AudioFlinger.cpp revision d3cee2f0f649c01e1153d593cbe723887b8e0ba0
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <cpustats/ThreadCpuUsage.h> 65#include <powermanager/PowerManager.h> 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67 68#include <common_time/cc_helper.h> 69#include <common_time/local_clock.h> 70 71// ---------------------------------------------------------------------------- 72 73 74namespace android { 75 76static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 77static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 78 79static const float MAX_GAIN = 4096.0f; 80static const uint32_t MAX_GAIN_INT = 0x1000; 81 82// retry counts for buffer fill timeout 83// 50 * ~20msecs = 1 second 84static const int8_t kMaxTrackRetries = 50; 85static const int8_t kMaxTrackStartupRetries = 50; 86// allow less retry attempts on direct output thread. 87// direct outputs can be a scarce resource in audio hardware and should 88// be released as quickly as possible. 89static const int8_t kMaxTrackRetriesDirect = 2; 90 91static const int kDumpLockRetries = 50; 92static const int kDumpLockSleepUs = 20000; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait for setParameters to complete 101static const nsecs_t kSetParametersTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 109 110// ---------------------------------------------------------------------------- 111 112#ifdef ADD_BATTERY_DATA 113// To collect the amplifier usage 114static void addBatteryData(uint32_t params) { 115 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 116 if (service == NULL) { 117 // it already logged 118 return; 119 } 120 121 service->addBatteryData(params); 122} 123#endif 124 125static int load_audio_interface(const char *if_name, const hw_module_t **mod, 126 audio_hw_device_t **dev) 127{ 128 int rc; 129 130 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 131 if (rc) 132 goto out; 133 134 rc = audio_hw_device_open(*mod, dev); 135 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 136 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 137 if (rc) 138 goto out; 139 140 return 0; 141 142out: 143 *mod = NULL; 144 *dev = NULL; 145 return rc; 146} 147 148static const char * const audio_interfaces[] = { 149 "primary", 150 "a2dp", 151 "usb", 152}; 153#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 154 155// ---------------------------------------------------------------------------- 156 157AudioFlinger::AudioFlinger() 158 : BnAudioFlinger(), 159 mPrimaryHardwareDev(NULL), 160 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 161 mMasterVolume(1.0f), 162 mMasterVolumeSupportLvl(MVS_NONE), 163 mMasterMute(false), 164 mNextUniqueId(1), 165 mMode(AUDIO_MODE_INVALID), 166 mBtNrecIsOff(false) 167{ 168} 169 170void AudioFlinger::onFirstRef() 171{ 172 int rc = 0; 173 174 Mutex::Autolock _l(mLock); 175 176 /* TODO: move all this work into an Init() function */ 177 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 178 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 179 uint32_t int_val; 180 if (1 == sscanf(val_str, "%u", &int_val)) { 181 mStandbyTimeInNsecs = milliseconds(int_val); 182 ALOGI("Using %u mSec as standby time.", int_val); 183 } else { 184 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 185 ALOGI("Using default %u mSec as standby time.", 186 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 187 } 188 } 189 190 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 191 const hw_module_t *mod; 192 audio_hw_device_t *dev; 193 194 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 195 if (rc) 196 continue; 197 198 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 199 mod->name, mod->id); 200 mAudioHwDevs.push(dev); 201 202 if (mPrimaryHardwareDev == NULL) { 203 mPrimaryHardwareDev = dev; 204 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 205 mod->name, mod->id, audio_interfaces[i]); 206 } 207 } 208 209 if (mPrimaryHardwareDev == NULL) { 210 ALOGE("Primary audio interface not found"); 211 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 212 } 213 214 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 215 // primary HW dev is selected can change so these conditions might not always be equivalent. 216 // When that happens, re-visit all the code that assumes this. 217 218 AutoMutex lock(mHardwareLock); 219 220 // Determine the level of master volume support the primary audio HAL has, 221 // and set the initial master volume at the same time. 222 float initialVolume = 1.0; 223 mMasterVolumeSupportLvl = MVS_NONE; 224 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 225 audio_hw_device_t *dev = mPrimaryHardwareDev; 226 227 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 228 if ((NULL != dev->get_master_volume) && 229 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 230 mMasterVolumeSupportLvl = MVS_FULL; 231 } else { 232 mMasterVolumeSupportLvl = MVS_SETONLY; 233 initialVolume = 1.0; 234 } 235 236 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 237 if ((NULL == dev->set_master_volume) || 238 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 239 mMasterVolumeSupportLvl = MVS_NONE; 240 } 241 mHardwareStatus = AUDIO_HW_IDLE; 242 } 243 244 // Set the mode for each audio HAL, and try to set the initial volume (if 245 // supported) for all of the non-primary audio HALs. 246 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 247 audio_hw_device_t *dev = mAudioHwDevs[i]; 248 249 mHardwareStatus = AUDIO_HW_INIT; 250 rc = dev->init_check(dev); 251 mHardwareStatus = AUDIO_HW_IDLE; 252 if (rc == 0) { 253 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 254 mHardwareStatus = AUDIO_HW_SET_MODE; 255 dev->set_mode(dev, mMode); 256 257 if ((dev != mPrimaryHardwareDev) && 258 (NULL != dev->set_master_volume)) { 259 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 260 dev->set_master_volume(dev, initialVolume); 261 } 262 263 mHardwareStatus = AUDIO_HW_IDLE; 264 } 265 } 266 267 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 268 ? initialVolume 269 : 1.0; 270 mMasterVolume = initialVolume; 271 mHardwareStatus = AUDIO_HW_IDLE; 272} 273 274AudioFlinger::~AudioFlinger() 275{ 276 277 while (!mRecordThreads.isEmpty()) { 278 // closeInput() will remove first entry from mRecordThreads 279 closeInput(mRecordThreads.keyAt(0)); 280 } 281 while (!mPlaybackThreads.isEmpty()) { 282 // closeOutput() will remove first entry from mPlaybackThreads 283 closeOutput(mPlaybackThreads.keyAt(0)); 284 } 285 286 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 287 // no mHardwareLock needed, as there are no other references to this 288 audio_hw_device_close(mAudioHwDevs[i]); 289 } 290} 291 292audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 293{ 294 /* first matching HW device is returned */ 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 audio_hw_device_t *dev = mAudioHwDevs[i]; 297 if ((dev->get_supported_devices(dev) & devices) == devices) 298 return dev; 299 } 300 return NULL; 301} 302 303status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 304{ 305 const size_t SIZE = 256; 306 char buffer[SIZE]; 307 String8 result; 308 309 result.append("Clients:\n"); 310 for (size_t i = 0; i < mClients.size(); ++i) { 311 sp<Client> client = mClients.valueAt(i).promote(); 312 if (client != 0) { 313 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 314 result.append(buffer); 315 } 316 } 317 318 result.append("Global session refs:\n"); 319 result.append(" session pid count\n"); 320 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 321 AudioSessionRef *r = mAudioSessionRefs[i]; 322 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 323 result.append(buffer); 324 } 325 write(fd, result.string(), result.size()); 326 return NO_ERROR; 327} 328 329 330status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 331{ 332 const size_t SIZE = 256; 333 char buffer[SIZE]; 334 String8 result; 335 hardware_call_state hardwareStatus = mHardwareStatus; 336 337 snprintf(buffer, SIZE, "Hardware status: %d\n" 338 "Standby Time mSec: %u\n", 339 hardwareStatus, 340 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 341 result.append(buffer); 342 write(fd, result.string(), result.size()); 343 return NO_ERROR; 344} 345 346status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 347{ 348 const size_t SIZE = 256; 349 char buffer[SIZE]; 350 String8 result; 351 snprintf(buffer, SIZE, "Permission Denial: " 352 "can't dump AudioFlinger from pid=%d, uid=%d\n", 353 IPCThreadState::self()->getCallingPid(), 354 IPCThreadState::self()->getCallingUid()); 355 result.append(buffer); 356 write(fd, result.string(), result.size()); 357 return NO_ERROR; 358} 359 360static bool tryLock(Mutex& mutex) 361{ 362 bool locked = false; 363 for (int i = 0; i < kDumpLockRetries; ++i) { 364 if (mutex.tryLock() == NO_ERROR) { 365 locked = true; 366 break; 367 } 368 usleep(kDumpLockSleepUs); 369 } 370 return locked; 371} 372 373status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 374{ 375 if (!dumpAllowed()) { 376 dumpPermissionDenial(fd, args); 377 } else { 378 // get state of hardware lock 379 bool hardwareLocked = tryLock(mHardwareLock); 380 if (!hardwareLocked) { 381 String8 result(kHardwareLockedString); 382 write(fd, result.string(), result.size()); 383 } else { 384 mHardwareLock.unlock(); 385 } 386 387 bool locked = tryLock(mLock); 388 389 // failed to lock - AudioFlinger is probably deadlocked 390 if (!locked) { 391 String8 result(kDeadlockedString); 392 write(fd, result.string(), result.size()); 393 } 394 395 dumpClients(fd, args); 396 dumpInternals(fd, args); 397 398 // dump playback threads 399 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 400 mPlaybackThreads.valueAt(i)->dump(fd, args); 401 } 402 403 // dump record threads 404 for (size_t i = 0; i < mRecordThreads.size(); i++) { 405 mRecordThreads.valueAt(i)->dump(fd, args); 406 } 407 408 // dump all hardware devs 409 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 410 audio_hw_device_t *dev = mAudioHwDevs[i]; 411 dev->dump(dev, fd); 412 } 413 if (locked) mLock.unlock(); 414 } 415 return NO_ERROR; 416} 417 418sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 419{ 420 // If pid is already in the mClients wp<> map, then use that entry 421 // (for which promote() is always != 0), otherwise create a new entry and Client. 422 sp<Client> client = mClients.valueFor(pid).promote(); 423 if (client == 0) { 424 client = new Client(this, pid); 425 mClients.add(pid, client); 426 } 427 428 return client; 429} 430 431// IAudioFlinger interface 432 433 434sp<IAudioTrack> AudioFlinger::createTrack( 435 pid_t pid, 436 audio_stream_type_t streamType, 437 uint32_t sampleRate, 438 audio_format_t format, 439 uint32_t channelMask, 440 int frameCount, 441 // FIXME dead, remove from IAudioFlinger 442 uint32_t flags, 443 const sp<IMemory>& sharedBuffer, 444 audio_io_handle_t output, 445 bool isTimed, 446 int *sessionId, 447 status_t *status) 448{ 449 sp<PlaybackThread::Track> track; 450 sp<TrackHandle> trackHandle; 451 sp<Client> client; 452 status_t lStatus; 453 int lSessionId; 454 455 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 456 // but if someone uses binder directly they could bypass that and cause us to crash 457 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 458 ALOGE("createTrack() invalid stream type %d", streamType); 459 lStatus = BAD_VALUE; 460 goto Exit; 461 } 462 463 { 464 Mutex::Autolock _l(mLock); 465 PlaybackThread *thread = checkPlaybackThread_l(output); 466 PlaybackThread *effectThread = NULL; 467 if (thread == NULL) { 468 ALOGE("unknown output thread"); 469 lStatus = BAD_VALUE; 470 goto Exit; 471 } 472 473 client = registerPid_l(pid); 474 475 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 476 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 477 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 478 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 479 if (mPlaybackThreads.keyAt(i) != output) { 480 // prevent same audio session on different output threads 481 uint32_t sessions = t->hasAudioSession(*sessionId); 482 if (sessions & PlaybackThread::TRACK_SESSION) { 483 ALOGE("createTrack() session ID %d already in use", *sessionId); 484 lStatus = BAD_VALUE; 485 goto Exit; 486 } 487 // check if an effect with same session ID is waiting for a track to be created 488 if (sessions & PlaybackThread::EFFECT_SESSION) { 489 effectThread = t.get(); 490 } 491 } 492 } 493 lSessionId = *sessionId; 494 } else { 495 // if no audio session id is provided, create one here 496 lSessionId = nextUniqueId(); 497 if (sessionId != NULL) { 498 *sessionId = lSessionId; 499 } 500 } 501 ALOGV("createTrack() lSessionId: %d", lSessionId); 502 503 track = thread->createTrack_l(client, streamType, sampleRate, format, 504 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 505 506 // move effect chain to this output thread if an effect on same session was waiting 507 // for a track to be created 508 if (lStatus == NO_ERROR && effectThread != NULL) { 509 Mutex::Autolock _dl(thread->mLock); 510 Mutex::Autolock _sl(effectThread->mLock); 511 moveEffectChain_l(lSessionId, effectThread, thread, true); 512 } 513 } 514 if (lStatus == NO_ERROR) { 515 trackHandle = new TrackHandle(track); 516 } else { 517 // remove local strong reference to Client before deleting the Track so that the Client 518 // destructor is called by the TrackBase destructor with mLock held 519 client.clear(); 520 track.clear(); 521 } 522 523Exit: 524 if (status != NULL) { 525 *status = lStatus; 526 } 527 return trackHandle; 528} 529 530uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 531{ 532 Mutex::Autolock _l(mLock); 533 PlaybackThread *thread = checkPlaybackThread_l(output); 534 if (thread == NULL) { 535 ALOGW("sampleRate() unknown thread %d", output); 536 return 0; 537 } 538 return thread->sampleRate(); 539} 540 541int AudioFlinger::channelCount(audio_io_handle_t output) const 542{ 543 Mutex::Autolock _l(mLock); 544 PlaybackThread *thread = checkPlaybackThread_l(output); 545 if (thread == NULL) { 546 ALOGW("channelCount() unknown thread %d", output); 547 return 0; 548 } 549 return thread->channelCount(); 550} 551 552audio_format_t AudioFlinger::format(audio_io_handle_t output) const 553{ 554 Mutex::Autolock _l(mLock); 555 PlaybackThread *thread = checkPlaybackThread_l(output); 556 if (thread == NULL) { 557 ALOGW("format() unknown thread %d", output); 558 return AUDIO_FORMAT_INVALID; 559 } 560 return thread->format(); 561} 562 563size_t AudioFlinger::frameCount(audio_io_handle_t output) const 564{ 565 Mutex::Autolock _l(mLock); 566 PlaybackThread *thread = checkPlaybackThread_l(output); 567 if (thread == NULL) { 568 ALOGW("frameCount() unknown thread %d", output); 569 return 0; 570 } 571 return thread->frameCount(); 572} 573 574uint32_t AudioFlinger::latency(audio_io_handle_t output) const 575{ 576 Mutex::Autolock _l(mLock); 577 PlaybackThread *thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 ALOGW("latency() unknown thread %d", output); 580 return 0; 581 } 582 return thread->latency(); 583} 584 585status_t AudioFlinger::setMasterVolume(float value) 586{ 587 status_t ret = initCheck(); 588 if (ret != NO_ERROR) { 589 return ret; 590 } 591 592 // check calling permissions 593 if (!settingsAllowed()) { 594 return PERMISSION_DENIED; 595 } 596 597 float swmv = value; 598 599 // when hw supports master volume, don't scale in sw mixer 600 if (MVS_NONE != mMasterVolumeSupportLvl) { 601 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 602 AutoMutex lock(mHardwareLock); 603 audio_hw_device_t *dev = mAudioHwDevs[i]; 604 605 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 606 if (NULL != dev->set_master_volume) { 607 dev->set_master_volume(dev, value); 608 } 609 mHardwareStatus = AUDIO_HW_IDLE; 610 } 611 612 swmv = 1.0; 613 } 614 615 Mutex::Autolock _l(mLock); 616 mMasterVolume = value; 617 mMasterVolumeSW = swmv; 618 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 619 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 620 621 return NO_ERROR; 622} 623 624status_t AudioFlinger::setMode(audio_mode_t mode) 625{ 626 status_t ret = initCheck(); 627 if (ret != NO_ERROR) { 628 return ret; 629 } 630 631 // check calling permissions 632 if (!settingsAllowed()) { 633 return PERMISSION_DENIED; 634 } 635 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 636 ALOGW("Illegal value: setMode(%d)", mode); 637 return BAD_VALUE; 638 } 639 640 { // scope for the lock 641 AutoMutex lock(mHardwareLock); 642 mHardwareStatus = AUDIO_HW_SET_MODE; 643 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 644 mHardwareStatus = AUDIO_HW_IDLE; 645 } 646 647 if (NO_ERROR == ret) { 648 Mutex::Autolock _l(mLock); 649 mMode = mode; 650 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 651 mPlaybackThreads.valueAt(i)->setMode(mode); 652 } 653 654 return ret; 655} 656 657status_t AudioFlinger::setMicMute(bool state) 658{ 659 status_t ret = initCheck(); 660 if (ret != NO_ERROR) { 661 return ret; 662 } 663 664 // check calling permissions 665 if (!settingsAllowed()) { 666 return PERMISSION_DENIED; 667 } 668 669 AutoMutex lock(mHardwareLock); 670 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 671 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 672 mHardwareStatus = AUDIO_HW_IDLE; 673 return ret; 674} 675 676bool AudioFlinger::getMicMute() const 677{ 678 status_t ret = initCheck(); 679 if (ret != NO_ERROR) { 680 return false; 681 } 682 683 bool state = AUDIO_MODE_INVALID; 684 AutoMutex lock(mHardwareLock); 685 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 686 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 687 mHardwareStatus = AUDIO_HW_IDLE; 688 return state; 689} 690 691status_t AudioFlinger::setMasterMute(bool muted) 692{ 693 // check calling permissions 694 if (!settingsAllowed()) { 695 return PERMISSION_DENIED; 696 } 697 698 Mutex::Autolock _l(mLock); 699 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 700 mMasterMute = muted; 701 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 702 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 703 704 return NO_ERROR; 705} 706 707float AudioFlinger::masterVolume() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolume_l(); 711} 712 713float AudioFlinger::masterVolumeSW() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterVolumeSW_l(); 717} 718 719bool AudioFlinger::masterMute() const 720{ 721 Mutex::Autolock _l(mLock); 722 return masterMute_l(); 723} 724 725float AudioFlinger::masterVolume_l() const 726{ 727 if (MVS_FULL == mMasterVolumeSupportLvl) { 728 float ret_val; 729 AutoMutex lock(mHardwareLock); 730 731 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 732 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 733 (NULL != mPrimaryHardwareDev->get_master_volume), 734 "can't get master volume"); 735 736 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 737 mHardwareStatus = AUDIO_HW_IDLE; 738 return ret_val; 739 } 740 741 return mMasterVolume; 742} 743 744status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 745 audio_io_handle_t output) 746{ 747 // check calling permissions 748 if (!settingsAllowed()) { 749 return PERMISSION_DENIED; 750 } 751 752 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 753 ALOGE("setStreamVolume() invalid stream %d", stream); 754 return BAD_VALUE; 755 } 756 757 AutoMutex lock(mLock); 758 PlaybackThread *thread = NULL; 759 if (output) { 760 thread = checkPlaybackThread_l(output); 761 if (thread == NULL) { 762 return BAD_VALUE; 763 } 764 } 765 766 mStreamTypes[stream].volume = value; 767 768 if (thread == NULL) { 769 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 770 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 771 } 772 } else { 773 thread->setStreamVolume(stream, value); 774 } 775 776 return NO_ERROR; 777} 778 779status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 780{ 781 // check calling permissions 782 if (!settingsAllowed()) { 783 return PERMISSION_DENIED; 784 } 785 786 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 787 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 788 ALOGE("setStreamMute() invalid stream %d", stream); 789 return BAD_VALUE; 790 } 791 792 AutoMutex lock(mLock); 793 mStreamTypes[stream].mute = muted; 794 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 795 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 796 797 return NO_ERROR; 798} 799 800float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 801{ 802 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 803 return 0.0f; 804 } 805 806 AutoMutex lock(mLock); 807 float volume; 808 if (output) { 809 PlaybackThread *thread = checkPlaybackThread_l(output); 810 if (thread == NULL) { 811 return 0.0f; 812 } 813 volume = thread->streamVolume(stream); 814 } else { 815 volume = streamVolume_l(stream); 816 } 817 818 return volume; 819} 820 821bool AudioFlinger::streamMute(audio_stream_type_t stream) const 822{ 823 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 824 return true; 825 } 826 827 AutoMutex lock(mLock); 828 return streamMute_l(stream); 829} 830 831status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 832{ 833 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 834 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 835 // check calling permissions 836 if (!settingsAllowed()) { 837 return PERMISSION_DENIED; 838 } 839 840 // ioHandle == 0 means the parameters are global to the audio hardware interface 841 if (ioHandle == 0) { 842 status_t final_result = NO_ERROR; 843 { 844 AutoMutex lock(mHardwareLock); 845 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 846 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 847 audio_hw_device_t *dev = mAudioHwDevs[i]; 848 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 849 final_result = result ?: final_result; 850 } 851 mHardwareStatus = AUDIO_HW_IDLE; 852 } 853 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 854 AudioParameter param = AudioParameter(keyValuePairs); 855 String8 value; 856 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 857 Mutex::Autolock _l(mLock); 858 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 859 if (mBtNrecIsOff != btNrecIsOff) { 860 for (size_t i = 0; i < mRecordThreads.size(); i++) { 861 sp<RecordThread> thread = mRecordThreads.valueAt(i); 862 RecordThread::RecordTrack *track = thread->track(); 863 if (track != NULL) { 864 audio_devices_t device = (audio_devices_t)( 865 thread->device() & AUDIO_DEVICE_IN_ALL); 866 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 867 thread->setEffectSuspended(FX_IID_AEC, 868 suspend, 869 track->sessionId()); 870 thread->setEffectSuspended(FX_IID_NS, 871 suspend, 872 track->sessionId()); 873 } 874 } 875 mBtNrecIsOff = btNrecIsOff; 876 } 877 } 878 return final_result; 879 } 880 881 // hold a strong ref on thread in case closeOutput() or closeInput() is called 882 // and the thread is exited once the lock is released 883 sp<ThreadBase> thread; 884 { 885 Mutex::Autolock _l(mLock); 886 thread = checkPlaybackThread_l(ioHandle); 887 if (thread == NULL) { 888 thread = checkRecordThread_l(ioHandle); 889 } else if (thread == primaryPlaybackThread_l()) { 890 // indicate output device change to all input threads for pre processing 891 AudioParameter param = AudioParameter(keyValuePairs); 892 int value; 893 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 894 for (size_t i = 0; i < mRecordThreads.size(); i++) { 895 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 896 } 897 } 898 } 899 } 900 if (thread != 0) { 901 return thread->setParameters(keyValuePairs); 902 } 903 return BAD_VALUE; 904} 905 906String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 907{ 908// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 909// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 910 911 if (ioHandle == 0) { 912 String8 out_s8; 913 914 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 915 char *s; 916 { 917 AutoMutex lock(mHardwareLock); 918 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 919 audio_hw_device_t *dev = mAudioHwDevs[i]; 920 s = dev->get_parameters(dev, keys.string()); 921 mHardwareStatus = AUDIO_HW_IDLE; 922 } 923 out_s8 += String8(s ? s : ""); 924 free(s); 925 } 926 return out_s8; 927 } 928 929 Mutex::Autolock _l(mLock); 930 931 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 932 if (playbackThread != NULL) { 933 return playbackThread->getParameters(keys); 934 } 935 RecordThread *recordThread = checkRecordThread_l(ioHandle); 936 if (recordThread != NULL) { 937 return recordThread->getParameters(keys); 938 } 939 return String8(""); 940} 941 942size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 943{ 944 status_t ret = initCheck(); 945 if (ret != NO_ERROR) { 946 return 0; 947 } 948 949 AutoMutex lock(mHardwareLock); 950 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 951 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 952 mHardwareStatus = AUDIO_HW_IDLE; 953 return size; 954} 955 956unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 957{ 958 if (ioHandle == 0) { 959 return 0; 960 } 961 962 Mutex::Autolock _l(mLock); 963 964 RecordThread *recordThread = checkRecordThread_l(ioHandle); 965 if (recordThread != NULL) { 966 return recordThread->getInputFramesLost(); 967 } 968 return 0; 969} 970 971status_t AudioFlinger::setVoiceVolume(float value) 972{ 973 status_t ret = initCheck(); 974 if (ret != NO_ERROR) { 975 return ret; 976 } 977 978 // check calling permissions 979 if (!settingsAllowed()) { 980 return PERMISSION_DENIED; 981 } 982 983 AutoMutex lock(mHardwareLock); 984 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 985 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 986 mHardwareStatus = AUDIO_HW_IDLE; 987 988 return ret; 989} 990 991status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 992 audio_io_handle_t output) const 993{ 994 status_t status; 995 996 Mutex::Autolock _l(mLock); 997 998 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 999 if (playbackThread != NULL) { 1000 return playbackThread->getRenderPosition(halFrames, dspFrames); 1001 } 1002 1003 return BAD_VALUE; 1004} 1005 1006void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1007{ 1008 1009 Mutex::Autolock _l(mLock); 1010 1011 pid_t pid = IPCThreadState::self()->getCallingPid(); 1012 if (mNotificationClients.indexOfKey(pid) < 0) { 1013 sp<NotificationClient> notificationClient = new NotificationClient(this, 1014 client, 1015 pid); 1016 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1017 1018 mNotificationClients.add(pid, notificationClient); 1019 1020 sp<IBinder> binder = client->asBinder(); 1021 binder->linkToDeath(notificationClient); 1022 1023 // the config change is always sent from playback or record threads to avoid deadlock 1024 // with AudioSystem::gLock 1025 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1026 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1027 } 1028 1029 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1030 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1031 } 1032 } 1033} 1034 1035void AudioFlinger::removeNotificationClient(pid_t pid) 1036{ 1037 Mutex::Autolock _l(mLock); 1038 1039 mNotificationClients.removeItem(pid); 1040 1041 ALOGV("%d died, releasing its sessions", pid); 1042 size_t num = mAudioSessionRefs.size(); 1043 bool removed = false; 1044 for (size_t i = 0; i< num; ) { 1045 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1046 ALOGV(" pid %d @ %d", ref->mPid, i); 1047 if (ref->mPid == pid) { 1048 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1049 mAudioSessionRefs.removeAt(i); 1050 delete ref; 1051 removed = true; 1052 num--; 1053 } else { 1054 i++; 1055 } 1056 } 1057 if (removed) { 1058 purgeStaleEffects_l(); 1059 } 1060} 1061 1062// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1063void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1064{ 1065 size_t size = mNotificationClients.size(); 1066 for (size_t i = 0; i < size; i++) { 1067 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1068 param2); 1069 } 1070} 1071 1072// removeClient_l() must be called with AudioFlinger::mLock held 1073void AudioFlinger::removeClient_l(pid_t pid) 1074{ 1075 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1076 mClients.removeItem(pid); 1077} 1078 1079 1080// ---------------------------------------------------------------------------- 1081 1082AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1083 uint32_t device, type_t type) 1084 : Thread(false), 1085 mType(type), 1086 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1087 // mChannelMask 1088 mChannelCount(0), 1089 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1090 mParamStatus(NO_ERROR), 1091 mStandby(false), mId(id), 1092 mDevice(device), 1093 mDeathRecipient(new PMDeathRecipient(this)) 1094{ 1095} 1096 1097AudioFlinger::ThreadBase::~ThreadBase() 1098{ 1099 mParamCond.broadcast(); 1100 // do not lock the mutex in destructor 1101 releaseWakeLock_l(); 1102 if (mPowerManager != 0) { 1103 sp<IBinder> binder = mPowerManager->asBinder(); 1104 binder->unlinkToDeath(mDeathRecipient); 1105 } 1106} 1107 1108void AudioFlinger::ThreadBase::exit() 1109{ 1110 ALOGV("ThreadBase::exit"); 1111 { 1112 // This lock prevents the following race in thread (uniprocessor for illustration): 1113 // if (!exitPending()) { 1114 // // context switch from here to exit() 1115 // // exit() calls requestExit(), what exitPending() observes 1116 // // exit() calls signal(), which is dropped since no waiters 1117 // // context switch back from exit() to here 1118 // mWaitWorkCV.wait(...); 1119 // // now thread is hung 1120 // } 1121 AutoMutex lock(mLock); 1122 requestExit(); 1123 mWaitWorkCV.signal(); 1124 } 1125 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1126 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1127 requestExitAndWait(); 1128} 1129 1130status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1131{ 1132 status_t status; 1133 1134 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1135 Mutex::Autolock _l(mLock); 1136 1137 mNewParameters.add(keyValuePairs); 1138 mWaitWorkCV.signal(); 1139 // wait condition with