audio_hw.c revision fdf296a35ba3deb8490522c834037e5e977b05cf
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 * 1078 audio_stream_out_frame_size((const struct audio_stream_out *)stream); 1079} 1080 1081static uint32_t out_get_channels(const struct audio_stream *stream) 1082{ 1083 struct stream_out *out = (struct stream_out *)stream; 1084 1085 return out->channel_mask; 1086} 1087 1088static audio_format_t out_get_format(const struct audio_stream *stream) 1089{ 1090 struct stream_out *out = (struct stream_out *)stream; 1091 1092 return out->format; 1093} 1094 1095static int out_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) 1096{ 1097 return -ENOSYS; 1098} 1099 1100static int out_standby(struct audio_stream *stream) 1101{ 1102 struct stream_out *out = (struct stream_out *)stream; 1103 struct audio_device *adev = out->dev; 1104 1105 ALOGV("%s: enter: usecase(%d: %s)", __func__, 1106 out->usecase, use_case_table[out->usecase]); 1107 1108 pthread_mutex_lock(&out->lock); 1109 if (!out->standby) { 1110 pthread_mutex_lock(&adev->lock); 1111 out->standby = true; 1112 if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1113 if (out->pcm) { 1114 pcm_close(out->pcm); 1115 out->pcm = NULL; 1116 } 1117 } else { 1118 stop_compressed_output_l(out); 1119 out->gapless_mdata.encoder_delay = 0; 1120 out->gapless_mdata.encoder_padding = 0; 1121 if (out->compr != NULL) { 1122 compress_close(out->compr); 1123 out->compr = NULL; 1124 } 1125 } 1126 stop_output_stream(out); 1127 pthread_mutex_unlock(&adev->lock); 1128 } 1129 pthread_mutex_unlock(&out->lock); 1130 ALOGV("%s: exit", __func__); 1131 return 0; 1132} 1133 1134static int out_dump(const struct audio_stream *stream __unused, int fd __unused) 1135{ 1136 return 0; 1137} 1138 1139static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms) 1140{ 1141 int ret = 0; 1142 char value[32]; 1143 struct compr_gapless_mdata tmp_mdata; 1144 1145 if (!out || !parms) { 1146 return -EINVAL; 1147 } 1148 1149 ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value)); 1150 if (ret >= 0) { 1151 tmp_mdata.encoder_delay = atoi(value); //whats a good limit check? 1152 } else { 1153 return -EINVAL; 1154 } 1155 1156 ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value)); 1157 if (ret >= 0) { 1158 tmp_mdata.encoder_padding = atoi(value); 1159 } else { 1160 return -EINVAL; 1161 } 1162 1163 out->gapless_mdata = tmp_mdata; 1164 out->send_new_metadata = 1; 1165 ALOGV("%s new encoder delay %u and padding %u", __func__, 1166 out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding); 1167 1168 return 0; 1169} 1170 1171 1172static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) 1173{ 1174 struct stream_out *out = (struct stream_out *)stream; 1175 struct audio_device *adev = out->dev; 1176 struct audio_usecase *usecase; 1177 struct listnode *node; 1178 struct str_parms *parms; 1179 char value[32]; 1180 int ret, val = 0; 1181 bool select_new_device = false; 1182 int status = 0; 1183 1184 ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s", 1185 __func__, out->usecase, use_case_table[out->usecase], kvpairs); 1186 parms = str_parms_create_str(kvpairs); 1187 ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); 1188 if (ret >= 0) { 1189 val = atoi(value); 1190 pthread_mutex_lock(&out->lock); 1191 pthread_mutex_lock(&adev->lock); 1192 1193 /* 1194 * When HDMI cable is unplugged the music playback is paused and 1195 * the policy manager sends routing=0. But the audioflinger 1196 * continues to write data until standby time (3sec). 1197 * As the HDMI core is turned off, the write gets blocked. 1198 * Avoid this by routing audio to speaker until standby. 1199 */ 1200 if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL && 1201 val == AUDIO_DEVICE_NONE) { 1202 val = AUDIO_DEVICE_OUT_SPEAKER; 1203 } 1204 1205 /* 1206 * select_devices() call below switches all the usecases on the same 1207 * backend to the new device. Refer to check_usecases_codec_backend() in 1208 * the select_devices(). But how do we undo this? 1209 * 1210 * For example, music playback is active on headset (deep-buffer usecase) 1211 * and if we go to ringtones and select a ringtone, low-latency usecase 1212 * will be started on headset+speaker. As we can't enable headset+speaker 1213 * and headset devices at the same time, select_devices() switches the music 1214 * playback to headset+speaker while starting low-lateny usecase for ringtone. 1215 * So when the ringtone playback is completed, how do we undo the same? 1216 * 1217 * We are relying on the out_set_parameters() call on deep-buffer output, 1218 * once the ringtone playback is ended. 1219 * NOTE: We should not check if the current devices are same as new devices. 1220 * Because select_devices() must be called to switch back the music 1221 * playback to headset. 1222 */ 1223 if (val != 0) { 1224 out->devices = val; 1225 1226 if (!out->standby) 1227 select_devices(adev, out->usecase); 1228 1229 if ((adev->mode == AUDIO_MODE_IN_CALL) && 1230 !voice_is_in_call(adev) && 1231 (out == adev->primary_output)) { 1232 ret = voice_start_call(adev); 1233 } else if ((adev->mode == AUDIO_MODE_IN_CALL) && 1234 voice_is_in_call(adev) && 1235 (out == adev->primary_output)) { 1236 voice_update_devices_for_all_voice_usecases(adev); 1237 } 1238 } 1239 1240 if ((adev->mode == AUDIO_MODE_NORMAL) && 1241 voice_is_in_call(adev) && 1242 (out == adev->primary_output)) { 1243 ret = voice_stop_call(adev); 1244 } 1245 1246 pthread_mutex_unlock(&adev->lock); 1247 pthread_mutex_unlock(&out->lock); 1248 } 1249 1250 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1251 parse_compress_metadata(out, parms); 1252 } 1253 1254 str_parms_destroy(parms); 1255 ALOGV("%s: exit: code(%d)", __func__, status); 1256 return status; 1257} 1258 1259static char* out_get_parameters(const struct audio_stream *stream, const char *keys) 1260{ 1261 struct stream_out *out = (struct stream_out *)stream; 1262 struct str_parms *query = str_parms_create_str(keys); 1263 char *str; 1264 char