AudioFlinger.cpp revision 810280460da5000785662f6c5b0c7ff3ee0a4cb3
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 <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75#include "FastMixer.h" 76 77// NBAIO implementations 78#include "AudioStreamOutSink.h" 79#include "MonoPipe.h" 80#include "MonoPipeReader.h" 81#include "SourceAudioBufferProvider.h" 82 83#ifdef HAVE_REQUEST_PRIORITY 84#include "SchedulingPolicyService.h" 85#endif 86 87#ifdef SOAKER 88#include "Soaker.h" 89#endif 90 91// ---------------------------------------------------------------------------- 92 93// Note: the following macro is used for extremely verbose logging message. In 94// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 95// 0; but one side effect of this is to turn all LOGV's as well. Some messages 96// are so verbose that we want to suppress them even when we have ALOG_ASSERT 97// turned on. Do not uncomment the #def below unless you really know what you 98// are doing and want to see all of the extremely verbose messages. 99//#define VERY_VERY_VERBOSE_LOGGING 100#ifdef VERY_VERY_VERBOSE_LOGGING 101#define ALOGVV ALOGV 102#else 103#define ALOGVV(a...) do { } while(0) 104#endif 105 106namespace android { 107 108static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 109static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 110 111static const float MAX_GAIN = 4096.0f; 112static const uint32_t MAX_GAIN_INT = 0x1000; 113 114// retry counts for buffer fill timeout 115// 50 * ~20msecs = 1 second 116static const int8_t kMaxTrackRetries = 50; 117static const int8_t kMaxTrackStartupRetries = 50; 118// allow less retry attempts on direct output thread. 119// direct outputs can be a scarce resource in audio hardware and should 120// be released as quickly as possible. 121static const int8_t kMaxTrackRetriesDirect = 2; 122 123static const int kDumpLockRetries = 50; 124static const int kDumpLockSleepUs = 20000; 125 126// don't warn about blocked writes or record buffer overflows more often than this 127static const nsecs_t kWarningThrottleNs = seconds(5); 128 129// RecordThread loop sleep time upon application overrun or audio HAL read error 130static const int kRecordThreadSleepUs = 5000; 131 132// maximum time to wait for setParameters to complete 133static const nsecs_t kSetParametersTimeoutNs = seconds(2); 134 135// minimum sleep time for the mixer thread loop when tracks are active but in underrun 136static const uint32_t kMinThreadSleepTimeUs = 5000; 137// maximum divider applied to the active sleep time in the mixer thread loop 138static const uint32_t kMaxThreadSleepTimeShift = 2; 139 140// minimum normal mix buffer size, expressed in milliseconds rather than frames 141static const uint32_t kMinNormalMixBufferSizeMs = 20; 142 143nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 144 145// Whether to use fast mixer 146static const enum { 147 FastMixer_Never, // never initialize or use: for debugging only 148 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 149 // normal mixer multiplier is 1 150 FastMixer_Static, // initialize if needed, then use all the time if initialized, 151 // multipler is calculated based on minimum normal mixer buffer size 152 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 153 // multipler is calculated based on minimum normal mixer buffer size 154 // FIXME for FastMixer_Dynamic: 155 // Supporting this option will require fixing HALs that can't handle large writes. 156 // For example, one HAL implementation returns an error from a large write, 157 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 158 // We could either fix the HAL implementations, or provide a wrapper that breaks 159 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 160} kUseFastMixer = FastMixer_Static; 161 162// ---------------------------------------------------------------------------- 163 164#ifdef ADD_BATTERY_DATA 165// To collect the amplifier usage 166static void addBatteryData(uint32_t params) { 167 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 168 if (service == NULL) { 169 // it already logged 170 return; 171 } 172 173 service->addBatteryData(params); 174} 175#endif 176 177static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 178{ 179 const hw_module_t *mod; 180 int rc; 181 182 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 183 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 184 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 185 if (rc) { 186 goto out; 187 } 188 rc = audio_hw_device_open(mod, dev); 189 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 190 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 191 if (rc) { 192 goto out; 193 } 194 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 195 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 196 rc = BAD_VALUE; 197 goto out; 198 } 199 return 0; 200 201out: 202 *dev = NULL; 203 return rc; 204} 205 206// ---------------------------------------------------------------------------- 207 208AudioFlinger::AudioFlinger() 209 : BnAudioFlinger(), 210 mPrimaryHardwareDev(NULL), 211 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 212 mMasterVolume(1.0f), 213 mMasterVolumeSupportLvl(MVS_NONE), 214 mMasterMute(false), 215 mNextUniqueId(1), 216 mMode(AUDIO_MODE_INVALID), 217 mBtNrecIsOff(false) 218{ 219} 220 221void AudioFlinger::onFirstRef() 222{ 223 int rc = 0; 224 225 Mutex::Autolock _l(mLock); 226 227 /* TODO: move all this work into an Init() function */ 228 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 229 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 230 uint32_t int_val; 231 if (1 == sscanf(val_str, "%u", &int_val)) { 232 mStandbyTimeInNsecs = milliseconds(int_val); 233 ALOGI("Using %u mSec as standby time.", int_val); 234 } else { 235 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 236 ALOGI("Using default %u mSec as standby time.", 237 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 238 } 239 } 240 241 mMode = AUDIO_MODE_NORMAL; 242 mMasterVolumeSW = 1.0; 243 mMasterVolume = 1.0; 244 mHardwareStatus = AUDIO_HW_IDLE; 245} 246 247AudioFlinger::~AudioFlinger() 248{ 249 250 while (!mRecordThreads.isEmpty()) { 251 // closeInput() will remove first entry from mRecordThreads 252 closeInput(mRecordThreads.keyAt(0)); 253 } 254 while (!mPlaybackThreads.isEmpty()) { 255 // closeOutput() will remove first entry from mPlaybackThreads 256 closeOutput(mPlaybackThreads.keyAt(0)); 257 } 258 259 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 260 // no mHardwareLock needed, as there are no other references to this 261 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 262 delete mAudioHwDevs.valueAt(i); 263 } 264} 265 266static const char * const audio_interfaces[] = { 267 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 268 AUDIO_HARDWARE_MODULE_ID_A2DP, 269 AUDIO_HARDWARE_MODULE_ID_USB, 270}; 271#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 272 273audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 274{ 275 // if module is 0, the request comes from an old policy manager and we should load 276 // well known modules 277 if (module == 0) { 278 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 279 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 280 loadHwModule_l(audio_interfaces[i]); 281 } 282 } else { 283 // check a match for the requested module handle 284 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 285 if (audioHwdevice != NULL) { 286 return audioHwdevice->hwDevice(); 287 } 288 } 289 // then try to find a module supporting the requested device. 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 292 if ((dev->get_supported_devices(dev) & devices) == devices) 293 return dev; 294 } 295 296 return NULL; 297} 298 299status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 300{ 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 305 result.append("Clients:\n"); 306 for (size_t i = 0; i < mClients.size(); ++i) { 307 sp<Client> client = mClients.valueAt(i).promote(); 308 if (client != 0) { 309 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 310 result.append(buffer); 311 } 312 } 313 314 result.append("Global session refs:\n"); 315 result.append(" session pid count\n"); 316 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 317 AudioSessionRef *r = mAudioSessionRefs[i]; 318 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 319 result.append(buffer); 320 } 321 write(fd, result.string(), result.size()); 322 return NO_ERROR; 323} 324 325 326status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 327{ 328 const size_t SIZE = 256; 329 char buffer[SIZE]; 330 String8 result; 331 hardware_call_state hardwareStatus = mHardwareStatus; 332 333 snprintf(buffer, SIZE, "Hardware status: %d\n" 334 "Standby Time mSec: %u\n", 335 hardwareStatus, 336 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 337 result.append(buffer); 338 write(fd, result.string(), result.size()); 339 return NO_ERROR; 340} 341 342status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 343{ 344 const size_t SIZE = 256; 345 char buffer[SIZE]; 346 String8 result; 347 snprintf(buffer, SIZE, "Permission Denial: " 348 "can't dump AudioFlinger from pid=%d, uid=%d\n", 349 IPCThreadState::self()->getCallingPid(), 350 IPCThreadState::self()->getCallingUid()); 351 result.append(buffer); 352 write(fd, result.string(), result.size()); 353 return NO_ERROR; 354} 355 356static bool tryLock(Mutex& mutex) 357{ 358 bool locked = false; 359 for (int i = 0; i < kDumpLockRetries; ++i) { 360 if (mutex.tryLock() == NO_ERROR) { 361 locked = true; 362 break; 363 } 364 usleep(kDumpLockSleepUs); 365 } 366 return locked; 367} 368 369status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 370{ 371 if (!dumpAllowed()) { 372 dumpPermissionDenial(fd, args); 373 } else { 374 // get state of hardware lock 375 bool hardwareLocked = tryLock(mHardwareLock); 376 if (!hardwareLocked) { 377 String8 result(kHardwareLockedString); 378 write(fd, result.string(), result.size()); 379 } else { 380 mHardwareLock.unlock(); 381 } 382 383 bool locked = tryLock(mLock); 384 385 // failed to lock - AudioFlinger is probably deadlocked 386 if (!locked) { 387 String8 result(kDeadlockedString); 388 write(fd, result.string(), result.size()); 389 } 390 391 dumpClients(fd, args); 392 dumpInternals(fd, args); 393 394 // dump playback threads 395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 396 mPlaybackThreads.valueAt(i)->dump(fd, args); 397 } 398 399 // dump record threads 400 for (size_t i = 0; i < mRecordThreads.size(); i++) { 401 mRecordThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump all hardware devs 405 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 406 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 407 dev->dump(dev, fd); 408 } 409 if (locked) mLock.unlock(); 410 } 411 return NO_ERROR; 412} 413 414sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 415{ 416 // If pid is already in the mClients wp<> map, then use that entry 417 // (for which promote() is always != 0), otherwise create a new entry and Client. 418 sp<Client> client = mClients.valueFor(pid).promote(); 419 if (client == 0) { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 return client; 425} 426 427// IAudioFlinger interface 428 429 430sp<IAudioTrack> AudioFlinger::createTrack( 431 pid_t pid, 432 audio_stream_type_t streamType, 433 uint32_t sampleRate, 434 audio_format_t format, 435 uint32_t channelMask, 436 int frameCount, 437 IAudioFlinger::track_flags_t flags, 438 const sp<IMemory>& sharedBuffer, 439 audio_io_handle_t output, 440 pid_t tid, 441 int *sessionId, 442 status_t *status) 443{ 444 sp<PlaybackThread::Track> track; 445 sp<TrackHandle> trackHandle; 446 sp<Client> client; 447 status_t lStatus; 448 int lSessionId; 449 450 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 451 // but if someone uses binder directly they could bypass that and cause us to crash 452 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 453 ALOGE("createTrack() invalid stream type %d", streamType); 454 lStatus = BAD_VALUE; 455 goto Exit; 456 } 457 458 { 459 Mutex::Autolock _l(mLock); 460 PlaybackThread *thread = checkPlaybackThread_l(output); 461 PlaybackThread *effectThread = NULL; 462 if (thread == NULL) { 463 ALOGE("unknown output thread"); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 client = registerPid_l(pid); 469 470 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 471 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 472 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 473 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 474 if (mPlaybackThreads.keyAt(i) != output) { 475 // prevent same audio session on different output threads 476 uint32_t sessions = t->hasAudioSession(*sessionId); 477 if (sessions & PlaybackThread::TRACK_SESSION) { 478 ALOGE("createTrack() session ID %d already in use", *sessionId); 479 lStatus = BAD_VALUE; 480 goto Exit; 481 } 482 // check if an effect with same session ID is waiting for a track to be created 483 if (sessions & PlaybackThread::EFFECT_SESSION) { 484 effectThread = t.get(); 485 } 486 } 487 } 488 lSessionId = *sessionId; 489 } else { 490 // if no audio session id is provided, create one here 491 lSessionId = nextUniqueId(); 492 if (sessionId != NULL) { 493 *sessionId = lSessionId; 494 } 495 } 496 ALOGV("createTrack() lSessionId: %d", lSessionId); 497 498 track = thread->createTrack_l(client, streamType, sampleRate, format, 499 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 500 501 // move effect chain to this output thread if an effect on same session was waiting 502 // for a track to be created 503 if (lStatus == NO_ERROR && effectThread != NULL) { 504 Mutex::Autolock _dl(thread->mLock); 505 Mutex::Autolock _sl(effectThread->mLock); 506 moveEffectChain_l(lSessionId, effectThread, thread, true); 507 } 508 509 // Look for sync events awaiting for a session to be used. 510 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 511 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 512 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 513 track->setSyncEvent(mPendingSyncEvents[i]); 514 mPendingSyncEvents.removeAt(i); 515 i--; 516 } 517 } 518 } 519 } 520 if (lStatus == NO_ERROR) { 521 trackHandle = new TrackHandle(track); 522 } else { 523 // remove local strong reference to Client before deleting the Track so that the Client 524 // destructor is called by the TrackBase destructor with mLock held 525 client.clear(); 526 track.clear(); 527 } 528 529Exit: 530 if (status != NULL) { 531 *status = lStatus; 532 } 533 return trackHandle; 534} 535 536uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 537{ 538 Mutex::Autolock _l(mLock); 539 PlaybackThread *thread = checkPlaybackThread_l(output); 540 if (thread == NULL) { 541 ALOGW("sampleRate() unknown thread %d", output); 542 return 0; 543 } 544 return thread->sampleRate(); 545} 546 547int AudioFlinger::channelCount(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("channelCount() unknown thread %d", output); 553 return 0; 554 } 555 return thread->channelCount(); 556} 557 558audio_format_t AudioFlinger::format(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("format() unknown thread %d", output); 564 return AUDIO_FORMAT_INVALID; 565 } 566 return thread->format(); 567} 568 569size_t AudioFlinger::frameCount(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("frameCount() unknown thread %d", output); 575 return 0; 576 } 577 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 578 // should examine all callers and fix them to handle smaller counts 579 return thread->frameCount(); 580} 581 582uint32_t AudioFlinger::latency(audio_io_handle_t output) const 583{ 584 Mutex::Autolock _l(mLock); 585 PlaybackThread *thread = checkPlaybackThread_l(output); 586 if (thread == NULL) { 587 ALOGW("latency() unknown thread %d", output); 588 return 0; 589 } 590 return thread->latency(); 591} 592 593status_t AudioFlinger::setMasterVolume(float value) 594{ 595 status_t ret = initCheck(); 596 if (ret != NO_ERROR) { 597 return ret; 598 } 599 600 // check calling permissions 601 if (!settingsAllowed()) { 602 return PERMISSION_DENIED; 603 } 604 605 float swmv = value; 606 607 Mutex::Autolock _l(mLock); 608 609 // when hw supports master volume, don't scale in sw mixer 610 if (MVS_NONE != mMasterVolumeSupportLvl) { 611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 612 AutoMutex lock(mHardwareLock); 613 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 614 615 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 616 if (NULL != dev->set_master_volume) { 617 dev->set_master_volume(dev, value); 618 } 619 mHardwareStatus = AUDIO_HW_IDLE; 620 } 621 622 swmv = 1.0; 623 } 624 625 mMasterVolume = value; 626 mMasterVolumeSW = swmv; 627 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 628 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 629 630 return NO_ERROR; 631} 632 633status_t AudioFlinger::setMode(audio_mode_t mode) 634{ 635 status_t ret = initCheck(); 636 if (ret != NO_ERROR) { 637 return ret; 638 } 639 640 // check calling permissions 641 if (!settingsAllowed()) { 642 return PERMISSION_DENIED; 643 } 644 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 645 ALOGW("Illegal value: setMode(%d)", mode); 646 return BAD_VALUE; 647 } 648 649 { // scope for the lock 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_HW_SET_MODE; 652 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 653 mHardwareStatus = AUDIO_HW_IDLE; 654 } 655 656 if (NO_ERROR == ret) { 657 Mutex::Autolock _l(mLock); 658 mMode = mode; 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 660 mPlaybackThreads.valueAt(i)->setMode(mode); 661 } 662 663 return ret; 664} 665 666status_t AudioFlinger::setMicMute(bool state) 667{ 668 status_t ret = initCheck(); 669 if (ret != NO_ERROR) { 670 return ret; 671 } 672 673 // check calling permissions 674 if (!settingsAllowed()) { 675 return PERMISSION_DENIED; 676 } 677 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 680 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return ret; 683} 684 685bool AudioFlinger::getMicMute() const 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return false; 690 } 691 692 bool state = AUDIO_MODE_INVALID; 693 AutoMutex lock(mHardwareLock); 694 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 695 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return state; 698} 699 700status_t AudioFlinger::setMasterMute(bool muted) 701{ 702 // check calling permissions 703 if (!settingsAllowed()) { 704 return PERMISSION_DENIED; 705 } 706 707 Mutex::Autolock _l(mLock); 708 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 709 mMasterMute = muted; 710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::masterVolume() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolume_l(); 720} 721 722float AudioFlinger::masterVolumeSW() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterVolumeSW_l(); 726} 727 728bool AudioFlinger::masterMute() const 729{ 730 Mutex::Autolock _l(mLock); 731 return masterMute_l(); 732} 733 734float AudioFlinger::masterVolume_l() const 735{ 736 if (MVS_FULL == mMasterVolumeSupportLvl) { 737 float ret_val; 738 AutoMutex lock(mHardwareLock); 739 740 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 741 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 742 (NULL != mPrimaryHardwareDev->get_master_volume), 743 "can't get master volume"); 744 745 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 746 mHardwareStatus = AUDIO_HW_IDLE; 747 return ret_val; 748 } 749 750 return mMasterVolume; 751} 752 753status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 754 audio_io_handle_t output) 755{ 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 761 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 762 ALOGE("setStreamVolume() invalid stream %d", stream); 763 return BAD_VALUE; 764 } 765 766 AutoMutex lock(mLock); 767 PlaybackThread *thread = NULL; 768 if (output) { 769 thread = checkPlaybackThread_l(output); 770 if (thread == NULL) { 771 return BAD_VALUE; 772 } 773 } 774 775 mStreamTypes[stream].volume = value; 776 777 if (thread == NULL) { 778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 779 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 780 } 781 } else { 782 thread->setStreamVolume(stream, value); 783 } 784 785 return NO_ERROR; 786} 787 788status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 796 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 797 ALOGE("setStreamMute() invalid stream %d", stream); 798 return BAD_VALUE; 799 } 800 801 AutoMutex lock(mLock); 802 mStreamTypes[stream].mute = muted; 803 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 805 806 return NO_ERROR; 807} 808 809float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 810{ 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 return 0.0f; 813 } 814 815 AutoMutex lock(mLock); 816 float volume; 817 if (output) { 818 PlaybackThread *thread = checkPlaybackThread_l(output); 819 if (thread == NULL) { 820 return 0.0f; 821 } 822 volume = thread->streamVolume(stream); 823 } else { 824 volume = streamVolume_l(stream); 825 } 826 827 return volume; 828} 829 830bool AudioFlinger::streamMute(audio_stream_type_t stream) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return true; 834 } 835 836 AutoMutex lock(mLock); 837 return streamMute_l(stream); 838} 839 840status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 841{ 842 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 843 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 844 // check calling permissions 845 if (!settingsAllowed()) { 846 return PERMISSION_DENIED; 847 } 848 849 // ioHandle == 0 means the parameters are global to the audio hardware interface 850 if (ioHandle == 0) { 851 Mutex::Autolock _l(mLock); 852 status_t final_result = NO_ERROR; 853 { 854 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 856 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 857 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 858 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 859 final_result = result ?: final_result; 860 } 861 mHardwareStatus = AUDIO_HW_IDLE; 862 } 863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 864 AudioParameter param = AudioParameter(keyValuePairs); 865 String8 value; 866 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 867 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 868 if (mBtNrecIsOff != btNrecIsOff) { 869 for (size_t i = 0; i < mRecordThreads.size(); i++) { 870 sp<RecordThread> thread = mRecordThreads.valueAt(i); 871 RecordThread::RecordTrack *track = thread->track(); 872 if (track != NULL) { 873 audio_devices_t device = (audio_devices_t)( 874 thread->device() & AUDIO_DEVICE_IN_ALL); 875 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 876 thread->setEffectSuspended(FX_IID_AEC, 877 suspend, 878 track->sessionId()); 879 thread->setEffectSuspended(FX_IID_NS, 880 suspend, 881 track->sessionId()); 882 } 883 } 884 mBtNrecIsOff = btNrecIsOff; 885 } 886 } 887 return final_result; 888 } 889 890 // hold a strong ref on thread in case closeOutput() or closeInput() is called 891 // and the thread is exited once the lock is released 892 sp<ThreadBase> thread; 893 { 894 Mutex::Autolock _l(mLock); 895 thread = checkPlaybackThread_l(ioHandle); 896 if (thread == NULL) { 897 thread = checkRecordThread_l(ioHandle); 898 } else if (thread == primaryPlaybackThread_l()) { 899 // indicate output device change to all input threads for pre processing 900 AudioParameter param = AudioParameter(keyValuePairs); 901 int value; 902 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 903 (value != 0)) { 904 for (size_t i = 0; i < mRecordThreads.size(); i++) { 905 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 906 } 907 } 908 } 909 } 910 if (thread != 0) { 911 return thread->setParameters(keyValuePairs); 912 } 913 return BAD_VALUE; 914} 915 916String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 917{ 918// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 919// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 920 921 Mutex::Autolock _l(mLock); 922 923 if (ioHandle == 0) { 924 String8 out_s8; 925 926 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 927 char *s; 928 { 929 AutoMutex lock(mHardwareLock); 930 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 931 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 932 s = dev->get_parameters(dev, keys.string()); 933 mHardwareStatus = AUDIO_HW_IDLE; 934 } 935 out_s8 += String8(s ? s : ""); 936 free(s); 937 } 938 return out_s8; 939 } 940 941 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 942 if (playbackThread != NULL) { 943 return playbackThread->getParameters(keys); 944 } 945 RecordThread *recordThread = checkRecordThread_l(ioHandle); 946 if (recordThread != NULL) { 947 return recordThread->getParameters(keys); 948 } 949 return String8(""); 950} 951 952size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 953{ 954 status_t ret = initCheck(); 955 if (ret != NO_ERROR) { 956 return 0; 957 } 958 959 AutoMutex lock(mHardwareLock); 960 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 961 struct audio_config config = { 962 sample_rate: sampleRate, 963 channel_mask: audio_channel_in_mask_from_count(channelCount), 964 format: format, 965 }; 966 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 967 mHardwareStatus = AUDIO_HW_IDLE; 968 return size; 969} 970 971unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 972{ 973 if (ioHandle == 0) { 974 return 0; 975 } 976 977 Mutex::Autolock _l(mLock); 978 979 RecordThread *recordThread = checkRecordThread_l(ioHandle); 980 if (recordThread != NULL) { 981 return recordThread->getInputFramesLost(); 982 } 983 return 0; 984} 985 986status_t AudioFlinger::setVoiceVolume(float value) 987{ 988 status_t ret = initCheck(); 989 if (ret != NO_ERROR) { 990 return ret; 991 } 992 993 // check calling permissions 994 if (!settingsAllowed()) { 995 return PERMISSION_DENIED; 996 } 997 998 AutoMutex lock(mHardwareLock); 999 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1000 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 1003 return ret; 1004} 1005 1006status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1007 audio_io_handle_t output) const 1008{ 1009 status_t status; 1010 1011 Mutex::Autolock _l(mLock); 1012 1013 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1014 if (playbackThread != NULL) { 1015 return playbackThread->getRenderPosition(halFrames, dspFrames); 1016 } 1017 1018 return BAD_VALUE; 1019} 1020 1021void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1022{ 1023 1024 Mutex::Autolock _l(mLock); 1025 1026 pid_t pid = IPCThreadState::self()->getCallingPid(); 1027 if (mNotificationClients.indexOfKey(pid) < 0) { 1028 sp<NotificationClient> notificationClient = new NotificationClient(this, 1029 client, 1030 pid); 1031 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1032 1033 mNotificationClients.add(pid, notificationClient); 1034 1035 sp<IBinder> binder = client->asBinder(); 1036 binder->linkToDeath(notificationClient); 1037 1038 // the config change is always sent from playback or record threads to avoid deadlock 1039 // with AudioSystem::gLock 1040 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1041 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1042 } 1043 1044 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1045 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1046 } 1047 } 1048} 1049 1050void AudioFlinger::removeNotificationClient(pid_t pid) 1051{ 1052 Mutex::Autolock _l(mLock); 1053 1054 mNotificationClients.removeItem(pid); 1055 1056 ALOGV("%d died, releasing its sessions", pid); 1057 size_t num = mAudioSessionRefs.size(); 1058 bool removed = false; 1059 for (size_t i = 0; i< num; ) { 1060 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1061 ALOGV(" pid %d @ %d", ref->mPid, i); 1062 if (ref->mPid == pid) { 1063 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1064 mAudioSessionRefs.removeAt(i); 1065 delete ref; 1066 removed = true; 1067 num--; 1068 } else { 1069 i++; 1070 } 1071 } 1072 if (removed) { 1073 purgeStaleEffects_l(); 1074 } 1075} 1076 1077// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1078void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1079{ 1080 size_t size = mNotificationClients.size(); 1081 for (size_t i = 0; i < size; i++) { 1082 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1083 param2); 1084 } 1085} 1086 1087// removeClient_l() must be called with AudioFlinger::mLock held 1088void AudioFlinger::removeClient_l(pid_t pid) 1089{ 1090 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1091 mClients.removeItem(pid); 1092} 1093 1094 1095// ---------------------------------------------------------------------------- 1096 1097AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1098 uint32_t device, type_t type) 1099 : Thread(false), 1100 mType(type), 1101 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1102 // mChannelMask 1103 mChannelCount(0), 1104 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1105 mParamStatus(NO_ERROR), 1106 mStandby(false), mId(id), 1107 mDevice(device), 1108 mDeathRecipient(new PMDeathRecipient(this)) 1109{ 1110} 1111 1112AudioFlinger::ThreadBase::~ThreadBase() 1113{ 1114 mParamCond.broadcast(); 1115 // do not lock the mutex in destructor 1116 releaseWakeLock_l(); 1117 if (mPowerManager != 0) { 1118 sp<IBinder> binder = mPowerManager->asBinder(); 1119 binder->unlinkToDeath(mDeathRecipient); 1120 } 1121} 1122 1123void AudioFlinger::ThreadBase::exit() 1124{ 1125 ALOGV("ThreadBase::exit"); 1126 { 1127 // This lock prevents the following race in thread (uniprocessor for illustration): 1128 // if (!exitPending()) { 1129 // // context switch from here to exit() 1130 // // exit() calls requestExit(), what exitPending() observes 1131 // // exit() calls signal(), which is dropped since no waiters 1132 // // context switch back from exit() to here 1133 // mWaitWorkCV.wait(...); 1134 // // now thread is hung 1135 // } 1136 AutoMutex lock(mLock); 1137 requestExit(); 1138 mWaitWorkCV.signal(); 1139 } 1140 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1141 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1142 requestExitAndWait(); 1143} 1144 1145status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1146{ 1147 status_t status; 1148 1149 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1150 Mutex::Autolock _l(mLock); 1151 1152 mNewParameters.add(keyValuePairs); 1153 mWaitWorkCV.signal(); 1154 // wait condition with timeout in case the thread loop has exited 1155 // before the request could be processed 1156 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1157 status = mParamStatus; 1158 mWaitWorkCV.signal(); 1159 } else { 1160 status = TIMED_OUT; 1161 } 1162 return status; 1163} 1164 1165void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1166{ 1167 Mutex::Autolock _l(mLock); 1168 sendConfigEvent_l(event, param); 1169} 1170 1171// sendConfigEvent_l() must be called with ThreadBase::mLock held 1172void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1173{ 1174 ConfigEvent configEvent; 1175 configEvent.mEvent = event; 1176 configEvent.mParam = param; 1177 mConfigEvents.add(configEvent); 1178 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1179 mWaitWorkCV.signal(); 1180} 1181 1182void AudioFlinger::ThreadBase::processConfigEvents() 1183{ 1184 mLock.lock(); 1185 while (!mConfigEvents.isEmpty()) { 1186 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1187 ConfigEvent configEvent = mConfigEvents[0]; 1188 mConfigEvents.removeAt(0); 1189 // release mLock before locking AudioFlinger mLock: lock order is always 1190 // AudioFlinger then ThreadBase to avoid cross deadlock 1191 mLock.unlock(); 1192 mAudioFlinger->mLock.lock(); 1193 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1194 mAudioFlinger->mLock.unlock(); 1195 mLock.lock(); 1196 } 1197 mLock.unlock(); 1198} 1199 1200status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1201{ 1202 const size_t SIZE = 256; 1203 char buffer[SIZE]; 1204 String8 result; 1205 1206 bool locked = tryLock(mLock); 1207 if (!locked) { 1208 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1209 write(fd, buffer, strlen(buffer)); 1210 } 1211 1212 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1217 result.append(buffer); 1218 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1219 result.append(buffer); 1220 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1221 result.append(buffer); 1222 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1223 result.append(buffer); 1224 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1225 result.append(buffer); 1226 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1227 result.append(buffer); 1228 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1229 result.append(buffer); 1230 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1231 result.append(buffer); 1232 1233 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1234 result.append(buffer); 1235 result.append(" Index Command"); 1236 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1237 snprintf(buffer, SIZE, "\n %02d ", i); 1238 result.append(buffer); 1239 result.append(mNewParameters[i]); 1240 } 1241 1242 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, " Index event param\n"); 1245 result.append(buffer); 1246 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1247 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1248 result.append(buffer); 1249 } 1250 result.append("\n"); 1251 1252 write(fd, result.string(), result.size()); 1253 1254 if (locked) { 1255 mLock.unlock(); 1256 } 1257 return NO_ERROR; 1258} 1259 1260status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1261{ 1262 const size_t SIZE = 256; 1263 char buffer[SIZE]; 1264 String8 result; 1265 1266 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1267 write(fd, buffer, strlen(buffer)); 1268 1269 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1270 sp<EffectChain> chain = mEffectChains[i]; 1271 if (chain != 0) { 1272 chain->dump(fd, args); 1273 } 1274 } 1275 return NO_ERROR; 1276} 1277 1278void AudioFlinger::ThreadBase::acquireWakeLock() 1279{ 1280 Mutex::Autolock _l(mLock); 1281 acquireWakeLock_l(); 1282} 1283 1284void AudioFlinger::ThreadBase::acquireWakeLock_l() 1285{ 1286 if (mPowerManager == 0) { 1287 // use checkService() to avoid blocking if power service is not up yet 1288 sp<IBinder> binder = 1289 defaultServiceManager()->checkService(String16("power")); 1290 if (binder == 0) { 1291 ALOGW("Thread %s cannot connect to the power manager service", mName); 1292 } else { 1293 mPowerManager = interface_cast<IPowerManager>(binder); 1294 binder->linkToDeath(mDeathRecipient); 1295 } 1296 } 1297 if (mPowerManager != 0) { 1298 sp<IBinder> binder = new BBinder(); 1299 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1300 binder, 1301 String16(mName)); 1302 if (status == NO_ERROR) { 1303 mWakeLockToken = binder; 1304 } 1305 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1306 } 1307} 1308 1309void AudioFlinger::ThreadBase::releaseWakeLock() 1310{ 1311 Mutex::Autolock _l(mLock); 1312 releaseWakeLock_l(); 1313} 1314 1315void AudioFlinger::ThreadBase::releaseWakeLock_l() 1316{ 1317 if (mWakeLockToken != 0) { 1318 ALOGV("releaseWakeLock_l() %s", mName); 1319 if (mPowerManager != 0) { 1320 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1321 } 1322 mWakeLockToken.clear(); 1323 } 1324} 1325 1326void AudioFlinger::ThreadBase::clearPowerManager() 1327{ 1328 Mutex::Autolock _l(mLock); 1329 releaseWakeLock_l(); 1330 mPowerManager.clear(); 1331} 1332 1333void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1334{ 1335 sp<ThreadBase> thread = mThread.promote(); 1336 if (thread != 0) { 1337 thread->clearPowerManager(); 1338 } 1339 ALOGW("power manager service died !!!"); 1340} 1341 1342void AudioFlinger::ThreadBase::setEffectSuspended( 1343 const effect_uuid_t *type, bool suspend, int sessionId) 1344{ 1345 Mutex::Autolock _l(mLock); 1346 setEffectSuspended_l(type, suspend, sessionId); 1347} 1348 1349void AudioFlinger::ThreadBase::setEffectSuspended_l( 1350 const effect_uuid_t *type, bool suspend, int sessionId) 1351{ 1352 sp<EffectChain> chain = getEffectChain_l(sessionId); 1353 if (chain != 0) { 1354 if (type != NULL) { 1355 chain->setEffectSuspended_l(type, suspend); 1356 } else { 1357 chain->setEffectSuspendedAll_l(suspend); 1358 } 1359 } 1360 1361 updateSuspendedSessions_l(type, suspend, sessionId); 1362} 1363 1364void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1365{ 1366 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1367 if (index < 0) { 1368 return; 1369 } 1370 1371 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1372 mSuspendedSessions.editValueAt(index); 1373 1374 for (size_t i = 0; i < sessionEffects.size(); i++) { 1375 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1376 for (int j = 0; j < desc->mRefCount; j++) { 1377 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1378 chain->setEffectSuspendedAll_l(true); 1379 } else { 1380 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1381 desc->mType.timeLow); 1382 chain->setEffectSuspended_l(&desc->mType, true); 1383 } 1384 } 1385 } 1386} 1387 1388void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1389 bool suspend, 1390 int sessionId) 1391{ 1392 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1393 1394 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1395 1396 if (suspend) { 1397 if (index >= 0) { 1398 sessionEffects = mSuspendedSessions.editValueAt(index); 1399 } else { 1400 mSuspendedSessions.add(sessionId, sessionEffects); 1401 } 1402 } else { 1403 if (index < 0) { 1404 return; 1405 } 1406 sessionEffects = mSuspendedSessions.editValueAt(index); 1407 } 1408 1409 1410 int key = EffectChain::kKeyForSuspendAll; 1411 if (type != NULL) { 1412 key = type->timeLow; 1413 } 1414 index = sessionEffects.indexOfKey(key); 1415 1416 sp<SuspendedSessionDesc> desc; 1417 if (suspend) { 1418 if (index >= 0) { 1419 desc = sessionEffects.valueAt(index); 1420 } else { 1421 desc = new SuspendedSessionDesc(); 1422 if (type != NULL) { 1423 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1424 } 1425 sessionEffects.add(key, desc); 1426 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1427 } 1428 desc->mRefCount++; 1429 } else { 1430 if (index < 0) { 1431 return; 1432 } 1433 desc = sessionEffects.valueAt(index); 1434 if (--desc->mRefCount == 0) { 1435 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1436 sessionEffects.removeItemsAt(index); 1437 if (sessionEffects.isEmpty()) { 1438 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1439 sessionId); 1440 mSuspendedSessions.removeItem(sessionId); 1441 } 1442 } 1443 } 1444 if (!sessionEffects.isEmpty()) { 1445 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1446 } 1447} 1448 1449void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1450 bool enabled, 1451 int sessionId) 1452{ 1453 Mutex::Autolock _l(mLock); 1454 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1455} 1456 1457void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1458 bool enabled, 1459 int sessionId) 1460{ 1461 if (mType != RECORD) { 1462 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1463 // another session. This gives the priority to well behaved effect control panels 1464 // and applications not using global effects. 