw device lock. Should restore the devices 2255 * back to "normal state" upon completion. 2256 */ 2257static int adev_verify_devices(struct audio_device *adev) 2258{ 2259 /* enumeration is a bit difficult because one really wants to pull 2260 * the use_case, device id, etc from the hidden pcm_device_table[]. 2261 * In this case there are the following use cases and device ids. 2262 * 2263 * [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {0, 0}, 2264 * [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = {15, 15}, 2265 * [USECASE_AUDIO_PLAYBACK_MULTI_CH] = {1, 1}, 2266 * [USECASE_AUDIO_PLAYBACK_OFFLOAD] = {9, 9}, 2267 * [USECASE_AUDIO_RECORD] = {0, 0}, 2268 * [USECASE_AUDIO_RECORD_LOW_LATENCY] = {15, 15}, 2269 * [USECASE_VOICE_CALL] = {2, 2}, 2270 * 2271 * USECASE_AUDIO_PLAYBACK_OFFLOAD, USECASE_AUDIO_PLAYBACK_MULTI_CH omitted. 2272 * USECASE_VOICE_CALL omitted, but possible for either input or output. 2273 */ 2274 2275 /* should be the usecases enabled in adev_open_input_stream() */ 2276 static const int test_in_usecases[] = { 2277 USECASE_AUDIO_RECORD, 2278 USECASE_AUDIO_RECORD_LOW_LATENCY, /* does not appear to be used */ 2279 }; 2280 /* should be the usecases enabled in adev_open_output_stream()*/ 2281 static const int test_out_usecases[] = { 2282 USECASE_AUDIO_PLAYBACK_DEEP_BUFFER, 2283 USECASE_AUDIO_PLAYBACK_LOW_LATENCY, 2284 }; 2285 static const usecase_type_t usecase_type_by_dir[] = { 2286 PCM_PLAYBACK, 2287 PCM_CAPTURE, 2288 }; 2289 static const unsigned flags_by_dir[] = { 2290 PCM_OUT, 2291 PCM_IN, 2292 }; 2293 2294 size_t i; 2295 unsigned dir; 2296 const unsigned card_id = adev->snd_card; 2297 char info[512]; /* for possible debug info */ 2298 2299 for (dir = 0; dir < 2; ++dir) { 2300 const usecase_type_t usecase_type = usecase_type_by_dir[dir]; 2301 const unsigned flags_dir = flags_by_dir[dir]; 2302 const size_t testsize = 2303 dir ? ARRAY_SIZE(test_in_usecases) : ARRAY_SIZE(test_out_usecases); 2304 const int *testcases = 2305 dir ? test_in_usecases : test_out_usecases; 2306 const audio_devices_t audio_device = 2307 dir ? AUDIO_DEVICE_IN_BUILTIN_MIC : AUDIO_DEVICE_OUT_SPEAKER; 2308 2309 for (i = 0; i < testsize; ++i) { 2310 const audio_usecase_t audio_usecase = testcases[i]; 2311 int device_id; 2312 snd_device_t snd_device; 2313 struct pcm_params **pparams; 2314 struct stream_out out; 2315 struct stream_in in; 2316 struct audio_usecase uc_info; 2317 int retval; 2318 2319 pparams = &adev->use_case_table[audio_usecase]; 2320 pcm_params_free(*pparams); /* can accept null input */ 2321 *pparams = NULL; 2322 2323 /* find the device ID for the use case (signed, for error) */ 2324 device_id = platform_get_pcm_device_id(audio_usecase, usecase_type); 2325 if (device_id < 0) 2326 continue; 2327 2328 /* prepare structures for device probing */ 2329 memset(&uc_info, 0, sizeof(uc_info)); 2330 uc_info.id = audio_usecase; 2331 uc_info.type = usecase_type; 2332 if (dir) { 2333 adev->active_input = ∈ 2334 memset(&in, 0, sizeof(in)); 2335 in.device = audio_device; 2336 in.source = AUDIO_SOURCE_VOICE_COMMUNICATION; 2337 uc_info.stream.in = ∈ 2338 } else { 2339 adev->active_input = NULL; 2340 } 2341 memset(&out, 0, sizeof(out)); 2342 out.devices = audio_device; /* only field