AudioFlinger.cpp revision 1127d65d536ebbe447ee17ce0926a7ce4a2a3c08
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 <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include <media/nbaio/AudioStreamOutSink.h> 80#include <media/nbaio/MonoPipe.h> 81#include <media/nbaio/MonoPipeReader.h> 82#include <media/nbaio/Pipe.h> 83#include <media/nbaio/PipeReader.h> 84#include <media/nbaio/SourceAudioBufferProvider.h> 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 169// for the track. The client then sub-divides this into smaller buffers for its use. 170// Currently the client uses double-buffering by default, but doesn't tell us about that. 171// So for now we just assume that client is double-buffered. 172// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 173// N-buffering, so AudioFlinger could allocate the right amount of memory. 174// See the client's minBufCount and mNotificationFramesAct calculations for details. 175static const int kFastTrackMultiplier = 2; 176 177// ---------------------------------------------------------------------------- 178 179#ifdef ADD_BATTERY_DATA 180// To collect the amplifier usage 181static void addBatteryData(uint32_t params) { 182 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 183 if (service == NULL) { 184 // it already logged 185 return; 186 } 187 188 service->addBatteryData(params); 189} 190#endif 191 192static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 193{ 194 const hw_module_t *mod; 195 int rc; 196 197 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 198 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 199 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 200 if (rc) { 201 goto out; 202 } 203 rc = audio_hw_device_open(mod, dev); 204 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 205 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 206 if (rc) { 207 goto out; 208 } 209 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 210 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 211 rc = BAD_VALUE; 212 goto out; 213 } 214 return 0; 215 216out: 217 *dev = NULL; 218 return rc; 219} 220 221// ---------------------------------------------------------------------------- 222 223AudioFlinger::AudioFlinger() 224 : BnAudioFlinger(), 225 mPrimaryHardwareDev(NULL), 226 mHardwareStatus(AUDIO_HW_IDLE), 227 mMasterVolume(1.0f), 228 mMasterMute(false), 229 mNextUniqueId(1), 230 mMode(AUDIO_MODE_INVALID), 231 mBtNrecIsOff(false) 232{ 233} 234 235void AudioFlinger::onFirstRef() 236{ 237 int rc = 0; 238 239 Mutex::Autolock _l(mLock); 240 241 /* TODO: move all this work into an Init() function */ 242 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 243 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 244 uint32_t int_val; 245 if (1 == sscanf(val_str, "%u", &int_val)) { 246 mStandbyTimeInNsecs = milliseconds(int_val); 247 ALOGI("Using %u mSec as standby time.", int_val); 248 } else { 249 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 250 ALOGI("Using default %u mSec as standby time.", 251 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 252 } 253 } 254 255 mMode = AUDIO_MODE_NORMAL; 256} 257 258AudioFlinger::~AudioFlinger() 259{ 260 while (!mRecordThreads.isEmpty()) { 261 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 262 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 263 } 264 while (!mPlaybackThreads.isEmpty()) { 265 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 266 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 267 } 268 269 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 270 // no mHardwareLock needed, as there are no other references to this 271 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 272 delete mAudioHwDevs.valueAt(i); 273 } 274} 275 276static const char * const audio_interfaces[] = { 277 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 278 AUDIO_HARDWARE_MODULE_ID_A2DP, 279 AUDIO_HARDWARE_MODULE_ID_USB, 280}; 281#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 282 283AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 284 audio_module_handle_t module, 285 audio_devices_t devices) 286{ 287 // if module is 0, the request comes from an old policy manager and we should load 288 // well known modules 289 if (module == 0) { 290 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 291 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 292 loadHwModule_l(audio_interfaces[i]); 293 } 294 // then try to find a module supporting the requested device. 295 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 297 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 298 if ((dev->get_supported_devices != NULL) && 299 (dev->get_supported_devices(dev) & devices) == devices) 300 return audioHwDevice; 301 } 302 } else { 303 // check a match for the requested module handle 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 305 if (audioHwDevice != NULL) { 306 return audioHwDevice; 307 } 308 } 309 310 return NULL; 311} 312 313void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 314{ 315 const size_t SIZE = 256; 316 char buffer[SIZE]; 317 String8 result; 318 319 result.append("Clients:\n"); 320 for (size_t i = 0; i < mClients.size(); ++i) { 321 sp<Client> client = mClients.valueAt(i).promote(); 322 if (client != 0) { 323 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 324 result.append(buffer); 325 } 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367static bool tryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = tryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = tryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 dumpClients(fd, args); 403 dumpInternals(fd, args); 404 405 // dump playback threads 406 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 407 mPlaybackThreads.valueAt(i)->dump(fd, args); 408 } 409 410 // dump record threads 411 for (size_t i = 0; i < mRecordThreads.size(); i++) { 412 mRecordThreads.valueAt(i)->dump(fd, args); 413 } 414 415 // dump all hardware devs 416 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 417 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 418 dev->dump(dev, fd); 419 } 420 421 // dump the serially shared record tee sink 422 if (mRecordTeeSource != 0) { 423 dumpTee(fd, mRecordTeeSource); 424 } 425 426 if (locked) mLock.unlock(); 427 } 428 return NO_ERROR; 429} 430 431sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 432{ 433 // If pid is already in the mClients wp<> map, then use that entry 434 // (for which promote() is always != 0), otherwise create a new entry and Client. 435 sp<Client> client = mClients.valueFor(pid).promote(); 436 if (client == 0) { 437 client = new Client(this, pid); 438 mClients.add(pid, client); 439 } 440 441 return client; 442} 443 444// IAudioFlinger interface 445 446 447sp<IAudioTrack> AudioFlinger::createTrack( 448 pid_t pid, 449 audio_stream_type_t streamType, 450 uint32_t sampleRate, 451 audio_format_t format, 452 audio_channel_mask_t channelMask, 453 int frameCount, 454 IAudioFlinger::track_flags_t *flags, 455 const sp<IMemory>& sharedBuffer, 456 audio_io_handle_t output, 457 pid_t tid, 458 int *sessionId, 459 status_t *status) 460{ 461 sp<PlaybackThread::Track> track; 462 sp<TrackHandle> trackHandle; 463 sp<Client> client; 464 status_t lStatus; 465 int lSessionId; 466 467 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 468 // but if someone uses binder directly they could bypass that and cause us to crash 469 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 470 ALOGE("createTrack() invalid stream type %d", streamType); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 { 476 Mutex::Autolock _l(mLock); 477 PlaybackThread *thread = checkPlaybackThread_l(output); 478 PlaybackThread *effectThread = NULL; 479 if (thread == NULL) { 480 ALOGE("unknown output thread"); 481 lStatus = BAD_VALUE; 482 goto Exit; 483 } 484 485 client = registerPid_l(pid); 486 487 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 488 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 489 // check if an effect chain with the same session ID is present on another 490 // output thread and move it here. 491 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 492 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 493 if (mPlaybackThreads.keyAt(i) != output) { 494 uint32_t sessions = t->hasAudioSession(*sessionId); 495 if (sessions & PlaybackThread::EFFECT_SESSION) { 496 effectThread = t.get(); 497 break; 498 } 499 } 500 } 501 lSessionId = *sessionId; 502 } else { 503 // if no audio session id is provided, create one here 504 lSessionId = nextUniqueId(); 505 if (sessionId != NULL) { 506 *sessionId = lSessionId; 507 } 508 } 509 ALOGV("createTrack() lSessionId: %d", lSessionId); 510 511 track = thread->createTrack_l(client, streamType, sampleRate, format, 512 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 513 514 // move effect chain to this output thread if an effect on same session was waiting 515 // for a track to be created 516 if (lStatus == NO_ERROR && effectThread != NULL) { 517 Mutex::Autolock _dl(thread->mLock); 518 Mutex::Autolock _sl(effectThread->mLock); 519 moveEffectChain_l(lSessionId, effectThread, thread, true); 520 } 521 522 // Look for sync events awaiting for a session to be used. 523 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 524 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 525 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 526 if (lStatus == NO_ERROR) { 527 (void) track->setSyncEvent(mPendingSyncEvents[i]); 528 } else { 529 mPendingSyncEvents[i]->cancel(); 530 } 531 mPendingSyncEvents.removeAt(i); 532 i--; 533 } 534 } 535 } 536 } 537 if (lStatus == NO_ERROR) { 538 trackHandle = new TrackHandle(track); 539 } else { 540 // remove local strong reference to Client before deleting the Track so that the Client 541 // destructor is called by the TrackBase destructor with mLock held 542 client.clear(); 543 track.clear(); 544 } 545 546Exit: 547 if (status != NULL) { 548 *status = lStatus; 549 } 550 return trackHandle; 551} 552 553uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 554{ 555 Mutex::Autolock _l(mLock); 556 PlaybackThread *thread = checkPlaybackThread_l(output); 557 if (thread == NULL) { 558 ALOGW("sampleRate() unknown thread %d", output); 559 return 0; 560 } 561 return thread->sampleRate(); 562} 563 564int AudioFlinger::channelCount(audio_io_handle_t output) const 565{ 566 Mutex::Autolock _l(mLock); 567 PlaybackThread *thread = checkPlaybackThread_l(output); 568 if (thread == NULL) { 569 ALOGW("channelCount() unknown thread %d", output); 570 return 0; 571 } 572 return thread->channelCount(); 573} 574 575audio_format_t AudioFlinger::format(audio_io_handle_t output) const 576{ 577 Mutex::Autolock _l(mLock); 578 PlaybackThread *thread = checkPlaybackThread_l(output); 579 if (thread == NULL) { 580 ALOGW("format() unknown thread %d", output); 581 return AUDIO_FORMAT_INVALID; 582 } 583 return thread->format(); 584} 585 586size_t AudioFlinger::frameCount(audio_io_handle_t output) const 587{ 588 Mutex::Autolock _l(mLock); 589 PlaybackThread *thread = checkPlaybackThread_l(output); 590 if (thread == NULL) { 591 ALOGW("frameCount() unknown thread %d", output); 592 return 0; 593 } 594 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 595 // should examine all callers and fix them to handle smaller counts 596 return thread->frameCount(); 597} 598 599uint32_t AudioFlinger::latency(audio_io_handle_t output) const 600{ 601 Mutex::Autolock _l(mLock); 602 PlaybackThread *thread = checkPlaybackThread_l(output); 603 if (thread == NULL) { 604 ALOGW("latency() unknown thread %d", output); 605 return 0; 606 } 607 return thread->latency(); 608} 609 610status_t AudioFlinger::setMasterVolume(float value) 611{ 612 status_t ret = initCheck(); 613 if (ret != NO_ERROR) { 614 return ret; 615 } 616 617 // check calling permissions 618 if (!settingsAllowed()) { 619 return PERMISSION_DENIED; 620 } 621 622 Mutex::Autolock _l(mLock); 623 mMasterVolume = value; 624 625 // Set master volume in the HALs which support it. 626 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 627 AutoMutex lock(mHardwareLock); 628 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 629 630 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 631 if (dev->canSetMasterVolume()) { 632 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 633 } 634 mHardwareStatus = AUDIO_HW_IDLE; 635 } 636 637 // Now set the master volume in each playback thread. Playback threads 638 // assigned to HALs which do not have master volume support will apply 639 // master volume during the mix operation. Threads with HALs which do 640 // support master volume will simply ignore the setting. 641 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 642 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 643 644 return NO_ERROR; 645} 646 647status_t AudioFlinger::setMode(audio_mode_t mode) 648{ 649 status_t ret = initCheck(); 650 if (ret != NO_ERROR) { 651 return ret; 652 } 653 654 // check calling permissions 655 if (!settingsAllowed()) { 656 return PERMISSION_DENIED; 657 } 658 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 659 ALOGW("Illegal value: setMode(%d)", mode); 660 return BAD_VALUE; 661 } 662 663 { // scope for the lock 664 AutoMutex lock(mHardwareLock); 665 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 666 mHardwareStatus = AUDIO_HW_SET_MODE; 667 ret = dev->set_mode(dev, mode); 668 mHardwareStatus = AUDIO_HW_IDLE; 669 } 670 671 if (NO_ERROR == ret) { 672 Mutex::Autolock _l(mLock); 673 mMode = mode; 674 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 675 mPlaybackThreads.valueAt(i)->setMode(mode); 676 } 677 678 return ret; 679} 680 681status_t AudioFlinger::setMicMute(bool state) 682{ 683 status_t ret = initCheck(); 684 if (ret != NO_ERROR) { 685 return ret; 686 } 687 688 // check calling permissions 689 if (!settingsAllowed()) { 690 return PERMISSION_DENIED; 691 } 692 693 AutoMutex lock(mHardwareLock); 694 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 695 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 696 ret = dev->set_mic_mute(dev, state); 697 mHardwareStatus = AUDIO_HW_IDLE; 698 return ret; 699} 700 701bool AudioFlinger::getMicMute() const 702{ 703 status_t ret = initCheck(); 704 if (ret != NO_ERROR) { 705 return false; 706 } 707 708 bool state = AUDIO_MODE_INVALID; 709 AutoMutex lock(mHardwareLock); 710 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 711 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 712 dev->get_mic_mute(dev, &state); 713 mHardwareStatus = AUDIO_HW_IDLE; 714 return state; 715} 716 717status_t AudioFlinger::setMasterMute(bool muted) 718{ 719 status_t ret = initCheck(); 720 if (ret != NO_ERROR) { 721 return ret; 722 } 723 724 // check calling permissions 725 if (!settingsAllowed()) { 726 return PERMISSION_DENIED; 727 } 728 729 Mutex::Autolock _l(mLock); 730 mMasterMute = muted; 731 732 // Set master mute in the HALs which support it. 733 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 734 AutoMutex lock(mHardwareLock); 735 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 736 737 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 738 if (dev->canSetMasterMute()) { 739 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 740 } 741 mHardwareStatus = AUDIO_HW_IDLE; 742 } 743 744 // Now set the master mute in each playback thread. Playback threads 745 // assigned to HALs which do not have master mute support will apply master 746 // mute during the mix operation. Threads with HALs which do support master 747 // mute will simply ignore the setting. 748 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 749 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 750 751 return NO_ERROR; 752} 753 754float AudioFlinger::masterVolume() const 755{ 756 Mutex::Autolock _l(mLock); 757 return masterVolume_l(); 758} 759 760bool AudioFlinger::masterMute() const 761{ 762 Mutex::Autolock _l(mLock); 763 return masterMute_l(); 764} 765 766float AudioFlinger::masterVolume_l() const 767{ 768 return mMasterVolume; 769} 770 771bool AudioFlinger::masterMute_l() const 772{ 773 return mMasterMute; 774} 775 776status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 777 audio_io_handle_t output) 778{ 779 // check calling permissions 780 if (!settingsAllowed()) { 781 return PERMISSION_DENIED; 782 } 783 784 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 785 ALOGE("setStreamVolume() invalid stream %d", stream); 786 return BAD_VALUE; 787 } 788 789 AutoMutex lock(mLock); 790 PlaybackThread *thread = NULL; 791 if (output) { 792 thread = checkPlaybackThread_l(output); 793 if (thread == NULL) { 794 return BAD_VALUE; 795 } 796 } 797 798 mStreamTypes[stream].volume = value; 799 800 if (thread == NULL) { 801 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 802 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 803 } 804 } else { 805 thread->setStreamVolume(stream, value); 806 } 807 808 return NO_ERROR; 809} 810 811status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 812{ 813 // check calling permissions 814 if (!settingsAllowed()) { 815 return PERMISSION_DENIED; 816 } 817 818 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 819 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 820 ALOGE("setStreamMute() invalid stream %d", stream); 821 return BAD_VALUE; 822 } 823 824 AutoMutex lock(mLock); 825 mStreamTypes[stream].mute = muted; 826 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 827 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 828 829 return NO_ERROR; 830} 831 832float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 833{ 834 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 835 return 0.0f; 836 } 837 838 AutoMutex lock(mLock); 839 float volume; 840 if (output) { 841 PlaybackThread *thread = checkPlaybackThread_l(output); 842 if (thread == NULL) { 843 return 0.0f; 844 } 845 volume = thread->streamVolume(stream); 846 } else { 847 volume = streamVolume_l(stream); 848 } 849 850 return volume; 851} 852 853bool AudioFlinger::streamMute(audio_stream_type_t stream) const 854{ 855 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 856 return true; 857 } 858 859 AutoMutex lock(mLock); 860 return streamMute_l(stream); 861} 862 863status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 864{ 865 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 866 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 867 // check calling permissions 868 if (!settingsAllowed()) { 869 return PERMISSION_DENIED; 870 } 871 872 // ioHandle == 0 means the parameters are global to the audio hardware interface 873 if (ioHandle == 0) { 874 Mutex::Autolock _l(mLock); 875 status_t final_result = NO_ERROR; 876 { 877 AutoMutex lock(mHardwareLock); 878 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 879 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 880 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 881 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 882 final_result = result ?: final_result; 883 } 884 mHardwareStatus = AUDIO_HW_IDLE; 885 } 886 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 887 AudioParameter param = AudioParameter(keyValuePairs); 888 String8 value; 889 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 890 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 891 if (mBtNrecIsOff != btNrecIsOff) { 892 for (size_t i = 0; i < mRecordThreads.size(); i++) { 893 sp<RecordThread> thread = mRecordThreads.valueAt(i); 894 audio_devices_t device = thread->inDevice(); 895 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 896 // collect all of the thread's session IDs 897 KeyedVector<int, bool> ids = thread->sessionIds(); 898 // suspend effects associated with those session IDs 899 for (size_t j = 0; j < ids.size(); ++j) { 900 int sessionId = ids.keyAt(j); 901 thread->setEffectSuspended(FX_IID_AEC, 902 suspend, 903 sessionId); 904 thread->setEffectSuspended(FX_IID_NS, 905 suspend, 906 sessionId); 907 } 908 } 909 mBtNrecIsOff = btNrecIsOff; 910 } 911 } 912 String8 screenState; 913 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 914 bool isOff = screenState == "off"; 915 if (isOff != (gScreenState & 1)) { 916 gScreenState = ((gScreenState & ~1) + 2) | isOff; 917 } 918 } 919 return final_result; 920 } 921 922 // hold a strong ref on thread in case closeOutput() or closeInput() is called 923 // and the thread is exited once the lock is released 924 sp<ThreadBase> thread; 925 { 926 Mutex::Autolock _l(mLock); 927 thread = checkPlaybackThread_l(ioHandle); 928 if (thread == 0) { 929 thread = checkRecordThread_l(ioHandle); 930 } else if (thread == primaryPlaybackThread_l()) { 931 // indicate output device change to all input threads for pre processing 932 AudioParameter param = AudioParameter(keyValuePairs); 933 int value; 934 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 935 (value != 0)) { 936 for (size_t i = 0; i < mRecordThreads.size(); i++) { 937 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 938 } 939 } 940 } 941 } 942 if (thread != 0) { 943 return thread->setParameters(keyValuePairs); 944 } 945 return BAD_VALUE; 946} 947 948String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 949{ 950 ALOGVV("getParameters() io %d, keys %s, tid %d, calling pid %d", 951 ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 952 953 Mutex::Autolock _l(mLock); 954 955 if (ioHandle == 0) { 956 String8 out_s8; 957 958 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 959 char *s; 960 { 961 AutoMutex lock(mHardwareLock); 962 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 963 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 964 s = dev->get_parameters(dev, keys.string()); 965 mHardwareStatus = AUDIO_HW_IDLE; 966 } 967 out_s8 += String8(s ? s : ""); 968 free(s); 969 } 970 return out_s8; 971 } 972 973 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 974 if (playbackThread != NULL) { 975 return playbackThread->getParameters(keys); 976 } 977 RecordThread *recordThread = checkRecordThread_l(ioHandle); 978 if (recordThread != NULL) { 979 return recordThread->getParameters(keys); 980 } 981 return String8(""); 982} 983 984size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 985 audio_channel_mask_t channelMask) const 986{ 987 status_t ret = initCheck(); 988 if (ret != NO_ERROR) { 989 return 0; 990 } 991 992 AutoMutex lock(mHardwareLock); 993 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 994 struct audio_config config = { 995 sample_rate: sampleRate, 996 channel_mask: channelMask, 997 format: format, 998 }; 999 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1000 size_t size = dev->get_input_buffer_size(dev, &config); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 return size; 1003} 1004 1005unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1006{ 1007 Mutex::Autolock _l(mLock); 1008 1009 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1010 if (recordThread != NULL) { 1011 return recordThread->getInputFramesLost(); 1012 } 1013 return 0; 1014} 1015 1016status_t AudioFlinger::setVoiceVolume(float value) 1017{ 1018 status_t ret = initCheck(); 1019 if (ret != NO_ERROR) { 1020 return ret; 1021 } 1022 1023 // check calling permissions 1024 if (!settingsAllowed()) { 1025 return PERMISSION_DENIED; 1026 } 1027 1028 AutoMutex lock(mHardwareLock); 1029 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1030 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1031 ret = dev->set_voice_volume(dev, value); 1032 mHardwareStatus = AUDIO_HW_IDLE; 1033 1034 return ret; 1035} 1036 1037status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1038 audio_io_handle_t output) const 1039{ 1040 status_t status; 1041 1042 Mutex::Autolock _l(mLock); 1043 1044 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1045 if (playbackThread != NULL) { 1046 return playbackThread->getRenderPosition(halFrames, dspFrames); 1047 } 1048 1049 return BAD_VALUE; 1050} 1051 1052void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1053{ 1054 1055 Mutex::Autolock _l(mLock); 1056 1057 pid_t pid = IPCThreadState::self()->getCallingPid(); 1058 if (mNotificationClients.indexOfKey(pid) < 0) { 1059 sp<NotificationClient> notificationClient = new NotificationClient(this, 1060 client, 1061 pid); 1062 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1063 1064 mNotificationClients.add(pid, notificationClient); 1065 1066 sp<IBinder> binder = client->asBinder(); 1067 binder->linkToDeath(notificationClient); 1068 1069 // the config change is always sent from playback or record threads to avoid deadlock 1070 // with AudioSystem::gLock 1071 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1072 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1073 } 1074 1075 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1076 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1077 } 1078 } 1079} 1080 1081void AudioFlinger::removeNotificationClient(pid_t pid) 1082{ 1083 Mutex::Autolock _l(mLock); 1084 1085 mNotificationClients.removeItem(pid); 1086 1087 ALOGV("%d died, releasing its sessions", pid); 1088 size_t num = mAudioSessionRefs.size(); 1089 bool removed = false; 1090 for (size_t i = 0; i< num; ) { 1091 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1092 ALOGV(" pid %d @ %d", ref->mPid, i); 1093 if (ref->mPid == pid) { 1094 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1095 mAudioSessionRefs.removeAt(i); 1096 delete ref; 1097 removed = true; 1098 num--; 1099 } else { 1100 i++; 1101 } 1102 } 1103 if (removed) { 1104 purgeStaleEffects_l(); 1105 } 1106} 1107 1108// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1109void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1110{ 1111 size_t size = mNotificationClients.size(); 1112 for (size_t i = 0; i < size; i++) { 1113 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1114 param2); 1115 } 1116} 1117 1118// removeClient_l() must be called with AudioFlinger::mLock held 1119void AudioFlinger::removeClient_l(pid_t pid) 1120{ 1121 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), 1122 IPCThreadState::self()->getCallingPid()); 1123 mClients.removeItem(pid); 1124} 1125 1126// getEffectThread_l() must be called with AudioFlinger::mLock held 1127sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1128{ 1129 sp<PlaybackThread> thread; 1130 1131 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1132 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1133 ALOG_ASSERT(thread == 0); 1134 thread = mPlaybackThreads.valueAt(i); 1135 } 1136 } 1137 1138 return thread; 1139} 1140 1141// ---------------------------------------------------------------------------- 1142 1143AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1144 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 1145 : Thread(false /*canCallJava*/), 1146 mType(type), 1147 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1148 // mChannelMask 1149 mChannelCount(0), 1150 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1151 mParamStatus(NO_ERROR), 1152 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 1153 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1154 // mName will be set by concrete (non-virtual) subclass 1155 mDeathRecipient(new PMDeathRecipient(this)) 1156{ 1157} 1158 1159AudioFlinger::ThreadBase::~ThreadBase() 1160{ 1161 mParamCond.broadcast(); 1162 // do not lock the mutex in destructor 1163 releaseWakeLock_l(); 1164 if (mPowerManager != 0) { 1165 sp<IBinder> binder = mPowerManager->asBinder(); 1166 binder->unlinkToDeath(mDeathRecipient); 1167 } 1168} 1169 1170void AudioFlinger::ThreadBase::exit() 1171{ 1172 ALOGV("ThreadBase::exit"); 1173 // do any cleanup required for exit to succeed 1174 preExit(); 1175 { 1176 // This lock prevents the following race in thread (uniprocessor for illustration): 1177 // if (!exitPending()) { 1178 // // context switch from here to exit() 1179 // // exit() calls requestExit(), what exitPending() observes 1180 // // exit() calls signal(), which is dropped since no waiters 1181 // // context switch back from exit() to here 1182 // mWaitWorkCV.wait(...); 1183 // // now thread is hung 1184 // } 1185 AutoMutex lock(mLock); 1186 requestExit(); 1187 mWaitWorkCV.broadcast(); 1188 } 1189 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1190 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1191 requestExitAndWait(); 1192} 1193 1194status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1195{ 1196 status_t status; 1197 1198 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1199 Mutex::Autolock _l(mLock); 1200 1201 mNewParameters.add(keyValuePairs); 1202 mWaitWorkCV.signal(); 1203 // wait condition with timeout in case the thread loop has exited 1204 // before the request could be processed 1205 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1206 status = mParamStatus; 1207 mWaitWorkCV.signal(); 1208 } else { 1209 status = TIMED_OUT; 1210 } 1211 return status; 1212} 1213 1214void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 1215{ 1216 Mutex::Autolock _l(mLock); 1217 sendIoConfigEvent_l(event, param); 1218} 1219 1220// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 1221void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 1222{ 1223 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 1224 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 1225 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 1226 param); 1227 mWaitWorkCV.signal(); 1228} 1229 1230// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 1231void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 1232{ 1233 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 1234 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 1235 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 1236 mConfigEvents.size(), pid, tid, prio); 1237 mWaitWorkCV.signal(); 1238} 1239 1240void AudioFlinger::ThreadBase::processConfigEvents() 1241{ 1242 mLock.lock(); 1243 while (!mConfigEvents.isEmpty()) { 1244 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1245 ConfigEvent *event = mConfigEvents[0]; 1246 mConfigEvents.removeAt(0); 1247 // release mLock before locking AudioFlinger mLock: lock order is always 1248 // AudioFlinger then ThreadBase to avoid cross deadlock 1249 mLock.unlock(); 1250 