1465 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1466 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1467 } 1468 } 1469 1470 sp<EffectChain> chain = getEffectChain_l(sessionId); 1471 if (chain != 0) { 1472 chain->checkSuspendOnEffectEnabled(effect, enabled); 1473 } 1474} 1475 1476// ---------------------------------------------------------------------------- 1477 1478AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1479 AudioStreamOut* output, 1480 audio_io_handle_t id, 1481 uint32_t device, 1482 type_t type) 1483 : ThreadBase(audioFlinger, id, device, type), 1484 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1485 // Assumes constructor is called by AudioFlinger with it's mLock held, 1486 // but it would be safer to explicitly pass initial masterMute as parameter 1487 mMasterMute(audioFlinger->masterMute_l()), 1488 // mStreamTypes[] initialized in constructor body 1489 mOutput(output), 1490 // Assumes constructor is called by AudioFlinger with it's mLock held, 1491 // but it would be safer to explicitly pass initial masterVolume as parameter 1492 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1493 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1494 mMixerStatus(MIXER_IDLE), 1495 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1496 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1497 // index 0 is reserved for normal mixer's submix 1498 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1499{ 1500 snprintf(mName, kNameLength, "AudioOut_%X", id); 1501 1502 readOutputParameters(); 1503 1504 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1505 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1506 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1507 stream = (audio_stream_type_t) (stream + 1)) { 1508 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1509 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1510 } 1511 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1512 // because mAudioFlinger doesn't have one to copy from 1513} 1514 1515AudioFlinger::PlaybackThread::~PlaybackThread() 1516{ 1517 delete [] mMixBuffer; 1518} 1519 1520status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1521{ 1522 dumpInternals(fd, args); 1523 dumpTracks(fd, args); 1524 dumpEffectChains(fd, args); 1525 return NO_ERROR; 1526} 1527 1528status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1529{ 1530 const size_t SIZE = 256; 1531 char buffer[SIZE]; 1532 String8 result; 1533 1534 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1535 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1536 const stream_type_t *st = &mStreamTypes[i]; 1537 if (i > 0) { 1538 result.appendFormat(", "); 1539 } 1540 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1541 if (st->mute) { 1542 result.append("M"); 1543 } 1544 } 1545 result.append("\n"); 1546 write(fd, result.string(), result.length()); 1547 result.clear(); 1548 1549 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1550 result.append(buffer); 1551 Track::appendDumpHeader(result); 1552 for (size_t i = 0; i < mTracks.size(); ++i) { 1553 sp<Track> track = mTracks[i]; 1554 if (track != 0) { 1555 track->dump(buffer, SIZE); 1556 result.append(buffer); 1557 } 1558 } 1559 1560 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1561 result.append(buffer); 1562 Track::appendDumpHeader(result); 1563 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1564 sp<Track> track = mActiveTracks[i].promote(); 1565 if (track != 0) { 1566 track->dump(buffer, SIZE); 1567 result.append(buffer); 1568 } 1569 } 1570 write(fd, result.string(), result.size()); 1571 return NO_ERROR; 1572} 1573 1574status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1575{ 1576 const size_t SIZE = 256; 1577 char buffer[SIZE]; 1578 String8 result; 1579 1580 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1581 result.append(buffer); 1582 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1583 result.append(buffer); 1584 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1585 result.append(buffer); 1586 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1587 result.append(buffer); 1588 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1589 result.append(buffer); 1590 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1591 result.append(buffer); 1592 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1593 result.append(buffer); 1594 write(fd, result.string(), result.size()); 1595 1596 dumpBase(fd, args); 1597 1598 return NO_ERROR; 1599} 1600 1601// Thread virtuals 1602status_t AudioFlinger::PlaybackThread::readyToRun() 1603{ 1604 status_t status = initCheck(); 1605 if (status == NO_ERROR) { 1606 ALOGI("AudioFlinger's thread %p ready to run", this); 1607 } else { 1608 ALOGE("No working audio driver found."); 1609 } 1610 return status; 1611} 1612 1613void AudioFlinger::PlaybackThread::onFirstRef() 1614{ 1615 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1616} 1617 1618// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1619sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1620 const sp<AudioFlinger::Client>& client, 1621 audio_stream_type_t streamType, 1622 uint32_t sampleRate, 1623 audio_format_t format, 1624 uint32_t channelMask, 1625 int frameCount, 1626 const sp<IMemory>& sharedBuffer, 1627 int sessionId, 1628 IAudioFlinger::track_flags_t flags, 1629 pid_t tid, 1630 status_t *status) 1631{ 1632 sp<Track> track; 1633 status_t lStatus; 1634 1635 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1636 1637 // client expresses a preference for FAST, but we get the final say 1638 if (flags & IAudioFlinger::TRACK_FAST) { 1639 if ( 1640 // not timed 1641 (!isTimed) && 1642 // either of these use cases: 1643 ( 1644 // use case 1: shared buffer with any frame count 1645 ( 1646 (sharedBuffer != 0) 1647 ) || 1648 // use case 2: callback handler and frame count is default or at least as large as HAL 1649 ( 1650 (tid != -1) && 1651 ((frameCount == 0) || 1652 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1653 ) 1654 ) && 1655 // PCM data 1656 audio_is_linear_pcm(format) && 1657 // mono or stereo 1658 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1659 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1660#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1661 // hardware sample rate 1662 (sampleRate == mSampleRate) && 1663#endif 1664 // normal mixer has an associated fast mixer 1665 hasFastMixer() && 1666 // there are sufficient fast track slots available 1667 (mFastTrackAvailMask != 0) 1668 // FIXME test that MixerThread for this fast track has a capable output HAL 1669 // FIXME add a permission test also? 1670 ) { 1671 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1672 if (frameCount == 0) { 1673 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1674 } 1675 ALOGI("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1676 frameCount, mFrameCount); 1677 } else { 1678 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1679 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1680 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1681 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1682 audio_is_linear_pcm(format), 1683 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1684 flags &= ~IAudioFlinger::TRACK_FAST; 1685 // For compatibility with AudioTrack calculation, buffer depth is forced 1686 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1687 // This is probably too conservative, but legacy application code may depend on it. 1688 // If you change this calculation, also review the start threshold which is related. 1689 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1690 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1691 if (minBufCount < 2) { 1692 minBufCount = 2; 1693 } 1694 int minFrameCount = mNormalFrameCount * minBufCount; 1695 if (frameCount < minFrameCount) { 1696 frameCount = minFrameCount; 1697 } 1698 } 1699 } 1700 1701 if (mType == DIRECT) { 1702 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1703 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1704 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1705 "for output %p with format %d", 1706 sampleRate, format, channelMask, mOutput, mFormat); 1707 lStatus = BAD_VALUE; 1708 goto Exit; 1709 } 1710 } 1711 } else { 1712 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1713 if (sampleRate > mSampleRate*2) { 1714 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1715 lStatus = BAD_VALUE; 1716 goto Exit; 1717 } 1718 } 1719 1720 lStatus = initCheck(); 1721 if (lStatus != NO_ERROR) { 1722 ALOGE("Audio driver not initialized."); 1723 goto Exit; 1724 } 1725 1726 { // scope for mLock 1727 Mutex::Autolock _l(mLock); 1728 1729 // all tracks in same audio session must share the same routing strategy otherwise 1730 // conflicts will happen when tracks are moved from one output to another by audio policy 1731 // manager 1732 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1733 for (size_t i = 0; i < mTracks.size(); ++i) { 1734 sp<Track> t = mTracks[i]; 1735 if (t != 0 && !t->isOutputTrack()) { 1736 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1737 if (sessionId == t->sessionId() && strategy != actual) { 1738 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1739 strategy, actual); 1740 lStatus = BAD_VALUE; 1741 goto Exit; 1742 } 1743 } 1744 } 1745 1746 if (!isTimed) { 1747 track = new Track(this, client, streamType, sampleRate, format, 1748 channelMask, frameCount, sharedBuffer, sessionId, flags); 1749 } else { 1750 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1751 channelMask, frameCount, sharedBuffer, sessionId); 1752 } 1753 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1754 lStatus = NO_MEMORY; 1755 goto Exit; 1756 } 1757 mTracks.add(track); 1758 1759 sp<EffectChain> chain = getEffectChain_l(sessionId); 1760 if (chain != 0) { 1761 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1762 track->setMainBuffer(chain->inBuffer()); 1763 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1764 chain->incTrackCnt(); 1765 } 1766 } 1767 1768#ifdef HAVE_REQUEST_PRIORITY 1769 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1770 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1771 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1772 // so ask activity manager to do this on our behalf 1773 int err = requestPriority(callingPid, tid, 1); 1774 if (err != 0) { 1775 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1776 1, callingPid, tid, err); 1777 } 1778 } 1779#endif 1780 1781 lStatus = NO_ERROR; 1782 1783Exit: 1784 if (status) { 1785 *status = lStatus; 1786 } 1787 return track; 1788} 1789 1790uint32_t AudioFlinger::PlaybackThread::latency() const 1791{ 1792 Mutex::Autolock _l(mLock); 1793 if (initCheck() == NO_ERROR) { 1794 return mOutput->stream->get_latency(mOutput->stream); 1795 } else { 1796 return 0; 1797 } 1798} 1799 1800void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1801{ 1802 Mutex::Autolock _l(mLock); 1803 mMasterVolume = value; 1804} 1805 1806void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1807{ 1808 Mutex::Autolock _l(mLock); 1809 setMasterMute_l(muted); 1810} 1811 1812void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1813{ 1814 Mutex::Autolock _l(mLock); 1815 mStreamTypes[stream].volume = value; 1816} 1817 1818void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1819{ 1820 Mutex::Autolock _l(mLock); 1821 mStreamTypes[stream].mute = muted; 1822} 1823 1824float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1825{ 1826 Mutex::Autolock _l(mLock); 1827 return mStreamTypes[stream].volume; 1828} 1829 1830// addTrack_l() must be called with ThreadBase::mLock held 1831status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1832{ 1833 status_t status = ALREADY_EXISTS; 1834 1835 // set retry count for buffer fill 1836 track->mRetryCount = kMaxTrackStartupRetries; 1837 if (mActiveTracks.indexOf(track) < 0) { 1838 // the track is newly added, make sure it fills up all its 1839 // buffers before playing. This is to ensure the client will 1840 // effectively get the latency it requested. 1841 track->mFillingUpStatus = Track::FS_FILLING; 1842 track->mResetDone = false; 1843 mActiveTracks.add(track); 1844 if (track->mainBuffer() != mMixBuffer) { 1845 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1846 if (chain != 0) { 1847 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1848 chain->incActiveTrackCnt(); 1849 } 1850 } 1851 1852 status = NO_ERROR; 1853 } 1854 1855 ALOGV("mWaitWorkCV.broadcast"); 1856 mWaitWorkCV.broadcast(); 1857 1858 return status; 1859} 1860 1861// destroyTrack_l() must be called with ThreadBase::mLock held 1862void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1863{ 1864 track->mState = TrackBase::TERMINATED; 1865 // active tracks are removed by threadLoop() 1866 if (mActiveTracks.indexOf(track) < 0) { 1867 removeTrack_l(track); 1868 } 1869} 1870 1871void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1872{ 1873 mTracks.remove(track); 1874 deleteTrackName_l(track->name()); 1875 // redundant as track is about to be destroyed, for dumpsys only 1876 track->mName = -1; 1877 if (track->isFastTrack()) { 1878 int index = track->mFastIndex; 1879 ALOG_ASSERT(0 < index && index < FastMixerState::kMaxFastTracks); 1880 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1881 mFastTrackAvailMask |= 1 << index; 1882 // redundant as track is about to be destroyed, for dumpsys only 1883 track->mFastIndex = -1; 1884 } 1885 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1886 if (chain != 0) { 1887 chain->decTrackCnt(); 1888 } 1889} 1890 1891String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1892{ 1893 String8 out_s8 = String8(""); 1894 char *s; 1895 1896 Mutex::Autolock _l(mLock); 1897 if (initCheck() != NO_ERROR) { 1898 return out_s8; 1899 } 1900 1901 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1902 out_s8 = String8(s); 1903 free(s); 1904 return out_s8; 1905} 1906 1907// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1908void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1909 AudioSystem::OutputDescriptor desc; 1910 void *param2 = NULL; 1911 1912 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1913 1914 switch (event) { 1915 case AudioSystem::OUTPUT_OPENED: 1916 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1917 desc.channels = mChannelMask; 1918 desc.samplingRate = mSampleRate; 1919 desc.format = mFormat; 1920 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1921 desc.latency = latency(); 1922 param2 = &desc; 1923 break; 1924 1925 case AudioSystem::STREAM_CONFIG_CHANGED: 1926 param2 = ¶m; 1927 case AudioSystem::OUTPUT_CLOSED: 1928 default: 1929 break; 1930 } 1931 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1932} 1933 1934void AudioFlinger::PlaybackThread::readOutputParameters() 1935{ 1936 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1937 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1938 mChannelCount = (uint16_t)popcount(mChannelMask); 1939 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1940 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1941 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1942 if (mFrameCount & 15) { 1943 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1944 mFrameCount); 1945 } 1946 1947 // Calculate size of normal mix buffer relative to the HAL output buffer size 1948 uint32_t multiple = 1; 1949 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1950 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1951 multiple = (minNormalFrameCount + mFrameCount - 1) / mFrameCount; 1952 // force multiple to be even, for compatibility with doubling of fast tracks due to HAL SRC 1953 // (it would be unusual for the normal mix buffer size to not be a multiple of fast track) 1954 // FIXME this rounding up should not be done if no HAL SRC 1955 if ((multiple > 2) && (multiple & 1)) { 1956 ++multiple; 1957 } 1958 } 1959 mNormalFrameCount = multiple * mFrameCount; 1960 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 1961 1962 // FIXME - Current mixer implementation only supports stereo output: Always 1963 // Allocate a stereo buffer even if HW output is mono. 1964 delete[] mMixBuffer; 1965 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 1966 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 1967 1968 // force reconfiguration of effect chains and engines to take new buffer size and audio 1969 // parameters into account 1970 // Note that mLock is not held when readOutputParameters() is called from the constructor 1971 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1972 // matter. 1973 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1974 Vector< sp<EffectChain> > effectChains = mEffectChains; 1975 for (size_t i = 0; i < effectChains.size(); i ++) { 1976 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1977 } 1978} 1979 1980status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1981{ 1982 if (halFrames == NULL || dspFrames == NULL) { 1983 return BAD_VALUE; 1984 } 1985 Mutex::Autolock _l(mLock); 1986 if (initCheck() != NO_ERROR) { 1987 return INVALID_OPERATION; 1988 } 1989 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1990 1991 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1992} 1993 1994uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1995{ 1996 Mutex::Autolock _l(mLock); 1997 uint32_t result = 0; 1998 if (getEffectChain_l(sessionId) != 0) { 1999 result = EFFECT_SESSION; 2000 } 2001 2002 for (size_t i = 0; i < mTracks.size(); ++i) { 2003 sp<Track> track = mTracks[i]; 2004 if (sessionId == track->sessionId() && 2005 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2006 result |= TRACK_SESSION; 2007 break; 2008 } 2009 } 2010 2011 return result; 2012} 2013 2014uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2015{ 2016 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2017 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2018 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2019 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2020 } 2021 for (size_t i = 0; i < mTracks.size(); i++) { 2022 sp<Track> track = mTracks[i]; 2023 if (sessionId == track->sessionId() && 2024 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2025 return AudioSystem::getStrategyForStream(track->streamType()); 2026 } 2027 } 2028 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2029} 2030 2031 2032AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2033{ 2034 Mutex::Autolock _l(mLock); 2035 return mOutput; 2036} 2037 2038AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2039{ 2040 Mutex::Autolock _l(mLock); 2041 AudioStreamOut *output = mOutput; 2042 mOutput = NULL; 2043 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2044 // must push a NULL and wait for ack 2045 mOutputSink.clear(); 2046 mPipeSink.clear(); 2047 mNormalSink.clear(); 2048 return output; 2049} 2050 2051// this method must always be called either with ThreadBase mLock held or inside the thread loop 2052audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2053{ 2054 if (mOutput == NULL) { 2055 return NULL; 2056 } 2057 return &mOutput->stream->common; 2058} 2059 2060uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2061{ 2062 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2063 // decoding and transfer time. So sleeping for half of the latency would likely cause 2064 // underruns 2065 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2066 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2067 } else { 2068 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2069 } 2070} 2071 2072status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2073{ 2074 if (!isValidSyncEvent(event)) { 2075 return BAD_VALUE; 2076 } 2077 2078 Mutex::Autolock _l(mLock); 2079 2080 for (size_t i = 0; i < mTracks.size(); ++i) { 2081 sp<Track> track = mTracks[i]; 2082 if (event->triggerSession() == track->sessionId()) { 2083 track->setSyncEvent(event); 2084 return NO_ERROR; 2085 } 2086 } 2087 2088 return NAME_NOT_FOUND; 2089} 2090 2091bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2092{ 2093 switch (event->type()) { 2094 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2095 return true; 2096 default: 2097 break; 2098 } 2099 return false; 2100} 2101 2102// ---------------------------------------------------------------------------- 2103 2104AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2105 audio_io_handle_t id, uint32_t device, type_t type) 2106 : PlaybackThread(audioFlinger, output, id, device, type), 2107 // mAudioMixer below 2108#ifdef SOAKER 2109 mSoaker(NULL), 2110#endif 2111 // mFastMixer below 2112 mFastMixerFutex(0) 2113 // mOutputSink below 2114 // mPipeSink below 2115 // mNormalSink below 2116{ 2117 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2118 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2119 "mFrameCount=%d, mNormalFrameCount=%d", 2120 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2121 mNormalFrameCount); 2122 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2123 2124 // FIXME - Current mixer implementation only supports stereo output 2125 if (mChannelCount == 1) { 2126 ALOGE("Invalid audio hardware channel count"); 2127 } 2128 2129 // create an NBAIO sink for the HAL output stream, and negotiate 2130 mOutputSink = new AudioStreamOutSink(output->stream); 2131 size_t numCounterOffers = 0; 2132 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2133 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2134 ALOG_ASSERT(index == 0); 2135 2136 // initialize fast mixer depending on configuration 2137 bool initFastMixer; 2138 switch (kUseFastMixer) { 2139 case FastMixer_Never: 2140 initFastMixer = false; 2141 break; 2142 case FastMixer_Always: 2143 initFastMixer = true; 2144 break; 2145 case FastMixer_Static: 2146 case FastMixer_Dynamic: 2147 initFastMixer = mFrameCount < mNormalFrameCount; 2148 break; 2149 } 2150 if (initFastMixer) { 2151 2152 // create a MonoPipe to connect our submix to FastMixer 2153 NBAIO_Format format = mOutputSink->format(); 2154 // frame count will be rounded up to a power of 2, so this formula should work well 2155 MonoPipe *monoPipe = new MonoPipe((mNormalFrameCount * 3) / 2, format, 2156 true /*writeCanBlock*/); 2157 const NBAIO_Format offers[1] = {format}; 2158 size_t numCounterOffers = 0; 2159 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2160 ALOG_ASSERT(index == 0); 2161 mPipeSink = monoPipe; 2162 2163#ifdef SOAKER 2164 // create a soaker as workaround for governor issues 2165 mSoaker = new Soaker(); 2166 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2167 mSoaker->run("Soaker", PRIORITY_LOWEST); 2168#endif 2169 2170 // create fast mixer and configure it initially with just one fast track for our submix 2171 mFastMixer = new FastMixer(); 2172 FastMixerStateQueue *sq = mFastMixer->sq(); 2173 FastMixerState *state = sq->begin(); 2174 FastTrack *fastTrack = &state->mFastTracks[0]; 2175 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2176 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2177 fastTrack->mVolumeProvider = NULL; 2178 fastTrack->mGeneration++; 2179 state->mFastTracksGen++; 2180 state->mTrackMask = 1; 2181 // fast mixer will use the HAL output sink 2182 state->mOutputSink = mOutputSink.get(); 2183 state->mOutputSinkGen++; 2184 state->mFrameCount = mFrameCount; 2185 state->mCommand = FastMixerState::COLD_IDLE; 2186 // already done in constructor initialization list 2187 //mFastMixerFutex = 0; 2188 state->mColdFutexAddr = &mFastMixerFutex; 2189 state->mColdGen++; 2190 state->mDumpState = &mFastMixerDumpState; 2191 sq->end(); 2192 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2193 2194 // start the fast mixer 2195 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2196#ifdef HAVE_REQUEST_PRIORITY 2197 pid_t tid = mFastMixer->getTid(); 2198 int err = requestPriority(getpid_cached, tid, 2); 2199 if (err != 0) { 2200 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2201 2, getpid_cached, tid, err); 2202 } 2203#endif 2204 2205 } else { 2206 mFastMixer = NULL; 2207 } 2208 2209 switch (kUseFastMixer) { 2210 case FastMixer_Never: 2211 case FastMixer_Dynamic: 2212 mNormalSink = mOutputSink; 2213 break; 2214 case FastMixer_Always: 2215 mNormalSink = mPipeSink; 2216 break; 2217 case FastMixer_Static: 2218 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2219 break; 2220 } 2221} 2222 2223AudioFlinger::MixerThread::~MixerThread() 2224{ 2225 if (mFastMixer != NULL) { 2226 FastMixerStateQueue *sq = mFastMixer->sq(); 2227 FastMixerState *state = sq->begin(); 2228 if (state->mCommand == FastMixerState::COLD_IDLE) { 2229 int32_t old = android_atomic_inc(&mFastMixerFutex); 2230 if (old == -1) { 2231 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2232 } 2233 } 2234 state->mCommand = FastMixerState::EXIT; 2235 sq->end(); 2236 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2237 mFastMixer->join(); 2238 // Though the fast mixer thread has exited, it's state queue is still valid. 2239 // We'll use that extract the final state which contains one remaining fast track 2240 // corresponding to our sub-mix. 2241 state = sq->begin(); 2242 ALOG_ASSERT(state->mTrackMask == 1); 2243 FastTrack *fastTrack = &state->mFastTracks[0]; 2244 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2245 delete fastTrack->mBufferProvider; 2246 sq->end(false /*didModify*/); 2247 delete mFastMixer; 2248#ifdef SOAKER 2249 if (mSoaker != NULL) { 2250 mSoaker->requestExitAndWait(); 2251 } 2252 delete mSoaker; 2253#endif 2254 } 2255 delete mAudioMixer; 2256} 2257 2258class CpuStats { 2259public: 2260 CpuStats(); 2261 void sample(const String8 &title); 2262#ifdef DEBUG_CPU_USAGE 2263private: 2264 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2265 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2266 2267 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2268 2269 int mCpuNum; // thread's current CPU number 2270 int mCpukHz; // frequency of thread's current CPU in kHz 2271#endif 2272}; 2273 2274CpuStats::CpuStats() 2275#ifdef DEBUG_CPU_USAGE 2276 : mCpuNum(-1), mCpukHz(-1) 2277#endif 2278{ 2279} 2280 2281void CpuStats::sample(const String8 &title) { 2282#ifdef DEBUG_CPU_USAGE 2283 // get current thread's delta CPU time in wall clock ns 2284 double wcNs; 2285 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2286 2287 // record sample for wall clock statistics 2288 if (valid) { 2289 mWcStats.sample(wcNs); 2290 } 2291 2292 // get the current CPU number 2293 int cpuNum = sched_getcpu(); 2294 2295 // get the current CPU frequency in kHz 2296 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2297 2298 // check if either CPU number or frequency changed 2299 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2300 mCpuNum = cpuNum; 2301 mCpukHz = cpukHz; 2302 // ignore sample for purposes of cycles 2303 valid = false; 2304 } 2305 2306 // if no change in CPU number or frequency, then record sample for cycle statistics 2307 if (valid && mCpukHz > 0) { 2308 double cycles = wcNs * cpukHz * 0.000001; 2309 mHzStats.sample(cycles); 2310 } 2311 2312 unsigned n = mWcStats.n(); 2313 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2314 if ((n & 127) == 1) { 2315 long long elapsed = mCpuUsage.elapsed(); 2316 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2317 double perLoop = elapsed / (double) n; 2318 double perLoop100 = perLoop * 0.01; 2319 double perLoop1k = perLoop * 0.001; 2320 double mean = mWcStats.mean(); 2321 double stddev = mWcStats.stddev(); 2322 double minimum = mWcStats.minimum(); 2323 double maximum = mWcStats.maximum(); 2324 double meanCycles = mHzStats.mean(); 2325 double stddevCycles = mHzStats.stddev(); 2326 double minCycles = mHzStats.minimum(); 2327 double maxCycles = mHzStats.maximum(); 2328 mCpuUsage.resetElapsed(); 2329 mWcStats.reset(); 2330 mHzStats.reset(); 2331 ALOGD("CPU usage for %s over past %.1f secs\n" 2332 " (%u mixer loops at %.1f mean ms per loop):\n" 2333 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2334 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2335 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2336 title.string(), 2337 elapsed * .000000001, n, perLoop * .000001, 2338 mean * .001, 2339 stddev * .001, 2340 minimum * .001, 2341 maximum * .001, 2342 mean / perLoop100, 2343 stddev / perLoop100, 2344 minimum / perLoop100, 2345 maximum / perLoop100, 2346 meanCycles / perLoop1k, 2347 stddevCycles / perLoop1k, 2348 minCycles / perLoop1k, 2349 maxCycles / perLoop1k); 2350 2351 } 2352 } 2353#endif 2354}; 2355 2356void AudioFlinger::PlaybackThread::checkSilentMode_l() 2357{ 2358 if (!mMasterMute) { 2359 char value[PROPERTY_VALUE_MAX]; 2360 if (property_get("ro.audio.silent", value, "0") > 0) { 2361 char *endptr; 2362 unsigned long ul = strtoul(value, &endptr, 0); 2363 if (*endptr == '\0' && ul != 0) { 2364 ALOGD("Silence is golden"); 2365 // The setprop command will not allow a property to be changed after 2366 // the first time it is set, so we don't have to worry about un-muting. 2367 setMasterMute_l(true); 2368 } 2369 } 2370 } 2371} 2372 2373bool AudioFlinger::PlaybackThread::threadLoop() 2374{ 2375 Vector< sp<Track> > tracksToRemove; 2376 2377 standbyTime = systemTime(); 2378 2379 // MIXER 2380 nsecs_t lastWarning = 0; 2381if (mType == MIXER) { 2382 longStandbyExit = false; 2383} 2384 2385 // DUPLICATING 2386 // FIXME could this be made local to while loop? 2387 writeFrames = 0; 2388 2389 cacheParameters_l(); 2390 sleepTime = idleSleepTime; 2391 2392if (mType == MIXER) { 2393 sleepTimeShift = 0; 2394} 2395 2396 CpuStats cpuStats; 2397 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2398 2399 acquireWakeLock(); 2400 2401 while (!exitPending()) 2402 { 2403 cpuStats.sample(myName); 2404 2405 Vector< sp<EffectChain> > effectChains; 2406 2407 processConfigEvents(); 2408 2409 { // scope for mLock 2410 2411 Mutex::Autolock _l(mLock); 2412 2413 if (checkForNewParameters_l()) { 2414 cacheParameters_l(); 2415 } 2416 2417 saveOutputTracks(); 2418 2419 // put audio hardware into standby after short delay 2420 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2421 mSuspended > 0)) { 2422 if (!mStandby) { 2423 2424 threadLoop_standby(); 2425 2426 mStandby = true; 2427 mBytesWritten = 0; 2428 } 2429 2430 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2431 // we're about to wait, flush the binder command buffer 2432 IPCThreadState::self()->flushCommands(); 2433 2434 clearOutputTracks(); 2435 2436 if (exitPending()) break; 2437 2438 releaseWakeLock_l(); 2439 // wait until we have something to do... 2440 ALOGV("%s going to sleep", myName.string()); 2441 mWaitWorkCV.wait(mLock); 2442 ALOGV("%s waking up", myName.string()); 2443 acquireWakeLock_l(); 2444 2445 mMixerStatus = MIXER_IDLE; 2446 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2447 2448 checkSilentMode_l(); 2449 2450 standbyTime = systemTime() + standbyDelay; 2451 sleepTime = idleSleepTime; 2452 if (mType == MIXER) { 2453 sleepTimeShift = 0; 2454 } 2455 2456 continue; 2457 } 2458 } 2459 2460 // mMixerStatusIgnoringFastTracks is also updated internally 2461 mMixerStatus = prepareTracks_l(&tracksToRemove); 2462 2463 // prevent any changes in effect chain list and in each effect chain 2464 // during mixing and effect process as the audio buffers could be deleted 2465 // or modified if an effect is created or deleted 2466 lockEffectChains_l(effectChains); 2467 } 2468 2469 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2470 threadLoop_mix(); 2471 } else { 2472 threadLoop_sleepTime(); 2473 } 2474 2475 if (mSuspended > 0) { 2476 sleepTime = suspendSleepTimeUs(); 2477 } 2478 2479 // only process effects if we're going to write 2480 if (sleepTime == 0) { 2481 for (size_t i = 0; i < effectChains.size(); i ++) { 2482 effectChains[i]->process_l(); 2483 } 2484 } 2485 2486 // enable changes in effect chain 2487 unlockEffectChains(effectChains); 2488 2489 // sleepTime == 0 means we must write to audio hardware 2490 if (sleepTime == 0) { 2491 2492 threadLoop_write(); 2493 2494if (mType == MIXER) { 2495 // write blocked detection 2496 nsecs_t now = systemTime(); 2497 nsecs_t delta = now - mLastWriteTime; 2498 if (!mStandby && delta > maxPeriod) { 2499 mNumDelayedWrites++; 2500 if ((now - lastWarning) > kWarningThrottleNs) { 2501 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2502 ns2ms(delta), mNumDelayedWrites, this); 2503 lastWarning = now; 2504 } 2505 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2506 // a different threshold. Or completely removed for what it is worth anyway... 