value[256]; 1265 struct str_parms *reply = str_parms_create(); 1266 size_t i, j; 1267 int ret; 1268 bool first = true; 1269 ALOGV("%s: enter: keys - %s", __func__, keys); 1270 ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value)); 1271 if (ret >= 0) { 1272 value[0] = '\0'; 1273 i = 0; 1274 while (out->supported_channel_masks[i] != 0) { 1275 for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) { 1276 if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) { 1277 if (!first) { 1278 strcat(value, "|"); 1279 } 1280 strcat(value, out_channels_name_to_enum_table[j].name); 1281 first = false; 1282 break; 1283 } 1284 } 1285 i++; 1286 } 1287 str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value); 1288 str = str_parms_to_str(reply); 1289 } else { 1290 str = strdup(keys); 1291 } 1292 str_parms_destroy(query); 1293 str_parms_destroy(reply); 1294 ALOGV("%s: exit: returns - %s", __func__, str); 1295 return str; 1296} 1297 1298static uint32_t out_get_latency(const struct audio_stream_out *stream) 1299{ 1300 struct stream_out *out = (struct stream_out *)stream; 1301 1302 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) 1303 return COMPRESS_OFFLOAD_PLAYBACK_LATENCY; 1304 1305 return (out->config.period_count * out->config.period_size * 1000) / 1306 (out->config.rate); 1307} 1308 1309static int out_set_volume(struct audio_stream_out *stream, float left, 1310 float right) 1311{ 1312 struct stream_out *out = (struct stream_out *)stream; 1313 int volume[2]; 1314 1315 if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) { 1316 /* only take left channel into account: the API is for stereo anyway */ 1317 out->muted = (left == 0.0f); 1318 return 0; 1319 } else if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1320 const char *mixer_ctl_name = "Compress Playback Volume"; 1321 struct audio_device *adev = out->dev; 1322 struct mixer_ctl *ctl; 1323 1324 ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); 1325 if (!ctl) { 1326 /* try with the control based on device id */ 1327 int pcm_device_id = platform_get_pcm_device_id(out->usecase, 1328 PCM_PLAYBACK); 1329 char ctl_name[128] = {0}; 1330 snprintf(ctl_name, sizeof(ctl_name), 1331 "Compress Playback %d Volume", pcm_device_id); 1332 ctl = mixer_get_ctl_by_name(adev->mixer, ctl_name); 1333 if (!ctl) { 1334 ALOGE("%s: Could not get volume ctl mixer cmd", __func__); 1335 return -EINVAL; 1336 } 1337 } 1338 volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX); 1339 volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX); 1340 mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0])); 1341 return 0; 1342 } 1343 1344 return -ENOSYS; 1345} 1346 1347static ssize_t out_write(struct audio_stream_out *stream, const void *buffer, 1348 size_t bytes) 1349{ 1350 struct stream_out *out = (struct stream_out *)stream; 1351 struct audio_device *adev = out->dev; 1352 ssize_t ret = 0; 1353 1354 pthread_mutex_lock(&out->lock); 1355 if (out->standby) { 1356 out->standby = false; 1357 pthread_mutex_lock(&adev->lock); 1358 ret = start_output_stream(out); 1359 pthread_mutex_unlock(&adev->lock); 1360 /* ToDo: If use case is compress offload should return 0 */ 1361 if (ret != 0) { 1362 out->standby = true; 1363 goto exit; 1364 } 1365 } 1366 1367 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1368 ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes); 1369 if (out->send_new_metadata) { 1370 ALOGVV("send new gapless metadata"); 1371 compress_set_gapless_metadata(out->compr, &out->gapless_mdata); 1372 out->send_new_metadata = 0; 1373 } 1374 1375 ret = compress_write(out->compr, buffer, bytes); 1376 ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret); 1377 if (ret >= 0 && ret < (ssize_t)bytes) { 1378 send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); 1379 } 1380 if (!out->playback_started) { 1381 compress_start(out->compr); 1382 out->playback_started = 1; 1383 out->offload_state = OFFLOAD_STATE_PLAYING; 1384 } 1385 pthread_mutex_unlock(&out->lock); 1386 return ret; 1387 } else { 1388 if (out->pcm) { 1389 if (out->muted) 1390 memset((void *)buffer, 0, bytes); 1391 ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes); 1392 ret = pcm_write(out->pcm, (void *)buffer, bytes); 1393 if (ret == 0) 1394 out->written += bytes / (out->config.channels * sizeof(short)); 1395 } 1396 } 1397 1398exit: 1399 pthread_mutex_unlock(&out->lock); 1400 1401 if (ret != 0) { 1402 if (out->pcm) 1403 ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(out->pcm)); 1404 out_standby(&out->stream.common); 1405 usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) / 1406 out_get_sample_rate(&out->stream.common)); 1407 } 1408 return bytes; 1409} 1410 1411static int out_get_render_position(const struct audio_stream_out *stream, 1412 uint32_t *dsp_frames) 1413{ 1414 struct stream_out *out = (struct stream_out *)stream; 1415 *dsp_frames = 0; 1416 if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) { 1417 pthread_mutex_lock(&out->lock); 1418 if (out->compr != NULL) { 1419 compress_get_tstamp(out->compr, (unsigned long *)dsp_frames, 1420 &out->sample_rate); 1421 ALOGVV("%s rendered frames %d sample_rate %d", 1422 __func__, *dsp_frames, out->sample_rate); 1423 } 1424 pthread_mutex_unlock(&out->lock); 1425 return 0; 1426 } else 1427 return -EINVAL; 1428} 1429 1430static int out_add_audio_effect(const struct audio_stream *stream __unused, 1431 effect_handle_t effect __unused) 1432{ 1433 return 0; 1434} 1435 1436static int out_remove_audio_effect(const struct audio_stream *stream __unused, 1437 effect_handle_t effect __unused) 1438{ 1439 return 0; 1440} 1441 1442static int out_get_next_write_timestamp(const struct audio_stream_out *stream __unused, 1443 int64_t *timestamp __unused) 1444{ 1445 return -EINVAL; 1446} 1447 1448static int out_get_presentation_position(const struct audio_stream_out *stream, 1449 uint64_t *frames, struct timespec *timestamp) 1450{ 1451 struct stream_out *out = (struct stream_out *)stream; 1452 int ret = -1; 1453 unsigned long dsp_frames; 1454 1455 pthread_mutex_lock(&out->lock); 1456 1457 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1458 if (out->compr != NULL) { 1459 compress_get_tstamp(out->compr, &dsp_frames, 1460 &out->sample_rate); 1461 ALOGVV("%s rendered frames %ld sample_rate %d", 1462 __func__, dsp_frames, out->sample_rate); 1463 *frames = dsp_frames; 1464 ret = 0; 1465 /* this is the best we can do */ 1466 clock_gettime(CLOCK_MONOTONIC, timestamp); 1467 } 1468 } else { 1469 if (out->pcm) { 1470 size_t avail; 1471 if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) { 1472 size_t kernel_buffer_size = out->config.period_size * out->config.period_count; 1473 int64_t signed_frames = out->written - kernel_buffer_size + avail; 1474 // This adjustment accounts for buffering after app processor. 