needed in select_devices */ 2343 uc_info.stream.out = &out; 2344 uc_info.devices = audio_device; 2345 uc_info.in_snd_device = SND_DEVICE_NONE; 2346 uc_info.out_snd_device = SND_DEVICE_NONE; 2347 list_add_tail(&adev->usecase_list, &uc_info.list); 2348 2349 /* select device - similar to start_(in/out)put_stream() */ 2350 retval = select_devices(adev, audio_usecase); 2351 if (retval >= 0) { 2352 *pparams = pcm_params_get(card_id, device_id, flags_dir); 2353#if LOG_NDEBUG == 0 2354 if (*pparams) { 2355 ALOGV("%s: (%s) card %d device %d", __func__, 2356 dir ? "input" : "output", card_id, device_id); 2357 pcm_params_to_string(*pparams, info, ARRAY_SIZE(info)); 2358 ALOGV(info); /* print parameters */ 2359 } else { 2360 ALOGV("%s: cannot locate card %d device %d", __func__, card_id, device_id); 2361 } 2362#endif 2363 } 2364 2365 /* deselect device - similar to stop_(in/out)put_stream() */ 2366 /* 1. Get and set stream specific mixer controls */ 2367 retval = disable_audio_route(adev, &uc_info); 2368 /* 2. Disable the rx device */ 2369 retval = disable_snd_device(adev, 2370 dir ? uc_info.in_snd_device : uc_info.out_snd_device); 2371 list_remove(&uc_info.list); 2372 } 2373 } 2374 adev->active_input = NULL; /* restore adev state */ 2375 return 0; 2376} 2377 2378static int adev_close(hw_device_t *device) 2379{ 2380 size_t i; 2381 struct audio_device *adev = (struct audio_device *)device; 2382 audio_route_free(adev->audio_route); 2383 free(adev->snd_dev_ref_cnt); 2384 platform_deinit(adev->platform); 2385 for (i = 0; i < ARRAY_SIZE(adev->use_case_table); ++i) { 2386 pcm_params_free(adev->use_case_table[i]); 2387 } 2388 free(device); 2389 return 0; 2390} 2391 2392/* This returns 1 if the input parameter looks at all plausible as a low latency period size, 2393 * or 0 otherwise. A return value of 1 doesn't mean the value is guaranteed to work, 2394 * just that it _might_ work. 2395 */ 2396static int period_size_is_plausible_for_low_latency(int period_size) 2397{ 2398 switch (period_size) { 2399 case 160: 2400 case 240: 2401 case 320: 2402 case 480: 2403 return 1; 2404 default: 2405 return 0; 2406 } 2407} 2408 2409static int adev_open(const hw_module_t *module, const char *name, 2410 hw_device_t **device) 2411{ 2412 struct audio_device *adev; 2413 int i, ret; 2414 2415 ALOGD("%s: enter", __func__); 2416 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL; 2417 2418 adev = calloc(1, sizeof(struct audio_device)); 2419 2420 pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL); 2421 2422 adev->device.common.tag = HARDWARE_DEVICE_TAG; 2423 adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 2424 adev->device.common.module = (struct hw_module_t *)module; 2425 adev->device.common.close = adev_close; 2426 2427 adev->device.init_check = adev_init_check; 2428 adev->device.set_voice_volume = adev_set_voice_volume; 2429 adev->device.set_master_volume = adev_set_master_volume; 2430 adev->device.get_master_volume = adev_get_master_volume; 2431 adev->device.set_master_mute = adev_set_master_mute; 2432 adev->device.get_master_mute = adev_get_master_mute; 2433 adev->device.set_mode = adev_set_mode; 2434 adev->device.set_mic_mute = adev_set_mic_mute; 2435 adev->device.get_mic_mute = adev_get_mic_mute; 