2507 if (mStandby) { 2508 longStandbyExit = true; 2509 } 2510 } 2511} 2512 2513 mStandby = false; 2514 } else { 2515 usleep(sleepTime); 2516 } 2517 2518 // Finally let go of removed track(s), without the lock held 2519 // since we can't guarantee the destructors won't acquire that 2520 // same lock. This will also mutate and push a new fast mixer state. 2521 threadLoop_removeTracks(tracksToRemove); 2522 tracksToRemove.clear(); 2523 2524 // FIXME I don't understand the need for this here; 2525 // it was in the original code but maybe the 2526 // assignment in saveOutputTracks() makes this unnecessary? 2527 clearOutputTracks(); 2528 2529 // Effect chains will be actually deleted here if they were removed from 2530 // mEffectChains list during mixing or effects processing 2531 effectChains.clear(); 2532 2533 // FIXME Note that the above .clear() is no longer necessary since effectChains 2534 // is now local to this block, but will keep it for now (at least until merge done). 2535 } 2536 2537if (mType == MIXER || mType == DIRECT) { 2538 // put output stream into standby mode 2539 if (!mStandby) { 2540 mOutput->stream->common.standby(&mOutput->stream->common); 2541 } 2542} 2543if (mType == DUPLICATING) { 2544 // for DuplicatingThread, standby mode is handled by the outputTracks 2545} 2546 2547 releaseWakeLock(); 2548 2549 ALOGV("Thread %p type %d exiting", this, mType); 2550 return false; 2551} 2552 2553// returns (via tracksToRemove) a set of tracks to remove. 2554void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2555{ 2556 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2557} 2558 2559void AudioFlinger::MixerThread::threadLoop_write() 2560{ 2561 // FIXME we should only do one push per cycle; confirm this is true 2562 // Start the fast mixer if it's not already running 2563 if (mFastMixer != NULL) { 2564 FastMixerStateQueue *sq = mFastMixer->sq(); 2565 FastMixerState *state = sq->begin(); 2566 if (state->mCommand != FastMixerState::MIX_WRITE && 2567 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2568 if (state->mCommand == FastMixerState::COLD_IDLE) { 2569 int32_t old = android_atomic_inc(&mFastMixerFutex); 2570 if (old == -1) { 2571 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2572 } 2573 } 2574 state->mCommand = FastMixerState::MIX_WRITE; 2575 sq->end(); 2576 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2577 if (kUseFastMixer == FastMixer_Dynamic) { 2578 mNormalSink = mPipeSink; 2579 } 2580 } else { 2581 sq->end(false /*didModify*/); 2582 } 2583 } 2584 PlaybackThread::threadLoop_write(); 2585} 2586 2587// shared by MIXER and DIRECT, overridden by DUPLICATING 2588void AudioFlinger::PlaybackThread::threadLoop_write() 2589{ 2590 // FIXME rewrite to reduce number of system calls 2591 mLastWriteTime = systemTime(); 2592 mInWrite = true; 2593 2594#define mBitShift 2 // FIXME 2595 size_t count = mixBufferSize >> mBitShift; 2596 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2597 if (framesWritten > 0) { 2598 size_t bytesWritten = framesWritten << mBitShift; 2599 mBytesWritten += bytesWritten; 2600 } 2601 2602 mNumWrites++; 2603 mInWrite = false; 2604} 2605 2606void AudioFlinger::MixerThread::threadLoop_standby() 2607{ 2608 // Idle the fast mixer if it's currently running 2609 if (mFastMixer != NULL) { 2610 FastMixerStateQueue *sq = mFastMixer->sq(); 2611 FastMixerState *state = sq->begin(); 2612 if (!(state->mCommand & FastMixerState::IDLE)) { 2613 state->mCommand = FastMixerState::COLD_IDLE; 2614 state->mColdFutexAddr = &mFastMixerFutex; 2615 state->mColdGen++; 2616 mFastMixerFutex = 0; 2617 sq->end(); 2618 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2619 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2620 if (kUseFastMixer == FastMixer_Dynamic) { 2621 mNormalSink = mOutputSink; 2622 } 2623 } else { 2624 sq->end(false /*didModify*/); 2625 } 2626 } 2627 PlaybackThread::threadLoop_standby(); 2628} 2629 2630// shared by MIXER and DIRECT, overridden by DUPLICATING 2631void AudioFlinger::PlaybackThread::threadLoop_standby() 2632{ 2633 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2634 mOutput->stream->common.standby(&mOutput->stream->common); 2635} 2636 2637void AudioFlinger::MixerThread::threadLoop_mix() 2638{ 2639 // obtain the presentation timestamp of the next output buffer 2640 int64_t pts; 2641 status_t status = INVALID_OPERATION; 2642 2643 if (NULL != mOutput->stream->get_next_write_timestamp) { 2644 status = mOutput->stream->get_next_write_timestamp( 2645 mOutput->stream, &pts); 2646 } 2647 2648 if (status != NO_ERROR) { 2649 pts = AudioBufferProvider::kInvalidPTS; 2650 } 2651 2652 // mix buffers... 2653 mAudioMixer->process(pts); 2654 // increase sleep time progressively when application underrun condition clears. 2655 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2656 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2657 // such that we would underrun the audio HAL. 2658 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2659 sleepTimeShift--; 2660 } 2661 sleepTime = 0; 2662 standbyTime = systemTime() + standbyDelay; 2663 //TODO: delay standby when effects have a tail 2664} 2665 2666void AudioFlinger::MixerThread::threadLoop_sleepTime() 2667{ 2668 // If no tracks are ready, sleep once for the duration of an output 2669 // buffer size, then write 0s to the output 2670 if (sleepTime == 0) { 2671 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2672 sleepTime = activeSleepTime >> sleepTimeShift; 2673 if (sleepTime < kMinThreadSleepTimeUs) { 2674 sleepTime = kMinThreadSleepTimeUs; 2675 } 2676 // reduce sleep time in case of consecutive application underruns to avoid 2677 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2678 // duration we would end up writing less data than needed by the audio HAL if 2679 // the condition persists. 2680 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2681 sleepTimeShift++; 2682 } 2683 } else { 2684 sleepTime = idleSleepTime; 2685 } 2686 } else if (mBytesWritten != 0 || 2687 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2688 memset (mMixBuffer, 0, mixBufferSize); 2689 sleepTime = 0; 2690 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2691 } 2692 // TODO add standby time extension fct of effect tail 2693} 2694 2695// prepareTracks_l() must be called with ThreadBase::mLock held 2696AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2697 Vector< sp<Track> > *tracksToRemove) 2698{ 2699 2700 mixer_state mixerStatus = MIXER_IDLE; 2701 // find out which tracks need to be processed 2702 size_t count = mActiveTracks.size(); 2703 size_t mixedTracks = 0; 2704 size_t tracksWithEffect = 0; 2705 // counts only _active_ fast tracks 2706 size_t fastTracks = 0; 2707 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2708 2709 float masterVolume = mMasterVolume; 2710 bool masterMute = mMasterMute; 2711 2712 if (masterMute) { 2713 masterVolume = 0; 2714 } 2715 // Delegate master volume control to effect in output mix effect chain if needed 2716 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2717 if (chain != 0) { 2718 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2719 chain->setVolume_l(&v, &v); 2720 masterVolume = (float)((v + (1 << 23)) >> 24); 2721 chain.clear(); 2722 } 2723 2724 // prepare a new state to push 2725 FastMixerStateQueue *sq = NULL; 2726 FastMixerState *state = NULL; 2727 bool didModify = false; 2728 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2729 if (mFastMixer != NULL) { 2730 sq = mFastMixer->sq(); 2731 state = sq->begin(); 2732 } 2733 2734 for (size_t i=0 ; i<count ; i++) { 2735 sp<Track> t = mActiveTracks[i].promote(); 2736 if (t == 0) continue; 2737 2738 // this const just means the local variable doesn't change 2739 Track* const track = t.get(); 2740 2741 // process fast tracks 2742 if (track->isFastTrack()) { 2743 2744 // It's theoretically possible (though unlikely) for a fast track to be created 2745 // and then removed within the same normal mix cycle. This is not a problem, as 2746 // the track never becomes active so it's fast mixer slot is never touched. 2747 // The converse, of removing an (active) track and then creating a new track 2748 // at the identical fast mixer slot within the same normal mix cycle, 2749 // is impossible because the slot isn't marked available until the end of each cycle. 2750 int j = track->mFastIndex; 2751 FastTrack *fastTrack = &state->mFastTracks[j]; 2752 2753 // Determine whether the track is currently in underrun condition, 2754 // and whether it had a recent underrun. 2755 uint32_t underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2756 uint32_t recentUnderruns = (underruns - (track->mObservedUnderruns & ~1)) >> 1; 2757 // don't count underruns that occur while stopping or pausing 2758 if (!(track->isStopped() || track->isPausing())) { 2759 track->mUnderrunCount += recentUnderruns; 2760 } 2761 track->mObservedUnderruns = underruns; 2762 2763 // This is similar to the formula for normal tracks, 2764 // with a few modifications for fast tracks. 2765 bool isActive; 2766 if (track->isStopped()) { 2767 // track stays active after stop() until first underrun 2768 isActive = recentUnderruns == 0; 2769 } else if (track->isPaused() || track->isTerminated()) { 2770 isActive = false; 2771 } else if (track->isPausing()) { 2772 // ramp down is not yet implemented 2773 isActive = true; 2774 track->setPaused(); 2775 } else if (track->isResuming()) { 2776 // ramp up is not yet implemented 2777 isActive = true; 2778 track->mState = TrackBase::ACTIVE; 2779 } else { 2780 // no minimum frame count for fast tracks; continual underrun is allowed, 2781 // but later could implement automatic pause after several consecutive underruns, 2782 // or auto-mute yet still consider the track active and continue to service it 2783 isActive = true; 2784 } 2785 2786 if (isActive) { 2787 // was it previously inactive? 2788 if (!(state->mTrackMask & (1 << j))) { 2789 ExtendedAudioBufferProvider *eabp = track; 2790 VolumeProvider *vp = track; 2791 fastTrack->mBufferProvider = eabp; 2792 fastTrack->mVolumeProvider = vp; 2793 fastTrack->mSampleRate = track->mSampleRate; 2794 fastTrack->mChannelMask = track->mChannelMask; 2795 fastTrack->mGeneration++; 2796 state->mTrackMask |= 1 << j; 2797 didModify = true; 2798 // no acknowledgement required for newly active tracks 2799 } 2800 // cache the combined master volume and stream type volume for fast mixer; this 2801 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2802 track->mCachedVolume = track->isMuted() ? 2803 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2804 ++fastTracks; 2805 } else { 2806 // was it previously active? 2807 if (state->mTrackMask & (1 << j)) { 2808 fastTrack->mBufferProvider = NULL; 2809 fastTrack->mGeneration++; 2810 state->mTrackMask &= ~(1 << j); 2811 didModify = true; 2812 // If any fast tracks were removed, we must wait for acknowledgement 2813 // because we're about to decrement the last sp<> on those tracks. 2814 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2815 } 2816 // Remainder of this block is copied from similar code for normal tracks 2817 if (track->isStopped()) { 2818 // Can't reset directly, as fast mixer is still polling this track 2819 // track->reset(); 2820 // So instead mark this track as needing to be reset after push with ack 2821 resetMask |= 1 << i; 2822 } 2823 // This would be incomplete if we auto-paused on underrun 2824 size_t audioHALFrames = 2825 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2826 size_t framesWritten = 2827 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2828 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2829 tracksToRemove->add(track); 2830 } 2831 // Avoids a misleading display in dumpsys 2832 track->mObservedUnderruns &= ~1; 2833 } 2834 continue; 2835 } 2836 2837 { // local variable scope to avoid goto warning 2838 2839 audio_track_cblk_t* cblk = track->cblk(); 2840 2841 // The first time a track is added we wait 2842 // for all its buffers to be filled before processing it 2843 int name = track->name(); 2844 // make sure that we have enough frames to mix one full buffer. 2845 // enforce this condition only once to enable draining the buffer in case the client 2846 // app does not call stop() and relies on underrun to stop: 2847 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2848 // during last round 2849 uint32_t minFrames = 1; 2850 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2851 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2852 if (t->sampleRate() == (int)mSampleRate) { 2853 minFrames = mNormalFrameCount; 2854 } else { 2855 // +1 for rounding and +1 for additional sample needed for interpolation 2856 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2857 // add frames already consumed but not yet released by the resampler 2858 // because cblk->framesReady() will include these frames 2859 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2860 // the minimum track buffer size is normally twice the number of frames necessary 2861 // to fill one buffer and the resampler should not leave more than one buffer worth 2862 // of unreleased frames after each pass, but just in case... 2863 ALOG_ASSERT(minFrames <= cblk->frameCount); 2864 } 2865 } 2866 if ((track->framesReady() >= minFrames) && track->isReady() && 2867 !track->isPaused() && !track->isTerminated()) 2868 { 2869 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2870 2871 mixedTracks++; 2872 2873 // track->mainBuffer() != mMixBuffer means there is an effect chain 2874 // connected to the track 2875 chain.clear(); 2876 if (track->mainBuffer() != mMixBuffer) { 2877 chain = getEffectChain_l(track->sessionId()); 2878 // Delegate volume control to effect in track effect chain if needed 2879 if (chain != 0) { 2880 tracksWithEffect++; 2881 } else { 2882 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2883 name, track->sessionId()); 2884 } 2885 } 2886 2887 2888 int param = AudioMixer::VOLUME; 2889 if (track->mFillingUpStatus == Track::FS_FILLED) { 2890 // no ramp for the first volume setting 2891 track->mFillingUpStatus = Track::FS_ACTIVE; 2892 if (track->mState == TrackBase::RESUMING) { 2893 track->mState = TrackBase::ACTIVE; 2894 param = AudioMixer::RAMP_VOLUME; 2895 } 2896 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2897 } else if (cblk->server != 0) { 2898 // If the track is stopped before the first frame was mixed, 2899 // do not apply ramp 2900 param = AudioMixer::RAMP_VOLUME; 2901 } 2902 2903 // compute volume for this track 2904 uint32_t vl, vr, va; 2905 if (track->isMuted() || track->isPausing() || 2906 mStreamTypes[track->streamType()].mute) { 2907 vl = vr = va = 0; 2908 if (track->isPausing()) { 2909 track->setPaused(); 2910 } 2911 } else { 2912 2913 // read original volumes with volume control 2914 float typeVolume = mStreamTypes[track->streamType()].volume; 2915 float v = masterVolume * typeVolume; 2916 uint32_t vlr = cblk->getVolumeLR(); 2917 vl = vlr & 0xFFFF; 2918 vr = vlr >> 16; 2919 // track volumes come from shared memory, so can't be trusted and must be clamped 2920 if (vl > MAX_GAIN_INT) { 2921 ALOGV("Track left volume out of range: %04X", vl); 2922 vl = MAX_GAIN_INT; 2923 } 2924 if (vr > MAX_GAIN_INT) { 2925 ALOGV("Track right volume out of range: %04X", vr); 2926 vr = MAX_GAIN_INT; 2927 } 2928 // now apply the master volume and stream type volume 2929 vl = (uint32_t)(v * vl) << 12; 2930 vr = (uint32_t)(v * vr) << 12; 2931 // assuming master volume and stream type volume each go up to 1.0, 2932 // vl and vr are now in 8.24 format 2933 2934 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2935 // send level comes from shared memory and so may be corrupt 2936 if (sendLevel > MAX_GAIN_INT) { 2937 ALOGV("Track send level out of range: %04X", sendLevel); 2938 sendLevel = MAX_GAIN_INT; 2939 } 2940 va = (uint32_t)(v * sendLevel); 2941 } 2942 // Delegate volume control to effect in track effect chain if needed 2943 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2944 // Do not ramp volume if volume is controlled by effect 2945 param = AudioMixer::VOLUME; 2946 track->mHasVolumeController = true; 2947 } else { 2948 // force no volume ramp when volume controller was just disabled or removed 2949 // from effect chain to avoid volume spike 2950 if (track->mHasVolumeController) { 2951 param = AudioMixer::VOLUME; 2952 } 2953 track->mHasVolumeController = false; 2954 } 2955 2956 // Convert volumes from 8.24 to 4.12 format 2957 // This additional clamping is needed in case chain->setVolume_l() overshot 2958 vl = (vl + (1 << 11)) >> 12; 2959 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2960 vr = (vr + (1 << 11)) >> 12; 2961 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2962 2963 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2964 2965 // XXX: these things DON'T need to be done each time 2966 mAudioMixer->setBufferProvider(name, track); 2967 mAudioMixer->enable(name); 2968 2969 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2970 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2971 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2972 mAudioMixer->setParameter( 2973 name, 2974 AudioMixer::TRACK, 2975 AudioMixer::FORMAT, (void *)track->format()); 2976 mAudioMixer->setParameter( 2977 name, 2978 AudioMixer::TRACK, 2979 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2980 mAudioMixer->setParameter( 2981 name, 2982 AudioMixer::RESAMPLE, 2983 AudioMixer::SAMPLE_RATE, 2984 (void *)(cblk->sampleRate)); 2985 mAudioMixer->setParameter( 2986 name, 2987 AudioMixer::TRACK, 2988 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2989 mAudioMixer->setParameter( 2990 name, 2991 AudioMixer::TRACK, 2992 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2993 2994 // reset retry count 2995 track->mRetryCount = kMaxTrackRetries; 2996 2997 // If one track is ready, set the mixer ready if: 2998 // - the mixer was not ready during previous round OR 2999 // - no other track is not ready 3000 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3001 mixerStatus != MIXER_TRACKS_ENABLED) { 3002 mixerStatus = MIXER_TRACKS_READY; 3003 } 3004 } else { 3005 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3006 if (track->isStopped()) { 3007 track->reset(); 3008 } 3009 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3010 track->isStopped() || track->isPaused()) { 3011 // We have consumed all the buffers of this track. 3012 // Remove it from the list of active tracks. 3013 // TODO: use actual buffer filling status instead of latency when available from 3014 // audio HAL 3015 size_t audioHALFrames = 3016 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3017 size_t framesWritten = 3018 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3019 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3020 tracksToRemove->add(track); 3021 } 3022 } else { 3023 // No buffers for this track. Give it a few chances to 3024 // fill a buffer, then remove it from active list. 3025 if (--(track->mRetryCount) <= 0) { 3026 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3027 tracksToRemove->add(track); 3028 // indicate to client process that the track was disabled because of underrun; 3029 // it will then automatically call start() when data is available 3030 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3031 // If one track is not ready, mark the mixer also not ready if: 3032 // - the mixer was ready during previous round OR 3033 // - no other track is ready 3034 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3035 mixerStatus != MIXER_TRACKS_READY) { 3036 mixerStatus = MIXER_TRACKS_ENABLED; 3037 } 3038 } 3039 mAudioMixer->disable(name); 3040 } 3041 3042 } // local variable scope to avoid goto warning 3043track_is_ready: ; 3044 3045 } 3046 3047 // Push the new FastMixer state if necessary 3048 if (didModify) { 3049 state->mFastTracksGen++; 3050 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3051 if (kUseFastMixer == FastMixer_Dynamic && 3052 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3053 state->mCommand = FastMixerState::COLD_IDLE; 3054 state->mColdFutexAddr = &mFastMixerFutex; 3055 state->mColdGen++; 3056 mFastMixerFutex = 0; 3057 if (kUseFastMixer == FastMixer_Dynamic) { 3058 mNormalSink = mOutputSink; 3059 } 3060 // If we go into cold idle, need to wait for acknowledgement 3061 // so that fast mixer stops doing I/O. 3062 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3063 } 3064 sq->end(); 3065 } 3066 if (sq != NULL) { 3067 sq->end(didModify); 3068 sq->push(block); 3069 } 3070 3071 // Now perform the deferred reset on fast tracks that have stopped 3072 while (resetMask != 0) { 3073 size_t i = __builtin_ctz(resetMask); 3074 ALOG_ASSERT(i < count); 3075 resetMask &= ~(1 << i); 3076 sp<Track> t = mActiveTracks[i].promote(); 3077 if (t == 0) continue; 3078 Track* track = t.get(); 3079 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3080 track->reset(); 3081 } 3082 3083 // remove all the tracks that need to be... 3084 count = tracksToRemove->size(); 3085 if (CC_UNLIKELY(count)) { 3086 for (size_t i=0 ; i<count ; i++) { 3087 const sp<Track>& track = tracksToRemove->itemAt(i); 3088 mActiveTracks.remove(track); 3089 if (track->mainBuffer() != mMixBuffer) { 3090 chain = getEffectChain_l(track->sessionId()); 3091 if (chain != 0) { 3092 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3093 chain->decActiveTrackCnt(); 3094 } 3095 } 3096 if (track->isTerminated()) { 3097 removeTrack_l(track); 3098 } 3099 } 3100 } 3101 3102 // mix buffer must be cleared if all tracks are connected to an 3103 // effect chain as in this case the mixer will not write to 3104 // mix buffer and track effects will accumulate into it 3105 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3106 // FIXME as a performance optimization, should remember previous zero status 3107 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3108 } 3109 3110 // if any fast tracks, then status is ready 3111 mMixerStatusIgnoringFastTracks = mixerStatus; 3112 if (fastTracks > 0) { 3113 mixerStatus = MIXER_TRACKS_READY; 3114 } 3115 return mixerStatus; 3116} 3117 3118/* 3119The derived values that are cached: 3120 - mixBufferSize from frame count * frame size 3121 - activeSleepTime from activeSleepTimeUs() 3122 - idleSleepTime from idleSleepTimeUs() 3123 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3124 - maxPeriod from frame count and sample rate (MIXER only) 3125 3126The parameters that affect these derived values are: 3127 - frame count 3128 - frame size 3129 - sample rate 3130 - device type: A2DP or not 3131 - device latency 3132 - format: PCM or not 3133 - active sleep time 3134 - idle sleep time 3135*/ 3136 3137void AudioFlinger::PlaybackThread::cacheParameters_l() 3138{ 3139 mixBufferSize = mNormalFrameCount * mFrameSize; 3140 activeSleepTime = activeSleepTimeUs(); 3141 idleSleepTime = idleSleepTimeUs(); 3142} 3143 3144void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3145{ 3146 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3147 this, streamType, mTracks.size()); 3148 Mutex::Autolock _l(mLock); 3149 3150 size_t size = mTracks.size(); 3151 for (size_t i = 0; i < size; i++) { 3152 sp<Track> t = mTracks[i]; 3153 if (t->streamType() == streamType) { 3154 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3155 t->mCblk->cv.signal(); 3156 } 3157 } 3158} 3159 3160// getTrackName_l() must be called with ThreadBase::mLock held 3161int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3162{ 3163 return mAudioMixer->getTrackName(channelMask); 3164} 3165 3166// deleteTrackName_l() must be called with ThreadBase::mLock held 3167void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3168{ 3169 ALOGV("remove track (%d) and delete from mixer", name); 3170 mAudioMixer->deleteTrackName(name); 3171} 3172 3173// checkForNewParameters_l() must be called with ThreadBase::mLock held 3174bool AudioFlinger::MixerThread::checkForNewParameters_l() 3175{ 3176 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3177 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3178 bool reconfig = false; 3179 3180 while (!mNewParameters.isEmpty()) { 3181 3182 if (mFastMixer != NULL) { 3183 FastMixerStateQueue *sq = mFastMixer->sq(); 3184 FastMixerState *state = sq->begin(); 3185 if (!(state->mCommand & FastMixerState::IDLE)) { 3186 previousCommand = state->mCommand; 3187 state->mCommand = FastMixerState::HOT_IDLE; 3188 sq->end(); 3189 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3190 } else { 3191 sq->end(false /*didModify*/); 3192 } 3193 } 3194 3195 status_t status = NO_ERROR; 3196 String8 keyValuePair = mNewParameters[0]; 3197 AudioParameter param = AudioParameter(keyValuePair); 3198 int value; 3199 3200 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3201 reconfig = true; 3202 } 3203 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3204 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3205 status = BAD_VALUE; 3206 } else { 3207 reconfig = true; 3208 } 3209 } 3210 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3211 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3212 status = BAD_VALUE; 3213 } else { 3214 reconfig = true; 3215 } 3216 } 3217 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3218 // do not accept frame count changes if tracks are open as the track buffer 3219 // size depends on frame count and correct behavior would not be guaranteed 3220 // if frame count is changed after track creation 3221 if (!mTracks.isEmpty()) { 3222 status = INVALID_OPERATION; 3223 } else { 3224 reconfig = true; 3225 } 3226 } 3227 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3228#ifdef ADD_BATTERY_DATA 3229 // when changing the audio output device, call addBatteryData to notify 3230 // the change 3231 if ((int)mDevice != value) { 3232 uint32_t params = 0; 3233 // check whether speaker is on 3234 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3235 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3236 } 3237 3238 int deviceWithoutSpeaker 3239 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3240 // check if any other device (except speaker) is on 3241 if (value & deviceWithoutSpeaker ) { 3242 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3243 } 3244 3245 if (params != 0) { 3246 addBatteryData(params); 3247 } 3248 } 3249#endif 3250 3251 // forward device change to effects that have requested to be 3252 // aware of attached audio device. 3253 mDevice = (uint32_t)value; 3254 for (size_t i = 0; i < mEffectChains.size(); i++) { 3255 mEffectChains[i]->setDevice_l(mDevice); 3256 } 3257 } 3258 3259 if (status == NO_ERROR) { 3260 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3261 keyValuePair.string()); 3262 if (!mStandby && status == INVALID_OPERATION) { 3263 mOutput->stream->common.standby(&mOutput->stream->common); 3264 mStandby = true; 3265 mBytesWritten = 0; 3266 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3267 keyValuePair.string()); 3268 } 3269 if (status == NO_ERROR && reconfig) { 3270 delete mAudioMixer; 3271 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3272 mAudioMixer = NULL; 3273 readOutputParameters(); 3274 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3275 for (size_t i = 0; i < mTracks.size() ; i++) { 3276 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3277 if (name < 0) break; 3278 mTracks[i]->mName = name; 3279 // limit track sample rate to 2 x new output sample rate 3280 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3281 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3282 } 3283 } 3284 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3285 } 3286 } 3287 3288 mNewParameters.removeAt(0); 3289 3290 mParamStatus = status; 3291 mParamCond.signal(); 3292 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3293 // already timed out waiting for the status and will never signal the condition. 3294 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3295 } 3296 3297 if (!(previousCommand & FastMixerState::IDLE)) { 3298 ALOG_ASSERT(mFastMixer != NULL); 3299 FastMixerStateQueue *sq = mFastMixer->sq(); 3300 FastMixerState *state = sq->begin(); 3301 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3302 state->mCommand = previousCommand; 3303 sq->end(); 3304 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3305 } 3306 3307 return reconfig; 3308} 3309 3310status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3311{ 3312 const size_t SIZE = 256; 3313 char buffer[SIZE]; 3314 String8 result; 3315 3316 PlaybackThread::dumpInternals(fd, args); 3317 3318 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3319 result.append(buffer); 3320 write(fd, result.string(), result.size()); 3321 3322 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3323 FastMixerDumpState copy = mFastMixerDumpState; 3324 copy.dump(fd); 3325 3326 return NO_ERROR; 3327} 3328 3329uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3330{ 3331 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3332} 3333 3334uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3335{ 3336 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3337} 3338 3339void AudioFlinger::MixerThread::cacheParameters_l() 3340{ 3341 PlaybackThread::cacheParameters_l(); 3342 3343 // FIXME: Relaxed timing because of a certain device that can't meet latency 3344 // Should be reduced to 2x after the vendor fixes the driver issue 3345 // increase threshold again due to low power audio mode. The way this warning 3346 // threshold is calculated and its usefulness should be reconsidered anyway. 3347 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3348} 3349 3350// ---------------------------------------------------------------------------- 3351AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3352 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3353 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3354 // mLeftVolFloat, mRightVolFloat 3355 // mLeftVolShort, mRightVolShort 3356{ 3357} 3358 3359AudioFlinger::DirectOutputThread::~DirectOutputThread() 3360{ 3361} 3362 3363AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3364 Vector< sp<Track> > *tracksToRemove 3365) 3366{ 3367 sp<Track> trackToRemove; 3368 3369 mixer_state mixerStatus = MIXER_IDLE; 3370 3371 // find out which tracks need to be processed 3372 if (mActiveTracks.size() != 0) { 3373 sp<Track> t = mActiveTracks[0].promote(); 3374 // The track died recently 3375 if (t == 0) return MIXER_IDLE; 3376 3377 Track* const track = t.get(); 3378 audio_track_cblk_t* cblk = track->cblk(); 3379 3380 // The first time a track is added we wait 3381 // for all its buffers to be filled before processing it 3382 if (cblk->framesReady() && track->isReady() && 3383 !track->isPaused() && !track->isTerminated()) 3384 { 3385 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3386 3387 if (track->mFillingUpStatus == Track::FS_FILLED) { 3388 track->mFillingUpStatus = Track::FS_ACTIVE; 3389 mLeftVolFloat = mRightVolFloat = 0; 3390 mLeftVolShort = mRightVolShort = 0; 3391 if (track->mState == TrackBase::RESUMING) { 3392 track->mState = TrackBase::ACTIVE; 3393 rampVolume = true; 3394 } 3395 } else if (cblk->server != 0) { 3396 // If the track is stopped before the first frame was mixed, 3397 // do not apply ramp 3398 rampVolume = true; 3399 } 3400 // compute volume for this track 3401 float left, right; 3402 if (track->isMuted() || mMasterMute || track->isPausing() || 3403 mStreamTypes[track->streamType()].mute) { 3404 left = right = 0; 3405 if (track->isPausing()) { 3406 track->setPaused(); 3407 } 3408 } else { 3409 float typeVolume = mStreamTypes[track->streamType()].volume; 3410 float v = mMasterVolume * typeVolume; 3411 uint32_t vlr = cblk->getVolumeLR(); 3412 float v_clamped = v * (vlr & 0xFFFF); 3413 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3414 left = v_clamped/MAX_GAIN; 3415 v_clamped = v * (vlr >> 16); 3416 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3417 right = v_clamped/MAX_GAIN; 3418 } 3419 3420 if (left != mLeftVolFloat || right != mRightVolFloat) { 3421 mLeftVolFloat = left; 3422 mRightVolFloat = right; 3423 3424 // If audio HAL implements volume control, 3425 // force software volume to nominal value 3426 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3427 left = 1.0f; 3428 right = 1.0f; 3429 } 3430 3431 // Convert volumes from float to 8.24 3432 uint32_t vl = (uint32_t)(left * (1 << 24)); 3433 uint32_t vr = (uint32_t)(right * (1 << 24)); 3434 3435 // Delegate volume control to effect in track effect chain if needed 3436 // only one effect chain can be present on DirectOutputThread, so if 3437 // there is one, the track is connected to it 3438 if (!mEffectChains.isEmpty()) { 3439 // Do not ramp volume if volume is controlled by effect 3440 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3441 rampVolume = false; 3442 } 3443 } 3444 3445 // Convert volumes from 8.24 to 4.12 format 3446 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3447 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3448 leftVol = (uint16_t)v_clamped; 3449 v_clamped = (vr + (1 << 11)) >> 12; 3450 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3451 rightVol = (uint16_t)v_clamped; 3452 } else { 3453 leftVol = mLeftVolShort; 3454 rightVol = mRightVolShort; 3455 rampVolume = false; 3456 } 3457 3458 // reset retry count 3459 track->mRetryCount = kMaxTrackRetriesDirect; 3460 mActiveTrack = t; 3461 mixerStatus = MIXER_TRACKS_READY; 3462 } else { 3463 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3464 if (track->isStopped()) { 3465 track->reset(); 3466 } 3467 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3468 // We have consumed all the buffers of this track. 