1475 // It is based on estimated DSP latency per use case, rather than exact. 1476 signed_frames -= 1477 (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL); 1478 1479 // It would be unusual for this value to be negative, but check just in case ... 1480 if (signed_frames >= 0) { 1481 *frames = signed_frames; 1482 ret = 0; 1483 } 1484 } 1485 } 1486 } 1487 1488 pthread_mutex_unlock(&out->lock); 1489 1490 return ret; 1491} 1492 1493static int out_set_callback(struct audio_stream_out *stream, 1494 stream_callback_t callback, void *cookie) 1495{ 1496 struct stream_out *out = (struct stream_out *)stream; 1497 1498 ALOGV("%s", __func__); 1499 pthread_mutex_lock(&out->lock); 1500 out->offload_callback = callback; 1501 out->offload_cookie = cookie; 1502 pthread_mutex_unlock(&out->lock); 1503 return 0; 1504} 1505 1506static int out_pause(struct audio_stream_out* stream) 1507{ 1508 struct stream_out *out = (struct stream_out *)stream; 1509 int status = -ENOSYS; 1510 ALOGV("%s", __func__); 1511 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1512 pthread_mutex_lock(&out->lock); 1513 if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) { 1514 status = compress_pause(out->compr); 1515 out->offload_state = OFFLOAD_STATE_PAUSED; 1516 } 1517 pthread_mutex_unlock(&out->lock); 1518 } 1519 return status; 1520} 1521 1522static int out_resume(struct audio_stream_out* stream) 1523{ 1524 struct stream_out *out = (struct stream_out *)stream; 1525 int status = -ENOSYS; 1526 ALOGV("%s", __func__); 1527 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1528 status = 0; 1529 pthread_mutex_lock(&out->lock); 1530 if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) { 1531 status = compress_resume(out->compr); 1532 out->offload_state = OFFLOAD_STATE_PLAYING; 1533 } 1534 pthread_mutex_unlock(&out->lock); 1535 } 1536 return status; 1537} 1538 1539static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type ) 1540{ 1541 struct stream_out *out = (struct stream_out *)stream; 1542 int status = -ENOSYS; 1543 ALOGV("%s", __func__); 1544 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1545 pthread_mutex_lock(&out->lock); 1546 if (type == AUDIO_DRAIN_EARLY_NOTIFY) 1547 status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN); 1548 else 1549 status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN); 1550 pthread_mutex_unlock(&out->lock); 1551 } 1552 return status; 1553} 1554 1555static int out_flush(struct audio_stream_out* stream) 1556{ 1557 struct stream_out *out = (struct stream_out *)stream; 1558 ALOGV("%s", __func__); 1559 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1560 pthread_mutex_lock(&out->lock); 1561 stop_compressed_output_l(out); 1562 pthread_mutex_unlock(&out->lock); 1563 return 0; 1564 } 1565 return -ENOSYS; 1566} 1567 1568/** audio_stream_in implementation **/ 1569static uint32_t in_get_sample_rate(const struct audio_stream *stream) 1570{ 1571 struct stream_in *in = (struct stream_in *)stream; 1572 1573 return in->config.rate; 1574} 1575 1576static int in_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused) 1577{ 1578 return -ENOSYS; 1579} 1580 1581static size_t in_get_buffer_size(const struct audio_stream *stream) 1582{ 1583 struct stream_in *in = (struct stream_in *)stream; 1584 1585 return in->config.period_size * 1586 audio_stream_in_frame_size((const struct audio_stream_in *)stream); 1587} 1588 1589static uint32_t in_get_channels(const struct audio_stream *stream) 1590{ 1591 struct stream_in *in = (struct stream_in *)stream; 1592 1593 return in->channel_mask; 1594} 1595 1596static audio_format_t in_get_format(const struct audio_stream *stream __unused) 1597{ 1598 return AUDIO_FORMAT_PCM_16_BIT; 1599} 1600 1601static int in_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) 1602{ 1603 return -ENOSYS; 1604} 1605 1606static int in_standby(struct audio_stream *stream) 1607{ 1608 struct stream_in *in = (struct stream_in *)stream; 1609 struct audio_device *adev = in->dev; 1610 int status = 0; 1611 ALOGV("%s: enter", __func__); 1612 pthread_mutex_lock(&in->lock); 1613 if (!in->standby) { 1614 pthread_mutex_lock(&adev->lock); 1615 in->standby = true; 1616 if (in->pcm) { 1617 pcm_close(in->pcm); 1618 in->pcm = NULL; 1619 } 1620 status = stop_input_stream(in); 1621 pthread_mutex_unlock(&adev->lock); 1622 } 1623 pthread_mutex_unlock(&in->lock); 1624 ALOGV("%s: exit: status(%d)", __func__, status); 1625 return status; 1626} 1627 1628static int in_dump(const struct audio_stream *stream __unused, int fd __unused) 1629{ 1630 return 0; 1631} 1632 1633static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) 1634{ 1635 struct stream_in *in = (struct stream_in *)stream; 1636 struct audio_device *adev = in->dev; 1637 struct str_parms *parms; 1638 char *str; 1639 char value[32]; 1640 int ret, val = 0; 1641 int status = 0; 1642 1643 ALOGV("%s: enter: kvpairs=%s", __func__, kvpairs); 1644 parms = str_parms_create_str(kvpairs); 1645 1646 ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value)); 1647 1648 pthread_mutex_lock(&in->lock); 1649 pthread_mutex_lock(&adev->lock); 1650 if (ret >= 0) { 1651 val = atoi(value); 1652 /* no audio source uses val == 0 */ 1653 if ((in->source != val) && (val != 