timeout in case the thread loop has exited 1140 // before the request could be processed 1141 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1142 status = mParamStatus; 1143 mWaitWorkCV.signal(); 1144 } else { 1145 status = TIMED_OUT; 1146 } 1147 return status; 1148} 1149 1150void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1151{ 1152 Mutex::Autolock _l(mLock); 1153 sendConfigEvent_l(event, param); 1154} 1155 1156// sendConfigEvent_l() must be called with ThreadBase::mLock held 1157void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1158{ 1159 ConfigEvent configEvent; 1160 configEvent.mEvent = event; 1161 configEvent.mParam = param; 1162 mConfigEvents.add(configEvent); 1163 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1164 mWaitWorkCV.signal(); 1165} 1166 1167void AudioFlinger::ThreadBase::processConfigEvents() 1168{ 1169 mLock.lock(); 1170 while (!mConfigEvents.isEmpty()) { 1171 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1172 ConfigEvent configEvent = mConfigEvents[0]; 1173 mConfigEvents.removeAt(0); 1174 // release mLock before locking AudioFlinger mLock: lock order is always 1175 // AudioFlinger then ThreadBase to avoid cross deadlock 1176 mLock.unlock(); 1177 mAudioFlinger->mLock.lock(); 1178 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1179 mAudioFlinger->mLock.unlock(); 1180 mLock.lock(); 1181 } 1182 mLock.unlock(); 1183} 1184 1185status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1186{ 1187 const size_t SIZE = 256; 1188 char buffer[SIZE]; 1189 String8 result; 1190 1191 bool locked = tryLock(mLock); 1192 if (!locked) { 1193 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1194 write(fd, buffer, strlen(buffer)); 1195 } 1196 1197 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1204 result.append(buffer); 1205 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1206 result.append(buffer); 1207 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1208 result.append(buffer); 1209 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1210 result.append(buffer); 1211 1212 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1213 result.append(buffer); 1214 result.append(" Index Command"); 1215 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1216 snprintf(buffer, SIZE, "\n %02d ", i); 1217 result.append(buffer); 1218 result.append(mNewParameters[i]); 1219 } 1220 1221 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, " Index event param\n"); 1224 result.append(buffer); 1225 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1226 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1227 result.append(buffer); 1228 } 1229 result.append("\n"); 1230 1231 write(fd, result.string(), result.size()); 1232 1233 if (locked) { 1234 mLock.unlock(); 1235 } 1236 return NO_ERROR; 1237} 1238 1239status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1240{ 1241 const size_t SIZE = 256; 1242 char buffer[SIZE]; 1243 String8 result; 1244 1245 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1246 write(fd, buffer, strlen(buffer)); 1247 1248 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1249 sp<EffectChain> chain = mEffectChains[i]; 1250 if (chain != 0) { 1251 chain->dump(fd, args); 1252 } 1253 } 1254 return NO_ERROR; 1255} 1256 1257void AudioFlinger::ThreadBase::acquireWakeLock() 1258{ 1259 Mutex::Autolock _l(mLock); 1260 acquireWakeLock_l(); 1261} 1262 1263void AudioFlinger::ThreadBase::acquireWakeLock_l() 1264{ 1265 if (mPowerManager == 0) { 1266 // use checkService() to avoid blocking if power service is not up yet 1267 sp<IBinder> binder = 1268 defaultServiceManager()->checkService(String16("power")); 1269 if (binder == 0) { 1270 ALOGW("Thread %s cannot connect to the power manager service", mName); 1271 } else { 1272 mPowerManager = interface_cast<IPowerManager>(binder); 1273 binder->linkToDeath(mDeathRecipient); 1274 } 1275 } 1276 if (mPowerManager != 0) { 1277 sp<IBinder> binder = new BBinder(); 1278 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1279 binder, 1280 String16(mName)); 1281 if (status == NO_ERROR) { 1282 mWakeLockToken = binder; 1283 } 1284 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1285 } 1286} 1287 1288void AudioFlinger::ThreadBase::releaseWakeLock() 1289{ 1290 Mutex::Autolock _l(mLock); 1291 releaseWakeLock_l(); 1292} 1293 1294void AudioFlinger::ThreadBase::releaseWakeLock_l() 1295{ 1296 if (mWakeLockToken != 0) { 1297 ALOGV("releaseWakeLock_l() %s", mName); 1298 if (mPowerManager != 0) { 1299 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1300 } 1301 mWakeLockToken.clear(); 1302 } 1303} 1304 1305void AudioFlinger::ThreadBase::clearPowerManager() 1306{ 1307 Mutex::Autolock _l(mLock); 1308 releaseWakeLock_l(); 1309 mPowerManager.clear(); 1310} 1311 1312void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1313{ 1314 sp<ThreadBase> thread = mThread.promote(); 1315 if (thread != 0) { 1316 thread->clearPowerManager(); 1317 } 1318 ALOGW("power manager service died !!!"); 1319} 1320 1321void AudioFlinger::ThreadBase::setEffectSuspended( 1322 const effect_uuid_t *type, bool suspend, int sessionId) 1323{ 1324 Mutex::Autolock _l(mLock); 1325 setEffectSuspended_l(type, suspend, sessionId); 1326} 1327 1328void AudioFlinger::ThreadBase::setEffectSuspended_l( 1329 const effect_uuid_t *type, bool suspend, int sessionId) 1330{ 1331 sp<EffectChain> chain = getEffectChain_l(sessionId); 1332 if (chain != 0) { 1333 if (type != NULL) { 1334 chain->setEffectSuspended_l(type, suspend); 1335 } else { 1336 chain->setEffectSuspendedAll_l(suspend); 1337 } 1338 } 1339 1340 updateSuspendedSessions_l(type, suspend, sessionId); 1341} 1342 1343void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1344{ 1345 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1346 if (index < 0) { 1347 return; 1348 } 1349 1350 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1351 mSuspendedSessions.editValueAt(index); 1352 1353 for (size_t i = 0; i < sessionEffects.size(); i++) { 1354 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1355 for (int j = 0; j < desc->mRefCount; j++) { 1356 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1357 chain->setEffectSuspendedAll_l(true); 1358 } else { 1359 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1360 desc->mType.timeLow); 1361 chain->setEffectSuspended_l(&desc->mType, true); 1362 } 1363 } 1364 } 1365} 1366 1367void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1368 bool suspend, 1369 int sessionId) 1370{ 1371 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1372 1373 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1374 1375 if (suspend) { 1376 if (index >= 0) { 1377 sessionEffects = mSuspendedSessions.editValueAt(index); 1378 } else { 1379 mSuspendedSessions.add(sessionId, sessionEffects); 1380 } 1381 } else { 1382 if (index < 0) { 1383 return; 1384 } 1385 sessionEffects = mSuspendedSessions.editValueAt(index); 1386 } 1387 1388 1389 int key = EffectChain::kKeyForSuspendAll; 1390 if (type != NULL) { 1391 key = type->timeLow; 1392 } 1393 index = sessionEffects.indexOfKey(key); 1394 1395 sp<SuspendedSessionDesc> desc; 1396 if (suspend) { 1397 if (index >= 0) { 1398 desc = sessionEffects.valueAt(index); 1399 } else { 1400 desc = new SuspendedSessionDesc(); 1401 if (type != NULL) { 1402 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1403 } 1404 sessionEffects.add(key, desc); 1405 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1406 } 1407 desc->mRefCount++; 1408 } else { 1409 if (index < 0) { 1410 return; 1411 } 1412 desc = sessionEffects.valueAt(index); 1413 if (--desc->mRefCount == 0) { 1414 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1415 sessionEffects.removeItemsAt(index); 1416 if (sessionEffects.isEmpty()) { 1417 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1418 sessionId); 1419 mSuspendedSessions.removeItem(sessionId); 1420 } 1421 } 1422 } 1423 if (!sessionEffects.isEmpty()) { 1424 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1425 } 1426} 1427 1428void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1429 bool enabled, 1430 int sessionId) 1431{ 1432 Mutex::Autolock _l(mLock); 1433 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1434} 1435 1436void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1437 bool enabled, 1438 int sessionId) 1439{ 1440 if (mType != RECORD) { 1441 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1442 // another session. This gives the priority to well behaved effect control panels 1443 // and applications not using global effects. 1444 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1445 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1446 } 1447 } 1448 1449 sp<EffectChain> chain = getEffectChain_l(sessionId); 1450 if (chain != 0) { 1451 chain->checkSuspendOnEffectEnabled(effect, enabled); 1452 } 1453} 1454 1455// ---------------------------------------------------------------------------- 1456 1457AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1458 AudioStreamOut* output, 1459 audio_io_handle_t id, 1460 uint32_t device, 1461 type_t type) 1462 : ThreadBase(audioFlinger, id, device, type), 1463 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1464 // Assumes constructor is called by AudioFlinger with it's mLock held, 1465 // but it would be safer to explicitly pass initial masterMute as parameter 1466 mMasterMute(audioFlinger->masterMute_l()), 1467 // mStreamTypes[] initialized in constructor body 1468 mOutput(output), 1469 // Assumes constructor is called by AudioFlinger with it's mLock held, 1470 // but it would be safer to explicitly pass initial masterVolume as parameter 1471 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1472 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1473 mMixerStatus(MIXER_IDLE), 1474 mPrevMixerStatus(MIXER_IDLE), 1475 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1476{ 1477 snprintf(mName, kNameLength, "AudioOut_%X", id); 1478 1479 readOutputParameters(); 1480 1481 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1482 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1483 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1484 stream = (audio_stream_type_t) (stream + 1)) { 1485 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1486 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1487 // initialized by stream_type_t default constructor 1488 // mStreamTypes[stream].valid = true; 1489 } 1490 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1491 // because mAudioFlinger doesn't have one to copy from 1492} 1493 1494AudioFlinger::PlaybackThread::~PlaybackThread() 1495{ 1496 delete [] mMixBuffer; 1497} 1498 1499status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1500{ 1501 dumpInternals(fd, args); 1502 dumpTracks(fd, args); 1503 dumpEffectChains(fd, args); 1504 return NO_ERROR; 1505} 1506 1507status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1508{ 1509 const size_t SIZE = 256; 1510 char buffer[SIZE]; 1511 String8 result; 1512 1513 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1514 result.append(buffer); 1515 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1516 for (size_t i = 0; i < mTracks.size(); ++i) { 1517 sp<Track> track = mTracks[i]; 1518 if (track != 0) { 1519 track->dump(buffer, SIZE); 1520 result.append(buffer); 1521 } 1522 } 1523 1524 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1525 result.append(buffer); 1526 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1527 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1528 sp<Track> track = mActiveTracks[i].promote(); 1529 if (track != 0) { 1530 track->dump(buffer, SIZE); 1531 result.append(buffer); 1532 } 1533 } 1534 write(fd, result.string(), result.size()); 1535 return NO_ERROR; 1536} 1537 1538status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1539{ 1540 const size_t SIZE = 256; 1541 char buffer[SIZE]; 1542 String8 result; 1543 1544 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1547 result.append(buffer); 1548 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1549 result.append(buffer); 1550 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1551 result.append(buffer); 1552 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1553 result.append(buffer); 1554 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1555 result.append(buffer); 1556 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1557 result.append(buffer); 1558 write(fd, result.string(), result.size()); 1559 1560 dumpBase(fd, args); 1561 1562 return NO_ERROR; 1563} 1564 1565// Thread virtuals 1566status_t AudioFlinger::PlaybackThread::readyToRun() 1567{ 1568 status_t status = initCheck(); 1569 if (status == NO_ERROR) { 1570 ALOGI("AudioFlinger's thread %p ready to run", this); 1571 } else { 1572 ALOGE("No working audio driver found."); 1573 } 1574 return status; 1575} 1576 1577void AudioFlinger::PlaybackThread::onFirstRef() 1578{ 1579 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1580} 1581 1582// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1583sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1584 const sp<AudioFlinger::Client>& client, 1585 audio_stream_type_t streamType, 1586 uint32_t sampleRate, 1587 audio_format_t format, 1588 uint32_t channelMask, 1589 int frameCount, 1590 const sp<IMemory>& sharedBuffer, 1591 int sessionId, 1592 bool isTimed, 1593 status_t *status) 1594{ 1595 sp<Track> track; 1596 status_t lStatus; 1597 1598 if (mType == DIRECT) { 1599 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1600 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1601 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1602 "for output %p with format %d", 1603 sampleRate, format, channelMask, mOutput, mFormat); 1604 lStatus = BAD_VALUE; 1605 goto Exit; 1606 } 1607 } 1608 } else { 1609 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1610 if (sampleRate > mSampleRate*2) { 1611 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1612 lStatus = BAD_VALUE; 1613 goto Exit; 1614 } 1615 } 1616 1617 lStatus = initCheck(); 1618 if (lStatus != NO_ERROR) { 1619 ALOGE("Audio driver not initialized."); 1620 goto Exit; 1621 } 1622 1623 { // scope for mLock 1624 Mutex::Autolock _l(mLock); 1625 1626 // all tracks in same audio session must share the same routing strategy otherwise 1627 // conflicts will happen when tracks are moved from one output to another by audio policy 1628 // manager 1629 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1630 for (size_t i = 0; i < mTracks.size(); ++i) { 1631 sp<Track> t = mTracks[i]; 1632 if (t != 0 && !t->isOutputTrack()) { 1633 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1634 if (sessionId == t->sessionId() && strategy != actual) { 1635 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1636 strategy, actual); 1637 lStatus = BAD_VALUE; 1638 goto Exit; 1639 } 1640 } 1641 } 1642 1643 if (!isTimed) { 1644 track = new Track(this, client, streamType, sampleRate, format, 1645 channelMask, frameCount, sharedBuffer, sessionId); 1646 } else { 1647 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1648 channelMask, frameCount, sharedBuffer, sessionId); 1649 } 1650 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1651 lStatus = NO_MEMORY; 1652 goto Exit; 1653 } 1654 mTracks.add(track); 1655 1656 sp<EffectChain> chain = getEffectChain_l(sessionId); 1657 if (chain != 0) { 1658 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1659 track->setMainBuffer(chain->inBuffer()); 1660 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1661 chain->incTrackCnt(); 1662 } 1663 1664 // invalidate track immediately if the stream type was moved to another thread since 1665 // createTrack() was called by the client process. 1666 if (!mStreamTypes[streamType].valid) { 1667 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1668 this, streamType); 1669 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1670 } 1671 } 1672 lStatus = NO_ERROR; 1673 1674Exit: 1675 if (status) { 1676 *status = lStatus; 1677 } 1678 return track; 1679} 1680 1681uint32_t AudioFlinger::PlaybackThread::latency() const 1682{ 1683 Mutex::Autolock _l(mLock); 1684 if (initCheck() == NO_ERROR) { 1685 return mOutput->stream->get_latency(mOutput->stream); 1686 } else { 1687 return 0; 1688 } 1689} 1690 1691void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 mMasterVolume = value; 1695} 1696 1697void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1698{ 1699 Mutex::Autolock _l(mLock); 1700 setMasterMute_l(muted); 1701} 1702 1703void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1704{ 1705 Mutex::Autolock _l(mLock); 1706 mStreamTypes[stream].volume = value; 1707} 1708 1709void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1710{ 1711 Mutex::Autolock _l(mLock); 1712 mStreamTypes[stream].mute = muted; 1713} 1714 1715float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1716{ 1717 Mutex::Autolock _l(mLock); 1718 return mStreamTypes[stream].volume; 1719} 1720 1721// addTrack_l() must be called with ThreadBase::mLock held 1722status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1723{ 1724 status_t status = ALREADY_EXISTS; 1725 1726 // set retry count for buffer fill 1727 track->mRetryCount = kMaxTrackStartupRetries; 1728 if (mActiveTracks.indexOf(track) < 0) { 1729 // the track is newly added, make sure it fills up all its 1730 // buffers before playing. This is to ensure the client will 1731 // effectively get the latency it requested. 1732 track->mFillingUpStatus = Track::FS_FILLING; 1733 track->mResetDone = false; 1734 mActiveTracks.add(track); 1735 if (track->mainBuffer() != mMixBuffer) { 1736 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1737 if (chain != 0) { 1738 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1739 chain->incActiveTrackCnt(); 1740 } 1741 } 1742 1743 status = NO_ERROR; 1744 } 1745 1746 ALOGV("mWaitWorkCV.broadcast"); 1747 mWaitWorkCV.broadcast(); 1748 1749 return status; 1750} 1751 1752// destroyTrack_l() must be called with ThreadBase::mLock held 1753void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1754{ 1755 track->mState = TrackBase::TERMINATED; 1756 if (mActiveTracks.indexOf(track) < 0) { 1757 removeTrack_l(track); 1758 } 1759} 1760 1761void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1762{ 1763 mTracks.remove(track); 1764 deleteTrackName_l(track->name()); 1765 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1766 if (chain != 0) { 1767 chain->decTrackCnt(); 1768 } 1769} 1770 1771String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1772{ 1773 String8 out_s8 = String8(""); 1774 char *s; 1775 1776 Mutex::Autolock _l(mLock); 1777 if (initCheck() != NO_ERROR) { 1778 return out_s8; 1779 } 1780 1781 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1782 out_s8 = String8(s); 1783 free(s); 1784 return out_s8; 1785} 1786 1787// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1788void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1789 AudioSystem::OutputDescriptor desc; 1790 void *param2 = NULL; 1791 1792 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1793 1794 switch (event) { 1795 case AudioSystem::OUTPUT_OPENED: 1796 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1797 desc.channels = mChannelMask; 1798 desc.samplingRate = mSampleRate; 1799 desc.format = mFormat; 1800 desc.frameCount = mFrameCount; 1801 desc.latency = latency(); 1802 param2 = &desc; 1803 break; 1804 1805 case AudioSystem::STREAM_CONFIG_CHANGED: 1806 param2 = ¶m; 1807 case AudioSystem::OUTPUT_CLOSED: 1808 default: 1809 break; 1810 } 1811 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1812} 1813 1814void AudioFlinger::PlaybackThread::readOutputParameters() 1815{ 1816 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1817 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1818 mChannelCount = (uint16_t)popcount(mChannelMask); 1819 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1820 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1821 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1822 1823 // FIXME - Current mixer implementation only supports stereo output: Always 1824 // Allocate a stereo buffer even if HW output is mono. 1825 delete[] mMixBuffer; 1826 mMixBuffer = new int16_t[mFrameCount * 2]; 1827 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1828 1829 // force reconfiguration of effect chains and engines to take new buffer size and audio 1830 // parameters into account 1831 // Note that mLock is not held when readOutputParameters() is called from the constructor 1832 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1833 // matter. 1834 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1835 Vector< sp<EffectChain> > effectChains = mEffectChains; 1836 for (size_t i = 0; i < effectChains.size(); i ++) { 1837 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1838 } 1839} 1840 1841status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1842{ 1843 if (halFrames == NULL || dspFrames == NULL) { 1844 return BAD_VALUE; 1845 } 1846 Mutex::Autolock _l(mLock); 1847 if (initCheck() != NO_ERROR) { 1848 return INVALID_OPERATION; 1849 } 1850 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1851 1852 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1853} 1854 1855uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1856{ 1857 Mutex::Autolock _l(mLock); 1858 uint32_t result = 0; 1859 if (getEffectChain_l(sessionId) != 0) { 1860 result = EFFECT_SESSION; 1861 } 1862 1863 for (size_t i = 0; i < mTracks.size(); ++i) { 1864 sp<Track> track = mTracks[i]; 1865 if (sessionId == track->sessionId() && 1866 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1867 result |= TRACK_SESSION; 1868 break; 1869 } 1870 } 1871 1872 return result; 1873} 1874 1875uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1876{ 1877 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1878 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1879 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1880 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1881 } 1882 for (size_t i = 0; i < mTracks.size(); i++) { 1883 sp<Track> track = mTracks[i]; 1884 if (sessionId == track->sessionId() && 1885 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1886 return AudioSystem::getStrategyForStream(track->streamType()); 1887 } 1888 } 1889 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1890} 1891 1892 1893AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1894{ 1895 Mutex::Autolock _l(mLock); 1896 return mOutput; 1897} 1898 1899AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1900{ 1901 Mutex::Autolock _l(mLock); 1902 AudioStreamOut *output = mOutput; 1903 mOutput = NULL; 1904 return output; 1905} 1906 1907// this method must always be called either with ThreadBase mLock held or inside the thread loop 1908audio_stream_t* AudioFlinger::PlaybackThread::stream() 1909{ 1910 if (mOutput == NULL) { 1911 return NULL; 1912 } 1913 return &mOutput->stream->common; 1914} 1915 1916uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1917{ 1918 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1919 // decoding and transfer time. So sleeping for half of the latency would likely cause 1920 // underruns 1921 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1922 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1923 } else { 1924 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1925 } 1926} 1927 1928// ---------------------------------------------------------------------------- 1929 1930AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1931 audio_io_handle_t id, uint32_t device, type_t type) 1932 : PlaybackThread(audioFlinger, output, id, device, type) 1933{ 1934 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1935 // FIXME - Current mixer implementation only supports stereo output 1936 if (mChannelCount == 1) { 1937 ALOGE("Invalid audio hardware channel count"); 1938 } 1939} 1940 1941AudioFlinger::MixerThread::~MixerThread() 1942{ 1943 delete mAudioMixer; 1944} 1945 1946class CpuStats { 1947public: 1948 void sample(); 1949#ifdef DEBUG_CPU_USAGE 1950private: 1951 ThreadCpuUsage mCpu; 1952#endif 1953}; 1954 1955void CpuStats::sample() { 1956#ifdef DEBUG_CPU_USAGE 1957 const CentralTendencyStatistics& stats = mCpu.statistics(); 1958 mCpu.sampleAndEnable(); 1959 unsigned n = stats.n(); 1960 // mCpu.elapsed() is expensive, so don't call it every loop 1961 if ((n & 127) == 1) { 1962 long long elapsed = mCpu.elapsed(); 1963 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1964 double perLoop = elapsed / (double) n; 1965 double perLoop100 = perLoop * 0.01; 1966 double mean = stats.mean(); 1967 double stddev = stats.stddev(); 1968 double minimum = stats.minimum(); 1969 double maximum = stats.maximum(); 1970 mCpu.resetStatistics(); 1971 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1972 elapsed * .000000001, n, perLoop * .000001, 1973 mean * .001, 1974 stddev * .001, 1975 minimum * .001, 1976 maximum * .001, 1977 mean / perLoop100, 1978 stddev / perLoop100, 1979 minimum / perLoop100, 1980 maximum / perLoop100); 1981 } 1982 } 1983#endif 1984}; 1985 1986void AudioFlinger::PlaybackThread::checkSilentMode_l() 1987{ 1988 if (!mMasterMute) { 1989 char value[PROPERTY_VALUE_MAX]; 1990 if (property_get("ro.audio.silent", value, "0") > 0) { 1991 char *endptr; 1992 unsigned long ul = strtoul(value, &endptr, 0); 1993 if (*endptr == '\0' && ul != 0) { 1994 ALOGD("Silence is golden"); 1995 // The setprop command will not allow a property to be changed after 1996 // the first time it is set, so we don't have to worry about un-muting. 1997 setMasterMute_l(true); 1998 } 1999 } 2000 } 2001} 2002 2003bool AudioFlinger::PlaybackThread::threadLoop() 2004{ 2005 Vector< sp<Track> > tracksToRemove; 2006 2007 standbyTime = systemTime(); 2008 2009 // MIXER 2010 nsecs_t lastWarning = 0; 2011if (mType == MIXER) { 2012 longStandbyExit = false; 2013} 2014 2015 // DUPLICATING 2016 // FIXME could this be made local to while loop? 2017 writeFrames = 0; 2018 2019 cacheParameters_l(); 2020 sleepTime = idleSleepTime; 2021 2022if (mType == MIXER) { 2023 sleepTimeShift = 0; 2024} 2025 2026 // MIXER 2027 CpuStats cpuStats; 2028 2029 acquireWakeLock(); 2030 2031 while (!exitPending()) 2032 { 2033if (mType == MIXER) { 2034 cpuStats.sample(); 2035} 2036 2037 Vector< sp<EffectChain> > effectChains; 2038 2039 processConfigEvents(); 2040 2041 { // scope for mLock 2042 2043 Mutex::Autolock _l(mLock); 2044 2045 if (checkForNewParameters_l()) { 2046 cacheParameters_l(); 2047 } 2048 2049 saveOutputTracks(); 2050 2051 // put audio hardware into standby after short delay 2052 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2053 mSuspended > 0)) { 2054 if (!mStandby) { 2055 2056 threadLoop_standby(); 2057 2058 mStandby = true; 2059 mBytesWritten = 0; 2060 } 2061 2062 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2063 // we're about to wait, flush the binder command buffer 2064 IPCThreadState::self()->flushCommands(); 2065 2066 clearOutputTracks(); 2067 2068 if (exitPending()) break; 2069 2070 releaseWakeLock_l(); 2071 // wait until we have something to do... 2072 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2073 mWaitWorkCV.wait(mLock); 2074 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2075 acquireWakeLock_l(); 2076 2077 mPrevMixerStatus = MIXER_IDLE; 2078 2079 checkSilentMode_l(); 2080 2081 standbyTime = systemTime() + standbyDelay; 2082 sleepTime = idleSleepTime; 2083 if (mType == MIXER) { 2084 sleepTimeShift = 0; 2085 } 2086 2087 continue; 2088 } 2089 } 2090 2091 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2092 // Shift in the new status; this could be a queue if it's 2093 // useful to filter the mixer status over several cycles. 2094 mPrevMixerStatus = mMixerStatus; 2095 mMixerStatus = newMixerStatus; 2096 2097 // prevent any changes in effect chain list and in each effect chain 2098 // during mixing and effect process as the audio buffers could be deleted 2099 // or modified if an effect is created or deleted 2100 lockEffectChains_l(effectChains); 2101 } 2102 2103 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2104 threadLoop_mix(); 2105 } else { 2106 threadLoop_sleepTime(); 2107 } 2108 2109 if (mSuspended > 0) { 2110 sleepTime = suspendSleepTimeUs(); 2111 } 2112 2113 // only process effects if we're going to write 2114 if (sleepTime == 0) { 2115 for (size_t i = 0; i < effectChains.size(); i ++) { 2116 effectChains[i]->process_l(); 2117 } 2118 } 2119 2120 // enable changes in effect chain 2121 unlockEffectChains(effectChains); 2122 2123 // sleepTime == 0 means we must write to audio hardware 2124 if (sleepTime == 0) { 2125 2126 threadLoop_write(); 2127 2128if (mType == MIXER) { 2129 // write blocked detection 2130 nsecs_t now = systemTime(); 2131 nsecs_t delta = now - mLastWriteTime; 2132 if (!mStandby && delta > maxPeriod) { 2133 mNumDelayedWrites++; 2134 if ((now - lastWarning) > kWarningThrottleNs) { 2135 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2136 ns2ms(delta), mNumDelayedWrites, this); 2137 lastWarning = now; 2138 } 2139 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2140 // a different threshold. Or completely removed for what it is worth anyway... 2141 if (mStandby) { 2142 longStandbyExit = true; 2143 } 2144 } 2145} 2146 2147 mStandby = false; 2148 } else { 2149 usleep(sleepTime); 2150 } 2151 2152 // finally let go of removed track(s), without the lock held 2153 // since we can't guarantee the destructors won't acquire that 2154 // same lock. 2155 tracksToRemove.clear(); 2156 2157 // FIXME I don't understand the need for this here; 2158 // it was in the original code but maybe the 2159 // assignment in saveOutputTracks() makes this unnecessary? 2160 clearOutputTracks(); 2161 2162 // Effect chains will be actually deleted here if they were removed from 2163 // mEffectChains list during mixing or effects processing 2164 effectChains.clear(); 2165 2166 // FIXME Note that the above .clear() is no longer necessary since effectChains 2167 // is now local to this block, but will keep it for now (at least until merge done). 