switch(event->type()) { 1251 case CFG_EVENT_PRIO: { 1252 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 1253 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 1254 if (err != 0) { 1255 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 1256 "error %d", 1257 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 1258 } 1259 } break; 1260 case CFG_EVENT_IO: { 1261 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 1262 mAudioFlinger->mLock.lock(); 1263 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 1264 mAudioFlinger->mLock.unlock(); 1265 } break; 1266 default: 1267 ALOGE("processConfigEvents() unknown event type %d", event->type()); 1268 break; 1269 } 1270 delete event; 1271 mLock.lock(); 1272 } 1273 mLock.unlock(); 1274} 1275 1276void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1277{ 1278 const size_t SIZE = 256; 1279 char buffer[SIZE]; 1280 String8 result; 1281 1282 bool locked = tryLock(mLock); 1283 if (!locked) { 1284 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1285 write(fd, buffer, strlen(buffer)); 1286 } 1287 1288 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1289 result.append(buffer); 1290 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1291 result.append(buffer); 1292 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1293 result.append(buffer); 1294 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 1295 result.append(buffer); 1296 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1297 result.append(buffer); 1298 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1299 result.append(buffer); 1300 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1301 result.append(buffer); 1302 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1303 result.append(buffer); 1304 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1305 result.append(buffer); 1306 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1307 result.append(buffer); 1308 1309 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1310 result.append(buffer); 1311 result.append(" Index Command"); 1312 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1313 snprintf(buffer, SIZE, "\n %02d ", i); 1314 result.append(buffer); 1315 result.append(mNewParameters[i]); 1316 } 1317 1318 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1319 result.append(buffer); 1320 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1321 mConfigEvents[i]->dump(buffer, SIZE); 1322 result.append(buffer); 1323 } 1324 result.append("\n"); 1325 1326 write(fd, result.string(), result.size()); 1327 1328 if (locked) { 1329 mLock.unlock(); 1330 } 1331} 1332 1333void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1334{ 1335 const size_t SIZE = 256; 1336 char buffer[SIZE]; 1337 String8 result; 1338 1339 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1340 write(fd, buffer, strlen(buffer)); 1341 1342 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1343 sp<EffectChain> chain = mEffectChains[i]; 1344 if (chain != 0) { 1345 chain->dump(fd, args); 1346 } 1347 } 1348} 1349 1350void AudioFlinger::ThreadBase::acquireWakeLock() 1351{ 1352 Mutex::Autolock _l(mLock); 1353 acquireWakeLock_l(); 1354} 1355 1356void AudioFlinger::ThreadBase::acquireWakeLock_l() 1357{ 1358 if (mPowerManager == 0) { 1359 // use checkService() to avoid blocking if power service is not up yet 1360 sp<IBinder> binder = 1361 defaultServiceManager()->checkService(String16("power")); 1362 if (binder == 0) { 1363 ALOGW("Thread %s cannot connect to the power manager service", mName); 1364 } else { 1365 mPowerManager = interface_cast<IPowerManager>(binder); 1366 binder->linkToDeath(mDeathRecipient); 1367 } 1368 } 1369 if (mPowerManager != 0) { 1370 sp<IBinder> binder = new BBinder(); 1371 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1372 binder, 1373 String16(mName)); 1374 if (status == NO_ERROR) { 1375 mWakeLockToken = binder; 1376 } 1377 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1378 } 1379} 1380 1381void AudioFlinger::ThreadBase::releaseWakeLock() 1382{ 1383 Mutex::Autolock _l(mLock); 1384 releaseWakeLock_l(); 1385} 1386 1387void AudioFlinger::ThreadBase::releaseWakeLock_l() 1388{ 1389 if (mWakeLockToken != 0) { 1390 ALOGV("releaseWakeLock_l() %s", mName); 1391 if (mPowerManager != 0) { 1392 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1393 } 1394 mWakeLockToken.clear(); 1395 } 1396} 1397 1398void AudioFlinger::ThreadBase::clearPowerManager() 1399{ 1400 Mutex::Autolock _l(mLock); 1401 releaseWakeLock_l(); 1402 mPowerManager.clear(); 1403} 1404 1405void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1406{ 1407 sp<ThreadBase> thread = mThread.promote(); 1408 if (thread != 0) { 1409 thread->clearPowerManager(); 1410 } 1411 ALOGW("power manager service died !!!"); 1412} 1413 1414void AudioFlinger::ThreadBase::setEffectSuspended( 1415 const effect_uuid_t *type, bool suspend, int sessionId) 1416{ 1417 Mutex::Autolock _l(mLock); 1418 setEffectSuspended_l(type, suspend, sessionId); 1419} 1420 1421void AudioFlinger::ThreadBase::setEffectSuspended_l( 1422 const effect_uuid_t *type, bool suspend, int sessionId) 1423{ 1424 sp<EffectChain> chain = getEffectChain_l(sessionId); 1425 if (chain != 0) { 1426 if (type != NULL) { 1427 chain->setEffectSuspended_l(type, suspend); 1428 } else { 1429 chain->setEffectSuspendedAll_l(suspend); 1430 } 1431 } 1432 1433 updateSuspendedSessions_l(type, suspend, sessionId); 1434} 1435 1436void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1437{ 1438 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1439 if (index < 0) { 1440 return; 1441 } 1442 1443 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1444 mSuspendedSessions.valueAt(index); 1445 1446 for (size_t i = 0; i < sessionEffects.size(); i++) { 1447 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1448 for (int j = 0; j < desc->mRefCount; j++) { 1449 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1450 chain->setEffectSuspendedAll_l(true); 1451 } else { 1452 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1453 desc->mType.timeLow); 1454 chain->setEffectSuspended_l(&desc->mType, true); 1455 } 1456 } 1457 } 1458} 1459 1460void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1461 bool suspend, 1462 int sessionId) 1463{ 1464 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1465 1466 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1467 1468 if (suspend) { 1469 if (index >= 0) { 1470 sessionEffects = mSuspendedSessions.valueAt(index); 1471 } else { 1472 mSuspendedSessions.add(sessionId, sessionEffects); 1473 } 1474 } else { 1475 if (index < 0) { 1476 return; 1477 } 1478 sessionEffects = mSuspendedSessions.valueAt(index); 1479 } 1480 1481 1482 int key = EffectChain::kKeyForSuspendAll; 1483 if (type != NULL) { 1484 key = type->timeLow; 1485 } 1486 index = sessionEffects.indexOfKey(key); 1487 1488 sp<SuspendedSessionDesc> desc; 1489 if (suspend) { 1490 if (index >= 0) { 1491 desc = sessionEffects.valueAt(index); 1492 } else { 1493 desc = new SuspendedSessionDesc(); 1494 if (type != NULL) { 1495 desc->mType = *type; 1496 } 1497 sessionEffects.add(key, desc); 1498 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1499 } 1500 desc->mRefCount++; 1501 } else { 1502 if (index < 0) { 1503 return; 1504 } 1505 desc = sessionEffects.valueAt(index); 1506 if (--desc->mRefCount == 0) { 1507 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1508 sessionEffects.removeItemsAt(index); 1509 if (sessionEffects.isEmpty()) { 1510 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1511 sessionId); 1512 mSuspendedSessions.removeItem(sessionId); 1513 } 1514 } 1515 } 1516 if (!sessionEffects.isEmpty()) { 1517 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1518 } 1519} 1520 1521void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1522 bool enabled, 1523 int sessionId) 1524{ 1525 Mutex::Autolock _l(mLock); 1526 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1527} 1528 1529void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1530 bool enabled, 1531 int sessionId) 1532{ 1533 if (mType != RECORD) { 1534 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1535 // another session. This gives the priority to well behaved effect control panels 1536 // and applications not using global effects. 1537 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1538 // global effects 1539 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1540 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1541 } 1542 } 1543 1544 sp<EffectChain> chain = getEffectChain_l(sessionId); 1545 if (chain != 0) { 1546 chain->checkSuspendOnEffectEnabled(effect, enabled); 1547 } 1548} 1549 1550// ---------------------------------------------------------------------------- 1551 1552AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1553 AudioStreamOut* output, 1554 audio_io_handle_t id, 1555 audio_devices_t device, 1556 type_t type) 1557 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1558 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1559 // mStreamTypes[] initialized in constructor body 1560 mOutput(output), 1561 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1562 mMixerStatus(MIXER_IDLE), 1563 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1564 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1565 mScreenState(gScreenState), 1566 // index 0 is reserved for normal mixer's submix 1567 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1568{ 1569 snprintf(mName, kNameLength, "AudioOut_%X", id); 1570 1571 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1572 // it would be safer to explicitly pass initial masterVolume/masterMute as 1573 // parameter. 1574 // 1575 // If the HAL we are using has support for master volume or master mute, 1576 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1577 // and the mute set to false). 1578 mMasterVolume = audioFlinger->masterVolume_l(); 1579 mMasterMute = audioFlinger->masterMute_l(); 1580 if (mOutput && mOutput->audioHwDev) { 1581 if (mOutput->audioHwDev->canSetMasterVolume()) { 1582 mMasterVolume = 1.0; 1583 } 1584 1585 if (mOutput->audioHwDev->canSetMasterMute()) { 1586 mMasterMute = false; 1587 } 1588 } 1589 1590 readOutputParameters(); 1591 1592 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1593 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1594 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1595 stream = (audio_stream_type_t) (stream + 1)) { 1596 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1597 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1598 } 1599 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1600 // because mAudioFlinger doesn't have one to copy from 1601} 1602 1603AudioFlinger::PlaybackThread::~PlaybackThread() 1604{ 1605 delete [] mMixBuffer; 1606} 1607 1608void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1609{ 1610 dumpInternals(fd, args); 1611 dumpTracks(fd, args); 1612 dumpEffectChains(fd, args); 1613} 1614 1615void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1616{ 1617 const size_t SIZE = 256; 1618 char buffer[SIZE]; 1619 String8 result; 1620 1621 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1622 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1623 const stream_type_t *st = &mStreamTypes[i]; 1624 if (i > 0) { 1625 result.appendFormat(", "); 1626 } 1627 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1628 if (st->mute) { 1629 result.append("M"); 1630 } 1631 } 1632 result.append("\n"); 1633 write(fd, result.string(), result.length()); 1634 result.clear(); 1635 1636 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1637 result.append(buffer); 1638 Track::appendDumpHeader(result); 1639 for (size_t i = 0; i < mTracks.size(); ++i) { 1640 sp<Track> track = mTracks[i]; 1641 if (track != 0) { 1642 track->dump(buffer, SIZE); 1643 result.append(buffer); 1644 } 1645 } 1646 1647 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1648 result.append(buffer); 1649 Track::appendDumpHeader(result); 1650 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1651 sp<Track> track = mActiveTracks[i].promote(); 1652 if (track != 0) { 1653 track->dump(buffer, SIZE); 1654 result.append(buffer); 1655 } 1656 } 1657 write(fd, result.string(), result.size()); 1658 1659 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1660 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1661 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1662 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1663} 1664 1665void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1666{ 1667 const size_t SIZE = 256; 1668 char buffer[SIZE]; 1669 String8 result; 1670 1671 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1672 result.append(buffer); 1673 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1674 ns2ms(systemTime() - mLastWriteTime)); 1675 result.append(buffer); 1676 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1677 result.append(buffer); 1678 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1679 result.append(buffer); 1680 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1681 result.append(buffer); 1682 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1683 result.append(buffer); 1684 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1685 result.append(buffer); 1686 write(fd, result.string(), result.size()); 1687 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1688 1689 dumpBase(fd, args); 1690} 1691 1692// Thread virtuals 1693status_t AudioFlinger::PlaybackThread::readyToRun() 1694{ 1695 status_t status = initCheck(); 1696 if (status == NO_ERROR) { 1697 ALOGI("AudioFlinger's thread %p ready to run", this); 1698 } else { 1699 ALOGE("No working audio driver found."); 1700 } 1701 return status; 1702} 1703 1704void AudioFlinger::PlaybackThread::onFirstRef() 1705{ 1706 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1707} 1708 1709// ThreadBase virtuals 1710void AudioFlinger::PlaybackThread::preExit() 1711{ 1712 ALOGV(" preExit()"); 1713 // FIXME this is using hard-coded strings but in the future, this functionality will be 1714 // converted to use audio HAL extensions required to support tunneling 1715 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1716} 1717 1718// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1719sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1720 const sp<AudioFlinger::Client>& client, 1721 audio_stream_type_t streamType, 1722 uint32_t sampleRate, 1723 audio_format_t format, 1724 audio_channel_mask_t channelMask, 1725 int frameCount, 1726 const sp<IMemory>& sharedBuffer, 1727 int sessionId, 1728 IAudioFlinger::track_flags_t *flags, 1729 pid_t tid, 1730 status_t *status) 1731{ 1732 sp<Track> track; 1733 status_t lStatus; 1734 1735 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1736 1737 // client expresses a preference for FAST, but we get the final say 1738 if (*flags & IAudioFlinger::TRACK_FAST) { 1739 if ( 1740 // not timed 1741 (!isTimed) && 1742 // either of these use cases: 1743 ( 1744 // use case 1: shared buffer with any frame count 1745 ( 1746 (sharedBuffer != 0) 1747 ) || 1748 // use case 2: callback handler and frame count is default or at least as large as HAL 1749 ( 1750 (tid != -1) && 1751 ((frameCount == 0) || 1752 (frameCount >= (int) (mFrameCount * kFastTrackMultiplier))) 1753 ) 1754 ) && 1755 // PCM data 1756 audio_is_linear_pcm(format) && 1757 // mono or stereo 1758 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1759 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1760#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1761 // hardware sample rate 1762 (sampleRate == mSampleRate) && 1763#endif 1764 // normal mixer has an associated fast mixer 1765 hasFastMixer() && 1766 // there are sufficient fast track slots available 1767 (mFastTrackAvailMask != 0) 1768 // FIXME test that MixerThread for this fast track has a capable output HAL 1769 // FIXME add a permission test also? 1770 ) { 1771 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1772 if (frameCount == 0) { 1773 frameCount = mFrameCount * kFastTrackMultiplier; 1774 } 1775 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1776 frameCount, mFrameCount); 1777 } else { 1778 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1779 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1780 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1781 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1782 audio_is_linear_pcm(format), 1783 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1784 *flags &= ~IAudioFlinger::TRACK_FAST; 1785 // For compatibility with AudioTrack calculation, buffer depth is forced 1786 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1787 // This is probably too conservative, but legacy application code may depend on it. 1788 // If you change this calculation, also review the start threshold which is related. 1789 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1790 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1791 if (minBufCount < 2) { 1792 minBufCount = 2; 1793 } 1794 int minFrameCount = mNormalFrameCount * minBufCount; 1795 if (frameCount < minFrameCount) { 1796 frameCount = minFrameCount; 1797 } 1798 } 1799 } 1800 1801 if (mType == DIRECT) { 1802 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1803 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1804 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1805 "for output %p with format %d", 1806 sampleRate, format, channelMask, mOutput, mFormat); 1807 lStatus = BAD_VALUE; 1808 goto Exit; 1809 } 1810 } 1811 } else { 1812 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1813 if (sampleRate > mSampleRate*2) { 1814 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1815 lStatus = BAD_VALUE; 1816 goto Exit; 1817 } 1818 } 1819 1820 lStatus = initCheck(); 1821 if (lStatus != NO_ERROR) { 1822 ALOGE("Audio driver not initialized."); 1823 goto Exit; 1824 } 1825 1826 { // scope for mLock 1827 Mutex::Autolock _l(mLock); 1828 1829 // all tracks in same audio session must share the same routing strategy otherwise 1830 // conflicts will happen when tracks are moved from one output to another by audio policy 1831 // manager 1832 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1833 for (size_t i = 0; i < mTracks.size(); ++i) { 1834 sp<Track> t = mTracks[i]; 1835 if (t != 0 && !t->isOutputTrack()) { 1836 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1837 if (sessionId == t->sessionId() && strategy != actual) { 1838 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1839 strategy, actual); 1840 lStatus = BAD_VALUE; 1841 goto Exit; 1842 } 1843 } 1844 } 1845 1846 if (!isTimed) { 1847 track = new Track(this, client, streamType, sampleRate, format, 1848 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1849 } else { 1850 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1851 channelMask, frameCount, sharedBuffer, sessionId); 1852 } 1853 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1854 lStatus = NO_MEMORY; 1855 goto Exit; 1856 } 1857 mTracks.add(track); 1858 1859 sp<EffectChain> chain = getEffectChain_l(sessionId); 1860 if (chain != 0) { 1861 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1862 track->setMainBuffer(chain->inBuffer()); 1863 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1864 chain->incTrackCnt(); 1865 } 1866 1867 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1868 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1869 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1870 // so ask activity manager to do this on our behalf 1871 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1872 } 1873 } 1874 1875 lStatus = NO_ERROR; 1876 1877Exit: 1878 if (status) { 1879 *status = lStatus; 1880 } 1881 return track; 1882} 1883 1884uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1885{ 1886 if (mFastMixer != NULL) { 1887 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1888 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1889 } 1890 return latency; 1891} 1892 1893uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1894{ 1895 return latency; 1896} 1897 1898uint32_t AudioFlinger::PlaybackThread::latency() const 1899{ 1900 Mutex::Autolock _l(mLock); 1901 return latency_l(); 1902} 1903uint32_t AudioFlinger::PlaybackThread::latency_l() const 1904{ 1905 if (initCheck() == NO_ERROR) { 1906 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1907 } else { 1908 return 0; 1909 } 1910} 1911 1912void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1913{ 1914 Mutex::Autolock _l(mLock); 1915 // Don't apply master volume in SW if our HAL can do it for us. 1916 if (mOutput && mOutput->audioHwDev && 1917 mOutput->audioHwDev->canSetMasterVolume()) { 1918 mMasterVolume = 1.0; 1919 } else { 1920 mMasterVolume = value; 1921 } 1922} 1923 1924void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1925{ 1926 Mutex::Autolock _l(mLock); 1927 // Don't apply master mute in SW if our HAL can do it for us. 1928 if (mOutput && mOutput->audioHwDev && 1929 mOutput->audioHwDev->canSetMasterMute()) { 1930 mMasterMute = false; 1931 } else { 1932 mMasterMute = muted; 1933 } 1934} 1935 1936void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1937{ 1938 Mutex::Autolock _l(mLock); 1939 mStreamTypes[stream].volume = value; 1940} 1941 1942void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1943{ 1944 Mutex::Autolock _l(mLock); 1945 mStreamTypes[stream].mute = muted; 1946} 1947 1948float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1949{ 1950 Mutex::Autolock _l(mLock); 1951 return mStreamTypes[stream].volume; 1952} 1953 1954// addTrack_l() must be called with ThreadBase::mLock held 1955status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1956{ 1957 status_t status = ALREADY_EXISTS; 1958 1959 // set retry count for buffer fill 1960 track->mRetryCount = kMaxTrackStartupRetries; 1961 if (mActiveTracks.indexOf(track) < 0) { 1962 // the track is newly added, make sure it fills up all its 1963 // buffers before playing. This is to ensure the client will 1964 // effectively get the latency it requested. 1965 track->mFillingUpStatus = Track::FS_FILLING; 1966 track->mResetDone = false; 1967 track->mPresentationCompleteFrames = 0; 1968 mActiveTracks.add(track); 1969 if (track->mainBuffer() != mMixBuffer) { 1970 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1971 if (chain != 0) { 1972 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1973 track->sessionId()); 1974 chain->incActiveTrackCnt(); 1975 } 1976 } 1977 1978 status = NO_ERROR; 1979 } 1980 1981 ALOGV("mWaitWorkCV.broadcast"); 1982 mWaitWorkCV.broadcast(); 1983 1984 return status; 1985} 1986 1987// destroyTrack_l() must be called with ThreadBase::mLock held 1988void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1989{ 1990 track->mState = TrackBase::TERMINATED; 1991 // active tracks are removed by threadLoop() 1992 if (mActiveTracks.indexOf(track) < 0) { 1993 removeTrack_l(track); 1994 } 1995} 1996 1997void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1998{ 1999 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2000 mTracks.remove(track); 2001 deleteTrackName_l(track->name()); 2002 // redundant as track is about to be destroyed, for dumpsys only 2003 track->mName = -1; 2004 if (track->isFastTrack()) { 2005 int index = track->mFastIndex; 2006 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2007 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2008 mFastTrackAvailMask |= 1 << index; 2009 // redundant as track is about to be destroyed, for dumpsys only 2010 track->mFastIndex = -1; 2011 } 2012 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2013 if (chain != 0) { 2014 chain->decTrackCnt(); 2015 } 2016} 2017 2018String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2019{ 2020 String8 out_s8 = String8(""); 2021 char *s; 2022 2023 Mutex::Autolock _l(mLock); 2024 if (initCheck() != NO_ERROR) { 2025 return out_s8; 2026 } 2027 2028 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2029 out_s8 = String8(s); 2030 free(s); 2031 return out_s8; 2032} 2033 2034// audioConfigChanged_l() must be called with AudioFlinger::mLock held 2035void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 2036 AudioSystem::OutputDescriptor desc; 2037 void *param2 = NULL; 2038 2039 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 2040 param); 2041 2042 switch (event) { 2043 case AudioSystem::OUTPUT_OPENED: 2044 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2045 desc.channels = mChannelMask; 2046 desc.samplingRate = mSampleRate; 2047 desc.format = mFormat; 2048 desc.frameCount = mNormalFrameCount; // FIXME see 2049 // AudioFlinger::frameCount(audio_io_handle_t) 2050 desc.latency = latency(); 2051 param2 = &desc; 2052 break; 2053 2054 case AudioSystem::STREAM_CONFIG_CHANGED: 2055 param2 = ¶m; 2056 case AudioSystem::OUTPUT_CLOSED: 2057 default: 2058 break; 2059 } 2060 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2061} 2062 2063void AudioFlinger::PlaybackThread::readOutputParameters() 2064{ 2065 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2066 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2067 mChannelCount = (uint16_t)popcount(mChannelMask); 2068 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2069 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2070 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2071 if (mFrameCount & 15) { 2072 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2073 mFrameCount); 2074 } 2075 2076 // Calculate size of normal mix buffer relative to the HAL output buffer size 2077 double multiplier = 1.0; 2078 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2079 kUseFastMixer == FastMixer_Dynamic)) { 2080 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2081 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2082 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2083 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2084 maxNormalFrameCount = maxNormalFrameCount & ~15; 2085 if (maxNormalFrameCount < minNormalFrameCount) { 2086 maxNormalFrameCount = minNormalFrameCount; 2087 } 2088 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2089 if (multiplier <= 1.0) { 2090 multiplier = 1.0; 2091 } else if (multiplier <= 2.0) { 2092 if (2 * mFrameCount <= maxNormalFrameCount) { 2093 multiplier = 2.0; 2094 } else { 2095 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2096 } 2097 } else { 2098 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2099 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 2100 // track, but we sometimes have to do this to satisfy the maximum frame count 2101 // constraint) 2102 // FIXME this rounding up should not be done if no HAL SRC 2103 uint32_t truncMult = (uint32_t) multiplier; 2104 if ((truncMult & 1)) { 2105 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2106 ++truncMult; 2107 } 2108 } 2109 multiplier = (double) truncMult; 2110 } 2111 } 2112 mNormalFrameCount = multiplier * mFrameCount; 2113 // round up to nearest 16 frames to satisfy AudioMixer 2114 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2115 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 2116 mNormalFrameCount); 2117 2118 delete[] mMixBuffer; 2119 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2120 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2121 2122 // force reconfiguration of effect chains and engines to take new buffer size and audio 2123 // parameters into account 2124 // Note that mLock is not held when readOutputParameters() is called from the constructor 2125 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2126 // matter. 2127 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2128 Vector< sp<EffectChain> > effectChains = mEffectChains; 2129 for (size_t i = 0; i < effectChains.size(); i ++) { 2130 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2131 } 2132} 2133 2134 2135status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2136{ 2137 if (halFrames == NULL || dspFrames == NULL) { 2138 return BAD_VALUE; 2139 } 2140 Mutex::Autolock _l(mLock); 2141 if (initCheck() != NO_ERROR) { 2142 return INVALID_OPERATION; 2143 } 2144 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2145 2146 if (isSuspended()) { 2147 // return an estimation of rendered frames when the output is suspended 2148 int32_t frames = mBytesWritten - latency_l(); 2149 if (frames < 0) { 2150 frames = 0; 2151 } 2152 *dspFrames = (uint32_t)frames; 2153 return NO_ERROR; 2154 } else { 2155 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2156 } 2157} 2158 2159uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2160{ 2161 Mutex::Autolock _l(mLock); 2162 uint32_t result = 0; 2163 if (getEffectChain_l(sessionId) != 0) { 2164 result = EFFECT_SESSION; 2165 } 2166 2167 for (size_t i = 0; i < mTracks.size(); ++i) { 2168 sp<Track> track = mTracks[i]; 2169 if (sessionId == track->sessionId() && 2170 !(track->mCblk->flags & CBLK_INVALID)) { 2171 result |= TRACK_SESSION; 2172 break; 2173 } 2174 } 2175 2176 return result; 2177} 2178 2179uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2180{ 2181 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2182 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2183 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2184 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2185 } 2186 for (size_t i = 0; i < mTracks.size(); i++) { 2187 sp<Track> track = mTracks[i]; 2188 if (sessionId == track->sessionId() && 2189 !