0)) { 1654 in->source = val; 1655 } 1656 } 1657 1658 ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); 1659 1660 if (ret >= 0) { 1661 val = atoi(value); 1662 if ((in->device != val) && (val != 0)) { 1663 in->device = val; 1664 /* If recording is in progress, change the tx device to new device */ 1665 if (!in->standby) 1666 status = select_devices(adev, in->usecase); 1667 } 1668 } 1669 1670 pthread_mutex_unlock(&adev->lock); 1671 pthread_mutex_unlock(&in->lock); 1672 1673 str_parms_destroy(parms); 1674 ALOGV("%s: exit: status(%d)", __func__, status); 1675 return status; 1676} 1677 1678static char* in_get_parameters(const struct audio_stream *stream __unused, 1679 const char *keys __unused) 1680{ 1681 return strdup(""); 1682} 1683 1684static int in_set_gain(struct audio_stream_in *stream __unused, float gain __unused) 1685{ 1686 return 0; 1687} 1688 1689static ssize_t in_read(struct audio_stream_in *stream, void *buffer, 1690 size_t bytes) 1691{ 1692 struct stream_in *in = (struct stream_in *)stream; 1693 struct audio_device *adev = in->dev; 1694 int i, ret = -1; 1695 1696 pthread_mutex_lock(&in->lock); 1697 if (in->standby) { 1698 pthread_mutex_lock(&adev->lock); 1699 ret = start_input_stream(in); 1700 pthread_mutex_unlock(&adev->lock); 1701 if (ret != 0) { 1702 goto exit; 1703 } 1704 in->standby = 0; 1705 } 1706 1707 if (in->pcm) { 1708 ret = pcm_read(in->pcm, buffer, bytes); 1709 } 1710 1711 /* 1712 * Instead of writing zeroes here, we could trust the hardware 1713 * to always provide zeroes when muted. 1714 */ 1715 if (ret == 0 && voice_get_mic_mute(adev) && !voice_is_in_call(adev)) 1716 memset(buffer, 0, bytes); 1717 1718exit: 1719 pthread_mutex_unlock(&in->lock); 1720 1721 if (ret != 0) { 1722 in_standby(&in->stream.common); 1723 ALOGV("%s: read failed - sleeping for buffer duration", __func__); 1724 usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) / 1725 in_get_sample_rate(&in->stream.common)); 1726 } 1727 return bytes; 1728} 1729 1730static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused) 1731{ 1732 return 0; 1733} 1734 1735static int add_remove_audio_effect(const struct audio_stream *stream, 1736 effect_handle_t effect, 1737 bool enable) 1738{ 1739 struct stream_in *in = (struct stream_in *)stream; 1740 int status = 0; 1741 effect_descriptor_t desc; 1742 1743 status = (*effect)->get_descriptor(effect, &desc); 1744 if (status != 0) 1745 return status; 1746 1747 pthread_mutex_lock(&in->lock); 1748 pthread_mutex_lock(&in->dev->lock); 1749 if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) && 1750 in->enable_aec != enable && 1751 (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) { 1752 in->enable_aec = enable; 1753 if (!in->standby) 1754 select_devices(in->dev, in->usecase); 1755 } 1756 pthread_mutex_unlock(&in->dev->lock); 1757 pthread_mutex_unlock(&in->lock); 1758 1759 return 0; 1760} 1761 1762static int in_add_audio_effect(const struct audio_stream *stream, 1763 effect_handle_t effect) 1764{ 1765 ALOGV("%s: effect %p", __func__, effect); 1766 return add_remove_audio_effect(stream, effect, true); 1767} 1768 1769static int in_remove_audio_effect(const struct audio_stream *stream, 1770 effect_handle_t effect) 1771{ 1772 ALOGV("%s: effect %p", __func__, effect); 1773 return add_remove_audio_effect(stream, effect, false); 1774} 1775 1776static int adev_open_output_stream(struct audio_hw_device *dev, 1777 audio_io_handle_t handle, 1778 audio_devices_t devices, 1779 audio_output_flags_t flags, 1780 struct audio_config *config, 1781 struct audio_stream_out **stream_out) 1782{ 1783 struct audio_device *adev = (struct audio_device *)dev; 1784 struct stream_out *out; 1785 int i, ret; 1786 1787 ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)", 1788 __func__, config->sample_rate, config->channel_mask, devices, flags); 1789 *stream_out = NULL; 1790 out = (struct stream_out *)calloc(1, sizeof(struct stream_out)); 1791 1792 if (devices == AUDIO_DEVICE_NONE) 1793 devices = AUDIO_DEVICE_OUT_SPEAKER; 1794 1795 out->flags = flags; 1796 out->devices = devices; 1797 out->dev = adev; 1798 out->format = config->format; 1799 out->sample_rate = config->sample_rate; 1800 out->channel_mask = AUDIO_CHANNEL_OUT_STEREO; 1801 out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO; 1802 out->handle = handle; 1803 1804 /* Init use case and pcm_config */ 1805 if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT && 1806 !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && 1807 out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { 1808 pthread_mutex_lock(&adev->lock); 1809 ret = read_hdmi_channel_masks(out); 1810 pthread_mutex_unlock(&adev->lock); 1811 if (ret != 0) 1812 goto error_open; 1813 1814 if (config->sample_rate == 0) 1815 config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE; 1816 if (config->channel_mask == 0) 1817 config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1; 1818 1819 out->channel_mask = config->channel_mask; 1820 out->sample_rate = config->sample_rate; 1821 out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH; 1822 out->config = pcm_config_hdmi_multi; 1823 out->config.rate = config->sample_rate; 1824 out->config.channels = audio_channel_count_from_out_mask(out->channel_mask); 1825 out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2); 1826 } else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1827 if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version || 1828 config->offload_info.size != AUDIO_INFO_INITIALIZER.size) { 1829 ALOGE("%s: Unsupported Offload information", __func__); 1830 ret = -EINVAL; 1831 goto error_open; 1832 } 1833 if (!is_supported_format(config->offload_info.format)) { 1834 ALOGE("%s: Unsupported audio format", __func__); 1835 ret = -EINVAL; 1836 goto error_open; 1837 } 1838 1839 out->compr_config.codec = (struct snd_codec *) 1840 calloc(1, sizeof(struct snd_codec)); 1841 1842 out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD; 1843 if (config->offload_info.channel_mask) 1844 