2168 } 2169 2170if (mType == MIXER || mType == DIRECT) { 2171 // put output stream into standby mode 2172 if (!mStandby) { 2173 mOutput->stream->common.standby(&mOutput->stream->common); 2174 } 2175} 2176if (mType == DUPLICATING) { 2177 // for DuplicatingThread, standby mode is handled by the outputTracks 2178} 2179 2180 releaseWakeLock(); 2181 2182 ALOGV("Thread %p type %d exiting", this, mType); 2183 return false; 2184} 2185 2186// shared by MIXER and DIRECT, overridden by DUPLICATING 2187void AudioFlinger::PlaybackThread::threadLoop_write() 2188{ 2189 // FIXME rewrite to reduce number of system calls 2190 mLastWriteTime = systemTime(); 2191 mInWrite = true; 2192 mBytesWritten += mixBufferSize; 2193 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2194 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2195 mNumWrites++; 2196 mInWrite = false; 2197} 2198 2199// shared by MIXER and DIRECT, overridden by DUPLICATING 2200void AudioFlinger::PlaybackThread::threadLoop_standby() 2201{ 2202 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2203 mOutput->stream->common.standby(&mOutput->stream->common); 2204} 2205 2206void AudioFlinger::MixerThread::threadLoop_mix() 2207{ 2208 // obtain the presentation timestamp of the next output buffer 2209 int64_t pts; 2210 status_t status = INVALID_OPERATION; 2211 2212 if (NULL != mOutput->stream->get_next_write_timestamp) { 2213 status = mOutput->stream->get_next_write_timestamp( 2214 mOutput->stream, &pts); 2215 } 2216 2217 if (status != NO_ERROR) { 2218 pts = AudioBufferProvider::kInvalidPTS; 2219 } 2220 2221 // mix buffers... 2222 mAudioMixer->process(pts); 2223 // increase sleep time progressively when application underrun condition clears. 2224 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2225 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2226 // such that we would underrun the audio HAL. 2227 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2228 sleepTimeShift--; 2229 } 2230 sleepTime = 0; 2231 standbyTime = systemTime() + standbyDelay; 2232 //TODO: delay standby when effects have a tail 2233} 2234 2235void AudioFlinger::MixerThread::threadLoop_sleepTime() 2236{ 2237 // If no tracks are ready, sleep once for the duration of an output 2238 // buffer size, then write 0s to the output 2239 if (sleepTime == 0) { 2240 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2241 sleepTime = activeSleepTime >> sleepTimeShift; 2242 if (sleepTime < kMinThreadSleepTimeUs) { 2243 sleepTime = kMinThreadSleepTimeUs; 2244 } 2245 // reduce sleep time in case of consecutive application underruns to avoid 2246 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2247 // duration we would end up writing less data than needed by the audio HAL if 2248 // the condition persists. 2249 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2250 sleepTimeShift++; 2251 } 2252 } else { 2253 sleepTime = idleSleepTime; 2254 } 2255 } else if (mBytesWritten != 0 || 2256 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2257 memset (mMixBuffer, 0, mixBufferSize); 2258 sleepTime = 0; 2259 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2260 } 2261 // TODO add standby time extension fct of effect tail 2262} 2263 2264// prepareTracks_l() must be called with ThreadBase::mLock held 2265AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2266 Vector< sp<Track> > *tracksToRemove) 2267{ 2268 2269 mixer_state mixerStatus = MIXER_IDLE; 2270 // find out which tracks need to be processed 2271 size_t count = mActiveTracks.size(); 2272 size_t mixedTracks = 0; 2273 size_t tracksWithEffect = 0; 2274 2275 float masterVolume = mMasterVolume; 2276 bool masterMute = mMasterMute; 2277 2278 if (masterMute) { 2279 masterVolume = 0; 2280 } 2281 // Delegate master volume control to effect in output mix effect chain if needed 2282 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2283 if (chain != 0) { 2284 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2285 chain->setVolume_l(&v, &v); 2286 masterVolume = (float)((v + (1 << 23)) >> 24); 2287 chain.clear(); 2288 } 2289 2290 for (size_t i=0 ; i<count ; i++) { 2291 sp<Track> t = mActiveTracks[i].promote(); 2292 if (t == 0) continue; 2293 2294 // this const just means the local variable doesn't change 2295 Track* const track = t.get(); 2296 audio_track_cblk_t* cblk = track->cblk(); 2297 2298 // The first time a track is added we wait 2299 // for all its buffers to be filled before processing it 2300 int name = track->name(); 2301 // make sure that we have enough frames to mix one full buffer. 2302 // enforce this condition only once to enable draining the buffer in case the client 2303 // app does not call stop() and relies on underrun to stop: 2304 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2305 // during last round 2306 uint32_t minFrames = 1; 2307 if (!track->isStopped() && !track->isPausing() && 2308 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2309 if (t->sampleRate() == (int)mSampleRate) { 2310 minFrames = mFrameCount; 2311 } else { 2312 // +1 for rounding and +1 for additional sample needed for interpolation 2313 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2314 // add frames already consumed but not yet released by the resampler 2315 // because cblk->framesReady() will include these frames 2316 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2317 // the minimum track buffer size is normally twice the number of frames necessary 2318 // to fill one buffer and the resampler should not leave more than one buffer worth 2319 // of unreleased frames after each pass, but just in case... 2320 ALOG_ASSERT(minFrames <= cblk->frameCount); 2321 } 2322 } 2323 if ((track->framesReady() >= minFrames) && track->isReady() && 2324 !track->isPaused() && !track->isTerminated()) 2325 { 2326 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2327 2328 mixedTracks++; 2329 2330 // track->mainBuffer() != mMixBuffer means there is an effect chain 2331 // connected to the track 2332 chain.clear(); 2333 if (track->mainBuffer() != mMixBuffer) { 2334 chain = getEffectChain_l(track->sessionId()); 2335 // Delegate volume control to effect in track effect chain if needed 2336 if (chain != 0) { 2337 tracksWithEffect++; 2338 } else { 2339 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2340 name, track->sessionId()); 2341 } 2342 } 2343 2344 2345 int param = AudioMixer::VOLUME; 2346 if (track->mFillingUpStatus == Track::FS_FILLED) { 2347 // no ramp for the first volume setting 2348 track->mFillingUpStatus = Track::FS_ACTIVE; 2349 if (track->mState == TrackBase::RESUMING) { 2350 track->mState = TrackBase::ACTIVE; 2351 param = AudioMixer::RAMP_VOLUME; 2352 } 2353 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2354 } else if (cblk->server != 0) { 2355 // If the track is stopped before the first frame was mixed, 2356 // do not apply ramp 2357 param = AudioMixer::RAMP_VOLUME; 2358 } 2359 2360 // compute volume for this track 2361 uint32_t vl, vr, va; 2362 if (track->isMuted() || track->isPausing() || 2363 mStreamTypes[track->streamType()].mute) { 2364 vl = vr = va = 0; 2365 if (track->isPausing()) { 2366 track->setPaused(); 2367 } 2368 } else { 2369 2370 // read original volumes with volume control 2371 float typeVolume = mStreamTypes[track->streamType()].volume; 2372 float v = masterVolume * typeVolume; 2373 uint32_t vlr = cblk->getVolumeLR(); 2374 vl = vlr & 0xFFFF; 2375 vr = vlr >> 16; 2376 // track volumes come from shared memory, so can't be trusted and must be clamped 2377 if (vl > MAX_GAIN_INT) { 2378 ALOGV("Track left volume out of range: %04X", vl); 2379 vl = MAX_GAIN_INT; 2380 } 2381 if (vr > MAX_GAIN_INT) { 2382 ALOGV("Track right volume out of range: %04X", vr); 2383 vr = MAX_GAIN_INT; 2384 } 2385 // now apply the master volume and stream type volume 2386 vl = (uint32_t)(v * vl) << 12; 2387 vr = (uint32_t)(v * vr) << 12; 2388 // assuming master volume and stream type volume each go up to 1.0, 2389 // vl and vr are now in 8.24 format 2390 2391 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2392 // send level comes from shared memory and so may be corrupt 2393 if (sendLevel > MAX_GAIN_INT) { 2394 ALOGV("Track send level out of range: %04X", sendLevel); 2395 sendLevel = MAX_GAIN_INT; 2396 } 2397 va = (uint32_t)(v * sendLevel); 2398 } 2399 // Delegate volume control to effect in track effect chain if needed 2400 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2401 // Do not ramp volume if volume is controlled by effect 2402 param = AudioMixer::VOLUME; 2403 track->mHasVolumeController = true; 2404 } else { 2405 // force no volume ramp when volume controller was just disabled or removed 2406 // from effect chain to avoid volume spike 2407 if (track->mHasVolumeController) { 2408 param = AudioMixer::VOLUME; 2409 } 2410 track->mHasVolumeController = false; 2411 } 2412 2413 // Convert volumes from 8.24 to 4.12 format 2414 // This additional clamping is needed in case chain->setVolume_l() overshot 2415 vl = (vl + (1 << 11)) >> 12; 2416 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2417 vr = (vr + (1 << 11)) >> 12; 2418 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2419 2420 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2421 2422 // XXX: these things DON'T need to be done each time 2423 mAudioMixer->setBufferProvider(name, track); 2424 mAudioMixer->enable(name); 2425 2426 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2427 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2428 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2429 mAudioMixer->setParameter( 2430 name, 2431 AudioMixer::TRACK, 2432 AudioMixer::FORMAT, (void *)track->format()); 2433 mAudioMixer->setParameter( 2434 name, 2435 AudioMixer::TRACK, 2436 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2437 mAudioMixer->setParameter( 2438 name, 2439 AudioMixer::RESAMPLE, 2440 AudioMixer::SAMPLE_RATE, 2441 (void *)(cblk->sampleRate)); 2442 mAudioMixer->setParameter( 2443 name, 2444 AudioMixer::TRACK, 2445 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2446 mAudioMixer->setParameter( 2447 name, 2448 AudioMixer::TRACK, 2449 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2450 2451 // reset retry count 2452 track->mRetryCount = kMaxTrackRetries; 2453 // If one track is ready, set the mixer ready if: 2454 // - the mixer was not ready during previous round OR 2455 // - no other track is not ready 2456 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2457 mixerStatus != MIXER_TRACKS_ENABLED) { 2458 mixerStatus = MIXER_TRACKS_READY; 2459 } 2460 } else { 2461 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2462 if (track->isStopped()) { 2463 track->reset(); 2464 } 2465 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2466 // We have consumed all the buffers of this track. 2467 // Remove it from the list of active tracks. 2468 tracksToRemove->add(track); 2469 } else { 2470 // No buffers for this track. Give it a few chances to 2471 // fill a buffer, then remove it from active list. 2472 if (--(track->mRetryCount) <= 0) { 2473 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2474 tracksToRemove->add(track); 2475 // indicate to client process that the track was disabled because of underrun 2476 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2477 // If one track is not ready, mark the mixer also not ready if: 2478 // - the mixer was ready during previous round OR 2479 // - no other track is ready 2480 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2481 mixerStatus != MIXER_TRACKS_READY) { 2482 mixerStatus = MIXER_TRACKS_ENABLED; 2483 } 2484 } 2485 mAudioMixer->disable(name); 2486 } 2487 } 2488 2489 // remove all the tracks that need to be... 2490 count = tracksToRemove->size(); 2491 if (CC_UNLIKELY(count)) { 2492 for (size_t i=0 ; i<count ; i++) { 2493 const sp<Track>& track = tracksToRemove->itemAt(i); 2494 mActiveTracks.remove(track); 2495 if (track->mainBuffer() != mMixBuffer) { 2496 chain = getEffectChain_l(track->sessionId()); 2497 if (chain != 0) { 2498 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2499 chain->decActiveTrackCnt(); 2500 } 2501 } 2502 if (track->isTerminated()) { 2503 removeTrack_l(track); 2504 } 2505 } 2506 } 2507 2508 // mix buffer must be cleared if all tracks are connected to an 2509 // effect chain as in this case the mixer will not write to 2510 // mix buffer and track effects will accumulate into it 2511 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2512 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2513 } 2514 2515 return mixerStatus; 2516} 2517 2518/* 2519The derived values that are cached: 2520 - mixBufferSize from frame count * frame size 2521 - activeSleepTime from activeSleepTimeUs() 2522 - idleSleepTime from idleSleepTimeUs() 2523 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2524 - maxPeriod from frame count and sample rate (MIXER only) 2525 2526The parameters that affect these derived values are: 2527 - frame count 2528 - frame size 2529 - sample rate 2530 - device type: A2DP or not 2531 - device latency 2532 - format: PCM or not 2533 - active sleep time 2534 - idle sleep time 2535*/ 2536 2537void AudioFlinger::PlaybackThread::cacheParameters_l() 2538{ 2539 mixBufferSize = mFrameCount * mFrameSize; 2540 activeSleepTime = activeSleepTimeUs(); 2541 idleSleepTime = idleSleepTimeUs(); 2542} 2543 2544void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2545{ 2546 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2547 this, streamType, mTracks.size()); 2548 Mutex::Autolock _l(mLock); 2549 2550 size_t size = mTracks.size(); 2551 for (size_t i = 0; i < size; i++) { 2552 sp<Track> t = mTracks[i]; 2553 if (t->streamType() == streamType) { 2554 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2555 t->mCblk->cv.signal(); 2556 } 2557 } 2558} 2559 2560void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2561{ 2562 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2563 this, streamType, valid); 2564 Mutex::Autolock _l(mLock); 2565 2566 mStreamTypes[streamType].valid = valid; 2567} 2568 2569// getTrackName_l() must be called with ThreadBase::mLock held 2570int AudioFlinger::MixerThread::getTrackName_l() 2571{ 2572 return mAudioMixer->getTrackName(); 2573} 2574 2575// deleteTrackName_l() must be called with ThreadBase::mLock held 2576void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2577{ 2578 ALOGV("remove track (%d) and delete from mixer", name); 2579 mAudioMixer->deleteTrackName(name); 2580} 2581 2582// checkForNewParameters_l() must be called with ThreadBase::mLock held 2583bool AudioFlinger::MixerThread::checkForNewParameters_l() 2584{ 2585 bool reconfig = false; 2586 2587 while (!mNewParameters.isEmpty()) { 2588 status_t status = NO_ERROR; 2589 String8 keyValuePair = mNewParameters[0]; 2590 AudioParameter param = AudioParameter(keyValuePair); 2591 int value; 2592 2593 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2594 reconfig = true; 2595 } 2596 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2597 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2598 status = BAD_VALUE; 2599 } else { 2600 reconfig = true; 2601 } 2602 } 2603 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2604 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2605 status = BAD_VALUE; 2606 } else { 2607 reconfig = true; 2608 } 2609 } 2610 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2611 // do not accept frame count changes if tracks are open as the track buffer 2612 // size depends on frame count and correct behavior would not be guaranteed 2613 // if frame count is changed after track creation 2614 if (!mTracks.isEmpty()) { 2615 status = INVALID_OPERATION; 2616 } else { 2617 reconfig = true; 2618 } 2619 } 2620 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2621#ifdef ADD_BATTERY_DATA 2622 // when changing the audio output device, call addBatteryData to notify 2623 // the change 2624 if ((int)mDevice != value) { 2625 uint32_t params = 0; 2626 // check whether speaker is on 2627 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2628 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2629 } 2630 2631 int deviceWithoutSpeaker 2632 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2633 // check if any other device (except speaker) is on 2634 if (value & deviceWithoutSpeaker ) { 2635 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2636 } 2637 2638 if (params != 0) { 2639 addBatteryData(params); 2640 } 2641 } 2642#endif 2643 2644 // forward device change to effects that have requested to be 2645 // aware of attached audio device. 2646 mDevice = (uint32_t)value; 2647 for (size_t i = 0; i < mEffectChains.size(); i++) { 2648 mEffectChains[i]->setDevice_l(mDevice); 2649 } 2650 } 2651 2652 if (status == NO_ERROR) { 2653 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2654 keyValuePair.string()); 2655 if (!mStandby && status == INVALID_OPERATION) { 2656 mOutput->stream->common.standby(&mOutput->stream->common); 2657 mStandby = true; 2658 mBytesWritten = 0; 2659 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2660 keyValuePair.string()); 2661 } 2662 if (status == NO_ERROR && reconfig) { 2663 delete mAudioMixer; 2664 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2665 mAudioMixer = NULL; 2666 readOutputParameters(); 2667 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2668 for (size_t i = 0; i < mTracks.size() ; i++) { 2669 int name = getTrackName_l(); 2670 if (name < 0) break; 2671 mTracks[i]->mName = name; 2672 // limit track sample rate to 2 x new output sample rate 2673 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2674 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2675 } 2676 } 2677 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2678 } 2679 } 2680 2681 mNewParameters.removeAt(0); 2682 2683 mParamStatus = status; 2684 mParamCond.signal(); 2685 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2686 // already timed out waiting for the status and will never signal the condition. 2687 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2688 } 2689 return reconfig; 2690} 2691 2692status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2693{ 2694 const size_t SIZE = 256; 2695 char buffer[SIZE]; 2696 String8 result; 2697 2698 PlaybackThread::dumpInternals(fd, args); 2699 2700 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2701 result.append(buffer); 2702 write(fd, result.string(), result.size()); 2703 return NO_ERROR; 2704} 2705 2706uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2707{ 2708 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2709} 2710 2711uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2712{ 2713 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2714} 2715 2716void AudioFlinger::MixerThread::cacheParameters_l() 2717{ 2718 PlaybackThread::cacheParameters_l(); 2719 2720 // FIXME: Relaxed timing because of a certain device that can't meet latency 2721 // Should be reduced to 2x after the vendor fixes the driver issue 2722 // increase threshold again due to low power audio mode. The way this warning 2723 // threshold is calculated and its usefulness should be reconsidered anyway. 2724 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2725} 2726 2727// ---------------------------------------------------------------------------- 2728AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2729 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2730 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2731 // mLeftVolFloat, mRightVolFloat 2732 // mLeftVolShort, mRightVolShort 2733{ 2734} 2735 2736AudioFlinger::DirectOutputThread::~DirectOutputThread() 2737{ 2738} 2739 2740AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2741 Vector< sp<Track> > *tracksToRemove 2742) 2743{ 2744 sp<Track> trackToRemove; 2745 2746 mixer_state mixerStatus = MIXER_IDLE; 2747 2748 // find out which tracks need to be processed 2749 if (mActiveTracks.size() != 0) { 2750 sp<Track> t = mActiveTracks[0].promote(); 2751 // The track died recently 2752 if (t == 0) return MIXER_IDLE; 2753 2754 Track* const track = t.get(); 2755 audio_track_cblk_t* cblk = track->cblk(); 2756 2757 // The first time a track is added we wait 2758 // for all its buffers to be filled before processing it 2759 if (cblk->framesReady() && track->isReady() && 2760 !track->isPaused() && !track->isTerminated()) 2761 { 2762 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2763 2764 if (track->mFillingUpStatus == Track::FS_FILLED) { 2765 track->mFillingUpStatus = Track::FS_ACTIVE; 2766 mLeftVolFloat = mRightVolFloat = 0; 2767 mLeftVolShort = mRightVolShort = 0; 2768 if (track->mState == TrackBase::RESUMING) { 2769 track->mState = TrackBase::ACTIVE; 2770 rampVolume = true; 2771 } 2772 } else if (cblk->server != 0) { 2773 // If the track is stopped before the first frame was mixed, 2774 // do not apply ramp 2775 rampVolume = true; 2776 } 2777 // compute volume for this track 2778 float left, right; 2779 if (track->isMuted() || mMasterMute || track->isPausing() || 2780 mStreamTypes[track->streamType()].mute) { 2781 left = right = 0; 2782 if (track->isPausing()) { 2783 track->setPaused(); 2784 } 2785 } else { 2786 float typeVolume = mStreamTypes[track->streamType()].volume; 2787 float v = mMasterVolume * typeVolume; 2788 uint32_t vlr = cblk->getVolumeLR(); 2789 float v_clamped = v * (vlr & 0xFFFF); 2790 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2791 left = v_clamped/MAX_GAIN; 2792 v_clamped = v * (vlr >> 16); 2793 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2794 right = v_clamped/MAX_GAIN; 2795 } 2796 2797 if (left != mLeftVolFloat || right != mRightVolFloat) { 2798 mLeftVolFloat = left; 2799 mRightVolFloat = right; 2800 2801 // If audio HAL implements volume control, 2802 // force software volume to nominal value 2803 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2804 left = 1.0f; 2805 right = 1.0f; 2806 } 2807 2808 // Convert volumes from float to 8.24 2809 uint32_t vl = (uint32_t)(left * (1 << 24)); 2810 uint32_t vr = (uint32_t)(right * (1 << 24)); 2811 2812 // Delegate volume control to effect in track effect chain if needed 2813 // only one effect chain can be present on DirectOutputThread, so if 2814 // there is one, the track is connected to it 2815 if (!mEffectChains.isEmpty()) { 2816 // Do not ramp volume if volume is controlled by effect 2817 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2818 rampVolume = false; 2819 } 2820 } 2821 2822 // Convert volumes from 8.24 to 4.12 format 2823 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2824 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2825 leftVol = (uint16_t)v_clamped; 2826 v_clamped = (vr + (1 << 11)) >> 12; 2827 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2828 rightVol = (uint16_t)v_clamped; 2829 } else { 2830 leftVol = mLeftVolShort; 2831 rightVol = mRightVolShort; 2832 rampVolume = false; 2833 } 2834 2835 // reset retry count 2836 track->mRetryCount = kMaxTrackRetriesDirect; 2837 mActiveTrack = t; 2838 mixerStatus = MIXER_TRACKS_READY; 2839 } else { 2840 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2841 if (track->isStopped()) { 2842 track->reset(); 2843 } 2844 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2845 // We have consumed all the buffers of this track. 2846 // Remove it from the list of active tracks. 2847 trackToRemove = track; 2848 } else { 2849 // No buffers for this track. Give it a few chances to 2850 // fill a buffer, then remove it from active list. 2851 if (--(track->mRetryCount) <= 0) { 2852 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2853 trackToRemove = track; 2854 } else { 2855 mixerStatus = MIXER_TRACKS_ENABLED; 2856 } 2857 } 2858 } 2859 } 2860 2861 // FIXME merge this with similar code for removing multiple tracks 2862 // remove all the tracks that need to be... 2863 if (CC_UNLIKELY(trackToRemove != 0)) { 2864 tracksToRemove->add(trackToRemove); 2865 mActiveTracks.remove(trackToRemove); 2866 if (!mEffectChains.isEmpty()) { 2867 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2868 trackToRemove->sessionId()); 2869 mEffectChains[0]->decActiveTrackCnt(); 2870 } 2871 if (trackToRemove->isTerminated()) { 2872 removeTrack_l(trackToRemove); 2873 } 2874 } 2875 2876 return mixerStatus; 2877} 2878 2879void AudioFlinger::DirectOutputThread::threadLoop_mix() 2880{ 2881 AudioBufferProvider::Buffer buffer; 2882 size_t frameCount = mFrameCount; 2883 int8_t *curBuf = (int8_t *)mMixBuffer; 2884 // output audio to hardware 2885 while (frameCount) { 2886 buffer.frameCount = frameCount; 2887 mActiveTrack->getNextBuffer(&buffer); 2888 if (CC_UNLIKELY(buffer.raw == NULL)) { 2889 memset(curBuf, 0, frameCount * mFrameSize); 2890 break; 2891 } 2892 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2893 frameCount -= buffer.frameCount; 2894 curBuf += buffer.frameCount * mFrameSize; 2895 mActiveTrack->releaseBuffer(&buffer); 2896 } 2897 sleepTime = 0; 2898 standbyTime = systemTime() + standbyDelay; 2899 mActiveTrack.clear(); 2900 2901 // apply volume 2902 2903 // Do not apply volume on compressed audio 2904 if (!audio_is_linear_pcm(mFormat)) { 2905 return; 2906 } 2907 2908 // convert to signed 16 bit before volume calculation 2909 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2910 size_t count = mFrameCount * mChannelCount; 2911 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2912 int16_t *dst = mMixBuffer + count-1; 2913 while (count--) { 2914 *dst-- = (int16_t)(*src--^0x80) << 8; 2915 } 2916 } 2917 2918 frameCount = mFrameCount; 2919 int16_t *out = mMixBuffer; 2920 if (rampVolume) { 2921 if (mChannelCount == 1) { 2922 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2923 int32_t vlInc = d / (int32_t)frameCount; 2924 int32_t vl = ((int32_t)mLeftVolShort << 16); 2925 do { 2926 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2927 out++; 2928 vl += vlInc; 2929 } while (--frameCount); 2930 2931 } else { 2932 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2933 int32_t vlInc = d / (int32_t)frameCount; 2934 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2935 int32_t vrInc = d / (int32_t)frameCount; 2936 int32_t vl = ((int32_t)mLeftVolShort << 16); 2937 int32_t vr = ((int32_t)mRightVolShort << 16); 2938 do { 2939 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2940 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2941 out += 2; 2942 vl += vlInc; 2943 vr += vrInc; 2944 } while (--frameCount); 2945 } 2946 } else { 2947 if (mChannelCount == 1) { 2948 do { 2949 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2950 out++; 2951 } while (--frameCount); 2952 } else { 2953 do { 2954 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2955 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2956 out += 2; 2957 } while (--frameCount); 2958 } 2959 } 2960 2961 // convert back to unsigned 8 bit after volume calculation 2962 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2963 size_t count = mFrameCount * mChannelCount; 2964 int16_t *src = mMixBuffer; 2965 uint8_t *dst = (uint8_t *)mMixBuffer; 2966 while (count--) { 2967 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2968 } 2969 } 2970 2971 mLeftVolShort = leftVol; 2972 mRightVolShort = rightVol; 2973} 2974 2975void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 2976{ 2977 if (sleepTime == 0) { 2978 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2979 sleepTime = activeSleepTime; 2980 } else { 2981 sleepTime = idleSleepTime; 2982 } 2983 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2984 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2985 sleepTime = 0; 2986 } 2987} 2988 2989// getTrackName_l() must be called with ThreadBase::mLock held 2990int AudioFlinger::DirectOutputThread::getTrackName_l() 2991{ 2992 return 0; 2993} 2994 2995// deleteTrackName_l() must be called with ThreadBase::mLock held 2996void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2997{ 2998} 2999 3000// checkForNewParameters_l() must be called with ThreadBase::mLock held 3001bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3002{ 3003 bool reconfig = false; 3004 3005 while (!mNewParameters.isEmpty()) { 3006 status_t status = NO_ERROR; 3007 String8 keyValuePair = mNewParameters[0]; 3008 AudioParameter param = AudioParameter(keyValuePair); 3009 int value; 3010 3011 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3012 // do not accept frame count changes if tracks are open as the track buffer 3013 // size depends on frame count and correct behavior would not be garantied 3014 // if frame count is changed after track creation 3015 if (!mTracks.isEmpty()) { 3016 status = INVALID_OPERATION; 3017 } else { 3018 reconfig = true; 3019 } 3020 } 3021 if (status == NO_ERROR) { 3022 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3023 keyValuePair.string()); 3024 if (!mStandby && status == INVALID_OPERATION) { 3025 mOutput->stream->common.standby(&mOutput->stream->common); 3026 mStandby = true; 3027 mBytesWritten = 0; 3028 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3029 keyValuePair.string()); 3030 } 3031 if (status == NO_ERROR && reconfig) { 3032 readOutputParameters(); 3033 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3034 } 3035 } 3036 3037 mNewParameters.removeAt(0); 3038 3039 mParamStatus = status; 3040 mParamCond.signal(); 3041 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3042 // already timed out waiting for the status and will never signal the condition. 