3469 // Remove it from the list of active tracks. 3470 // TODO: implement behavior for compressed audio 3471 size_t audioHALFrames = 3472 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3473 size_t framesWritten = 3474 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3475 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3476 trackToRemove = track; 3477 } 3478 } else { 3479 // No buffers for this track. Give it a few chances to 3480 // fill a buffer, then remove it from active list. 3481 if (--(track->mRetryCount) <= 0) { 3482 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3483 trackToRemove = track; 3484 } else { 3485 mixerStatus = MIXER_TRACKS_ENABLED; 3486 } 3487 } 3488 } 3489 } 3490 3491 // FIXME merge this with similar code for removing multiple tracks 3492 // remove all the tracks that need to be... 3493 if (CC_UNLIKELY(trackToRemove != 0)) { 3494 tracksToRemove->add(trackToRemove); 3495 mActiveTracks.remove(trackToRemove); 3496 if (!mEffectChains.isEmpty()) { 3497 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3498 trackToRemove->sessionId()); 3499 mEffectChains[0]->decActiveTrackCnt(); 3500 } 3501 if (trackToRemove->isTerminated()) { 3502 removeTrack_l(trackToRemove); 3503 } 3504 } 3505 3506 return mixerStatus; 3507} 3508 3509void AudioFlinger::DirectOutputThread::threadLoop_mix() 3510{ 3511 AudioBufferProvider::Buffer buffer; 3512 size_t frameCount = mFrameCount; 3513 int8_t *curBuf = (int8_t *)mMixBuffer; 3514 // output audio to hardware 3515 while (frameCount) { 3516 buffer.frameCount = frameCount; 3517 mActiveTrack->getNextBuffer(&buffer); 3518 if (CC_UNLIKELY(buffer.raw == NULL)) { 3519 memset(curBuf, 0, frameCount * mFrameSize); 3520 break; 3521 } 3522 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3523 frameCount -= buffer.frameCount; 3524 curBuf += buffer.frameCount * mFrameSize; 3525 mActiveTrack->releaseBuffer(&buffer); 3526 } 3527 sleepTime = 0; 3528 standbyTime = systemTime() + standbyDelay; 3529 mActiveTrack.clear(); 3530 3531 // apply volume 3532 3533 // Do not apply volume on compressed audio 3534 if (!audio_is_linear_pcm(mFormat)) { 3535 return; 3536 } 3537 3538 // convert to signed 16 bit before volume calculation 3539 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3540 size_t count = mFrameCount * mChannelCount; 3541 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3542 int16_t *dst = mMixBuffer + count-1; 3543 while (count--) { 3544 *dst-- = (int16_t)(*src--^0x80) << 8; 3545 } 3546 } 3547 3548 frameCount = mFrameCount; 3549 int16_t *out = mMixBuffer; 3550 if (rampVolume) { 3551 if (mChannelCount == 1) { 3552 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3553 int32_t vlInc = d / (int32_t)frameCount; 3554 int32_t vl = ((int32_t)mLeftVolShort << 16); 3555 do { 3556 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3557 out++; 3558 vl += vlInc; 3559 } while (--frameCount); 3560 3561 } else { 3562 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3563 int32_t vlInc = d / (int32_t)frameCount; 3564 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3565 int32_t vrInc = d / (int32_t)frameCount; 3566 int32_t vl = ((int32_t)mLeftVolShort << 16); 3567 int32_t vr = ((int32_t)mRightVolShort << 16); 3568 do { 3569 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3570 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3571 out += 2; 3572 vl += vlInc; 3573 vr += vrInc; 3574 } while (--frameCount); 3575 } 3576 } else { 3577 if (mChannelCount == 1) { 3578 do { 3579 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3580 out++; 3581 } while (--frameCount); 3582 } else { 3583 do { 3584 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3585 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3586 out += 2; 3587 } while (--frameCount); 3588 } 3589 } 3590 3591 // convert back to unsigned 8 bit after volume calculation 3592 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3593 size_t count = mFrameCount * mChannelCount; 3594 int16_t *src = mMixBuffer; 3595 uint8_t *dst = (uint8_t *)mMixBuffer; 3596 while (count--) { 3597 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3598 } 3599 } 3600 3601 mLeftVolShort = leftVol; 3602 mRightVolShort = rightVol; 3603} 3604 3605void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3606{ 3607 if (sleepTime == 0) { 3608 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3609 sleepTime = activeSleepTime; 3610 } else { 3611 sleepTime = idleSleepTime; 3612 } 3613 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3614 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3615 sleepTime = 0; 3616 } 3617} 3618 3619// getTrackName_l() must be called with ThreadBase::mLock held 3620int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3621{ 3622 return 0; 3623} 3624 3625// deleteTrackName_l() must be called with ThreadBase::mLock held 3626void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3627{ 3628} 3629 3630// checkForNewParameters_l() must be called with ThreadBase::mLock held 3631bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3632{ 3633 bool reconfig = false; 3634 3635 while (!mNewParameters.isEmpty()) { 3636 status_t status = NO_ERROR; 3637 String8 keyValuePair = mNewParameters[0]; 3638 AudioParameter param = AudioParameter(keyValuePair); 3639 int value; 3640 3641 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3642 // do not accept frame count changes if tracks are open as the track buffer 3643 // size depends on frame count and correct behavior would not be garantied 3644 // if frame count is changed after track creation 3645 if (!mTracks.isEmpty()) { 3646 status = INVALID_OPERATION; 3647 } else { 3648 reconfig = true; 3649 } 3650 } 3651 if (status == NO_ERROR) { 3652 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3653 keyValuePair.string()); 3654 if (!mStandby && status == INVALID_OPERATION) { 3655 mOutput->stream->common.standby(&mOutput->stream->common); 3656 mStandby = true; 3657 mBytesWritten = 0; 3658 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3659 keyValuePair.string()); 3660 } 3661 if (status == NO_ERROR && reconfig) { 3662 readOutputParameters(); 3663 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3664 } 3665 } 3666 3667 mNewParameters.removeAt(0); 3668 3669 mParamStatus = status; 3670 mParamCond.signal(); 3671 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3672 // already timed out waiting for the status and will never signal the condition. 3673 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3674 } 3675 return reconfig; 3676} 3677 3678uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3679{ 3680 uint32_t time; 3681 if (audio_is_linear_pcm(mFormat)) { 3682 time = PlaybackThread::activeSleepTimeUs(); 3683 } else { 3684 time = 10000; 3685 } 3686 return time; 3687} 3688 3689uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3690{ 3691 uint32_t time; 3692 if (audio_is_linear_pcm(mFormat)) { 3693 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3694 } else { 3695 time = 10000; 3696 } 3697 return time; 3698} 3699 3700uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3701{ 3702 uint32_t time; 3703 if (audio_is_linear_pcm(mFormat)) { 3704 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3705 } else { 3706 time = 10000; 3707 } 3708 return time; 3709} 3710 3711void AudioFlinger::DirectOutputThread::cacheParameters_l() 3712{ 3713 PlaybackThread::cacheParameters_l(); 3714 3715 // use shorter standby delay as on normal output to release 3716 // hardware resources as soon as possible 3717 standbyDelay = microseconds(activeSleepTime*2); 3718} 3719 3720// ---------------------------------------------------------------------------- 3721 3722AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3723 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3724 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3725 mWaitTimeMs(UINT_MAX) 3726{ 3727 addOutputTrack(mainThread); 3728} 3729 3730AudioFlinger::DuplicatingThread::~DuplicatingThread() 3731{ 3732 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3733 mOutputTracks[i]->destroy(); 3734 } 3735} 3736 3737void AudioFlinger::DuplicatingThread::threadLoop_mix() 3738{ 3739 // mix buffers... 3740 if (outputsReady(outputTracks)) { 3741 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3742 } else { 3743 memset(mMixBuffer, 0, mixBufferSize); 3744 } 3745 sleepTime = 0; 3746 writeFrames = mNormalFrameCount; 3747} 3748 3749void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3750{ 3751 if (sleepTime == 0) { 3752 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3753 sleepTime = activeSleepTime; 3754 } else { 3755 sleepTime = idleSleepTime; 3756 } 3757 } else if (mBytesWritten != 0) { 3758 // flush remaining overflow buffers in output tracks 3759 for (size_t i = 0; i < outputTracks.size(); i++) { 3760 if (outputTracks[i]->isActive()) { 3761 sleepTime = 0; 3762 writeFrames = 0; 3763 memset(mMixBuffer, 0, mixBufferSize); 3764 break; 3765 } 3766 } 3767 } 3768} 3769 3770void AudioFlinger::DuplicatingThread::threadLoop_write() 3771{ 3772 standbyTime = systemTime() + standbyDelay; 3773 for (size_t i = 0; i < outputTracks.size(); i++) { 3774 outputTracks[i]->write(mMixBuffer, writeFrames); 3775 } 3776 mBytesWritten += mixBufferSize; 3777} 3778 3779void AudioFlinger::DuplicatingThread::threadLoop_standby() 3780{ 3781 // DuplicatingThread implements standby by stopping all tracks 3782 for (size_t i = 0; i < outputTracks.size(); i++) { 3783 outputTracks[i]->stop(); 3784 } 3785} 3786 3787void AudioFlinger::DuplicatingThread::saveOutputTracks() 3788{ 3789 outputTracks = mOutputTracks; 3790} 3791 3792void AudioFlinger::DuplicatingThread::clearOutputTracks() 3793{ 3794 outputTracks.clear(); 3795} 3796 3797void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3798{ 3799 Mutex::Autolock _l(mLock); 3800 // FIXME explain this formula 3801 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3802 OutputTrack *outputTrack = new OutputTrack(thread, 3803 this, 3804 mSampleRate, 3805 mFormat, 3806 mChannelMask, 3807 frameCount); 3808 if (outputTrack->cblk() != NULL) { 3809 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3810 mOutputTracks.add(outputTrack); 3811 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3812 updateWaitTime_l(); 3813 } 3814} 3815 3816void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3817{ 3818 Mutex::Autolock _l(mLock); 3819 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3820 if (mOutputTracks[i]->thread() == thread) { 3821 mOutputTracks[i]->destroy(); 3822 mOutputTracks.removeAt(i); 3823 updateWaitTime_l(); 3824 return; 3825 } 3826 } 3827 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3828} 3829 3830// caller must hold mLock 3831void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3832{ 3833 mWaitTimeMs = UINT_MAX; 3834 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3835 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3836 if (strong != 0) { 3837 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3838 if (waitTimeMs < mWaitTimeMs) { 3839 mWaitTimeMs = waitTimeMs; 3840 } 3841 } 3842 } 3843} 3844 3845 3846bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3847{ 3848 for (size_t i = 0; i < outputTracks.size(); i++) { 3849 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3850 if (thread == 0) { 3851 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3852 return false; 3853 } 3854 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3855 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3856 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3857 return false; 3858 } 3859 } 3860 return true; 3861} 3862 3863uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3864{ 3865 return (mWaitTimeMs * 1000) / 2; 3866} 3867 3868void AudioFlinger::DuplicatingThread::cacheParameters_l() 3869{ 3870 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3871 updateWaitTime_l(); 3872 3873 MixerThread::cacheParameters_l(); 3874} 3875 3876// ---------------------------------------------------------------------------- 3877 3878// TrackBase constructor must be called with AudioFlinger::mLock held 3879AudioFlinger::ThreadBase::TrackBase::TrackBase( 3880 ThreadBase *thread, 3881 const sp<Client>& client, 3882 uint32_t sampleRate, 3883 audio_format_t format, 3884 uint32_t channelMask, 3885 int frameCount, 3886 const sp<IMemory>& sharedBuffer, 3887 int sessionId) 3888 : RefBase(), 3889 mThread(thread), 3890 mClient(client), 3891 mCblk(NULL), 3892 // mBuffer 3893 // mBufferEnd 3894 mFrameCount(0), 3895 mState(IDLE), 3896 mSampleRate(sampleRate), 3897 mFormat(format), 3898 mStepServerFailed(false), 3899 mSessionId(sessionId) 3900 // mChannelCount 3901 // mChannelMask 3902{ 3903 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3904 3905 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3906 size_t size = sizeof(audio_track_cblk_t); 3907 uint8_t channelCount = popcount(channelMask); 3908 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3909 if (sharedBuffer == 0) { 3910 size += bufferSize; 3911 } 3912 3913 if (client != NULL) { 3914 mCblkMemory = client->heap()->allocate(size); 3915 if (mCblkMemory != 0) { 3916 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3917 if (mCblk != NULL) { // construct the shared structure in-place. 3918 new(mCblk) audio_track_cblk_t(); 3919 // clear all buffers 3920 mCblk->frameCount = frameCount; 3921 mCblk->sampleRate = sampleRate; 3922// uncomment the following lines to quickly test 32-bit wraparound 3923// mCblk->user = 0xffff0000; 3924// mCblk->server = 0xffff0000; 3925// mCblk->userBase = 0xffff0000; 3926// mCblk->serverBase = 0xffff0000; 3927 mChannelCount = channelCount; 3928 mChannelMask = channelMask; 3929 if (sharedBuffer == 0) { 3930 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3931 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3932 // Force underrun condition to avoid false underrun callback until first data is 3933 // written to buffer (other flags are cleared) 3934 mCblk->flags = CBLK_UNDERRUN_ON; 3935 } else { 3936 mBuffer = sharedBuffer->pointer(); 3937 } 3938 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3939 } 3940 } else { 3941 ALOGE("not enough memory for AudioTrack size=%u", size); 3942 client->heap()->dump("AudioTrack"); 3943 return; 3944 } 3945 } else { 3946 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3947 // construct the shared structure in-place. 3948 new(mCblk) audio_track_cblk_t(); 3949 // clear all buffers 3950 mCblk->frameCount = frameCount; 3951 mCblk->sampleRate = sampleRate; 3952// uncomment the following lines to quickly test 32-bit wraparound 3953// mCblk->user = 0xffff0000; 3954// mCblk->server = 0xffff0000; 3955// mCblk->userBase = 0xffff0000; 3956// mCblk->serverBase = 0xffff0000; 3957 mChannelCount = channelCount; 3958 mChannelMask = channelMask; 3959 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3960 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3961 // Force underrun condition to avoid false underrun callback until first data is 3962 // written to buffer (other flags are cleared) 3963 mCblk->flags = CBLK_UNDERRUN_ON; 3964 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3965 } 3966} 3967 3968AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3969{ 3970 if (mCblk != NULL) { 3971 if (mClient == 0) { 3972 delete mCblk; 3973 } else { 3974 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3975 } 3976 } 3977 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3978 if (mClient != 0) { 3979 // Client destructor must run with AudioFlinger mutex locked 3980 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3981 // If the client's reference count drops to zero, the associated destructor 3982 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3983 // relying on the automatic clear() at end of scope. 3984 mClient.clear(); 3985 } 3986} 3987 3988// AudioBufferProvider interface 3989// getNextBuffer() = 0; 3990// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3991void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3992{ 3993 buffer->raw = NULL; 3994 mFrameCount = buffer->frameCount; 3995 // FIXME See note at getNextBuffer() 3996 (void) step(); // ignore return value of step() 3997 buffer->frameCount = 0; 3998} 3999 4000bool AudioFlinger::ThreadBase::TrackBase::step() { 4001 bool result; 4002 audio_track_cblk_t* cblk = this->cblk(); 4003 4004 result = cblk->stepServer(mFrameCount); 4005 if (!result) { 4006 ALOGV("stepServer failed acquiring cblk mutex"); 4007 mStepServerFailed = true; 4008 } 4009 return result; 4010} 4011 4012void AudioFlinger::ThreadBase::TrackBase::reset() { 4013 audio_track_cblk_t* cblk = this->cblk(); 4014 4015 cblk->user = 0; 4016 cblk->server = 0; 4017 cblk->userBase = 0; 4018 cblk->serverBase = 0; 4019 mStepServerFailed = false; 4020 ALOGV("TrackBase::reset"); 4021} 4022 4023int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4024 return (int)mCblk->sampleRate; 4025} 4026 4027void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4028 audio_track_cblk_t* cblk = this->cblk(); 4029 size_t frameSize = cblk->frameSize; 4030 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4031 int8_t *bufferEnd = bufferStart + frames * frameSize; 4032 4033 // Check validity of returned pointer in case the track control block would have been corrupted. 4034 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4035 "TrackBase::getBuffer buffer out of range:\n" 4036 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4037 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4038 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4039 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4040 4041 return bufferStart; 4042} 4043 4044status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4045{ 4046 mSyncEvents.add(event); 4047 return NO_ERROR; 4048} 4049 4050// ---------------------------------------------------------------------------- 4051 4052// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4053AudioFlinger::PlaybackThread::Track::Track( 4054 PlaybackThread *thread, 4055 const sp<Client>& client, 4056 audio_stream_type_t streamType, 4057 uint32_t sampleRate, 4058 audio_format_t format, 4059 uint32_t channelMask, 4060 int frameCount, 4061 const sp<IMemory>& sharedBuffer, 4062 int sessionId, 4063 IAudioFlinger::track_flags_t flags) 4064 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4065 mMute(false), 4066 mFillingUpStatus(FS_INVALID), 4067 // mRetryCount initialized later when needed 4068 mSharedBuffer(sharedBuffer), 4069 mStreamType(streamType), 4070 mName(-1), // see note below 4071 mMainBuffer(thread->mixBuffer()), 4072 mAuxBuffer(NULL), 4073 mAuxEffectId(0), mHasVolumeController(false), 4074 mPresentationCompleteFrames(0), 4075 mFlags(flags), 4076 mFastIndex(-1), 4077 mObservedUnderruns(0), 4078 mUnderrunCount(0), 4079 mCachedVolume(1.0) 4080{ 4081 if (mCblk != NULL) { 4082 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4083 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4084 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4085 if (flags & IAudioFlinger::TRACK_FAST) { 4086 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4087 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4088 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4089 ALOG_ASSERT(0 < i && i < FastMixerState::kMaxFastTracks); 4090 // FIXME This is too eager. We allocate a fast track index before the 4091 // fast track becomes active. Since fast tracks are a scarce resource, 4092 // this means we are potentially denying other more important fast tracks from 4093 // being created. It would be better to allocate the index dynamically. 4094 mFastIndex = i; 4095 // Read the initial underruns because this field is never cleared by the fast mixer 4096 mObservedUnderruns = thread->getFastTrackUnderruns(i) & ~1; 4097 thread->mFastTrackAvailMask &= ~(1 << i); 4098 } 4099 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4100 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4101 if (mName < 0) { 4102 ALOGE("no more track names available"); 4103 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4104 // then we leak a fast track index. Should swap these two sections, or better yet 4105 // only allocate a normal mixer name for normal tracks. 4106 } 4107 } 4108 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4109} 4110 4111AudioFlinger::PlaybackThread::Track::~Track() 4112{ 4113 ALOGV("PlaybackThread::Track destructor"); 4114 sp<ThreadBase> thread = mThread.promote(); 4115 if (thread != 0) { 4116 Mutex::Autolock _l(thread->mLock); 4117 mState = TERMINATED; 4118 } 4119} 4120 4121void AudioFlinger::PlaybackThread::Track::destroy() 4122{ 4123 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4124 // by removing it from mTracks vector, so there is a risk that this Tracks's 4125 // destructor is called. As the destructor needs to lock mLock, 4126 // we must acquire a strong reference on this Track before locking mLock 4127 // here so that the destructor is called only when exiting this function. 4128 // On the other hand, as long as Track::destroy() is only called by 4129 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4130 // this Track with its member mTrack. 4131 sp<Track> keep(this); 4132 { // scope for mLock 4133 sp<ThreadBase> thread = mThread.promote(); 4134 if (thread != 0) { 4135 if (!isOutputTrack()) { 4136 if (mState == ACTIVE || mState == RESUMING) { 4137 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4138 4139#ifdef ADD_BATTERY_DATA 4140 // to track the speaker usage 4141 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4142#endif 4143 } 4144 AudioSystem::releaseOutput(thread->id()); 4145 } 4146 Mutex::Autolock _l(thread->mLock); 4147 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4148 playbackThread->destroyTrack_l(this); 4149 } 4150 } 4151} 4152 4153/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4154{ 4155 result.append(" Name Client Type Fmt Chn mask Session Frames S M F SRate L dB R dB " 4156 " Server User Main buf Aux Buf FastUnder\n"); 4157 4158} 4159 4160void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4161{ 4162 uint32_t vlr = mCblk->getVolumeLR(); 4163 if (isFastTrack()) { 4164 sprintf(buffer, " F %2d", mFastIndex); 4165 } else { 4166 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4167 } 4168 track_state state = mState; 4169 char stateChar; 4170 switch (state) { 4171 case IDLE: 4172 stateChar = 'I'; 4173 break; 4174 case TERMINATED: 4175 stateChar = 'T'; 4176 break; 4177 case STOPPED: 4178 stateChar = 'S'; 4179 break; 4180 case RESUMING: 4181 stateChar = 'R'; 4182 break; 4183 case ACTIVE: 4184 stateChar = 'A'; 4185 break; 4186 case PAUSING: 4187 stateChar = 'p'; 4188 break; 4189 case PAUSED: 4190 stateChar = 'P'; 4191 break; 4192 default: 4193 stateChar = '?'; 4194 break; 4195 } 4196 bool nowInUnderrun = mObservedUnderruns & 1; 4197 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %1c %1d %1d %5u %5.2g %5.2g " 4198 "0x%08x 0x%08x 0x%08x 0x%08x %9u%c\n", 4199 (mClient == 0) ? getpid_cached : mClient->pid(), 4200 mStreamType, 4201 mFormat, 4202 mChannelMask, 4203 mSessionId, 4204 mFrameCount, 4205 stateChar, 4206 mMute, 4207 mFillingUpStatus, 4208 mCblk->sampleRate, 4209 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4210 20.0 * log10((vlr >> 16) / 4096.0), 4211 mCblk->server, 4212 mCblk->user, 4213 (int)mMainBuffer, 4214 (int)mAuxBuffer, 4215 mUnderrunCount, 4216 nowInUnderrun ? '*' : ' '); 4217} 4218 4219// AudioBufferProvider interface 4220status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4221 AudioBufferProvider::Buffer* buffer, int64_t pts) 4222{ 4223 audio_track_cblk_t* cblk = this->cblk(); 4224 uint32_t framesReady; 4225 uint32_t framesReq = buffer->frameCount; 4226 4227 // Check if last stepServer failed, try to step now 4228 if (mStepServerFailed) { 4229 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4230 // Since the fast mixer is higher priority than client callback thread, 4231 // it does not result in priority inversion for client. 4232 // But a non-blocking solution would be preferable to avoid 4233 // fast mixer being unable to tryLock(), and 4234 // to avoid the extra context switches if the client wakes up, 4235 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4236 if (!step()) goto getNextBuffer_exit; 4237 ALOGV("stepServer recovered"); 4238 mStepServerFailed = false; 4239 } 4240 4241 // FIXME Same as above 4242 framesReady = cblk->framesReady(); 4243 4244 if (CC_LIKELY(framesReady)) { 4245 uint32_t s = cblk->server; 4246 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4247 4248 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4249 if (framesReq > framesReady) { 4250 framesReq = framesReady; 4251 } 4252 if (framesReq > bufferEnd - s) { 4253 framesReq = bufferEnd - s; 4254 } 4255 4256 buffer->raw = getBuffer(s, framesReq); 4257 if (buffer->raw == NULL) goto getNextBuffer_exit; 4258 4259 buffer->frameCount = framesReq; 4260 return NO_ERROR; 4261 } 4262 4263getNextBuffer_exit: 4264 buffer->raw = NULL; 4265 buffer->frameCount = 0; 4266 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4267 return NOT_ENOUGH_DATA; 4268} 4269 4270// Note that framesReady() takes a mutex on the control block using tryLock(). 4271// This could result in priority inversion if framesReady() is called by the normal mixer, 4272// as the normal mixer thread runs at lower 4273// priority than the client's callback thread: there is a short window within framesReady() 4274// during which the normal mixer could be preempted, and the client callback would block. 4275// Another problem can occur if framesReady() is called by the fast mixer: 4276// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4277// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4278size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4279 return mCblk->framesReady(); 4280} 4281 4282// Don't call for fast tracks; the framesReady() could result in priority inversion 4283bool AudioFlinger::PlaybackThread::Track::isReady() const { 4284 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4285 4286 if (framesReady() >= mCblk->frameCount || 4287 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4288 mFillingUpStatus = FS_FILLED; 4289 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4290 return true; 4291 } 4292 return false; 4293} 4294 4295status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4296 int triggerSession) 4297{ 4298 status_t status = NO_ERROR; 4299 ALOGV("start(%d), calling pid %d session %d", 4300 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4301 4302 sp<ThreadBase> thread = mThread.promote(); 4303 if (thread != 0) { 4304 Mutex::Autolock _l(thread->mLock); 4305 track_state state = mState; 4306 // here the track could be either new, or restarted 4307 // in both cases "unstop" the track 4308 if (mState == PAUSED) { 4309 mState = TrackBase::RESUMING; 4310 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4311 } else { 4312 mState = TrackBase::ACTIVE; 4313 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4314 } 4315 4316 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4317 thread->mLock.unlock(); 4318 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4319 thread->mLock.lock(); 4320 4321#ifdef ADD_BATTERY_DATA 4322 // to track the speaker usage 4323 if (status == NO_ERROR) { 4324 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4325 } 4326#endif 4327 } 4328 if (status == NO_ERROR) { 4329 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4330 playbackThread->addTrack_l(this); 4331 } else { 4332 mState = state; 4333 } 4334 } else { 4335 status = BAD_VALUE; 4336 } 4337 return status; 4338} 4339 4340void AudioFlinger::PlaybackThread::Track::stop() 4341{ 4342 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4343 sp<ThreadBase> thread = mThread.promote(); 4344 if (thread != 0) { 4345 Mutex::Autolock _l(thread->mLock); 4346 track_state state = mState; 4347 if (mState > STOPPED) { 4348 mState = STOPPED; 4349 // If the track is not active (PAUSED and buffers full), flush buffers 4350 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4351 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4352 reset(); 4353 } 4354 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 4355 } 4356 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4357 thread->mLock.unlock(); 4358 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4359 thread->mLock.lock(); 4360 4361#ifdef ADD_BATTERY_DATA 4362 // to track the speaker usage 4363 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4364#endif 4365 } 4366 } 4367} 4368 4369void AudioFlinger::PlaybackThread::Track::pause() 4370{ 4371 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4372 sp<ThreadBase> thread = mThread.promote(); 4373 if (thread != 0) { 4374 Mutex::Autolock _l(thread->mLock); 4375 if (mState == ACTIVE || mState == RESUMING) { 4376 mState = PAUSING; 4377 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4378 if (!isOutputTrack()) { 4379 thread->mLock.unlock(); 4380 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4381 thread->mLock.lock(); 4382 4383#ifdef ADD_BATTERY_DATA 4384 // to track the speaker usage 4385 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4386#endif 4387 } 4388 } 4389 } 4390} 4391 4392void AudioFlinger::PlaybackThread::Track::flush() 4393{ 4394 ALOGV("flush(%d)", mName); 4395 sp<ThreadBase> thread = mThread.promote(); 4396 if (thread != 0) { 4397 Mutex::Autolock _l(thread->mLock); 4398 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 4399 return; 4400 } 4401 // No point remaining in PAUSED state after a flush => go to 4402 // STOPPED state 4403 mState = STOPPED; 4404 4405 // do not reset the track if it is still in the process of being stopped or paused. 4406 // this will be done by prepareTracks_l() when the track is stopped. 4407 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4408 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4409 reset(); 4410 } 4411 } 4412} 4413 4414void AudioFlinger::PlaybackThread::Track::reset() 4415{ 4416 // Do not reset twice to avoid discarding data written just after a flush and before 4417 // the audioflinger thread detects the track is stopped. 4418 if (!mResetDone) { 4419 TrackBase::reset(); 4420 // Force underrun condition to avoid false underrun callback until first data is 4421 // written to buffer 4422 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4423 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4424 mFillingUpStatus = FS_FILLING; 4425 mResetDone = true; 4426 mPresentationCompleteFrames = 0; 4427 } 4428} 4429 4430void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4431{ 4432 mMute = muted; 4433} 4434 4435status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4436{ 4437 status_t status = DEAD_OBJECT; 4438 sp<ThreadBase> thread = mThread.promote(); 4439 if (thread != 0) { 4440 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4441 status = playbackThread->attachAuxEffect(this, EffectId); 4442 } 4443 return status; 4444} 4445 4446void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4447{ 4448 mAuxEffectId = EffectId; 4449 mAuxBuffer = buffer; 4450} 4451 4452bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4453 size_t audioHalFrames) 4454{ 4455 // a track is considered presented when the total number of frames written to audio HAL 4456 // corresponds to the number of frames written when presentationComplete() is called for the 4457 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4458 if (mPresentationCompleteFrames == 0) { 4459 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4460 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4461 mPresentationCompleteFrames, audioHalFrames); 4462 } 4463 if (framesWritten >= mPresentationCompleteFrames) { 4464 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4465 mSessionId, framesWritten); 4466 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4467 mPresentationCompleteFrames = 0; 4468 return true; 4469 } 4470 return false; 4471} 4472 4473void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4474{ 4475 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4476 if (mSyncEvents[i]->type() == type) { 4477 mSyncEvents[i]->trigger(); 4478 mSyncEvents.removeAt(i); 4479 i--; 4480 } 4481 } 4482} 4483 4484// implement VolumeBufferProvider interface 4485 4486uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4487{ 4488 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4489 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4490 uint32_t vlr = mCblk->getVolumeLR(); 4491 uint32_t vl = vlr & 0xFFFF; 4492 uint32_t vr = vlr >> 16; 4493 // track volumes come from shared memory, so can't be trusted and must be clamped 4494 if (vl > MAX_GAIN_INT) { 4495 vl = MAX_GAIN_INT; 4496 } 4497 if (vr > MAX_GAIN_INT) { 4498 vr = MAX_GAIN_INT; 4499 } 4500 // now apply the cached master volume and stream type volume; 4501 // this is trusted but lacks any synchronization or barrier so may be stale 4502 float v = mCachedVolume; 4503 vl *= v; 4504 vr *= v; 4505 // re-combine into U4.16 4506 vlr = (vr << 16) | (vl & 0xFFFF); 4507 // FIXME look at mute, pause, and stop flags 4508 return vlr; 4509} 4510 4511// timed audio tracks 4512 4513sp<AudioFlinger::PlaybackThread::TimedTrack> 4514AudioFlinger::PlaybackThread::TimedTrack::create( 4515 PlaybackThread *thread, 4516 const sp<Client>& client, 4517 audio_stream_type_t streamType, 4518 uint32_t sampleRate, 4519 audio_format_t format, 4520 uint32_t channelMask, 4521 int frameCount, 4522 const sp<IMemory>& sharedBuffer, 4523 int sessionId) { 4524 if (!client->reserveTimedTrack()) 4525 return NULL; 4526 4527 return new TimedTrack( 4528 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4529 sharedBuffer, sessionId); 4530} 4531 4532AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4533 PlaybackThread *thread, 4534 const sp<Client>& client, 4535 audio_stream_type_t streamType, 4536 uint32_t sampleRate, 4537 audio_format_t format, 4538 uint32_t channelMask, 4539 int frameCount, 4540 const sp<IMemory>& sharedBuffer, 4541 int sessionId) 4542 : Track(thread, client, streamType, sampleRate, format, channelMask, 4543 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4544 mQueueHeadInFlight(false), 4545 mTrimQueueHeadOnRelease(false), 4546 mFramesPendingInQueue(0), 4547 mTimedSilenceBuffer(NULL), 4548 mTimedSilenceBufferSize(0), 4549 mTimedAudioOutputOnTime(false), 4550 mMediaTimeTransformValid(false) 4551{ 4552 LocalClock lc; 4553 mLocalTimeFreq = lc.getLocalFreq(); 4554 4555 mLocalTimeToSampleTransform.a_zero = 0; 4556 mLocalTimeToSampleTransform.b_zero = 0; 4557 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4558 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4559 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4560 &mLocalTimeToSampleTransform.a_to_b_denom); 4561 4562 mMediaTimeToSampleTransform.a_zero = 0; 4563 mMediaTimeToSampleTransform.b_zero = 0; 4564 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4565 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4566 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4567 &mMediaTimeToSampleTransform.a_to_b_denom); 4568} 4569 4570AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4571 mClient->releaseTimedTrack(); 4572 delete [] mTimedSilenceBuffer; 4573} 4574 4575status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4576 size_t size, sp<IMemory>* buffer) { 4577 4578 Mutex::Autolock _l(mTimedBufferQueueLock); 4579 4580 trimTimedBufferQueue_l(); 4581 4582 // lazily initialize the shared memory heap for timed buffers 4583 if (mTimedMemoryDealer == NULL) { 4584 const int kTimedBufferHeapSize = 512 << 10; 4585 4586 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4587 "AudioFlingerTimed"); 4588 if (mTimedMemoryDealer == NULL) 4589 return NO_MEMORY; 4590 } 4591 4592 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4593 if (newBuffer == NULL) { 4594 newBuffer = mTimedMemoryDealer->allocate(size); 4595 if (newBuffer == NULL) 4596 return NO_MEMORY; 4597 } 4598 4599 *buffer = newBuffer; 4600 return NO_ERROR; 4601} 4602 4603// caller must hold mTimedBufferQueueLock 4604void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4605 int64_t mediaTimeNow; 4606 { 4607 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4608 if (!mMediaTimeTransformValid) 4609 return; 4610 4611 int64_t targetTimeNow; 4612 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4613 ? mCCHelper.getCommonTime(&targetTimeNow) 4614 : mCCHelper.getLocalTime(&targetTimeNow); 4615 4616 if (OK != res) 4617 return; 4618 4619 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4620 &mediaTimeNow)) { 4621 return; 4622 } 4623 } 4624 4625 size_t trimEnd; 4626 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4627 int64_t bufEnd; 4628 4629 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4630 // We have a next buffer. Just use its PTS as the PTS of the frame 4631 // following the last frame in this buffer. If the stream is sparse 4632 // (ie, there are deliberate gaps left in the stream which should be 4633 // filled with silence by the TimedAudioTrack), then this can result 4634 // in one extra buffer being left un-trimmed when it could have 4635 // been. In general, this is not typical, and we would rather 4636 // optimized away the TS calculation below for the more common case 4637 // where PTSes are contiguous. 