out->channel_mask = config->offload_info.channel_mask; 1845 else if (config->channel_mask) 1846 out->channel_mask = config->channel_mask; 1847 out->format = config->offload_info.format; 1848 out->sample_rate = config->offload_info.sample_rate; 1849 1850 out->stream.set_callback = out_set_callback; 1851 out->stream.pause = out_pause; 1852 out->stream.resume = out_resume; 1853 out->stream.drain = out_drain; 1854 out->stream.flush = out_flush; 1855 1856 out->compr_config.codec->id = 1857 get_snd_codec_id(config->offload_info.format); 1858 out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; 1859 out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS; 1860 out->compr_config.codec->sample_rate = 1861 compress_get_alsa_rate(config->offload_info.sample_rate); 1862 out->compr_config.codec->bit_rate = 1863 config->offload_info.bit_rate; 1864 out->compr_config.codec->ch_in = 1865 audio_channel_count_from_out_mask(config->channel_mask); 1866 out->compr_config.codec->ch_out = out->compr_config.codec->ch_in; 1867 1868 if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) 1869 out->non_blocking = 1; 1870 1871 out->send_new_metadata = 1; 1872 create_offload_callback_thread(out); 1873 ALOGV("%s: offloaded output offload_info version %04x bit rate %d", 1874 __func__, config->offload_info.version, 1875 config->offload_info.bit_rate); 1876 } else { 1877 if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) { 1878 out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER; 1879 out->config = pcm_config_deep_buffer; 1880 } else { 1881 out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY; 1882 out->config = pcm_config_low_latency; 1883 } 1884 if (config->format != audio_format_from_pcm_format(out->config.format)) { 1885 if (k_enable_extended_precision 1886 && pcm_params_format_test(adev->use_case_table[out->usecase], 1887 pcm_format_from_audio_format(config->format))) { 1888 out->config.format = pcm_format_from_audio_format(config->format); 1889 /* out->format already set to config->format */ 1890 } else { 1891 /* deny the externally proposed config format 1892 * and use the one specified in audio_hw layer configuration. 1893 * Note: out->format is returned by out->stream.common.get_format() 1894 * and is used to set config->format in the code several lines below. 1895 */ 1896 out->format = audio_format_from_pcm_format(out->config.format); 1897 } 1898 } 1899 out->sample_rate = out->config.rate; 1900 } 1901 ALOGV("%s: Usecase(%s) config->format %#x out->config.format %#x\n", 1902 __func__, use_case_table[out->usecase], config->format, out->config.format); 1903 1904 if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) { 1905 if(adev->primary_output == NULL) 1906 adev->primary_output = out; 1907 else { 1908 ALOGE("%s: Primary output is already opened", __func__); 1909 ret = -EEXIST; 1910 goto error_open; 1911 } 1912 } 1913 1914 /* Check if this usecase is already existing */ 1915 pthread_mutex_lock(&adev->lock); 1916 if (get_usecase_from_list(adev, out->usecase) != NULL) { 1917 ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase); 1918 pthread_mutex_unlock(&adev->lock); 1919 ret = -EEXIST; 1920 goto error_open; 1921 } 1922 pthread_mutex_unlock(&adev->lock); 1923 1924 out->stream.common.get_sample_rate = out_get_sample_rate; 1925 out->stream.common.set_sample_rate = out_set_sample_rate; 1926 out->stream.common.get_buffer_size = out_get_buffer_size; 1927 out->stream.common.get_channels = out_get_channels; 1928 out->stream.common.get_format = out_get_format; 1929 out->stream.common.set_format = out_set_format; 1930 out->stream.common.standby = out_standby; 1931 out->stream.common.dump = out_dump; 1932 out->stream.common.set_parameters = out_set_parameters; 1933 out->stream.common.get_parameters = out_get_parameters; 1934 out->stream.common.add_audio_effect = out_add_audio_effect; 1935 out->stream.common.remove_audio_effect = out_remove_audio_effect; 1936 out->stream.get_latency = out_get_latency; 1937 out->stream.set_volume = out_set_volume; 1938 out->stream.write = out_write; 1939 out->stream.get_render_position = out_get_render_position; 1940 out->stream.get_next_write_timestamp = out_get_next_write_timestamp; 1941 out->stream.get_presentation_position = out_get_presentation_position; 1942 1943 out->standby = 1; 1944 /* out->muted = false; by calloc() */ 1945 /* out->written = 0; by calloc() */ 1946 1947 pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL); 1948 pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL); 1949 1950 config->format = out->stream.common.get_format(&out->stream.common); 1951 config->channel_mask = out->stream.common.get_channels(&out->stream.common); 1952 config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common); 1953 1954 *stream_out = &out->stream; 1955 ALOGV("%s: exit", __func__); 1956 return 0; 1957 1958error_open: 1959 free(out); 1960 *stream_out = NULL; 1961 ALOGD("%s: exit: ret %d", __func__, ret); 1962 return ret; 1963} 1964 1965static void adev_close_output_stream(struct audio_hw_device *dev __unused, 1966 struct audio_stream_out *stream) 1967{ 1968 struct stream_out *out = (struct stream_out *)stream; 1969 struct audio_device *adev = out->dev; 1970 1971 ALOGV("%s: enter", __func__); 1972 out_standby(&stream->common); 1973 if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { 1974 destroy_offload_callback_thread(out); 1975 1976 if (out->compr_config.codec != NULL) 1977 free(out->compr_config.codec); 1978 } 1979 pthread_cond_destroy(&out->cond); 1980 pthread_mutex_destroy(&out->lock); 1981 free(stream); 1982 ALOGV("%s: exit", __func__); 1983} 1984 1985static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) 1986{ 1987 struct audio_device *adev = (struct audio_device *)dev; 1988 struct str_parms *parms; 1989 char *str; 1990 char value[32]; 1991 int val; 1992 int ret; 1993 int status = 0; 1994 1995 ALOGD("%s: enter: %s", __func__, kvpairs); 1996 1997 pthread_mutex_lock(&adev->lock); 1998 1999 parms = str_parms_create_str(kvpairs); 2000 status = voice_set_parameters(adev, parms); 2001 if (status != 0) { 2002 goto done; 2003 } 2004 2005 ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value)); 2006 if (ret >= 0) { 2007 /* When set to false, HAL should disable EC and NS 2008 * But it is currently not supported. 