3043 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3044 } 3045 return reconfig; 3046} 3047 3048uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3049{ 3050 uint32_t time; 3051 if (audio_is_linear_pcm(mFormat)) { 3052 time = PlaybackThread::activeSleepTimeUs(); 3053 } else { 3054 time = 10000; 3055 } 3056 return time; 3057} 3058 3059uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3060{ 3061 uint32_t time; 3062 if (audio_is_linear_pcm(mFormat)) { 3063 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3064 } else { 3065 time = 10000; 3066 } 3067 return time; 3068} 3069 3070uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3071{ 3072 uint32_t time; 3073 if (audio_is_linear_pcm(mFormat)) { 3074 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3075 } else { 3076 time = 10000; 3077 } 3078 return time; 3079} 3080 3081void AudioFlinger::DirectOutputThread::cacheParameters_l() 3082{ 3083 PlaybackThread::cacheParameters_l(); 3084 3085 // use shorter standby delay as on normal output to release 3086 // hardware resources as soon as possible 3087 standbyDelay = microseconds(activeSleepTime*2); 3088} 3089 3090// ---------------------------------------------------------------------------- 3091 3092AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3093 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3094 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3095 mWaitTimeMs(UINT_MAX) 3096{ 3097 addOutputTrack(mainThread); 3098} 3099 3100AudioFlinger::DuplicatingThread::~DuplicatingThread() 3101{ 3102 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3103 mOutputTracks[i]->destroy(); 3104 } 3105} 3106 3107void AudioFlinger::DuplicatingThread::threadLoop_mix() 3108{ 3109 // mix buffers... 3110 if (outputsReady(outputTracks)) { 3111 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3112 } else { 3113 memset(mMixBuffer, 0, mixBufferSize); 3114 } 3115 sleepTime = 0; 3116 writeFrames = mFrameCount; 3117} 3118 3119void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3120{ 3121 if (sleepTime == 0) { 3122 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3123 sleepTime = activeSleepTime; 3124 } else { 3125 sleepTime = idleSleepTime; 3126 } 3127 } else if (mBytesWritten != 0) { 3128 // flush remaining overflow buffers in output tracks 3129 for (size_t i = 0; i < outputTracks.size(); i++) { 3130 if (outputTracks[i]->isActive()) { 3131 sleepTime = 0; 3132 writeFrames = 0; 3133 memset(mMixBuffer, 0, mixBufferSize); 3134 break; 3135 } 3136 } 3137 } 3138} 3139 3140void AudioFlinger::DuplicatingThread::threadLoop_write() 3141{ 3142 standbyTime = systemTime() + standbyDelay; 3143 for (size_t i = 0; i < outputTracks.size(); i++) { 3144 outputTracks[i]->write(mMixBuffer, writeFrames); 3145 } 3146 mBytesWritten += mixBufferSize; 3147} 3148 3149void AudioFlinger::DuplicatingThread::threadLoop_standby() 3150{ 3151 // DuplicatingThread implements standby by stopping all tracks 3152 for (size_t i = 0; i < outputTracks.size(); i++) { 3153 outputTracks[i]->stop(); 3154 } 3155} 3156 3157void AudioFlinger::DuplicatingThread::saveOutputTracks() 3158{ 3159 outputTracks = mOutputTracks; 3160} 3161 3162void AudioFlinger::DuplicatingThread::clearOutputTracks() 3163{ 3164 outputTracks.clear(); 3165} 3166 3167void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3168{ 3169 Mutex::Autolock _l(mLock); 3170 // FIXME explain this formula 3171 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3172 OutputTrack *outputTrack = new OutputTrack(thread, 3173 this, 3174 mSampleRate, 3175 mFormat, 3176 mChannelMask, 3177 frameCount); 3178 if (outputTrack->cblk() != NULL) { 3179 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3180 mOutputTracks.add(outputTrack); 3181 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3182 updateWaitTime_l(); 3183 } 3184} 3185 3186void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3187{ 3188 Mutex::Autolock _l(mLock); 3189 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3190 if (mOutputTracks[i]->thread() == thread) { 3191 mOutputTracks[i]->destroy(); 3192 mOutputTracks.removeAt(i); 3193 updateWaitTime_l(); 3194 return; 3195 } 3196 } 3197 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3198} 3199 3200// caller must hold mLock 3201void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3202{ 3203 mWaitTimeMs = UINT_MAX; 3204 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3205 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3206 if (strong != 0) { 3207 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3208 if (waitTimeMs < mWaitTimeMs) { 3209 mWaitTimeMs = waitTimeMs; 3210 } 3211 } 3212 } 3213} 3214 3215 3216bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3217{ 3218 for (size_t i = 0; i < outputTracks.size(); i++) { 3219 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3220 if (thread == 0) { 3221 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3222 return false; 3223 } 3224 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3225 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3226 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3227 return false; 3228 } 3229 } 3230 return true; 3231} 3232 3233uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3234{ 3235 return (mWaitTimeMs * 1000) / 2; 3236} 3237 3238void AudioFlinger::DuplicatingThread::cacheParameters_l() 3239{ 3240 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3241 updateWaitTime_l(); 3242 3243 MixerThread::cacheParameters_l(); 3244} 3245 3246// ---------------------------------------------------------------------------- 3247 3248// TrackBase constructor must be called with AudioFlinger::mLock held 3249AudioFlinger::ThreadBase::TrackBase::TrackBase( 3250 ThreadBase *thread, 3251 const sp<Client>& client, 3252 uint32_t sampleRate, 3253 audio_format_t format, 3254 uint32_t channelMask, 3255 int frameCount, 3256 const sp<IMemory>& sharedBuffer, 3257 int sessionId) 3258 : RefBase(), 3259 mThread(thread), 3260 mClient(client), 3261 mCblk(NULL), 3262 // mBuffer 3263 // mBufferEnd 3264 mFrameCount(0), 3265 mState(IDLE), 3266 mFormat(format), 3267 mStepServerFailed(false), 3268 mSessionId(sessionId) 3269 // mChannelCount 3270 // mChannelMask 3271{ 3272 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3273 3274 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3275 size_t size = sizeof(audio_track_cblk_t); 3276 uint8_t channelCount = popcount(channelMask); 3277 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3278 if (sharedBuffer == 0) { 3279 size += bufferSize; 3280 } 3281 3282 if (client != NULL) { 3283 mCblkMemory = client->heap()->allocate(size); 3284 if (mCblkMemory != 0) { 3285 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3286 if (mCblk != NULL) { // construct the shared structure in-place. 3287 new(mCblk) audio_track_cblk_t(); 3288 // clear all buffers 3289 mCblk->frameCount = frameCount; 3290 mCblk->sampleRate = sampleRate; 3291 mChannelCount = channelCount; 3292 mChannelMask = channelMask; 3293 if (sharedBuffer == 0) { 3294 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3295 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3296 // Force underrun condition to avoid false underrun callback until first data is 3297 // written to buffer (other flags are cleared) 3298 mCblk->flags = CBLK_UNDERRUN_ON; 3299 } else { 3300 mBuffer = sharedBuffer->pointer(); 3301 } 3302 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3303 } 3304 } else { 3305 ALOGE("not enough memory for AudioTrack size=%u", size); 3306 client->heap()->dump("AudioTrack"); 3307 return; 3308 } 3309 } else { 3310 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3311 // construct the shared structure in-place. 3312 new(mCblk) audio_track_cblk_t(); 3313 // clear all buffers 3314 mCblk->frameCount = frameCount; 3315 mCblk->sampleRate = sampleRate; 3316 mChannelCount = channelCount; 3317 mChannelMask = channelMask; 3318 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3319 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3320 // Force underrun condition to avoid false underrun callback until first data is 3321 // written to buffer (other flags are cleared) 3322 mCblk->flags = CBLK_UNDERRUN_ON; 3323 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3324 } 3325} 3326 3327AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3328{ 3329 if (mCblk != NULL) { 3330 if (mClient == 0) { 3331 delete mCblk; 3332 } else { 3333 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3334 } 3335 } 3336 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3337 if (mClient != 0) { 3338 // Client destructor must run with AudioFlinger mutex locked 3339 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3340 // If the client's reference count drops to zero, the associated destructor 3341 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3342 // relying on the automatic clear() at end of scope. 3343 mClient.clear(); 3344 } 3345} 3346 3347// AudioBufferProvider interface 3348// getNextBuffer() = 0; 3349// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3350void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3351{ 3352 buffer->raw = NULL; 3353 mFrameCount = buffer->frameCount; 3354 (void) step(); // ignore return value of step() 3355 buffer->frameCount = 0; 3356} 3357 3358bool AudioFlinger::ThreadBase::TrackBase::step() { 3359 bool result; 3360 audio_track_cblk_t* cblk = this->cblk(); 3361 3362 result = cblk->stepServer(mFrameCount); 3363 if (!result) { 3364 ALOGV("stepServer failed acquiring cblk mutex"); 3365 mStepServerFailed = true; 3366 } 3367 return result; 3368} 3369 3370void AudioFlinger::ThreadBase::TrackBase::reset() { 3371 audio_track_cblk_t* cblk = this->cblk(); 3372 3373 cblk->user = 0; 3374 cblk->server = 0; 3375 cblk->userBase = 0; 3376 cblk->serverBase = 0; 3377 mStepServerFailed = false; 3378 ALOGV("TrackBase::reset"); 3379} 3380 3381int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3382 return (int)mCblk->sampleRate; 3383} 3384 3385void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3386 audio_track_cblk_t* cblk = this->cblk(); 3387 size_t frameSize = cblk->frameSize; 3388 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3389 int8_t *bufferEnd = bufferStart + frames * frameSize; 3390 3391 // Check validity of returned pointer in case the track control block would have been corrupted. 3392 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3393 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3394 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3395 server %d, serverBase %d, user %d, userBase %d", 3396 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3397 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3398 return NULL; 3399 } 3400 3401 return bufferStart; 3402} 3403 3404// ---------------------------------------------------------------------------- 3405 3406// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3407AudioFlinger::PlaybackThread::Track::Track( 3408 PlaybackThread *thread, 3409 const sp<Client>& client, 3410 audio_stream_type_t streamType, 3411 uint32_t sampleRate, 3412 audio_format_t format, 3413 uint32_t channelMask, 3414 int frameCount, 3415 const sp<IMemory>& sharedBuffer, 3416 int sessionId) 3417 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3418 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3419 mAuxEffectId(0), mHasVolumeController(false) 3420{ 3421 if (mCblk != NULL) { 3422 if (thread != NULL) { 3423 mName = thread->getTrackName_l(); 3424 mMainBuffer = thread->mixBuffer(); 3425 } 3426 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3427 if (mName < 0) { 3428 ALOGE("no more track names available"); 3429 } 3430 mStreamType = streamType; 3431 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3432 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3433 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3434 } 3435} 3436 3437AudioFlinger::PlaybackThread::Track::~Track() 3438{ 3439 ALOGV("PlaybackThread::Track destructor"); 3440 sp<ThreadBase> thread = mThread.promote(); 3441 if (thread != 0) { 3442 Mutex::Autolock _l(thread->mLock); 3443 mState = TERMINATED; 3444 } 3445} 3446 3447void AudioFlinger::PlaybackThread::Track::destroy() 3448{ 3449 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3450 // by removing it from mTracks vector, so there is a risk that this Tracks's 3451 // destructor is called. As the destructor needs to lock mLock, 3452 // we must acquire a strong reference on this Track before locking mLock 3453 // here so that the destructor is called only when exiting this function. 3454 // On the other hand, as long as Track::destroy() is only called by 3455 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3456 // this Track with its member mTrack. 3457 sp<Track> keep(this); 3458 { // scope for mLock 3459 sp<ThreadBase> thread = mThread.promote(); 3460 if (thread != 0) { 3461 if (!isOutputTrack()) { 3462 if (mState == ACTIVE || mState == RESUMING) { 3463 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3464 3465#ifdef ADD_BATTERY_DATA 3466 // to track the speaker usage 3467 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3468#endif 3469 } 3470 AudioSystem::releaseOutput(thread->id()); 3471 } 3472 Mutex::Autolock _l(thread->mLock); 3473 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3474 playbackThread->destroyTrack_l(this); 3475 } 3476 } 3477} 3478 3479void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3480{ 3481 uint32_t vlr = mCblk->getVolumeLR(); 3482 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3483 mName - AudioMixer::TRACK0, 3484 (mClient == 0) ? getpid_cached : mClient->pid(), 3485 mStreamType, 3486 mFormat, 3487 mChannelMask, 3488 mSessionId, 3489 mFrameCount, 3490 mState, 3491 mMute, 3492 mFillingUpStatus, 3493 mCblk->sampleRate, 3494 vlr & 0xFFFF, 3495 vlr >> 16, 3496 mCblk->server, 3497 mCblk->user, 3498 (int)mMainBuffer, 3499 (int)mAuxBuffer); 3500} 3501 3502// AudioBufferProvider interface 3503status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3504 AudioBufferProvider::Buffer* buffer, int64_t pts) 3505{ 3506 audio_track_cblk_t* cblk = this->cblk(); 3507 uint32_t framesReady; 3508 uint32_t framesReq = buffer->frameCount; 3509 3510 // Check if last stepServer failed, try to step now 3511 if (mStepServerFailed) { 3512 if (!step()) goto getNextBuffer_exit; 3513 ALOGV("stepServer recovered"); 3514 mStepServerFailed = false; 3515 } 3516 3517 framesReady = cblk->framesReady(); 3518 3519 if (CC_LIKELY(framesReady)) { 3520 uint32_t s = cblk->server; 3521 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3522 3523 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3524 if (framesReq > framesReady) { 3525 framesReq = framesReady; 3526 } 3527 if (s + framesReq > bufferEnd) { 3528 framesReq = bufferEnd - s; 3529 } 3530 3531 buffer->raw = getBuffer(s, framesReq); 3532 if (buffer->raw == NULL) goto getNextBuffer_exit; 3533 3534 buffer->frameCount = framesReq; 3535 return NO_ERROR; 3536 } 3537 3538getNextBuffer_exit: 3539 buffer->raw = NULL; 3540 buffer->frameCount = 0; 3541 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3542 return NOT_ENOUGH_DATA; 3543} 3544 3545uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3546 return mCblk->framesReady(); 3547} 3548 3549bool AudioFlinger::PlaybackThread::Track::isReady() const { 3550 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3551 3552 if (framesReady() >= mCblk->frameCount || 3553 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3554 mFillingUpStatus = FS_FILLED; 3555 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3556 return true; 3557 } 3558 return false; 3559} 3560 3561status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3562{ 3563 status_t status = NO_ERROR; 3564 ALOGV("start(%d), calling pid %d session %d tid %d", 3565 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3566 sp<ThreadBase> thread = mThread.promote(); 3567 if (thread != 0) { 3568 Mutex::Autolock _l(thread->mLock); 3569 track_state state = mState; 3570 // here the track could be either new, or restarted 3571 // in both cases "unstop" the track 3572 if (mState == PAUSED) { 3573 mState = TrackBase::RESUMING; 3574 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3575 } else { 3576 mState = TrackBase::ACTIVE; 3577 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3578 } 3579 3580 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3581 thread->mLock.unlock(); 3582 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3583 thread->mLock.lock(); 3584 3585#ifdef ADD_BATTERY_DATA 3586 // to track the speaker usage 3587 if (status == NO_ERROR) { 3588 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3589 } 3590#endif 3591 } 3592 if (status == NO_ERROR) { 3593 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3594 playbackThread->addTrack_l(this); 3595 } else { 3596 mState = state; 3597 } 3598 } else { 3599 status = BAD_VALUE; 3600 } 3601 return status; 3602} 3603 3604void AudioFlinger::PlaybackThread::Track::stop() 3605{ 3606 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3607 sp<ThreadBase> thread = mThread.promote(); 3608 if (thread != 0) { 3609 Mutex::Autolock _l(thread->mLock); 3610 track_state state = mState; 3611 if (mState > STOPPED) { 3612 mState = STOPPED; 3613 // If the track is not active (PAUSED and buffers full), flush buffers 3614 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3615 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3616 reset(); 3617 } 3618 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3619 } 3620 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3621 thread->mLock.unlock(); 3622 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3623 thread->mLock.lock(); 3624 3625#ifdef ADD_BATTERY_DATA 3626 // to track the speaker usage 3627 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3628#endif 3629 } 3630 } 3631} 3632 3633void AudioFlinger::PlaybackThread::Track::pause() 3634{ 3635 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3636 sp<ThreadBase> thread = mThread.promote(); 3637 if (thread != 0) { 3638 Mutex::Autolock _l(thread->mLock); 3639 if (mState == ACTIVE || mState == RESUMING) { 3640 mState = PAUSING; 3641 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3642 if (!isOutputTrack()) { 3643 thread->mLock.unlock(); 3644 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3645 thread->mLock.lock(); 3646 3647#ifdef ADD_BATTERY_DATA 3648 // to track the speaker usage 3649 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3650#endif 3651 } 3652 } 3653 } 3654} 3655 3656void AudioFlinger::PlaybackThread::Track::flush() 3657{ 3658 ALOGV("flush(%d)", mName); 3659 sp<ThreadBase> thread = mThread.promote(); 3660 if (thread != 0) { 3661 Mutex::Autolock _l(thread->mLock); 3662 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3663 return; 3664 } 3665 // No point remaining in PAUSED state after a flush => go to 3666 // STOPPED state 3667 mState = STOPPED; 3668 3669 // do not reset the track if it is still in the process of being stopped or paused. 3670 // this will be done by prepareTracks_l() when the track is stopped. 3671 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3672 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3673 reset(); 3674 } 3675 } 3676} 3677 3678void AudioFlinger::PlaybackThread::Track::reset() 3679{ 3680 // Do not reset twice to avoid discarding data written just after a flush and before 3681 // the audioflinger thread detects the track is stopped. 3682 if (!mResetDone) { 3683 TrackBase::reset(); 3684 // Force underrun condition to avoid false underrun callback until first data is 3685 // written to buffer 3686 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3687 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3688 mFillingUpStatus = FS_FILLING; 3689 mResetDone = true; 3690 } 3691} 3692 3693void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3694{ 3695 mMute = muted; 3696} 3697 3698status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3699{ 3700 status_t status = DEAD_OBJECT; 3701 sp<ThreadBase> thread = mThread.promote(); 3702 if (thread != 0) { 3703 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3704 status = playbackThread->attachAuxEffect(this, EffectId); 3705 } 3706 return status; 3707} 3708 3709void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3710{ 3711 mAuxEffectId = EffectId; 3712 mAuxBuffer = buffer; 3713} 3714 3715// timed audio tracks 3716 3717sp<AudioFlinger::PlaybackThread::TimedTrack> 3718AudioFlinger::PlaybackThread::TimedTrack::create( 3719 PlaybackThread *thread, 3720 const sp<Client>& client, 3721 audio_stream_type_t streamType, 3722 uint32_t sampleRate, 3723 audio_format_t format, 3724 uint32_t channelMask, 3725 int frameCount, 3726 const sp<IMemory>& sharedBuffer, 3727 int sessionId) { 3728 if (!client->reserveTimedTrack()) 3729 return NULL; 3730 3731 sp<TimedTrack> track = new TimedTrack( 3732 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3733 sharedBuffer, sessionId); 3734 3735 if (track == NULL) { 3736 client->releaseTimedTrack(); 3737 return NULL; 3738 } 3739 3740 return track; 3741} 3742 3743AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3744 PlaybackThread *thread, 3745 const sp<Client>& client, 3746 audio_stream_type_t streamType, 3747 uint32_t sampleRate, 3748 audio_format_t format, 3749 uint32_t channelMask, 3750 int frameCount, 3751 const sp<IMemory>& sharedBuffer, 3752 int sessionId) 3753 : Track(thread, client, streamType, sampleRate, format, channelMask, 3754 frameCount, sharedBuffer, sessionId), 3755 mTimedSilenceBuffer(NULL), 3756 mTimedSilenceBufferSize(0), 3757 mTimedAudioOutputOnTime(false), 3758 mMediaTimeTransformValid(false) 3759{ 3760 LocalClock lc; 3761 mLocalTimeFreq = lc.getLocalFreq(); 3762 3763 mLocalTimeToSampleTransform.a_zero = 0; 3764 mLocalTimeToSampleTransform.b_zero = 0; 3765 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3766 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3767 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3768 &mLocalTimeToSampleTransform.a_to_b_denom); 3769} 3770 3771AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3772 mClient->releaseTimedTrack(); 3773 delete [] mTimedSilenceBuffer; 3774} 3775 3776status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3777 size_t size, sp<IMemory>* buffer) { 3778 3779 Mutex::Autolock _l(mTimedBufferQueueLock); 3780 3781 trimTimedBufferQueue_l(); 3782 3783 // lazily initialize the shared memory heap for timed buffers 3784 if (mTimedMemoryDealer == NULL) { 3785 const int kTimedBufferHeapSize = 512 << 10; 3786 3787 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3788 "AudioFlingerTimed"); 3789 if (mTimedMemoryDealer == NULL) 3790 return NO_MEMORY; 3791 } 3792 3793 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3794 if (newBuffer == NULL) { 3795 newBuffer = mTimedMemoryDealer->allocate(size); 3796 if (newBuffer == NULL) 3797 return NO_MEMORY; 3798 } 3799 3800 *buffer = newBuffer; 3801 return NO_ERROR; 3802} 3803 3804// caller must hold mTimedBufferQueueLock 3805void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3806 int64_t mediaTimeNow; 3807 { 3808 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3809 if (!mMediaTimeTransformValid) 3810 return; 3811 3812 int64_t targetTimeNow; 3813 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3814 ? mCCHelper.getCommonTime(&targetTimeNow) 3815 : mCCHelper.getLocalTime(&targetTimeNow); 3816 3817 if (OK != res) 3818 return; 3819 3820 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3821 &mediaTimeNow)) { 3822 return; 3823 } 3824 } 3825 3826 size_t trimIndex; 3827 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3828 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3829 break; 3830 } 3831 3832 if (trimIndex) { 3833 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3834 } 3835} 3836 3837status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3838 const sp<IMemory>& buffer, int64_t pts) { 3839 3840 { 3841 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3842 if (!mMediaTimeTransformValid) 3843 return INVALID_OPERATION; 3844 } 3845 3846 Mutex::Autolock _l(mTimedBufferQueueLock); 3847 3848 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3849 3850 return NO_ERROR; 3851} 3852 3853status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3854 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3855 3856 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3857 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3858 target); 3859 3860 if (!(target == TimedAudioTrack::LOCAL_TIME || 3861 target == TimedAudioTrack::COMMON_TIME)) { 3862 return BAD_VALUE; 3863 } 3864 3865 Mutex::Autolock lock(mMediaTimeTransformLock); 3866 mMediaTimeTransform = xform; 3867 mMediaTimeTransformTarget = target; 3868 mMediaTimeTransformValid = true; 3869 3870 return NO_ERROR; 3871} 3872 3873#define min(a, b) ((a) < (b) ? (a) : (b)) 3874 3875// implementation of getNextBuffer for tracks whose buffers have timestamps 3876status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3877 AudioBufferProvider::Buffer* buffer, int64_t pts) 3878{ 3879 if (pts == AudioBufferProvider::kInvalidPTS) { 3880 buffer->raw = 0; 3881 buffer->frameCount = 0; 3882 return INVALID_OPERATION; 3883 } 3884 3885 Mutex::Autolock _l(mTimedBufferQueueLock); 3886 3887 while (true) { 3888 3889 // if we have no timed buffers, then fail 3890 if (mTimedBufferQueue.isEmpty()) { 3891 buffer->raw = 0; 3892 buffer->frameCount = 0; 3893 return NOT_ENOUGH_DATA; 3894 } 3895 3896 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3897 3898 // calculate the PTS of the head of the timed buffer queue expressed in 3899 // local time 3900 int64_t headLocalPTS; 3901 { 3902 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3903 3904 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 3905 3906 if (mMediaTimeTransform.a_to_b_denom == 0) { 3907 // the transform represents a pause, so yield silence 3908 timedYieldSilence(buffer->frameCount, buffer); 3909 return NO_ERROR; 3910 } 3911 3912 int64_t transformedPTS; 3913 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3914 &transformedPTS)) { 3915 // the transform failed. this shouldn't happen, but if it does 3916 // then just drop this buffer 3917 ALOGW("timedGetNextBuffer transform failed"); 3918 buffer->raw = 0; 3919 buffer->frameCount = 0; 3920 mTimedBufferQueue.removeAt(0); 3921 return NO_ERROR; 3922 } 3923 3924 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3925 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3926 &headLocalPTS)) { 3927 buffer->raw = 0; 3928 buffer->frameCount = 0; 3929 return INVALID_OPERATION; 3930 } 3931 } else { 3932 headLocalPTS = transformedPTS; 3933 } 3934 } 3935 3936 // adjust the head buffer's PTS to reflect the portion of the head buffer 3937 // that has already been consumed 3938 int64_t effectivePTS = headLocalPTS + 3939 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3940 3941 // Calculate the delta in samples between the head of the input buffer 3942 // queue and the start of the next output buffer that will be written. 3943 // If the transformation fails because of over or underflow, it means 3944 // that the sample's position in the output stream is so far out of 3945 // whack that it should just be dropped. 3946 int64_t sampleDelta; 3947 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3948 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3949 mTimedBufferQueue.removeAt(0); 3950 continue; 3951 } 3952 if (!mLocalTimeToSampleTransform.doForwardTransform( 3953 (effectivePTS - pts) << 32, &sampleDelta)) { 3954 ALOGV("*** too late during sample rate transform: dropped buffer"); 3955 mTimedBufferQueue.removeAt(0); 3956 continue; 3957 } 3958 3959 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3960 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3961 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3962 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3963 3964 // if the delta between the ideal placement for the next input sample and 3965 // the current output position is within this threshold, then we will 3966 // concatenate the next input samples to the previous output 3967 const int64_t kSampleContinuityThreshold = 3968 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3969 3970 // if this is the first buffer of audio that we're emitting from this track 3971 // then it should be almost exactly on time. 3972 const int64_t kSampleStartupThreshold = 1LL << 32; 3973 3974 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3975 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3976 // the next input is close enough to being on time, so concatenate it 3977 // with the last output 3978 timedYieldSamples(buffer); 3979 3980 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3981 return NO_ERROR; 3982 } else if (sampleDelta > 0) { 3983 // the gap between the current output position and the proper start of 3984 // the next input sample is too big, so fill it with silence 3985 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3986 3987 timedYieldSilence(framesUntilNextInput, buffer); 3988 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3989 return NO_ERROR; 3990 } else { 3991 // the next input sample is late 3992 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3993 size_t onTimeSamplePosition = 3994 head.position() + lateFrames * mCblk->frameSize; 3995 3996 if (onTimeSamplePosition > head.buffer()->size()) { 3997 // all the remaining samples in the head are too late, so 3998 // drop it and move on 3999 ALOGV("*** too late: dropped buffer"); 4000 mTimedBufferQueue.removeAt(0); 4001 continue; 4002 } else { 4003 // skip over the late samples 4004 head.setPosition(onTimeSamplePosition); 4005 4006 // yield the available samples 4007 timedYieldSamples(buffer); 4008 4009 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4010 return NO_ERROR; 4011 } 4012 } 4013 } 4014} 4015 4016// Yield samples from the timed buffer queue head up to the given output 4017// buffer's capacity. 4018// 4019// Caller must hold mTimedBufferQueueLock 4020void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4021 AudioBufferProvider::Buffer* buffer) { 4022 4023 const TimedBuffer& head = mTimedBufferQueue[0]; 4024 4025 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4026 head.position()); 4027 4028 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4029 mCblk->frameSize); 4030 size_t framesRequested = buffer->frameCount; 4031 buffer->frameCount = min(framesLeftInHead, framesRequested); 4032 4033 mTimedAudioOutputOnTime = true; 4034} 4035 4036// Yield samples of silence up to the given output buffer's capacity 4037// 4038// Caller must hold mTimedBufferQueueLock 4039void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4040 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4041 4042 // lazily allocate a buffer filled with silence 4043 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4044 delete [] mTimedSilenceBuffer; 4045 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4046 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4047 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4048 } 4049 4050 buffer->raw = mTimedSilenceBuffer; 4051 size_t framesRequested = buffer->frameCount; 4052 buffer->frameCount = min(numFrames, framesRequested); 4053 4054 mTimedAudioOutputOnTime = false; 4055} 4056 4057// AudioBufferProvider interface 4058void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4059 AudioBufferProvider::Buffer* buffer) { 4060 4061 Mutex::Autolock _l(mTimedBufferQueueLock); 4062 4063 // If the buffer which was just released is part of the buffer at the head 4064 // of the queue, be sure to update the amt of the buffer which has been 4065 // consumed. If the buffer being returned is not part of the head of the 4066 // queue, its either because the buffer is part of the silence buffer, or 4067 // because the head of the timed queue was trimmed after the mixer called 4068 // getNextBuffer but before the mixer called releaseBuffer. 