2436 adev->device.set_parameters = adev_set_parameters; 2437 adev->device.get_parameters = adev_get_parameters; 2438 adev->device.get_input_buffer_size = adev_get_input_buffer_size; 2439 adev->device.open_output_stream = adev_open_output_stream; 2440 adev->device.close_output_stream = adev_close_output_stream; 2441 adev->device.open_input_stream = adev_open_input_stream; 2442 adev->device.close_input_stream = adev_close_input_stream; 2443 adev->device.dump = adev_dump; 2444 2445 /* Set the default route before the PCM stream is opened */ 2446 pthread_mutex_lock(&adev->lock); 2447 adev->mode = AUDIO_MODE_NORMAL; 2448 adev->active_input = NULL; 2449 adev->primary_output = NULL; 2450 adev->bluetooth_nrec = true; 2451 adev->acdb_settings = TTY_MODE_OFF; 2452 /* adev->cur_hdmi_channels = 0; by calloc() */ 2453 adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int)); 2454 voice_init(adev); 2455 list_init(&adev->usecase_list); 2456 pthread_mutex_unlock(&adev->lock); 2457 2458 /* Loads platform specific libraries dynamically */ 2459 adev->platform = platform_init(adev); 2460 if (!adev->platform) { 2461 free(adev->snd_dev_ref_cnt); 2462 free(adev); 2463 ALOGE("%s: Failed to init platform data, aborting.", __func__); 2464 *device = NULL; 2465 return -EINVAL; 2466 } 2467 2468 if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) { 2469 adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW); 2470 if (adev->visualizer_lib == NULL) { 2471 ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH); 2472 } else { 2473 ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH); 2474 adev->visualizer_start_output = 2475 (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib, 2476 "visualizer_hal_start_output"); 2477 adev->visualizer_stop_output = 2478 (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib, 2479 "visualizer_hal_stop_output"); 2480 } 2481 } 2482 2483 adev->bt_wb_speech_enabled = false; 2484 2485 *device = &adev->device.common; 2486 if (k_enable_extended_precision) 2487 adev_verify_devices(adev); 2488 2489 char value[PROPERTY_VALUE_MAX]; 2490 int trial; 2491 if (property_get("audio_hal.period_size", value, NULL) > 0) { 2492 trial = atoi(value); 2493 if (period_size_is_plausible_for_low_latency(trial)) { 2494 pcm_config_low_latency.period_size = trial; 2495 pcm_config_low_latency.start_threshold = trial / 4; 2496 pcm_config_low_latency.avail_min = trial / 4; 2497 configured_low_latency_capture_period_size = trial; 2498 } 2499 } 2500 if (property_get("audio_hal.in_period_size", value, NULL) > 0) { 2501 trial = atoi(value); 2502 if (period_size_is_plausible_for_low_latency(trial)) { 2503 configured_low_latency_capture_period_size = trial; 2504 } 2505 } 2506 2507 ALOGV("%s: exit", __func__); 2508 return 0; 2509} 2510 2511static struct hw_module_methods_t hal_module_methods = { 2512 .open = adev_open, 2513}; 2514 2515struct audio_module HAL_MODULE_INFO_SYM = { 2516 .common = { 2517 .tag = HARDWARE_MODULE_TAG, 2518 .module_api_version = AUDIO_MODULE_API_VERSION_0_1, 2519 .hal_api_version = HARDWARE_HAL_API_VERSION, 2520 .id = AUDIO_HARDWARE_MODULE_ID, 2521 .name = "QCOM Audio HAL", 2522 .author = "Code Aurora Forum", 2523 .methods = &hal_module_methods, 2524 }, 2525}; 2526