2009 */ 2010 if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) 2011 adev->bluetooth_nrec = true; 2012 else 2013 adev->bluetooth_nrec = false; 2014 } 2015 2016 ret = str_parms_get_str(parms, "screen_state", value, sizeof(value)); 2017 if (ret >= 0) { 2018 if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) 2019 adev->screen_off = false; 2020 else 2021 adev->screen_off = true; 2022 } 2023 2024 ret = str_parms_get_int(parms, "rotation", &val); 2025 if (ret >= 0) { 2026 bool reverse_speakers = false; 2027 switch(val) { 2028 // FIXME: note that the code below assumes that the speakers are in the correct placement 2029 // relative to the user when the device is rotated 90deg from its default rotation. This 2030 // assumption is device-specific, not platform-specific like this code. 2031 case 270: 2032 reverse_speakers = true; 2033 break; 2034 case 0: 2035 case 90: 2036 case 180: 2037 break; 2038 default: 2039 ALOGE("%s: unexpected rotation of %d", __func__, val); 2040 status = -EINVAL; 2041 } 2042 if (status == 0) { 2043 if (adev->speaker_lr_swap != reverse_speakers) { 2044 adev->speaker_lr_swap = reverse_speakers; 2045 // only update the selected device if there is active pcm playback 2046 struct audio_usecase *usecase; 2047 struct listnode *node; 2048 list_for_each(node, &adev->usecase_list) { 2049 usecase = node_to_item(node, struct audio_usecase, list); 2050 if (usecase->type == PCM_PLAYBACK) { 2051 select_devices(adev, usecase->id); 2052 break; 2053 } 2054 } 2055 } 2056 } 2057 } 2058 2059 ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value)); 2060 if (ret >= 0) { 2061 adev->bt_wb_speech_enabled = !strcmp(value, AUDIO_PARAMETER_VALUE_ON); 2062 } 2063 2064 audio_extn_hfp_set_parameters(adev, parms); 2065done: 2066 str_parms_destroy(parms); 2067 pthread_mutex_unlock(&adev->lock); 2068 ALOGV("%s: exit with code(%d)", __func__, status); 2069 return status; 2070} 2071 2072static char* adev_get_parameters(const struct audio_hw_device *dev, 2073 const char *keys) 2074{ 2075 struct audio_device *adev = (struct audio_device *)dev; 2076 struct str_parms *reply = str_parms_create(); 2077 struct str_parms *query = str_parms_create_str(keys); 2078 char *str; 2079 2080 pthread_mutex_lock(&adev->lock); 2081 2082 voice_get_parameters(adev, query, reply); 2083 str = str_parms_to_str(reply); 2084 str_parms_destroy(query); 2085 str_parms_destroy(reply); 2086 2087 pthread_mutex_unlock(&adev->lock); 2088 ALOGV("%s: exit: returns - %s", __func__, str); 2089 return str; 2090} 2091 2092static int adev_init_check(const struct audio_hw_device *dev __unused) 2093{ 2094 return 0; 2095} 2096 2097static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) 2098{ 2099 int ret; 2100 struct audio_device *adev = (struct audio_device *)dev; 2101 2102 pthread_mutex_lock(&adev->lock); 2103 ret = voice_set_volume(adev, volume); 2104 pthread_mutex_unlock(&adev->lock); 2105 2106 return ret; 2107} 2108 2109static int adev_set_master_volume(struct audio_hw_device *dev __unused, float volume __unused) 2110{ 2111 return -ENOSYS; 2112} 2113 2114static int adev_get_master_volume(struct audio_hw_device *dev __unused, 2115 float *volume __unused) 2116{ 2117 return -ENOSYS; 2118} 2119 2120static int adev_set_master_mute(struct audio_hw_device *dev __unused, bool muted __unused) 2121{ 2122 return -ENOSYS; 2123} 2124 2125static int adev_get_master_mute(struct audio_hw_device *dev __unused, bool *muted __unused) 2126{ 2127 return -ENOSYS; 2128} 2129 2130static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) 2131{ 2132 struct audio_device *adev = (struct audio_device *)dev; 2133 2134 pthread_mutex_lock(&adev->lock); 2135 if (adev->mode != mode) { 2136 ALOGD("%s: mode %d\n", __func__, mode); 2137 adev->mode = mode; 2138 } 2139 pthread_mutex_unlock(&adev->lock); 2140 return 0; 2141} 2142 2143static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) 2144{ 2145 int ret; 2146 struct audio_device *adev = (struct audio_device *)dev; 2147 2148 ALOGD("%s: state %d\n", __func__, state); 2149 pthread_mutex_lock(&adev->lock); 2150 ret = voice_set_mic_mute(adev, state); 2151 pthread_mutex_unlock(&adev->lock); 2152 2153 return ret; 2154} 2155 2156static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) 2157{ 2158 *state = voice_get_mic_mute((struct audio_device *)dev); 2159 return 0; 2160} 2161 2162static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unused, 2163 const struct audio_config *config) 2164{ 2165 int channel_count = audio_channel_count_from_in_mask(config->channel_mask); 2166 2167 return get_input_buffer_size(config->sample_rate, config->format, channel_count); 2168} 2169 2170static int adev_open_input_stream(struct audio_hw_device *dev, 2171 audio_io_handle_t handle __unused, 2172 audio_devices_t devices, 2173 struct audio_config *config, 2174 struct audio_stream_in **stream_in) 2175{ 2176 struct audio_device *adev = (struct audio_device *)dev; 2177 struct stream_in *in; 2178 int ret = 0, buffer_size, frame_size; 2179 int channel_count = audio_channel_count_from_in_mask(config->channel_mask); 2180 2181 ALOGV("%s: enter", __func__); 2182 *stream_in = NULL; 2183 if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) 2184 return -EINVAL; 2185 2186 in = (struct stream_in *)calloc(1, sizeof(struct stream_in)); 2187 2188 pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL); 2189 2190 in->stream.common.get_sample_rate = in_get_sample_rate; 2191 in->stream.common.set_sample_rate = in_set_sample_rate; 2192 in->stream.common.get_buffer_size = in_get_buffer_size; 2193 in->stream.common.get_channels = in_get_channels; 2194 in->stream.common.get_format = in_get_format; 2195 in->stream.common.set_format = in_set_format; 2196 in->stream.common.standby = in_standby; 2197 in->stream.common.dump = in_dump; 2198 in->stream.common.set_parameters = in_set_parameters; 2199 in->stream.common.get_parameters = in_get_parameters; 2200 in->stream.common.add_audio_effect = in_add_audio_effect; 2201 in->stream.common.remove_audio_effect = in_remove_audio_effect; 2202 in->stream.set_gain = in_set_gain; 2203 in->stream.read = in_read; 2204 in->stream.get_input_frames_lost = in_get_input_frames_lost; 2205 2206 in->device = devices; 2207 in->source = AUDIO_SOURCE_DEFAULT; 2208 in->dev = adev; 2209 in->standby = 1; 2210 in->channel_mask = config->channel_mask; 2211 2212 /* Update config params with the requested sample rate and channels */ 2213 in->usecase = USECASE_AUDIO_RECORD; 2214#if LOW_LATENCY_CAPTURE_USE_CASE 2215 if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE) 2216 in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY; 2217#endif 2218 in->config = pcm_config_audio_capture; 2219 in->config.channels = channel_count; 2220 in->config.rate = config->sample_rate; 2221 2222 frame_size = audio_stream_in_frame_size(&in->stream); 2223 buffer_size = get_input_buffer_size(config->sample_rate, 2224 config->format, 2225 channel_count); 2226 in->config.period_size = buffer_size / frame_size; 2227 2228 *stream_in = &in->stream; 2229 ALOGV("%s: exit", __func__); 2230 return 0; 2231 2232err_open: 2233 free(in); 2234 *stream_in = NULL; 2235 return ret; 2236} 2237 2238static void adev_close_input_stream(struct audio_hw_device *dev __unused, 2239 struct audio_stream_in *stream) 2240{ 2241 ALOGV("%s", __func__); 2242 2243 in_standby(&stream->common); 2244 free(stream); 2245 2246 return; 2247} 2248 2249static int adev_dump(const audio_hw_device_t *device __unused, int fd __unused) 2250{ 2251 return 0; 2252} 2253 2254/* verifies input and output devices and their capabilities. 2255 * 2256 * This verification is required when enabling extended bit-depth or 2257 * sampling rates, as not all qcom products support it. 2258 * 2259 * Suitable for calling only on initialization such as adev_open(). 2260 * It fills the audio_device use_case_table[] array. 2261 * 2262 * Has a side-effect that it needs to configure audio routing / devices 2263 * in order to power up the devices and read the device parameters. 2264 * It does not acquire any hw device lock. Should restore the devices 2265 * back to "normal state" upon completion. 2266 */ 2267static int adev_verify_devices(struct audio_device *adev) 2268{ 2269 /* enumeration is a bit difficult because one really wants to pull 2270 * the use_case, device id, etc from the hidden pcm_device_table[]. 2271 * In this case there are the following use cases and device ids. 2272 * 2273 * [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {0, 0}, 2274 * [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = {15, 15}, 2275 * [USECASE_AUDIO_PLAYBACK_MULTI_CH] = {1, 1}, 2276 * [USECASE_AUDIO_PLAYBACK_OFFLOAD] = {9, 9}, 2277 * [USECASE_AUDIO_RECORD] = {0, 0}, 2278 * [USECASE_AUDIO_RECORD_LOW_LATENCY] = {15, 15}, 2279 * [USECASE_VOICE_CALL] = {2, 2}, 2280 * 2281 * USECASE_AUDIO_PLAYBACK_OFFLOAD, USECASE_AUDIO_PLAYBACK_MULTI_CH omitted. 2282 * USECASE_VOICE_CALL omitted, but possible for either input or output. 2283 */ 2284 2285 /* should be the usecases enabled in adev_open_input_stream() */ 2286 static const int test_in_usecases[] = { 2287 USECASE_AUDIO_RECORD, 2288 USECASE_AUDIO_RECORD_LOW_LATENCY, /* does not appear to be used */ 2289 }; 2290 /* should be the usecases enabled in adev_open_output_stream()*/ 2291 static const int test_out_usecases[] = { 2292 USECASE_AUDIO_PLAYBACK_DEEP_BUFFER, 2293 USECASE_AUDIO_PLAYBACK_LOW_LATENCY, 2294 }; 2295 static const usecase_type_t usecase_type_by_dir[] = { 2296 PCM_PLAYBACK, 2297 PCM_CAPTURE, 2298 }; 2299 static const unsigned flags_by_dir[] = { 2300 PCM_OUT, 2301 PCM_IN, 2302 }; 2303 2304 size_t i; 2305 unsigned dir; 2306 const unsigned card_id = adev->snd_card; 2307 char info[512]; /* for possible debug info */ 2308 2309 for (dir = 0; dir < 2; ++dir) { 2310 const usecase_type_t usecase_type = usecase_type_by_dir[dir]; 2311 const unsigned flags_dir = flags_by_dir[dir]; 2312 const size_t testsize = 2313 dir ? ARRAY_SIZE(test_in_usecases) : ARRAY_SIZE(test_out_usecases); 2314 const int *testcases = 2315 dir ? test_in_usecases : test_out_usecases; 2316 const audio_devices_t audio_device = 2317 dir ? AUDIO_DEVICE_IN_BUILTIN_MIC : AUDIO_DEVICE_OUT_SPEAKER; 2318 2319 for (i = 0; i < testsize; ++i) { 2320 const audio_usecase_t audio_usecase = testcases[i]; 2321 int device_id; 2322 snd_device_t snd_device; 2323 struct pcm_params **pparams; 2324 struct stream_out out; 2325 struct stream_in in; 2326 struct audio_usecase uc_info; 2327 int retval; 2328 2329 pparams = &adev->use_case_table[audio_usecase]; 2330 pcm_params_free(*pparams); /* can accept null input */ 2331 *pparams = NULL; 2332 2333 /* find the device ID for the use case (signed, for error) */ 2334 device_id = platform_get_pcm_device_id(audio_usecase, usecase_type); 2335 if (device_id < 0) 2336 continue; 2337 2338 /* prepare structures for device probing */ 2339 memset(&uc_info, 0, sizeof(uc_info)); 2340 uc_info.id = audio_usecase; 2341 uc_info.type = usecase_type; 2342 if (dir) { 2343 adev->active_input = ∈ 2344 memset(&in, 0, sizeof(in)); 2345 in.device = audio_device; 2346 in.source = AUDIO_SOURCE_VOICE_COMMUNICATION; 2347 uc_info.stream.in = ∈ 2348 } else { 2349 adev->active_input = NULL; 2350 } 2351 memset(&out, 0, sizeof(out)); 2352 out.devices = audio_device; /* only field needed in select_devices */ 2353 uc_info.stream.out = &out; 2354 uc_info.devices = audio_device; 2355 uc_info.in_snd_device = SND_DEVICE_NONE; 2356 uc_info.out_snd_device = SND_DEVICE_NONE; 2357 list_add_tail(&adev->usecase_list, &uc_info.list); 2358 2359 /* select device - similar to start_(in/out)put_stream() */ 2360 retval = select_devices(adev, audio_usecase); 2361 if (retval >= 0) { 2362 *pparams = pcm_params_get(card_id, device_id, flags_dir); 2363#if