4069 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4070 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4071 4072 void* start = head.buffer()->pointer(); 4073 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4074 4075 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4076 head.setPosition(head.position() + 4077 (buffer->frameCount * mCblk->frameSize)); 4078 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4079 mTimedBufferQueue.removeAt(0); 4080 } 4081 } 4082 } 4083 4084 buffer->raw = 0; 4085 buffer->frameCount = 0; 4086} 4087 4088uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4089 Mutex::Autolock _l(mTimedBufferQueueLock); 4090 4091 uint32_t frames = 0; 4092 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4093 const TimedBuffer& tb = mTimedBufferQueue[i]; 4094 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4095 } 4096 4097 return frames; 4098} 4099 4100AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4101 : mPTS(0), mPosition(0) {} 4102 4103AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4104 const sp<IMemory>& buffer, int64_t pts) 4105 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4106 4107// ---------------------------------------------------------------------------- 4108 4109// RecordTrack constructor must be called with AudioFlinger::mLock held 4110AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4111 RecordThread *thread, 4112 const sp<Client>& client, 4113 uint32_t sampleRate, 4114 audio_format_t format, 4115 uint32_t channelMask, 4116 int frameCount, 4117 int sessionId) 4118 : TrackBase(thread, client, sampleRate, format, 4119 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4120 mOverflow(false) 4121{ 4122 if (mCblk != NULL) { 4123 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4124 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4125 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4126 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4127 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4128 } else { 4129 mCblk->frameSize = sizeof(int8_t); 4130 } 4131 } 4132} 4133 4134AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4135{ 4136 sp<ThreadBase> thread = mThread.promote(); 4137 if (thread != 0) { 4138 AudioSystem::releaseInput(thread->id()); 4139 } 4140} 4141 4142// AudioBufferProvider interface 4143status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4144{ 4145 audio_track_cblk_t* cblk = this->cblk(); 4146 uint32_t framesAvail; 4147 uint32_t framesReq = buffer->frameCount; 4148 4149 // Check if last stepServer failed, try to step now 4150 if (mStepServerFailed) { 4151 if (!step()) goto getNextBuffer_exit; 4152 ALOGV("stepServer recovered"); 4153 mStepServerFailed = false; 4154 } 4155 4156 framesAvail = cblk->framesAvailable_l(); 4157 4158 if (CC_LIKELY(framesAvail)) { 4159 uint32_t s = cblk->server; 4160 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4161 4162 if (framesReq > framesAvail) { 4163 framesReq = framesAvail; 4164 } 4165 if (s + framesReq > bufferEnd) { 4166 framesReq = bufferEnd - s; 4167 } 4168 4169 buffer->raw = getBuffer(s, framesReq); 4170 if (buffer->raw == NULL) goto getNextBuffer_exit; 4171 4172 buffer->frameCount = framesReq; 4173 return NO_ERROR; 4174 } 4175 4176getNextBuffer_exit: 4177 buffer->raw = NULL; 4178 buffer->frameCount = 0; 4179 return NOT_ENOUGH_DATA; 4180} 4181 4182status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4183{ 4184 sp<ThreadBase> thread = mThread.promote(); 4185 if (thread != 0) { 4186 RecordThread *recordThread = (RecordThread *)thread.get(); 4187 return recordThread->start(this, tid); 4188 } else { 4189 return BAD_VALUE; 4190 } 4191} 4192 4193void AudioFlinger::RecordThread::RecordTrack::stop() 4194{ 4195 sp<ThreadBase> thread = mThread.promote(); 4196 if (thread != 0) { 4197 RecordThread *recordThread = (RecordThread *)thread.get(); 4198 recordThread->stop(this); 4199 TrackBase::reset(); 4200 // Force overerrun condition to avoid false overrun callback until first data is 4201 // read from buffer 4202 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4203 } 4204} 4205 4206void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4207{ 4208 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4209 (mClient == 0) ? getpid_cached : mClient->pid(), 4210 mFormat, 4211 mChannelMask, 4212 mSessionId, 4213 mFrameCount, 4214 mState, 4215 mCblk->sampleRate, 4216 mCblk->server, 4217 mCblk->user); 4218} 4219 4220 4221// ---------------------------------------------------------------------------- 4222 4223AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4224 PlaybackThread *playbackThread, 4225 DuplicatingThread *sourceThread, 4226 uint32_t sampleRate, 4227 audio_format_t format, 4228 uint32_t channelMask, 4229 int frameCount) 4230 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4231 mActive(false), mSourceThread(sourceThread) 4232{ 4233 4234 if (mCblk != NULL) { 4235 mCblk->flags |= CBLK_DIRECTION_OUT; 4236 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4237 mOutBuffer.frameCount = 0; 4238 playbackThread->mTracks.add(this); 4239 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4240 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4241 mCblk, mBuffer, mCblk->buffers, 4242 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4243 } else { 4244 ALOGW("Error creating output track on thread %p", playbackThread); 4245 } 4246} 4247 4248AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4249{ 4250 clearBufferQueue(); 4251} 4252 4253status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4254{ 4255 status_t status = Track::start(tid); 4256 if (status != NO_ERROR) { 4257 return status; 4258 } 4259 4260 mActive = true; 4261 mRetryCount = 127; 4262 return status; 4263} 4264 4265void AudioFlinger::PlaybackThread::OutputTrack::stop() 4266{ 4267 Track::stop(); 4268 clearBufferQueue(); 4269 mOutBuffer.frameCount = 0; 4270 mActive = false; 4271} 4272 4273bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4274{ 4275 Buffer *pInBuffer; 4276 Buffer inBuffer; 4277 uint32_t channelCount = mChannelCount; 4278 bool outputBufferFull = false; 4279 inBuffer.frameCount = frames; 4280 inBuffer.i16 = data; 4281 4282 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4283 4284 if (!mActive && frames != 0) { 4285 start(0); 4286 sp<ThreadBase> thread = mThread.promote(); 4287 if (thread != 0) { 4288 MixerThread *mixerThread = (MixerThread *)thread.get(); 4289 if (mCblk->frameCount > frames){ 4290 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4291 uint32_t startFrames = (mCblk->frameCount - frames); 4292 pInBuffer = new Buffer; 4293 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4294 pInBuffer->frameCount = startFrames; 4295 pInBuffer->i16 = pInBuffer->mBuffer; 4296 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4297 mBufferQueue.add(pInBuffer); 4298 } else { 4299 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4300 } 4301 } 4302 } 4303 } 4304 4305 while (waitTimeLeftMs) { 4306 // First write pending buffers, then new data 4307 if (mBufferQueue.size()) { 4308 pInBuffer = mBufferQueue.itemAt(0); 4309 } else { 4310 pInBuffer = &inBuffer; 4311 } 4312 4313 if (pInBuffer->frameCount == 0) { 4314 break; 4315 } 4316 4317 if (mOutBuffer.frameCount == 0) { 4318 mOutBuffer.frameCount = pInBuffer->frameCount; 4319 nsecs_t startTime = systemTime(); 4320 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4321 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4322 outputBufferFull = true; 4323 break; 4324 } 4325 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4326 if (waitTimeLeftMs >= waitTimeMs) { 4327 waitTimeLeftMs -= waitTimeMs; 4328 } else { 4329 waitTimeLeftMs = 0; 4330 } 4331 } 4332 4333 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4334 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4335 mCblk->stepUser(outFrames); 4336 pInBuffer->frameCount -= outFrames; 4337 pInBuffer->i16 += outFrames * channelCount; 4338 mOutBuffer.frameCount -= outFrames; 4339 mOutBuffer.i16 += outFrames * channelCount; 4340 4341 if (pInBuffer->frameCount == 0) { 4342 if (mBufferQueue.size()) { 4343 mBufferQueue.removeAt(0); 4344 delete [] pInBuffer->mBuffer; 4345 delete pInBuffer; 4346 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4347 } else { 4348 break; 4349 } 4350 } 4351 } 4352 4353 // If we could not write all frames, allocate a buffer and queue it for next time. 4354 if (inBuffer.frameCount) { 4355 sp<ThreadBase> thread = mThread.promote(); 4356 if (thread != 0 && !thread->standby()) { 4357 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4358 pInBuffer = new Buffer; 4359 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4360 pInBuffer->frameCount = inBuffer.frameCount; 4361 pInBuffer->i16 = pInBuffer->mBuffer; 4362 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4363 mBufferQueue.add(pInBuffer); 4364 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4365 } else { 4366 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4367 } 4368 } 4369 } 4370 4371 // Calling write() with a 0 length buffer, means that no more data will be written: 4372 // If no more buffers are pending, fill output track buffer to make sure it is started 4373 // by output mixer. 4374 if (frames == 0 && mBufferQueue.size() == 0) { 4375 if (mCblk->user < mCblk->frameCount) { 4376 frames = mCblk->frameCount - mCblk->user; 4377 pInBuffer = new Buffer; 4378 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4379 pInBuffer->frameCount = frames; 4380 pInBuffer->i16 = pInBuffer->mBuffer; 4381 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4382 mBufferQueue.add(pInBuffer); 4383 } else if (mActive) { 4384 stop(); 4385 } 4386 } 4387 4388 return outputBufferFull; 4389} 4390 4391status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4392{ 4393 int active; 4394 status_t result; 4395 audio_track_cblk_t* cblk = mCblk; 4396 uint32_t framesReq = buffer->frameCount; 4397 4398// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4399 buffer->frameCount = 0; 4400 4401 uint32_t framesAvail = cblk->framesAvailable(); 4402 4403 4404 if (framesAvail == 0) { 4405 Mutex::Autolock _l(cblk->lock); 4406 goto start_loop_here; 4407 while (framesAvail == 0) { 4408 active = mActive; 4409 if (CC_UNLIKELY(!active)) { 4410 ALOGV("Not active and NO_MORE_BUFFERS"); 4411 return NO_MORE_BUFFERS; 4412 } 4413 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4414 if (result != NO_ERROR) { 4415 return NO_MORE_BUFFERS; 4416 } 4417 // read the server count again 4418 start_loop_here: 4419 framesAvail = cblk->framesAvailable_l(); 4420 } 4421 } 4422 4423// if (framesAvail < framesReq) { 4424// return NO_MORE_BUFFERS; 4425// } 4426 4427 if (framesReq > framesAvail) { 4428 framesReq = framesAvail; 4429 } 4430 4431 uint32_t u = cblk->user; 4432 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4433 4434 if (u + framesReq > bufferEnd) { 4435 framesReq = bufferEnd - u; 4436 } 4437 4438 buffer->frameCount = framesReq; 4439 buffer->raw = (void *)cblk->buffer(u); 4440 return NO_ERROR; 4441} 4442 4443 4444void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4445{ 4446 size_t size = mBufferQueue.size(); 4447 4448 for (size_t i = 0; i < size; i++) { 4449 Buffer *pBuffer = mBufferQueue.itemAt(i); 4450 delete [] pBuffer->mBuffer; 4451 delete pBuffer; 4452 } 4453 mBufferQueue.clear(); 4454} 4455 4456// ---------------------------------------------------------------------------- 4457 4458AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4459 : RefBase(), 4460 mAudioFlinger(audioFlinger), 4461 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4462 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4463 mPid(pid), 4464 mTimedTrackCount(0) 4465{ 4466 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4467} 4468 4469// Client destructor must be called with AudioFlinger::mLock held 4470AudioFlinger::Client::~Client() 4471{ 4472 mAudioFlinger->removeClient_l(mPid); 4473} 4474 4475sp<MemoryDealer> AudioFlinger::Client::heap() const 4476{ 4477 return mMemoryDealer; 4478} 4479 4480// Reserve one of the limited slots for a timed audio track associated 4481// with this client 4482bool AudioFlinger::Client::reserveTimedTrack() 4483{ 4484 const int kMaxTimedTracksPerClient = 4; 4485 4486 Mutex::Autolock _l(mTimedTrackLock); 4487 4488 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4489 ALOGW("can not create timed track - pid %d has exceeded the limit", 4490 mPid); 4491 return false; 4492 } 4493 4494 mTimedTrackCount++; 4495 return true; 4496} 4497 4498// Release a slot for a timed audio track 4499void AudioFlinger::Client::releaseTimedTrack() 4500{ 4501 Mutex::Autolock _l(mTimedTrackLock); 4502 mTimedTrackCount--; 4503} 4504 4505// ---------------------------------------------------------------------------- 4506 4507AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4508 const sp<IAudioFlingerClient>& client, 4509 pid_t pid) 4510 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4511{ 4512} 4513 4514AudioFlinger::NotificationClient::~NotificationClient() 4515{ 4516} 4517 4518void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4519{ 4520 sp<NotificationClient> keep(this); 4521 mAudioFlinger->removeNotificationClient(mPid); 4522} 4523 4524// ---------------------------------------------------------------------------- 4525 4526AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4527 : BnAudioTrack(), 4528 mTrack(track) 4529{ 4530} 4531 4532AudioFlinger::TrackHandle::~TrackHandle() { 4533 // just stop the track on deletion, associated resources 4534 // will be freed from the main thread once all pending buffers have 4535 // been played. Unless it's not in the active track list, in which 4536 // case we free everything now... 4537 mTrack->destroy(); 4538} 4539 4540sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4541 return mTrack->getCblk(); 4542} 4543 4544status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4545 return mTrack->start(tid); 4546} 4547 4548void AudioFlinger::TrackHandle::stop() { 4549 mTrack->stop(); 4550} 4551 4552void AudioFlinger::TrackHandle::flush() { 4553 mTrack->flush(); 4554} 4555 4556void AudioFlinger::TrackHandle::mute(bool e) { 4557 mTrack->mute(e); 4558} 4559 4560void AudioFlinger::TrackHandle::pause() { 4561 mTrack->pause(); 4562} 4563 4564status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4565{ 4566 return mTrack->attachAuxEffect(EffectId); 4567} 4568 4569status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4570 sp<IMemory>* buffer) { 4571 if (!mTrack->isTimedTrack()) 4572 return INVALID_OPERATION; 4573 4574 PlaybackThread::TimedTrack* tt = 4575 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4576 return tt->allocateTimedBuffer(size, buffer); 4577} 4578 4579status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4580 int64_t pts) { 4581 if (!mTrack->isTimedTrack()) 4582 return INVALID_OPERATION; 4583 4584 PlaybackThread::TimedTrack* tt = 4585 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4586 return tt->queueTimedBuffer(buffer, pts); 4587} 4588 4589status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4590 const LinearTransform& xform, int target) { 4591 4592 if (!mTrack->isTimedTrack()) 4593 return INVALID_OPERATION; 4594 4595 PlaybackThread::TimedTrack* tt = 4596 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4597 return tt->setMediaTimeTransform( 4598 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4599} 4600 4601status_t AudioFlinger::TrackHandle::onTransact( 4602 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4603{ 4604 return BnAudioTrack::onTransact(code, data, reply, flags); 4605} 4606 4607// ---------------------------------------------------------------------------- 4608 4609sp<IAudioRecord> AudioFlinger::openRecord( 4610 pid_t pid, 4611 audio_io_handle_t input, 4612 uint32_t sampleRate, 4613 audio_format_t format, 4614 uint32_t channelMask, 4615 int frameCount, 4616 // FIXME dead, remove from IAudioFlinger 4617 uint32_t flags, 4618 int *sessionId, 4619 status_t *status) 4620{ 4621 sp<RecordThread::RecordTrack> recordTrack; 4622 sp<RecordHandle> recordHandle; 4623 sp<Client> client; 4624 status_t lStatus; 4625 RecordThread *thread; 4626 size_t inFrameCount; 4627 int lSessionId; 4628 4629 // check calling permissions 4630 if (!recordingAllowed()) { 4631 lStatus = PERMISSION_DENIED; 4632 goto Exit; 4633 } 4634 4635 // add client to list 4636 { // scope for mLock 4637 Mutex::Autolock _l(mLock); 4638 thread = checkRecordThread_l(input); 4639 if (thread == NULL) { 4640 lStatus = BAD_VALUE; 4641 goto Exit; 4642 } 4643 4644 client = registerPid_l(pid); 4645 4646 // If no audio session id is provided, create one here 4647 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4648 lSessionId = *sessionId; 4649 } else { 4650 lSessionId = nextUniqueId(); 4651 if (sessionId != NULL) { 4652 *sessionId = lSessionId; 4653 } 4654 } 4655 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4656 recordTrack = thread->createRecordTrack_l(client, 4657 sampleRate, 4658 format, 4659 channelMask, 4660 frameCount, 4661 lSessionId, 4662 &lStatus); 4663 } 4664 if (lStatus != NO_ERROR) { 4665 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4666 // destructor is called by the TrackBase destructor with mLock held 4667 client.clear(); 4668 recordTrack.clear(); 4669 goto Exit; 4670 } 4671 4672 // return to handle to client 4673 recordHandle = new RecordHandle(recordTrack); 4674 lStatus = NO_ERROR; 4675 4676Exit: 4677 if (status) { 4678 *status = lStatus; 4679 } 4680 return recordHandle; 4681} 4682 4683// ---------------------------------------------------------------------------- 4684 4685AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4686 : BnAudioRecord(), 4687 mRecordTrack(recordTrack) 4688{ 4689} 4690 4691AudioFlinger::RecordHandle::~RecordHandle() { 4692 stop(); 4693} 4694 4695sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4696 return mRecordTrack->getCblk(); 4697} 4698 4699status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4700 ALOGV("RecordHandle::start()"); 4701 return mRecordTrack->start(tid); 4702} 4703 4704void AudioFlinger::RecordHandle::stop() { 4705 ALOGV("RecordHandle::stop()"); 4706 mRecordTrack->stop(); 4707} 4708 4709status_t AudioFlinger::RecordHandle::onTransact( 4710 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4711{ 4712 return BnAudioRecord::onTransact(code, data, reply, flags); 4713} 4714 4715// ---------------------------------------------------------------------------- 4716 4717AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4718 AudioStreamIn *input, 4719 uint32_t sampleRate, 4720 uint32_t channels, 4721 audio_io_handle_t id, 4722 uint32_t device) : 4723 ThreadBase(audioFlinger, id, device, RECORD), 4724 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4725 // mRsmpInIndex and mInputBytes set by readInputParameters() 4726 mReqChannelCount(popcount(channels)), 4727 mReqSampleRate(sampleRate) 4728 // mBytesRead is only meaningful while active, and so is cleared in start() 4729 // (but might be better to also clear here for dump?) 4730{ 4731 snprintf(mName, kNameLength, "AudioIn_%X", id); 4732 4733 readInputParameters(); 4734} 4735 4736 4737AudioFlinger::RecordThread::~RecordThread() 4738{ 4739 delete[] mRsmpInBuffer; 4740 delete mResampler; 4741 delete[] mRsmpOutBuffer; 4742} 4743 4744void AudioFlinger::RecordThread::onFirstRef() 4745{ 4746 run(mName, PRIORITY_URGENT_AUDIO); 4747} 4748 4749status_t AudioFlinger::RecordThread::readyToRun() 4750{ 4751 status_t status = initCheck(); 4752 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4753 return status; 4754} 4755 4756bool AudioFlinger::RecordThread::threadLoop() 4757{ 4758 AudioBufferProvider::Buffer buffer; 4759 sp<RecordTrack> activeTrack; 4760 Vector< sp<EffectChain> > effectChains; 4761 4762 nsecs_t lastWarning = 0; 4763 4764 acquireWakeLock(); 4765 4766 // start recording 4767 while (!exitPending()) { 4768 4769 processConfigEvents(); 4770 4771 { // scope for mLock 4772 Mutex::Autolock _l(mLock); 4773 checkForNewParameters_l(); 4774 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4775 if (!mStandby) { 4776 mInput->stream->common.standby(&mInput->stream->common); 4777 mStandby = true; 4778 } 4779 4780 if (exitPending()) break; 4781 4782 releaseWakeLock_l(); 4783 ALOGV("RecordThread: loop stopping"); 4784 // go to sleep 4785 mWaitWorkCV.wait(mLock); 4786 ALOGV("RecordThread: loop starting"); 4787 acquireWakeLock_l(); 4788 continue; 4789 } 4790 if (mActiveTrack != 0) { 4791 if (mActiveTrack->mState == TrackBase::PAUSING) { 4792 if (!mStandby) { 4793 mInput->stream->common.standby(&mInput->stream->common); 4794 mStandby = true; 4795 } 4796 mActiveTrack.clear(); 4797 mStartStopCond.broadcast(); 4798 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4799 if (mReqChannelCount != mActiveTrack->channelCount()) { 4800 mActiveTrack.clear(); 4801 mStartStopCond.broadcast(); 4802 } else if (mBytesRead != 0) { 4803 // record start succeeds only if first read from audio input 4804 // succeeds 4805 if (mBytesRead > 0) { 4806 mActiveTrack->mState = TrackBase::ACTIVE; 4807 } else { 4808 mActiveTrack.clear(); 4809 } 4810 mStartStopCond.broadcast(); 4811 } 4812 mStandby = false; 4813 } 4814 } 4815 lockEffectChains_l(effectChains); 4816 } 4817 4818 if (mActiveTrack != 0) { 4819 if (mActiveTrack->mState != TrackBase::ACTIVE && 4820 mActiveTrack->mState != TrackBase::RESUMING) { 4821 unlockEffectChains(effectChains); 4822 usleep(kRecordThreadSleepUs); 4823 continue; 4824 } 4825 for (size_t i = 0; i < effectChains.size(); i ++) { 4826 effectChains[i]->process_l(); 4827 } 4828 4829 buffer.frameCount = mFrameCount; 4830 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4831 size_t framesOut = buffer.frameCount; 4832 if (mResampler == NULL) { 4833 // no resampling 4834 while (framesOut) { 4835 size_t framesIn = mFrameCount - mRsmpInIndex; 4836 if (framesIn) { 4837 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4838 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4839 if (framesIn > framesOut) 4840 framesIn = framesOut; 4841 mRsmpInIndex += framesIn; 4842 framesOut -= framesIn; 4843 if ((int)mChannelCount == mReqChannelCount || 4844 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4845 memcpy(dst, src, framesIn * mFrameSize); 4846 } else { 4847 int16_t *src16 = (int16_t *)src; 4848 int16_t *dst16 = (int16_t *)dst; 4849 if (mChannelCount == 1) { 4850 while (framesIn--) { 4851 *dst16++ = *src16; 4852 *dst16++ = *src16++; 4853 } 4854 } else { 4855 while (framesIn--) { 4856 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4857 src16 += 2; 4858 } 4859 } 4860 } 4861 } 4862 if (framesOut && mFrameCount == mRsmpInIndex) { 4863 if (framesOut == mFrameCount && 4864 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4865 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4866 framesOut = 0; 4867 } else { 4868 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4869 mRsmpInIndex = 0; 4870 } 4871 if (mBytesRead < 0) { 4872 ALOGE("Error reading audio input"); 4873 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4874 // Force input into standby so that it tries to 4875 // recover at next read attempt 4876 mInput->stream->common.standby(&mInput->stream->common); 4877 usleep(kRecordThreadSleepUs); 4878 } 4879 mRsmpInIndex = mFrameCount; 4880 framesOut = 0; 4881 buffer.frameCount = 0; 4882 } 4883 } 4884 } 4885 } else { 4886 // resampling 4887 4888 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4889 // alter output frame count as if we were expecting stereo samples 4890 if (mChannelCount == 1 && mReqChannelCount == 1) { 4891 framesOut >>= 1; 4892 } 4893 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4894 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4895 // are 32 bit aligned which should be always true. 4896 if (mChannelCount == 2 && mReqChannelCount == 1) { 4897 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4898 // the resampler always outputs stereo samples: do post stereo to mono conversion 4899 int16_t *src = (int16_t *)mRsmpOutBuffer; 4900 int16_t *dst = buffer.i16; 4901 while (framesOut--) { 4902 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4903 src += 2; 4904 } 4905 } else { 4906 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4907 } 4908 4909 } 4910 mActiveTrack->releaseBuffer(&buffer); 4911 mActiveTrack->overflow(); 4912 } 4913 // client isn't retrieving buffers fast enough 4914 else { 4915 if (!mActiveTrack->setOverflow()) { 4916 nsecs_t now = systemTime(); 4917 if ((now - lastWarning) > kWarningThrottleNs) { 4918 ALOGW("RecordThread: buffer overflow"); 4919 lastWarning = now; 4920 } 4921 } 4922 // Release the processor for a while before asking for a new buffer. 4923 // This will give the application more chance to read from the buffer and 4924 // clear the overflow. 4925 usleep(kRecordThreadSleepUs); 4926 } 4927 } 4928 // enable changes in effect chain 4929 unlockEffectChains(effectChains); 4930 effectChains.clear(); 4931 } 4932 4933 if (!mStandby) { 4934 mInput->stream->common.standby(&mInput->stream->common); 4935 } 4936 mActiveTrack.clear(); 4937 4938 mStartStopCond.broadcast(); 4939 4940 releaseWakeLock(); 4941 4942 ALOGV("RecordThread %p exiting", this); 4943 return false; 4944} 4945 4946 4947sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4948 const sp<AudioFlinger::Client>& client, 4949 uint32_t sampleRate, 4950 audio_format_t format, 4951 int channelMask, 4952 int frameCount, 4953 int sessionId, 4954 status_t *status) 4955{ 4956 sp<RecordTrack> track; 4957 status_t lStatus; 4958 4959 lStatus = initCheck(); 4960 if (lStatus != NO_ERROR) { 4961 ALOGE("Audio driver not initialized."); 4962 goto Exit; 4963 } 4964 4965 { // scope for mLock 4966 Mutex::Autolock _l(mLock); 4967 4968 track = new RecordTrack(this, client, sampleRate, 4969 format, channelMask, frameCount, sessionId); 4970 4971 if (track->getCblk() == 0) { 4972 lStatus = NO_MEMORY; 4973 goto Exit; 4974 } 4975 4976 mTrack = track.get(); 4977 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4978 bool suspend = audio_is_bluetooth_sco_device( 4979 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4980 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4981 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4982 } 4983 lStatus = NO_ERROR; 4984 4985Exit: 4986 if (status) { 4987 *status = lStatus; 4988 } 4989 return track; 4990} 4991 4992status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4993{ 4994 ALOGV("RecordThread::start tid=%d", tid); 4995 sp<ThreadBase> strongMe = this; 4996 status_t status = NO_ERROR; 4997 { 4998 AutoMutex lock(mLock); 4999 if (mActiveTrack != 0) { 5000 if (recordTrack != mActiveTrack.get()) { 5001 status = -EBUSY; 5002 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5003 mActiveTrack->mState = TrackBase::ACTIVE; 5004 } 5005 return status; 5006 } 5007 5008 recordTrack->mState = TrackBase::IDLE; 5009 mActiveTrack = recordTrack; 5010 mLock.unlock(); 5011 status_t status = AudioSystem::startInput(mId); 5012 mLock.lock(); 5013 if (status != NO_ERROR) { 5014 mActiveTrack.clear(); 5015 return status; 5016 } 5017 mRsmpInIndex = mFrameCount; 5018 mBytesRead = 0; 5019 if (mResampler != NULL) { 5020 mResampler->reset(); 5021 } 5022 mActiveTrack->mState = TrackBase::RESUMING; 5023 // signal thread to start 5024 ALOGV("Signal record thread"); 5025 mWaitWorkCV.signal(); 5026 // do not wait for mStartStopCond if exiting 5027 if (exitPending()) { 5028 mActiveTrack.clear(); 5029 status = INVALID_OPERATION; 5030 goto startError; 5031 } 5032 mStartStopCond.wait(mLock); 5033 if (mActiveTrack == 0) { 5034 ALOGV("Record failed to start"); 5035 status = BAD_VALUE; 5036 goto startError; 5037 } 5038 ALOGV("Record started OK"); 5039 return status; 5040 } 5041startError: 5042 AudioSystem::stopInput(mId); 5043 return status; 5044} 5045 5046void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5047 ALOGV("RecordThread::stop"); 5048 sp<ThreadBase> strongMe = this; 5049 { 5050 AutoMutex lock(mLock); 5051 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5052 mActiveTrack->mState = TrackBase::PAUSING; 5053 // do not wait for mStartStopCond if exiting 5054 if (exitPending()) { 5055 return; 5056 } 5057 mStartStopCond.wait(mLock); 5058 // if we have been restarted, recordTrack == mActiveTrack.get() here 5059 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5060 mLock.unlock(); 5061 AudioSystem::stopInput(mId); 5062 mLock.lock(); 5063 ALOGV("Record stopped OK"); 5064 } 5065 } 5066 } 5067} 5068 5069status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5070{ 5071 const size_t SIZE = 256; 5072 char buffer[SIZE]; 5073 String8 result; 5074 5075 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5076 result.append(buffer); 5077 5078 if (mActiveTrack != 0) { 5079 result.append("Active Track:\n"); 5080 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5081 mActiveTrack->dump(buffer, SIZE); 5082 result.append(buffer); 5083 5084 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5085 result.append(buffer); 5086 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5087 result.append(buffer); 5088 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5089 result.append(buffer); 5090 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5091 result.append(buffer); 5092 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5093 result.append(buffer); 5094 5095 5096 } else { 5097 result.append("No record client\n"); 5098 } 5099 write(fd, result.string(), result.size()); 5100 5101 dumpBase(fd, args); 5102 dumpEffectChains(fd, args); 5103 5104 return NO_ERROR; 5105} 5106 5107// AudioBufferProvider interface 5108status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5109{ 5110 size_t framesReq = buffer->frameCount; 5111 size_t framesReady = mFrameCount - mRsmpInIndex; 5112 int channelCount; 5113 5114 if (framesReady == 0) { 5115 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5116 if (mBytesRead < 0) { 5117 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5118 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5119 // Force input into standby so that it tries to 5120 // recover at next read attempt 5121 mInput->stream->common.standby(&mInput->stream->common); 5122 usleep(kRecordThreadSleepUs); 5123 } 5124 buffer->raw = NULL; 5125 buffer->frameCount = 0; 5126 return NOT_ENOUGH_DATA; 5127 } 5128 mRsmpInIndex = 0; 5129 framesReady = mFrameCount; 5130 } 5131 5132 if (framesReq > framesReady) { 5133 framesReq = framesReady; 5134 } 5135 5136 if (mChannelCount == 1 && mReqChannelCount == 2) { 5137 channelCount = 1; 5138 } else { 5139 channelCount = 2; 5140 } 5141 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5142 buffer->frameCount = framesReq; 5143 return NO_ERROR; 5144} 5145 5146// AudioBufferProvider interface 5147void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5148{ 5149 mRsmpInIndex += buffer->frameCount; 5150 buffer->frameCount = 0; 5151} 5152 5153bool AudioFlinger::RecordThread::checkForNewParameters_l() 5154{ 5155 bool reconfig = false; 5156 5157 while (!mNewParameters.isEmpty()) { 5158 status_t status = NO_ERROR; 5159 String8 keyValuePair = mNewParameters[0]; 5160 AudioParameter param = AudioParameter(keyValuePair); 5161 int value; 5162 audio_format_t reqFormat = mFormat; 5163 int reqSamplingRate = mReqSampleRate; 5164 int reqChannelCount = mReqChannelCount; 5165 5166 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5167 reqSamplingRate = value; 5168 reconfig = true; 5169 } 5170 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5171 reqFormat = (audio_format_t) value; 5172 reconfig = true; 5173 } 5174 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5175 reqChannelCount = popcount(value); 5176 reconfig = true; 5177 } 5178 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5179 // do not accept frame count changes if tracks are open as the track buffer 5180 // size depends on frame count and correct behavior would not be guaranteed 5181 // if frame count is changed after track creation 5182 if (mActiveTrack != 0) { 5183 status = INVALID_OPERATION; 5184 } else { 5185 reconfig = true; 5186 } 5187 } 5188 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5189 // forward device change to effects that have requested to be 5190 // aware of attached audio device. 