4638 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4639 } else { 4640 // We have no next buffer. Compute the PTS of the frame following 4641 // the last frame in this buffer by computing the duration of of 4642 // this frame in media time units and adding it to the PTS of the 4643 // buffer. 4644 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4645 / mCblk->frameSize; 4646 4647 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4648 &bufEnd)) { 4649 ALOGE("Failed to convert frame count of %lld to media time" 4650 " duration" " (scale factor %d/%u) in %s", 4651 frameCount, 4652 mMediaTimeToSampleTransform.a_to_b_numer, 4653 mMediaTimeToSampleTransform.a_to_b_denom, 4654 __PRETTY_FUNCTION__); 4655 break; 4656 } 4657 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4658 } 4659 4660 if (bufEnd > mediaTimeNow) 4661 break; 4662 4663 // Is the buffer we want to use in the middle of a mix operation right 4664 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4665 // from the mixer which should be coming back shortly. 4666 if (!trimEnd && mQueueHeadInFlight) { 4667 mTrimQueueHeadOnRelease = true; 4668 } 4669 } 4670 4671 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4672 if (trimStart < trimEnd) { 4673 // Update the bookkeeping for framesReady() 4674 for (size_t i = trimStart; i < trimEnd; ++i) { 4675 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4676 } 4677 4678 // Now actually remove the buffers from the queue. 4679 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4680 } 4681} 4682 4683void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4684 const char* logTag) { 4685 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4686 "%s called (reason \"%s\"), but timed buffer queue has no" 4687 " elements to trim.", __FUNCTION__, logTag); 4688 4689 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4690 mTimedBufferQueue.removeAt(0); 4691} 4692 4693void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4694 const TimedBuffer& buf, 4695 const char* logTag) { 4696 uint32_t bufBytes = buf.buffer()->size(); 4697 uint32_t consumedAlready = buf.position(); 4698 4699 ALOG_ASSERT(consumedAlready <= bufBytes, 4700 "Bad bookkeeping while updating frames pending. Timed buffer is" 4701 " only %u bytes long, but claims to have consumed %u" 4702 " bytes. (update reason: \"%s\")", 4703 bufBytes, consumedAlready, logTag); 4704 4705 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4706 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4707 "Bad bookkeeping while updating frames pending. Should have at" 4708 " least %u queued frames, but we think we have only %u. (update" 4709 " reason: \"%s\")", 4710 bufFrames, mFramesPendingInQueue, logTag); 4711 4712 mFramesPendingInQueue -= bufFrames; 4713} 4714 4715status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4716 const sp<IMemory>& buffer, int64_t pts) { 4717 4718 { 4719 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4720 if (!mMediaTimeTransformValid) 4721 return INVALID_OPERATION; 4722 } 4723 4724 Mutex::Autolock _l(mTimedBufferQueueLock); 4725 4726 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4727 mFramesPendingInQueue += bufFrames; 4728 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4729 4730 return NO_ERROR; 4731} 4732 4733status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4734 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4735 4736 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4737 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4738 target); 4739 4740 if (!(target == TimedAudioTrack::LOCAL_TIME || 4741 target == TimedAudioTrack::COMMON_TIME)) { 4742 return BAD_VALUE; 4743 } 4744 4745 Mutex::Autolock lock(mMediaTimeTransformLock); 4746 mMediaTimeTransform = xform; 4747 mMediaTimeTransformTarget = target; 4748 mMediaTimeTransformValid = true; 4749 4750 return NO_ERROR; 4751} 4752 4753#define min(a, b) ((a) < (b) ? (a) : (b)) 4754 4755// implementation of getNextBuffer for tracks whose buffers have timestamps 4756status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4757 AudioBufferProvider::Buffer* buffer, int64_t pts) 4758{ 4759 if (pts == AudioBufferProvider::kInvalidPTS) { 4760 buffer->raw = 0; 4761 buffer->frameCount = 0; 4762 mTimedAudioOutputOnTime = false; 4763 return INVALID_OPERATION; 4764 } 4765 4766 Mutex::Autolock _l(mTimedBufferQueueLock); 4767 4768 ALOG_ASSERT(!mQueueHeadInFlight, 4769 "getNextBuffer called without releaseBuffer!"); 4770 4771 while (true) { 4772 4773 // if we have no timed buffers, then fail 4774 if (mTimedBufferQueue.isEmpty()) { 4775 buffer->raw = 0; 4776 buffer->frameCount = 0; 4777 return NOT_ENOUGH_DATA; 4778 } 4779 4780 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4781 4782 // calculate the PTS of the head of the timed buffer queue expressed in 4783 // local time 4784 int64_t headLocalPTS; 4785 { 4786 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4787 4788 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4789 4790 if (mMediaTimeTransform.a_to_b_denom == 0) { 4791 // the transform represents a pause, so yield silence 4792 timedYieldSilence_l(buffer->frameCount, buffer); 4793 return NO_ERROR; 4794 } 4795 4796 int64_t transformedPTS; 4797 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4798 &transformedPTS)) { 4799 // the transform failed. this shouldn't happen, but if it does 4800 // then just drop this buffer 4801 ALOGW("timedGetNextBuffer transform failed"); 4802 buffer->raw = 0; 4803 buffer->frameCount = 0; 4804 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4805 return NO_ERROR; 4806 } 4807 4808 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4809 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4810 &headLocalPTS)) { 4811 buffer->raw = 0; 4812 buffer->frameCount = 0; 4813 return INVALID_OPERATION; 4814 } 4815 } else { 4816 headLocalPTS = transformedPTS; 4817 } 4818 } 4819 4820 // adjust the head buffer's PTS to reflect the portion of the head buffer 4821 // that has already been consumed 4822 int64_t effectivePTS = headLocalPTS + 4823 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4824 4825 // Calculate the delta in samples between the head of the input buffer 4826 // queue and the start of the next output buffer that will be written. 4827 // If the transformation fails because of over or underflow, it means 4828 // that the sample's position in the output stream is so far out of 4829 // whack that it should just be dropped. 4830 int64_t sampleDelta; 4831 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4832 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4833 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4834 " mix"); 4835 continue; 4836 } 4837 if (!mLocalTimeToSampleTransform.doForwardTransform( 4838 (effectivePTS - pts) << 32, &sampleDelta)) { 4839 ALOGV("*** too late during sample rate transform: dropped buffer"); 4840 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 4841 continue; 4842 } 4843 4844 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 4845 " sampleDelta=[%d.%08x]", 4846 head.pts(), head.position(), pts, 4847 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 4848 + (sampleDelta >> 32)), 4849 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4850 4851 // if the delta between the ideal placement for the next input sample and 4852 // the current output position is within this threshold, then we will 4853 // concatenate the next input samples to the previous output 4854 const int64_t kSampleContinuityThreshold = 4855 (static_cast<int64_t>(sampleRate()) << 32) / 250; 4856 4857 // if this is the first buffer of audio that we're emitting from this track 4858 // then it should be almost exactly on time. 4859 const int64_t kSampleStartupThreshold = 1LL << 32; 4860 4861 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4862 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4863 // the next input is close enough to being on time, so concatenate it 4864 // with the last output 4865 timedYieldSamples_l(buffer); 4866 4867 ALOGVV("*** on time: head.pos=%d frameCount=%u", 4868 head.position(), buffer->frameCount); 4869 return NO_ERROR; 4870 } 4871 4872 // Looks like our output is not on time. Reset our on timed status. 4873 // Next time we mix samples from our input queue, then should be within 4874 // the StartupThreshold. 4875 mTimedAudioOutputOnTime = false; 4876 if (sampleDelta > 0) { 4877 // the gap between the current output position and the proper start of 4878 // the next input sample is too big, so fill it with silence 4879 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4880 4881 timedYieldSilence_l(framesUntilNextInput, buffer); 4882 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4883 return NO_ERROR; 4884 } else { 4885 // the next input sample is late 4886 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4887 size_t onTimeSamplePosition = 4888 head.position() + lateFrames * mCblk->frameSize; 4889 4890 if (onTimeSamplePosition > head.buffer()->size()) { 4891 // all the remaining samples in the head are too late, so 4892 // drop it and move on 4893 ALOGV("*** too late: dropped buffer"); 4894 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 4895 continue; 4896 } else { 4897 // skip over the late samples 4898 head.setPosition(onTimeSamplePosition); 4899 4900 // yield the available samples 4901 timedYieldSamples_l(buffer); 4902 4903 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4904 return NO_ERROR; 4905 } 4906 } 4907 } 4908} 4909 4910// Yield samples from the timed buffer queue head up to the given output 4911// buffer's capacity. 4912// 4913// Caller must hold mTimedBufferQueueLock 4914void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 4915 AudioBufferProvider::Buffer* buffer) { 4916 4917 const TimedBuffer& head = mTimedBufferQueue[0]; 4918 4919 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4920 head.position()); 4921 4922 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4923 mCblk->frameSize); 4924 size_t framesRequested = buffer->frameCount; 4925 buffer->frameCount = min(framesLeftInHead, framesRequested); 4926 4927 mQueueHeadInFlight = true; 4928 mTimedAudioOutputOnTime = true; 4929} 4930 4931// Yield samples of silence up to the given output buffer's capacity 4932// 4933// Caller must hold mTimedBufferQueueLock 4934void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 4935 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4936 4937 // lazily allocate a buffer filled with silence 4938 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4939 delete [] mTimedSilenceBuffer; 4940 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4941 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4942 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4943 } 4944 4945 buffer->raw = mTimedSilenceBuffer; 4946 size_t framesRequested = buffer->frameCount; 4947 buffer->frameCount = min(numFrames, framesRequested); 4948 4949 mTimedAudioOutputOnTime = false; 4950} 4951 4952// AudioBufferProvider interface 4953void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4954 AudioBufferProvider::Buffer* buffer) { 4955 4956 Mutex::Autolock _l(mTimedBufferQueueLock); 4957 4958 // If the buffer which was just released is part of the buffer at the head 4959 // of the queue, be sure to update the amt of the buffer which has been 4960 // consumed. If the buffer being returned is not part of the head of the 4961 // queue, its either because the buffer is part of the silence buffer, or 4962 // because the head of the timed queue was trimmed after the mixer called 4963 // getNextBuffer but before the mixer called releaseBuffer. 4964 if (buffer->raw == mTimedSilenceBuffer) { 4965 ALOG_ASSERT(!mQueueHeadInFlight, 4966 "Queue head in flight during release of silence buffer!"); 4967 goto done; 4968 } 4969 4970 ALOG_ASSERT(mQueueHeadInFlight, 4971 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 4972 " head in flight."); 4973 4974 if (mTimedBufferQueue.size()) { 4975 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4976 4977 void* start = head.buffer()->pointer(); 4978 void* end = reinterpret_cast<void*>( 4979 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 4980 + head.buffer()->size()); 4981 4982 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 4983 "released buffer not within the head of the timed buffer" 4984 " queue; qHead = [%p, %p], released buffer = %p", 4985 start, end, buffer->raw); 4986 4987 head.setPosition(head.position() + 4988 (buffer->frameCount * mCblk->frameSize)); 4989 mQueueHeadInFlight = false; 4990 4991 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 4992 "Bad bookkeeping during releaseBuffer! Should have at" 4993 " least %u queued frames, but we think we have only %u", 4994 buffer->frameCount, mFramesPendingInQueue); 4995 4996 mFramesPendingInQueue -= buffer->frameCount; 4997 4998 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 4999 || mTrimQueueHeadOnRelease) { 5000 trimTimedBufferQueueHead_l("releaseBuffer"); 5001 mTrimQueueHeadOnRelease = false; 5002 } 5003 } else { 5004 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5005 " buffers in the timed buffer queue"); 5006 } 5007 5008done: 5009 buffer->raw = 0; 5010 buffer->frameCount = 0; 5011} 5012 5013size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5014 Mutex::Autolock _l(mTimedBufferQueueLock); 5015 return mFramesPendingInQueue; 5016} 5017 5018AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5019 : mPTS(0), mPosition(0) {} 5020 5021AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5022 const sp<IMemory>& buffer, int64_t pts) 5023 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5024 5025// ---------------------------------------------------------------------------- 5026 5027// RecordTrack constructor must be called with AudioFlinger::mLock held 5028AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5029 RecordThread *thread, 5030 const sp<Client>& client, 5031 uint32_t sampleRate, 5032 audio_format_t format, 5033 uint32_t channelMask, 5034 int frameCount, 5035 int sessionId) 5036 : TrackBase(thread, client, sampleRate, format, 5037 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5038 mOverflow(false) 5039{ 5040 if (mCblk != NULL) { 5041 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5042 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5043 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5044 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5045 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5046 } else { 5047 mCblk->frameSize = sizeof(int8_t); 5048 } 5049 } 5050} 5051 5052AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5053{ 5054 sp<ThreadBase> thread = mThread.promote(); 5055 if (thread != 0) { 5056 AudioSystem::releaseInput(thread->id()); 5057 } 5058} 5059 5060// AudioBufferProvider interface 5061status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5062{ 5063 audio_track_cblk_t* cblk = this->cblk(); 5064 uint32_t framesAvail; 5065 uint32_t framesReq = buffer->frameCount; 5066 5067 // Check if last stepServer failed, try to step now 5068 if (mStepServerFailed) { 5069 if (!step()) goto getNextBuffer_exit; 5070 ALOGV("stepServer recovered"); 5071 mStepServerFailed = false; 5072 } 5073 5074 framesAvail = cblk->framesAvailable_l(); 5075 5076 if (CC_LIKELY(framesAvail)) { 5077 uint32_t s = cblk->server; 5078 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5079 5080 if (framesReq > framesAvail) { 5081 framesReq = framesAvail; 5082 } 5083 if (framesReq > bufferEnd - s) { 5084 framesReq = bufferEnd - s; 5085 } 5086 5087 buffer->raw = getBuffer(s, framesReq); 5088 if (buffer->raw == NULL) goto getNextBuffer_exit; 5089 5090 buffer->frameCount = framesReq; 5091 return NO_ERROR; 5092 } 5093 5094getNextBuffer_exit: 5095 buffer->raw = NULL; 5096 buffer->frameCount = 0; 5097 return NOT_ENOUGH_DATA; 5098} 5099 5100status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5101 int triggerSession) 5102{ 5103 sp<ThreadBase> thread = mThread.promote(); 5104 if (thread != 0) { 5105 RecordThread *recordThread = (RecordThread *)thread.get(); 5106 return recordThread->start(this, event, triggerSession); 5107 } else { 5108 return BAD_VALUE; 5109 } 5110} 5111 5112void AudioFlinger::RecordThread::RecordTrack::stop() 5113{ 5114 sp<ThreadBase> thread = mThread.promote(); 5115 if (thread != 0) { 5116 RecordThread *recordThread = (RecordThread *)thread.get(); 5117 recordThread->stop(this); 5118 TrackBase::reset(); 5119 // Force overrun condition to avoid false overrun callback until first data is 5120 // read from buffer 5121 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5122 } 5123} 5124 5125void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5126{ 5127 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5128 (mClient == 0) ? getpid_cached : mClient->pid(), 5129 mFormat, 5130 mChannelMask, 5131 mSessionId, 5132 mFrameCount, 5133 mState, 5134 mCblk->sampleRate, 5135 mCblk->server, 5136 mCblk->user); 5137} 5138 5139 5140// ---------------------------------------------------------------------------- 5141 5142AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5143 PlaybackThread *playbackThread, 5144 DuplicatingThread *sourceThread, 5145 uint32_t sampleRate, 5146 audio_format_t format, 5147 uint32_t channelMask, 5148 int frameCount) 5149 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5150 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5151 mActive(false), mSourceThread(sourceThread) 5152{ 5153 5154 if (mCblk != NULL) { 5155 mCblk->flags |= CBLK_DIRECTION_OUT; 5156 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5157 mOutBuffer.frameCount = 0; 5158 playbackThread->mTracks.add(this); 5159 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5160 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5161 mCblk, mBuffer, mCblk->buffers, 5162 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5163 } else { 5164 ALOGW("Error creating output track on thread %p", playbackThread); 5165 } 5166} 5167 5168AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5169{ 5170 clearBufferQueue(); 5171} 5172 5173status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5174 int triggerSession) 5175{ 5176 status_t status = Track::start(event, triggerSession); 5177 if (status != NO_ERROR) { 5178 return status; 5179 } 5180 5181 mActive = true; 5182 mRetryCount = 127; 5183 return status; 5184} 5185 5186void AudioFlinger::PlaybackThread::OutputTrack::stop() 5187{ 5188 Track::stop(); 5189 clearBufferQueue(); 5190 mOutBuffer.frameCount = 0; 5191 mActive = false; 5192} 5193 5194bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5195{ 5196 Buffer *pInBuffer; 5197 Buffer inBuffer; 5198 uint32_t channelCount = mChannelCount; 5199 bool outputBufferFull = false; 5200 inBuffer.frameCount = frames; 5201 inBuffer.i16 = data; 5202 5203 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5204 5205 if (!mActive && frames != 0) { 5206 start(); 5207 sp<ThreadBase> thread = mThread.promote(); 5208 if (thread != 0) { 5209 MixerThread *mixerThread = (MixerThread *)thread.get(); 5210 if (mCblk->frameCount > frames){ 5211 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5212 uint32_t startFrames = (mCblk->frameCount - frames); 5213 pInBuffer = new Buffer; 5214 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5215 pInBuffer->frameCount = startFrames; 5216 pInBuffer->i16 = pInBuffer->mBuffer; 5217 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5218 mBufferQueue.add(pInBuffer); 5219 } else { 5220 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5221 } 5222 } 5223 } 5224 } 5225 5226 while (waitTimeLeftMs) { 5227 // First write pending buffers, then new data 5228 if (mBufferQueue.size()) { 5229 pInBuffer = mBufferQueue.itemAt(0); 5230 } else { 5231 pInBuffer = &inBuffer; 5232 } 5233 5234 if (pInBuffer->frameCount == 0) { 5235 break; 5236 } 5237 5238 if (mOutBuffer.frameCount == 0) { 5239 mOutBuffer.frameCount = pInBuffer->frameCount; 5240 nsecs_t startTime = systemTime(); 5241 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5242 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5243 outputBufferFull = true; 5244 break; 5245 } 5246 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5247 if (waitTimeLeftMs >= waitTimeMs) { 5248 waitTimeLeftMs -= waitTimeMs; 5249 } else { 5250 waitTimeLeftMs = 0; 5251 } 5252 } 5253 5254 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5255 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5256 mCblk->stepUser(outFrames); 5257 pInBuffer->frameCount -= outFrames; 5258 pInBuffer->i16 += outFrames * channelCount; 5259 mOutBuffer.frameCount -= outFrames; 5260 mOutBuffer.i16 += outFrames * channelCount; 5261 5262 if (pInBuffer->frameCount == 0) { 5263 if (mBufferQueue.size()) { 5264 mBufferQueue.removeAt(0); 5265 delete [] pInBuffer->mBuffer; 5266 delete pInBuffer; 5267 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5268 } else { 5269 break; 5270 } 5271 } 5272 } 5273 5274 // If we could not write all frames, allocate a buffer and queue it for next time. 5275 if (inBuffer.frameCount) { 5276 sp<ThreadBase> thread = mThread.promote(); 5277 if (thread != 0 && !thread->standby()) { 5278 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5279 pInBuffer = new Buffer; 5280 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5281 pInBuffer->frameCount = inBuffer.frameCount; 5282 pInBuffer->i16 = pInBuffer->mBuffer; 5283 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5284 mBufferQueue.add(pInBuffer); 5285 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5286 } else { 5287 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5288 } 5289 } 5290 } 5291 5292 // Calling write() with a 0 length buffer, means that no more data will be written: 5293 // If no more buffers are pending, fill output track buffer to make sure it is started 5294 // by output mixer. 5295 if (frames == 0 && mBufferQueue.size() == 0) { 5296 if (mCblk->user < mCblk->frameCount) { 5297 frames = mCblk->frameCount - mCblk->user; 5298 pInBuffer = new Buffer; 5299 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5300 pInBuffer->frameCount = frames; 5301 pInBuffer->i16 = pInBuffer->mBuffer; 5302 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5303 mBufferQueue.add(pInBuffer); 5304 } else if (mActive) { 5305 stop(); 5306 } 5307 } 5308 5309 return outputBufferFull; 5310} 5311 5312status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5313{ 5314 int active; 5315 status_t result; 5316 audio_track_cblk_t* cblk = mCblk; 5317 uint32_t framesReq = buffer->frameCount; 5318 5319// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5320 buffer->frameCount = 0; 5321 5322 uint32_t framesAvail = cblk->framesAvailable(); 5323 5324 5325 if (framesAvail == 0) { 5326 Mutex::Autolock _l(cblk->lock); 5327 goto start_loop_here; 5328 while (framesAvail == 0) { 5329 active = mActive; 5330 if (CC_UNLIKELY(!active)) { 5331 ALOGV("Not active and NO_MORE_BUFFERS"); 5332 return NO_MORE_BUFFERS; 5333 } 5334 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5335 if (result != NO_ERROR) { 5336 return NO_MORE_BUFFERS; 5337 } 5338 // read the server count again 5339 start_loop_here: 5340 framesAvail = cblk->framesAvailable_l(); 5341 } 5342 } 5343 5344// if (framesAvail < framesReq) { 5345// return NO_MORE_BUFFERS; 5346// } 5347 5348 if (framesReq > framesAvail) { 5349 framesReq = framesAvail; 5350 } 5351 5352 uint32_t u = cblk->user; 5353 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5354 5355 if (framesReq > bufferEnd - u) { 5356 framesReq = bufferEnd - u; 5357 } 5358 5359 buffer->frameCount = framesReq; 5360 buffer->raw = (void *)cblk->buffer(u); 5361 return NO_ERROR; 5362} 5363 5364 5365void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5366{ 5367 size_t size = mBufferQueue.size(); 5368 5369 for (size_t i = 0; i < size; i++) { 5370 Buffer *pBuffer = mBufferQueue.itemAt(i); 5371 delete [] pBuffer->mBuffer; 5372 delete pBuffer; 5373 } 5374 mBufferQueue.clear(); 5375} 5376 5377// ---------------------------------------------------------------------------- 5378 5379AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5380 : RefBase(), 5381 mAudioFlinger(audioFlinger), 5382 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5383 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5384 mPid(pid), 5385 mTimedTrackCount(0) 5386{ 5387 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5388} 5389 5390// Client destructor must be called with AudioFlinger::mLock held 5391AudioFlinger::Client::~Client() 5392{ 5393 mAudioFlinger->removeClient_l(mPid); 5394} 5395 5396sp<MemoryDealer> AudioFlinger::Client::heap() const 5397{ 5398 return mMemoryDealer; 5399} 5400 5401// Reserve one of the limited slots for a timed audio track associated 5402// with this client 5403bool AudioFlinger::Client::reserveTimedTrack() 5404{ 5405 const int kMaxTimedTracksPerClient = 4; 5406 5407 Mutex::Autolock _l(mTimedTrackLock); 5408 5409 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5410 ALOGW("can not create timed track - pid %d has exceeded the limit", 5411 mPid); 5412 return false; 5413 } 5414 5415 mTimedTrackCount++; 5416 return true; 5417} 5418 5419// Release a slot for a timed audio track 5420void AudioFlinger::Client::releaseTimedTrack() 5421{ 5422 Mutex::Autolock _l(mTimedTrackLock); 5423 mTimedTrackCount--; 5424} 5425 5426// ---------------------------------------------------------------------------- 5427 5428AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5429 const sp<IAudioFlingerClient>& client, 5430 pid_t pid) 5431 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5432{ 5433} 5434 5435AudioFlinger::NotificationClient::~NotificationClient() 5436{ 5437} 5438 5439void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5440{ 5441 sp<NotificationClient> keep(this); 5442 mAudioFlinger->removeNotificationClient(mPid); 5443} 5444 5445// ---------------------------------------------------------------------------- 5446 5447AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5448 : BnAudioTrack(), 5449 mTrack(track) 5450{ 5451} 5452 5453AudioFlinger::TrackHandle::~TrackHandle() { 5454 // just stop the track on deletion, associated resources 5455 // will be freed from the main thread once all pending buffers have 5456 // been played. Unless it's not in the active track list, in which 5457 // case we free everything now... 5458 mTrack->destroy(); 5459} 5460 5461sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5462 return mTrack->getCblk(); 5463} 5464 5465status_t AudioFlinger::TrackHandle::start() { 5466 return mTrack->start(); 5467} 5468 5469void AudioFlinger::TrackHandle::stop() { 5470 mTrack->stop(); 5471} 5472 5473void AudioFlinger::TrackHandle::flush() { 5474 mTrack->flush(); 5475} 5476 5477void AudioFlinger::TrackHandle::mute(bool e) { 5478 mTrack->mute(e); 5479} 5480 5481void AudioFlinger::TrackHandle::pause() { 5482 mTrack->pause(); 5483} 5484 5485status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5486{ 5487 return mTrack->attachAuxEffect(EffectId); 5488} 5489 5490status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5491 sp<IMemory>* buffer) { 5492 if (!mTrack->isTimedTrack()) 5493 return INVALID_OPERATION; 5494 5495 PlaybackThread::TimedTrack* tt = 5496 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5497 return tt->allocateTimedBuffer(size, buffer); 5498} 5499 5500status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5501 int64_t pts) { 5502 if (!mTrack->isTimedTrack()) 5503 return INVALID_OPERATION; 5504 5505 PlaybackThread::TimedTrack* tt = 5506 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5507 return tt->queueTimedBuffer(buffer, pts); 5508} 5509 5510status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5511 const LinearTransform& xform, int target) { 5512 5513 if (!mTrack->isTimedTrack()) 5514 return INVALID_OPERATION; 5515 5516 PlaybackThread::TimedTrack* tt = 5517 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5518 return tt->setMediaTimeTransform( 5519 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5520} 5521 5522status_t AudioFlinger::TrackHandle::onTransact( 5523 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5524{ 5525 return BnAudioTrack::onTransact(code, data, reply, flags); 5526} 5527 5528// ---------------------------------------------------------------------------- 5529 5530sp<IAudioRecord> AudioFlinger::openRecord( 5531 pid_t pid, 5532 audio_io_handle_t input, 5533 uint32_t sampleRate, 5534 audio_format_t format, 5535 uint32_t channelMask, 5536 int frameCount, 5537 IAudioFlinger::track_flags_t flags, 5538 int *sessionId, 5539 status_t *status) 5540{ 5541 sp<RecordThread::RecordTrack> recordTrack; 5542 sp<RecordHandle> recordHandle; 5543 sp<Client> client; 5544 status_t lStatus; 5545 RecordThread *thread; 5546 size_t inFrameCount; 5547 int lSessionId; 5548 5549 // check calling permissions 5550 if (!recordingAllowed()) { 5551 lStatus = PERMISSION_DENIED; 5552 goto Exit; 5553 } 5554 5555 // add client to list 5556 { // scope for mLock 5557 Mutex::Autolock _l(mLock); 5558 thread = checkRecordThread_l(input); 5559 if (thread == NULL) { 5560 lStatus = BAD_VALUE; 5561 goto Exit; 5562 } 5563 5564 client = registerPid_l(pid); 5565 5566 // If no audio session id is provided, create one here 5567 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5568 lSessionId = *sessionId; 5569 } else { 5570 lSessionId = nextUniqueId(); 5571 if (sessionId != NULL) { 5572 *sessionId = lSessionId; 5573 } 5574 } 5575 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5576 recordTrack = thread->createRecordTrack_l(client, 5577 sampleRate, 5578 format, 5579 channelMask, 5580 frameCount, 5581 lSessionId, 5582 &lStatus); 5583 } 5584 if (lStatus != NO_ERROR) { 5585 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5586 // destructor is called by the TrackBase destructor with mLock held 5587 client.clear(); 5588 recordTrack.clear(); 5589 goto Exit; 5590 } 5591 5592 // return to handle to client 5593 recordHandle = new RecordHandle(recordTrack); 5594 lStatus = NO_ERROR; 5595 5596Exit: 5597 if (status) { 5598 *status = lStatus; 5599 } 5600 return recordHandle; 5601} 5602 5603// ---------------------------------------------------------------------------- 5604 5605AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5606 : BnAudioRecord(), 5607 mRecordTrack(recordTrack) 5608{ 5609} 5610 5611AudioFlinger::RecordHandle::~RecordHandle() { 5612 stop(); 5613} 5614 5615sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5616 return mRecordTrack->getCblk(); 5617} 5618 5619status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5620 ALOGV("RecordHandle::start()"); 5621 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5622} 5623 5624void AudioFlinger::RecordHandle::stop() { 5625 ALOGV("RecordHandle::stop()"); 5626 mRecordTrack->stop(); 5627} 5628 5629status_t AudioFlinger::RecordHandle::onTransact( 5630 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5631{ 5632 return BnAudioRecord::onTransact(code, data, reply, flags); 5633} 5634 5635// ---------------------------------------------------------------------------- 5636 5637AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5638 AudioStreamIn *input, 5639 uint32_t sampleRate, 5640 uint32_t channels, 5641 audio_io_handle_t id, 5642 uint32_t device) : 5643 ThreadBase(audioFlinger, id, device, RECORD), 5644 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5645 // mRsmpInIndex and mInputBytes set by readInputParameters() 5646 mReqChannelCount(popcount(channels)), 5647 mReqSampleRate(sampleRate) 5648 // mBytesRead is only meaningful while active, and so is cleared in start() 5649 // (but might be better to also clear here for dump?) 5650{ 5651 snprintf(mName, kNameLength, "AudioIn_%X", id); 5652 5653 readInputParameters(); 5654} 5655 5656 5657AudioFlinger::RecordThread::~RecordThread() 5658{ 5659 delete[] mRsmpInBuffer; 5660 delete mResampler; 5661 delete[] mRsmpOutBuffer; 5662} 5663 5664void AudioFlinger::RecordThread::onFirstRef() 5665{ 5666 run(mName, PRIORITY_URGENT_AUDIO); 5667} 5668 5669status_t AudioFlinger::RecordThread::readyToRun() 5670{ 5671 status_t status = initCheck(); 5672 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5673 return status; 5674} 5675 5676bool AudioFlinger::RecordThread::threadLoop() 5677{ 5678 AudioBufferProvider::Buffer buffer; 5679 sp<RecordTrack> activeTrack; 5680 Vector< sp<EffectChain> > effectChains; 5681 5682 nsecs_t lastWarning = 0; 5683 5684 acquireWakeLock(); 5685 5686 // start recording 5687 while (!exitPending()) { 5688 5689 processConfigEvents(); 5690 5691 { // scope for mLock 5692 Mutex::Autolock _l(mLock); 5693 checkForNewParameters_l(); 5694 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5695 if (!mStandby) { 5696 mInput->stream->common.standby(&mInput->stream->common); 5697 mStandby = true; 5698 } 5699 5700 if (exitPending()) break; 5701 5702 releaseWakeLock_l(); 5703 ALOGV("RecordThread: loop stopping"); 5704 // go to sleep 5705 mWaitWorkCV.wait(mLock); 5706 ALOGV("RecordThread: loop starting"); 5707 acquireWakeLock_l(); 5708 continue; 5709 } 5710 if (mActiveTrack != 0) { 5711 if (mActiveTrack->mState == TrackBase::PAUSING) { 5712 if (!mStandby) { 5713 mInput->stream->common.standby(&mInput->stream->common); 5714 mStandby = true; 5715 } 5716 mActiveTrack.clear(); 5717 mStartStopCond.broadcast(); 5718 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5719 if (mReqChannelCount != mActiveTrack->channelCount()) { 5720 mActiveTrack.clear(); 5721 mStartStopCond.broadcast(); 5722 } else if (mBytesRead != 0) { 5723 // record start succeeds only if first read from audio input 5724 // succeeds 5725 if (mBytesRead > 0) { 5726 mActiveTrack->mState = TrackBase::ACTIVE; 5727 } else { 5728 mActiveTrack.clear(); 5729 } 5730 mStartStopCond.broadcast(); 5731 } 5732 mStandby = false; 5733 } 5734 } 5735 lockEffectChains_l(effectChains); 5736 } 5737 5738 if (mActiveTrack != 0) { 5739 if (mActiveTrack->mState != TrackBase::ACTIVE && 5740 mActiveTrack->mState != TrackBase::RESUMING) { 5741 unlockEffectChains(effectChains); 5742 usleep(kRecordThreadSleepUs); 5743 continue; 5744 } 5745 for (size_t i = 0; i < effectChains.size(); i ++) { 5746 effectChains[i]->process_l(); 5747 } 5748 5749 buffer.frameCount = mFrameCount; 5750 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5751 size_t framesOut = buffer.frameCount; 5752 if (mResampler == NULL) { 5753 // no resampling 5754 while (framesOut) { 5755 size_t framesIn = mFrameCount - mRsmpInIndex; 5756 if (framesIn) { 5757 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5758 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5759 if (framesIn > framesOut) 5760 framesIn = framesOut; 5761 mRsmpInIndex += framesIn; 5762 framesOut -= framesIn; 5763 if ((int)mChannelCount == mReqChannelCount || 5764 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5765 memcpy(dst, src, framesIn * mFrameSize); 5766 } else { 5767 int16_t *src16 = (int16_t *)src; 5768 int16_t *dst16 = (int16_t *)dst; 5769 if (mChannelCount == 1) { 5770 while (framesIn--) { 5771 *dst16++ = *src16; 5772 *dst16++ = *src16++; 5773 } 5774 } else { 5775 while (framesIn--) { 5776 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5777 src16 += 2; 5778 } 5779 } 5780 } 5781 } 5782 if (framesOut && mFrameCount == mRsmpInIndex) { 5783 if (framesOut == mFrameCount && 5784 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5785 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5786 framesOut = 0; 5787 } else { 5788 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5789 mRsmpInIndex = 0; 5790 } 5791 if (mBytesRead < 0) { 5792 ALOGE("Error reading audio input"); 5793 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5794 // Force input into standby so that it tries to 5795 // recover at next read attempt 5796 mInput->stream->common.standby(&mInput->stream->common); 5797 usleep(kRecordThreadSleepUs); 5798 } 5799 mRsmpInIndex = mFrameCount; 5800 framesOut = 0; 5801 buffer.frameCount = 0; 5802 } 5803 } 5804 } 5805 } else { 5806 // resampling 5807 5808 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5809 // alter output frame count as if we were expecting stereo samples 5810 if (mChannelCount == 1 && mReqChannelCount == 1) { 5811 framesOut >>= 1; 5812 } 5813 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5814 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5815 // are 32 bit aligned which should be always true. 5816 if (mChannelCount == 2 && mReqChannelCount == 1) { 5817 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5818 // the resampler always outputs stereo samples: do post stereo to mono conversion 5819 int16_t *src = (int16_t *)mRsmpOutBuffer; 5820 int16_t *dst = buffer.i16; 5821 while (framesOut--) { 5822 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5823 src += 2; 5824 } 5825 } else { 5826 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5827 } 5828 5829 } 5830 if (mFramestoDrop == 0) { 5831 mActiveTrack->releaseBuffer(&buffer); 5832 } else { 5833 if (mFramestoDrop > 0) { 5834 mFramestoDrop -= buffer.frameCount; 5835 if (mFramestoDrop < 0) { 5836 mFramestoDrop = 0; 5837 } 5838 } 5839 } 5840 mActiveTrack->overflow(); 5841 } 5842 // client isn't retrieving buffers fast enough 5843 else { 5844 if (!mActiveTrack->setOverflow()) { 5845 nsecs_t now = systemTime(); 5846 if ((now - lastWarning) > kWarningThrottleNs) { 5847 ALOGW("RecordThread: buffer overflow"); 5848 lastWarning = now; 5849 } 5850 } 5851 // Release the processor for a while before asking for a new buffer. 