LOG_NDEBUG == 0 2364 if (*pparams) { 2365 ALOGV("%s: (%s) card %d device %d", __func__, 2366 dir ? "input" : "output", card_id, device_id); 2367 pcm_params_to_string(*pparams, info, ARRAY_SIZE(info)); 2368 ALOGV(info); /* print parameters */ 2369 } else { 2370 ALOGV("%s: cannot locate card %d device %d", __func__, card_id, device_id); 2371 } 2372#endif 2373 } 2374 2375 /* deselect device - similar to stop_(in/out)put_stream() */ 2376 /* 1. Get and set stream specific mixer controls */ 2377 retval = disable_audio_route(adev, &uc_info); 2378 /* 2. Disable the rx device */ 2379 retval = disable_snd_device(adev, 2380 dir ? uc_info.in_snd_device : uc_info.out_snd_device); 2381 list_remove(&uc_info.list); 2382 } 2383 } 2384 adev->active_input = NULL; /* restore adev state */ 2385 return 0; 2386} 2387 2388static int adev_close(hw_device_t *device) 2389{ 2390 size_t i; 2391 struct audio_device *adev = (struct audio_device *)device; 2392 audio_route_free(adev->audio_route); 2393 free(adev->snd_dev_ref_cnt); 2394 platform_deinit(adev->platform); 2395 for (i = 0; i < ARRAY_SIZE(adev->use_case_table); ++i) { 2396 pcm_params_free(adev->use_case_table[i]); 2397 } 2398 free(device); 2399 return 0; 2400} 2401 2402/* This returns 1 if the input parameter looks at all plausible as a low latency period size, 2403 * or 0 otherwise. A return value of 1 doesn't mean the value is guaranteed to work, 2404 * just that it _might_ work. 2405 */ 2406static int period_size_is_plausible_for_low_latency(int period_size) 2407{ 2408 switch (period_size) { 2409 case 160: 2410 case 240: 2411 case 320: 2412 case 480: 2413 return 1; 2414 default: 2415 return 0; 2416 } 2417} 2418 2419static int adev_open(const hw_module_t *module, const char *name, 2420 hw_device_t **device) 2421{ 2422 struct audio_device *adev; 2423 int i, ret; 2424 2425 ALOGD("%s: enter", __func__); 2426 if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL; 2427 2428 adev = calloc(1, sizeof(struct audio_device)); 2429 2430 pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL); 2431 2432 adev->device.common.tag = HARDWARE_DEVICE_TAG; 2433 adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0; 2434 adev->device.common.module = (struct hw_module_t *)module; 2435 adev->device.common.close = adev_close; 2436 2437 adev->device.init_check = adev_init_check; 2438 adev->device.set_voice_volume = adev_set_voice_volume; 2439 adev->device.set_master_volume = adev_set_master_volume; 2440 adev->device.get_master_volume = adev_get_master_volume; 2441 adev->device.set_master_mute = adev_set_master_mute; 2442 adev->device.get_master_mute = adev_get_master_mute; 2443 adev->device.set_mode = adev_set_mode; 2444 adev->device.set_mic_mute = adev_set_mic_mute; 2445 adev->device.get_mic_mute = adev_get_mic_mute; 2446 adev->device.set_parameters = adev_set_parameters; 2447 adev->device.get_parameters = adev_get_parameters; 2448 adev->device.get_input_buffer_size = adev_get_input_buffer_size; 2449 adev->device.open_output_stream = adev_open_output_stream; 2450 adev->device.close_output_stream = adev_close_output_stream; 2451 adev->device.open_input_stream = adev_open_input_stream; 2452 adev->device.close_input_stream = adev_close_input_stream; 2453 adev->device.dump = adev_dump; 2454 2455 /* Set the default route before the PCM stream is opened */ 2456 pthread_mutex_lock(&adev->lock); 2457 adev->mode = AUDIO_MODE_NORMAL; 2458 adev->active_input = NULL; 2459 adev->primary_output = NULL; 2460 adev->bluetooth_nrec = true; 2461 adev->acdb_settings = TTY_MODE_OFF; 2462 /* adev->cur_hdmi_channels = 0; by calloc() */ 2463 adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int)); 2464 voice_init(adev); 2465 list_init(&adev->usecase_list); 2466 pthread_mutex_unlock(&adev->lock); 2467 2468 /* Loads platform specific libraries dynamically */ 2469 adev->platform = platform_init(adev); 2470 if (!adev->platform) { 2471 free(adev->snd_dev_ref_cnt); 2472 free(adev); 2473 ALOGE("%s: Failed to init platform data, aborting.", __func__); 2474 *device = NULL; 2475 return -EINVAL; 2476 } 2477 2478 if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) { 2479 adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW); 2480 if (adev->visualizer_lib == NULL) { 2481 ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH); 2482 } else { 2483 ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH); 2484 adev->visualizer_start_output = 2485 (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib, 2486 "visualizer_hal_start_output"); 2487 adev->visualizer_stop_output = 2488 (int (*)(audio_io_handle_t))dlsym(adev->visualizer_lib, 2489 "visualizer_hal_stop_output"); 2490 } 2491 } 2492 2493 adev->bt_wb_speech_enabled = false; 2494 2495 *device = &adev->device.common; 2496 if (k_enable_extended_precision) 2497 adev_verify_devices(adev); 2498 2499 char value[PROPERTY_VALUE_MAX]; 2500 int trial; 2501 if (property_get("audio_hal.period_size", value, NULL) > 0) { 2502 trial = atoi(value); 2503 if (period_size_is_plausible_for_low_latency(trial)) { 2504 pcm_config_low_latency.period_size = trial; 2505 pcm_config_low_latency.start_threshold = trial / 4; 2506 pcm_config_low_latency.avail_min = trial / 4; 2507 configured_low_latency_capture_period_size = trial; 2508 } 2509 } 2510 if (property_get("audio_hal.in_period_size", value, NULL) > 0) { 2511 trial = atoi(value); 2512 if (period_size_is_plausible_for_low_latency(trial)) { 2513 configured_low_latency_capture_period_size = trial; 2514 } 2515 } 2516 2517 ALOGV("%s: exit", __func__); 2518 return 0; 2519} 2520 2521static struct hw_module_methods_t hal_module_methods = { 2522 .open = adev_open, 2523}; 2524 2525struct audio_module HAL_MODULE_INFO_SYM = { 2526 .common = { 2527 .tag = HARDWARE_MODULE_TAG, 2528 .module_api_version = AUDIO_MODULE_API_VERSION_0_1, 2529 .hal_api_version = HARDWARE_HAL_API_VERSION, 2530 .id = AUDIO_HARDWARE_MODULE_ID, 2531 .name = "QCOM Audio HAL", 2532 .author = "Code Aurora Forum", 2533 .methods = &hal_module_methods, 2534 }, 2535}; 2536