(track->mCblk->flags & CBLK_INVALID)) { 2190 return AudioSystem::getStrategyForStream(track->streamType()); 2191 } 2192 } 2193 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2194} 2195 2196 2197AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2198{ 2199 Mutex::Autolock _l(mLock); 2200 return mOutput; 2201} 2202 2203AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2204{ 2205 Mutex::Autolock _l(mLock); 2206 AudioStreamOut *output = mOutput; 2207 mOutput = NULL; 2208 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2209 // must push a NULL and wait for ack 2210 mOutputSink.clear(); 2211 mPipeSink.clear(); 2212 mNormalSink.clear(); 2213 return output; 2214} 2215 2216// this method must always be called either with ThreadBase mLock held or inside the thread loop 2217audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2218{ 2219 if (mOutput == NULL) { 2220 return NULL; 2221 } 2222 return &mOutput->stream->common; 2223} 2224 2225uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2226{ 2227 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2228} 2229 2230status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2231{ 2232 if (!isValidSyncEvent(event)) { 2233 return BAD_VALUE; 2234 } 2235 2236 Mutex::Autolock _l(mLock); 2237 2238 for (size_t i = 0; i < mTracks.size(); ++i) { 2239 sp<Track> track = mTracks[i]; 2240 if (event->triggerSession() == track->sessionId()) { 2241 (void) track->setSyncEvent(event); 2242 return NO_ERROR; 2243 } 2244 } 2245 2246 return NAME_NOT_FOUND; 2247} 2248 2249bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2250{ 2251 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2252} 2253 2254void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2255 const Vector< sp<Track> >& tracksToRemove) 2256{ 2257 size_t count = tracksToRemove.size(); 2258 if (CC_UNLIKELY(count)) { 2259 for (size_t i = 0 ; i < count ; i++) { 2260 const sp<Track>& track = tracksToRemove.itemAt(i); 2261 if ((track->sharedBuffer() != 0) && 2262 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2263 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2264 } 2265 } 2266 } 2267 2268} 2269 2270// ---------------------------------------------------------------------------- 2271 2272AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2273 audio_io_handle_t id, audio_devices_t device, type_t type) 2274 : PlaybackThread(audioFlinger, output, id, device, type), 2275 // mAudioMixer below 2276 // mFastMixer below 2277 mFastMixerFutex(0) 2278 // mOutputSink below 2279 // mPipeSink below 2280 // mNormalSink below 2281{ 2282 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2283 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2284 "mFrameCount=%d, mNormalFrameCount=%d", 2285 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2286 mNormalFrameCount); 2287 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2288 2289 // FIXME - Current mixer implementation only supports stereo output 2290 if (mChannelCount != FCC_2) { 2291 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2292 } 2293 2294 // create an NBAIO sink for the HAL output stream, and negotiate 2295 mOutputSink = new AudioStreamOutSink(output->stream); 2296 size_t numCounterOffers = 0; 2297 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2298 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2299 ALOG_ASSERT(index == 0); 2300 2301 // initialize fast mixer depending on configuration 2302 bool initFastMixer; 2303 switch (kUseFastMixer) { 2304 case FastMixer_Never: 2305 initFastMixer = false; 2306 break; 2307 case FastMixer_Always: 2308 initFastMixer = true; 2309 break; 2310 case FastMixer_Static: 2311 case FastMixer_Dynamic: 2312 initFastMixer = mFrameCount < mNormalFrameCount; 2313 break; 2314 } 2315 if (initFastMixer) { 2316 2317 // create a MonoPipe to connect our submix to FastMixer 2318 NBAIO_Format format = mOutputSink->format(); 2319 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2320 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2321 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2322 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2323 const NBAIO_Format offers[1] = {format}; 2324 size_t numCounterOffers = 0; 2325 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2326 ALOG_ASSERT(index == 0); 2327 monoPipe->setAvgFrames((mScreenState & 1) ? 2328 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2329 mPipeSink = monoPipe; 2330 2331#ifdef TEE_SINK_FRAMES 2332 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2333 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2334 numCounterOffers = 0; 2335 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2336 ALOG_ASSERT(index == 0); 2337 mTeeSink = teeSink; 2338 PipeReader *teeSource = new PipeReader(*teeSink); 2339 numCounterOffers = 0; 2340 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2341 ALOG_ASSERT(index == 0); 2342 mTeeSource = teeSource; 2343#endif 2344 2345 // create fast mixer and configure it initially with just one fast track for our submix 2346 mFastMixer = new FastMixer(); 2347 FastMixerStateQueue *sq = mFastMixer->sq(); 2348#ifdef STATE_QUEUE_DUMP 2349 sq->setObserverDump(&mStateQueueObserverDump); 2350 sq->setMutatorDump(&mStateQueueMutatorDump); 2351#endif 2352 FastMixerState *state = sq->begin(); 2353 FastTrack *fastTrack = &state->mFastTracks[0]; 2354 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2355 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2356 fastTrack->mVolumeProvider = NULL; 2357 fastTrack->mGeneration++; 2358 state->mFastTracksGen++; 2359 state->mTrackMask = 1; 2360 // fast mixer will use the HAL output sink 2361 state->mOutputSink = mOutputSink.get(); 2362 state->mOutputSinkGen++; 2363 state->mFrameCount = mFrameCount; 2364 state->mCommand = FastMixerState::COLD_IDLE; 2365 // already done in constructor initialization list 2366 //mFastMixerFutex = 0; 2367 state->mColdFutexAddr = &mFastMixerFutex; 2368 state->mColdGen++; 2369 state->mDumpState = &mFastMixerDumpState; 2370 state->mTeeSink = mTeeSink.get(); 2371 sq->end(); 2372 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2373 2374 // start the fast mixer 2375 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2376 pid_t tid = mFastMixer->getTid(); 2377 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2378 if (err != 0) { 2379 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2380 kPriorityFastMixer, getpid_cached, tid, err); 2381 } 2382 2383#ifdef AUDIO_WATCHDOG 2384 // create and start the watchdog 2385 mAudioWatchdog = new AudioWatchdog(); 2386 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2387 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2388 tid = mAudioWatchdog->getTid(); 2389 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2390 if (err != 0) { 2391 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2392 kPriorityFastMixer, getpid_cached, tid, err); 2393 } 2394#endif 2395 2396 } else { 2397 mFastMixer = NULL; 2398 } 2399 2400 switch (kUseFastMixer) { 2401 case FastMixer_Never: 2402 case FastMixer_Dynamic: 2403 mNormalSink = mOutputSink; 2404 break; 2405 case FastMixer_Always: 2406 mNormalSink = mPipeSink; 2407 break; 2408 case FastMixer_Static: 2409 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2410 break; 2411 } 2412} 2413 2414AudioFlinger::MixerThread::~MixerThread() 2415{ 2416 if (mFastMixer != NULL) { 2417 FastMixerStateQueue *sq = mFastMixer->sq(); 2418 FastMixerState *state = sq->begin(); 2419 if (state->mCommand == FastMixerState::COLD_IDLE) { 2420 int32_t old = android_atomic_inc(&mFastMixerFutex); 2421 if (old == -1) { 2422 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2423 } 2424 } 2425 state->mCommand = FastMixerState::EXIT; 2426 sq->end(); 2427 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2428 mFastMixer->join(); 2429 // Though the fast mixer thread has exited, it's state queue is still valid. 2430 // We'll use that extract the final state which contains one remaining fast track 2431 // corresponding to our sub-mix. 2432 state = sq->begin(); 2433 ALOG_ASSERT(state->mTrackMask == 1); 2434 FastTrack *fastTrack = &state->mFastTracks[0]; 2435 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2436 delete fastTrack->mBufferProvider; 2437 sq->end(false /*didModify*/); 2438 delete mFastMixer; 2439#ifdef AUDIO_WATCHDOG 2440 if (mAudioWatchdog != 0) { 2441 mAudioWatchdog->requestExit(); 2442 mAudioWatchdog->requestExitAndWait(); 2443 mAudioWatchdog.clear(); 2444 } 2445#endif 2446 } 2447 delete mAudioMixer; 2448} 2449 2450class CpuStats { 2451public: 2452 CpuStats(); 2453 void sample(const String8 &title); 2454#ifdef DEBUG_CPU_USAGE 2455private: 2456 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2457 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2458 2459 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2460 2461 int mCpuNum; // thread's current CPU number 2462 int mCpukHz; // frequency of thread's current CPU in kHz 2463#endif 2464}; 2465 2466CpuStats::CpuStats() 2467#ifdef DEBUG_CPU_USAGE 2468 : mCpuNum(-1), mCpukHz(-1) 2469#endif 2470{ 2471} 2472 2473void CpuStats::sample(const String8 &title) { 2474#ifdef DEBUG_CPU_USAGE 2475 // get current thread's delta CPU time in wall clock ns 2476 double wcNs; 2477 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2478 2479 // record sample for wall clock statistics 2480 if (valid) { 2481 mWcStats.sample(wcNs); 2482 } 2483 2484 // get the current CPU number 2485 int cpuNum = sched_getcpu(); 2486 2487 // get the current CPU frequency in kHz 2488 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2489 2490 // check if either CPU number or frequency changed 2491 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2492 mCpuNum = cpuNum; 2493 mCpukHz = cpukHz; 2494 // ignore sample for purposes of cycles 2495 valid = false; 2496 } 2497 2498 // if no change in CPU number or frequency, then record sample for cycle statistics 2499 if (valid && mCpukHz > 0) { 2500 double cycles = wcNs * cpukHz * 0.000001; 2501 mHzStats.sample(cycles); 2502 } 2503 2504 unsigned n = mWcStats.n(); 2505 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2506 if ((n & 127) == 1) { 2507 long long elapsed = mCpuUsage.elapsed(); 2508 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2509 double perLoop = elapsed / (double) n; 2510 double perLoop100 = perLoop * 0.01; 2511 double perLoop1k = perLoop * 0.001; 2512 double mean = mWcStats.mean(); 2513 double stddev = mWcStats.stddev(); 2514 double minimum = mWcStats.minimum(); 2515 double maximum = mWcStats.maximum(); 2516 double meanCycles = mHzStats.mean(); 2517 double stddevCycles = mHzStats.stddev(); 2518 double minCycles = mHzStats.minimum(); 2519 double maxCycles = mHzStats.maximum(); 2520 mCpuUsage.resetElapsed(); 2521 mWcStats.reset(); 2522 mHzStats.reset(); 2523 ALOGD("CPU usage for %s over past %.1f secs\n" 2524 " (%u mixer loops at %.1f mean ms per loop):\n" 2525 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2526 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2527 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2528 title.string(), 2529 elapsed * .000000001, n, perLoop * .000001, 2530 mean * .001, 2531 stddev * .001, 2532 minimum * .001, 2533 maximum * .001, 2534 mean / perLoop100, 2535 stddev / perLoop100, 2536 minimum / perLoop100, 2537 maximum / perLoop100, 2538 meanCycles / perLoop1k, 2539 stddevCycles / perLoop1k, 2540 minCycles / perLoop1k, 2541 maxCycles / perLoop1k); 2542 2543 } 2544 } 2545#endif 2546}; 2547 2548void AudioFlinger::PlaybackThread::checkSilentMode_l() 2549{ 2550 if (!mMasterMute) { 2551 char value[PROPERTY_VALUE_MAX]; 2552 if (property_get("ro.audio.silent", value, "0") > 0) { 2553 char *endptr; 2554 unsigned long ul = strtoul(value, &endptr, 0); 2555 if (*endptr == '\0' && ul != 0) { 2556 ALOGD("Silence is golden"); 2557 // The setprop command will not allow a property to be changed after 2558 // the first time it is set, so we don't have to worry about un-muting. 2559 setMasterMute_l(true); 2560 } 2561 } 2562 } 2563} 2564 2565bool AudioFlinger::PlaybackThread::threadLoop() 2566{ 2567 Vector< sp<Track> > tracksToRemove; 2568 2569 standbyTime = systemTime(); 2570 2571 // MIXER 2572 nsecs_t lastWarning = 0; 2573 2574 // DUPLICATING 2575 // FIXME could this be made local to while loop? 2576 writeFrames = 0; 2577 2578 cacheParameters_l(); 2579 sleepTime = idleSleepTime; 2580 2581 if (mType == MIXER) { 2582 sleepTimeShift = 0; 2583 } 2584 2585 CpuStats cpuStats; 2586 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2587 2588 acquireWakeLock(); 2589 2590 while (!exitPending()) 2591 { 2592 cpuStats.sample(myName); 2593 2594 Vector< sp<EffectChain> > effectChains; 2595 2596 processConfigEvents(); 2597 2598 { // scope for mLock 2599 2600 Mutex::Autolock _l(mLock); 2601 2602 if (checkForNewParameters_l()) { 2603 cacheParameters_l(); 2604 } 2605 2606 saveOutputTracks(); 2607 2608 // put audio hardware into standby after short delay 2609 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2610 isSuspended())) { 2611 if (!mStandby) { 2612 2613 threadLoop_standby(); 2614 2615 mStandby = true; 2616 } 2617 2618 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2619 // we're about to wait, flush the binder command buffer 2620 IPCThreadState::self()->flushCommands(); 2621 2622 clearOutputTracks(); 2623 2624 if (exitPending()) break; 2625 2626 releaseWakeLock_l(); 2627 // wait until we have something to do... 2628 ALOGV("%s going to sleep", myName.string()); 2629 mWaitWorkCV.wait(mLock); 2630 ALOGV("%s waking up", myName.string()); 2631 acquireWakeLock_l(); 2632 2633 mMixerStatus = MIXER_IDLE; 2634 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2635 mBytesWritten = 0; 2636 2637 checkSilentMode_l(); 2638 2639 standbyTime = systemTime() + standbyDelay; 2640 sleepTime = idleSleepTime; 2641 if (mType == MIXER) { 2642 sleepTimeShift = 0; 2643 } 2644 2645 continue; 2646 } 2647 } 2648 2649 // mMixerStatusIgnoringFastTracks is also updated internally 2650 mMixerStatus = prepareTracks_l(&tracksToRemove); 2651 2652 // prevent any changes in effect chain list and in each effect chain 2653 // during mixing and effect process as the audio buffers could be deleted 2654 // or modified if an effect is created or deleted 2655 lockEffectChains_l(effectChains); 2656 } 2657 2658 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2659 threadLoop_mix(); 2660 } else { 2661 threadLoop_sleepTime(); 2662 } 2663 2664 if (isSuspended()) { 2665 sleepTime = suspendSleepTimeUs(); 2666 mBytesWritten += mixBufferSize; 2667 } 2668 2669 // only process effects if we're going to write 2670 if (sleepTime == 0) { 2671 for (size_t i = 0; i < effectChains.size(); i ++) { 2672 effectChains[i]->process_l(); 2673 } 2674 } 2675 2676 // enable changes in effect chain 2677 unlockEffectChains(effectChains); 2678 2679 // sleepTime == 0 means we must write to audio hardware 2680 if (sleepTime == 0) { 2681 2682 threadLoop_write(); 2683 2684if (mType == MIXER) { 2685 // write blocked detection 2686 nsecs_t now = systemTime(); 2687 nsecs_t delta = now - mLastWriteTime; 2688 if (!mStandby && delta > maxPeriod) { 2689 mNumDelayedWrites++; 2690 if ((now - lastWarning) > kWarningThrottleNs) { 2691#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2692 ScopedTrace st(ATRACE_TAG, "underrun"); 2693#endif 2694 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2695 ns2ms(delta), mNumDelayedWrites, this); 2696 lastWarning = now; 2697 } 2698 } 2699} 2700 2701 mStandby = false; 2702 } else { 2703 usleep(sleepTime); 2704 } 2705 2706 // Finally let go of removed track(s), without the lock held 2707 // since we can't guarantee the destructors won't acquire that 2708 // same lock. This will also mutate and push a new fast mixer state. 2709 threadLoop_removeTracks(tracksToRemove); 2710 tracksToRemove.clear(); 2711 2712 // FIXME I don't understand the need for this here; 2713 // it was in the original code but maybe the 2714 // assignment in saveOutputTracks() makes this unnecessary? 2715 clearOutputTracks(); 2716 2717 // Effect chains will be actually deleted here if they were removed from 2718 // mEffectChains list during mixing or effects processing 2719 effectChains.clear(); 2720 2721 // FIXME Note that the above .clear() is no longer necessary since effectChains 2722 // is now local to this block, but will keep it for now (at least until merge done). 2723 } 2724 2725 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2726 if (mType == MIXER || mType == DIRECT) { 2727 // put output stream into standby mode 2728 if (!mStandby) { 2729 mOutput->stream->common.standby(&mOutput->stream->common); 2730 } 2731 } 2732 2733 releaseWakeLock(); 2734 2735 ALOGV("Thread %p type %d exiting", this, mType); 2736 return false; 2737} 2738 2739void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2740{ 2741 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2742} 2743 2744void AudioFlinger::MixerThread::threadLoop_write() 2745{ 2746 // FIXME we should only do one push per cycle; confirm this is true 2747 // Start the fast mixer if it's not already running 2748 if (mFastMixer != NULL) { 2749 FastMixerStateQueue *sq = mFastMixer->sq(); 2750 FastMixerState *state = sq->begin(); 2751 if (state->mCommand != FastMixerState::MIX_WRITE && 2752 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2753 if (state->mCommand == FastMixerState::COLD_IDLE) { 2754 int32_t old = android_atomic_inc(&mFastMixerFutex); 2755 if (old == -1) { 2756 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2757 } 2758#ifdef AUDIO_WATCHDOG 2759 if (mAudioWatchdog != 0) { 2760 mAudioWatchdog->resume(); 2761 } 2762#endif 2763 } 2764 state->mCommand = FastMixerState::MIX_WRITE; 2765 sq->end(); 2766 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2767 if (kUseFastMixer == FastMixer_Dynamic) { 2768 mNormalSink = mPipeSink; 2769 } 2770 } else { 2771 sq->end(false /*didModify*/); 2772 } 2773 } 2774 PlaybackThread::threadLoop_write(); 2775} 2776 2777// shared by MIXER and DIRECT, overridden by DUPLICATING 2778void AudioFlinger::PlaybackThread::threadLoop_write() 2779{ 2780 // FIXME rewrite to reduce number of system calls 2781 mLastWriteTime = systemTime(); 2782 mInWrite = true; 2783 int bytesWritten; 2784 2785 // If an NBAIO sink is present, use it to write the normal mixer's submix 2786 if (mNormalSink != 0) { 2787#define mBitShift 2 // FIXME 2788 size_t count = mixBufferSize >> mBitShift; 2789#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2790 Tracer::traceBegin(ATRACE_TAG, "write"); 2791#endif 2792 // update the setpoint when gScreenState changes 2793 uint32_t screenState = gScreenState; 2794 if (screenState != mScreenState) { 2795 mScreenState = screenState; 2796 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2797 if (pipe != NULL) { 2798 pipe->setAvgFrames((mScreenState & 1) ? 2799 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2800 } 2801 } 2802 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2803#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2804 Tracer::traceEnd(ATRACE_TAG); 2805#endif 2806 if (framesWritten > 0) { 2807 bytesWritten = framesWritten << mBitShift; 2808 } else { 2809 bytesWritten = framesWritten; 2810 } 2811 // otherwise use the HAL / AudioStreamOut directly 2812 } else { 2813 // Direct output thread. 2814 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2815 } 2816 2817 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2818 mNumWrites++; 2819 mInWrite = false; 2820} 2821 2822void AudioFlinger::MixerThread::threadLoop_standby() 2823{ 2824 // Idle the fast mixer if it's currently running 2825 if (mFastMixer != NULL) { 2826 FastMixerStateQueue *sq = mFastMixer->sq(); 2827 FastMixerState *state = sq->begin(); 2828 if (!(state->mCommand & FastMixerState::IDLE)) { 2829 state->mCommand = FastMixerState::COLD_IDLE; 2830 state->mColdFutexAddr = &mFastMixerFutex; 2831 state->mColdGen++; 2832 mFastMixerFutex = 0; 2833 sq->end(); 2834 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2835 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2836 if (kUseFastMixer == FastMixer_Dynamic) { 2837 mNormalSink = mOutputSink; 2838 } 2839#ifdef AUDIO_WATCHDOG 2840 if (mAudioWatchdog != 0) { 2841 mAudioWatchdog->pause(); 2842 } 2843#endif 2844 } else { 2845 sq->end(false /*didModify*/); 2846 } 2847 } 2848 PlaybackThread::threadLoop_standby(); 2849} 2850 2851// shared by MIXER and DIRECT, overridden by DUPLICATING 2852void AudioFlinger::PlaybackThread::threadLoop_standby() 2853{ 2854 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2855 mOutput->stream->common.standby(&mOutput->stream->common); 2856} 2857 2858void AudioFlinger::MixerThread::threadLoop_mix() 2859{ 2860 // obtain the presentation timestamp of the next output buffer 2861 int64_t pts; 2862 status_t status = INVALID_OPERATION; 2863 2864 if (mNormalSink != 0) { 2865 status = mNormalSink->getNextWriteTimestamp(&pts); 2866 } else { 2867 status = mOutputSink->getNextWriteTimestamp(&pts); 2868 } 2869 2870 if (status != NO_ERROR) { 2871 pts = AudioBufferProvider::kInvalidPTS; 2872 } 2873 2874 // mix buffers... 2875 mAudioMixer->process(pts); 2876 // increase sleep time progressively when application underrun condition clears. 2877 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2878 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2879 // such that we would underrun the audio HAL. 2880 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2881 sleepTimeShift--; 2882 } 2883 sleepTime = 0; 2884 standbyTime = systemTime() + standbyDelay; 2885 //TODO: delay standby when effects have a tail 2886} 2887 2888void AudioFlinger::MixerThread::threadLoop_sleepTime() 2889{ 2890 // If no tracks are ready, sleep once for the duration of an output 2891 // buffer size, then write 0s to the output 2892 if (sleepTime == 0) { 2893 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2894 sleepTime = activeSleepTime >> sleepTimeShift; 2895 if (sleepTime < kMinThreadSleepTimeUs) { 2896 sleepTime = kMinThreadSleepTimeUs; 2897 } 2898 // reduce sleep time in case of consecutive application underruns to avoid 2899 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2900 // duration we would end up writing less data than needed by the audio HAL if 2901 // the condition persists. 2902 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2903 sleepTimeShift++; 2904 } 2905 } else { 2906 sleepTime = idleSleepTime; 2907 } 2908 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2909 memset (mMixBuffer, 0, mixBufferSize); 2910 sleepTime = 0; 2911 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), 2912 "anticipated start"); 2913 } 2914 // TODO add standby time extension fct of effect tail 2915} 2916 2917// prepareTracks_l() must be called with ThreadBase::mLock held 2918AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2919 Vector< sp<Track> > *tracksToRemove) 2920{ 2921 2922 mixer_state mixerStatus = MIXER_IDLE; 2923 // find out which tracks need to be processed 2924 size_t count = mActiveTracks.size(); 2925 size_t mixedTracks = 0; 2926 size_t tracksWithEffect = 0; 2927 // counts only _active_ fast tracks 2928 size_t fastTracks = 0; 2929 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2930 2931 float masterVolume = mMasterVolume; 2932 bool masterMute = mMasterMute; 2933 2934 if (masterMute) { 2935 masterVolume = 0; 2936 } 2937 // Delegate master volume control to effect in output mix effect chain if needed 2938 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2939 if (chain != 0) { 2940 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2941 chain->setVolume_l(&v, &v); 2942 masterVolume = (float)((v + (1 << 23)) >> 24); 2943 chain.clear(); 2944 } 2945 2946 // prepare a new state to push 2947 FastMixerStateQueue *sq = NULL; 2948 FastMixerState *state = NULL; 2949 bool didModify = false; 2950 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2951 if (mFastMixer != NULL) { 2952 sq = mFastMixer->sq(); 2953 state = sq->begin(); 2954 } 2955 2956 for (size_t i=0 ; i<count ; i++) { 2957 sp<Track> t = mActiveTracks[i].promote(); 2958 if (t == 0) continue; 2959 2960 // this const just means the local variable doesn't change 2961 Track* const track = t.get(); 2962 2963 // process fast tracks 2964 if (track->isFastTrack()) { 2965 2966 // It's theoretically possible (though unlikely) for a fast track to be created 2967 // and then removed within the same normal mix cycle. This is not a problem, as 2968 // the track never becomes active so it's fast mixer slot is never touched. 2969 // The converse, of removing an (active) track and then creating a new track 2970 // at the identical fast mixer slot within the same normal mix cycle, 2971 // is impossible because the slot isn't marked available until the end of each cycle. 2972 int j = track->mFastIndex; 2973 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2974 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2975 FastTrack *fastTrack = &state->mFastTracks[j]; 2976 2977 // Determine whether the track is currently in underrun condition, 2978 // and whether it had a recent underrun. 2979 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2980 FastTrackUnderruns underruns = ftDump->mUnderruns; 2981 uint32_t recentFull = (underruns.mBitFields.mFull - 2982 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2983 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2984 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2985 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2986 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2987 uint32_t recentUnderruns = recentPartial + recentEmpty; 2988 track->mObservedUnderruns = underruns; 2989 // don't count underruns that occur while stopping or pausing 2990 // or stopped which can occur when flush() is called while active 2991 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2992 track->mUnderrunCount += recentUnderruns; 2993 } 2994 2995 // This is similar to the state machine for normal tracks, 2996 // with a few modifications for fast tracks. 2997 bool isActive = true; 2998 switch (track->mState) { 2999 case TrackBase::STOPPING_1: 3000 // track stays active in STOPPING_1 state until first underrun 3001 if (recentUnderruns > 0) { 3002 track->mState = TrackBase::STOPPING_2; 3003 } 3004 break; 3005 case TrackBase::PAUSING: 3006 // ramp down is not yet implemented 3007 track->setPaused(); 3008 break; 3009 case TrackBase::RESUMING: 3010 // ramp up is not yet implemented 3011 track->mState = TrackBase::ACTIVE; 3012 break; 3013 case TrackBase::ACTIVE: 3014 if (recentFull > 0 || recentPartial > 0) { 3015 // track has provided at least some frames recently: reset retry count 3016 track->mRetryCount = kMaxTrackRetries; 3017 } 3018 if (recentUnderruns == 0) { 3019 // no recent underruns: stay active 3020 break; 3021 } 3022 // there has recently been an underrun of some kind 3023 if (track->sharedBuffer() == 0) { 3024 // were any of the recent underruns "empty" (no frames available)? 3025 if (recentEmpty == 0) { 3026 // no, then ignore the partial underruns as they are allowed indefinitely 3027 break; 3028 } 3029 // there has recently been an "empty" underrun: decrement the retry counter 3030 if (--(track->mRetryCount) > 0) { 3031 break; 3032 } 3033 // indicate to client process that the track was disabled because of underrun; 3034 // it will then automatically call start() when data is available 3035 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 3036 // remove from active list, but state remains ACTIVE [confusing but true] 3037 isActive = false; 3038 break; 3039 } 3040 // fall through 3041 case TrackBase::STOPPING_2: 3042 case TrackBase::PAUSED: 3043 case TrackBase::TERMINATED: 3044 case TrackBase::STOPPED: 3045 case TrackBase::FLUSHED: // flush() while active 3046 // Check for presentation complete if track is inactive 3047 // We have consumed all the buffers of this track. 3048 // This would be incomplete if we auto-paused on underrun 3049 { 3050 size_t audioHALFrames = 3051 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3052 size_t framesWritten = 3053 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3054 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3055 // track stays in active list until presentation is complete 3056 break; 3057 } 3058 } 3059 if (track->isStopping_2()) { 3060 track->mState = TrackBase::STOPPED; 3061 } 3062 if (track->isStopped()) { 3063 // Can't reset directly, as fast mixer is still polling this track 3064 // track->reset(); 3065 // So instead mark this track as needing to be reset after push with ack 3066 resetMask |= 1 << i; 3067 } 3068 isActive = false; 3069 break; 3070 case TrackBase::IDLE: 3071 default: 3072 LOG_FATAL("unexpected track state %d", track->mState); 3073 } 3074 3075 if (isActive) { 3076 // was it previously inactive? 3077 if (!(state->mTrackMask & (1 << j))) { 3078 ExtendedAudioBufferProvider *eabp = track; 3079 VolumeProvider *vp = track; 3080 fastTrack->mBufferProvider = eabp; 3081 fastTrack->mVolumeProvider = vp; 3082 fastTrack->mSampleRate = track->mSampleRate; 3083 fastTrack->mChannelMask = track->mChannelMask; 3084 fastTrack->mGeneration++; 3085 state->mTrackMask |= 1 << j; 3086 didModify = true; 3087 // no acknowledgement required for newly active tracks 3088 } 3089 // cache the combined master volume and stream type volume for fast mixer; this 3090 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3091 track->mCachedVolume = track->isMuted() ? 3092 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3093 ++fastTracks; 3094 } else { 3095 // was it previously active? 3096 if (state->mTrackMask & (1 << j)) { 3097 fastTrack->mBufferProvider = NULL; 3098 fastTrack->mGeneration++; 3099 state->mTrackMask &= ~(1 << j); 3100 didModify = true; 3101 // If any fast tracks were removed, we must wait for acknowledgement 3102 // because we're about to decrement the last sp<> on those tracks. 3103 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3104 } else { 3105 LOG_FATAL("fast track %d should have been active", j); 3106 } 3107 tracksToRemove->add(track); 3108 // Avoids a misleading display in dumpsys 3109 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3110 } 3111 continue; 3112 } 3113 3114 { // local variable scope to avoid goto warning 3115 3116 audio_track_cblk_t* cblk = track->cblk(); 3117 3118 // The first time a track is added we wait 3119 // for all its buffers to be filled before processing it 3120 int name = track->name(); 3121 // make sure that we have enough frames to mix one full buffer. 