5852 // This will give the application more chance to read from the buffer and 5853 // clear the overflow. 5854 usleep(kRecordThreadSleepUs); 5855 } 5856 } 5857 // enable changes in effect chain 5858 unlockEffectChains(effectChains); 5859 effectChains.clear(); 5860 } 5861 5862 if (!mStandby) { 5863 mInput->stream->common.standby(&mInput->stream->common); 5864 } 5865 mActiveTrack.clear(); 5866 5867 mStartStopCond.broadcast(); 5868 5869 releaseWakeLock(); 5870 5871 ALOGV("RecordThread %p exiting", this); 5872 return false; 5873} 5874 5875 5876sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5877 const sp<AudioFlinger::Client>& client, 5878 uint32_t sampleRate, 5879 audio_format_t format, 5880 int channelMask, 5881 int frameCount, 5882 int sessionId, 5883 status_t *status) 5884{ 5885 sp<RecordTrack> track; 5886 status_t lStatus; 5887 5888 lStatus = initCheck(); 5889 if (lStatus != NO_ERROR) { 5890 ALOGE("Audio driver not initialized."); 5891 goto Exit; 5892 } 5893 5894 { // scope for mLock 5895 Mutex::Autolock _l(mLock); 5896 5897 track = new RecordTrack(this, client, sampleRate, 5898 format, channelMask, frameCount, sessionId); 5899 5900 if (track->getCblk() == 0) { 5901 lStatus = NO_MEMORY; 5902 goto Exit; 5903 } 5904 5905 mTrack = track.get(); 5906 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5907 bool suspend = audio_is_bluetooth_sco_device( 5908 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5909 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5910 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5911 } 5912 lStatus = NO_ERROR; 5913 5914Exit: 5915 if (status) { 5916 *status = lStatus; 5917 } 5918 return track; 5919} 5920 5921status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5922 AudioSystem::sync_event_t event, 5923 int triggerSession) 5924{ 5925 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5926 sp<ThreadBase> strongMe = this; 5927 status_t status = NO_ERROR; 5928 5929 if (event == AudioSystem::SYNC_EVENT_NONE) { 5930 mSyncStartEvent.clear(); 5931 mFramestoDrop = 0; 5932 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5933 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5934 triggerSession, 5935 recordTrack->sessionId(), 5936 syncStartEventCallback, 5937 this); 5938 mFramestoDrop = -1; 5939 } 5940 5941 { 5942 AutoMutex lock(mLock); 5943 if (mActiveTrack != 0) { 5944 if (recordTrack != mActiveTrack.get()) { 5945 status = -EBUSY; 5946 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5947 mActiveTrack->mState = TrackBase::ACTIVE; 5948 } 5949 return status; 5950 } 5951 5952 recordTrack->mState = TrackBase::IDLE; 5953 mActiveTrack = recordTrack; 5954 mLock.unlock(); 5955 status_t status = AudioSystem::startInput(mId); 5956 mLock.lock(); 5957 if (status != NO_ERROR) { 5958 mActiveTrack.clear(); 5959 clearSyncStartEvent(); 5960 return status; 5961 } 5962 mRsmpInIndex = mFrameCount; 5963 mBytesRead = 0; 5964 if (mResampler != NULL) { 5965 mResampler->reset(); 5966 } 5967 mActiveTrack->mState = TrackBase::RESUMING; 5968 // signal thread to start 5969 ALOGV("Signal record thread"); 5970 mWaitWorkCV.signal(); 5971 // do not wait for mStartStopCond if exiting 5972 if (exitPending()) { 5973 mActiveTrack.clear(); 5974 status = INVALID_OPERATION; 5975 goto startError; 5976 } 5977 mStartStopCond.wait(mLock); 5978 if (mActiveTrack == 0) { 5979 ALOGV("Record failed to start"); 5980 status = BAD_VALUE; 5981 goto startError; 5982 } 5983 ALOGV("Record started OK"); 5984 return status; 5985 } 5986startError: 5987 AudioSystem::stopInput(mId); 5988 clearSyncStartEvent(); 5989 return status; 5990} 5991 5992void AudioFlinger::RecordThread::clearSyncStartEvent() 5993{ 5994 if (mSyncStartEvent != 0) { 5995 mSyncStartEvent->cancel(); 5996 } 5997 mSyncStartEvent.clear(); 5998} 5999 6000void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6001{ 6002 sp<SyncEvent> strongEvent = event.promote(); 6003 6004 if (strongEvent != 0) { 6005 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6006 me->handleSyncStartEvent(strongEvent); 6007 } 6008} 6009 6010void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6011{ 6012 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 6013 mActiveTrack.get(), 6014 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 6015 event->listenerSession()); 6016 6017 if (mActiveTrack != 0 && 6018 event == mSyncStartEvent) { 6019 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6020 // from audio HAL 6021 mFramestoDrop = mFrameCount * 2; 6022 mSyncStartEvent.clear(); 6023 } 6024} 6025 6026void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6027 ALOGV("RecordThread::stop"); 6028 sp<ThreadBase> strongMe = this; 6029 { 6030 AutoMutex lock(mLock); 6031 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6032 mActiveTrack->mState = TrackBase::PAUSING; 6033 // do not wait for mStartStopCond if exiting 6034 if (exitPending()) { 6035 return; 6036 } 6037 mStartStopCond.wait(mLock); 6038 // if we have been restarted, recordTrack == mActiveTrack.get() here 6039 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6040 mLock.unlock(); 6041 AudioSystem::stopInput(mId); 6042 mLock.lock(); 6043 ALOGV("Record stopped OK"); 6044 } 6045 } 6046 } 6047} 6048 6049bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6050{ 6051 return false; 6052} 6053 6054status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6055{ 6056 if (!isValidSyncEvent(event)) { 6057 return BAD_VALUE; 6058 } 6059 6060 Mutex::Autolock _l(mLock); 6061 6062 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6063 mTrack->setSyncEvent(event); 6064 return NO_ERROR; 6065 } 6066 return NAME_NOT_FOUND; 6067} 6068 6069status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6070{ 6071 const size_t SIZE = 256; 6072 char buffer[SIZE]; 6073 String8 result; 6074 6075 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6076 result.append(buffer); 6077 6078 if (mActiveTrack != 0) { 6079 result.append("Active Track:\n"); 6080 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6081 mActiveTrack->dump(buffer, SIZE); 6082 result.append(buffer); 6083 6084 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6085 result.append(buffer); 6086 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6087 result.append(buffer); 6088 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6089 result.append(buffer); 6090 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6091 result.append(buffer); 6092 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6093 result.append(buffer); 6094 6095 6096 } else { 6097 result.append("No record client\n"); 6098 } 6099 write(fd, result.string(), result.size()); 6100 6101 dumpBase(fd, args); 6102 dumpEffectChains(fd, args); 6103 6104 return NO_ERROR; 6105} 6106 6107// AudioBufferProvider interface 6108status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6109{ 6110 size_t framesReq = buffer->frameCount; 6111 size_t framesReady = mFrameCount - mRsmpInIndex; 6112 int channelCount; 6113 6114 if (framesReady == 0) { 6115 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6116 if (mBytesRead < 0) { 6117 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6118 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6119 // Force input into standby so that it tries to 6120 // recover at next read attempt 6121 mInput->stream->common.standby(&mInput->stream->common); 6122 usleep(kRecordThreadSleepUs); 6123 } 6124 buffer->raw = NULL; 6125 buffer->frameCount = 0; 6126 return NOT_ENOUGH_DATA; 6127 } 6128 mRsmpInIndex = 0; 6129 framesReady = mFrameCount; 6130 } 6131 6132 if (framesReq > framesReady) { 6133 framesReq = framesReady; 6134 } 6135 6136 if (mChannelCount == 1 && mReqChannelCount == 2) { 6137 channelCount = 1; 6138 } else { 6139 channelCount = 2; 6140 } 6141 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6142 buffer->frameCount = framesReq; 6143 return NO_ERROR; 6144} 6145 6146// AudioBufferProvider interface 6147void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6148{ 6149 mRsmpInIndex += buffer->frameCount; 6150 buffer->frameCount = 0; 6151} 6152 6153bool AudioFlinger::RecordThread::checkForNewParameters_l() 6154{ 6155 bool reconfig = false; 6156 6157 while (!mNewParameters.isEmpty()) { 6158 status_t status = NO_ERROR; 6159 String8 keyValuePair = mNewParameters[0]; 6160 AudioParameter param = AudioParameter(keyValuePair); 6161 int value; 6162 audio_format_t reqFormat = mFormat; 6163 int reqSamplingRate = mReqSampleRate; 6164 int reqChannelCount = mReqChannelCount; 6165 6166 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6167 reqSamplingRate = value; 6168 reconfig = true; 6169 } 6170 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6171 reqFormat = (audio_format_t) value; 6172 reconfig = true; 6173 } 6174 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6175 reqChannelCount = popcount(value); 6176 reconfig = true; 6177 } 6178 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6179 // do not accept frame count changes if tracks are open as the track buffer 6180 // size depends on frame count and correct behavior would not be guaranteed 6181 // if frame count is changed after track creation 6182 if (mActiveTrack != 0) { 6183 status = INVALID_OPERATION; 6184 } else { 6185 reconfig = true; 6186 } 6187 } 6188 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6189 // forward device change to effects that have requested to be 6190 // aware of attached audio device. 6191 for (size_t i = 0; i < mEffectChains.size(); i++) { 6192 mEffectChains[i]->setDevice_l(value); 6193 } 6194 // store input device and output device but do not forward output device to audio HAL. 6195 // Note that status is ignored by the caller for output device 6196 // (see AudioFlinger::setParameters() 6197 if (value & AUDIO_DEVICE_OUT_ALL) { 6198 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6199 status = BAD_VALUE; 6200 } else { 6201 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6202 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6203 if (mTrack != NULL) { 6204 bool suspend = audio_is_bluetooth_sco_device( 6205 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6206 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6207 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6208 } 6209 } 6210 mDevice |= (uint32_t)value; 6211 } 6212 if (status == NO_ERROR) { 6213 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6214 if (status == INVALID_OPERATION) { 6215 mInput->stream->common.standby(&mInput->stream->common); 6216 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6217 keyValuePair.string()); 6218 } 6219 if (reconfig) { 6220 if (status == BAD_VALUE && 6221 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6222 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6223 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6224 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6225 (reqChannelCount <= FCC_2)) { 6226 status = NO_ERROR; 6227 } 6228 if (status == NO_ERROR) { 6229 readInputParameters(); 6230 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6231 } 6232 } 6233 } 6234 6235 mNewParameters.removeAt(0); 6236 6237 mParamStatus = status; 6238 mParamCond.signal(); 6239 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6240 // already timed out waiting for the status and will never signal the condition. 6241 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6242 } 6243 return reconfig; 6244} 6245 6246String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6247{ 6248 char *s; 6249 String8 out_s8 = String8(); 6250 6251 Mutex::Autolock _l(mLock); 6252 if (initCheck() != NO_ERROR) { 6253 return out_s8; 6254 } 6255 6256 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6257 out_s8 = String8(s); 6258 free(s); 6259 return out_s8; 6260} 6261 6262void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6263 AudioSystem::OutputDescriptor desc; 6264 void *param2 = NULL; 6265 6266 switch (event) { 6267 case AudioSystem::INPUT_OPENED: 6268 case AudioSystem::INPUT_CONFIG_CHANGED: 6269 desc.channels = mChannelMask; 6270 desc.samplingRate = mSampleRate; 6271 desc.format = mFormat; 6272 desc.frameCount = mFrameCount; 6273 desc.latency = 0; 6274 param2 = &desc; 6275 break; 6276 6277 case AudioSystem::INPUT_CLOSED: 6278 default: 6279 break; 6280 } 6281 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6282} 6283 6284void AudioFlinger::RecordThread::readInputParameters() 6285{ 6286 delete mRsmpInBuffer; 6287 // mRsmpInBuffer is always assigned a new[] below 6288 delete mRsmpOutBuffer; 6289 mRsmpOutBuffer = NULL; 6290 delete mResampler; 6291 mResampler = NULL; 6292 6293 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6294 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6295 mChannelCount = (uint16_t)popcount(mChannelMask); 6296 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6297 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6298 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6299 mFrameCount = mInputBytes / mFrameSize; 6300 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6301 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6302 6303 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6304 { 6305 int channelCount; 6306 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6307 // stereo to mono post process as the resampler always outputs stereo. 6308 if (mChannelCount == 1 && mReqChannelCount == 2) { 6309 channelCount = 1; 6310 } else { 6311 channelCount = 2; 6312 } 6313 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6314 mResampler->setSampleRate(mSampleRate); 6315 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6316 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6317 6318 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6319 if (mChannelCount == 1 && mReqChannelCount == 1) { 6320 mFrameCount >>= 1; 6321 } 6322 6323 } 6324 mRsmpInIndex = mFrameCount; 6325} 6326 6327unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6328{ 6329 Mutex::Autolock _l(mLock); 6330 if (initCheck() != NO_ERROR) { 6331 return 0; 6332 } 6333 6334 return mInput->stream->get_input_frames_lost(mInput->stream); 6335} 6336 6337uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6338{ 6339 Mutex::Autolock _l(mLock); 6340 uint32_t result = 0; 6341 if (getEffectChain_l(sessionId) != 0) { 6342 result = EFFECT_SESSION; 6343 } 6344 6345 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6346 result |= TRACK_SESSION; 6347 } 6348 6349 return result; 6350} 6351 6352AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6353{ 6354 Mutex::Autolock _l(mLock); 6355 return mTrack; 6356} 6357 6358AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6359{ 6360 Mutex::Autolock _l(mLock); 6361 return mInput; 6362} 6363 6364AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6365{ 6366 Mutex::Autolock _l(mLock); 6367 AudioStreamIn *input = mInput; 6368 mInput = NULL; 6369 return input; 6370} 6371 6372// this method must always be called either with ThreadBase mLock held or inside the thread loop 6373audio_stream_t* AudioFlinger::RecordThread::stream() const 6374{ 6375 if (mInput == NULL) { 6376 return NULL; 6377 } 6378 return &mInput->stream->common; 6379} 6380 6381 6382// ---------------------------------------------------------------------------- 6383 6384audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6385{ 6386 if (!settingsAllowed()) { 6387 return 0; 6388 } 6389 Mutex::Autolock _l(mLock); 6390 return loadHwModule_l(name); 6391} 6392 6393// loadHwModule_l() must be called with AudioFlinger::mLock held 6394audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6395{ 6396 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6397 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6398 ALOGW("loadHwModule() module %s already loaded", name); 6399 return mAudioHwDevs.keyAt(i); 6400 } 6401 } 6402 6403 audio_hw_device_t *dev; 6404 6405 int rc = load_audio_interface(name, &dev); 6406 if (rc) { 6407 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6408 return 0; 6409 } 6410 6411 mHardwareStatus = AUDIO_HW_INIT; 6412 rc = dev->init_check(dev); 6413 mHardwareStatus = AUDIO_HW_IDLE; 6414 if (rc) { 6415 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6416 return 0; 6417 } 6418 6419 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6420 (NULL != dev->set_master_volume)) { 6421 AutoMutex lock(mHardwareLock); 6422 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6423 dev->set_master_volume(dev, mMasterVolume); 6424 mHardwareStatus = AUDIO_HW_IDLE; 6425 } 6426 6427 audio_module_handle_t handle = nextUniqueId(); 6428 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6429 6430 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6431 name, dev->common.module->name, dev->common.module->id, handle); 6432 6433 return handle; 6434 6435} 6436 6437audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6438 audio_devices_t *pDevices, 6439 uint32_t *pSamplingRate, 6440 audio_format_t *pFormat, 6441 audio_channel_mask_t *pChannelMask, 6442 uint32_t *pLatencyMs, 6443 audio_output_flags_t flags) 6444{ 6445 status_t status; 6446 PlaybackThread *thread = NULL; 6447 struct audio_config config = { 6448 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6449 channel_mask: pChannelMask ? *pChannelMask : 0, 6450 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6451 }; 6452 audio_stream_out_t *outStream = NULL; 6453 audio_hw_device_t *outHwDev; 6454 6455 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6456 module, 6457 (pDevices != NULL) ? (int)*pDevices : 0, 6458 config.sample_rate, 6459 config.format, 6460 config.channel_mask, 6461 flags); 6462 6463 if (pDevices == NULL || *pDevices == 0) { 6464 return 0; 6465 } 6466 6467 Mutex::Autolock _l(mLock); 6468 6469 outHwDev = findSuitableHwDev_l(module, *pDevices); 6470 if (outHwDev == NULL) 6471 return 0; 6472 6473 audio_io_handle_t id = nextUniqueId(); 6474 6475 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6476 6477 status = outHwDev->open_output_stream(outHwDev, 6478 id, 6479 *pDevices, 6480 (audio_output_flags_t)flags, 6481 &config, 6482 &outStream); 6483 6484 mHardwareStatus = AUDIO_HW_IDLE; 6485 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6486 outStream, 6487 config.sample_rate, 6488 config.format, 6489 config.channel_mask, 6490 status); 6491 6492 if (status == NO_ERROR && outStream != NULL) { 6493 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6494 6495 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6496 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6497 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6498 thread = new DirectOutputThread(this, output, id, *pDevices); 6499 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6500 } else { 6501 thread = new MixerThread(this, output, id, *pDevices); 6502 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6503 } 6504 mPlaybackThreads.add(id, thread); 6505 6506 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6507 if (pFormat != NULL) *pFormat = config.format; 6508 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6509 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6510 6511 // notify client processes of the new output creation 6512 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6513 6514 // the first primary output opened designates the primary hw device 6515 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6516 ALOGI("Using module %d has the primary audio interface", module); 6517 mPrimaryHardwareDev = outHwDev; 6518 6519 AutoMutex lock(mHardwareLock); 6520 mHardwareStatus = AUDIO_HW_SET_MODE; 6521 outHwDev->set_mode(outHwDev, mMode); 6522 6523 // Determine the level of master volume support the primary audio HAL has, 6524 // and set the initial master volume at the same time. 6525 float initialVolume = 1.0; 6526 mMasterVolumeSupportLvl = MVS_NONE; 6527 6528 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6529 if ((NULL != outHwDev->get_master_volume) && 6530 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6531 mMasterVolumeSupportLvl = MVS_FULL; 6532 } else { 6533 mMasterVolumeSupportLvl = MVS_SETONLY; 6534 initialVolume = 1.0; 6535 } 6536 6537 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6538 if ((NULL == outHwDev->set_master_volume) || 6539 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6540 mMasterVolumeSupportLvl = MVS_NONE; 6541 } 6542 // now that we have a primary device, initialize master volume on other devices 6543 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6544 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6545 6546 if ((dev != mPrimaryHardwareDev) && 6547 (NULL != dev->set_master_volume)) { 6548 dev->set_master_volume(dev, initialVolume); 6549 } 6550 } 6551 mHardwareStatus = AUDIO_HW_IDLE; 6552 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6553 ? initialVolume 6554 : 1.0; 6555 mMasterVolume = initialVolume; 6556 } 6557 return id; 6558 } 6559 6560 return 0; 6561} 6562 6563audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6564 audio_io_handle_t output2) 6565{ 6566 Mutex::Autolock _l(mLock); 6567 MixerThread *thread1 = checkMixerThread_l(output1); 6568 MixerThread *thread2 = checkMixerThread_l(output2); 6569 6570 if (thread1 == NULL || thread2 == NULL) { 6571 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6572 return 0; 6573 } 6574 6575 audio_io_handle_t id = nextUniqueId(); 6576 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6577 thread->addOutputTrack(thread2); 6578 mPlaybackThreads.add(id, thread); 6579 // notify client processes of the new output creation 6580 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6581 return id; 6582} 6583 6584status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6585{ 6586 // keep strong reference on the playback thread so that 6587 // it is not destroyed while exit() is executed 6588 sp<PlaybackThread> thread; 6589 { 6590 Mutex::Autolock _l(mLock); 6591 thread = checkPlaybackThread_l(output); 6592 if (thread == NULL) { 6593 return BAD_VALUE; 6594 } 6595 6596 ALOGV("closeOutput() %d", output); 6597 6598 if (thread->type() == ThreadBase::MIXER) { 6599 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6600 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6601 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6602 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6603 } 6604 } 6605 } 6606 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6607 mPlaybackThreads.removeItem(output); 6608 } 6609 thread->exit(); 6610 // The thread entity (active unit of execution) is no longer running here, 6611 // but the ThreadBase container still exists. 6612 6613 if (thread->type() != ThreadBase::DUPLICATING) { 6614 AudioStreamOut *out = thread->clearOutput(); 6615 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6616 // from now on thread->mOutput is NULL 6617 out->hwDev->close_output_stream(out->hwDev, out->stream); 6618 delete out; 6619 } 6620 return NO_ERROR; 6621} 6622 6623status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6624{ 6625 Mutex::Autolock _l(mLock); 6626 PlaybackThread *thread = checkPlaybackThread_l(output); 6627 6628 if (thread == NULL) { 6629 return BAD_VALUE; 6630 } 6631 6632 ALOGV("suspendOutput() %d", output); 6633 thread->suspend(); 6634 6635 return NO_ERROR; 6636} 6637 6638status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6639{ 6640 Mutex::Autolock _l(mLock); 6641 PlaybackThread *thread = checkPlaybackThread_l(output); 6642 6643 if (thread == NULL) { 6644 return BAD_VALUE; 6645 } 6646 6647 ALOGV("restoreOutput() %d", output); 6648 6649 thread->restore(); 6650 6651 return NO_ERROR; 6652} 6653 6654audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6655 audio_devices_t *pDevices, 6656 uint32_t *pSamplingRate, 6657 audio_format_t *pFormat, 6658 uint32_t *pChannelMask) 6659{ 6660 status_t status; 6661 RecordThread *thread = NULL; 6662 struct audio_config config = { 6663 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6664 channel_mask: pChannelMask ? *pChannelMask : 0, 6665 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6666 }; 6667 uint32_t reqSamplingRate = config.sample_rate; 6668 audio_format_t reqFormat = config.format; 6669 audio_channel_mask_t reqChannels = config.channel_mask; 6670 audio_stream_in_t *inStream = NULL; 6671 audio_hw_device_t *inHwDev; 6672 6673 if (pDevices == NULL || *pDevices == 0) { 6674 return 0; 6675 } 6676 6677 Mutex::Autolock _l(mLock); 6678 6679 inHwDev = findSuitableHwDev_l(module, *pDevices); 6680 if (inHwDev == NULL) 6681 return 0; 6682 6683 audio_io_handle_t id = nextUniqueId(); 6684 6685 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6686 &inStream); 6687 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6688 inStream, 6689 config.sample_rate, 6690 config.format, 6691 config.channel_mask, 6692 status); 6693 6694 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6695 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6696 // or stereo to mono conversions on 16 bit PCM inputs. 6697 if (status == BAD_VALUE && 6698 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6699 (config.sample_rate <= 2 * reqSamplingRate) && 6700 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6701 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6702 inStream = NULL; 6703 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6704 } 6705 6706 if (status == NO_ERROR && inStream != NULL) { 6707 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6708 6709 // Start record thread 6710 // RecorThread require both input and output device indication to forward to audio 6711 // pre processing modules 6712 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6713 thread = new RecordThread(this, 6714 input, 6715 reqSamplingRate, 6716 reqChannels, 6717 id, 6718 device); 6719 mRecordThreads.add(id, thread); 6720 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6721 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6722 if (pFormat != NULL) *pFormat = config.format; 6723 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6724 6725 input->stream->common.standby(&input->stream->common); 6726 6727 // notify client processes of the new input creation 6728 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6729 return id; 6730 } 6731 6732 return 0; 6733} 6734 6735status_t AudioFlinger::closeInput(audio_io_handle_t input) 6736{ 6737 // keep strong reference on the record thread so that 6738 // it is not destroyed while exit() is executed 6739 sp<RecordThread> thread; 6740 { 6741 Mutex::Autolock _l(mLock); 6742 thread = checkRecordThread_l(input); 6743 if (thread == NULL) { 6744 return BAD_VALUE; 6745 } 6746 6747 ALOGV("closeInput() %d", input); 6748 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6749 mRecordThreads.removeItem(input); 6750 } 6751 thread->exit(); 6752 // The thread entity (active unit of execution) is no longer running here, 6753 // but the ThreadBase container still exists. 