5191 for (size_t i = 0; i < mEffectChains.size(); i++) { 5192 mEffectChains[i]->setDevice_l(value); 5193 } 5194 // store input device and output device but do not forward output device to audio HAL. 5195 // Note that status is ignored by the caller for output device 5196 // (see AudioFlinger::setParameters() 5197 if (value & AUDIO_DEVICE_OUT_ALL) { 5198 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5199 status = BAD_VALUE; 5200 } else { 5201 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5202 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5203 if (mTrack != NULL) { 5204 bool suspend = audio_is_bluetooth_sco_device( 5205 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5206 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5207 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5208 } 5209 } 5210 mDevice |= (uint32_t)value; 5211 } 5212 if (status == NO_ERROR) { 5213 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5214 if (status == INVALID_OPERATION) { 5215 mInput->stream->common.standby(&mInput->stream->common); 5216 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5217 keyValuePair.string()); 5218 } 5219 if (reconfig) { 5220 if (status == BAD_VALUE && 5221 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5222 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5223 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5224 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5225 (reqChannelCount <= FCC_2)) { 5226 status = NO_ERROR; 5227 } 5228 if (status == NO_ERROR) { 5229 readInputParameters(); 5230 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5231 } 5232 } 5233 } 5234 5235 mNewParameters.removeAt(0); 5236 5237 mParamStatus = status; 5238 mParamCond.signal(); 5239 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5240 // already timed out waiting for the status and will never signal the condition. 5241 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5242 } 5243 return reconfig; 5244} 5245 5246String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5247{ 5248 char *s; 5249 String8 out_s8 = String8(); 5250 5251 Mutex::Autolock _l(mLock); 5252 if (initCheck() != NO_ERROR) { 5253 return out_s8; 5254 } 5255 5256 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5257 out_s8 = String8(s); 5258 free(s); 5259 return out_s8; 5260} 5261 5262void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5263 AudioSystem::OutputDescriptor desc; 5264 void *param2 = NULL; 5265 5266 switch (event) { 5267 case AudioSystem::INPUT_OPENED: 5268 case AudioSystem::INPUT_CONFIG_CHANGED: 5269 desc.channels = mChannelMask; 5270 desc.samplingRate = mSampleRate; 5271 desc.format = mFormat; 5272 desc.frameCount = mFrameCount; 5273 desc.latency = 0; 5274 param2 = &desc; 5275 break; 5276 5277 case AudioSystem::INPUT_CLOSED: 5278 default: 5279 break; 5280 } 5281 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5282} 5283 5284void AudioFlinger::RecordThread::readInputParameters() 5285{ 5286 delete mRsmpInBuffer; 5287 // mRsmpInBuffer is always assigned a new[] below 5288 delete mRsmpOutBuffer; 5289 mRsmpOutBuffer = NULL; 5290 delete mResampler; 5291 mResampler = NULL; 5292 5293 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5294 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5295 mChannelCount = (uint16_t)popcount(mChannelMask); 5296 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5297 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5298 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5299 mFrameCount = mInputBytes / mFrameSize; 5300 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5301 5302 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5303 { 5304 int channelCount; 5305 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5306 // stereo to mono post process as the resampler always outputs stereo. 5307 if (mChannelCount == 1 && mReqChannelCount == 2) { 5308 channelCount = 1; 5309 } else { 5310 channelCount = 2; 5311 } 5312 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5313 mResampler->setSampleRate(mSampleRate); 5314 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5315 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5316 5317 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5318 if (mChannelCount == 1 && mReqChannelCount == 1) { 5319 mFrameCount >>= 1; 5320 } 5321 5322 } 5323 mRsmpInIndex = mFrameCount; 5324} 5325 5326unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5327{ 5328 Mutex::Autolock _l(mLock); 5329 if (initCheck() != NO_ERROR) { 5330 return 0; 5331 } 5332 5333 return mInput->stream->get_input_frames_lost(mInput->stream); 5334} 5335 5336uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5337{ 5338 Mutex::Autolock _l(mLock); 5339 uint32_t result = 0; 5340 if (getEffectChain_l(sessionId) != 0) { 5341 result = EFFECT_SESSION; 5342 } 5343 5344 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5345 result |= TRACK_SESSION; 5346 } 5347 5348 return result; 5349} 5350 5351AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5352{ 5353 Mutex::Autolock _l(mLock); 5354 return mTrack; 5355} 5356 5357AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5358{ 5359 Mutex::Autolock _l(mLock); 5360 return mInput; 5361} 5362 5363AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5364{ 5365 Mutex::Autolock _l(mLock); 5366 AudioStreamIn *input = mInput; 5367 mInput = NULL; 5368 return input; 5369} 5370 5371// this method must always be called either with ThreadBase mLock held or inside the thread loop 5372audio_stream_t* AudioFlinger::RecordThread::stream() 5373{ 5374 if (mInput == NULL) { 5375 return NULL; 5376 } 5377 return &mInput->stream->common; 5378} 5379 5380 5381// ---------------------------------------------------------------------------- 5382 5383audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5384 uint32_t *pSamplingRate, 5385 audio_format_t *pFormat, 5386 uint32_t *pChannels, 5387 uint32_t *pLatencyMs, 5388 audio_policy_output_flags_t flags) 5389{ 5390 status_t status; 5391 PlaybackThread *thread = NULL; 5392 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5393 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5394 uint32_t channels = pChannels ? *pChannels : 0; 5395 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5396 audio_stream_out_t *outStream; 5397 audio_hw_device_t *outHwDev; 5398 5399 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5400 pDevices ? *pDevices : 0, 5401 samplingRate, 5402 format, 5403 channels, 5404 flags); 5405 5406 if (pDevices == NULL || *pDevices == 0) { 5407 return 0; 5408 } 5409 5410 Mutex::Autolock _l(mLock); 5411 5412 outHwDev = findSuitableHwDev_l(*pDevices); 5413 if (outHwDev == NULL) 5414 return 0; 5415 5416 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5417 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5418 &channels, &samplingRate, &outStream); 5419 mHardwareStatus = AUDIO_HW_IDLE; 5420 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5421 outStream, 5422 samplingRate, 5423 format, 5424 channels, 5425 status); 5426 5427 if (outStream != NULL) { 5428 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5429 audio_io_handle_t id = nextUniqueId(); 5430 5431 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5432 (format != AUDIO_FORMAT_PCM_16_BIT) || 5433 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5434 thread = new DirectOutputThread(this, output, id, *pDevices); 5435 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5436 } else { 5437 thread = new MixerThread(this, output, id, *pDevices); 5438 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5439 } 5440 mPlaybackThreads.add(id, thread); 5441 5442 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5443 if (pFormat != NULL) *pFormat = format; 5444 if (pChannels != NULL) *pChannels = channels; 5445 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5446 5447 // notify client processes of the new output creation 5448 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5449 return id; 5450 } 5451 5452 return 0; 5453} 5454 5455audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5456 audio_io_handle_t output2) 5457{ 5458 Mutex::Autolock _l(mLock); 5459 MixerThread *thread1 = checkMixerThread_l(output1); 5460 MixerThread *thread2 = checkMixerThread_l(output2); 5461 5462 if (thread1 == NULL || thread2 == NULL) { 5463 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5464 return 0; 5465 } 5466 5467 audio_io_handle_t id = nextUniqueId(); 5468 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5469 thread->addOutputTrack(thread2); 5470 mPlaybackThreads.add(id, thread); 5471 // notify client processes of the new output creation 5472 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5473 return id; 5474} 5475 5476status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5477{ 5478 // keep strong reference on the playback thread so that 5479 // it is not destroyed while exit() is executed 5480 sp<PlaybackThread> thread; 5481 { 5482 Mutex::Autolock _l(mLock); 5483 thread = checkPlaybackThread_l(output); 5484 if (thread == NULL) { 5485 return BAD_VALUE; 5486 } 5487 5488 ALOGV("closeOutput() %d", output); 5489 5490 if (thread->type() == ThreadBase::MIXER) { 5491 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5492 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5493 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5494 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5495 } 5496 } 5497 } 5498 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5499 mPlaybackThreads.removeItem(output); 5500 } 5501 thread->exit(); 5502 // The thread entity (active unit of execution) is no longer running here, 5503 // but the ThreadBase container still exists. 5504 5505 if (thread->type() != ThreadBase::DUPLICATING) { 5506 AudioStreamOut *out = thread->clearOutput(); 5507 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5508 // from now on thread->mOutput is NULL 5509 out->hwDev->close_output_stream(out->hwDev, out->stream); 5510 delete out; 5511 } 5512 return NO_ERROR; 5513} 5514 5515status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5516{ 5517 Mutex::Autolock _l(mLock); 5518 PlaybackThread *thread = checkPlaybackThread_l(output); 5519 5520 if (thread == NULL) { 5521 return BAD_VALUE; 5522 } 5523 5524 ALOGV("suspendOutput() %d", output); 5525 thread->suspend(); 5526 5527 return NO_ERROR; 5528} 5529 5530status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5531{ 5532 Mutex::Autolock _l(mLock); 5533 PlaybackThread *thread = checkPlaybackThread_l(output); 5534 5535 if (thread == NULL) { 5536 return BAD_VALUE; 5537 } 5538 5539 ALOGV("restoreOutput() %d", output); 5540 5541 thread->restore(); 5542 5543 return NO_ERROR; 5544} 5545 5546audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5547 uint32_t *pSamplingRate, 5548 audio_format_t *pFormat, 5549 uint32_t *pChannels, 5550 audio_in_acoustics_t acoustics) 5551{ 5552 status_t status; 5553 RecordThread *thread = NULL; 5554 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5555 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5556 uint32_t channels = pChannels ? *pChannels : 0; 5557 uint32_t reqSamplingRate = samplingRate; 5558 audio_format_t reqFormat = format; 5559 uint32_t reqChannels = channels; 5560 audio_stream_in_t *inStream; 5561 audio_hw_device_t *inHwDev; 5562 5563 if (pDevices == NULL || *pDevices == 0) { 5564 return 0; 5565 } 5566 5567 Mutex::Autolock _l(mLock); 5568 5569 inHwDev = findSuitableHwDev_l(*pDevices); 5570 if (inHwDev == NULL) 5571 return 0; 5572 5573 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5574 &channels, &samplingRate, 5575 acoustics, 5576 &inStream); 5577 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5578 inStream, 5579 samplingRate, 5580 format, 5581 channels, 5582 acoustics, 5583 status); 5584 5585 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5586 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5587 // or stereo to mono conversions on 16 bit PCM inputs. 5588 if (inStream == NULL && status == BAD_VALUE && 5589 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5590 (samplingRate <= 2 * reqSamplingRate) && 5591 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5592 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5593 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5594 &channels, &samplingRate, 5595 acoustics, 5596 &inStream); 5597 } 5598 5599 if (inStream != NULL) { 5600 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5601 5602 audio_io_handle_t id = nextUniqueId(); 5603 // Start record thread 5604 // RecorThread require both input and output device indication to forward to audio 5605 // pre processing modules 5606 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5607 thread = new RecordThread(this, 5608 input, 5609 reqSamplingRate, 5610 reqChannels, 5611 id, 5612 device); 5613 mRecordThreads.add(id, thread); 5614 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5615 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5616 if (pFormat != NULL) *pFormat = format; 5617 if (pChannels != NULL) *pChannels = reqChannels; 5618 5619 input->stream->common.standby(&input->stream->common); 5620 5621 // notify client processes of the new input creation 5622 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5623 return id; 5624 } 5625 5626 return 0; 5627} 5628 5629status_t AudioFlinger::closeInput(audio_io_handle_t input) 5630{ 5631 // keep strong reference on the record thread so that 5632 // it is not destroyed while exit() is executed 5633 sp<RecordThread> thread; 5634 { 5635 Mutex::Autolock _l(mLock); 5636 thread = checkRecordThread_l(input); 5637 if (thread == NULL) { 5638 return BAD_VALUE; 5639 } 5640 5641 ALOGV("closeInput() %d", input); 5642 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5643 mRecordThreads.removeItem(input); 5644 } 5645 thread->exit(); 5646 // The thread entity (active unit of execution) is no longer running here, 5647 // but the ThreadBase container still exists. 5648 5649 AudioStreamIn *in = thread->clearInput(); 5650 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5651 // from now on thread->mInput is NULL 5652 in->hwDev->close_input_stream(in->hwDev, in->stream); 5653 delete in; 5654 5655 return NO_ERROR; 5656} 5657 5658status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5659{ 5660 Mutex::Autolock _l(mLock); 5661 MixerThread *dstThread = checkMixerThread_l(output); 5662 if (dstThread == NULL) { 5663 ALOGW("setStreamOutput() bad output id %d", output); 5664 return BAD_VALUE; 5665 } 5666 5667 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5668 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5669 5670 dstThread->setStreamValid(stream, true); 5671 5672 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5673 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5674 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5675 MixerThread *srcThread = (MixerThread *)thread; 5676 srcThread->setStreamValid(stream, false); 5677 srcThread->invalidateTracks(stream); 5678 } 5679 } 5680 5681 return NO_ERROR; 5682} 5683 5684 5685int AudioFlinger::newAudioSessionId() 5686{ 5687 return nextUniqueId(); 5688} 5689 5690void AudioFlinger::acquireAudioSessionId(int audioSession) 5691{ 5692 Mutex::Autolock _l(mLock); 5693 pid_t caller = IPCThreadState::self()->getCallingPid(); 5694 ALOGV("acquiring %d from %d", audioSession, caller); 5695 size_t num = mAudioSessionRefs.size(); 5696 for (size_t i = 0; i< num; i++) { 5697 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5698 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5699 ref->mCnt++; 5700 ALOGV(" incremented refcount to %d", ref->mCnt); 5701 return; 5702 } 5703 } 5704 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5705 ALOGV(" added new entry for %d", audioSession); 5706} 5707 5708void AudioFlinger::releaseAudioSessionId(int audioSession) 5709{ 5710 Mutex::Autolock _l(mLock); 5711 pid_t caller = IPCThreadState::self()->getCallingPid(); 5712 ALOGV("releasing %d from %d", audioSession, caller); 5713 size_t num = mAudioSessionRefs.size(); 5714 for (size_t i = 0; i< num; i++) { 5715 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5716 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5717 ref->mCnt--; 5718 ALOGV(" decremented refcount to %d", ref->mCnt); 5719 if (ref->mCnt == 0) { 5720 mAudioSessionRefs.removeAt(i); 5721 delete ref; 5722 purgeStaleEffects_l(); 5723 } 5724 return; 5725 } 5726 } 5727 ALOGW("session id %d not found for pid %d", audioSession, caller); 5728} 5729 5730void AudioFlinger::purgeStaleEffects_l() { 5731 5732 ALOGV("purging stale effects"); 5733 5734 Vector< sp<EffectChain> > chains; 5735 5736 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5737 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5738 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5739 sp<EffectChain> ec = t->mEffectChains[j]; 5740 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5741 chains.push(ec); 5742 } 5743 } 5744 } 5745 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5746 sp<RecordThread> t = mRecordThreads.valueAt(i); 5747 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5748 sp<EffectChain> ec = t->mEffectChains[j]; 5749 chains.push(ec); 5750 } 5751 } 5752 5753 for (size_t i = 0; i < chains.size(); i++) { 5754 sp<EffectChain> ec = chains[i]; 5755 int sessionid = ec->sessionId(); 5756 sp<ThreadBase> t = ec->mThread.promote(); 5757 if (t == 0) { 5758 continue; 5759 } 5760 size_t numsessionrefs = mAudioSessionRefs.size(); 5761 bool found = false; 5762 for (size_t k = 0; k < numsessionrefs; k++) { 5763 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5764 if (ref->mSessionid == sessionid) { 5765 ALOGV(" session %d still exists for %d with %d refs", 5766 sessionid, ref->mPid, ref->mCnt); 5767 found = true; 5768 break; 5769 } 5770 } 5771 if (!found) { 5772 // remove all effects from the chain 5773 while (ec->mEffects.size()) { 5774 sp<EffectModule> effect = ec->mEffects[0]; 5775 effect->unPin(); 5776 Mutex::Autolock _l (t->mLock); 5777 t->removeEffect_l(effect); 5778 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5779 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5780 if (handle != 0) { 5781 handle->mEffect.clear(); 5782 if (handle->mHasControl && handle->mEnabled) { 5783 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5784 } 5785 } 5786 } 5787 AudioSystem::unregisterEffect(effect->id()); 5788 } 5789 } 5790 } 5791 return; 5792} 5793 5794// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5795AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5796{ 5797 return mPlaybackThreads.valueFor(output).get(); 5798} 5799 5800// checkMixerThread_l() must be called with AudioFlinger::mLock held 5801AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5802{ 5803 PlaybackThread *thread = checkPlaybackThread_l(output); 5804 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5805} 5806 5807// checkRecordThread_l() must be called with AudioFlinger::mLock held 5808AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5809{ 5810 return mRecordThreads.valueFor(input).get(); 5811} 5812 5813uint32_t AudioFlinger::nextUniqueId() 5814{ 5815 return android_atomic_inc(&mNextUniqueId); 5816} 5817 5818AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5819{ 5820 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5821 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5822 AudioStreamOut *output = thread->getOutput(); 5823 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5824 return thread; 5825 } 5826 } 5827 return NULL; 5828} 5829 5830uint32_t AudioFlinger::primaryOutputDevice_l() const 5831{ 5832 PlaybackThread *thread = primaryPlaybackThread_l(); 5833 5834 if (thread == NULL) { 5835 return 0; 5836 } 5837 5838 return thread->device(); 5839} 5840 5841 5842// ---------------------------------------------------------------------------- 5843// Effect management 5844// ---------------------------------------------------------------------------- 5845 5846 5847status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5848{ 5849 Mutex::Autolock _l(mLock); 5850 return EffectQueryNumberEffects(numEffects); 5851} 5852 5853status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5854{ 5855 Mutex::Autolock _l(mLock); 5856 return EffectQueryEffect(index, descriptor); 5857} 5858 5859status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5860 effect_descriptor_t *descriptor) const 5861{ 5862 Mutex::Autolock _l(mLock); 5863 return EffectGetDescriptor(pUuid, descriptor); 5864} 5865 5866 5867sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5868 effect_descriptor_t *pDesc, 5869 const sp<IEffectClient>& effectClient, 5870 int32_t priority, 5871 audio_io_handle_t io, 5872 int sessionId, 5873 status_t *status, 5874 int *id, 5875 int *enabled) 5876{ 5877 status_t lStatus = NO_ERROR; 5878 sp<EffectHandle> handle; 5879 effect_descriptor_t desc; 5880 5881 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5882 pid, effectClient.get(), priority, sessionId, io); 5883 5884 if (pDesc == NULL) { 5885 lStatus = BAD_VALUE; 5886 goto Exit; 5887 } 5888 5889 // check audio settings permission for global effects 5890 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5891 lStatus = PERMISSION_DENIED; 5892 goto Exit; 5893 } 5894 5895 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5896 // that can only be created by audio policy manager (running in same process) 5897 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5898 lStatus = PERMISSION_DENIED; 5899 goto Exit; 5900 } 5901 5902 if (io == 0) { 5903 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5904 // output must be specified by AudioPolicyManager when using session 5905 // AUDIO_SESSION_OUTPUT_STAGE 5906 lStatus = BAD_VALUE; 5907 goto Exit; 5908 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5909 // if the output returned by getOutputForEffect() is removed before we lock the 5910 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5911 // and we will exit safely 5912 io = AudioSystem::getOutputForEffect(&desc); 5913 } 5914 } 5915 5916 { 5917 Mutex::Autolock _l(mLock); 5918 5919 5920 if (!EffectIsNullUuid(&pDesc->uuid)) { 5921 // if uuid is specified, request effect descriptor 5922 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5923 if (lStatus < 0) { 5924 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5925 goto Exit; 5926 } 5927 } else { 5928 // if uuid is not specified, look for an available implementation 5929 // of the required type in effect factory 5930 if (EffectIsNullUuid(&pDesc->type)) { 5931 ALOGW("createEffect() no effect type"); 5932 lStatus = BAD_VALUE; 5933 goto Exit; 5934 } 5935 uint32_t numEffects = 0; 5936 effect_descriptor_t d; 5937 d.flags = 0; // prevent compiler warning 5938 bool found = false; 5939 5940 lStatus = EffectQueryNumberEffects(&numEffects); 5941 if (lStatus < 0) { 5942 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5943 goto Exit; 5944 } 5945 for (uint32_t i = 0; i < numEffects; i++) { 5946 lStatus = EffectQueryEffect(i, &desc); 5947 if (lStatus < 0) { 5948 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5949 continue; 5950 } 5951 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5952 // If matching type found save effect descriptor. If the session is 5953 // 0 and the effect is not auxiliary, continue enumeration in case 5954 // an auxiliary version of this effect type is available 5955 found = true; 5956 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5957 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5958 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5959 break; 5960 } 5961 } 5962 } 5963 if (!found) { 5964 lStatus = BAD_VALUE; 5965 ALOGW("createEffect() effect not found"); 5966 goto Exit; 5967 } 5968 // For same effect type, chose auxiliary version over insert version if 5969 // connect to output mix (Compliance to OpenSL ES) 5970 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5971 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5972 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5973 } 5974 } 5975 5976 // Do not allow auxiliary effects on a session different from 0 (output mix) 5977 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5978 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5979 lStatus = INVALID_OPERATION; 5980 goto Exit; 5981 } 5982 5983 // check recording permission for visualizer 5984 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5985 !recordingAllowed()) { 5986 lStatus = PERMISSION_DENIED; 5987 goto Exit; 5988 } 5989 5990 // return effect descriptor 5991 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5992 5993 // If output is not specified try to find a matching audio session ID in one of the 5994 // output threads. 5995 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5996 // because of code checking output when entering the function. 5997 // Note: io is never 0 when creating an effect on an input 5998 if (io == 0) { 5999 // look for the thread where the specified audio session is present 6000 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6001 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6002 io = mPlaybackThreads.keyAt(i); 6003 break; 6004 } 6005 } 6006 if (io == 0) { 6007 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6008 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6009 io = mRecordThreads.keyAt(i); 6010 break; 6011 } 6012 } 6013 } 6014 // If no output thread contains the requested session ID, default to 6015 // first output. The effect chain will be moved to the correct output 6016 // thread when a track with the same session ID is created 6017 if (io == 0 && mPlaybackThreads.size()) { 6018 io = mPlaybackThreads.keyAt(0); 6019 } 6020 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6021 } 6022 ThreadBase *thread = checkRecordThread_l(io); 6023 if (thread == NULL) { 6024 thread = checkPlaybackThread_l(io); 6025 if (thread == NULL) { 6026 ALOGE("createEffect() unknown output thread"); 6027 lStatus = BAD_VALUE; 6028 goto Exit; 6029 } 6030 } 6031 6032 sp<Client> client = registerPid_l(pid); 6033 6034 // create effect on selected output thread 6035 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6036 &desc, enabled, &lStatus); 6037 if (handle != 0 && id != NULL) { 6038 *id = handle->id(); 6039 } 6040 } 6041 6042Exit: 6043 if (status != NULL) { 6044 *status = lStatus; 6045 } 6046 return handle; 6047} 6048 6049status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6050 audio_io_handle_t dstOutput) 6051{ 6052 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6053 sessionId, srcOutput, dstOutput); 6054 Mutex::Autolock _l(mLock); 6055 if (srcOutput == dstOutput) { 6056 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6057 return NO_ERROR; 6058 } 6059 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6060 if (srcThread == NULL) { 6061 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6062 return BAD_VALUE; 6063 } 6064 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6065 if (dstThread == NULL) { 6066 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6067 return BAD_VALUE; 6068 } 6069 6070 Mutex::Autolock _dl(dstThread->mLock); 6071 Mutex::Autolock _sl(srcThread->mLock); 6072 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6073 6074 return NO_ERROR; 6075} 6076 6077// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6078status_t AudioFlinger::moveEffectChain_l(int sessionId, 6079 AudioFlinger::PlaybackThread *srcThread, 6080 AudioFlinger::PlaybackThread *dstThread, 6081 bool reRegister) 6082{ 6083 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6084 sessionId, srcThread, dstThread); 6085 6086 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6087 if (chain == 0) { 6088 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6089 sessionId, srcThread); 6090 return INVALID_OPERATION; 6091 } 6092 6093 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6094 // so that a new chain is created with correct parameters when first effect is added. This is 6095 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6096 // removed. 