3122 // enforce this condition only once to enable draining the buffer in case the client 3123 // app does not call stop() and relies on underrun to stop: 3124 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3125 // during last round 3126 uint32_t minFrames = 1; 3127 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3128 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3129 if (t->sampleRate() == mSampleRate) { 3130 minFrames = mNormalFrameCount; 3131 } else { 3132 // +1 for rounding and +1 for additional sample needed for interpolation 3133 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3134 // add frames already consumed but not yet released by the resampler 3135 // because cblk->framesReady() will include these frames 3136 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3137 // the minimum track buffer size is normally twice the number of frames necessary 3138 // to fill one buffer and the resampler should not leave more than one buffer worth 3139 // of unreleased frames after each pass, but just in case... 3140 ALOG_ASSERT(minFrames <= cblk->frameCount); 3141 } 3142 } 3143 if ((track->framesReady() >= minFrames) && track->isReady() && 3144 !track->isPaused() && !track->isTerminated()) 3145 { 3146 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 3147 this); 3148 3149 mixedTracks++; 3150 3151 // track->mainBuffer() != mMixBuffer means there is an effect chain 3152 // connected to the track 3153 chain.clear(); 3154 if (track->mainBuffer() != mMixBuffer) { 3155 chain = getEffectChain_l(track->sessionId()); 3156 // Delegate volume control to effect in track effect chain if needed 3157 if (chain != 0) { 3158 tracksWithEffect++; 3159 } else { 3160 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3161 "session %d", 3162 name, track->sessionId()); 3163 } 3164 } 3165 3166 3167 int param = AudioMixer::VOLUME; 3168 if (track->mFillingUpStatus == Track::FS_FILLED) { 3169 // no ramp for the first volume setting 3170 track->mFillingUpStatus = Track::FS_ACTIVE; 3171 if (track->mState == TrackBase::RESUMING) { 3172 track->mState = TrackBase::ACTIVE; 3173 param = AudioMixer::RAMP_VOLUME; 3174 } 3175 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3176 } else if (cblk->server != 0) { 3177 // If the track is stopped before the first frame was mixed, 3178 // do not apply ramp 3179 param = AudioMixer::RAMP_VOLUME; 3180 } 3181 3182 // compute volume for this track 3183 uint32_t vl, vr, va; 3184 if (track->isMuted() || track->isPausing() || 3185 mStreamTypes[track->streamType()].mute) { 3186 vl = vr = va = 0; 3187 if (track->isPausing()) { 3188 track->setPaused(); 3189 } 3190 } else { 3191 3192 // read original volumes with volume control 3193 float typeVolume = mStreamTypes[track->streamType()].volume; 3194 float v = masterVolume * typeVolume; 3195 uint32_t vlr = cblk->getVolumeLR(); 3196 vl = vlr & 0xFFFF; 3197 vr = vlr >> 16; 3198 // track volumes come from shared memory, so can't be trusted and must be clamped 3199 if (vl > MAX_GAIN_INT) { 3200 ALOGV("Track left volume out of range: %04X", vl); 3201 vl = MAX_GAIN_INT; 3202 } 3203 if (vr > MAX_GAIN_INT) { 3204 ALOGV("Track right volume out of range: %04X", vr); 3205 vr = MAX_GAIN_INT; 3206 } 3207 // now apply the master volume and stream type volume 3208 vl = (uint32_t)(v * vl) << 12; 3209 vr = (uint32_t)(v * vr) << 12; 3210 // assuming master volume and stream type volume each go up to 1.0, 3211 // vl and vr are now in 8.24 format 3212 3213 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3214 // send level comes from shared memory and so may be corrupt 3215 if (sendLevel > MAX_GAIN_INT) { 3216 ALOGV("Track send level out of range: %04X", sendLevel); 3217 sendLevel = MAX_GAIN_INT; 3218 } 3219 va = (uint32_t)(v * sendLevel); 3220 } 3221 // Delegate volume control to effect in track effect chain if needed 3222 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3223 // Do not ramp volume if volume is controlled by effect 3224 param = AudioMixer::VOLUME; 3225 track->mHasVolumeController = true; 3226 } else { 3227 // force no volume ramp when volume controller was just disabled or removed 3228 // from effect chain to avoid volume spike 3229 if (track->mHasVolumeController) { 3230 param = AudioMixer::VOLUME; 3231 } 3232 track->mHasVolumeController = false; 3233 } 3234 3235 // Convert volumes from 8.24 to 4.12 format 3236 // This additional clamping is needed in case chain->setVolume_l() overshot 3237 vl = (vl + (1 << 11)) >> 12; 3238 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3239 vr = (vr + (1 << 11)) >> 12; 3240 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3241 3242 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3243 3244 // XXX: these things DON'T need to be done each time 3245 mAudioMixer->setBufferProvider(name, track); 3246 mAudioMixer->enable(name); 3247 3248 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3249 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3250 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3251 mAudioMixer->setParameter( 3252 name, 3253 AudioMixer::TRACK, 3254 AudioMixer::FORMAT, (void *)track->format()); 3255 mAudioMixer->setParameter( 3256 name, 3257 AudioMixer::TRACK, 3258 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3259 mAudioMixer->setParameter( 3260 name, 3261 AudioMixer::RESAMPLE, 3262 AudioMixer::SAMPLE_RATE, 3263 (void *)(cblk->sampleRate)); 3264 mAudioMixer->setParameter( 3265 name, 3266 AudioMixer::TRACK, 3267 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3268 mAudioMixer->setParameter( 3269 name, 3270 AudioMixer::TRACK, 3271 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3272 3273 // reset retry count 3274 track->mRetryCount = kMaxTrackRetries; 3275 3276 // If one track is ready, set the mixer ready if: 3277 // - the mixer was not ready during previous round OR 3278 // - no other track is not ready 3279 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3280 mixerStatus != MIXER_TRACKS_ENABLED) { 3281 mixerStatus = MIXER_TRACKS_READY; 3282 } 3283 } else { 3284 // clear effect chain input buffer if an active track underruns to avoid sending 3285 // previous audio buffer again to effects 3286 chain = getEffectChain_l(track->sessionId()); 3287 if (chain != 0) { 3288 chain->clearInputBuffer(); 3289 } 3290 3291 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 3292 cblk->server, this); 3293 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3294 track->isStopped() || track->isPaused()) { 3295 // We have consumed all the buffers of this track. 3296 // Remove it from the list of active tracks. 3297 // TODO: use actual buffer filling status instead of latency when available from 3298 // audio HAL 3299 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3300 size_t framesWritten = 3301 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3302 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3303 if (track->isStopped()) { 3304 track->reset(); 3305 } 3306 tracksToRemove->add(track); 3307 } 3308 } else { 3309 track->mUnderrunCount++; 3310 // No buffers for this track. Give it a few chances to 3311 // fill a buffer, then remove it from active list. 3312 if (--(track->mRetryCount) <= 0) { 3313 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3314 tracksToRemove->add(track); 3315 // indicate to client process that the track was disabled because of underrun; 3316 // it will then automatically call start() when data is available 3317 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3318 // If one track is not ready, mark the mixer also not ready if: 3319 // - the mixer was ready during previous round OR 3320 // - no other track is ready 3321 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3322 mixerStatus != MIXER_TRACKS_READY) { 3323 mixerStatus = MIXER_TRACKS_ENABLED; 3324 } 3325 } 3326 mAudioMixer->disable(name); 3327 } 3328 3329 } // local variable scope to avoid goto warning 3330track_is_ready: ; 3331 3332 } 3333 3334 // Push the new FastMixer state if necessary 3335 bool pauseAudioWatchdog = false; 3336 if (didModify) { 3337 state->mFastTracksGen++; 3338 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3339 if (kUseFastMixer == FastMixer_Dynamic && 3340 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3341 state->mCommand = FastMixerState::COLD_IDLE; 3342 state->mColdFutexAddr = &mFastMixerFutex; 3343 state->mColdGen++; 3344 mFastMixerFutex = 0; 3345 if (kUseFastMixer == FastMixer_Dynamic) { 3346 mNormalSink = mOutputSink; 3347 } 3348 // If we go into cold idle, need to wait for acknowledgement 3349 // so that fast mixer stops doing I/O. 3350 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3351 pauseAudioWatchdog = true; 3352 } 3353 sq->end(); 3354 } 3355 if (sq != NULL) { 3356 sq->end(didModify); 3357 sq->push(block); 3358 } 3359#ifdef AUDIO_WATCHDOG 3360 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3361 mAudioWatchdog->pause(); 3362 } 3363#endif 3364 3365 // Now perform the deferred reset on fast tracks that have stopped 3366 while (resetMask != 0) { 3367 size_t i = __builtin_ctz(resetMask); 3368 ALOG_ASSERT(i < count); 3369 resetMask &= ~(1 << i); 3370 sp<Track> t = mActiveTracks[i].promote(); 3371 if (t == 0) continue; 3372 Track* track = t.get(); 3373 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3374 track->reset(); 3375 } 3376 3377 // remove all the tracks that need to be... 3378 count = tracksToRemove->size(); 3379 if (CC_UNLIKELY(count)) { 3380 for (size_t i=0 ; i<count ; i++) { 3381 const sp<Track>& track = tracksToRemove->itemAt(i); 3382 mActiveTracks.remove(track); 3383 if (track->mainBuffer() != mMixBuffer) { 3384 chain = getEffectChain_l(track->sessionId()); 3385 if (chain != 0) { 3386 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3387 track->sessionId()); 3388 chain->decActiveTrackCnt(); 3389 } 3390 } 3391 if (track->isTerminated()) { 3392 removeTrack_l(track); 3393 } 3394 } 3395 } 3396 3397 // mix buffer must be cleared if all tracks are connected to an 3398 // effect chain as in this case the mixer will not write to 3399 // mix buffer and track effects will accumulate into it 3400 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3401 (mixedTracks == 0 && fastTracks > 0)) { 3402 // FIXME as a performance optimization, should remember previous zero status 3403 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3404 } 3405 3406 // if any fast tracks, then status is ready 3407 mMixerStatusIgnoringFastTracks = mixerStatus; 3408 if (fastTracks > 0) { 3409 mixerStatus = MIXER_TRACKS_READY; 3410 } 3411 return mixerStatus; 3412} 3413 3414/* 3415The derived values that are cached: 3416 - mixBufferSize from frame count * frame size 3417 - activeSleepTime from activeSleepTimeUs() 3418 - idleSleepTime from idleSleepTimeUs() 3419 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3420 - maxPeriod from frame count and sample rate (MIXER only) 3421 3422The parameters that affect these derived values are: 3423 - frame count 3424 - frame size 3425 - sample rate 3426 - device type: A2DP or not 3427 - device latency 3428 - format: PCM or not 3429 - active sleep time 3430 - idle sleep time 3431*/ 3432 3433void AudioFlinger::PlaybackThread::cacheParameters_l() 3434{ 3435 mixBufferSize = mNormalFrameCount * mFrameSize; 3436 activeSleepTime = activeSleepTimeUs(); 3437 idleSleepTime = idleSleepTimeUs(); 3438} 3439 3440void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3441{ 3442 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3443 this, streamType, mTracks.size()); 3444 Mutex::Autolock _l(mLock); 3445 3446 size_t size = mTracks.size(); 3447 for (size_t i = 0; i < size; i++) { 3448 sp<Track> t = mTracks[i]; 3449 if (t->streamType() == streamType) { 3450 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 3451 t->mCblk->cv.signal(); 3452 } 3453 } 3454} 3455 3456// getTrackName_l() must be called with ThreadBase::mLock held 3457int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3458{ 3459 return mAudioMixer->getTrackName(channelMask, sessionId); 3460} 3461 3462// deleteTrackName_l() must be called with ThreadBase::mLock held 3463void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3464{ 3465 ALOGV("remove track (%d) and delete from mixer", name); 3466 mAudioMixer->deleteTrackName(name); 3467} 3468 3469// checkForNewParameters_l() must be called with ThreadBase::mLock held 3470bool AudioFlinger::MixerThread::checkForNewParameters_l() 3471{ 3472 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3473 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3474 bool reconfig = false; 3475 3476 while (!mNewParameters.isEmpty()) { 3477 3478 if (mFastMixer != NULL) { 3479 FastMixerStateQueue *sq = mFastMixer->sq(); 3480 FastMixerState *state = sq->begin(); 3481 if (!(state->mCommand & FastMixerState::IDLE)) { 3482 previousCommand = state->mCommand; 3483 state->mCommand = FastMixerState::HOT_IDLE; 3484 sq->end(); 3485 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3486 } else { 3487 sq->end(false /*didModify*/); 3488 } 3489 } 3490 3491 status_t status = NO_ERROR; 3492 String8 keyValuePair = mNewParameters[0]; 3493 AudioParameter param = AudioParameter(keyValuePair); 3494 int value; 3495 3496 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3497 reconfig = true; 3498 } 3499 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3500 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3501 status = BAD_VALUE; 3502 } else { 3503 reconfig = true; 3504 } 3505 } 3506 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3507 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3508 status = BAD_VALUE; 3509 } else { 3510 reconfig = true; 3511 } 3512 } 3513 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3514 // do not accept frame count changes if tracks are open as the track buffer 3515 // size depends on frame count and correct behavior would not be guaranteed 3516 // if frame count is changed after track creation 3517 if (!mTracks.isEmpty()) { 3518 status = INVALID_OPERATION; 3519 } else { 3520 reconfig = true; 3521 } 3522 } 3523 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3524#ifdef ADD_BATTERY_DATA 3525 // when changing the audio output device, call addBatteryData to notify 3526 // the change 3527 if (mOutDevice != value) { 3528 uint32_t params = 0; 3529 // check whether speaker is on 3530 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3531 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3532 } 3533 3534 audio_devices_t deviceWithoutSpeaker 3535 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3536 // check if any other device (except speaker) is on 3537 if (value & deviceWithoutSpeaker ) { 3538 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3539 } 3540 3541 if (params != 0) { 3542 addBatteryData(params); 3543 } 3544 } 3545#endif 3546 3547 // forward device change to effects that have requested to be 3548 // aware of attached audio device. 3549 mOutDevice = value; 3550 for (size_t i = 0; i < mEffectChains.size(); i++) { 3551 mEffectChains[i]->setDevice_l(mOutDevice); 3552 } 3553 } 3554 3555 if (status == NO_ERROR) { 3556 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3557 keyValuePair.string()); 3558 if (!mStandby && status == INVALID_OPERATION) { 3559 mOutput->stream->common.standby(&mOutput->stream->common); 3560 mStandby = true; 3561 mBytesWritten = 0; 3562 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3563 keyValuePair.string()); 3564 } 3565 if (status == NO_ERROR && reconfig) { 3566 delete mAudioMixer; 3567 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3568 mAudioMixer = NULL; 3569 readOutputParameters(); 3570 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3571 for (size_t i = 0; i < mTracks.size() ; i++) { 3572 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3573 if (name < 0) break; 3574 mTracks[i]->mName = name; 3575 // limit track sample rate to 2 x new output sample rate 3576 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3577 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3578 } 3579 } 3580 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3581 } 3582 } 3583 3584 mNewParameters.removeAt(0); 3585 3586 mParamStatus = status; 3587 mParamCond.signal(); 3588 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3589 // already timed out waiting for the status and will never signal the condition. 3590 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3591 } 3592 3593 if (!(previousCommand & FastMixerState::IDLE)) { 3594 ALOG_ASSERT(mFastMixer != NULL); 3595 FastMixerStateQueue *sq = mFastMixer->sq(); 3596 FastMixerState *state = sq->begin(); 3597 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3598 state->mCommand = previousCommand; 3599 sq->end(); 3600 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3601 } 3602 3603 return reconfig; 3604} 3605 3606void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3607{ 3608 NBAIO_Source *teeSource = source.get(); 3609 if (teeSource != NULL) { 3610 char teeTime[16]; 3611 struct timeval tv; 3612 gettimeofday(&tv, NULL); 3613 struct tm tm; 3614 localtime_r(&tv.tv_sec, &tm); 3615 strftime(teeTime, sizeof(teeTime), "%T", &tm); 3616 char teePath[64]; 3617 sprintf(teePath, "/data/misc/media/%s_%d.wav", teeTime, id); 3618 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3619 if (teeFd >= 0) { 3620 char wavHeader[44]; 3621 memcpy(wavHeader, 3622 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3623 sizeof(wavHeader)); 3624 NBAIO_Format format = teeSource->format(); 3625 unsigned channelCount = Format_channelCount(format); 3626 ALOG_ASSERT(channelCount <= FCC_2); 3627 uint32_t sampleRate = Format_sampleRate(format); 3628 wavHeader[22] = channelCount; // number of channels 3629 wavHeader[24] = sampleRate; // sample rate 3630 wavHeader[25] = sampleRate >> 8; 3631 wavHeader[32] = channelCount * 2; // block alignment 3632 write(teeFd, wavHeader, sizeof(wavHeader)); 3633 size_t total = 0; 3634 bool firstRead = true; 3635 for (;;) { 3636#define TEE_SINK_READ 1024 3637 short buffer[TEE_SINK_READ * FCC_2]; 3638 size_t count = TEE_SINK_READ; 3639 ssize_t actual = teeSource->read(buffer, count, 3640 AudioBufferProvider::kInvalidPTS); 3641 bool wasFirstRead = firstRead; 3642 firstRead = false; 3643 if (actual <= 0) { 3644 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3645 continue; 3646 } 3647 break; 3648 } 3649 ALOG_ASSERT(actual <= (ssize_t)count); 3650 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3651 total += actual; 3652 } 3653 lseek(teeFd, (off_t) 4, SEEK_SET); 3654 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3655 write(teeFd, &temp, sizeof(temp)); 3656 lseek(teeFd, (off_t) 40, SEEK_SET); 3657 temp = total * channelCount * sizeof(short); 3658 write(teeFd, &temp, sizeof(temp)); 3659 close(teeFd); 3660 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3661 } else { 3662 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3663 } 3664 } 3665} 3666 3667void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3668{ 3669 const size_t SIZE = 256; 3670 char buffer[SIZE]; 3671 String8 result; 3672 3673 PlaybackThread::dumpInternals(fd, args); 3674 3675 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3676 result.append(buffer); 3677 write(fd, result.string(), result.size()); 3678 3679 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3680 FastMixerDumpState copy = mFastMixerDumpState; 3681 copy.dump(fd); 3682 3683#ifdef STATE_QUEUE_DUMP 3684 // Similar for state queue 3685 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3686 observerCopy.dump(fd); 3687 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3688 mutatorCopy.dump(fd); 3689#endif 3690 3691 // Write the tee output to a .wav file 3692 dumpTee(fd, mTeeSource, mId); 3693 3694#ifdef AUDIO_WATCHDOG 3695 if (mAudioWatchdog != 0) { 3696 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3697 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3698 wdCopy.dump(fd); 3699 } 3700#endif 3701} 3702 3703uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3704{ 3705 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3706} 3707 3708uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3709{ 3710 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3711} 3712 3713void AudioFlinger::MixerThread::cacheParameters_l() 3714{ 3715 PlaybackThread::cacheParameters_l(); 3716 3717 // FIXME: Relaxed timing because of a certain device that can't meet latency 3718 // Should be reduced to 2x after the vendor fixes the driver issue 3719 // increase threshold again due to low power audio mode. The way this warning 3720 // threshold is calculated and its usefulness should be reconsidered anyway. 3721 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3722} 3723 3724// ---------------------------------------------------------------------------- 3725AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3726 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3727 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3728 // mLeftVolFloat, mRightVolFloat 3729{ 3730} 3731 3732AudioFlinger::DirectOutputThread::~DirectOutputThread() 3733{ 3734} 3735 3736AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3737 Vector< sp<Track> > *tracksToRemove 3738) 3739{ 3740 sp<Track> trackToRemove; 3741 3742 mixer_state mixerStatus = MIXER_IDLE; 3743 3744 // find out which tracks need to be processed 3745 if (mActiveTracks.size() != 0) { 3746 sp<Track> t = mActiveTracks[0].promote(); 3747 // The track died recently 3748 if (t == 0) return MIXER_IDLE; 3749 3750 Track* const track = t.get(); 3751 audio_track_cblk_t* cblk = track->cblk(); 3752 3753 // The first time a track is added we wait 3754 // for all its buffers to be filled before processing it 3755 uint32_t minFrames; 3756 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3757 minFrames = mNormalFrameCount; 3758 } else { 3759 minFrames = 1; 3760 } 3761 if ((track->framesReady() >= minFrames) && track->isReady() && 3762 !track->isPaused() && !track->isTerminated()) 3763 { 3764 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3765 3766 if (track->mFillingUpStatus == Track::FS_FILLED) { 3767 track->mFillingUpStatus = Track::FS_ACTIVE; 3768 mLeftVolFloat = mRightVolFloat = 0; 3769 if (track->mState == TrackBase::RESUMING) { 3770 track->mState = TrackBase::ACTIVE; 3771 } 3772 } 3773 3774 // compute volume for this track 3775 float left, right; 3776 if (track->isMuted() || mMasterMute || track->isPausing() || 3777 mStreamTypes[track->streamType()].mute) { 3778 left = right = 0; 3779 if (track->isPausing()) { 3780 track->setPaused(); 3781 } 3782 } else { 3783 float typeVolume = mStreamTypes[track->streamType()].volume; 3784 float v = mMasterVolume * typeVolume; 3785 uint32_t vlr = cblk->getVolumeLR(); 3786 float v_clamped = v * (vlr & 0xFFFF); 3787 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3788 left = v_clamped/MAX_GAIN; 3789 v_clamped = v * (vlr >> 16); 3790 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3791 right = v_clamped/MAX_GAIN; 3792 } 3793 3794 if (left != mLeftVolFloat || right != mRightVolFloat) { 3795 mLeftVolFloat = left; 3796 mRightVolFloat = right; 3797 3798 // Convert volumes from float to 8.24 3799 uint32_t vl = (uint32_t)(left * (1 << 24)); 3800 uint32_t vr = (uint32_t)(right * (1 << 24)); 3801 3802 // Delegate volume control to effect in track effect chain if needed 3803 // only one effect chain can be present on DirectOutputThread, so if 3804 // there is one, the track is connected to it 3805 if (!mEffectChains.isEmpty()) { 3806 // Do not ramp volume if volume is controlled by effect 3807 mEffectChains[0]->setVolume_l(&vl, &vr); 3808 left = (float)vl / (1 << 24); 3809 right = (float)vr / (1 << 24); 3810 } 3811 mOutput->stream->set_volume(mOutput->stream, left, right); 3812 } 3813 3814 // reset retry count 3815 track->mRetryCount = kMaxTrackRetriesDirect; 3816 mActiveTrack = t; 3817 mixerStatus = MIXER_TRACKS_READY; 3818 } else { 3819 // clear effect chain input buffer if an active track underruns to avoid sending 3820 // previous audio buffer again to effects 3821 if (!mEffectChains.isEmpty()) { 3822 mEffectChains[0]->clearInputBuffer(); 3823 } 3824 3825 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3826 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3827 track->isStopped() || track->isPaused()) { 3828 // We have consumed all the buffers of this track. 3829 // Remove it from the list of active tracks. 3830 // TODO: implement behavior for compressed audio 3831 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3832 size_t framesWritten = 3833 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3834 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3835 if (track->isStopped()) { 3836 track->reset(); 3837 } 3838 trackToRemove = track; 3839 } 3840 } else { 3841 // No buffers for this track. Give it a few chances to 3842 // fill a buffer, then remove it from active list. 3843 if (--(track->mRetryCount) <= 0) { 3844 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3845 trackToRemove = track; 3846 } else { 3847 mixerStatus = MIXER_TRACKS_ENABLED; 3848 } 3849 } 3850 } 3851 } 3852 3853 // FIXME merge this with similar code for removing multiple tracks 3854 // remove all the tracks that need to be... 3855 if (CC_UNLIKELY(trackToRemove != 0)) { 3856 tracksToRemove->add(trackToRemove); 3857 mActiveTracks.remove(trackToRemove); 3858 if (!mEffectChains.isEmpty()) { 3859 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3860 trackToRemove->sessionId()); 3861 mEffectChains[0]->decActiveTrackCnt(); 3862 } 3863 if (trackToRemove->isTerminated()) { 3864 removeTrack_l(trackToRemove); 3865 } 3866 } 3867 3868 return mixerStatus; 3869} 3870 3871void AudioFlinger::DirectOutputThread::threadLoop_mix() 3872{ 3873 AudioBufferProvider::Buffer buffer; 3874 size_t frameCount = mFrameCount; 3875 int8_t *curBuf = (int8_t *)mMixBuffer; 3876 // output audio to hardware 3877 while (frameCount) { 3878 buffer.frameCount = frameCount; 3879 mActiveTrack->getNextBuffer(&buffer); 3880 if (CC_UNLIKELY(buffer.raw == NULL)) { 3881 memset(curBuf, 0, frameCount * mFrameSize); 3882 break; 3883 } 3884 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3885 frameCount -= buffer.frameCount; 3886 curBuf += buffer.frameCount * mFrameSize; 3887 mActiveTrack->releaseBuffer(&buffer); 3888 } 3889 sleepTime = 0; 3890 standbyTime = systemTime() + standbyDelay; 3891 mActiveTrack.clear(); 3892 3893} 3894 3895void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3896{ 3897 if (sleepTime == 0) { 3898 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3899 sleepTime = activeSleepTime; 3900 } else { 3901 sleepTime = idleSleepTime; 3902 } 3903 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3904 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3905 sleepTime = 0; 3906 } 3907} 3908 3909// getTrackName_l() must be called with ThreadBase::mLock held 3910int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3911 int sessionId) 3912{ 3913 return 0; 3914} 3915 3916// deleteTrackName_l() must be called with ThreadBase::mLock held 3917void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3918{ 3919} 3920 3921// checkForNewParameters_l() must be called with ThreadBase::mLock held 3922bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3923{ 3924 bool reconfig = false; 3925 3926 while (!mNewParameters.isEmpty()) { 3927 status_t status = NO_ERROR; 3928 String8 keyValuePair = mNewParameters[0]; 3929 AudioParameter param = AudioParameter(keyValuePair); 3930 int value; 3931 3932 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3933 // do not accept frame count changes if tracks are open as the track buffer 3934 // size depends on frame count and correct behavior would not be garantied 3935 // if frame count is changed after track creation 3936 if (!mTracks.isEmpty()) { 3937 status = INVALID_OPERATION; 3938 } else { 3939 reconfig = true; 3940 } 3941 } 3942 if (status == NO_ERROR) { 3943 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3944 keyValuePair.string()); 3945 if (!mStandby && status == INVALID_OPERATION) { 3946 mOutput->stream->common.standby(&mOutput->stream->common); 3947 mStandby = true; 3948 mBytesWritten = 0; 3949 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3950 keyValuePair.string()); 3951 } 3952 if (status == NO_ERROR && reconfig) { 3953 readOutputParameters(); 3954 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3955 } 3956 } 3957 3958 mNewParameters.removeAt(0); 3959 3960 mParamStatus = status; 3961 mParamCond.signal(); 3962 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3963 // already timed out waiting for the status and will never signal the condition. 3964 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3965 } 3966 return reconfig; 3967} 3968 3969uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3970{ 3971 uint32_t time; 3972 if (audio_is_linear_pcm(mFormat)) { 3973 time = PlaybackThread::activeSleepTimeUs(); 3974 } else { 3975 time = 10000; 3976 } 3977 return time; 3978} 3979 3980uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3981{ 3982 uint32_t time; 3983 if (audio_is_linear_pcm(mFormat)) { 3984 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3985 } else { 3986 time = 10000; 3987 } 3988 return time; 3989} 3990 3991uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3992{ 3993 uint32_t time; 3994 if (audio_is_linear_pcm(mFormat)) { 3995 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3996 } else { 3997 time = 10000; 3998 } 3999 return time; 4000} 4001 4002void AudioFlinger::DirectOutputThread::cacheParameters_l() 4003{ 4004 PlaybackThread::cacheParameters_l(); 4005 4006 // use shorter standby delay as on normal output to release 4007 // hardware resources as soon as possible 4008 standbyDelay = microseconds(activeSleepTime*2); 4009} 4010 4011// ---------------------------------------------------------------------------- 4012 4013AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4014 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4015 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4016 DUPLICATING), 4017 mWaitTimeMs(UINT_MAX) 4018{ 4019 addOutputTrack(mainThread); 4020} 4021 4022AudioFlinger::DuplicatingThread::~DuplicatingThread() 4023{ 4024 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4025 mOutputTracks[i]->destroy(); 4026 } 4027} 4028 4029void AudioFlinger::DuplicatingThread::threadLoop_mix() 4030{ 4031 // mix buffers... 