6754 6755 AudioStreamIn *in = thread->clearInput(); 6756 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6757 // from now on thread->mInput is NULL 6758 in->hwDev->close_input_stream(in->hwDev, in->stream); 6759 delete in; 6760 6761 return NO_ERROR; 6762} 6763 6764status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6765{ 6766 Mutex::Autolock _l(mLock); 6767 MixerThread *dstThread = checkMixerThread_l(output); 6768 if (dstThread == NULL) { 6769 ALOGW("setStreamOutput() bad output id %d", output); 6770 return BAD_VALUE; 6771 } 6772 6773 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6774 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6775 6776 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6777 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6778 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6779 MixerThread *srcThread = (MixerThread *)thread; 6780 srcThread->invalidateTracks(stream); 6781 } 6782 } 6783 6784 return NO_ERROR; 6785} 6786 6787 6788int AudioFlinger::newAudioSessionId() 6789{ 6790 return nextUniqueId(); 6791} 6792 6793void AudioFlinger::acquireAudioSessionId(int audioSession) 6794{ 6795 Mutex::Autolock _l(mLock); 6796 pid_t caller = IPCThreadState::self()->getCallingPid(); 6797 ALOGV("acquiring %d from %d", audioSession, caller); 6798 size_t num = mAudioSessionRefs.size(); 6799 for (size_t i = 0; i< num; i++) { 6800 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6801 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6802 ref->mCnt++; 6803 ALOGV(" incremented refcount to %d", ref->mCnt); 6804 return; 6805 } 6806 } 6807 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6808 ALOGV(" added new entry for %d", audioSession); 6809} 6810 6811void AudioFlinger::releaseAudioSessionId(int audioSession) 6812{ 6813 Mutex::Autolock _l(mLock); 6814 pid_t caller = IPCThreadState::self()->getCallingPid(); 6815 ALOGV("releasing %d from %d", audioSession, caller); 6816 size_t num = mAudioSessionRefs.size(); 6817 for (size_t i = 0; i< num; i++) { 6818 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6819 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6820 ref->mCnt--; 6821 ALOGV(" decremented refcount to %d", ref->mCnt); 6822 if (ref->mCnt == 0) { 6823 mAudioSessionRefs.removeAt(i); 6824 delete ref; 6825 purgeStaleEffects_l(); 6826 } 6827 return; 6828 } 6829 } 6830 ALOGW("session id %d not found for pid %d", audioSession, caller); 6831} 6832 6833void AudioFlinger::purgeStaleEffects_l() { 6834 6835 ALOGV("purging stale effects"); 6836 6837 Vector< sp<EffectChain> > chains; 6838 6839 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6840 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6841 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6842 sp<EffectChain> ec = t->mEffectChains[j]; 6843 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6844 chains.push(ec); 6845 } 6846 } 6847 } 6848 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6849 sp<RecordThread> t = mRecordThreads.valueAt(i); 6850 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6851 sp<EffectChain> ec = t->mEffectChains[j]; 6852 chains.push(ec); 6853 } 6854 } 6855 6856 for (size_t i = 0; i < chains.size(); i++) { 6857 sp<EffectChain> ec = chains[i]; 6858 int sessionid = ec->sessionId(); 6859 sp<ThreadBase> t = ec->mThread.promote(); 6860 if (t == 0) { 6861 continue; 6862 } 6863 size_t numsessionrefs = mAudioSessionRefs.size(); 6864 bool found = false; 6865 for (size_t k = 0; k < numsessionrefs; k++) { 6866 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6867 if (ref->mSessionid == sessionid) { 6868 ALOGV(" session %d still exists for %d with %d refs", 6869 sessionid, ref->mPid, ref->mCnt); 6870 found = true; 6871 break; 6872 } 6873 } 6874 if (!found) { 6875 // remove all effects from the chain 6876 while (ec->mEffects.size()) { 6877 sp<EffectModule> effect = ec->mEffects[0]; 6878 effect->unPin(); 6879 Mutex::Autolock _l (t->mLock); 6880 t->removeEffect_l(effect); 6881 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6882 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6883 if (handle != 0) { 6884 handle->mEffect.clear(); 6885 if (handle->mHasControl && handle->mEnabled) { 6886 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6887 } 6888 } 6889 } 6890 AudioSystem::unregisterEffect(effect->id()); 6891 } 6892 } 6893 } 6894 return; 6895} 6896 6897// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6898AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6899{ 6900 return mPlaybackThreads.valueFor(output).get(); 6901} 6902 6903// checkMixerThread_l() must be called with AudioFlinger::mLock held 6904AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6905{ 6906 PlaybackThread *thread = checkPlaybackThread_l(output); 6907 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6908} 6909 6910// checkRecordThread_l() must be called with AudioFlinger::mLock held 6911AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6912{ 6913 return mRecordThreads.valueFor(input).get(); 6914} 6915 6916uint32_t AudioFlinger::nextUniqueId() 6917{ 6918 return android_atomic_inc(&mNextUniqueId); 6919} 6920 6921AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6922{ 6923 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6924 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6925 AudioStreamOut *output = thread->getOutput(); 6926 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6927 return thread; 6928 } 6929 } 6930 return NULL; 6931} 6932 6933uint32_t AudioFlinger::primaryOutputDevice_l() const 6934{ 6935 PlaybackThread *thread = primaryPlaybackThread_l(); 6936 6937 if (thread == NULL) { 6938 return 0; 6939 } 6940 6941 return thread->device(); 6942} 6943 6944sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6945 int triggerSession, 6946 int listenerSession, 6947 sync_event_callback_t callBack, 6948 void *cookie) 6949{ 6950 Mutex::Autolock _l(mLock); 6951 6952 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6953 status_t playStatus = NAME_NOT_FOUND; 6954 status_t recStatus = NAME_NOT_FOUND; 6955 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6956 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6957 if (playStatus == NO_ERROR) { 6958 return event; 6959 } 6960 } 6961 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6962 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6963 if (recStatus == NO_ERROR) { 6964 return event; 6965 } 6966 } 6967 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6968 mPendingSyncEvents.add(event); 6969 } else { 6970 ALOGV("createSyncEvent() invalid event %d", event->type()); 6971 event.clear(); 6972 } 6973 return event; 6974} 6975 6976// ---------------------------------------------------------------------------- 6977// Effect management 6978// ---------------------------------------------------------------------------- 6979 6980 6981status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6982{ 6983 Mutex::Autolock _l(mLock); 6984 return EffectQueryNumberEffects(numEffects); 6985} 6986 6987status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6988{ 6989 Mutex::Autolock _l(mLock); 6990 return EffectQueryEffect(index, descriptor); 6991} 6992 6993status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6994 effect_descriptor_t *descriptor) const 6995{ 6996 Mutex::Autolock _l(mLock); 6997 return EffectGetDescriptor(pUuid, descriptor); 6998} 6999 7000 7001sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7002 effect_descriptor_t *pDesc, 7003 const sp<IEffectClient>& effectClient, 7004 int32_t priority, 7005 audio_io_handle_t io, 7006 int sessionId, 7007 status_t *status, 7008 int *id, 7009 int *enabled) 7010{ 7011 status_t lStatus = NO_ERROR; 7012 sp<EffectHandle> handle; 7013 effect_descriptor_t desc; 7014 7015 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7016 pid, effectClient.get(), priority, sessionId, io); 7017 7018 if (pDesc == NULL) { 7019 lStatus = BAD_VALUE; 7020 goto Exit; 7021 } 7022 7023 // check audio settings permission for global effects 7024 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7025 lStatus = PERMISSION_DENIED; 7026 goto Exit; 7027 } 7028 7029 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7030 // that can only be created by audio policy manager (running in same process) 7031 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7032 lStatus = PERMISSION_DENIED; 7033 goto Exit; 7034 } 7035 7036 if (io == 0) { 7037 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7038 // output must be specified by AudioPolicyManager when using session 7039 // AUDIO_SESSION_OUTPUT_STAGE 7040 lStatus = BAD_VALUE; 7041 goto Exit; 7042 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7043 // if the output returned by getOutputForEffect() is removed before we lock the 7044 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7045 // and we will exit safely 7046 io = AudioSystem::getOutputForEffect(&desc); 7047 } 7048 } 7049 7050 { 7051 Mutex::Autolock _l(mLock); 7052 7053 7054 if (!EffectIsNullUuid(&pDesc->uuid)) { 7055 // if uuid is specified, request effect descriptor 7056 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7057 if (lStatus < 0) { 7058 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7059 goto Exit; 7060 } 7061 } else { 7062 // if uuid is not specified, look for an available implementation 7063 // of the required type in effect factory 7064 if (EffectIsNullUuid(&pDesc->type)) { 7065 ALOGW("createEffect() no effect type"); 7066 lStatus = BAD_VALUE; 7067 goto Exit; 7068 } 7069 uint32_t numEffects = 0; 7070 effect_descriptor_t d; 7071 d.flags = 0; // prevent compiler warning 7072 bool found = false; 7073 7074 lStatus = EffectQueryNumberEffects(&numEffects); 7075 if (lStatus < 0) { 7076 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7077 goto Exit; 7078 } 7079 for (uint32_t i = 0; i < numEffects; i++) { 7080 lStatus = EffectQueryEffect(i, &desc); 7081 if (lStatus < 0) { 7082 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7083 continue; 7084 } 7085 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7086 // If matching type found save effect descriptor. If the session is 7087 // 0 and the effect is not auxiliary, continue enumeration in case 7088 // an auxiliary version of this effect type is available 7089 found = true; 7090 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7091 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7092 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7093 break; 7094 } 7095 } 7096 } 7097 if (!found) { 7098 lStatus = BAD_VALUE; 7099 ALOGW("createEffect() effect not found"); 7100 goto Exit; 7101 } 7102 // For same effect type, chose auxiliary version over insert version if 7103 // connect to output mix (Compliance to OpenSL ES) 7104 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7105 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7106 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7107 } 7108 } 7109 7110 // Do not allow auxiliary effects on a session different from 0 (output mix) 7111 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7112 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7113 lStatus = INVALID_OPERATION; 7114 goto Exit; 7115 } 7116 7117 // check recording permission for visualizer 7118 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7119 !recordingAllowed()) { 7120 lStatus = PERMISSION_DENIED; 7121 goto Exit; 7122 } 7123 7124 // return effect descriptor 7125 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7126 7127 // If output is not specified try to find a matching audio session ID in one of the 7128 // output threads. 7129 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7130 // because of code checking output when entering the function. 7131 // Note: io is never 0 when creating an effect on an input 7132 if (io == 0) { 7133 // look for the thread where the specified audio session is present 7134 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7135 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7136 io = mPlaybackThreads.keyAt(i); 7137 break; 7138 } 7139 } 7140 if (io == 0) { 7141 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7142 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7143 io = mRecordThreads.keyAt(i); 7144 break; 7145 } 7146 } 7147 } 7148 // If no output thread contains the requested session ID, default to 7149 // first output. The effect chain will be moved to the correct output 7150 // thread when a track with the same session ID is created 7151 if (io == 0 && mPlaybackThreads.size()) { 7152 io = mPlaybackThreads.keyAt(0); 7153 } 7154 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7155 } 7156 ThreadBase *thread = checkRecordThread_l(io); 7157 if (thread == NULL) { 7158 thread = checkPlaybackThread_l(io); 7159 if (thread == NULL) { 7160 ALOGE("createEffect() unknown output thread"); 7161 lStatus = BAD_VALUE; 7162 goto Exit; 7163 } 7164 } 7165 7166 sp<Client> client = registerPid_l(pid); 7167 7168 // create effect on selected output thread 7169 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7170 &desc, enabled, &lStatus); 7171 if (handle != 0 && id != NULL) { 7172 *id = handle->id(); 7173 } 7174 } 7175 7176Exit: 7177 if (status != NULL) { 7178 *status = lStatus; 7179 } 7180 return handle; 7181} 7182 7183status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7184 audio_io_handle_t dstOutput) 7185{ 7186 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7187 sessionId, srcOutput, dstOutput); 7188 Mutex::Autolock _l(mLock); 7189 if (srcOutput == dstOutput) { 7190 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7191 return NO_ERROR; 7192 } 7193 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7194 if (srcThread == NULL) { 7195 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7196 return BAD_VALUE; 7197 } 7198 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7199 if (dstThread == NULL) { 7200 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7201 return BAD_VALUE; 7202 } 7203 7204 Mutex::Autolock _dl(dstThread->mLock); 7205 Mutex::Autolock _sl(srcThread->mLock); 7206 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7207 7208 return NO_ERROR; 7209} 7210 7211// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7212status_t AudioFlinger::moveEffectChain_l(int sessionId, 7213 AudioFlinger::PlaybackThread *srcThread, 7214 AudioFlinger::PlaybackThread *dstThread, 7215 bool reRegister) 7216{ 7217 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7218 sessionId, srcThread, dstThread); 7219 7220 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7221 if (chain == 0) { 7222 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7223 sessionId, srcThread); 7224 return INVALID_OPERATION; 7225 } 7226 7227 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7228 // so that a new chain is created with correct parameters when first effect is added. This is 7229 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7230 // removed. 7231 srcThread->removeEffectChain_l(chain); 7232 7233 // transfer all effects one by one so that new effect chain is created on new thread with 7234 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7235 audio_io_handle_t dstOutput = dstThread->id(); 7236 sp<EffectChain> dstChain; 7237 uint32_t strategy = 0; // prevent compiler warning 7238 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7239 while (effect != 0) { 7240 srcThread->removeEffect_l(effect); 7241 dstThread->addEffect_l(effect); 7242 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7243 if (effect->state() == EffectModule::ACTIVE || 7244 effect->state() == EffectModule::STOPPING) { 7245 effect->start(); 7246 } 7247 // if the move request is not received from audio policy manager, the effect must be 7248 // re-registered with the new strategy and output 7249 if (dstChain == 0) { 7250 dstChain = effect->chain().promote(); 7251 if (dstChain == 0) { 7252 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7253 srcThread->addEffect_l(effect); 7254 return NO_INIT; 7255 } 7256 strategy = dstChain->strategy(); 7257 } 7258 if (reRegister) { 7259 AudioSystem::unregisterEffect(effect->id()); 7260 AudioSystem::registerEffect(&effect->desc(), 7261 dstOutput, 7262 strategy, 7263 sessionId, 7264 effect->id()); 7265 } 7266 effect = chain->getEffectFromId_l(0); 7267 } 7268 7269 return NO_ERROR; 7270} 7271 7272 7273// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7274sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7275 const sp<AudioFlinger::Client>& client, 7276 const sp<IEffectClient>& effectClient, 7277 int32_t priority, 7278 int sessionId, 7279 effect_descriptor_t *desc, 7280 int *enabled, 7281 status_t *status 7282 ) 7283{ 7284 sp<EffectModule> effect; 7285 sp<EffectHandle> handle; 7286 status_t lStatus; 7287 sp<EffectChain> chain; 7288 bool chainCreated = false; 7289 bool effectCreated = false; 7290 bool effectRegistered = false; 7291 7292 lStatus = initCheck(); 7293 if (lStatus != NO_ERROR) { 7294 ALOGW("createEffect_l() Audio driver not initialized."); 7295 goto Exit; 7296 } 7297 7298 // Do not allow effects with session ID 0 on direct output or duplicating threads 7299 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7300 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7301 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7302 desc->name, sessionId); 7303 lStatus = BAD_VALUE; 7304 goto Exit; 7305 } 7306 // Only Pre processor effects are allowed on input threads and only on input threads 7307 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7308 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7309 desc->name, desc->flags, mType); 7310 lStatus = BAD_VALUE; 7311 goto Exit; 7312 } 7313 7314 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7315 7316 { // scope for mLock 7317 Mutex::Autolock _l(mLock); 7318 7319 // check for existing effect chain with the requested audio session 7320 chain = getEffectChain_l(sessionId); 7321 if (chain == 0) { 7322 // create a new chain for this session 7323 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7324 chain = new EffectChain(this, sessionId); 7325 addEffectChain_l(chain); 7326 chain->setStrategy(getStrategyForSession_l(sessionId)); 7327 chainCreated = true; 7328 } else { 7329 effect = chain->getEffectFromDesc_l(desc); 7330 } 7331 7332 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7333 7334 if (effect == 0) { 7335 int id = mAudioFlinger->nextUniqueId(); 7336 // Check CPU and memory usage 7337 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7338 if (lStatus != NO_ERROR) { 7339 goto Exit; 7340 } 7341 effectRegistered = true; 7342 // create a new effect module if none present in the chain 7343 effect = new EffectModule(this, chain, desc, id, sessionId); 7344 lStatus = effect->status(); 7345 if (lStatus != NO_ERROR) { 7346 goto Exit; 7347 } 7348 lStatus = chain->addEffect_l(effect); 7349 if (lStatus != NO_ERROR) { 7350 goto Exit; 7351 } 7352 effectCreated = true; 7353 7354 effect->setDevice(mDevice); 7355 effect->setMode(mAudioFlinger->getMode()); 7356 } 7357 // create effect handle and connect it to effect module 7358 handle = new EffectHandle(effect, client, effectClient, priority); 7359 lStatus = effect->addHandle(handle); 7360 if (enabled != NULL) { 7361 *enabled = (int)effect->isEnabled(); 7362 } 7363 } 7364 7365Exit: 7366 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7367 Mutex::Autolock _l(mLock); 7368 if (effectCreated) { 7369 chain->removeEffect_l(effect); 7370 } 7371 if (effectRegistered) { 7372 AudioSystem::unregisterEffect(effect->id()); 7373 } 7374 if (chainCreated) { 7375 removeEffectChain_l(chain); 7376 } 7377 handle.clear(); 7378 } 7379 7380 if (status != NULL) { 7381 *status = lStatus; 7382 } 7383 return handle; 7384} 7385 7386sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7387{ 7388 sp<EffectChain> chain = getEffectChain_l(sessionId); 7389 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7390} 7391 7392// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7393// PlaybackThread::mLock held 7394status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7395{ 7396 // check for existing effect chain with the requested audio session 7397 int sessionId = effect->sessionId(); 7398 sp<EffectChain> chain = getEffectChain_l(sessionId); 7399 bool chainCreated = false; 7400 7401 if (chain == 0) { 7402 // create a new chain for this session 7403 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7404 chain = new EffectChain(this, sessionId); 7405 addEffectChain_l(chain); 7406 chain->setStrategy(getStrategyForSession_l(sessionId)); 7407 chainCreated = true; 7408 } 7409 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7410 7411 if (chain->getEffectFromId_l(effect->id()) != 0) { 7412 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7413 this, effect->desc().name, chain.get()); 7414 return BAD_VALUE; 7415 } 7416 7417 status_t status = chain->addEffect_l(effect); 7418 if (status != NO_ERROR) { 7419 if (chainCreated) { 7420 removeEffectChain_l(chain); 7421 } 7422 return status; 7423 } 7424 7425 effect->setDevice(mDevice); 7426 effect->setMode(mAudioFlinger->getMode()); 7427 return NO_ERROR; 7428} 7429 7430void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7431 7432 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7433 effect_descriptor_t desc = effect->desc(); 7434 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7435 detachAuxEffect_l(effect->id()); 7436 } 7437 7438 sp<EffectChain> chain = effect->chain().promote(); 7439 if (chain != 0) { 7440 // remove effect chain if removing last effect 7441 if (chain->removeEffect_l(effect) == 0) { 7442 removeEffectChain_l(chain); 7443 } 7444 } else { 7445 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7446 } 7447} 7448 7449void AudioFlinger::ThreadBase::lockEffectChains_l( 7450 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7451{ 7452 effectChains = mEffectChains; 7453 for (size_t i = 0; i < mEffectChains.size(); i++) { 7454 mEffectChains[i]->lock(); 7455 } 7456} 7457 7458void AudioFlinger::ThreadBase::unlockEffectChains( 7459 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7460{ 7461 for (size_t i = 0; i < effectChains.size(); i++) { 7462 effectChains[i]->unlock(); 7463 } 7464} 7465 7466sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7467{ 7468 Mutex::Autolock _l(mLock); 7469 return getEffectChain_l(sessionId); 7470} 7471 7472sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7473{ 7474 size_t size = mEffectChains.size(); 7475 for (size_t i = 0; i < size; i++) { 7476 if (mEffectChains[i]->sessionId() == sessionId) { 7477 return mEffectChains[i]; 7478 } 7479 } 7480 return 0; 7481} 7482 7483void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7484{ 7485 Mutex::Autolock _l(mLock); 7486 size_t size = mEffectChains.size(); 7487 for (size_t i = 0; i < size; i++) { 7488 mEffectChains[i]->setMode_l(mode); 7489 } 7490} 7491 7492void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7493 const wp<EffectHandle>& handle, 7494 bool unpinIfLast) { 7495 7496 Mutex::Autolock _l(mLock); 7497 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7498 // delete the effect module if removing last handle on it 7499 if (effect->removeHandle(handle) == 0) { 7500 if (!effect->isPinned() || unpinIfLast) { 7501 removeEffect_l(effect); 7502 AudioSystem::unregisterEffect(effect->id()); 7503 } 7504 } 7505} 7506 7507status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7508{ 7509 int session = chain->sessionId(); 7510 int16_t *buffer = mMixBuffer; 7511 bool ownsBuffer = false; 7512 7513 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7514 if (session > 0) { 7515 // Only one effect chain can be present in direct output thread and it uses 7516 // the mix buffer as input 7517 if (mType != DIRECT) { 7518 size_t numSamples = mNormalFrameCount * mChannelCount; 7519 buffer = new int16_t[numSamples]; 7520 memset(buffer, 0, numSamples * sizeof(int16_t)); 7521 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7522 ownsBuffer = true; 7523 } 7524 7525 // Attach all tracks with same session ID to this chain. 7526 for (size_t i = 0; i < mTracks.size(); ++i) { 7527 sp<Track> track = mTracks[i]; 7528 if (session == track->sessionId()) { 7529 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7530 track->setMainBuffer(buffer); 7531 chain->incTrackCnt(); 7532 } 7533 } 7534 7535 // indicate all active tracks in the chain 7536 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7537 sp<Track> track = mActiveTracks[i].promote(); 7538 if (track == 0) continue; 7539 if (session == track->sessionId()) { 7540 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7541 chain->incActiveTrackCnt(); 7542 } 7543 } 7544 } 7545 7546 chain->setInBuffer(buffer, ownsBuffer); 7547 chain->setOutBuffer(mMixBuffer); 7548 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7549 // chains list in order to be processed last as it contains output stage effects 7550 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7551 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7552 // after track specific effects and before output stage 7553 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7554 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7555 // Effect chain for other sessions are inserted at beginning of effect 7556 // chains list to be processed before output mix effects. Relative order between other 7557 // sessions is not important 7558 size_t size = mEffectChains.size(); 7559 size_t i = 0; 7560 for (i = 0; i < size; i++) { 7561 if (mEffectChains[i]->sessionId() < session) break; 7562 } 7563 mEffectChains.insertAt(chain, i); 7564 checkSuspendOnAddEffectChain_l(chain); 7565 7566 return NO_ERROR; 7567} 7568 7569size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7570{ 7571 int session = chain->sessionId(); 7572 7573 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7574 7575 for (size_t i = 0; i < mEffectChains.size(); i++) { 7576 if (chain == mEffectChains[i]) { 7577 mEffectChains.removeAt(i); 7578 // detach all active tracks from the chain 7579 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7580 sp<Track> track = mActiveTracks[i].promote(); 7581 if (track == 0) continue; 7582 if (session == track->sessionId()) { 7583 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7584 chain.get(), session); 7585 chain->decActiveTrackCnt(); 7586 } 7587 } 7588 7589 // detach all tracks with same session ID from this chain 7590 for (size_t i = 0; i < mTracks.size(); ++i) { 7591 sp<Track> track = mTracks[i]; 7592 if (session == track->sessionId()) { 7593 track->setMainBuffer(mMixBuffer); 7594 chain->decTrackCnt(); 7595 } 7596 } 7597 break; 7598 } 7599 } 7600 return mEffectChains.size(); 7601} 7602 7603status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7604 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7605{ 7606 Mutex::Autolock _l(mLock); 7607 return attachAuxEffect_l(track, EffectId); 7608} 7609 7610status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7611 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7612{ 7613 status_t status = NO_ERROR; 7614 7615 if (EffectId == 0) { 7616 track->setAuxBuffer(0, NULL); 7617 } else { 7618 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7619 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7620 if (effect != 0) { 7621 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7622 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7623 } else { 7624 status = INVALID_OPERATION; 7625 } 7626 } else { 7627 status = BAD_VALUE; 7628 } 7629 } 7630 return status; 7631} 7632 7633void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7634{ 7635 for (size_t i = 0; i < mTracks.size(); ++i) { 7636 sp<Track> track = mTracks[i]; 7637 if (track->auxEffectId() == effectId) { 7638 attachAuxEffect_l(track, 0); 7639 } 7640 } 7641} 7642 7643status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7644{ 7645 // only one chain per input thread 7646 if (mEffectChains.size() != 0) { 7647 return INVALID_OPERATION; 7648 } 7649 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7650 7651 chain->setInBuffer(NULL); 7652 chain->setOutBuffer(NULL); 7653 7654 checkSuspendOnAddEffectChain_l(chain); 7655 7656 mEffectChains.add(chain); 7657 7658 return NO_ERROR; 7659} 7660 7661size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7662{ 7663 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7664 ALOGW_IF(mEffectChains.size() != 1, 7665 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7666 chain.get(), mEffectChains.size(), this); 7667 if (mEffectChains.size() == 1) { 7668 mEffectChains.removeAt(0); 7669 } 7670 return 0; 7671} 7672 7673// ---------------------------------------------------------------------------- 7674// EffectModule implementation 7675// ---------------------------------------------------------------------------- 7676 7677#undef LOG_TAG 7678#define LOG_TAG "AudioFlinger::EffectModule" 7679 7680AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7681 const wp<AudioFlinger::EffectChain>& chain, 7682 effect_descriptor_t *desc, 7683 int id, 7684 int sessionId) 7685 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7686 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7687{ 7688 ALOGV("Constructor %p", this); 7689 int lStatus; 7690 if (thread == NULL) { 7691 return; 7692 } 7693 7694 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7695 7696 // create effect engine from effect factory 7697 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7698 7699 if (mStatus != NO_ERROR) { 7700 return; 7701 } 7702 lStatus = init(); 7703 if (lStatus < 0) { 7704 mStatus = lStatus; 7705 goto Error; 7706 } 7707 7708 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7709 mPinned = true; 7710 } 7711 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7712 return; 7713Error: 7714 EffectRelease(mEffectInterface); 7715 mEffectInterface = NULL; 7716 ALOGV("Constructor Error %d", mStatus); 7717} 7718 7719AudioFlinger::EffectModule::~EffectModule() 7720{ 7721 ALOGV("Destructor %p", this); 7722 if (mEffectInterface != NULL) { 7723 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7724 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7725 sp<ThreadBase> thread = mThread.promote(); 7726 if (thread != 0) { 7727 audio_stream_t *stream = thread->stream(); 7728 if (stream != NULL) { 7729 stream->remove_audio_effect(stream, mEffectInterface); 7730 } 7731 } 7732 } 7733 // release effect engine 7734 EffectRelease(mEffectInterface); 7735 } 7736} 7737 7738status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7739{ 7740 status_t status; 7741 7742 Mutex::Autolock _l(mLock); 7743 int priority = handle->priority(); 7744 size_t size = mHandles.size(); 7745 sp<EffectHandle> h; 7746 size_t i; 7747 for (i = 0; i < size; i++) { 7748 h = mHandles[i].promote(); 7749 if (h == 0) continue; 7750 if (h->priority() <= priority) break; 7751 } 7752 // if inserted in first place, move effect control from previous owner to this handle 7753 if (i == 0) { 7754 bool enabled = false; 7755 if (h != 0) { 7756 enabled = h->enabled(); 7757 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7758 } 7759 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7760 status = NO_ERROR; 7761 } else { 7762 status = ALREADY_EXISTS; 7763 } 7764 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7765 mHandles.insertAt(handle, i); 7766 return status; 7767} 7768 7769size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7770{ 7771 Mutex::Autolock _l(mLock); 7772 size_t size = mHandles.size(); 7773 size_t i; 7774 for (i = 0; i < size; i++) { 7775 if (mHandles[i] == handle) break; 7776 } 7777 if (i == size) { 7778 return size; 7779 } 7780 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7781 7782 bool enabled = false; 7783 EffectHandle *hdl = handle.unsafe_get(); 7784 if (hdl != NULL) { 7785 ALOGV("removeHandle() unsafe_get OK"); 7786 enabled = hdl->enabled(); 7787 } 7788 mHandles.removeAt(i); 7789 size = mHandles.size(); 7790 // if removed from first place, move effect control from this handle to next in line 7791 if (i == 0 && size != 0) { 7792 sp<EffectHandle> h = mHandles[0].promote(); 7793 if (h != 0) { 7794 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7795 } 7796 } 7797 7798 // Prevent calls to process() and other functions on effect interface from now on. 7799 // The effect engine will be released by the destructor when the last strong reference on 7800 // this object is released which can happen after next process is called. 7801 if (size == 0 && !mPinned) { 7802 mState = DESTROYED; 7803 } 7804 7805 return size; 7806} 7807 7808sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7809{ 7810 Mutex::Autolock _l(mLock); 7811 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7812} 7813 7814void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7815{ 7816 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7817 // keep a strong reference on this EffectModule to avoid calling the 7818 // destructor before we exit 7819 sp<EffectModule> keep(this); 7820 { 7821 sp<ThreadBase> thread = mThread.promote(); 7822 if (thread != 0) { 7823 thread->disconnectEffect(keep, handle, unpinIfLast); 7824 } 7825 } 7826} 7827 7828void AudioFlinger::EffectModule::updateState() { 7829 Mutex::Autolock _l(mLock); 7830 7831 switch (mState) { 7832 case RESTART: 7833 reset_l(); 7834 // FALL THROUGH 7835 7836 case STARTING: 7837 // clear auxiliary effect input buffer for next accumulation 7838 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7839 memset(mConfig.inputCfg.buffer.raw, 7840 0, 7841 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7842 } 7843 start_l(); 7844 mState = ACTIVE; 7845 break; 7846 case STOPPING: 7847 stop_l(); 7848 mDisableWaitCnt = mMaxDisableWaitCnt; 7849 mState = STOPPED; 7850 break; 7851 case STOPPED: 7852 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7853 // turn off sequence. 