6097 srcThread->removeEffectChain_l(chain); 6098 6099 // transfer all effects one by one so that new effect chain is created on new thread with 6100 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6101 audio_io_handle_t dstOutput = dstThread->id(); 6102 sp<EffectChain> dstChain; 6103 uint32_t strategy = 0; // prevent compiler warning 6104 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6105 while (effect != 0) { 6106 srcThread->removeEffect_l(effect); 6107 dstThread->addEffect_l(effect); 6108 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6109 if (effect->state() == EffectModule::ACTIVE || 6110 effect->state() == EffectModule::STOPPING) { 6111 effect->start(); 6112 } 6113 // if the move request is not received from audio policy manager, the effect must be 6114 // re-registered with the new strategy and output 6115 if (dstChain == 0) { 6116 dstChain = effect->chain().promote(); 6117 if (dstChain == 0) { 6118 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6119 srcThread->addEffect_l(effect); 6120 return NO_INIT; 6121 } 6122 strategy = dstChain->strategy(); 6123 } 6124 if (reRegister) { 6125 AudioSystem::unregisterEffect(effect->id()); 6126 AudioSystem::registerEffect(&effect->desc(), 6127 dstOutput, 6128 strategy, 6129 sessionId, 6130 effect->id()); 6131 } 6132 effect = chain->getEffectFromId_l(0); 6133 } 6134 6135 return NO_ERROR; 6136} 6137 6138 6139// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6140sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6141 const sp<AudioFlinger::Client>& client, 6142 const sp<IEffectClient>& effectClient, 6143 int32_t priority, 6144 int sessionId, 6145 effect_descriptor_t *desc, 6146 int *enabled, 6147 status_t *status 6148 ) 6149{ 6150 sp<EffectModule> effect; 6151 sp<EffectHandle> handle; 6152 status_t lStatus; 6153 sp<EffectChain> chain; 6154 bool chainCreated = false; 6155 bool effectCreated = false; 6156 bool effectRegistered = false; 6157 6158 lStatus = initCheck(); 6159 if (lStatus != NO_ERROR) { 6160 ALOGW("createEffect_l() Audio driver not initialized."); 6161 goto Exit; 6162 } 6163 6164 // Do not allow effects with session ID 0 on direct output or duplicating threads 6165 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6166 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6167 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6168 desc->name, sessionId); 6169 lStatus = BAD_VALUE; 6170 goto Exit; 6171 } 6172 // Only Pre processor effects are allowed on input threads and only on input threads 6173 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6174 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6175 desc->name, desc->flags, mType); 6176 lStatus = BAD_VALUE; 6177 goto Exit; 6178 } 6179 6180 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6181 6182 { // scope for mLock 6183 Mutex::Autolock _l(mLock); 6184 6185 // check for existing effect chain with the requested audio session 6186 chain = getEffectChain_l(sessionId); 6187 if (chain == 0) { 6188 // create a new chain for this session 6189 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6190 chain = new EffectChain(this, sessionId); 6191 addEffectChain_l(chain); 6192 chain->setStrategy(getStrategyForSession_l(sessionId)); 6193 chainCreated = true; 6194 } else { 6195 effect = chain->getEffectFromDesc_l(desc); 6196 } 6197 6198 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6199 6200 if (effect == 0) { 6201 int id = mAudioFlinger->nextUniqueId(); 6202 // Check CPU and memory usage 6203 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6204 if (lStatus != NO_ERROR) { 6205 goto Exit; 6206 } 6207 effectRegistered = true; 6208 // create a new effect module if none present in the chain 6209 effect = new EffectModule(this, chain, desc, id, sessionId); 6210 lStatus = effect->status(); 6211 if (lStatus != NO_ERROR) { 6212 goto Exit; 6213 } 6214 lStatus = chain->addEffect_l(effect); 6215 if (lStatus != NO_ERROR) { 6216 goto Exit; 6217 } 6218 effectCreated = true; 6219 6220 effect->setDevice(mDevice); 6221 effect->setMode(mAudioFlinger->getMode()); 6222 } 6223 // create effect handle and connect it to effect module 6224 handle = new EffectHandle(effect, client, effectClient, priority); 6225 lStatus = effect->addHandle(handle); 6226 if (enabled != NULL) { 6227 *enabled = (int)effect->isEnabled(); 6228 } 6229 } 6230 6231Exit: 6232 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6233 Mutex::Autolock _l(mLock); 6234 if (effectCreated) { 6235 chain->removeEffect_l(effect); 6236 } 6237 if (effectRegistered) { 6238 AudioSystem::unregisterEffect(effect->id()); 6239 } 6240 if (chainCreated) { 6241 removeEffectChain_l(chain); 6242 } 6243 handle.clear(); 6244 } 6245 6246 if (status != NULL) { 6247 *status = lStatus; 6248 } 6249 return handle; 6250} 6251 6252sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6253{ 6254 sp<EffectChain> chain = getEffectChain_l(sessionId); 6255 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6256} 6257 6258// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6259// PlaybackThread::mLock held 6260status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6261{ 6262 // check for existing effect chain with the requested audio session 6263 int sessionId = effect->sessionId(); 6264 sp<EffectChain> chain = getEffectChain_l(sessionId); 6265 bool chainCreated = false; 6266 6267 if (chain == 0) { 6268 // create a new chain for this session 6269 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6270 chain = new EffectChain(this, sessionId); 6271 addEffectChain_l(chain); 6272 chain->setStrategy(getStrategyForSession_l(sessionId)); 6273 chainCreated = true; 6274 } 6275 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6276 6277 if (chain->getEffectFromId_l(effect->id()) != 0) { 6278 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6279 this, effect->desc().name, chain.get()); 6280 return BAD_VALUE; 6281 } 6282 6283 status_t status = chain->addEffect_l(effect); 6284 if (status != NO_ERROR) { 6285 if (chainCreated) { 6286 removeEffectChain_l(chain); 6287 } 6288 return status; 6289 } 6290 6291 effect->setDevice(mDevice); 6292 effect->setMode(mAudioFlinger->getMode()); 6293 return NO_ERROR; 6294} 6295 6296void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6297 6298 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6299 effect_descriptor_t desc = effect->desc(); 6300 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6301 detachAuxEffect_l(effect->id()); 6302 } 6303 6304 sp<EffectChain> chain = effect->chain().promote(); 6305 if (chain != 0) { 6306 // remove effect chain if removing last effect 6307 if (chain->removeEffect_l(effect) == 0) { 6308 removeEffectChain_l(chain); 6309 } 6310 } else { 6311 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6312 } 6313} 6314 6315void AudioFlinger::ThreadBase::lockEffectChains_l( 6316 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6317{ 6318 effectChains = mEffectChains; 6319 for (size_t i = 0; i < mEffectChains.size(); i++) { 6320 mEffectChains[i]->lock(); 6321 } 6322} 6323 6324void AudioFlinger::ThreadBase::unlockEffectChains( 6325 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6326{ 6327 for (size_t i = 0; i < effectChains.size(); i++) { 6328 effectChains[i]->unlock(); 6329 } 6330} 6331 6332sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6333{ 6334 Mutex::Autolock _l(mLock); 6335 return getEffectChain_l(sessionId); 6336} 6337 6338sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6339{ 6340 size_t size = mEffectChains.size(); 6341 for (size_t i = 0; i < size; i++) { 6342 if (mEffectChains[i]->sessionId() == sessionId) { 6343 return mEffectChains[i]; 6344 } 6345 } 6346 return 0; 6347} 6348 6349void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6350{ 6351 Mutex::Autolock _l(mLock); 6352 size_t size = mEffectChains.size(); 6353 for (size_t i = 0; i < size; i++) { 6354 mEffectChains[i]->setMode_l(mode); 6355 } 6356} 6357 6358void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6359 const wp<EffectHandle>& handle, 6360 bool unpinIfLast) { 6361 6362 Mutex::Autolock _l(mLock); 6363 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6364 // delete the effect module if removing last handle on it 6365 if (effect->removeHandle(handle) == 0) { 6366 if (!effect->isPinned() || unpinIfLast) { 6367 removeEffect_l(effect); 6368 AudioSystem::unregisterEffect(effect->id()); 6369 } 6370 } 6371} 6372 6373status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6374{ 6375 int session = chain->sessionId(); 6376 int16_t *buffer = mMixBuffer; 6377 bool ownsBuffer = false; 6378 6379 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6380 if (session > 0) { 6381 // Only one effect chain can be present in direct output thread and it uses 6382 // the mix buffer as input 6383 if (mType != DIRECT) { 6384 size_t numSamples = mFrameCount * mChannelCount; 6385 buffer = new int16_t[numSamples]; 6386 memset(buffer, 0, numSamples * sizeof(int16_t)); 6387 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6388 ownsBuffer = true; 6389 } 6390 6391 // Attach all tracks with same session ID to this chain. 6392 for (size_t i = 0; i < mTracks.size(); ++i) { 6393 sp<Track> track = mTracks[i]; 6394 if (session == track->sessionId()) { 6395 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6396 track->setMainBuffer(buffer); 6397 chain->incTrackCnt(); 6398 } 6399 } 6400 6401 // indicate all active tracks in the chain 6402 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6403 sp<Track> track = mActiveTracks[i].promote(); 6404 if (track == 0) continue; 6405 if (session == track->sessionId()) { 6406 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6407 chain->incActiveTrackCnt(); 6408 } 6409 } 6410 } 6411 6412 chain->setInBuffer(buffer, ownsBuffer); 6413 chain->setOutBuffer(mMixBuffer); 6414 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6415 // chains list in order to be processed last as it contains output stage effects 6416 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6417 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6418 // after track specific effects and before output stage 6419 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6420 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6421 // Effect chain for other sessions are inserted at beginning of effect 6422 // chains list to be processed before output mix effects. Relative order between other 6423 // sessions is not important 6424 size_t size = mEffectChains.size(); 6425 size_t i = 0; 6426 for (i = 0; i < size; i++) { 6427 if (mEffectChains[i]->sessionId() < session) break; 6428 } 6429 mEffectChains.insertAt(chain, i); 6430 checkSuspendOnAddEffectChain_l(chain); 6431 6432 return NO_ERROR; 6433} 6434 6435size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6436{ 6437 int session = chain->sessionId(); 6438 6439 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6440 6441 for (size_t i = 0; i < mEffectChains.size(); i++) { 6442 if (chain == mEffectChains[i]) { 6443 mEffectChains.removeAt(i); 6444 // detach all active tracks from the chain 6445 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6446 sp<Track> track = mActiveTracks[i].promote(); 6447 if (track == 0) continue; 6448 if (session == track->sessionId()) { 6449 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6450 chain.get(), session); 6451 chain->decActiveTrackCnt(); 6452 } 6453 } 6454 6455 // detach all tracks with same session ID from this chain 6456 for (size_t i = 0; i < mTracks.size(); ++i) { 6457 sp<Track> track = mTracks[i]; 6458 if (session == track->sessionId()) { 6459 track->setMainBuffer(mMixBuffer); 6460 chain->decTrackCnt(); 6461 } 6462 } 6463 break; 6464 } 6465 } 6466 return mEffectChains.size(); 6467} 6468 6469status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6470 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6471{ 6472 Mutex::Autolock _l(mLock); 6473 return attachAuxEffect_l(track, EffectId); 6474} 6475 6476status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6477 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6478{ 6479 status_t status = NO_ERROR; 6480 6481 if (EffectId == 0) { 6482 track->setAuxBuffer(0, NULL); 6483 } else { 6484 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6485 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6486 if (effect != 0) { 6487 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6488 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6489 } else { 6490 status = INVALID_OPERATION; 6491 } 6492 } else { 6493 status = BAD_VALUE; 6494 } 6495 } 6496 return status; 6497} 6498 6499void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6500{ 6501 for (size_t i = 0; i < mTracks.size(); ++i) { 6502 sp<Track> track = mTracks[i]; 6503 if (track->auxEffectId() == effectId) { 6504 attachAuxEffect_l(track, 0); 6505 } 6506 } 6507} 6508 6509status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6510{ 6511 // only one chain per input thread 6512 if (mEffectChains.size() != 0) { 6513 return INVALID_OPERATION; 6514 } 6515 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6516 6517 chain->setInBuffer(NULL); 6518 chain->setOutBuffer(NULL); 6519 6520 checkSuspendOnAddEffectChain_l(chain); 6521 6522 mEffectChains.add(chain); 6523 6524 return NO_ERROR; 6525} 6526 6527size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6528{ 6529 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6530 ALOGW_IF(mEffectChains.size() != 1, 6531 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6532 chain.get(), mEffectChains.size(), this); 6533 if (mEffectChains.size() == 1) { 6534 mEffectChains.removeAt(0); 6535 } 6536 return 0; 6537} 6538 6539// ---------------------------------------------------------------------------- 6540// EffectModule implementation 6541// ---------------------------------------------------------------------------- 6542 6543#undef LOG_TAG 6544#define LOG_TAG "AudioFlinger::EffectModule" 6545 6546AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6547 const wp<AudioFlinger::EffectChain>& chain, 6548 effect_descriptor_t *desc, 6549 int id, 6550 int sessionId) 6551 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6552 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6553{ 6554 ALOGV("Constructor %p", this); 6555 int lStatus; 6556 if (thread == NULL) { 6557 return; 6558 } 6559 6560 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6561 6562 // create effect engine from effect factory 6563 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6564 6565 if (mStatus != NO_ERROR) { 6566 return; 6567 } 6568 lStatus = init(); 6569 if (lStatus < 0) { 6570 mStatus = lStatus; 6571 goto Error; 6572 } 6573 6574 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6575 mPinned = true; 6576 } 6577 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6578 return; 6579Error: 6580 EffectRelease(mEffectInterface); 6581 mEffectInterface = NULL; 6582 ALOGV("Constructor Error %d", mStatus); 6583} 6584 6585AudioFlinger::EffectModule::~EffectModule() 6586{ 6587 ALOGV("Destructor %p", this); 6588 if (mEffectInterface != NULL) { 6589 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6590 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6591 sp<ThreadBase> thread = mThread.promote(); 6592 if (thread != 0) { 6593 audio_stream_t *stream = thread->stream(); 6594 if (stream != NULL) { 6595 stream->remove_audio_effect(stream, mEffectInterface); 6596 } 6597 } 6598 } 6599 // release effect engine 6600 EffectRelease(mEffectInterface); 6601 } 6602} 6603 6604status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6605{ 6606 status_t status; 6607 6608 Mutex::Autolock _l(mLock); 6609 int priority = handle->priority(); 6610 size_t size = mHandles.size(); 6611 sp<EffectHandle> h; 6612 size_t i; 6613 for (i = 0; i < size; i++) { 6614 h = mHandles[i].promote(); 6615 if (h == 0) continue; 6616 if (h->priority() <= priority) break; 6617 } 6618 // if inserted in first place, move effect control from previous owner to this handle 6619 if (i == 0) { 6620 bool enabled = false; 6621 if (h != 0) { 6622 enabled = h->enabled(); 6623 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6624 } 6625 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6626 status = NO_ERROR; 6627 } else { 6628 status = ALREADY_EXISTS; 6629 } 6630 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6631 mHandles.insertAt(handle, i); 6632 return status; 6633} 6634 6635size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6636{ 6637 Mutex::Autolock _l(mLock); 6638 size_t size = mHandles.size(); 6639 size_t i; 6640 for (i = 0; i < size; i++) { 6641 if (mHandles[i] == handle) break; 6642 } 6643 if (i == size) { 6644 return size; 6645 } 6646 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6647 6648 bool enabled = false; 6649 EffectHandle *hdl = handle.unsafe_get(); 6650 if (hdl != NULL) { 6651 ALOGV("removeHandle() unsafe_get OK"); 6652 enabled = hdl->enabled(); 6653 } 6654 mHandles.removeAt(i); 6655 size = mHandles.size(); 6656 // if removed from first place, move effect control from this handle to next in line 6657 if (i == 0 && size != 0) { 6658 sp<EffectHandle> h = mHandles[0].promote(); 6659 if (h != 0) { 6660 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6661 } 6662 } 6663 6664 // Prevent calls to process() and other functions on effect interface from now on. 6665 // The effect engine will be released by the destructor when the last strong reference on 6666 // this object is released which can happen after next process is called. 6667 if (size == 0 && !mPinned) { 6668 mState = DESTROYED; 6669 } 6670 6671 return size; 6672} 6673 6674sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6675{ 6676 Mutex::Autolock _l(mLock); 6677 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6678} 6679 6680void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6681{ 6682 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6683 // keep a strong reference on this EffectModule to avoid calling the 6684 // destructor before we exit 6685 sp<EffectModule> keep(this); 6686 { 6687 sp<ThreadBase> thread = mThread.promote(); 6688 if (thread != 0) { 6689 thread->disconnectEffect(keep, handle, unpinIfLast); 6690 } 6691 } 6692} 6693 6694void AudioFlinger::EffectModule::updateState() { 6695 Mutex::Autolock _l(mLock); 6696 6697 switch (mState) { 6698 case RESTART: 6699 reset_l(); 6700 // FALL THROUGH 6701 6702 case STARTING: 6703 // clear auxiliary effect input buffer for next accumulation 6704 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6705 memset(mConfig.inputCfg.buffer.raw, 6706 0, 6707 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6708 } 6709 start_l(); 6710 mState = ACTIVE; 6711 break; 6712 case STOPPING: 6713 stop_l(); 6714 mDisableWaitCnt = mMaxDisableWaitCnt; 6715 mState = STOPPED; 6716 break; 6717 case STOPPED: 6718 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6719 // turn off sequence. 6720 if (--mDisableWaitCnt == 0) { 6721 reset_l(); 6722 mState = IDLE; 6723 } 6724 break; 6725 default: //IDLE , ACTIVE, DESTROYED 6726 break; 6727 } 6728} 6729 6730void AudioFlinger::EffectModule::process() 6731{ 6732 Mutex::Autolock _l(mLock); 6733 6734 if (mState == DESTROYED || mEffectInterface == NULL || 6735 mConfig.inputCfg.buffer.raw == NULL || 6736 mConfig.outputCfg.buffer.raw == NULL) { 6737 return; 6738 } 6739 6740 if (isProcessEnabled()) { 6741 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6742 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6743 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6744 mConfig.inputCfg.buffer.s32, 6745 mConfig.inputCfg.buffer.frameCount/2); 6746 } 6747 6748 // do the actual processing in the effect engine 6749 int ret = (*mEffectInterface)->process(mEffectInterface, 6750 &mConfig.inputCfg.buffer, 6751 &mConfig.outputCfg.buffer); 6752 6753 // force transition to IDLE state when engine is ready 6754 if (mState == STOPPED && ret == -ENODATA) { 6755 mDisableWaitCnt = 1; 6756 } 6757 6758 // clear auxiliary effect input buffer for next accumulation 6759 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6760 memset(mConfig.inputCfg.buffer.raw, 0, 6761 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6762 } 6763 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6764 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6765 // If an insert effect is idle and input buffer is different from output buffer, 6766 // accumulate input onto output 6767 sp<EffectChain> chain = mChain.promote(); 6768 if (chain != 0 && chain->activeTrackCnt() != 0) { 6769 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6770 int16_t *in = mConfig.inputCfg.buffer.s16; 6771 int16_t *out = mConfig.outputCfg.buffer.s16; 6772 for (size_t i = 0; i < frameCnt; i++) { 6773 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6774 } 6775 } 6776 } 6777} 6778 6779void AudioFlinger::EffectModule::reset_l() 6780{ 6781 if (mEffectInterface == NULL) { 6782 return; 6783 } 6784 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6785} 6786 6787status_t AudioFlinger::EffectModule::configure() 6788{ 6789 uint32_t channels; 6790 if (mEffectInterface == NULL) { 6791 return NO_INIT; 6792 } 6793 6794 sp<ThreadBase> thread = mThread.promote(); 6795 if (thread == 0) { 6796 return DEAD_OBJECT; 6797 } 6798 6799 // TODO: handle configuration of effects replacing track process 6800 if (thread->channelCount() == 1) { 6801 channels = AUDIO_CHANNEL_OUT_MONO; 6802 } else { 6803 channels = AUDIO_CHANNEL_OUT_STEREO; 6804 } 6805 6806 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6807 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6808 } else { 6809 mConfig.inputCfg.channels = channels; 6810 } 6811 mConfig.outputCfg.channels = channels; 6812 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6813 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6814 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6815 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6816 mConfig.inputCfg.bufferProvider.cookie = NULL; 6817 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6818 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6819 mConfig.outputCfg.bufferProvider.cookie = NULL; 6820 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6821 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6822 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6823 // Insert effect: 6824 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6825 // always overwrites output buffer: input buffer == output buffer 6826 // - in other sessions: 6827 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6828 // other effect: overwrites output buffer: input buffer == output buffer 6829 // Auxiliary effect: 6830 // accumulates in output buffer: input buffer != output buffer 6831 // Therefore: accumulate <=> input buffer != output buffer 6832 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6833 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6834 } else { 6835 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6836 } 6837 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6838 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6839 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6840 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6841 6842 ALOGV("configure() %p thread %p buffer %p framecount %d", 6843 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6844 6845 status_t cmdStatus; 6846 uint32_t size = sizeof(int); 6847 status_t status = (*mEffectInterface)->command(mEffectInterface, 6848 EFFECT_CMD_SET_CONFIG, 6849 sizeof(effect_config_t), 6850 &mConfig, 6851 &size, 6852 &cmdStatus); 6853 if (status == 0) { 6854 status = cmdStatus; 6855 } 6856 6857 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6858 (1000 * mConfig.outputCfg.buffer.frameCount); 6859 6860 return status; 6861} 6862 6863status_t AudioFlinger::EffectModule::init() 6864{ 6865 Mutex::Autolock _l(mLock); 6866 if (mEffectInterface == NULL) { 6867 return NO_INIT; 6868 } 6869 status_t cmdStatus; 6870 uint32_t size = sizeof(status_t); 6871 status_t status = (*mEffectInterface)->command(mEffectInterface, 6872 EFFECT_CMD_INIT, 6873 0, 6874 NULL, 6875 &size, 6876 &cmdStatus); 6877 if (status == 0) { 6878 status = cmdStatus; 6879 } 6880 return status; 6881} 6882 6883status_t AudioFlinger::EffectModule::start() 6884{ 6885 Mutex::Autolock _l(mLock); 6886 return start_l(); 6887} 6888 6889status_t AudioFlinger::EffectModule::start_l() 6890{ 6891 if (mEffectInterface == NULL) { 6892 return NO_INIT; 6893 } 6894 status_t cmdStatus; 6895 uint32_t size = sizeof(status_t); 6896 status_t status = (*mEffectInterface)->command(mEffectInterface, 6897 EFFECT_CMD_ENABLE, 6898 0, 6899 NULL, 6900 &size, 6901 &cmdStatus); 6902 if (status == 0) { 6903 status = cmdStatus; 6904 } 6905 if (status == 0 && 6906 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6907 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6908 sp<ThreadBase> thread = mThread.promote(); 6909 if (thread != 0) { 6910 audio_stream_t *stream = thread->stream(); 6911 if (stream != NULL) { 6912 stream->add_audio_effect(stream, mEffectInterface); 6913 } 6914 } 6915 } 6916 return status; 6917} 6918 6919status_t AudioFlinger::EffectModule::stop() 6920{ 6921 Mutex::Autolock _l(mLock); 6922 return stop_l(); 6923} 6924 6925status_t AudioFlinger::EffectModule::stop_l() 6926{ 6927 if (mEffectInterface == NULL) { 6928 return NO_INIT; 6929 } 6930 status_t cmdStatus; 6931 uint32_t size = sizeof(status_t); 6932 status_t status = (*mEffectInterface)->command(mEffectInterface, 6933 EFFECT_CMD_DISABLE, 6934 0, 6935 NULL, 6936 &size, 6937 &cmdStatus); 6938 if (status == 0) { 6939 status = cmdStatus; 6940 } 6941 if (status == 0 && 6942 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6943 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6944 sp<ThreadBase> thread = mThread.promote(); 6945 if (thread != 0) { 6946 audio_stream_t *stream = thread->stream(); 6947 if (stream != NULL) { 6948 stream->remove_audio_effect(stream, mEffectInterface); 6949 } 6950 } 6951 } 6952 return status; 6953} 6954 6955status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6956 uint32_t cmdSize, 6957 void *pCmdData, 6958 uint32_t *replySize, 6959 void *pReplyData) 6960{ 6961 Mutex::Autolock _l(mLock); 6962// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6963 6964 if (mState == DESTROYED || mEffectInterface == NULL) { 6965 return NO_INIT; 6966 } 6967 status_t status = (*mEffectInterface)->command(mEffectInterface, 6968 cmdCode, 6969 cmdSize, 6970 pCmdData, 6971 replySize, 6972 pReplyData); 6973 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6974 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6975 for (size_t i = 1; i < mHandles.size(); i++) { 6976 sp<EffectHandle> h = mHandles[i].promote(); 6977 if (h != 0) { 6978 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6979 } 6980 } 6981 } 6982 return status; 6983} 6984 6985status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6986{ 6987 6988 Mutex::Autolock _l(mLock); 6989 ALOGV("setEnabled %p enabled %d", this, enabled); 6990 6991 if (enabled != isEnabled()) { 6992 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6993 if (enabled && status != NO_ERROR) { 6994 return status; 6995 } 6996 6997 switch (mState) { 6998 // going from disabled to enabled 6999 case IDLE: 7000 mState = STARTING; 7001 break; 7002 case STOPPED: 7003 mState = RESTART; 7004 break; 7005 case STOPPING: 7006 mState = ACTIVE; 7007 break; 7008 7009 // going from enabled to disabled 7010 case RESTART: 7011 mState = STOPPED; 7012 break; 7013 case STARTING: 7014 mState = IDLE; 7015 break; 7016 case ACTIVE: 7017 mState = STOPPING; 7018 break; 7019 case DESTROYED: 7020 return NO_ERROR; // simply ignore as we are being destroyed 7021 } 7022 for (size_t i = 1; i < mHandles.size(); i++) { 7023 sp<EffectHandle> h = mHandles[i].promote(); 7024 if (h != 0) { 7025 h->setEnabled(enabled); 7026 } 7027 } 7028 } 7029 return NO_ERROR; 7030} 7031 7032bool AudioFlinger::EffectModule::isEnabled() const 7033{ 7034 switch (mState) { 7035 case RESTART: 7036 case STARTING: 7037 case ACTIVE: 7038 return true; 7039 case IDLE: 7040 case STOPPING: 7041 case STOPPED: 7042 case DESTROYED: 7043 default: 7044 return false; 7045 } 7046} 7047 7048bool AudioFlinger::EffectModule::isProcessEnabled() const 7049{ 7050 switch (mState) { 7051 case RESTART: 7052 case ACTIVE: 7053 case STOPPING: 7054 case STOPPED: 7055 return true; 7056 case IDLE: 7057 case STARTING: 7058 case DESTROYED: 7059 default: 7060 return false; 7061 } 7062} 7063 7064status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7065{ 7066 Mutex::Autolock _l(mLock); 7067 status_t status = NO_ERROR; 7068 7069 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7070 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7071 if (isProcessEnabled() && 7072 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7073 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7074 status_t cmdStatus; 7075 uint32_t volume[2]; 7076 uint32_t *pVolume = NULL; 7077 uint32_t size = sizeof(volume); 7078 volume[0] = *left; 7079 volume[1] = *right; 7080 if (controller) { 7081 pVolume = volume; 7082 } 7083 status = (*mEffectInterface)->command(mEffectInterface, 7084 EFFECT_CMD_SET_VOLUME, 7085 size, 7086 volume, 7087 &size, 7088 pVolume); 7089 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7090 *left = volume[0]; 7091 *right = volume[1]; 7092 } 7093 } 7094 return status; 7095} 7096 7097status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7098{ 7099 Mutex::Autolock _l(mLock); 7100 status_t status = NO_ERROR; 7101 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7102 // audio pre processing modules on RecordThread can receive both output and 7103 // input device indication in the same call 7104 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7105 if (dev) { 7106 status_t cmdStatus; 7107 uint32_t size = sizeof(status_t); 7108 7109 status = (*mEffectInterface)->command(mEffectInterface, 7110 EFFECT_CMD_SET_DEVICE, 7111 sizeof(uint32_t), 7112 &dev, 7113 &size, 7114 &cmdStatus); 7115 if (status == NO_ERROR) { 7116 status = cmdStatus; 7117 } 7118 } 7119 dev = device & AUDIO_DEVICE_IN_ALL; 7120 if (dev) { 7121 status_t cmdStatus; 7122 uint32_t size = sizeof(status_t); 7123 7124 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7125 EFFECT_CMD_SET_INPUT_DEVICE, 7126 sizeof(uint32_t), 7127 &dev, 7128 &size, 7129 &cmdStatus); 7130 if (status2 == NO_ERROR) { 7131 status2 = cmdStatus; 7132 } 7133 if (status == NO_ERROR) { 7134 status = status2; 7135 } 7136 } 7137 } 7138 return status; 7139} 7140 7141status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7142{ 7143 Mutex::Autolock _l(mLock); 7144 status_t status = NO_ERROR; 7145 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7146 status_t cmdStatus; 7147 uint32_t size = sizeof(status_t); 7148 status = (*mEffectInterface)->command(mEffectInterface, 7149 EFFECT_CMD_SET_AUDIO_MODE, 7150 sizeof(audio_mode_t), 7151 &mode, 7152 &size, 7153 &cmdStatus); 7154 if (status == NO_ERROR) { 7155 status = cmdStatus; 7156 } 7157 } 7158 return status; 7159} 7160 7161void AudioFlinger::EffectModule::setSuspended(bool suspended) 7162{ 7163 Mutex::Autolock _l(mLock); 7164 mSuspended = suspended; 7165} 7166 7167bool AudioFlinger::EffectModule::suspended() const 7168{ 7169 Mutex::Autolock _l(mLock); 7170 return mSuspended; 7171} 7172 7173status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7174{ 7175 const size_t SIZE = 256; 7176 char buffer[SIZE]; 7177 String8 result; 7178 7179 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7180 result.append(buffer); 7181 7182 bool locked = tryLock(mLock); 7183 // failed to lock - AudioFlinger is probably deadlocked 7184 if (!locked) { 7185 result.append("\t\tCould not lock Fx mutex:\n"); 7186 } 7187 7188 result.append("\t\tSession Status State Engine:\n"); 7189 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7190 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7191 result.append(buffer); 7192 7193 result.append("\t\tDescriptor:\n"); 7194 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7195 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7196 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7197 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7198 result.append(buffer); 7199 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7200 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7201 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7202 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7203 result.append(buffer); 7204 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7205 mDescriptor.apiVersion, 7206 mDescriptor.flags); 7207 result.append(buffer); 7208 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7209 mDescriptor.name); 7210 result.append(buffer); 7211 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7212 mDescriptor.implementor); 7213 result.append(buffer); 7214 7215 result.append("\t\t- Input configuration:\n"); 7216 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7217 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7218 (uint32_t)mConfig.inputCfg.buffer.raw, 7219 mConfig.inputCfg.buffer.frameCount, 7220 mConfig.inputCfg.samplingRate, 7221 mConfig.inputCfg.channels, 7222 mConfig.inputCfg.format); 7223 result.append(buffer); 7224 7225 result.append("\t\t- Output configuration:\n"); 7226 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7227 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7228 (uint32_t)mConfig.outputCfg.buffer.raw, 7229 mConfig.outputCfg.buffer.frameCount, 7230 mConfig.outputCfg.samplingRate, 7231 mConfig.outputCfg.channels, 7232 mConfig.outputCfg.format); 7233 result.append(buffer); 7234 7235 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7236 result.append(buffer); 7237 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7238 for (size_t i = 0; i < mHandles.size(); ++i) { 7239 sp<EffectHandle> handle = mHandles[i].promote(); 7240 if (handle != 0) { 7241 handle->dump(buffer, SIZE); 7242 result.append(buffer); 7243 } 7244 } 7245 7246 result.append("\n"); 7247 7248 write(fd, result.string(), result.length()); 7249 7250 if (locked) { 7251 mLock.unlock(); 7252 } 7253 7254 return NO_ERROR; 7255} 7256 7257// ---------------------------------------------------------------------------- 7258// EffectHandle implementation 7259// ---------------------------------------------------------------------------- 7260 7261#undef LOG_TAG 7262#define LOG_TAG "AudioFlinger::EffectHandle" 7263 7264AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7265 const sp<AudioFlinger::Client>& client, 7266 const sp<IEffectClient>& effectClient, 7267 int32_t priority) 7268 : BnEffect(), 7269 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7270 mPriority(priority), mHasControl(false), mEnabled(false) 7271{ 7272 ALOGV("constructor %p", this); 7273 7274 if (client == 0) { 7275 return; 7276 } 7277 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7278 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7279 if (mCblkMemory != 0) { 7280 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7281 7282 if (mCblk != NULL) { 7283 new(mCblk) effect_param_cblk_t(); 7284 mBuffer = (uint8_t *)mCblk + bufOffset; 7285 } 7286 } else { 7287 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7288 return; 7289 } 7290} 7291 7292AudioFlinger::EffectHandle::~EffectHandle() 7293{ 7294 ALOGV("Destructor %p", this); 7295 disconnect(false); 7296 ALOGV("Destructor DONE %p", this); 7297} 7298 7299status_t AudioFlinger::EffectHandle::enable() 7300{ 7301 ALOGV("enable %p", this); 7302 if (!mHasControl) return INVALID_OPERATION; 7303 if (mEffect == 0) return DEAD_OBJECT; 7304 7305 if (mEnabled) { 7306 return NO_ERROR; 7307 } 7308 7309 mEnabled = true; 7310 7311 sp<ThreadBase> thread = mEffect->thread().promote(); 7312 if (thread != 0) { 7313 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7314 } 7315 7316 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7317 if (mEffect->suspended()) { 7318 return NO_ERROR; 7319 } 7320 7321 status_t status = mEffect->setEnabled(true); 7322 if (status != NO_ERROR) { 7323 if (thread != 0) { 7324 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7325 } 7326 mEnabled = false; 7327 } 7328 return status; 7329} 7330 7331status_t AudioFlinger::EffectHandle::disable() 7332{ 7333 ALOGV("disable %p", this); 7334 if (!mHasControl) return INVALID_OPERATION; 7335 if (mEffect == 0) return DEAD_OBJECT; 7336 7337 if (!mEnabled) { 7338 return NO_ERROR; 7339 } 7340 mEnabled = false; 7341 7342 if (mEffect->suspended()) { 7343 return NO_ERROR; 7344 } 7345 7346 status_t status = mEffect->setEnabled(false); 7347 7348 sp<ThreadBase> thread = mEffect->thread().promote(); 7349 if (thread != 0) { 7350 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7351 } 7352 7353 return status; 7354} 7355 7356void AudioFlinger::EffectHandle::disconnect() 7357{ 7358 disconnect(true); 7359} 7360 7361void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7362{ 7363 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7364 if (mEffect == 0) { 7365 return; 7366 } 7367 mEffect->disconnect(this, unpinIfLast); 7368 7369 if (mHasControl && mEnabled) { 7370 sp<ThreadBase> thread = mEffect->thread().promote(); 7371 if (thread != 0) { 7372 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7373 } 7374 } 7375 7376 // release sp on module => module destructor can be called now 7377 mEffect.clear(); 7378 if (mClient != 0) { 7379 if (mCblk != NULL) { 7380 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7381 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7382 } 7383 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7384 // Client destructor must run with AudioFlinger mutex locked 7385 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7386 mClient.clear(); 7387 } 7388} 7389 7390status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7391 uint32_t cmdSize, 7392 void *pCmdData, 7393 uint32_t *replySize, 7394 void *pReplyData) 7395{ 7396// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7397// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7398 7399 // only get parameter command is permitted for applications not controlling the effect 7400 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7401 return INVALID_OPERATION; 7402 } 7403 if (mEffect == 0) return DEAD_OBJECT; 7404 if (mClient == 0) return INVALID_OPERATION; 7405 7406 // handle commands that are not forwarded transparently to effect engine 7407 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7408 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7409 // no risk to block the whole media server process or mixer threads is we are stuck here 7410 Mutex::Autolock _l(mCblk->lock); 7411 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7412 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7413 mCblk->serverIndex = 0; 7414 mCblk->clientIndex = 0; 7415 return BAD_VALUE; 7416 } 7417 status_t status = NO_ERROR; 7418 while (mCblk->serverIndex < mCblk->clientIndex) { 7419 int reply; 7420 uint32_t rsize = sizeof(int); 7421 int *p = (int *)(mBuffer + mCblk->serverIndex); 7422 int size = *p++; 7423 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7424 ALOGW("command(): invalid parameter block size"); 7425 break; 7426 } 7427 effect_param_t *param = (effect_param_t *)p; 7428 if (param->psize == 0 || param->vsize == 0) { 7429 ALOGW("command(): null parameter or value size"); 7430 mCblk->serverIndex += size; 7431 continue; 7432 } 7433 uint32_t psize = sizeof(effect_param_t) + 7434 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7435 param->vsize; 7436 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7437 psize, 7438 p, 7439 &rsize, 7440 &reply); 7441 // stop at first error encountered 7442 if (ret != NO_ERROR) { 7443 status = ret; 7444 *(int *)pReplyData = reply; 7445 break; 7446 } else if (reply != NO_ERROR) { 7447 *(int *)pReplyData = reply; 7448 break; 7449 } 7450 mCblk->serverIndex += size; 7451 } 7452 mCblk->serverIndex = 0; 7453 mCblk->clientIndex = 0; 7454 return status; 7455 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7456 *(int *)pReplyData = NO_ERROR; 7457 return enable(); 7458 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7459 *(int *)pReplyData = NO_ERROR; 7460 return disable(); 7461 } 7462 7463 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7464} 7465 7466void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7467{ 7468 ALOGV("setControl %p control %d", this, hasControl); 7469 7470 mHasControl = hasControl; 7471 mEnabled = enabled; 7472 7473 if (signal && mEffectClient != 0) { 7474 mEffectClient->controlStatusChanged(hasControl); 7475 } 7476} 7477 7478void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7479 uint32_t cmdSize, 7480 void *pCmdData, 7481 uint32_t replySize, 7482 void *pReplyData) 7483{ 7484 if (mEffectClient != 0) { 7485 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7486 } 7487} 7488 7489 7490 7491void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7492{ 7493 if (mEffectClient != 0) { 7494 mEffectClient->enableStatusChanged(enabled); 7495 } 7496} 7497 7498status_t AudioFlinger::EffectHandle::onTransact( 7499 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7500{ 7501 return BnEffect::onTransact(code, data, reply, flags); 7502} 7503 7504 7505void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7506{ 7507 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7508 7509 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7510 (mClient == 0) ? getpid_cached : mClient->pid(), 7511 mPriority, 7512 mHasControl, 7513 !locked, 7514 mCblk ? mCblk->clientIndex : 0, 7515 mCblk ? mCblk->serverIndex : 0 7516 ); 7517 7518 if (locked) { 7519 mCblk->lock.unlock(); 7520 } 7521} 7522 7523#undef LOG_TAG 7524#define LOG_TAG "AudioFlinger::EffectChain" 7525 7526AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7527 int sessionId) 7528 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7529 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7530 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7531{ 7532 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7533 if (thread == NULL) { 7534 return; 7535 } 7536 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7537 thread->frameCount(); 7538} 7539 7540AudioFlinger::EffectChain::~EffectChain() 7541{ 7542 if (mOwnInBuffer) { 7543 delete mInBuffer; 7544 } 7545 7546} 7547 7548// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7549sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7550{ 7551 size_t size = mEffects.size(); 7552 7553 for (size_t i = 0; i < size; i++) { 7554 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7555 return mEffects[i]; 7556 } 7557 } 7558 return 0; 7559} 7560 7561// getEffectFromId_l() must be called with ThreadBase::mLock held 7562sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7563{ 7564 size_t size = mEffects.size(); 7565 7566 for (size_t i = 0; i < size; i++) { 7567 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7568 if (id == 0 || mEffects[i]->id() == id) { 7569 return mEffects[i]; 7570 } 7571 } 7572 return 0; 7573} 7574 7575// getEffectFromType_l() must be called with ThreadBase::mLock held 7576sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7577 const effect_uuid_t *type) 7578{ 7579 size_t size = mEffects.size(); 7580 7581 for (size_t i = 0; i < size; i++) { 7582 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7583 return mEffects[i]; 7584 } 7585 } 7586 return 0; 7587} 7588 7589// Must be called with EffectChain::mLock locked 7590void AudioFlinger::EffectChain::process_l() 7591{ 7592 sp<ThreadBase> thread = mThread.promote(); 7593 if (thread == 0) { 7594 ALOGW("process_l(): cannot promote mixer thread"); 7595 return; 7596 } 7597 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7598 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7599 // always process effects unless no more tracks are on the session and the effect tail 7600 // has been rendered 7601 bool doProcess = true; 7602 if (!isGlobalSession) { 7603 bool tracksOnSession = (trackCnt() != 0); 7604 7605 if (!tracksOnSession && mTailBufferCount == 0) { 7606 doProcess = false; 7607 } 7608 7609 if (activeTrackCnt() == 0) { 7610 // if no track is active and the effect tail has not been rendered, 7611 // the input buffer must be cleared here as the mixer process will not do it 7612 if (tracksOnSession || mTailBufferCount > 0) { 7613 size_t numSamples = thread->frameCount() * thread->channelCount(); 7614 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7615 if (mTailBufferCount > 0) { 7616 mTailBufferCount--; 7617 } 7618 } 7619 } 7620 } 7621 7622 size_t size = mEffects.size(); 7623 if (doProcess) { 7624 for (size_t i = 0; i < size; i++) { 7625 mEffects[i]->process(); 7626 } 7627 } 7628 for (size_t i = 0; i < size; i++) { 7629 mEffects[i]->updateState(); 7630 } 7631} 7632 7633// addEffect_l() must be called with PlaybackThread::mLock held 7634status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7635{ 7636 effect_descriptor_t desc = effect->desc(); 7637 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7638 7639 Mutex::Autolock _l(mLock); 7640 effect->setChain(this); 7641 sp<ThreadBase> thread = mThread.promote(); 7642 if (thread == 0) { 7643 return NO_INIT; 7644 } 7645 effect->setThread(thread); 7646 7647 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7648 // Auxiliary effects are inserted at the beginning of mEffects vector as 7649 // they are processed first and accumulated in chain input buffer 7650 mEffects.insertAt(effect, 0); 7651 7652 // the input buffer for auxiliary effect contains mono samples in 7653 // 32 bit format. This is to avoid saturation in AudoMixer 7654 // accumulation stage. Saturation is done in EffectModule::process() before 7655 // calling the process in effect engine 7656 size_t numSamples = thread->frameCount(); 7657 int32_t *buffer = new int32_t[numSamples]; 7658 memset(buffer, 0, numSamples * sizeof(int32_t)); 7659 effect->setInBuffer((int16_t *)buffer); 7660 // auxiliary effects output samples to chain input buffer for further processing 7661 // by insert effects 7662 effect->setOutBuffer(mInBuffer); 7663 } else { 7664 // Insert effects are inserted at the end of mEffects vector as they are processed 7665 // after track and auxiliary effects. 7666 // Insert effect order as a function of indicated preference: 7667 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7668 // another effect is present 7669 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7670 // last effect claiming first position 7671 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7672 // first effect claiming last position 7673 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7674 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7675 // already present 7676 7677 size_t size = mEffects.size(); 7678 size_t idx_insert = size; 7679 ssize_t idx_insert_first = -1; 7680 ssize_t idx_insert_last = -1; 7681 7682 for (size_t i = 0; i < size; i++) { 7683 effect_descriptor_t d = mEffects[i]->desc(); 7684 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7685 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7686 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7687 // check invalid effect chaining combinations 7688 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7689 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7690 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7691 return INVALID_OPERATION; 7692 } 7693 // remember position of first insert effect and by default 7694 // select this as insert position for new effect 7695 if (idx_insert == size) { 7696 idx_insert = i; 7697 } 7698 // remember position of last insert effect claiming 7699 // first position 7700 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7701 idx_insert_first = i; 7702 } 7703 // remember position of first insert effect claiming 7704 // last position 7705 if (iPref == EFFECT_FLAG_INSERT_LAST && 7706 idx_insert_last == -1) { 7707 idx_insert_last = i; 7708 } 7709 } 7710 } 7711 7712 // modify idx_insert from first position if needed 7713 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7714 if (idx_insert_last != -1) { 7715 idx_insert = idx_insert_last; 7716 } else { 7717 idx_insert = size; 7718 } 7719 } else { 7720 if (idx_insert_first != -1) { 7721 idx_insert = idx_insert_first + 1; 7722 } 7723 } 7724 7725 // always read samples from chain input buffer 7726 effect->setInBuffer(mInBuffer); 7727 7728 // if last effect in the chain, output samples to chain 7729 // output buffer, otherwise to chain input buffer 7730 if (idx_insert == size) { 7731 if (idx_insert != 0) { 7732 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7733 mEffects[idx_insert-1]->configure(); 7734 } 7735 effect->setOutBuffer(mOutBuffer); 7736 } else { 7737 effect->setOutBuffer(mInBuffer); 7738 } 7739 mEffects.insertAt(effect, idx_insert); 7740 7741 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7742 } 7743 effect->configure(); 7744 return NO_ERROR; 7745} 7746 7747// removeEffect_l() must be called with PlaybackThread::mLock held 7748size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7749{ 7750 Mutex::Autolock _l(mLock); 7751 size_t size = mEffects.size(); 7752 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7753 7754 for (size_t i = 0; i < size; i++) { 7755 if (effect == mEffects[i]) { 7756 // calling stop here will remove pre-processing effect from the audio HAL. 7757 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7758 // the middle of a read from audio HAL 7759 if (mEffects[i]->state() == EffectModule::ACTIVE || 7760 mEffects[i]->state() == EffectModule::STOPPING) { 7761 mEffects[i]->stop(); 7762 } 7763 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7764 delete[] effect->inBuffer(); 7765 } else { 7766 if (i == size - 1 && i != 0) { 7767 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7768 mEffects[i - 1]->configure(); 7769 } 7770 } 7771 mEffects.removeAt(i); 7772 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7773 break; 7774 } 7775 } 7776 7777 return mEffects.size(); 7778} 7779 7780// setDevice_l() must be called with PlaybackThread::mLock held 7781void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7782{ 7783 size_t size = mEffects.size(); 7784 for (size_t i = 0; i < size; i++) { 7785 mEffects[i]->setDevice(device); 7786 } 7787} 7788 7789// setMode_l() must be called with PlaybackThread::mLock held 7790void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7791{ 7792 size_t size = mEffects.size(); 7793 for (size_t i = 0; i < size; i++) { 7794 mEffects[i]->setMode(mode); 7795 } 7796} 7797 7798// setVolume_l() must be called with PlaybackThread::mLock held 7799bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7800{ 7801 uint32_t newLeft = *left; 7802 uint32_t newRight = *right; 7803 bool hasControl = false; 7804 int ctrlIdx = -1; 7805 size_t size = mEffects.size(); 7806 7807 // first update volume controller 7808 for (size_t i = size; i > 0; i--) { 7809 if (mEffects[i - 1]->isProcessEnabled() && 7810 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7811 ctrlIdx = i - 1; 7812 hasControl = true; 7813 break; 7814 } 7815 } 7816 7817 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7818 if (hasControl) { 7819 *left = mNewLeftVolume; 7820 *right = mNewRightVolume; 7821 } 7822 return hasControl; 7823 } 7824 7825 mVolumeCtrlIdx = ctrlIdx; 7826 mLeftVolume = newLeft; 7827 mRightVolume = newRight; 7828 7829 // second get volume update from volume controller 7830 if (ctrlIdx >= 0) { 7831 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7832 mNewLeftVolume = newLeft; 7833 mNewRightVolume = newRight; 7834 } 7835 // then indicate volume to all other effects in chain. 7836 // Pass altered volume to effects before volume controller 7837 // and requested volume to effects after controller 7838 uint32_t lVol = newLeft; 7839 uint32_t rVol = newRight; 7840 7841 for (size_t i = 0; i < size; i++) { 7842 if ((int)i == ctrlIdx) continue; 7843 // this also works for ctrlIdx == -1 when there is no volume controller 7844 if ((int)i > ctrlIdx) { 7845 lVol = *left; 7846 rVol = *right; 7847 } 7848 mEffects[i]->setVolume(&lVol, &rVol, false); 7849 } 7850 *left = newLeft; 7851 *right = newRight; 7852 7853 return hasControl; 7854} 7855 7856status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7857{ 7858 const size_t SIZE = 256; 7859 char buffer[SIZE]; 7860 String8 result; 7861 7862 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7863 result.append(buffer); 7864 7865 bool locked = tryLock(mLock); 7866 // failed to lock - AudioFlinger is probably deadlocked 7867 if (!locked) { 7868 result.append("\tCould not lock mutex:\n"); 7869 } 7870 7871 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7872 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7873 mEffects.size(), 7874 (uint32_t)mInBuffer, 7875 (uint32_t)mOutBuffer, 7876 mActiveTrackCnt); 7877 result.append(buffer); 7878 write(fd, result.string(), result.size()); 7879 7880 for (size_t i = 0; i < mEffects.size(); ++i) { 7881 sp<EffectModule> effect = mEffects[i]; 7882 if (effect != 0) { 7883 effect->dump(fd, args); 7884 } 7885 } 7886 7887 if (locked) { 7888 mLock.unlock(); 7889 } 7890 7891 return NO_ERROR; 7892} 7893 7894// must be called with ThreadBase::mLock held 7895void AudioFlinger::EffectChain::setEffectSuspended_l( 7896 const effect_uuid_t *type, bool suspend) 7897{ 7898 sp<SuspendedEffectDesc> desc; 7899 // use effect type UUID timelow as key as there is no real risk of identical 7900 // timeLow fields among effect type UUIDs. 7901 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7902 if (suspend) { 7903 if (index >= 0) { 7904 desc = mSuspendedEffects.valueAt(index); 7905 } else { 7906 desc = new SuspendedEffectDesc(); 7907 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7908 mSuspendedEffects.add(type->timeLow, desc); 7909 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7910 } 7911 if (desc->mRefCount++ == 0) { 7912 sp<EffectModule> effect = getEffectIfEnabled(type); 7913 if (effect != 0) { 7914 desc->mEffect = effect; 7915 effect->setSuspended(true); 7916 effect->setEnabled(false); 7917 } 7918 } 7919 } else { 7920 if (index < 0) { 7921 return; 7922 } 7923 desc = mSuspendedEffects.valueAt(index); 7924 if (desc->mRefCount <= 0) { 7925 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7926 desc->mRefCount = 1; 7927 } 7928 if (--desc->mRefCount == 0) { 7929 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7930 if (desc->mEffect != 0) { 7931 sp<EffectModule> effect = desc->mEffect.promote(); 7932 if (effect != 0) { 7933 effect->setSuspended(false); 7934 sp<EffectHandle> handle = effect->controlHandle(); 7935 if (handle != 0) { 7936 effect->setEnabled(handle->enabled()); 7937 } 7938 } 7939 desc->mEffect.clear(); 7940 } 7941 mSuspendedEffects.removeItemsAt(index); 7942 } 7943 } 7944} 7945 7946// must be called with ThreadBase::mLock held 7947void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7948{ 7949 sp<SuspendedEffectDesc> desc; 7950 7951 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7952 if (suspend) { 7953 if (index >= 0) { 7954 desc = mSuspendedEffects.valueAt(index); 7955 } else { 7956 desc = new SuspendedEffectDesc(); 7957 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7958 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7959 } 7960 if (desc->mRefCount++ == 0) { 7961 Vector< sp<EffectModule> > effects; 7962 getSuspendEligibleEffects(effects); 7963 for (size_t i = 0; i < effects.size(); i++) { 7964 setEffectSuspended_l(&effects[i]->desc().type, true); 7965 } 7966 } 7967 } else { 7968 if (index < 0) { 7969 return; 7970 } 7971 desc = mSuspendedEffects.valueAt(index); 7972 if (desc->mRefCount <= 0) { 7973 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7974 desc->mRefCount = 1; 7975 } 7976 if (--desc->mRefCount == 0) { 7977 Vector<const effect_uuid_t *> types; 7978 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7979 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7980 continue; 7981 } 7982 types.add(&mSuspendedEffects.valueAt(i)->mType); 7983 } 7984 for (size_t i = 0; i < types.size(); i++) { 7985 setEffectSuspended_l(types[i], false); 7986 } 7987 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7988 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7989 } 7990 } 7991} 7992 7993 7994// The volume effect is used for automated tests only 7995#ifndef OPENSL_ES_H_ 7996static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7997 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7998const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7999#endif //OPENSL_ES_H_ 8000 8001bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8002{ 8003 // auxiliary effects and visualizer are never suspended on output mix 8004 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8005 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8006 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8007 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8008 return false; 8009 } 8010 return true; 8011} 8012 8013void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8014{ 8015 effects.clear(); 8016 for (size_t i = 0; i < mEffects.size(); i++) { 8017 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8018 effects.add(mEffects[i]); 8019 } 8020 } 8021} 8022 8023sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8024 const effect_uuid_t *type) 8025{ 8026 sp<EffectModule> effect = getEffectFromType_l(type); 8027 return effect != 0 && effect->isEnabled() ? effect : 0; 8028} 8029 8030void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8031 bool enabled) 8032{ 8033 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8034 if (enabled) { 8035 if (index < 0) { 8036 // if the effect is not suspend check if all effects are suspended 8037 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8038 if (index < 0) { 8039 return; 8040 } 8041 if (!isEffectEligibleForSuspend(effect->desc())) { 8042 return; 8043 } 8044 setEffectSuspended_l(&effect->desc().type, enabled); 8045 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8046 if (index < 0) { 8047 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8048 return; 8049 } 8050 } 8051 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8052 effect->desc().type.timeLow); 8053 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8054 // if effect is requested to suspended but was not yet enabled, supend it now. 8055 if (desc->mEffect == 0) { 8056 desc->mEffect = effect; 8057 effect->setEnabled(false); 8058 effect->setSuspended(true); 8059 } 8060 } else { 8061 if (index < 0) { 8062 return; 8063 } 8064 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8065 effect->desc().type.timeLow); 8066 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8067 desc->mEffect.clear(); 8068 effect->setSuspended(false); 8069 } 8070} 8071 8072#undef LOG_TAG 8073#define LOG_TAG "AudioFlinger" 8074 8075// ---------------------------------------------------------------------------- 8076 8077status_t AudioFlinger::onTransact( 8078 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8079{ 8080 return BnAudioFlinger::onTransact(code, data, reply, flags); 8081} 8082 8083}; // namespace android 8084