4032 if (outputsReady(outputTracks)) { 4033 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4034 } else { 4035 memset(mMixBuffer, 0, mixBufferSize); 4036 } 4037 sleepTime = 0; 4038 writeFrames = mNormalFrameCount; 4039 standbyTime = systemTime() + standbyDelay; 4040} 4041 4042void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4043{ 4044 if (sleepTime == 0) { 4045 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4046 sleepTime = activeSleepTime; 4047 } else { 4048 sleepTime = idleSleepTime; 4049 } 4050 } else if (mBytesWritten != 0) { 4051 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4052 writeFrames = mNormalFrameCount; 4053 memset(mMixBuffer, 0, mixBufferSize); 4054 } else { 4055 // flush remaining overflow buffers in output tracks 4056 writeFrames = 0; 4057 } 4058 sleepTime = 0; 4059 } 4060} 4061 4062void AudioFlinger::DuplicatingThread::threadLoop_write() 4063{ 4064 for (size_t i = 0; i < outputTracks.size(); i++) { 4065 outputTracks[i]->write(mMixBuffer, writeFrames); 4066 } 4067 mBytesWritten += mixBufferSize; 4068} 4069 4070void AudioFlinger::DuplicatingThread::threadLoop_standby() 4071{ 4072 // DuplicatingThread implements standby by stopping all tracks 4073 for (size_t i = 0; i < outputTracks.size(); i++) { 4074 outputTracks[i]->stop(); 4075 } 4076} 4077 4078void AudioFlinger::DuplicatingThread::saveOutputTracks() 4079{ 4080 outputTracks = mOutputTracks; 4081} 4082 4083void AudioFlinger::DuplicatingThread::clearOutputTracks() 4084{ 4085 outputTracks.clear(); 4086} 4087 4088void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4089{ 4090 Mutex::Autolock _l(mLock); 4091 // FIXME explain this formula 4092 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4093 OutputTrack *outputTrack = new OutputTrack(thread, 4094 this, 4095 mSampleRate, 4096 mFormat, 4097 mChannelMask, 4098 frameCount); 4099 if (outputTrack->cblk() != NULL) { 4100 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4101 mOutputTracks.add(outputTrack); 4102 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4103 updateWaitTime_l(); 4104 } 4105} 4106 4107void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4108{ 4109 Mutex::Autolock _l(mLock); 4110 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4111 if (mOutputTracks[i]->thread() == thread) { 4112 mOutputTracks[i]->destroy(); 4113 mOutputTracks.removeAt(i); 4114 updateWaitTime_l(); 4115 return; 4116 } 4117 } 4118 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4119} 4120 4121// caller must hold mLock 4122void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4123{ 4124 mWaitTimeMs = UINT_MAX; 4125 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4126 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4127 if (strong != 0) { 4128 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4129 if (waitTimeMs < mWaitTimeMs) { 4130 mWaitTimeMs = waitTimeMs; 4131 } 4132 } 4133 } 4134} 4135 4136 4137bool AudioFlinger::DuplicatingThread::outputsReady( 4138 const SortedVector< sp<OutputTrack> > &outputTracks) 4139{ 4140 for (size_t i = 0; i < outputTracks.size(); i++) { 4141 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4142 if (thread == 0) { 4143 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4144 outputTracks[i].get()); 4145 return false; 4146 } 4147 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4148 // see note at standby() declaration 4149 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4150 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4151 thread.get()); 4152 return false; 4153 } 4154 } 4155 return true; 4156} 4157 4158uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4159{ 4160 return (mWaitTimeMs * 1000) / 2; 4161} 4162 4163void AudioFlinger::DuplicatingThread::cacheParameters_l() 4164{ 4165 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4166 updateWaitTime_l(); 4167 4168 MixerThread::cacheParameters_l(); 4169} 4170 4171// ---------------------------------------------------------------------------- 4172 4173// TrackBase constructor must be called with AudioFlinger::mLock held 4174AudioFlinger::ThreadBase::TrackBase::TrackBase( 4175 ThreadBase *thread, 4176 const sp<Client>& client, 4177 uint32_t sampleRate, 4178 audio_format_t format, 4179 audio_channel_mask_t channelMask, 4180 int frameCount, 4181 const sp<IMemory>& sharedBuffer, 4182 int sessionId) 4183 : RefBase(), 4184 mThread(thread), 4185 mClient(client), 4186 mCblk(NULL), 4187 // mBuffer 4188 // mBufferEnd 4189 mStepCount(0), 4190 mState(IDLE), 4191 mSampleRate(sampleRate), 4192 mFormat(format), 4193 mFrameSize(0), // will be set to correct value in constructor 4194 mStepServerFailed(false), 4195 mSessionId(sessionId) 4196 // mChannelCount 4197 // mChannelMask 4198{ 4199 // client == 0 implies sharedBuffer == 0 4200 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 4201 4202 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 4203 sharedBuffer->size()); 4204 4205 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4206 size_t size = sizeof(audio_track_cblk_t); 4207 uint8_t channelCount = popcount(channelMask); 4208 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4209 if (sharedBuffer == 0) { 4210 size += bufferSize; 4211 } 4212 4213 if (client != 0) { 4214 mCblkMemory = client->heap()->allocate(size); 4215 if (mCblkMemory != 0) { 4216 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4217 // can't assume mCblk != NULL 4218 } else { 4219 ALOGE("not enough memory for AudioTrack size=%u", size); 4220 client->heap()->dump("AudioTrack"); 4221 return; 4222 } 4223 } else { 4224 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4225 // assume mCblk != NULL 4226 } 4227 4228 // construct the shared structure in-place. 4229 if (mCblk != NULL) { 4230 new(mCblk) audio_track_cblk_t(); 4231 // clear all buffers 4232 mCblk->frameCount = frameCount; 4233 mCblk->sampleRate = sampleRate; 4234// uncomment the following lines to quickly test 32-bit wraparound 4235// mCblk->user = 0xffff0000; 4236// mCblk->server = 0xffff0000; 4237// mCblk->userBase = 0xffff0000; 4238// mCblk->serverBase = 0xffff0000; 4239 mChannelCount = channelCount; 4240 mChannelMask = channelMask; 4241 if (sharedBuffer == 0) { 4242 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4243 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4244 // Force underrun condition to avoid false underrun callback until first data is 4245 // written to buffer (other flags are cleared) 4246 mCblk->flags = CBLK_UNDERRUN; 4247 } else { 4248 mBuffer = sharedBuffer->pointer(); 4249 } 4250 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4251 } 4252} 4253 4254AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4255{ 4256 if (mCblk != NULL) { 4257 if (mClient == 0) { 4258 delete mCblk; 4259 } else { 4260 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4261 } 4262 } 4263 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4264 if (mClient != 0) { 4265 // Client destructor must run with AudioFlinger mutex locked 4266 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4267 // If the client's reference count drops to zero, the associated destructor 4268 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4269 // relying on the automatic clear() at end of scope. 4270 mClient.clear(); 4271 } 4272} 4273 4274// AudioBufferProvider interface 4275// getNextBuffer() = 0; 4276// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4277void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4278{ 4279 buffer->raw = NULL; 4280 mStepCount = buffer->frameCount; 4281 // FIXME See note at getNextBuffer() 4282 (void) step(); // ignore return value of step() 4283 buffer->frameCount = 0; 4284} 4285 4286bool AudioFlinger::ThreadBase::TrackBase::step() { 4287 bool result; 4288 audio_track_cblk_t* cblk = this->cblk(); 4289 4290 result = cblk->stepServer(mStepCount, isOut()); 4291 if (!result) { 4292 ALOGV("stepServer failed acquiring cblk mutex"); 4293 mStepServerFailed = true; 4294 } 4295 return result; 4296} 4297 4298void AudioFlinger::ThreadBase::TrackBase::reset() { 4299 audio_track_cblk_t* cblk = this->cblk(); 4300 4301 cblk->user = 0; 4302 cblk->server = 0; 4303 cblk->userBase = 0; 4304 cblk->serverBase = 0; 4305 mStepServerFailed = false; 4306 ALOGV("TrackBase::reset"); 4307} 4308 4309uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4310 return mCblk->sampleRate; 4311} 4312 4313void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4314 audio_track_cblk_t* cblk = this->cblk(); 4315 size_t frameSize = mFrameSize; 4316 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4317 int8_t *bufferEnd = bufferStart + frames * frameSize; 4318 4319 // Check validity of returned pointer in case the track control block would have been corrupted. 4320 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4321 "TrackBase::getBuffer buffer out of range:\n" 4322 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4323 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4324 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4325 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4326 4327 return bufferStart; 4328} 4329 4330status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4331{ 4332 mSyncEvents.add(event); 4333 return NO_ERROR; 4334} 4335 4336// ---------------------------------------------------------------------------- 4337 4338// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4339AudioFlinger::PlaybackThread::Track::Track( 4340 PlaybackThread *thread, 4341 const sp<Client>& client, 4342 audio_stream_type_t streamType, 4343 uint32_t sampleRate, 4344 audio_format_t format, 4345 audio_channel_mask_t channelMask, 4346 int frameCount, 4347 const sp<IMemory>& sharedBuffer, 4348 int sessionId, 4349 IAudioFlinger::track_flags_t flags) 4350 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 4351 sessionId), 4352 mMute(false), 4353 mFillingUpStatus(FS_INVALID), 4354 // mRetryCount initialized later when needed 4355 mSharedBuffer(sharedBuffer), 4356 mStreamType(streamType), 4357 mName(-1), // see note below 4358 mMainBuffer(thread->mixBuffer()), 4359 mAuxBuffer(NULL), 4360 mAuxEffectId(0), mHasVolumeController(false), 4361 mPresentationCompleteFrames(0), 4362 mFlags(flags), 4363 mFastIndex(-1), 4364 mUnderrunCount(0), 4365 mCachedVolume(1.0) 4366{ 4367 // NOTE: frame size for 8 bit PCM data is based on a sample size of 4368 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4369 mFrameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : 4370 sizeof(uint8_t); 4371 if (mCblk != NULL) { 4372 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4373 mName = thread->getTrackName_l(channelMask, sessionId); 4374 mCblk->mName = mName; 4375 if (mName < 0) { 4376 ALOGE("no more track names available"); 4377 return; 4378 } 4379 // only allocate a fast track index if we were able to allocate a normal track name 4380 if (flags & IAudioFlinger::TRACK_FAST) { 4381 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4382 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4383 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4384 // FIXME This is too eager. We allocate a fast track index before the 4385 // fast track becomes active. Since fast tracks are a scarce resource, 4386 // this means we are potentially denying other more important fast tracks from 4387 // being created. It would be better to allocate the index dynamically. 4388 mFastIndex = i; 4389 mCblk->mName = i; 4390 // Read the initial underruns because this field is never cleared by the fast mixer 4391 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4392 thread->mFastTrackAvailMask &= ~(1 << i); 4393 } 4394 } 4395 ALOGV("Track constructor name %d, calling pid %d", mName, 4396 IPCThreadState::self()->getCallingPid()); 4397} 4398 4399AudioFlinger::PlaybackThread::Track::~Track() 4400{ 4401 ALOGV("PlaybackThread::Track destructor"); 4402} 4403 4404void AudioFlinger::PlaybackThread::Track::destroy() 4405{ 4406 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4407 // by removing it from mTracks vector, so there is a risk that this Tracks's 4408 // destructor is called. As the destructor needs to lock mLock, 4409 // we must acquire a strong reference on this Track before locking mLock 4410 // here so that the destructor is called only when exiting this function. 4411 // On the other hand, as long as Track::destroy() is only called by 4412 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4413 // this Track with its member mTrack. 4414 sp<Track> keep(this); 4415 { // scope for mLock 4416 sp<ThreadBase> thread = mThread.promote(); 4417 if (thread != 0) { 4418 if (!isOutputTrack()) { 4419 if (mState == ACTIVE || mState == RESUMING) { 4420 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4421 4422#ifdef ADD_BATTERY_DATA 4423 // to track the speaker usage 4424 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4425#endif 4426 } 4427 AudioSystem::releaseOutput(thread->id()); 4428 } 4429 Mutex::Autolock _l(thread->mLock); 4430 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4431 playbackThread->destroyTrack_l(this); 4432 } 4433 } 4434} 4435 4436/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4437{ 4438 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " 4439 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 4440} 4441 4442void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4443{ 4444 uint32_t vlr = mCblk->getVolumeLR(); 4445 if (isFastTrack()) { 4446 sprintf(buffer, " F %2d", mFastIndex); 4447 } else { 4448 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4449 } 4450 track_state state = mState; 4451 char stateChar; 4452 switch (state) { 4453 case IDLE: 4454 stateChar = 'I'; 4455 break; 4456 case TERMINATED: 4457 stateChar = 'T'; 4458 break; 4459 case STOPPING_1: 4460 stateChar = 's'; 4461 break; 4462 case STOPPING_2: 4463 stateChar = '5'; 4464 break; 4465 case STOPPED: 4466 stateChar = 'S'; 4467 break; 4468 case RESUMING: 4469 stateChar = 'R'; 4470 break; 4471 case ACTIVE: 4472 stateChar = 'A'; 4473 break; 4474 case PAUSING: 4475 stateChar = 'p'; 4476 break; 4477 case PAUSED: 4478 stateChar = 'P'; 4479 break; 4480 case FLUSHED: 4481 stateChar = 'F'; 4482 break; 4483 default: 4484 stateChar = '?'; 4485 break; 4486 } 4487 char nowInUnderrun; 4488 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4489 case UNDERRUN_FULL: 4490 nowInUnderrun = ' '; 4491 break; 4492 case UNDERRUN_PARTIAL: 4493 nowInUnderrun = '<'; 4494 break; 4495 case UNDERRUN_EMPTY: 4496 nowInUnderrun = '*'; 4497 break; 4498 default: 4499 nowInUnderrun = '?'; 4500 break; 4501 } 4502 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4503 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4504 (mClient == 0) ? getpid_cached : mClient->pid(), 4505 mStreamType, 4506 mFormat, 4507 mChannelMask, 4508 mSessionId, 4509 mStepCount, 4510 mCblk->frameCount, 4511 stateChar, 4512 mMute, 4513 mFillingUpStatus, 4514 mCblk->sampleRate, 4515 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4516 20.0 * log10((vlr >> 16) / 4096.0), 4517 mCblk->server, 4518 mCblk->user, 4519 (int)mMainBuffer, 4520 (int)mAuxBuffer, 4521 mCblk->flags, 4522 mUnderrunCount, 4523 nowInUnderrun); 4524} 4525 4526// AudioBufferProvider interface 4527status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4528 AudioBufferProvider::Buffer* buffer, int64_t pts) 4529{ 4530 audio_track_cblk_t* cblk = this->cblk(); 4531 uint32_t framesReady; 4532 uint32_t framesReq = buffer->frameCount; 4533 4534 // Check if last stepServer failed, try to step now 4535 if (mStepServerFailed) { 4536 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4537 // Since the fast mixer is higher priority than client callback thread, 4538 // it does not result in priority inversion for client. 4539 // But a non-blocking solution would be preferable to avoid 4540 // fast mixer being unable to tryLock(), and 4541 // to avoid the extra context switches if the client wakes up, 4542 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4543 if (!step()) goto getNextBuffer_exit; 4544 ALOGV("stepServer recovered"); 4545 mStepServerFailed = false; 4546 } 4547 4548 // FIXME Same as above 4549 framesReady = cblk->framesReadyOut(); 4550 4551 if (CC_LIKELY(framesReady)) { 4552 uint32_t s = cblk->server; 4553 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4554 4555 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4556 if (framesReq > framesReady) { 4557 framesReq = framesReady; 4558 } 4559 if (framesReq > bufferEnd - s) { 4560 framesReq = bufferEnd - s; 4561 } 4562 4563 buffer->raw = getBuffer(s, framesReq); 4564 buffer->frameCount = framesReq; 4565 return NO_ERROR; 4566 } 4567 4568getNextBuffer_exit: 4569 buffer->raw = NULL; 4570 buffer->frameCount = 0; 4571 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4572 return NOT_ENOUGH_DATA; 4573} 4574 4575// Note that framesReady() takes a mutex on the control block using tryLock(). 4576// This could result in priority inversion if framesReady() is called by the normal mixer, 4577// as the normal mixer thread runs at lower 4578// priority than the client's callback thread: there is a short window within framesReady() 4579// during which the normal mixer could be preempted, and the client callback would block. 4580// Another problem can occur if framesReady() is called by the fast mixer: 4581// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4582// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4583size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4584 return mCblk->framesReadyOut(); 4585} 4586 4587// Don't call for fast tracks; the framesReady() could result in priority inversion 4588bool AudioFlinger::PlaybackThread::Track::isReady() const { 4589 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4590 4591 if (framesReady() >= mCblk->frameCount || 4592 (mCblk->flags & CBLK_FORCEREADY)) { 4593 mFillingUpStatus = FS_FILLED; 4594 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4595 return true; 4596 } 4597 return false; 4598} 4599 4600status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4601 int triggerSession) 4602{ 4603 status_t status = NO_ERROR; 4604 ALOGV("start(%d), calling pid %d session %d", 4605 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4606 4607 sp<ThreadBase> thread = mThread.promote(); 4608 if (thread != 0) { 4609 Mutex::Autolock _l(thread->mLock); 4610 track_state state = mState; 4611 // here the track could be either new, or restarted 4612 // in both cases "unstop" the track 4613 if (mState == PAUSED) { 4614 mState = TrackBase::RESUMING; 4615 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4616 } else { 4617 mState = TrackBase::ACTIVE; 4618 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4619 } 4620 4621 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4622 thread->mLock.unlock(); 4623 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4624 thread->mLock.lock(); 4625 4626#ifdef ADD_BATTERY_DATA 4627 // to track the speaker usage 4628 if (status == NO_ERROR) { 4629 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4630 } 4631#endif 4632 } 4633 if (status == NO_ERROR) { 4634 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4635 playbackThread->addTrack_l(this); 4636 } else { 4637 mState = state; 4638 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4639 } 4640 } else { 4641 status = BAD_VALUE; 4642 } 4643 return status; 4644} 4645 4646void AudioFlinger::PlaybackThread::Track::stop() 4647{ 4648 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4649 sp<ThreadBase> thread = mThread.promote(); 4650 if (thread != 0) { 4651 Mutex::Autolock _l(thread->mLock); 4652 track_state state = mState; 4653 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4654 // If the track is not active (PAUSED and buffers full), flush buffers 4655 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4656 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4657 reset(); 4658 mState = STOPPED; 4659 } else if (!isFastTrack()) { 4660 mState = STOPPED; 4661 } else { 4662 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4663 // and then to STOPPED and reset() when presentation is complete 4664 mState = STOPPING_1; 4665 } 4666 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 4667 playbackThread); 4668 } 4669 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4670 thread->mLock.unlock(); 4671 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4672 thread->mLock.lock(); 4673 4674#ifdef ADD_BATTERY_DATA 4675 // to track the speaker usage 4676 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4677#endif 4678 } 4679 } 4680} 4681 4682void AudioFlinger::PlaybackThread::Track::pause() 4683{ 4684 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4685 sp<ThreadBase> thread = mThread.promote(); 4686 if (thread != 0) { 4687 Mutex::Autolock _l(thread->mLock); 4688 if (mState == ACTIVE || mState == RESUMING) { 4689 mState = PAUSING; 4690 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4691 if (!isOutputTrack()) { 4692 thread->mLock.unlock(); 4693 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4694 thread->mLock.lock(); 4695 4696#ifdef ADD_BATTERY_DATA 4697 // to track the speaker usage 4698 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4699#endif 4700 } 4701 } 4702 } 4703} 4704 4705void AudioFlinger::PlaybackThread::Track::flush() 4706{ 4707 ALOGV("flush(%d)", mName); 4708 sp<ThreadBase> thread = mThread.promote(); 4709 if (thread != 0) { 4710 Mutex::Autolock _l(thread->mLock); 4711 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4712 mState != PAUSING) { 4713 return; 4714 } 4715 // No point remaining in PAUSED state after a flush => go to 4716 // FLUSHED state 4717 mState = FLUSHED; 4718 // do not reset the track if it is still in the process of being stopped or paused. 4719 // this will be done by prepareTracks_l() when the track is stopped. 4720 // prepareTracks_l() will see mState == FLUSHED, then 4721 // remove from active track list, reset(), and trigger presentation complete 4722 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4723 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4724 reset(); 4725 } 4726 } 4727} 4728 4729void AudioFlinger::PlaybackThread::Track::reset() 4730{ 4731 // Do not reset twice to avoid discarding data written just after a flush and before 4732 // the audioflinger thread detects the track is stopped. 4733 if (!mResetDone) { 4734 TrackBase::reset(); 4735 // Force underrun condition to avoid false underrun callback until first data is 4736 // written to buffer 4737 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4738 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 4739 mFillingUpStatus = FS_FILLING; 4740 mResetDone = true; 4741 if (mState == FLUSHED) { 4742 mState = IDLE; 4743 } 4744 } 4745} 4746 4747void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4748{ 4749 mMute = muted; 4750} 4751 4752status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4753{ 4754 status_t status = DEAD_OBJECT; 4755 sp<ThreadBase> thread = mThread.promote(); 4756 if (thread != 0) { 4757 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4758 sp<AudioFlinger> af = mClient->audioFlinger(); 4759 4760 Mutex::Autolock _l(af->mLock); 4761 4762 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4763 4764 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4765 Mutex::Autolock _dl(playbackThread->mLock); 4766 Mutex::Autolock _sl(srcThread->mLock); 4767 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4768 if (chain == 0) { 4769 return INVALID_OPERATION; 4770 } 4771 4772 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4773 if (effect == 0) { 4774 return INVALID_OPERATION; 4775 } 4776 srcThread->removeEffect_l(effect); 4777 playbackThread->addEffect_l(effect); 4778 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4779 if (effect->state() == EffectModule::ACTIVE || 4780 effect->state() == EffectModule::STOPPING) { 4781 effect->start(); 4782 } 4783 4784 sp<EffectChain> dstChain = effect->chain().promote(); 4785 if (dstChain == 0) { 4786 srcThread->addEffect_l(effect); 4787 return INVALID_OPERATION; 4788 } 4789 AudioSystem::unregisterEffect(effect->id()); 4790 AudioSystem::registerEffect(&effect->desc(), 4791 srcThread->id(), 4792 dstChain->strategy(), 4793 AUDIO_SESSION_OUTPUT_MIX, 4794 effect->id()); 4795 } 4796 status = playbackThread->attachAuxEffect(this, EffectId); 4797 } 4798 return status; 4799} 4800 4801void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4802{ 4803 mAuxEffectId = EffectId; 4804 mAuxBuffer = buffer; 4805} 4806 4807bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4808 size_t audioHalFrames) 4809{ 4810 // a track is considered presented when the total number of frames written to audio HAL 4811 // corresponds to the number of frames written when presentationComplete() is called for the 4812 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4813 if (mPresentationCompleteFrames == 0) { 4814 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4815 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4816 mPresentationCompleteFrames, audioHalFrames); 4817 } 4818 if (framesWritten >= mPresentationCompleteFrames) { 4819 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4820 mSessionId, framesWritten); 4821 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4822 return true; 4823 } 4824 return false; 4825} 4826 4827void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4828{ 4829 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4830 if (mSyncEvents[i]->type() == type) { 4831 mSyncEvents[i]->trigger(); 4832 mSyncEvents.removeAt(i); 4833 i--; 4834 } 4835 } 4836} 4837 4838// implement VolumeBufferProvider interface 4839 4840uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4841{ 4842 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4843 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4844 uint32_t vlr = mCblk->getVolumeLR(); 4845 uint32_t vl = vlr & 0xFFFF; 4846 uint32_t vr = vlr >> 16; 4847 // track volumes come from shared memory, so can't be trusted and must be clamped 4848 if (vl > MAX_GAIN_INT) { 4849 vl = MAX_GAIN_INT; 4850 } 4851 if (vr > MAX_GAIN_INT) { 4852 vr = MAX_GAIN_INT; 4853 } 4854 // now apply the cached master volume and stream type volume; 4855 // this is trusted but lacks any synchronization or barrier so may be stale 4856 float v = mCachedVolume; 4857 vl *= v; 4858 vr *= v; 4859 // re-combine into U4.16 4860 vlr = (vr << 16) | (vl & 0xFFFF); 4861 // FIXME look at mute, pause, and stop flags 4862 return vlr; 4863} 4864 4865status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4866{ 4867 if (mState == TERMINATED || mState == PAUSED || 4868 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4869 (mState == STOPPED)))) { 4870 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4871 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4872 event->cancel(); 4873 return INVALID_OPERATION; 4874 } 4875 (void) TrackBase::setSyncEvent(event); 4876 return NO_ERROR; 4877} 4878 4879bool AudioFlinger::PlaybackThread::Track::isOut() const 4880{ 4881 return true; 4882} 4883 4884// timed audio tracks 4885 4886sp<AudioFlinger::PlaybackThread::TimedTrack> 4887AudioFlinger::PlaybackThread::TimedTrack::create( 4888 PlaybackThread *thread, 4889 const sp<Client>& client, 4890 audio_stream_type_t streamType, 4891 uint32_t sampleRate, 4892 audio_format_t format, 4893 audio_channel_mask_t channelMask, 4894 int frameCount, 4895 const sp<IMemory>& sharedBuffer, 4896 int sessionId) { 4897 if (!client->reserveTimedTrack()) 4898 return 0; 4899 4900 return new TimedTrack( 4901 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4902 sharedBuffer, sessionId); 4903} 4904 4905AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4906 PlaybackThread *thread, 4907 const sp<Client>& client, 4908 audio_stream_type_t streamType, 4909 uint32_t sampleRate, 4910 audio_format_t format, 4911 audio_channel_mask_t channelMask, 4912 int frameCount, 4913 const sp<IMemory>& sharedBuffer, 4914 int sessionId) 4915 : Track(thread, client, streamType, sampleRate, format, channelMask, 4916 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4917 mQueueHeadInFlight(false), 4918 mTrimQueueHeadOnRelease(false), 4919 mFramesPendingInQueue(0), 4920 mTimedSilenceBuffer(NULL), 4921 mTimedSilenceBufferSize(0), 4922 mTimedAudioOutputOnTime(false), 4923 mMediaTimeTransformValid(false) 4924{ 4925 LocalClock lc; 4926 mLocalTimeFreq = lc.getLocalFreq(); 4927 4928 mLocalTimeToSampleTransform.a_zero = 0; 4929 mLocalTimeToSampleTransform.b_zero = 0; 4930 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4931 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4932 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4933 &mLocalTimeToSampleTransform.a_to_b_denom); 4934 4935 mMediaTimeToSampleTransform.a_zero = 0; 4936 mMediaTimeToSampleTransform.b_zero = 0; 4937 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4938 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4939 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4940 &mMediaTimeToSampleTransform.a_to_b_denom); 4941} 4942 4943AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4944 mClient->releaseTimedTrack(); 4945 delete [] mTimedSilenceBuffer; 4946} 4947 4948status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4949 size_t size, sp<IMemory>* buffer) { 4950 4951 Mutex::Autolock _l(mTimedBufferQueueLock); 4952 4953 trimTimedBufferQueue_l(); 4954 4955 // lazily initialize the shared memory heap for timed buffers 4956 if (mTimedMemoryDealer == NULL) { 4957 const int kTimedBufferHeapSize = 512 << 10; 4958 4959 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4960 "AudioFlingerTimed"); 4961 if (mTimedMemoryDealer == NULL) 4962 return NO_MEMORY; 4963 } 4964 4965 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4966 if (newBuffer == NULL) { 4967 newBuffer = mTimedMemoryDealer->allocate(size); 4968 if (newBuffer == NULL) 4969 return NO_MEMORY; 4970 } 4971 4972 *buffer = newBuffer; 4973 return NO_ERROR; 4974} 4975 4976// caller must hold mTimedBufferQueueLock 4977void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4978 int64_t mediaTimeNow; 4979 { 4980 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4981 if (!mMediaTimeTransformValid) 4982 return; 4983 4984 int64_t targetTimeNow; 4985 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4986 ? mCCHelper.getCommonTime(&targetTimeNow) 4987 : mCCHelper.getLocalTime(&targetTimeNow); 4988 4989 if (OK != res) 4990 return; 4991 4992 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4993 &mediaTimeNow)) { 4994 return; 4995 } 4996 } 4997 4998 size_t trimEnd; 4999 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 5000 int64_t bufEnd; 5001 5002 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 5003 // We have a next buffer. Just use its PTS as the PTS of the frame 5004 // following the last frame in this buffer. If the stream is sparse 5005 // (ie, there are deliberate gaps left in the stream which should be 5006 // filled with silence by the TimedAudioTrack), then this can result 5007 // in one extra buffer being left un-trimmed when it could have 5008 // been. In general, this is not typical, and we would rather 5009 // optimized away the TS calculation below for the more common case 5010 // where PTSes are contiguous. 