7854 if (--mDisableWaitCnt == 0) { 7855 reset_l(); 7856 mState = IDLE; 7857 } 7858 break; 7859 default: //IDLE , ACTIVE, DESTROYED 7860 break; 7861 } 7862} 7863 7864void AudioFlinger::EffectModule::process() 7865{ 7866 Mutex::Autolock _l(mLock); 7867 7868 if (mState == DESTROYED || mEffectInterface == NULL || 7869 mConfig.inputCfg.buffer.raw == NULL || 7870 mConfig.outputCfg.buffer.raw == NULL) { 7871 return; 7872 } 7873 7874 if (isProcessEnabled()) { 7875 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7876 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7877 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7878 mConfig.inputCfg.buffer.s32, 7879 mConfig.inputCfg.buffer.frameCount/2); 7880 } 7881 7882 // do the actual processing in the effect engine 7883 int ret = (*mEffectInterface)->process(mEffectInterface, 7884 &mConfig.inputCfg.buffer, 7885 &mConfig.outputCfg.buffer); 7886 7887 // force transition to IDLE state when engine is ready 7888 if (mState == STOPPED && ret == -ENODATA) { 7889 mDisableWaitCnt = 1; 7890 } 7891 7892 // clear auxiliary effect input buffer for next accumulation 7893 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7894 memset(mConfig.inputCfg.buffer.raw, 0, 7895 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7896 } 7897 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7898 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7899 // If an insert effect is idle and input buffer is different from output buffer, 7900 // accumulate input onto output 7901 sp<EffectChain> chain = mChain.promote(); 7902 if (chain != 0 && chain->activeTrackCnt() != 0) { 7903 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7904 int16_t *in = mConfig.inputCfg.buffer.s16; 7905 int16_t *out = mConfig.outputCfg.buffer.s16; 7906 for (size_t i = 0; i < frameCnt; i++) { 7907 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7908 } 7909 } 7910 } 7911} 7912 7913void AudioFlinger::EffectModule::reset_l() 7914{ 7915 if (mEffectInterface == NULL) { 7916 return; 7917 } 7918 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7919} 7920 7921status_t AudioFlinger::EffectModule::configure() 7922{ 7923 uint32_t channels; 7924 if (mEffectInterface == NULL) { 7925 return NO_INIT; 7926 } 7927 7928 sp<ThreadBase> thread = mThread.promote(); 7929 if (thread == 0) { 7930 return DEAD_OBJECT; 7931 } 7932 7933 // TODO: handle configuration of effects replacing track process 7934 if (thread->channelCount() == 1) { 7935 channels = AUDIO_CHANNEL_OUT_MONO; 7936 } else { 7937 channels = AUDIO_CHANNEL_OUT_STEREO; 7938 } 7939 7940 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7941 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7942 } else { 7943 mConfig.inputCfg.channels = channels; 7944 } 7945 mConfig.outputCfg.channels = channels; 7946 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7947 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7948 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7949 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7950 mConfig.inputCfg.bufferProvider.cookie = NULL; 7951 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7952 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7953 mConfig.outputCfg.bufferProvider.cookie = NULL; 7954 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7955 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7956 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7957 // Insert effect: 7958 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7959 // always overwrites output buffer: input buffer == output buffer 7960 // - in other sessions: 7961 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7962 // other effect: overwrites output buffer: input buffer == output buffer 7963 // Auxiliary effect: 7964 // accumulates in output buffer: input buffer != output buffer 7965 // Therefore: accumulate <=> input buffer != output buffer 7966 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7967 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7968 } else { 7969 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7970 } 7971 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7972 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7973 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7974 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7975 7976 ALOGV("configure() %p thread %p buffer %p framecount %d", 7977 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7978 7979 status_t cmdStatus; 7980 uint32_t size = sizeof(int); 7981 status_t status = (*mEffectInterface)->command(mEffectInterface, 7982 EFFECT_CMD_SET_CONFIG, 7983 sizeof(effect_config_t), 7984 &mConfig, 7985 &size, 7986 &cmdStatus); 7987 if (status == 0) { 7988 status = cmdStatus; 7989 } 7990 7991 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7992 (1000 * mConfig.outputCfg.buffer.frameCount); 7993 7994 return status; 7995} 7996 7997status_t AudioFlinger::EffectModule::init() 7998{ 7999 Mutex::Autolock _l(mLock); 8000 if (mEffectInterface == NULL) { 8001 return NO_INIT; 8002 } 8003 status_t cmdStatus; 8004 uint32_t size = sizeof(status_t); 8005 status_t status = (*mEffectInterface)->command(mEffectInterface, 8006 EFFECT_CMD_INIT, 8007 0, 8008 NULL, 8009 &size, 8010 &cmdStatus); 8011 if (status == 0) { 8012 status = cmdStatus; 8013 } 8014 return status; 8015} 8016 8017status_t AudioFlinger::EffectModule::start() 8018{ 8019 Mutex::Autolock _l(mLock); 8020 return start_l(); 8021} 8022 8023status_t AudioFlinger::EffectModule::start_l() 8024{ 8025 if (mEffectInterface == NULL) { 8026 return NO_INIT; 8027 } 8028 status_t cmdStatus; 8029 uint32_t size = sizeof(status_t); 8030 status_t status = (*mEffectInterface)->command(mEffectInterface, 8031 EFFECT_CMD_ENABLE, 8032 0, 8033 NULL, 8034 &size, 8035 &cmdStatus); 8036 if (status == 0) { 8037 status = cmdStatus; 8038 } 8039 if (status == 0 && 8040 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8041 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8042 sp<ThreadBase> thread = mThread.promote(); 8043 if (thread != 0) { 8044 audio_stream_t *stream = thread->stream(); 8045 if (stream != NULL) { 8046 stream->add_audio_effect(stream, mEffectInterface); 8047 } 8048 } 8049 } 8050 return status; 8051} 8052 8053status_t AudioFlinger::EffectModule::stop() 8054{ 8055 Mutex::Autolock _l(mLock); 8056 return stop_l(); 8057} 8058 8059status_t AudioFlinger::EffectModule::stop_l() 8060{ 8061 if (mEffectInterface == NULL) { 8062 return NO_INIT; 8063 } 8064 status_t cmdStatus; 8065 uint32_t size = sizeof(status_t); 8066 status_t status = (*mEffectInterface)->command(mEffectInterface, 8067 EFFECT_CMD_DISABLE, 8068 0, 8069 NULL, 8070 &size, 8071 &cmdStatus); 8072 if (status == 0) { 8073 status = cmdStatus; 8074 } 8075 if (status == 0 && 8076 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8077 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8078 sp<ThreadBase> thread = mThread.promote(); 8079 if (thread != 0) { 8080 audio_stream_t *stream = thread->stream(); 8081 if (stream != NULL) { 8082 stream->remove_audio_effect(stream, mEffectInterface); 8083 } 8084 } 8085 } 8086 return status; 8087} 8088 8089status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8090 uint32_t cmdSize, 8091 void *pCmdData, 8092 uint32_t *replySize, 8093 void *pReplyData) 8094{ 8095 Mutex::Autolock _l(mLock); 8096// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8097 8098 if (mState == DESTROYED || mEffectInterface == NULL) { 8099 return NO_INIT; 8100 } 8101 status_t status = (*mEffectInterface)->command(mEffectInterface, 8102 cmdCode, 8103 cmdSize, 8104 pCmdData, 8105 replySize, 8106 pReplyData); 8107 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8108 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8109 for (size_t i = 1; i < mHandles.size(); i++) { 8110 sp<EffectHandle> h = mHandles[i].promote(); 8111 if (h != 0) { 8112 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8113 } 8114 } 8115 } 8116 return status; 8117} 8118 8119status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8120{ 8121 8122 Mutex::Autolock _l(mLock); 8123 ALOGV("setEnabled %p enabled %d", this, enabled); 8124 8125 if (enabled != isEnabled()) { 8126 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8127 if (enabled && status != NO_ERROR) { 8128 return status; 8129 } 8130 8131 switch (mState) { 8132 // going from disabled to enabled 8133 case IDLE: 8134 mState = STARTING; 8135 break; 8136 case STOPPED: 8137 mState = RESTART; 8138 break; 8139 case STOPPING: 8140 mState = ACTIVE; 8141 break; 8142 8143 // going from enabled to disabled 8144 case RESTART: 8145 mState = STOPPED; 8146 break; 8147 case STARTING: 8148 mState = IDLE; 8149 break; 8150 case ACTIVE: 8151 mState = STOPPING; 8152 break; 8153 case DESTROYED: 8154 return NO_ERROR; // simply ignore as we are being destroyed 8155 } 8156 for (size_t i = 1; i < mHandles.size(); i++) { 8157 sp<EffectHandle> h = mHandles[i].promote(); 8158 if (h != 0) { 8159 h->setEnabled(enabled); 8160 } 8161 } 8162 } 8163 return NO_ERROR; 8164} 8165 8166bool AudioFlinger::EffectModule::isEnabled() const 8167{ 8168 switch (mState) { 8169 case RESTART: 8170 case STARTING: 8171 case ACTIVE: 8172 return true; 8173 case IDLE: 8174 case STOPPING: 8175 case STOPPED: 8176 case DESTROYED: 8177 default: 8178 return false; 8179 } 8180} 8181 8182bool AudioFlinger::EffectModule::isProcessEnabled() const 8183{ 8184 switch (mState) { 8185 case RESTART: 8186 case ACTIVE: 8187 case STOPPING: 8188 case STOPPED: 8189 return true; 8190 case IDLE: 8191 case STARTING: 8192 case DESTROYED: 8193 default: 8194 return false; 8195 } 8196} 8197 8198status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8199{ 8200 Mutex::Autolock _l(mLock); 8201 status_t status = NO_ERROR; 8202 8203 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8204 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8205 if (isProcessEnabled() && 8206 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8207 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8208 status_t cmdStatus; 8209 uint32_t volume[2]; 8210 uint32_t *pVolume = NULL; 8211 uint32_t size = sizeof(volume); 8212 volume[0] = *left; 8213 volume[1] = *right; 8214 if (controller) { 8215 pVolume = volume; 8216 } 8217 status = (*mEffectInterface)->command(mEffectInterface, 8218 EFFECT_CMD_SET_VOLUME, 8219 size, 8220 volume, 8221 &size, 8222 pVolume); 8223 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8224 *left = volume[0]; 8225 *right = volume[1]; 8226 } 8227 } 8228 return status; 8229} 8230 8231status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8232{ 8233 Mutex::Autolock _l(mLock); 8234 status_t status = NO_ERROR; 8235 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8236 // audio pre processing modules on RecordThread can receive both output and 8237 // input device indication in the same call 8238 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8239 if (dev) { 8240 status_t cmdStatus; 8241 uint32_t size = sizeof(status_t); 8242 8243 status = (*mEffectInterface)->command(mEffectInterface, 8244 EFFECT_CMD_SET_DEVICE, 8245 sizeof(uint32_t), 8246 &dev, 8247 &size, 8248 &cmdStatus); 8249 if (status == NO_ERROR) { 8250 status = cmdStatus; 8251 } 8252 } 8253 dev = device & AUDIO_DEVICE_IN_ALL; 8254 if (dev) { 8255 status_t cmdStatus; 8256 uint32_t size = sizeof(status_t); 8257 8258 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8259 EFFECT_CMD_SET_INPUT_DEVICE, 8260 sizeof(uint32_t), 8261 &dev, 8262 &size, 8263 &cmdStatus); 8264 if (status2 == NO_ERROR) { 8265 status2 = cmdStatus; 8266 } 8267 if (status == NO_ERROR) { 8268 status = status2; 8269 } 8270 } 8271 } 8272 return status; 8273} 8274 8275status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8276{ 8277 Mutex::Autolock _l(mLock); 8278 status_t status = NO_ERROR; 8279 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8280 status_t cmdStatus; 8281 uint32_t size = sizeof(status_t); 8282 status = (*mEffectInterface)->command(mEffectInterface, 8283 EFFECT_CMD_SET_AUDIO_MODE, 8284 sizeof(audio_mode_t), 8285 &mode, 8286 &size, 8287 &cmdStatus); 8288 if (status == NO_ERROR) { 8289 status = cmdStatus; 8290 } 8291 } 8292 return status; 8293} 8294 8295void AudioFlinger::EffectModule::setSuspended(bool suspended) 8296{ 8297 Mutex::Autolock _l(mLock); 8298 mSuspended = suspended; 8299} 8300 8301bool AudioFlinger::EffectModule::suspended() const 8302{ 8303 Mutex::Autolock _l(mLock); 8304 return mSuspended; 8305} 8306 8307status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8308{ 8309 const size_t SIZE = 256; 8310 char buffer[SIZE]; 8311 String8 result; 8312 8313 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8314 result.append(buffer); 8315 8316 bool locked = tryLock(mLock); 8317 // failed to lock - AudioFlinger is probably deadlocked 8318 if (!locked) { 8319 result.append("\t\tCould not lock Fx mutex:\n"); 8320 } 8321 8322 result.append("\t\tSession Status State Engine:\n"); 8323 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8324 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8325 result.append(buffer); 8326 8327 result.append("\t\tDescriptor:\n"); 8328 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8329 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8330 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8331 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8332 result.append(buffer); 8333 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8334 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8335 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8336 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8337 result.append(buffer); 8338 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8339 mDescriptor.apiVersion, 8340 mDescriptor.flags); 8341 result.append(buffer); 8342 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8343 mDescriptor.name); 8344 result.append(buffer); 8345 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8346 mDescriptor.implementor); 8347 result.append(buffer); 8348 8349 result.append("\t\t- Input configuration:\n"); 8350 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8351 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8352 (uint32_t)mConfig.inputCfg.buffer.raw, 8353 mConfig.inputCfg.buffer.frameCount, 8354 mConfig.inputCfg.samplingRate, 8355 mConfig.inputCfg.channels, 8356 mConfig.inputCfg.format); 8357 result.append(buffer); 8358 8359 result.append("\t\t- Output configuration:\n"); 8360 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8361 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8362 (uint32_t)mConfig.outputCfg.buffer.raw, 8363 mConfig.outputCfg.buffer.frameCount, 8364 mConfig.outputCfg.samplingRate, 8365 mConfig.outputCfg.channels, 8366 mConfig.outputCfg.format); 8367 result.append(buffer); 8368 8369 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8370 result.append(buffer); 8371 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8372 for (size_t i = 0; i < mHandles.size(); ++i) { 8373 sp<EffectHandle> handle = mHandles[i].promote(); 8374 if (handle != 0) { 8375 handle->dump(buffer, SIZE); 8376 result.append(buffer); 8377 } 8378 } 8379 8380 result.append("\n"); 8381 8382 write(fd, result.string(), result.length()); 8383 8384 if (locked) { 8385 mLock.unlock(); 8386 } 8387 8388 return NO_ERROR; 8389} 8390 8391// ---------------------------------------------------------------------------- 8392// EffectHandle implementation 8393// ---------------------------------------------------------------------------- 8394 8395#undef LOG_TAG 8396#define LOG_TAG "AudioFlinger::EffectHandle" 8397 8398AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8399 const sp<AudioFlinger::Client>& client, 8400 const sp<IEffectClient>& effectClient, 8401 int32_t priority) 8402 : BnEffect(), 8403 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8404 mPriority(priority), mHasControl(false), mEnabled(false) 8405{ 8406 ALOGV("constructor %p", this); 8407 8408 if (client == 0) { 8409 return; 8410 } 8411 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8412 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8413 if (mCblkMemory != 0) { 8414 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8415 8416 if (mCblk != NULL) { 8417 new(mCblk) effect_param_cblk_t(); 8418 mBuffer = (uint8_t *)mCblk + bufOffset; 8419 } 8420 } else { 8421 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8422 return; 8423 } 8424} 8425 8426AudioFlinger::EffectHandle::~EffectHandle() 8427{ 8428 ALOGV("Destructor %p", this); 8429 disconnect(false); 8430 ALOGV("Destructor DONE %p", this); 8431} 8432 8433status_t AudioFlinger::EffectHandle::enable() 8434{ 8435 ALOGV("enable %p", this); 8436 if (!mHasControl) return INVALID_OPERATION; 8437 if (mEffect == 0) return DEAD_OBJECT; 8438 8439 if (mEnabled) { 8440 return NO_ERROR; 8441 } 8442 8443 mEnabled = true; 8444 8445 sp<ThreadBase> thread = mEffect->thread().promote(); 8446 if (thread != 0) { 8447 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8448 } 8449 8450 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8451 if (mEffect->suspended()) { 8452 return NO_ERROR; 8453 } 8454 8455 status_t status = mEffect->setEnabled(true); 8456 if (status != NO_ERROR) { 8457 if (thread != 0) { 8458 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8459 } 8460 mEnabled = false; 8461 } 8462 return status; 8463} 8464 8465status_t AudioFlinger::EffectHandle::disable() 8466{ 8467 ALOGV("disable %p", this); 8468 if (!mHasControl) return INVALID_OPERATION; 8469 if (mEffect == 0) return DEAD_OBJECT; 8470 8471 if (!mEnabled) { 8472 return NO_ERROR; 8473 } 8474 mEnabled = false; 8475 8476 if (mEffect->suspended()) { 8477 return NO_ERROR; 8478 } 8479 8480 status_t status = mEffect->setEnabled(false); 8481 8482 sp<ThreadBase> thread = mEffect->thread().promote(); 8483 if (thread != 0) { 8484 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8485 } 8486 8487 return status; 8488} 8489 8490void AudioFlinger::EffectHandle::disconnect() 8491{ 8492 disconnect(true); 8493} 8494 8495void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8496{ 8497 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8498 if (mEffect == 0) { 8499 return; 8500 } 8501 mEffect->disconnect(this, unpinIfLast); 8502 8503 if (mHasControl && mEnabled) { 8504 sp<ThreadBase> thread = mEffect->thread().promote(); 8505 if (thread != 0) { 8506 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8507 } 8508 } 8509 8510 // release sp on module => module destructor can be called now 8511 mEffect.clear(); 8512 if (mClient != 0) { 8513 if (mCblk != NULL) { 8514 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8515 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8516 } 8517 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8518 // Client destructor must run with AudioFlinger mutex locked 8519 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8520 mClient.clear(); 8521 } 8522} 8523 8524status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8525 uint32_t cmdSize, 8526 void *pCmdData, 8527 uint32_t *replySize, 8528 void *pReplyData) 8529{ 8530// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8531// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8532 8533 // only get parameter command is permitted for applications not controlling the effect 8534 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8535 return INVALID_OPERATION; 8536 } 8537 if (mEffect == 0) return DEAD_OBJECT; 8538 if (mClient == 0) return INVALID_OPERATION; 8539 8540 // handle commands that are not forwarded transparently to effect engine 8541 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8542 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8543 // no risk to block the whole media server process or mixer threads is we are stuck here 8544 Mutex::Autolock _l(mCblk->lock); 8545 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8546 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8547 mCblk->serverIndex = 0; 8548 mCblk->clientIndex = 0; 8549 return BAD_VALUE; 8550 } 8551 status_t status = NO_ERROR; 8552 while (mCblk->serverIndex < mCblk->clientIndex) { 8553 int reply; 8554 uint32_t rsize = sizeof(int); 8555 int *p = (int *)(mBuffer + mCblk->serverIndex); 8556 int size = *p++; 8557 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8558 ALOGW("command(): invalid parameter block size"); 8559 break; 8560 } 8561 effect_param_t *param = (effect_param_t *)p; 8562 if (param->psize == 0 || param->vsize == 0) { 8563 ALOGW("command(): null parameter or value size"); 8564 mCblk->serverIndex += size; 8565 continue; 8566 } 8567 uint32_t psize = sizeof(effect_param_t) + 8568 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8569 param->vsize; 8570 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8571 psize, 8572 p, 8573 &rsize, 8574 &reply); 8575 // stop at first error encountered 8576 if (ret != NO_ERROR) { 8577 status = ret; 8578 *(int *)pReplyData = reply; 8579 break; 8580 } else if (reply != NO_ERROR) { 8581 *(int *)pReplyData = reply; 8582 break; 8583 } 8584 mCblk->serverIndex += size; 8585 } 8586 mCblk->serverIndex = 0; 8587 mCblk->clientIndex = 0; 8588 return status; 8589 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8590 *(int *)pReplyData = NO_ERROR; 8591 return enable(); 8592 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8593 *(int *)pReplyData = NO_ERROR; 8594 return disable(); 8595 } 8596 8597 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8598} 8599 8600void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8601{ 8602 ALOGV("setControl %p control %d", this, hasControl); 8603 8604 mHasControl = hasControl; 8605 mEnabled = enabled; 8606 8607 if (signal && mEffectClient != 0) { 8608 mEffectClient->controlStatusChanged(hasControl); 8609 } 8610} 8611 8612void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8613 uint32_t cmdSize, 8614 void *pCmdData, 8615 uint32_t replySize, 8616 void *pReplyData) 8617{ 8618 if (mEffectClient != 0) { 8619 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8620 } 8621} 8622 8623 8624 8625void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8626{ 8627 if (mEffectClient != 0) { 8628 mEffectClient->enableStatusChanged(enabled); 8629 } 8630} 8631 8632status_t AudioFlinger::EffectHandle::onTransact( 8633 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8634{ 8635 return BnEffect::onTransact(code, data, reply, flags); 8636} 8637 8638 8639void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8640{ 8641 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8642 8643 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8644 (mClient == 0) ? getpid_cached : mClient->pid(), 8645 mPriority, 8646 mHasControl, 8647 !locked, 8648 mCblk ? mCblk->clientIndex : 0, 8649 mCblk ? mCblk->serverIndex : 0 8650 ); 8651 8652 if (locked) { 8653 mCblk->lock.unlock(); 8654 } 8655} 8656 8657#undef LOG_TAG 8658#define LOG_TAG "AudioFlinger::EffectChain" 8659 8660AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8661 int sessionId) 8662 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8663 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8664 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8665{ 8666 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8667 if (thread == NULL) { 8668 return; 8669 } 8670 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8671 thread->frameCount(); 8672} 8673 8674AudioFlinger::EffectChain::~EffectChain() 8675{ 8676 if (mOwnInBuffer) { 8677 delete mInBuffer; 8678 } 8679 8680} 8681 8682// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8683sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8684{ 8685 size_t size = mEffects.size(); 8686 8687 for (size_t i = 0; i < size; i++) { 8688 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8689 return mEffects[i]; 8690 } 8691 } 8692 return 0; 8693} 8694 8695// getEffectFromId_l() must be called with ThreadBase::mLock held 8696sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8697{ 8698 size_t size = mEffects.size(); 8699 8700 for (size_t i = 0; i < size; i++) { 8701 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8702 if (id == 0 || mEffects[i]->id() == id) { 8703 return mEffects[i]; 8704 } 8705 } 8706 return 0; 8707} 8708 8709// getEffectFromType_l() must be called with ThreadBase::mLock held 8710sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8711 const effect_uuid_t *type) 8712{ 8713 size_t size = mEffects.size(); 8714 8715 for (size_t i = 0; i < size; i++) { 8716 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8717 return mEffects[i]; 8718 } 8719 } 8720 return 0; 8721} 8722 8723// Must be called with EffectChain::mLock locked 8724void AudioFlinger::EffectChain::process_l() 8725{ 8726 sp<ThreadBase> thread = mThread.promote(); 8727 if (thread == 0) { 8728 ALOGW("process_l(): cannot promote mixer thread"); 8729 return; 8730 } 8731 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8732 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8733 // always process effects unless no more tracks are on the session and the effect tail 8734 // has been rendered 8735 bool doProcess = true; 8736 if (!isGlobalSession) { 8737 bool tracksOnSession = (trackCnt() != 0); 8738 8739 if (!tracksOnSession && mTailBufferCount == 0) { 8740 doProcess = false; 8741 } 8742 8743 if (activeTrackCnt() == 0) { 8744 // if no track is active and the effect tail has not been rendered, 8745 // the input buffer must be cleared here as the mixer process will not do it 8746 if (tracksOnSession || mTailBufferCount > 0) { 8747 size_t numSamples = thread->frameCount() * thread->channelCount(); 8748 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8749 if (mTailBufferCount > 0) { 8750 mTailBufferCount--; 8751 } 8752 } 8753 } 8754 } 8755 8756 size_t size = mEffects.size(); 8757 if (doProcess) { 8758 for (size_t i = 0; i < size; i++) { 8759 mEffects[i]->process(); 8760 } 8761 } 8762 for (size_t i = 0; i < size; i++) { 8763 mEffects[i]->updateState(); 8764 } 8765} 8766 8767// addEffect_l() must be called with PlaybackThread::mLock held 8768status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8769{ 8770 effect_descriptor_t desc = effect->desc(); 8771 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8772 8773 Mutex::Autolock _l(mLock); 8774 effect->setChain(this); 8775 sp<ThreadBase> thread = mThread.promote(); 8776 if (thread == 0) { 8777 return NO_INIT; 8778 } 8779 effect->setThread(thread); 8780 8781 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8782 // Auxiliary effects are inserted at the beginning of mEffects vector as 8783 // they are processed first and accumulated in chain input buffer 8784 mEffects.insertAt(effect, 0); 8785 8786 // the input buffer for auxiliary effect contains mono samples in 8787 // 32 bit format. This is to avoid saturation in AudoMixer 8788 // accumulation stage. Saturation is done in EffectModule::process() before 8789 // calling the process in effect engine 8790 size_t numSamples = thread->frameCount(); 8791 int32_t *buffer = new int32_t[numSamples]; 8792 memset(buffer, 0, numSamples * sizeof(int32_t)); 8793 effect->setInBuffer((int16_t *)buffer); 8794 // auxiliary effects output samples to chain input buffer for further processing 8795 // by insert effects 8796 effect->setOutBuffer(mInBuffer); 8797 } else { 8798 // Insert effects are inserted at the end of mEffects vector as they are processed 8799 // after track and auxiliary effects. 8800 // Insert effect order as a function of indicated preference: 8801 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8802 // another effect is present 8803 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8804 // last effect claiming first position 8805 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8806 // first effect claiming last position 8807 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8808 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8809 // already present 8810 8811 size_t size = mEffects.size(); 8812 size_t idx_insert = size; 8813 ssize_t idx_insert_first = -1; 8814 ssize_t idx_insert_last = -1; 8815 8816 for (size_t i = 0; i < size; i++) { 8817 effect_descriptor_t d = mEffects[i]->desc(); 8818 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8819 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8820 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8821 // check invalid effect chaining combinations 8822 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8823 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8824 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8825 return INVALID_OPERATION; 8826 } 8827 // remember position of first insert effect and by default 8828 // select this as insert position for new effect 8829 if (idx_insert == size) { 8830 idx_insert = i; 8831 } 8832 // remember position of last insert effect claiming 8833 // first position 8834 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8835 idx_insert_first = i; 8836 } 8837 // remember position of first insert effect claiming 8838 // last position 8839 if (iPref == EFFECT_FLAG_INSERT_LAST && 8840 idx_insert_last == -1) { 8841 idx_insert_last = i; 8842 } 8843 } 8844 } 8845 8846 // modify idx_insert from first position if needed 8847 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8848 if (idx_insert_last != -1) { 8849 idx_insert = idx_insert_last; 8850 } else { 8851 idx_insert = size; 8852 } 8853 } else { 8854 if (idx_insert_first != -1) { 8855 idx_insert = idx_insert_first + 1; 8856 } 8857 } 8858 8859 // always read samples from chain input buffer 8860 effect->setInBuffer(mInBuffer); 8861 8862 // if last effect in the chain, output samples to chain 8863 // output buffer, otherwise to chain input buffer 8864 if (idx_insert == size) { 8865 if (idx_insert != 0) { 8866 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8867 mEffects[idx_insert-1]->configure(); 8868 } 8869 effect->setOutBuffer(mOutBuffer); 8870 } else { 8871 effect->setOutBuffer(mInBuffer); 8872 } 8873 mEffects.insertAt(effect, idx_insert); 8874 8875 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8876 } 8877 effect->configure(); 8878 return NO_ERROR; 8879} 8880 8881// removeEffect_l() must be called with PlaybackThread::mLock held 8882size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8883{ 8884 Mutex::Autolock _l(mLock); 8885 size_t size = mEffects.size(); 8886 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8887 8888 for (size_t i = 0; i < size; i++) { 8889 if (effect == mEffects[i]) { 8890 // calling stop here will remove pre-processing effect from the audio HAL. 8891 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8892 // the middle of a read from audio HAL 8893 if (mEffects[i]->state() == EffectModule::ACTIVE || 8894 mEffects[i]->state() == EffectModule::STOPPING) { 8895 mEffects[i]->stop(); 8896 } 8897 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8898 delete[] effect->inBuffer(); 8899 } else { 8900 if (i == size - 1 && i != 0) { 8901 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8902 mEffects[i - 1]->configure(); 8903 } 8904 } 8905 mEffects.removeAt(i); 8906 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8907 break; 8908 } 8909 } 8910 8911 return mEffects.size(); 8912} 8913 8914// setDevice_l() must be called with PlaybackThread::mLock held 8915void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8916{ 8917 size_t size = mEffects.size(); 8918 for (size_t i = 0; i < size; i++) { 8919 mEffects[i]->setDevice(device); 8920 } 8921} 8922 8923// setMode_l() must be called with PlaybackThread::mLock held 8924void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8925{ 8926 size_t size = mEffects.size(); 8927 for (size_t i = 0; i < size; i++) { 8928 mEffects[i]->setMode(mode); 8929 } 8930} 8931 8932// setVolume_l() must be called with PlaybackThread::mLock held 8933bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8934{ 8935 uint32_t newLeft = *left; 8936 uint32_t newRight = *right; 8937 bool hasControl = false; 8938 int ctrlIdx = -1; 8939 size_t size = mEffects.size(); 8940 8941 // first update volume controller 8942 for (size_t i = size; i > 0; i--) { 8943 if (mEffects[i - 1]->isProcessEnabled() && 8944 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8945 ctrlIdx = i - 1; 8946 hasControl = true; 8947 break; 8948 } 8949 } 8950 8951 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8952 if (hasControl) { 8953 *left = mNewLeftVolume; 8954 *right = mNewRightVolume; 8955 } 8956 return hasControl; 8957 } 8958 8959 mVolumeCtrlIdx = ctrlIdx; 8960 mLeftVolume = newLeft; 8961 mRightVolume = newRight; 8962 8963 // second get volume update from volume controller 8964 if (ctrlIdx >= 0) { 8965 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8966 mNewLeftVolume = newLeft; 8967 mNewRightVolume = newRight; 8968 } 8969 // then indicate volume to all other effects in chain. 8970 // Pass altered volume to effects before volume controller 8971 // and requested volume to effects after controller 8972 uint32_t lVol = newLeft; 8973 uint32_t rVol = newRight; 8974 8975 for (size_t i = 0; i < size; i++) { 8976 if ((int)i == ctrlIdx) continue; 8977 // this also works for ctrlIdx == -1 when there is no volume controller 8978 if ((int)i > ctrlIdx) { 8979 lVol = *left; 8980 rVol = *right; 8981 } 8982 mEffects[i]->setVolume(&lVol, &rVol, false); 8983 } 8984 *left = newLeft; 8985 *right = newRight; 8986 8987 return hasControl; 8988} 8989 8990status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8991{ 8992 const size_t SIZE = 256; 8993 char buffer[SIZE]; 8994 String8 result; 8995 8996 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8997 result.append(buffer); 8998 8999 bool locked = tryLock(mLock); 9000 // failed to lock - AudioFlinger is probably deadlocked 9001 if (!locked) { 9002 result.append("\tCould not lock mutex:\n"); 9003 } 9004 9005 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9006 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9007 mEffects.size(), 9008 (uint32_t)mInBuffer, 9009 (uint32_t)mOutBuffer, 9010 mActiveTrackCnt); 9011 result.append(buffer); 9012 write(fd, result.string(), result.size()); 9013 9014 for (size_t i = 0; i < mEffects.size(); ++i) { 9015 sp<EffectModule> effect = mEffects[i]; 9016 if (effect != 0) { 9017 effect->dump(fd, args); 9018 } 9019 } 9020 9021 if (locked) { 9022 mLock.unlock(); 9023 } 9024 9025 return NO_ERROR; 9026} 9027 9028// must be called with ThreadBase::mLock held 9029void AudioFlinger::EffectChain::setEffectSuspended_l( 9030 const effect_uuid_t *type, bool suspend) 9031{ 9032 sp<SuspendedEffectDesc> desc; 9033 // use effect type UUID timelow as key as there is no real risk of identical 9034 // timeLow fields among effect type UUIDs. 9035 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9036 if (suspend) { 9037 if (index >= 0) { 9038 desc = mSuspendedEffects.valueAt(index); 9039 } else { 9040 desc = new SuspendedEffectDesc(); 9041 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9042 mSuspendedEffects.add(type->timeLow, desc); 9043 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9044 } 9045 if (desc->mRefCount++ == 0) { 9046 sp<EffectModule> effect = getEffectIfEnabled(type); 9047 if (effect != 0) { 9048 desc->mEffect = effect; 9049 effect->setSuspended(true); 9050 effect->setEnabled(false); 9051 } 9052 } 9053 } else { 9054 if (index < 0) { 9055 return; 9056 } 9057 desc = mSuspendedEffects.valueAt(index); 9058 if (desc->mRefCount <= 0) { 9059 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9060 desc->mRefCount = 1; 9061 } 9062 if (--desc->mRefCount == 0) { 9063 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9064 if (desc->mEffect != 0) { 9065 sp<EffectModule> effect = desc->mEffect.promote(); 9066 if (effect != 0) { 9067 effect->setSuspended(false); 9068 sp<EffectHandle> handle = effect->controlHandle(); 9069 if (handle != 0) { 9070 effect->setEnabled(handle->enabled()); 9071 } 9072 } 9073 desc->mEffect.clear(); 9074 } 9075 mSuspendedEffects.removeItemsAt(index); 9076 } 9077 } 9078} 9079 9080// must be called with ThreadBase::mLock held 9081void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9082{ 9083 sp<SuspendedEffectDesc> desc; 9084 9085 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9086 if (suspend) { 9087 if (index >= 0) { 9088 desc = mSuspendedEffects.valueAt(index); 9089 } else { 9090 desc = new SuspendedEffectDesc(); 9091 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9092 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9093 } 9094 if (desc->mRefCount++ == 0) { 9095 Vector< sp<EffectModule> > effects; 9096 getSuspendEligibleEffects(effects); 9097 for (size_t i = 0; i < effects.size(); i++) { 9098 setEffectSuspended_l(&effects[i]->desc().type, true); 9099 } 9100 } 9101 } else { 9102 if (index < 0) { 9103 return; 9104 } 9105 desc = mSuspendedEffects.valueAt(index); 9106 if (desc->mRefCount <= 0) { 9107 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9108 desc->mRefCount = 1; 9109 } 9110 if (--desc->mRefCount == 0) { 9111 Vector<const effect_uuid_t *> types; 9112 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9113 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9114 continue; 9115 } 9116 types.add(&mSuspendedEffects.valueAt(i)->mType); 9117 } 9118 for (size_t i = 0; i < types.size(); i++) { 9119 setEffectSuspended_l(types[i], false); 9120 } 9121 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9122 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9123 } 9124 } 9125} 9126 9127 9128// The volume effect is used for automated tests only 9129#ifndef OPENSL_ES_H_ 9130static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9131 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9132const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9133#endif //OPENSL_ES_H_ 9134 9135bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9136{ 9137 // auxiliary effects and visualizer are never suspended on output mix 9138 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9139 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9140 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9141 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9142 return false; 9143 } 9144 return true; 9145} 9146 9147void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9148{ 9149 effects.clear(); 9150 for (size_t i = 0; i < mEffects.size(); i++) { 9151 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9152 effects.add(mEffects[i]); 9153 } 9154 } 9155} 9156 9157sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9158 const effect_uuid_t *type) 9159{ 9160 sp<EffectModule> effect = getEffectFromType_l(type); 9161 return effect != 0 && effect->isEnabled() ? effect : 0; 9162} 9163 9164void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9165 bool enabled) 9166{ 9167 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9168 if (enabled) { 9169 if (index < 0) { 9170 // if the effect is not suspend check if all effects are suspended 9171 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9172 if (index < 0) { 9173 return; 9174 } 9175 if (!isEffectEligibleForSuspend(effect->desc())) { 9176 return; 9177 } 9178 setEffectSuspended_l(&effect->desc().type, enabled); 9179 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9180 if (index < 0) { 9181 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9182 return; 9183 } 9184 } 9185 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9186 effect->desc().type.timeLow); 9187 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9188 // if effect is requested to suspended but was not yet enabled, supend it now. 9189 if (desc->mEffect == 0) { 9190 desc->mEffect = effect; 9191 effect->setEnabled(false); 9192 effect->setSuspended(true); 9193 } 9194 } else { 9195 if (index < 0) { 9196 return; 9197 } 9198 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9199 effect->desc().type.timeLow); 9200 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9201 desc->mEffect.clear(); 9202 effect->setSuspended(false); 9203 } 9204} 9205 9206#undef LOG_TAG 9207#define LOG_TAG "AudioFlinger" 9208 9209// ---------------------------------------------------------------------------- 9210 9211status_t AudioFlinger::onTransact( 9212 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9213{ 9214 return BnAudioFlinger::onTransact(code, data, reply, flags); 9215} 9216 9217}; // namespace android 9218