5011 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 5012 } else { 5013 // We have no next buffer. Compute the PTS of the frame following 5014 // the last frame in this buffer by computing the duration of of 5015 // this frame in media time units and adding it to the PTS of the 5016 // buffer. 5017 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 5018 / mFrameSize; 5019 5020 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5021 &bufEnd)) { 5022 ALOGE("Failed to convert frame count of %lld to media time" 5023 " duration" " (scale factor %d/%u) in %s", 5024 frameCount, 5025 mMediaTimeToSampleTransform.a_to_b_numer, 5026 mMediaTimeToSampleTransform.a_to_b_denom, 5027 __PRETTY_FUNCTION__); 5028 break; 5029 } 5030 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5031 } 5032 5033 if (bufEnd > mediaTimeNow) 5034 break; 5035 5036 // Is the buffer we want to use in the middle of a mix operation right 5037 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5038 // from the mixer which should be coming back shortly. 5039 if (!trimEnd && mQueueHeadInFlight) { 5040 mTrimQueueHeadOnRelease = true; 5041 } 5042 } 5043 5044 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5045 if (trimStart < trimEnd) { 5046 // Update the bookkeeping for framesReady() 5047 for (size_t i = trimStart; i < trimEnd; ++i) { 5048 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5049 } 5050 5051 // Now actually remove the buffers from the queue. 5052 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5053 } 5054} 5055 5056void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5057 const char* logTag) { 5058 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5059 "%s called (reason \"%s\"), but timed buffer queue has no" 5060 " elements to trim.", __FUNCTION__, logTag); 5061 5062 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5063 mTimedBufferQueue.removeAt(0); 5064} 5065 5066void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5067 const TimedBuffer& buf, 5068 const char* logTag) { 5069 uint32_t bufBytes = buf.buffer()->size(); 5070 uint32_t consumedAlready = buf.position(); 5071 5072 ALOG_ASSERT(consumedAlready <= bufBytes, 5073 "Bad bookkeeping while updating frames pending. Timed buffer is" 5074 " only %u bytes long, but claims to have consumed %u" 5075 " bytes. (update reason: \"%s\")", 5076 bufBytes, consumedAlready, logTag); 5077 5078 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 5079 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5080 "Bad bookkeeping while updating frames pending. Should have at" 5081 " least %u queued frames, but we think we have only %u. (update" 5082 " reason: \"%s\")", 5083 bufFrames, mFramesPendingInQueue, logTag); 5084 5085 mFramesPendingInQueue -= bufFrames; 5086} 5087 5088status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5089 const sp<IMemory>& buffer, int64_t pts) { 5090 5091 { 5092 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5093 if (!mMediaTimeTransformValid) 5094 return INVALID_OPERATION; 5095 } 5096 5097 Mutex::Autolock _l(mTimedBufferQueueLock); 5098 5099 uint32_t bufFrames = buffer->size() / mFrameSize; 5100 mFramesPendingInQueue += bufFrames; 5101 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5102 5103 return NO_ERROR; 5104} 5105 5106status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5107 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5108 5109 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5110 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5111 target); 5112 5113 if (!(target == TimedAudioTrack::LOCAL_TIME || 5114 target == TimedAudioTrack::COMMON_TIME)) { 5115 return BAD_VALUE; 5116 } 5117 5118 Mutex::Autolock lock(mMediaTimeTransformLock); 5119 mMediaTimeTransform = xform; 5120 mMediaTimeTransformTarget = target; 5121 mMediaTimeTransformValid = true; 5122 5123 return NO_ERROR; 5124} 5125 5126#define min(a, b) ((a) < (b) ? (a) : (b)) 5127 5128// implementation of getNextBuffer for tracks whose buffers have timestamps 5129status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5130 AudioBufferProvider::Buffer* buffer, int64_t pts) 5131{ 5132 if (pts == AudioBufferProvider::kInvalidPTS) { 5133 buffer->raw = NULL; 5134 buffer->frameCount = 0; 5135 mTimedAudioOutputOnTime = false; 5136 return INVALID_OPERATION; 5137 } 5138 5139 Mutex::Autolock _l(mTimedBufferQueueLock); 5140 5141 ALOG_ASSERT(!mQueueHeadInFlight, 5142 "getNextBuffer called without releaseBuffer!"); 5143 5144 while (true) { 5145 5146 // if we have no timed buffers, then fail 5147 if (mTimedBufferQueue.isEmpty()) { 5148 buffer->raw = NULL; 5149 buffer->frameCount = 0; 5150 return NOT_ENOUGH_DATA; 5151 } 5152 5153 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5154 5155 // calculate the PTS of the head of the timed buffer queue expressed in 5156 // local time 5157 int64_t headLocalPTS; 5158 { 5159 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5160 5161 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5162 5163 if (mMediaTimeTransform.a_to_b_denom == 0) { 5164 // the transform represents a pause, so yield silence 5165 timedYieldSilence_l(buffer->frameCount, buffer); 5166 return NO_ERROR; 5167 } 5168 5169 int64_t transformedPTS; 5170 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5171 &transformedPTS)) { 5172 // the transform failed. this shouldn't happen, but if it does 5173 // then just drop this buffer 5174 ALOGW("timedGetNextBuffer transform failed"); 5175 buffer->raw = NULL; 5176 buffer->frameCount = 0; 5177 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5178 return NO_ERROR; 5179 } 5180 5181 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5182 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5183 &headLocalPTS)) { 5184 buffer->raw = NULL; 5185 buffer->frameCount = 0; 5186 return INVALID_OPERATION; 5187 } 5188 } else { 5189 headLocalPTS = transformedPTS; 5190 } 5191 } 5192 5193 // adjust the head buffer's PTS to reflect the portion of the head buffer 5194 // that has already been consumed 5195 int64_t effectivePTS = headLocalPTS + 5196 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 5197 5198 // Calculate the delta in samples between the head of the input buffer 5199 // queue and the start of the next output buffer that will be written. 5200 // If the transformation fails because of over or underflow, it means 5201 // that the sample's position in the output stream is so far out of 5202 // whack that it should just be dropped. 5203 int64_t sampleDelta; 5204 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5205 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5206 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5207 " mix"); 5208 continue; 5209 } 5210 if (!mLocalTimeToSampleTransform.doForwardTransform( 5211 (effectivePTS - pts) << 32, &sampleDelta)) { 5212 ALOGV("*** too late during sample rate transform: dropped buffer"); 5213 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5214 continue; 5215 } 5216 5217 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5218 " sampleDelta=[%d.%08x]", 5219 head.pts(), head.position(), pts, 5220 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5221 + (sampleDelta >> 32)), 5222 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5223 5224 // if the delta between the ideal placement for the next input sample and 5225 // the current output position is within this threshold, then we will 5226 // concatenate the next input samples to the previous output 5227 const int64_t kSampleContinuityThreshold = 5228 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5229 5230 // if this is the first buffer of audio that we're emitting from this track 5231 // then it should be almost exactly on time. 5232 const int64_t kSampleStartupThreshold = 1LL << 32; 5233 5234 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5235 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5236 // the next input is close enough to being on time, so concatenate it 5237 // with the last output 5238 timedYieldSamples_l(buffer); 5239 5240 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5241 head.position(), buffer->frameCount); 5242 return NO_ERROR; 5243 } 5244 5245 // Looks like our output is not on time. Reset our on timed status. 5246 // Next time we mix samples from our input queue, then should be within 5247 // the StartupThreshold. 5248 mTimedAudioOutputOnTime = false; 5249 if (sampleDelta > 0) { 5250 // the gap between the current output position and the proper start of 5251 // the next input sample is too big, so fill it with silence 5252 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5253 5254 timedYieldSilence_l(framesUntilNextInput, buffer); 5255 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5256 return NO_ERROR; 5257 } else { 5258 // the next input sample is late 5259 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5260 size_t onTimeSamplePosition = 5261 head.position() + lateFrames * mFrameSize; 5262 5263 if (onTimeSamplePosition > head.buffer()->size()) { 5264 // all the remaining samples in the head are too late, so 5265 // drop it and move on 5266 ALOGV("*** too late: dropped buffer"); 5267 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5268 continue; 5269 } else { 5270 // skip over the late samples 5271 head.setPosition(onTimeSamplePosition); 5272 5273 // yield the available samples 5274 timedYieldSamples_l(buffer); 5275 5276 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5277 return NO_ERROR; 5278 } 5279 } 5280 } 5281} 5282 5283// Yield samples from the timed buffer queue head up to the given output 5284// buffer's capacity. 5285// 5286// Caller must hold mTimedBufferQueueLock 5287void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5288 AudioBufferProvider::Buffer* buffer) { 5289 5290 const TimedBuffer& head = mTimedBufferQueue[0]; 5291 5292 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5293 head.position()); 5294 5295 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5296 mFrameSize); 5297 size_t framesRequested = buffer->frameCount; 5298 buffer->frameCount = min(framesLeftInHead, framesRequested); 5299 5300 mQueueHeadInFlight = true; 5301 mTimedAudioOutputOnTime = true; 5302} 5303 5304// Yield samples of silence up to the given output buffer's capacity 5305// 5306// Caller must hold mTimedBufferQueueLock 5307void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5308 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5309 5310 // lazily allocate a buffer filled with silence 5311 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 5312 delete [] mTimedSilenceBuffer; 5313 mTimedSilenceBufferSize = numFrames * mFrameSize; 5314 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5315 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5316 } 5317 5318 buffer->raw = mTimedSilenceBuffer; 5319 size_t framesRequested = buffer->frameCount; 5320 buffer->frameCount = min(numFrames, framesRequested); 5321 5322 mTimedAudioOutputOnTime = false; 5323} 5324 5325// AudioBufferProvider interface 5326void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5327 AudioBufferProvider::Buffer* buffer) { 5328 5329 Mutex::Autolock _l(mTimedBufferQueueLock); 5330 5331 // If the buffer which was just released is part of the buffer at the head 5332 // of the queue, be sure to update the amt of the buffer which has been 5333 // consumed. If the buffer being returned is not part of the head of the 5334 // queue, its either because the buffer is part of the silence buffer, or 5335 // because the head of the timed queue was trimmed after the mixer called 5336 // getNextBuffer but before the mixer called releaseBuffer. 5337 if (buffer->raw == mTimedSilenceBuffer) { 5338 ALOG_ASSERT(!mQueueHeadInFlight, 5339 "Queue head in flight during release of silence buffer!"); 5340 goto done; 5341 } 5342 5343 ALOG_ASSERT(mQueueHeadInFlight, 5344 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5345 " head in flight."); 5346 5347 if (mTimedBufferQueue.size()) { 5348 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5349 5350 void* start = head.buffer()->pointer(); 5351 void* end = reinterpret_cast<void*>( 5352 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5353 + head.buffer()->size()); 5354 5355 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5356 "released buffer not within the head of the timed buffer" 5357 " queue; qHead = [%p, %p], released buffer = %p", 5358 start, end, buffer->raw); 5359 5360 head.setPosition(head.position() + 5361 (buffer->frameCount * mFrameSize)); 5362 mQueueHeadInFlight = false; 5363 5364 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5365 "Bad bookkeeping during releaseBuffer! Should have at" 5366 " least %u queued frames, but we think we have only %u", 5367 buffer->frameCount, mFramesPendingInQueue); 5368 5369 mFramesPendingInQueue -= buffer->frameCount; 5370 5371 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5372 || mTrimQueueHeadOnRelease) { 5373 trimTimedBufferQueueHead_l("releaseBuffer"); 5374 mTrimQueueHeadOnRelease = false; 5375 } 5376 } else { 5377 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5378 " buffers in the timed buffer queue"); 5379 } 5380 5381done: 5382 buffer->raw = 0; 5383 buffer->frameCount = 0; 5384} 5385 5386size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5387 Mutex::Autolock _l(mTimedBufferQueueLock); 5388 return mFramesPendingInQueue; 5389} 5390 5391AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5392 : mPTS(0), mPosition(0) {} 5393 5394AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5395 const sp<IMemory>& buffer, int64_t pts) 5396 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5397 5398// ---------------------------------------------------------------------------- 5399 5400// RecordTrack constructor must be called with AudioFlinger::mLock held 5401AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5402 RecordThread *thread, 5403 const sp<Client>& client, 5404 uint32_t sampleRate, 5405 audio_format_t format, 5406 audio_channel_mask_t channelMask, 5407 int frameCount, 5408 int sessionId) 5409 : TrackBase(thread, client, sampleRate, format, 5410 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5411 mOverflow(false) 5412{ 5413 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5414 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5415 mFrameSize = mChannelCount * sizeof(int16_t); 5416 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5417 mFrameSize = mChannelCount * sizeof(int8_t); 5418 } else { 5419 mFrameSize = sizeof(int8_t); 5420 } 5421} 5422 5423AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5424{ 5425 ALOGV("%s", __func__); 5426} 5427 5428// AudioBufferProvider interface 5429status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 5430 int64_t pts) 5431{ 5432 audio_track_cblk_t* cblk = this->cblk(); 5433 uint32_t framesAvail; 5434 uint32_t framesReq = buffer->frameCount; 5435 5436 // Check if last stepServer failed, try to step now 5437 if (mStepServerFailed) { 5438 if (!step()) goto getNextBuffer_exit; 5439 ALOGV("stepServer recovered"); 5440 mStepServerFailed = false; 5441 } 5442 5443 // FIXME lock is not actually held, so overrun is possible 5444 framesAvail = cblk->framesAvailableIn_l(); 5445 5446 if (CC_LIKELY(framesAvail)) { 5447 uint32_t s = cblk->server; 5448 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5449 5450 if (framesReq > framesAvail) { 5451 framesReq = framesAvail; 5452 } 5453 if (framesReq > bufferEnd - s) { 5454 framesReq = bufferEnd - s; 5455 } 5456 5457 buffer->raw = getBuffer(s, framesReq); 5458 buffer->frameCount = framesReq; 5459 return NO_ERROR; 5460 } 5461 5462getNextBuffer_exit: 5463 buffer->raw = NULL; 5464 buffer->frameCount = 0; 5465 return NOT_ENOUGH_DATA; 5466} 5467 5468status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5469 int triggerSession) 5470{ 5471 sp<ThreadBase> thread = mThread.promote(); 5472 if (thread != 0) { 5473 RecordThread *recordThread = (RecordThread *)thread.get(); 5474 return recordThread->start(this, event, triggerSession); 5475 } else { 5476 return BAD_VALUE; 5477 } 5478} 5479 5480void AudioFlinger::RecordThread::RecordTrack::stop() 5481{ 5482 sp<ThreadBase> thread = mThread.promote(); 5483 if (thread != 0) { 5484 RecordThread *recordThread = (RecordThread *)thread.get(); 5485 recordThread->mLock.lock(); 5486 bool doStop = recordThread->stop_l(this); 5487 if (doStop) { 5488 TrackBase::reset(); 5489 // Force overrun condition to avoid false overrun callback until first data is 5490 // read from buffer 5491 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 5492 } 5493 recordThread->mLock.unlock(); 5494 if (doStop) { 5495 AudioSystem::stopInput(recordThread->id()); 5496 } 5497 } 5498} 5499 5500/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5501{ 5502 result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); 5503} 5504 5505void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5506{ 5507 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5508 (mClient == 0) ? getpid_cached : mClient->pid(), 5509 mFormat, 5510 mChannelMask, 5511 mSessionId, 5512 mStepCount, 5513 mState, 5514 mCblk->sampleRate, 5515 mCblk->server, 5516 mCblk->user, 5517 mCblk->frameCount); 5518} 5519 5520bool AudioFlinger::RecordThread::RecordTrack::isOut() const 5521{ 5522 return false; 5523} 5524 5525// ---------------------------------------------------------------------------- 5526 5527AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5528 PlaybackThread *playbackThread, 5529 DuplicatingThread *sourceThread, 5530 uint32_t sampleRate, 5531 audio_format_t format, 5532 audio_channel_mask_t channelMask, 5533 int frameCount) 5534 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5535 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5536 mActive(false), mSourceThread(sourceThread), mBuffers(NULL) 5537{ 5538 5539 if (mCblk != NULL) { 5540 mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5541 mOutBuffer.frameCount = 0; 5542 playbackThread->mTracks.add(this); 5543 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5544 "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", 5545 mCblk, mBuffer, mCblk->buffers, 5546 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5547 } else { 5548 ALOGW("Error creating output track on thread %p", playbackThread); 5549 } 5550} 5551 5552AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5553{ 5554 clearBufferQueue(); 5555} 5556 5557status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5558 int triggerSession) 5559{ 5560 status_t status = Track::start(event, triggerSession); 5561 if (status != NO_ERROR) { 5562 return status; 5563 } 5564 5565 mActive = true; 5566 mRetryCount = 127; 5567 return status; 5568} 5569 5570void AudioFlinger::PlaybackThread::OutputTrack::stop() 5571{ 5572 Track::stop(); 5573 clearBufferQueue(); 5574 mOutBuffer.frameCount = 0; 5575 mActive = false; 5576} 5577 5578bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5579{ 5580 Buffer *pInBuffer; 5581 Buffer inBuffer; 5582 uint32_t channelCount = mChannelCount; 5583 bool outputBufferFull = false; 5584 inBuffer.frameCount = frames; 5585 inBuffer.i16 = data; 5586 5587 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5588 5589 if (!mActive && frames != 0) { 5590 start(); 5591 sp<ThreadBase> thread = mThread.promote(); 5592 if (thread != 0) { 5593 MixerThread *mixerThread = (MixerThread *)thread.get(); 5594 if (mCblk->frameCount > frames){ 5595 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5596 uint32_t startFrames = (mCblk->frameCount - frames); 5597 pInBuffer = new Buffer; 5598 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5599 pInBuffer->frameCount = startFrames; 5600 pInBuffer->i16 = pInBuffer->mBuffer; 5601 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5602 mBufferQueue.add(pInBuffer); 5603 } else { 5604 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5605 } 5606 } 5607 } 5608 } 5609 5610 while (waitTimeLeftMs) { 5611 // First write pending buffers, then new data 5612 if (mBufferQueue.size()) { 5613 pInBuffer = mBufferQueue.itemAt(0); 5614 } else { 5615 pInBuffer = &inBuffer; 5616 } 5617 5618 if (pInBuffer->frameCount == 0) { 5619 break; 5620 } 5621 5622 if (mOutBuffer.frameCount == 0) { 5623 mOutBuffer.frameCount = pInBuffer->frameCount; 5624 nsecs_t startTime = systemTime(); 5625 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5626 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 5627 mThread.unsafe_get()); 5628 outputBufferFull = true; 5629 break; 5630 } 5631 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5632 if (waitTimeLeftMs >= waitTimeMs) { 5633 waitTimeLeftMs -= waitTimeMs; 5634 } else { 5635 waitTimeLeftMs = 0; 5636 } 5637 } 5638 5639 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 5640 pInBuffer->frameCount; 5641 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5642 mCblk->stepUserOut(outFrames); 5643 pInBuffer->frameCount -= outFrames; 5644 pInBuffer->i16 += outFrames * channelCount; 5645 mOutBuffer.frameCount -= outFrames; 5646 mOutBuffer.i16 += outFrames * channelCount; 5647 5648 if (pInBuffer->frameCount == 0) { 5649 if (mBufferQueue.size()) { 5650 mBufferQueue.removeAt(0); 5651 delete [] pInBuffer->mBuffer; 5652 delete pInBuffer; 5653 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 5654 mThread.unsafe_get(), mBufferQueue.size()); 5655 } else { 5656 break; 5657 } 5658 } 5659 } 5660 5661 // If we could not write all frames, allocate a buffer and queue it for next time. 5662 if (inBuffer.frameCount) { 5663 sp<ThreadBase> thread = mThread.promote(); 5664 if (thread != 0 && !thread->standby()) { 5665 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5666 pInBuffer = new Buffer; 5667 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5668 pInBuffer->frameCount = inBuffer.frameCount; 5669 pInBuffer->i16 = pInBuffer->mBuffer; 5670 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 5671 sizeof(int16_t)); 5672 mBufferQueue.add(pInBuffer); 5673 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 5674 mThread.unsafe_get(), mBufferQueue.size()); 5675 } else { 5676 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 5677 mThread.unsafe_get(), this); 5678 } 5679 } 5680 } 5681 5682 // Calling write() with a 0 length buffer, means that no more data will be written: 5683 // If no more buffers are pending, fill output track buffer to make sure it is started 5684 // by output mixer. 5685 if (frames == 0 && mBufferQueue.size() == 0) { 5686 if (mCblk->user < mCblk->frameCount) { 5687 frames = mCblk->frameCount - mCblk->user; 5688 pInBuffer = new Buffer; 5689 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5690 pInBuffer->frameCount = frames; 5691 pInBuffer->i16 = pInBuffer->mBuffer; 5692 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5693 mBufferQueue.add(pInBuffer); 5694 } else if (mActive) { 5695 stop(); 5696 } 5697 } 5698 5699 return outputBufferFull; 5700} 5701 5702status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 5703 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5704{ 5705 int active; 5706 status_t result; 5707 audio_track_cblk_t* cblk = mCblk; 5708 uint32_t framesReq = buffer->frameCount; 5709 5710 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5711 buffer->frameCount = 0; 5712 5713 uint32_t framesAvail = cblk->framesAvailableOut(); 5714 5715 5716 if (framesAvail == 0) { 5717 Mutex::Autolock _l(cblk->lock); 5718 goto start_loop_here; 5719 while (framesAvail == 0) { 5720 active = mActive; 5721 if (CC_UNLIKELY(!active)) { 5722 ALOGV("Not active and NO_MORE_BUFFERS"); 5723 return NO_MORE_BUFFERS; 5724 } 5725 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5726 if (result != NO_ERROR) { 5727 return NO_MORE_BUFFERS; 5728 } 5729 // read the server count again 5730 start_loop_here: 5731 framesAvail = cblk->framesAvailableOut_l(); 5732 } 5733 } 5734 5735// if (framesAvail < framesReq) { 5736// return NO_MORE_BUFFERS; 5737// } 5738 5739 if (framesReq > framesAvail) { 5740 framesReq = framesAvail; 5741 } 5742 5743 uint32_t u = cblk->user; 5744 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5745 5746 if (framesReq > bufferEnd - u) { 5747 framesReq = bufferEnd - u; 5748 } 5749 5750 buffer->frameCount = framesReq; 5751 buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); 5752 return NO_ERROR; 5753} 5754 5755 5756void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5757{ 5758 size_t size = mBufferQueue.size(); 5759 5760 for (size_t i = 0; i < size; i++) { 5761 Buffer *pBuffer = mBufferQueue.itemAt(i); 5762 delete [] pBuffer->mBuffer; 5763 delete pBuffer; 5764 } 5765 mBufferQueue.clear(); 5766} 5767 5768// ---------------------------------------------------------------------------- 5769 5770AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5771 : RefBase(), 5772 mAudioFlinger(audioFlinger), 5773 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5774 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5775 mPid(pid), 5776 mTimedTrackCount(0) 5777{ 5778 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5779} 5780 5781// Client destructor must be called with AudioFlinger::mLock held 5782AudioFlinger::Client::~Client() 5783{ 5784 mAudioFlinger->removeClient_l(mPid); 5785} 5786 5787sp<MemoryDealer> AudioFlinger::Client::heap() const 5788{ 5789 return mMemoryDealer; 5790} 5791 5792// Reserve one of the limited slots for a timed audio track associated 5793// with this client 5794bool AudioFlinger::Client::reserveTimedTrack() 5795{ 5796 const int kMaxTimedTracksPerClient = 4; 5797 5798 Mutex::Autolock _l(mTimedTrackLock); 5799 5800 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5801 ALOGW("can not create timed track - pid %d has exceeded the limit", 5802 mPid); 5803 return false; 5804 } 5805 5806 mTimedTrackCount++; 5807 return true; 5808} 5809 5810// Release a slot for a timed audio track 5811void AudioFlinger::Client::releaseTimedTrack() 5812{ 5813 Mutex::Autolock _l(mTimedTrackLock); 5814 mTimedTrackCount--; 5815} 5816 5817// ---------------------------------------------------------------------------- 5818 5819AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5820 const sp<IAudioFlingerClient>& client, 5821 pid_t pid) 5822 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5823{ 5824} 5825 5826AudioFlinger::NotificationClient::~NotificationClient() 5827{ 5828} 5829 5830void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5831{ 5832 sp<NotificationClient> keep(this); 5833 mAudioFlinger->removeNotificationClient(mPid); 5834} 5835 5836// ---------------------------------------------------------------------------- 5837 5838AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5839 : BnAudioTrack(), 5840 mTrack(track) 5841{ 5842} 5843 5844AudioFlinger::TrackHandle::~TrackHandle() { 5845 // just stop the track on deletion, associated resources 5846 // will be freed from the main thread once all pending buffers have 5847 // been played. Unless it's not in the active track list, in which 5848 // case we free everything now... 5849 mTrack->destroy(); 5850} 5851 5852sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5853 return mTrack->getCblk(); 5854} 5855 5856status_t AudioFlinger::TrackHandle::start() { 5857 return mTrack->start(); 5858} 5859 5860void AudioFlinger::TrackHandle::stop() { 5861 mTrack->stop(); 5862} 5863 5864void AudioFlinger::TrackHandle::flush() { 5865 mTrack->flush(); 5866} 5867 5868void AudioFlinger::TrackHandle::mute(bool e) { 5869 mTrack->mute(e); 5870} 5871 5872void AudioFlinger::TrackHandle::pause() { 5873 mTrack->pause(); 5874} 5875 5876status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5877{ 5878 return mTrack->attachAuxEffect(EffectId); 5879} 5880 5881status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5882 sp<IMemory>* buffer) { 5883 if (!mTrack->isTimedTrack()) 5884 return INVALID_OPERATION; 5885 5886 PlaybackThread::TimedTrack* tt = 5887 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5888 return tt->allocateTimedBuffer(size, buffer); 5889} 5890 5891status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5892 int64_t pts) { 5893 if (!mTrack->isTimedTrack()) 5894 return INVALID_OPERATION; 5895 5896 PlaybackThread::TimedTrack* tt = 5897 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5898 return tt->queueTimedBuffer(buffer, pts); 5899} 5900 5901status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5902 const LinearTransform& xform, int target) { 5903 5904 if (!mTrack->isTimedTrack()) 5905 return INVALID_OPERATION; 5906 5907 PlaybackThread::TimedTrack* tt = 5908 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5909 return tt->setMediaTimeTransform( 5910 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5911} 5912 5913status_t AudioFlinger::TrackHandle::onTransact( 5914 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5915{ 5916 return BnAudioTrack::onTransact(code, data, reply, flags); 5917} 5918 5919// ---------------------------------------------------------------------------- 5920 5921sp<IAudioRecord> AudioFlinger::openRecord( 5922 pid_t pid, 5923 audio_io_handle_t input, 5924 uint32_t sampleRate, 5925 audio_format_t format, 5926 audio_channel_mask_t channelMask, 5927 int frameCount, 5928 IAudioFlinger::track_flags_t flags, 5929 pid_t tid, 5930 int *sessionId, 5931 status_t *status) 5932{ 5933 sp<RecordThread::RecordTrack> recordTrack; 5934 sp<RecordHandle> recordHandle; 5935 sp<Client> client; 5936 status_t lStatus; 5937 RecordThread *thread; 5938 size_t inFrameCount; 5939 int lSessionId; 5940 5941 // check calling permissions 5942 if (!recordingAllowed()) { 5943 lStatus = PERMISSION_DENIED; 5944 goto Exit; 5945 } 5946 5947 // add client to list 5948 { // scope for mLock 5949 Mutex::Autolock _l(mLock); 5950 thread = checkRecordThread_l(input); 5951 if (thread == NULL) { 5952 lStatus = BAD_VALUE; 5953 goto Exit; 5954 } 5955 5956 client = registerPid_l(pid); 5957 5958 // If no audio session id is provided, create one here 5959 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5960 lSessionId = *sessionId; 5961 } else { 5962 lSessionId = nextUniqueId(); 5963 if (sessionId != NULL) { 5964 *sessionId = lSessionId; 5965 } 5966 } 5967 // create new record track. 5968 // The record track uses one track in mHardwareMixerThread by convention. 5969 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5970 frameCount, lSessionId, flags, tid, &lStatus); 5971 } 5972 if (lStatus != NO_ERROR) { 5973 // remove local strong reference to Client before deleting the RecordTrack so that the 5974 // Client destructor is called by the TrackBase destructor with mLock held 5975 client.clear(); 5976 recordTrack.clear(); 5977 goto Exit; 5978 } 5979 5980 // return to handle to client 5981 recordHandle = new RecordHandle(recordTrack); 5982 lStatus = NO_ERROR; 5983 5984Exit: 5985 if (status) { 5986 *status = lStatus; 5987 } 5988 return recordHandle; 5989} 5990 5991// ---------------------------------------------------------------------------- 5992 5993AudioFlinger::RecordHandle::RecordHandle( 5994 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5995 : BnAudioRecord(), 5996 mRecordTrack(recordTrack) 5997{ 5998} 5999 6000AudioFlinger::RecordHandle::~RecordHandle() { 6001 stop_nonvirtual(); 6002 mRecordTrack->destroy(); 6003} 6004 6005sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 6006 return mRecordTrack->getCblk(); 6007} 6008 6009status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 6010 int triggerSession) { 6011 ALOGV("RecordHandle::start()"); 6012 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 6013} 6014 6015void AudioFlinger::RecordHandle::stop() { 6016 stop_nonvirtual(); 6017} 6018 6019void AudioFlinger::RecordHandle::stop_nonvirtual() { 6020 ALOGV("RecordHandle::stop()"); 6021 mRecordTrack->stop(); 6022} 6023 6024status_t AudioFlinger::RecordHandle::onTransact( 6025 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6026{ 6027 return BnAudioRecord::onTransact(code, data, reply, flags); 6028} 6029 6030// ---------------------------------------------------------------------------- 6031 6032AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6033 AudioStreamIn *input, 6034 uint32_t sampleRate, 6035 audio_channel_mask_t channelMask, 6036 audio_io_handle_t id, 6037 audio_devices_t device, 6038 const sp<NBAIO_Sink>& teeSink) : 6039 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6040 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6041 // mRsmpInIndex and mInputBytes set by readInputParameters() 6042 mReqChannelCount(popcount(channelMask)), 6043 mReqSampleRate(sampleRate), 6044 // mBytesRead is only meaningful while active, and so is cleared in start() 6045 // (but might be better to also clear here for dump?) 6046 mTeeSink(teeSink) 6047{ 6048 snprintf(mName, kNameLength, "AudioIn_%X", id); 6049 6050 readInputParameters(); 6051 6052} 6053 6054 6055AudioFlinger::RecordThread::~RecordThread() 6056{ 6057 delete[] mRsmpInBuffer; 6058 delete mResampler; 6059 delete[] mRsmpOutBuffer; 6060} 6061 6062void AudioFlinger::RecordThread::onFirstRef() 6063{ 6064 run(mName, PRIORITY_URGENT_AUDIO); 6065} 6066 6067status_t AudioFlinger::RecordThread::readyToRun() 6068{ 6069 status_t status = initCheck(); 6070 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6071 return status; 6072} 6073 6074bool AudioFlinger::RecordThread::threadLoop() 6075{ 6076 AudioBufferProvider::Buffer buffer; 6077 sp<RecordTrack> activeTrack; 6078 Vector< sp<EffectChain> > effectChains; 6079 6080 nsecs_t lastWarning = 0; 6081 6082 inputStandBy(); 6083 acquireWakeLock(); 6084 6085 // used to verify we've read at least once before evaluating how many bytes were read 6086 bool readOnce = false; 6087 6088 // start recording 6089 while (!exitPending()) { 6090 6091 processConfigEvents(); 6092 6093 { // scope for mLock 6094 Mutex::Autolock _l(mLock); 6095 checkForNewParameters_l(); 6096 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6097 standby(); 6098 6099 if (exitPending()) break; 6100 6101 releaseWakeLock_l(); 6102 ALOGV("RecordThread: loop stopping"); 6103 // go to sleep 6104 mWaitWorkCV.wait(mLock); 6105 ALOGV("RecordThread: loop starting"); 6106 acquireWakeLock_l(); 6107 continue; 6108 } 6109 if (mActiveTrack != 0) { 6110 if (mActiveTrack->mState == TrackBase::PAUSING) { 6111 standby(); 6112 mActiveTrack.clear(); 6113 mStartStopCond.broadcast(); 6114 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6115 if (mReqChannelCount != mActiveTrack->channelCount()) { 6116 mActiveTrack.clear(); 6117 mStartStopCond.broadcast(); 6118 } else if (readOnce) { 6119 // record start succeeds only if first read from audio input 6120 // succeeds 6121 if (mBytesRead >= 0) { 6122 mActiveTrack->mState = TrackBase::ACTIVE; 6123 } else { 6124 mActiveTrack.clear(); 6125 } 6126 mStartStopCond.broadcast(); 6127 } 6128 mStandby = false; 6129 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6130 removeTrack_l(mActiveTrack); 6131 mActiveTrack.clear(); 6132 } 6133 } 6134 lockEffectChains_l(effectChains); 6135 } 6136 6137 if (mActiveTrack != 0) { 6138 if (mActiveTrack->mState != TrackBase::ACTIVE && 6139 mActiveTrack->mState != TrackBase::RESUMING) { 6140 unlockEffectChains(effectChains); 6141 usleep(kRecordThreadSleepUs); 6142 continue; 6143 } 6144 for (size_t i = 0; i < effectChains.size(); i ++) { 6145 effectChains[i]->process_l(); 6146 } 6147 6148 buffer.frameCount = mFrameCount; 6149 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6150 readOnce = true; 6151 size_t framesOut = buffer.frameCount; 6152 if (mResampler == NULL) { 6153 // no resampling 6154 while (framesOut) { 6155 size_t framesIn = mFrameCount - mRsmpInIndex; 6156 if (framesIn) { 6157 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6158 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 6159 mActiveTrack->mFrameSize; 6160 if (framesIn > framesOut) 6161 framesIn = framesOut; 6162 mRsmpInIndex += framesIn; 6163 framesOut -= framesIn; 6164 if ((int)mChannelCount == mReqChannelCount || 6165 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6166 memcpy(dst, src, framesIn * mFrameSize); 6167 } else { 6168 if (mChannelCount == 1) { 6169 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6170 (int16_t *)src, framesIn); 6171 } else { 6172 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6173 (int16_t *)src, framesIn); 6174 } 6175 } 6176 } 6177 if (framesOut && mFrameCount == mRsmpInIndex) { 6178 void *readInto; 6179 if (framesOut == mFrameCount && 6180 ((int)mChannelCount == mReqChannelCount || 6181 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6182 readInto = buffer.raw; 6183 framesOut = 0; 6184 } else { 6185 readInto = mRsmpInBuffer; 6186 mRsmpInIndex = 0; 6187 } 6188 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6189 if (mBytesRead <= 0) { 6190 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6191 { 6192 ALOGE("Error reading audio input"); 6193 // Force input into standby so that it tries to 6194 // recover at next read attempt 6195 inputStandBy(); 6196 usleep(kRecordThreadSleepUs); 6197 } 6198 mRsmpInIndex = mFrameCount; 6199 framesOut = 0; 6200 buffer.frameCount = 0; 6201 } else if (mTeeSink != 0) { 6202 (void) mTeeSink->write(readInto, 6203 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6204 } 6205 } 6206 } 6207 } else { 6208 // resampling 6209 6210 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6211 // alter output frame count as if we were expecting stereo samples 6212 if (mChannelCount == 1 && mReqChannelCount == 1) { 6213 framesOut >>= 1; 6214 } 6215 mResampler->resample(mRsmpOutBuffer, framesOut, 6216 this /* AudioBufferProvider* */); 6217 // ditherAndClamp() works as long as all buffers returned by 6218 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 6219 if (mChannelCount == 2 && mReqChannelCount == 1) { 6220 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6221 // the resampler always outputs stereo samples: 6222 // do post stereo to mono conversion 6223 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6224 framesOut); 6225 } else { 6226 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6227 } 6228 6229 } 6230 if (mFramestoDrop == 0) { 6231 mActiveTrack->releaseBuffer(&buffer); 6232 } else { 6233 if (mFramestoDrop > 0) { 6234 mFramestoDrop -= buffer.frameCount; 6235 if (mFramestoDrop <= 0) { 6236 clearSyncStartEvent(); 6237 } 6238 } else { 6239 mFramestoDrop += buffer.frameCount; 6240 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6241 mSyncStartEvent->isCancelled()) { 6242 ALOGW("Synced record %s, session %d, trigger session %d", 6243 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6244 mActiveTrack->sessionId(), 6245 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6246 clearSyncStartEvent(); 6247 } 6248 } 6249 } 6250 mActiveTrack->clearOverflow(); 6251 } 6252 // client isn't retrieving buffers fast enough 6253 else { 6254 if (!mActiveTrack->setOverflow()) { 6255 nsecs_t now = systemTime(); 6256 if ((now - lastWarning) > kWarningThrottleNs) { 6257 ALOGW("RecordThread: buffer overflow"); 6258 lastWarning = now; 6259 } 6260 } 6261 // Release the processor for a while before asking for a new buffer. 6262 // This will give the application more chance to read from the buffer and 6263 // clear the overflow. 6264 usleep(kRecordThreadSleepUs); 6265 } 6266 } 6267 // enable changes in effect chain 6268 unlockEffectChains(effectChains); 6269 effectChains.clear(); 6270 } 6271 6272 standby(); 6273 6274 { 6275 Mutex::Autolock _l(mLock); 6276 mActiveTrack.clear(); 6277 mStartStopCond.broadcast(); 6278 } 6279 6280 releaseWakeLock(); 6281 6282 ALOGV("RecordThread %p exiting", this); 6283 return false; 6284} 6285 6286void AudioFlinger::RecordThread::standby() 6287{ 6288 if (!mStandby) { 6289 inputStandBy(); 6290 mStandby = true; 6291 } 6292} 6293 6294void AudioFlinger::RecordThread::inputStandBy() 6295{ 6296 mInput->stream->common.standby(&mInput->stream->common); 6297} 6298 6299sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6300 const sp<AudioFlinger::Client>& client, 6301 uint32_t sampleRate, 6302 audio_format_t format, 6303 audio_channel_mask_t channelMask, 6304 int frameCount, 6305 int sessionId, 6306 IAudioFlinger::track_flags_t flags, 6307 pid_t tid, 6308 status_t *status) 6309{ 6310 sp<RecordTrack> track; 6311 status_t lStatus; 6312 6313 lStatus = initCheck(); 6314 if (lStatus != NO_ERROR) { 6315 ALOGE("Audio driver not initialized."); 6316 goto Exit; 6317 } 6318 6319 // FIXME use flags and tid similar to createTrack_l() 6320 6321 { // scope for mLock 6322 Mutex::Autolock _l(mLock); 6323 6324 track = new RecordTrack(this, client, sampleRate, 6325 format, channelMask, frameCount, sessionId); 6326 6327 if (track->getCblk() == 0) { 6328 lStatus = NO_MEMORY; 6329 goto Exit; 6330 } 6331 mTracks.add(track); 6332 6333 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6334 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6335 mAudioFlinger->btNrecIsOff(); 6336 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6337 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6338 } 6339 lStatus = NO_ERROR; 6340 6341Exit: 6342 if (status) { 6343 *status = lStatus; 6344 } 6345 return track; 6346} 6347 6348status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6349 AudioSystem::sync_event_t event, 6350 int triggerSession) 6351{ 6352 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6353 sp<ThreadBase> strongMe = this; 6354 status_t status = NO_ERROR; 6355 6356 if (event == AudioSystem::SYNC_EVENT_NONE) { 6357 clearSyncStartEvent(); 6358 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6359 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6360 triggerSession, 6361 recordTrack->sessionId(), 6362 syncStartEventCallback, 6363 this); 6364 // Sync event can be cancelled by the trigger session if the track is not in a 6365 // compatible state in which case we start record immediately 6366 if (mSyncStartEvent->isCancelled()) { 6367 clearSyncStartEvent(); 6368 } else { 6369 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6370 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6371 } 6372 } 6373 6374 { 6375 AutoMutex lock(mLock); 6376 if (mActiveTrack != 0) { 6377 if (recordTrack != mActiveTrack.get()) { 6378 status = -EBUSY; 6379 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6380 mActiveTrack->mState = TrackBase::ACTIVE; 6381 } 6382 return status; 6383 } 6384 6385 recordTrack->mState = TrackBase::IDLE; 6386 mActiveTrack = recordTrack; 6387 mLock.unlock(); 6388 status_t status = AudioSystem::startInput(mId); 6389 mLock.lock(); 6390 if (status != NO_ERROR) { 6391 mActiveTrack.clear(); 6392 clearSyncStartEvent(); 6393 return status; 6394 } 6395 mRsmpInIndex = mFrameCount; 6396 mBytesRead = 0; 6397 if (mResampler != NULL) { 6398 mResampler->reset(); 6399 } 6400 mActiveTrack->mState = TrackBase::RESUMING; 6401 // signal thread to start 6402 ALOGV("Signal record thread"); 6403 mWaitWorkCV.broadcast(); 6404 // do not wait for mStartStopCond if exiting 6405 if (exitPending()) { 6406 mActiveTrack.clear(); 6407 status = INVALID_OPERATION; 6408 goto startError; 6409 } 6410 mStartStopCond.wait(mLock); 6411 if (mActiveTrack == 0) { 6412 ALOGV("Record failed to start"); 6413 status = BAD_VALUE; 6414 goto startError; 6415 } 6416 ALOGV("Record started OK"); 6417 return status; 6418 } 6419startError: 6420 AudioSystem::stopInput(mId); 6421 clearSyncStartEvent(); 6422 return status; 6423} 6424 6425void AudioFlinger::RecordThread::clearSyncStartEvent() 6426{ 6427 if (mSyncStartEvent != 0) { 6428 mSyncStartEvent->cancel(); 6429 } 6430 mSyncStartEvent.clear(); 6431 mFramestoDrop = 0; 6432} 6433 6434void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6435{ 6436 sp<SyncEvent> strongEvent = event.promote(); 6437 6438 if (strongEvent != 0) { 6439 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6440 me->handleSyncStartEvent(strongEvent); 6441 } 6442} 6443 6444void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6445{ 6446 if (event == mSyncStartEvent) { 6447 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6448 // from audio HAL 6449 mFramestoDrop = mFrameCount * 2; 6450 } 6451} 6452 6453bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6454 ALOGV("RecordThread::stop"); 6455 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6456 return false; 6457 } 6458 recordTrack->mState = TrackBase::PAUSING; 6459 // do not wait for mStartStopCond if exiting 6460 if (exitPending()) { 6461 return true; 6462 } 6463 mStartStopCond.wait(mLock); 6464 // if we have been restarted, recordTrack == mActiveTrack.get() here 6465 if (exitPending() || recordTrack != mActiveTrack.get()) { 6466 ALOGV("Record stopped OK"); 6467 return true; 6468 } 6469 return false; 6470} 6471 6472bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6473{ 6474 return false; 6475} 6476 6477status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6478{ 6479#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6480 if (!isValidSyncEvent(event)) { 6481 return BAD_VALUE; 6482 } 6483 6484 int eventSession = event->triggerSession(); 6485 status_t ret = NAME_NOT_FOUND; 6486 6487 Mutex::Autolock _l(mLock); 6488 6489 for (size_t i = 0; i < mTracks.size(); i++) { 6490 sp<RecordTrack> track = mTracks[i]; 6491 if (eventSession == track->sessionId()) { 6492 (void) track->setSyncEvent(event); 6493 ret = NO_ERROR; 6494 } 6495 } 6496 return ret; 6497#else 6498 return BAD_VALUE; 6499#endif 6500} 6501 6502void AudioFlinger::RecordThread::RecordTrack::destroy() 6503{ 6504 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6505 sp<RecordTrack> keep(this); 6506 { 6507 sp<ThreadBase> thread = mThread.promote(); 6508 if (thread != 0) { 6509 if (mState == ACTIVE || mState == RESUMING) { 6510 AudioSystem::stopInput(thread->id()); 6511 } 6512 AudioSystem::releaseInput(thread->id()); 6513 Mutex::Autolock _l(thread->mLock); 6514 RecordThread *recordThread = (RecordThread *) thread.get(); 6515 recordThread->destroyTrack_l(this); 6516 } 6517 } 6518} 6519 6520// destroyTrack_l() must be called with ThreadBase::mLock held 6521void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6522{ 6523 track->mState = TrackBase::TERMINATED; 6524 // active tracks are removed by threadLoop() 6525 if (mActiveTrack != track) { 6526 removeTrack_l(track); 6527 } 6528} 6529 6530void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6531{ 6532 mTracks.remove(track); 6533 // need anything related to effects here? 6534} 6535 6536void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6537{ 6538 dumpInternals(fd, args); 6539 dumpTracks(fd, args); 6540 dumpEffectChains(fd, args); 6541} 6542 6543void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6544{ 6545 const size_t SIZE = 256; 6546 char buffer[SIZE]; 6547 String8 result; 6548 6549 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6550 result.append(buffer); 6551 6552 if (mActiveTrack != 0) { 6553 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6554 result.append(buffer); 6555 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6556 result.append(buffer); 6557 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6558 result.append(buffer); 6559 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6560 result.append(buffer); 6561 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 6562 result.append(buffer); 6563 } else { 6564 result.append("No active record client\n"); 6565 } 6566 6567 write(fd, result.string(), result.size()); 6568 6569 dumpBase(fd, args); 6570} 6571 6572void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6573{ 6574 const size_t SIZE = 256; 6575 char buffer[SIZE]; 6576 String8 result; 6577 6578 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6579 result.append(buffer); 6580 RecordTrack::appendDumpHeader(result); 6581 for (size_t i = 0; i < mTracks.size(); ++i) { 6582 sp<RecordTrack> track = mTracks[i]; 6583 if (track != 0) { 6584 track->dump(buffer, SIZE); 6585 result.append(buffer); 6586 } 6587 } 6588 6589 if (mActiveTrack != 0) { 6590 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6591 result.append(buffer); 6592 RecordTrack::appendDumpHeader(result); 6593 mActiveTrack->dump(buffer, SIZE); 6594 result.append(buffer); 6595 6596 } 6597 write(fd, result.string(), result.size()); 6598} 6599 6600// AudioBufferProvider interface 6601status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6602{ 6603 size_t framesReq = buffer->frameCount; 6604 size_t framesReady = mFrameCount - mRsmpInIndex; 6605 int channelCount; 6606 6607 if (framesReady == 0) { 6608 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6609 if (mBytesRead <= 0) { 6610 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6611 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6612 // Force input into standby so that it tries to 6613 // recover at next read attempt 6614 inputStandBy(); 6615 usleep(kRecordThreadSleepUs); 6616 } 6617 buffer->raw = NULL; 6618 buffer->frameCount = 0; 6619 return NOT_ENOUGH_DATA; 6620 } 6621 mRsmpInIndex = 0; 6622 framesReady = mFrameCount; 6623 } 6624 6625 if (framesReq > framesReady) { 6626 framesReq = framesReady; 6627 } 6628 6629 if (mChannelCount == 1 && mReqChannelCount == 2) { 6630 channelCount = 1; 6631 } else { 6632 channelCount = 2; 6633 } 6634 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6635 buffer->frameCount = framesReq; 6636 return NO_ERROR; 6637} 6638 6639// AudioBufferProvider interface 6640void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6641{ 6642 mRsmpInIndex += buffer->frameCount; 6643 buffer->frameCount = 0; 6644} 6645 6646bool AudioFlinger::RecordThread::checkForNewParameters_l() 6647{ 6648 bool reconfig = false; 6649 6650 while (!mNewParameters.isEmpty()) { 6651 status_t status = NO_ERROR; 6652 String8 keyValuePair = mNewParameters[0]; 6653 AudioParameter param = AudioParameter(keyValuePair); 6654 int value; 6655 audio_format_t reqFormat = mFormat; 6656 uint32_t reqSamplingRate = mReqSampleRate; 6657 int reqChannelCount = mReqChannelCount; 6658 6659 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6660 reqSamplingRate = value; 6661 reconfig = true; 6662 } 6663 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6664 reqFormat = (audio_format_t) value; 6665 reconfig = true; 6666 } 6667 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6668 reqChannelCount = popcount(value); 6669 reconfig = true; 6670 } 6671 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6672 // do not accept frame count changes if tracks are open as the track buffer 6673 // size depends on frame count and correct behavior would not be guaranteed 6674 // if frame count is changed after track creation 6675 if (mActiveTrack != 0) { 6676 status = INVALID_OPERATION; 6677 } else { 6678 reconfig = true; 6679 } 6680 } 6681 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6682 // forward device change to effects that have requested to be 6683 // aware of attached audio device. 6684 for (size_t i = 0; i < mEffectChains.size(); i++) { 6685 mEffectChains[i]->setDevice_l(value); 6686 } 6687 6688 // store input device and output device but do not forward output device to audio HAL. 6689 // Note that status is ignored by the caller for output device 6690 // (see AudioFlinger::setParameters() 6691 if (audio_is_output_devices(value)) { 6692 mOutDevice = value; 6693 status = BAD_VALUE; 6694 } else { 6695 mInDevice = value; 6696 // disable AEC and NS if the device is a BT SCO headset supporting those 6697 // pre processings 6698 if (mTracks.size() > 0) { 6699 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6700 mAudioFlinger->btNrecIsOff(); 6701 for (size_t i = 0; i < mTracks.size(); i++) { 6702 sp<RecordTrack> track = mTracks[i]; 6703 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6704 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6705 } 6706 } 6707 } 6708 } 6709 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6710 mAudioSource != (audio_source_t)value) { 6711 // forward device change to effects that have requested to be 6712 // aware of attached audio device. 6713 for (size_t i = 0; i < mEffectChains.size(); i++) { 6714 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6715 } 6716 mAudioSource = (audio_source_t)value; 6717 } 6718 if (status == NO_ERROR) { 6719 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6720 keyValuePair.string()); 6721 if (status == INVALID_OPERATION) { 6722 inputStandBy(); 6723 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6724 keyValuePair.string()); 6725 } 6726 if (reconfig) { 6727 if (status == BAD_VALUE && 6728 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6729 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6730 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 6731 <= (2 * reqSamplingRate)) && 6732 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 6733 <= FCC_2 && 6734 (reqChannelCount <= FCC_2)) { 6735 status = NO_ERROR; 6736 } 6737 if (status == NO_ERROR) { 6738 readInputParameters(); 6739 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6740 } 6741 } 6742 } 6743 6744 mNewParameters.removeAt(0); 6745 6746 mParamStatus = status; 6747 mParamCond.signal(); 6748 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6749 // already timed out waiting for the status and will never signal the condition. 6750 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6751 } 6752 return reconfig; 6753} 6754 6755String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6756{ 6757 char *s; 6758 String8 out_s8 = String8(); 6759 6760 Mutex::Autolock _l(mLock); 6761 if (initCheck() != NO_ERROR) { 6762 return out_s8; 6763 } 6764 6765 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6766 out_s8 = String8(s); 6767 free(s); 6768 return out_s8; 6769} 6770 6771void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6772 AudioSystem::OutputDescriptor desc; 6773 void *param2 = NULL; 6774 6775 switch (event) { 6776 case AudioSystem::INPUT_OPENED: 6777 case AudioSystem::INPUT_CONFIG_CHANGED: 6778 desc.channels = mChannelMask; 6779 desc.samplingRate = mSampleRate; 6780 desc.format = mFormat; 6781 desc.frameCount = mFrameCount; 6782 desc.latency = 0; 6783 param2 = &desc; 6784 break; 6785 6786 case AudioSystem::INPUT_CLOSED: 6787 default: 6788 break; 6789 } 6790 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6791} 6792 6793void AudioFlinger::RecordThread::readInputParameters() 6794{ 6795 delete mRsmpInBuffer; 6796 // mRsmpInBuffer is always assigned a new[] below 6797 delete mRsmpOutBuffer; 6798 mRsmpOutBuffer = NULL; 6799 delete mResampler; 6800 mResampler = NULL; 6801 6802 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6803 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6804 mChannelCount = (uint16_t)popcount(mChannelMask); 6805 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6806 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6807 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6808 mFrameCount = mInputBytes / mFrameSize; 6809 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6810 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6811 6812 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6813 { 6814 int channelCount; 6815 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6816 // stereo to mono post process as the resampler always outputs stereo. 6817 if (mChannelCount == 1 && mReqChannelCount == 2) { 6818 channelCount = 1; 6819 } else { 6820 channelCount = 2; 6821 } 6822 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6823 mResampler->setSampleRate(mSampleRate); 6824 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6825 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6826 6827 // optmization: if mono to mono, alter input frame count as if we were inputing 6828 // stereo samples 6829 if (mChannelCount == 1 && mReqChannelCount == 1) { 6830 mFrameCount >>= 1; 6831 } 6832 6833 } 6834 mRsmpInIndex = mFrameCount; 6835} 6836 6837unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6838{ 6839 Mutex::Autolock _l(mLock); 6840 if (initCheck() != NO_ERROR) { 6841 return 0; 6842 } 6843 6844 return mInput->stream->get_input_frames_lost(mInput->stream); 6845} 6846 6847uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6848{ 6849 Mutex::Autolock _l(mLock); 6850 uint32_t result = 0; 6851 if (getEffectChain_l(sessionId) != 0) { 6852 result = EFFECT_SESSION; 6853 } 6854 6855 for (size_t i = 0; i < mTracks.size(); ++i) { 6856 if (sessionId == mTracks[i]->sessionId()) { 6857 result |= TRACK_SESSION; 6858 break; 6859 } 6860 } 6861 6862 return result; 6863} 6864 6865KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6866{ 6867 KeyedVector<int, bool> ids; 6868 Mutex::Autolock _l(mLock); 6869 for (size_t j = 0; j < mTracks.size(); ++j) { 6870 sp<RecordThread::RecordTrack> track = mTracks[j]; 6871 int sessionId = track->sessionId(); 6872 if (ids.indexOfKey(sessionId) < 0) { 6873 ids.add(sessionId, true); 6874 } 6875 } 6876 return ids; 6877} 6878 6879AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6880{ 6881 Mutex::Autolock _l(mLock); 6882 AudioStreamIn *input = mInput; 6883 mInput = NULL; 6884 return input; 6885} 6886 6887// this method must always be called either with ThreadBase mLock held or inside the thread loop 6888audio_stream_t* AudioFlinger::RecordThread::stream() const 6889{ 6890 if (mInput == NULL) { 6891 return NULL; 6892 } 6893 return &mInput->stream->common; 6894} 6895 6896 6897// ---------------------------------------------------------------------------- 6898 6899audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6900{ 6901 if (!settingsAllowed()) { 6902 return 0; 6903 } 6904 Mutex::Autolock _l(mLock); 6905 return loadHwModule_l(name); 6906} 6907 6908// loadHwModule_l() must be called with AudioFlinger::mLock held 6909audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6910{ 6911 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6912 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6913 ALOGW("loadHwModule() module %s already loaded", name); 6914 return mAudioHwDevs.keyAt(i); 6915 } 6916 } 6917 6918 audio_hw_device_t *dev; 6919 6920 int rc = load_audio_interface(name, &dev); 6921 if (rc) { 6922 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6923 return 0; 6924 } 6925 6926 mHardwareStatus = AUDIO_HW_INIT; 6927 rc = dev->init_check(dev); 6928 mHardwareStatus = AUDIO_HW_IDLE; 6929 if (rc) { 6930 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6931 return 0; 6932 } 6933 6934 // Check and cache this HAL's level of support for master mute and master 6935 // volume. If this is the first HAL opened, and it supports the get 6936 // methods, use the initial values provided by the HAL as the current 6937 // master mute and volume settings. 6938 6939 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6940 { // scope for auto-lock pattern 6941 AutoMutex lock(mHardwareLock); 6942 6943 if (0 == mAudioHwDevs.size()) { 6944 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6945 if (NULL != dev->get_master_volume) { 6946 float mv; 6947 if (OK == dev->get_master_volume(dev, &mv)) { 6948 mMasterVolume = mv; 6949 } 6950 } 6951 6952 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6953 if (NULL != dev->get_master_mute) { 6954 bool mm; 6955 if (OK == dev->get_master_mute(dev, &mm)) { 6956 mMasterMute = mm; 6957 } 6958 } 6959 } 6960 6961 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6962 if ((NULL != dev->set_master_volume) && 6963 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6964 flags = static_cast<AudioHwDevice::Flags>(flags | 6965 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6966 } 6967 6968 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6969 if ((NULL != dev->set_master_mute) && 6970 (OK == dev->set_master_mute(dev, mMasterMute))) { 6971 flags = static_cast<AudioHwDevice::Flags>(flags | 6972 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6973 } 6974 6975 mHardwareStatus = AUDIO_HW_IDLE; 6976 } 6977 6978 audio_module_handle_t handle = nextUniqueId(); 6979 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6980 6981 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6982 name, dev->common.module->name, dev->common.module->id, handle); 6983 6984 return handle; 6985 6986} 6987 6988// ---------------------------------------------------------------------------- 6989 6990uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 6991{ 6992 Mutex::Autolock _l(mLock); 6993 PlaybackThread *thread = primaryPlaybackThread_l(); 6994 return thread != NULL ? thread->sampleRate() : 0; 6995} 6996 6997int32_t AudioFlinger::getPrimaryOutputFrameCount() 6998{ 6999 Mutex::Autolock _l(mLock); 7000 PlaybackThread *thread = primaryPlaybackThread_l(); 7001 return thread != NULL ? thread->frameCountHAL() : 0; 7002} 7003 7004// ---------------------------------------------------------------------------- 7005 7006audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 7007 audio_devices_t *pDevices, 7008 uint32_t *pSamplingRate, 7009 audio_format_t *pFormat, 7010 audio_channel_mask_t *pChannelMask, 7011 uint32_t *pLatencyMs, 7012 audio_output_flags_t flags) 7013{ 7014 status_t status; 7015 PlaybackThread *thread = NULL; 7016 struct audio_config config = { 7017 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7018 channel_mask: pChannelMask ? *pChannelMask : 0, 7019 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7020 }; 7021 audio_stream_out_t *outStream = NULL; 7022 AudioHwDevice *outHwDev; 7023 7024 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 7025 module, 7026 (pDevices != NULL) ? *pDevices : 0, 7027 config.sample_rate, 7028 config.format, 7029 config.channel_mask, 7030 flags); 7031 7032 if (pDevices == NULL || *pDevices == 0) { 7033 return 0; 7034 } 7035 7036 Mutex::Autolock _l(mLock); 7037 7038 outHwDev = findSuitableHwDev_l(module, *pDevices); 7039 if (outHwDev == NULL) 7040 return 0; 7041 7042 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7043 audio_io_handle_t id = nextUniqueId(); 7044 7045 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7046 7047 status = hwDevHal->open_output_stream(hwDevHal, 7048 id, 7049 *pDevices, 7050 (audio_output_flags_t)flags, 7051 &config, 7052 &outStream); 7053 7054 mHardwareStatus = AUDIO_HW_IDLE; 7055 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 7056 "Channels %x, status %d", 7057 outStream, 7058 config.sample_rate, 7059 config.format, 7060 config.channel_mask, 7061 status); 7062 7063 if (status == NO_ERROR && outStream != NULL) { 7064 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7065 7066 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7067 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7068 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7069 thread = new DirectOutputThread(this, output, id, *pDevices); 7070 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7071 } else { 7072 thread = new MixerThread(this, output, id, *pDevices); 7073 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7074 } 7075 mPlaybackThreads.add(id, thread); 7076 7077 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7078 if (pFormat != NULL) *pFormat = config.format; 7079 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7080 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7081 7082 // notify client processes of the new output creation 7083 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7084 7085 // the first primary output opened designates the primary hw device 7086 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7087 ALOGI("Using module %d has the primary audio interface", module); 7088 mPrimaryHardwareDev = outHwDev; 7089 7090 AutoMutex lock(mHardwareLock); 7091 mHardwareStatus = AUDIO_HW_SET_MODE; 7092 hwDevHal->set_mode(hwDevHal, mMode); 7093 mHardwareStatus = AUDIO_HW_IDLE; 7094 } 7095 return id; 7096 } 7097 7098 return 0; 7099} 7100 7101audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7102 audio_io_handle_t output2) 7103{ 7104 Mutex::Autolock _l(mLock); 7105 MixerThread *thread1 = checkMixerThread_l(output1); 7106 MixerThread *thread2 = checkMixerThread_l(output2); 7107 7108 if (thread1 == NULL || thread2 == NULL) { 7109 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 7110 output2); 7111 return 0; 7112 } 7113 7114 audio_io_handle_t id = nextUniqueId(); 7115 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7116 thread->addOutputTrack(thread2); 7117 mPlaybackThreads.add(id, thread); 7118 // notify client processes of the new output creation 7119 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7120 return id; 7121} 7122 7123status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7124{ 7125 return closeOutput_nonvirtual(output); 7126} 7127 7128status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7129{ 7130 // keep strong reference on the playback thread so that 7131 // it is not destroyed while exit() is executed 7132 sp<PlaybackThread> thread; 7133 { 7134 Mutex::Autolock _l(mLock); 7135 thread = checkPlaybackThread_l(output); 7136 if (thread == NULL) { 7137 return BAD_VALUE; 7138 } 7139 7140 ALOGV("closeOutput() %d", output); 7141 7142 if (thread->type() == ThreadBase::MIXER) { 7143 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7144 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7145 DuplicatingThread *dupThread = 7146 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7147 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7148 } 7149 } 7150 } 7151 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7152 mPlaybackThreads.removeItem(output); 7153 } 7154 thread->exit(); 7155 // The thread entity (active unit of execution) is no longer running here, 7156 // but the ThreadBase container still exists. 7157 7158 if (thread->type() != ThreadBase::DUPLICATING) { 7159 AudioStreamOut *out = thread->clearOutput(); 7160 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7161 // from now on thread->mOutput is NULL 7162 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7163 delete out; 7164 } 7165 return NO_ERROR; 7166} 7167 7168status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7169{ 7170 Mutex::Autolock _l(mLock); 7171 PlaybackThread *thread = checkPlaybackThread_l(output); 7172 7173 if (thread == NULL) { 7174 return BAD_VALUE; 7175 } 7176 7177 ALOGV("suspendOutput() %d", output); 7178 thread->suspend(); 7179 7180 return NO_ERROR; 7181} 7182 7183status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7184{ 7185 Mutex::Autolock _l(mLock); 7186 PlaybackThread *thread = checkPlaybackThread_l(output); 7187 7188 if (thread == NULL) { 7189 return BAD_VALUE; 7190 } 7191 7192 ALOGV("restoreOutput() %d", output); 7193 7194 thread->restore(); 7195 7196 return NO_ERROR; 7197} 7198 7199audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7200 audio_devices_t *pDevices, 7201 uint32_t *pSamplingRate, 7202 audio_format_t *pFormat, 7203 audio_channel_mask_t *pChannelMask) 7204{ 7205 status_t status; 7206 RecordThread *thread = NULL; 7207 struct audio_config config = { 7208 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7209 channel_mask: pChannelMask ? *pChannelMask : 0, 7210 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7211 }; 7212 uint32_t reqSamplingRate = config.sample_rate; 7213 audio_format_t reqFormat = config.format; 7214 audio_channel_mask_t reqChannels = config.channel_mask; 7215 audio_stream_in_t *inStream = NULL; 7216 AudioHwDevice *inHwDev; 7217 7218 if (pDevices == NULL || *pDevices == 0) { 7219 return 0; 7220 } 7221 7222 Mutex::Autolock _l(mLock); 7223 7224 inHwDev = findSuitableHwDev_l(module, *pDevices); 7225 if (inHwDev == NULL) 7226 return 0; 7227 7228 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7229 audio_io_handle_t id = nextUniqueId(); 7230 7231 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7232 &inStream); 7233 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 7234 "status %d", 7235 inStream, 7236 config.sample_rate, 7237 config.format, 7238 config.channel_mask, 7239 status); 7240 7241 // If the input could not be opened with the requested parameters and we can handle the 7242 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 7243 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 7244 if (status == BAD_VALUE && 7245 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7246 (config.sample_rate <= 2 * reqSamplingRate) && 7247 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7248 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7249 inStream = NULL; 7250 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7251 } 7252 7253 if (status == NO_ERROR && inStream != NULL) { 7254 7255 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7256 // or (re-)create if current Pipe is idle and does not match the new format 7257 sp<NBAIO_Sink> teeSink; 7258#ifdef TEE_SINK_INPUT_FRAMES 7259 enum { 7260 TEE_SINK_NO, // don't copy input 7261 TEE_SINK_NEW, // copy input using a new pipe 7262 TEE_SINK_OLD, // copy input using an existing pipe 7263 } kind; 7264 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7265 popcount(inStream->common.get_channels(&inStream->common))); 7266 if (format == Format_Invalid) { 7267 kind = TEE_SINK_NO; 7268 } else if (mRecordTeeSink == 0) { 7269 kind = TEE_SINK_NEW; 7270 } else if (mRecordTeeSink->getStrongCount() != 1) { 7271 kind = TEE_SINK_NO; 7272 } else if (format == mRecordTeeSink->format()) { 7273 kind = TEE_SINK_OLD; 7274 } else { 7275 kind = TEE_SINK_NEW; 7276 } 7277 switch (kind) { 7278 case TEE_SINK_NEW: { 7279 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7280 size_t numCounterOffers = 0; 7281 const NBAIO_Format offers[1] = {format}; 7282 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7283 ALOG_ASSERT(index == 0); 7284 PipeReader *pipeReader = new PipeReader(*pipe); 7285 numCounterOffers = 0; 7286 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7287 ALOG_ASSERT(index == 0); 7288 mRecordTeeSink = pipe; 7289 mRecordTeeSource = pipeReader; 7290 teeSink = pipe; 7291 } 7292 break; 7293 case TEE_SINK_OLD: 7294 teeSink = mRecordTeeSink; 7295 break; 7296 case TEE_SINK_NO: 7297 default: 7298 break; 7299 } 7300#endif 7301 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7302 7303 // Start record thread 7304 // RecorThread require both input and output device indication to forward to audio 7305 // pre processing modules 7306 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7307 7308 thread = new RecordThread(this, 7309 input, 7310 reqSamplingRate, 7311 reqChannels, 7312 id, 7313 device, teeSink); 7314 mRecordThreads.add(id, thread); 7315 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7316 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7317 if (pFormat != NULL) *pFormat = config.format; 7318 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7319 7320 // notify client processes of the new input creation 7321 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7322 return id; 7323 } 7324 7325 return 0; 7326} 7327 7328status_t AudioFlinger::closeInput(audio_io_handle_t input) 7329{ 7330 return closeInput_nonvirtual(input); 7331} 7332 7333status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7334{ 7335 // keep strong reference on the record thread so that 7336 // it is not destroyed while exit() is executed 7337 sp<RecordThread> thread; 7338 { 7339 Mutex::Autolock _l(mLock); 7340 thread = checkRecordThread_l(input); 7341 if (thread == 0) { 7342 return BAD_VALUE; 7343 } 7344 7345 ALOGV("closeInput() %d", input); 7346 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7347 mRecordThreads.removeItem(input); 7348 } 7349 thread->exit(); 7350 // The thread entity (active unit of execution) is no longer running here, 7351 // but the ThreadBase container still exists. 7352 7353 AudioStreamIn *in = thread->clearInput(); 7354 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7355 // from now on thread->mInput is NULL 7356 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7357 delete in; 7358 7359 return NO_ERROR; 7360} 7361 7362status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7363{ 7364 Mutex::Autolock _l(mLock); 7365 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7366 7367 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7368 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7369 thread->invalidateTracks(stream); 7370 } 7371 7372 return NO_ERROR; 7373} 7374 7375 7376int AudioFlinger::newAudioSessionId() 7377{ 7378 return nextUniqueId(); 7379} 7380 7381void AudioFlinger::acquireAudioSessionId(int audioSession) 7382{ 7383 Mutex::Autolock _l(mLock); 7384 pid_t caller = IPCThreadState::self()->getCallingPid(); 7385 ALOGV("acquiring %d from %d", audioSession, caller); 7386 size_t num = mAudioSessionRefs.size(); 7387 for (size_t i = 0; i< num; i++) { 7388 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7389 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7390 ref->mCnt++; 7391 ALOGV(" incremented refcount to %d", ref->mCnt); 7392 return; 7393 } 7394 } 7395 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7396 ALOGV(" added new entry for %d", audioSession); 7397} 7398 7399void AudioFlinger::releaseAudioSessionId(int audioSession) 7400{ 7401 Mutex::Autolock _l(mLock); 7402 pid_t caller = IPCThreadState::self()->getCallingPid(); 7403 ALOGV("releasing %d from %d", audioSession, caller); 7404 size_t num = mAudioSessionRefs.size(); 7405 for (size_t i = 0; i< num; i++) { 7406 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7407 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7408 ref->mCnt--; 7409 ALOGV(" decremented refcount to %d", ref->mCnt); 7410 if (ref->mCnt == 0) { 7411 mAudioSessionRefs.removeAt(i); 7412 delete ref; 7413 purgeStaleEffects_l(); 7414 } 7415 return; 7416 } 7417 } 7418 ALOGW("session id %d not found for pid %d", audioSession, caller); 7419} 7420 7421void AudioFlinger::purgeStaleEffects_l() { 7422 7423 ALOGV("purging stale effects"); 7424 7425 Vector< sp<EffectChain> > chains; 7426 7427 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7428 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7429 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7430 sp<EffectChain> ec = t->mEffectChains[j]; 7431 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7432 chains.push(ec); 7433 } 7434 } 7435 } 7436 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7437 sp<RecordThread> t = mRecordThreads.valueAt(i); 7438 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7439 sp<EffectChain> ec = t->mEffectChains[j]; 7440 chains.push(ec); 7441 } 7442 } 7443 7444 for (size_t i = 0; i < chains.size(); i++) { 7445 sp<EffectChain> ec = chains[i]; 7446 int sessionid = ec->sessionId(); 7447 sp<ThreadBase> t = ec->mThread.promote(); 7448 if (t == 0) { 7449 continue; 7450 } 7451 size_t numsessionrefs = mAudioSessionRefs.size(); 7452 bool found = false; 7453 for (size_t k = 0; k < numsessionrefs; k++) { 7454 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7455 if (ref->mSessionid == sessionid) { 7456 ALOGV(" session %d still exists for %d with %d refs", 7457 sessionid, ref->mPid, ref->mCnt); 7458 found = true; 7459 break; 7460 } 7461 } 7462 if (!found) { 7463 Mutex::Autolock _l (t->mLock); 7464 // remove all effects from the chain 7465 while (ec->mEffects.size()) { 7466 sp<EffectModule> effect = ec->mEffects[0]; 7467 effect->unPin(); 7468 t->removeEffect_l(effect); 7469 if (effect->purgeHandles()) { 7470 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7471 } 7472 AudioSystem::unregisterEffect(effect->id()); 7473 } 7474 } 7475 } 7476 return; 7477} 7478 7479// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7480AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7481{ 7482 return mPlaybackThreads.valueFor(output).get(); 7483} 7484 7485// checkMixerThread_l() must be called with AudioFlinger::mLock held 7486AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7487{ 7488 PlaybackThread *thread = checkPlaybackThread_l(output); 7489 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7490} 7491 7492// checkRecordThread_l() must be called with AudioFlinger::mLock held 7493AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7494{ 7495 return mRecordThreads.valueFor(input).get(); 7496} 7497 7498uint32_t AudioFlinger::nextUniqueId() 7499{ 7500 return android_atomic_inc(&mNextUniqueId); 7501} 7502 7503AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7504{ 7505 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7506 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7507 AudioStreamOut *output = thread->getOutput(); 7508 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7509 return thread; 7510 } 7511 } 7512 return NULL; 7513} 7514 7515audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7516{ 7517 PlaybackThread *thread = primaryPlaybackThread_l(); 7518 7519 if (thread == NULL) { 7520 return 0; 7521 } 7522 7523 return thread->outDevice(); 7524} 7525 7526sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7527 int triggerSession, 7528 int listenerSession, 7529 sync_event_callback_t callBack, 7530 void *cookie) 7531{ 7532 Mutex::Autolock _l(mLock); 7533 7534 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7535 status_t playStatus = NAME_NOT_FOUND; 7536 status_t recStatus = NAME_NOT_FOUND; 7537 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7538 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7539 if (playStatus == NO_ERROR) { 7540 return event; 7541 } 7542 } 7543 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7544 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7545 if (recStatus == NO_ERROR) { 7546 return event; 7547 } 7548 } 7549 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7550 mPendingSyncEvents.add(event); 7551 } else { 7552 ALOGV("createSyncEvent() invalid event %d", event->type()); 7553 event.clear(); 7554 } 7555 return event; 7556} 7557 7558// ---------------------------------------------------------------------------- 7559// Effect management 7560// ---------------------------------------------------------------------------- 7561 7562 7563status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7564{ 7565 Mutex::Autolock _l(mLock); 7566 return EffectQueryNumberEffects(numEffects); 7567} 7568 7569status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7570{ 7571 Mutex::Autolock _l(mLock); 7572 return EffectQueryEffect(index, descriptor); 7573} 7574 7575status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7576 effect_descriptor_t *descriptor) const 7577{ 7578 Mutex::Autolock _l(mLock); 7579 return EffectGetDescriptor(pUuid, descriptor); 7580} 7581 7582 7583sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7584 effect_descriptor_t *pDesc, 7585 const sp<IEffectClient>& effectClient, 7586 int32_t priority, 7587 audio_io_handle_t io, 7588 int sessionId, 7589 status_t *status, 7590 int *id, 7591 int *enabled) 7592{ 7593 status_t lStatus = NO_ERROR; 7594 sp<EffectHandle> handle; 7595 effect_descriptor_t desc; 7596 7597 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7598 pid, effectClient.get(), priority, sessionId, io); 7599 7600 if (pDesc == NULL) { 7601 lStatus = BAD_VALUE; 7602 goto Exit; 7603 } 7604 7605 // check audio settings permission for global effects 7606 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7607 lStatus = PERMISSION_DENIED; 7608 goto Exit; 7609 } 7610 7611 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7612 // that can only be created by audio policy manager (running in same process) 7613 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7614 lStatus = PERMISSION_DENIED; 7615 goto Exit; 7616 } 7617 7618 if (io == 0) { 7619 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7620 // output must be specified by AudioPolicyManager when using session 7621 // AUDIO_SESSION_OUTPUT_STAGE 7622 lStatus = BAD_VALUE; 7623 goto Exit; 7624 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7625 // if the output returned by getOutputForEffect() is removed before we lock the 7626 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7627 // and we will exit safely 7628 io = AudioSystem::getOutputForEffect(&desc); 7629 } 7630 } 7631 7632 { 7633 Mutex::Autolock _l(mLock); 7634 7635 7636 if (!EffectIsNullUuid(&pDesc->uuid)) { 7637 // if uuid is specified, request effect descriptor 7638 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7639 if (lStatus < 0) { 7640 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7641 goto Exit; 7642 } 7643 } else { 7644 // if uuid is not specified, look for an available implementation 7645 // of the required type in effect factory 7646 if (EffectIsNullUuid(&pDesc->type)) { 7647 ALOGW("createEffect() no effect type"); 7648 lStatus = BAD_VALUE; 7649 goto Exit; 7650 } 7651 uint32_t numEffects = 0; 7652 effect_descriptor_t d; 7653 d.flags = 0; // prevent compiler warning 7654 bool found = false; 7655 7656 lStatus = EffectQueryNumberEffects(&numEffects); 7657 if (lStatus < 0) { 7658 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7659 goto Exit; 7660 } 7661 for (uint32_t i = 0; i < numEffects; i++) { 7662 lStatus = EffectQueryEffect(i, &desc); 7663 if (lStatus < 0) { 7664 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7665 continue; 7666 } 7667 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7668 // If matching type found save effect descriptor. If the session is 7669 // 0 and the effect is not auxiliary, continue enumeration in case 7670 // an auxiliary version of this effect type is available 7671 found = true; 7672 d = desc; 7673 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7674 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7675 break; 7676 } 7677 } 7678 } 7679 if (!found) { 7680 lStatus = BAD_VALUE; 7681 ALOGW("createEffect() effect not found"); 7682 goto Exit; 7683 } 7684 // For same effect type, chose auxiliary version over insert version if 7685 // connect to output mix (Compliance to OpenSL ES) 7686 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7687 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7688 desc = d; 7689 } 7690 } 7691 7692 // Do not allow auxiliary effects on a session different from 0 (output mix) 7693 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7694 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7695 lStatus = INVALID_OPERATION; 7696 goto Exit; 7697 } 7698 7699 // check recording permission for visualizer 7700 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7701 !recordingAllowed()) { 7702 lStatus = PERMISSION_DENIED; 7703 goto Exit; 7704 } 7705 7706 // return effect descriptor 7707 *pDesc = desc; 7708 7709 // If output is not specified try to find a matching audio session ID in one of the 7710 // output threads. 7711 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7712 // because of code checking output when entering the function. 7713 // Note: io is never 0 when creating an effect on an input 7714 if (io == 0) { 7715 // look for the thread where the specified audio session is present 7716 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7717 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7718 io = mPlaybackThreads.keyAt(i); 7719 break; 7720 } 7721 } 7722 if (io == 0) { 7723 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7724 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7725 io = mRecordThreads.keyAt(i); 7726 break; 7727 } 7728 } 7729 } 7730 // If no output thread contains the requested session ID, default to 7731 // first output. The effect chain will be moved to the correct output 7732 // thread when a track with the same session ID is created 7733 if (io == 0 && mPlaybackThreads.size()) { 7734 io = mPlaybackThreads.keyAt(0); 7735 } 7736 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7737 } 7738 ThreadBase *thread = checkRecordThread_l(io); 7739 if (thread == NULL) { 7740 thread = checkPlaybackThread_l(io); 7741 if (thread == NULL) { 7742 ALOGE("createEffect() unknown output thread"); 7743 lStatus = BAD_VALUE; 7744 goto Exit; 7745 } 7746 } 7747 7748 sp<Client> client = registerPid_l(pid); 7749 7750 // create effect on selected output thread 7751 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7752 &desc, enabled, &lStatus); 7753 if (handle != 0 && id != NULL) { 7754 *id = handle->id(); 7755 } 7756 } 7757 7758Exit: 7759 if (status != NULL) { 7760 *status = lStatus; 7761 } 7762 return handle; 7763} 7764 7765status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7766 audio_io_handle_t dstOutput) 7767{ 7768 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7769 sessionId, srcOutput, dstOutput); 7770 Mutex::Autolock _l(mLock); 7771 if (srcOutput == dstOutput) { 7772 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7773 return NO_ERROR; 7774 } 7775 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7776 if (srcThread == NULL) { 7777 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7778 return BAD_VALUE; 7779 } 7780 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7781 if (dstThread == NULL) { 7782 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7783 return BAD_VALUE; 7784 } 7785 7786 Mutex::Autolock _dl(dstThread->mLock); 7787 Mutex::Autolock _sl(srcThread->mLock); 7788 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7789 7790 return NO_ERROR; 7791} 7792 7793// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7794status_t AudioFlinger::moveEffectChain_l(int sessionId, 7795 AudioFlinger::PlaybackThread *srcThread, 7796 AudioFlinger::PlaybackThread *dstThread, 7797 bool reRegister) 7798{ 7799 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7800 sessionId, srcThread, dstThread); 7801 7802 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7803 if (chain == 0) { 7804 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7805 sessionId, srcThread); 7806 return INVALID_OPERATION; 7807 } 7808 7809 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7810 // so that a new chain is created with correct parameters when first effect is added. This is 7811 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7812 // removed. 7813 srcThread->removeEffectChain_l(chain); 7814 7815 // transfer all effects one by one so that new effect chain is created on new thread with 7816 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7817 audio_io_handle_t dstOutput = dstThread->id(); 7818 sp<EffectChain> dstChain; 7819 uint32_t strategy = 0; // prevent compiler warning 7820 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7821 while (effect != 0) { 7822 srcThread->removeEffect_l(effect); 7823 dstThread->addEffect_l(effect); 7824 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7825 if (effect->state() == EffectModule::ACTIVE || 7826 effect->state() == EffectModule::STOPPING) { 7827 effect->start(); 7828 } 7829 // if the move request is not received from audio policy manager, the effect must be 7830 // re-registered with the new strategy and output 7831 if (dstChain == 0) { 7832 dstChain = effect->chain().promote(); 7833 if (dstChain == 0) { 7834 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7835 srcThread->addEffect_l(effect); 7836 return NO_INIT; 7837 } 7838 strategy = dstChain->strategy(); 7839 } 7840 if (reRegister) { 7841 AudioSystem::unregisterEffect(effect->id()); 7842 AudioSystem::registerEffect(&effect->desc(), 7843 dstOutput, 7844 strategy, 7845 sessionId, 7846 effect->id()); 7847 } 7848 effect = chain->getEffectFromId_l(0); 7849 } 7850 7851 return NO_ERROR; 7852} 7853 7854 7855// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7856sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7857 const sp<AudioFlinger::Client>& client, 7858 const sp<IEffectClient>& effectClient, 7859 int32_t priority, 7860 int sessionId, 7861 effect_descriptor_t *desc, 7862 int *enabled, 7863 status_t *status 7864 ) 7865{ 7866 sp<EffectModule> effect; 7867 sp<EffectHandle> handle; 7868 status_t lStatus; 7869 sp<EffectChain> chain; 7870 bool chainCreated = false; 7871 bool effectCreated = false; 7872 bool effectRegistered = false; 7873 7874 lStatus = initCheck(); 7875 if (lStatus != NO_ERROR) { 7876 ALOGW("createEffect_l() Audio driver not initialized."); 7877 goto Exit; 7878 } 7879 7880 // Do not allow effects with session ID 0 on direct output or duplicating threads 7881 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7882 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7883 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7884 desc->name, sessionId); 7885 lStatus = BAD_VALUE; 7886 goto Exit; 7887 } 7888 // Only Pre processor effects are allowed on input threads and only on input threads 7889 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7890 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7891 desc->name, desc->flags, mType); 7892 lStatus = BAD_VALUE; 7893 goto Exit; 7894 } 7895 7896 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7897 7898 { // scope for mLock 7899 Mutex::Autolock _l(mLock); 7900 7901 // check for existing effect chain with the requested audio session 7902 chain = getEffectChain_l(sessionId); 7903 if (chain == 0) { 7904 // create a new chain for this session 7905 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7906 chain = new EffectChain(this, sessionId); 7907 addEffectChain_l(chain); 7908 chain->setStrategy(getStrategyForSession_l(sessionId)); 7909 chainCreated = true; 7910 } else { 7911 effect = chain->getEffectFromDesc_l(desc); 7912 } 7913 7914 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7915 7916 if (effect == 0) { 7917 int id = mAudioFlinger->nextUniqueId(); 7918 // Check CPU and memory usage 7919 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7920 if (lStatus != NO_ERROR) { 7921 goto Exit; 7922 } 7923 effectRegistered = true; 7924 // create a new effect module if none present in the chain 7925 effect = new EffectModule(this, chain, desc, id, sessionId); 7926 lStatus = effect->status(); 7927 if (lStatus != NO_ERROR) { 7928 goto Exit; 7929 } 7930 lStatus = chain->addEffect_l(effect); 7931 if (lStatus != NO_ERROR) { 7932 goto Exit; 7933 } 7934 effectCreated = true; 7935 7936 effect->setDevice(mOutDevice); 7937 effect->setDevice(mInDevice); 7938 effect->setMode(mAudioFlinger->getMode()); 7939 effect->setAudioSource(mAudioSource); 7940 } 7941 // create effect handle and connect it to effect module 7942 handle = new EffectHandle(effect, client, effectClient, priority); 7943 lStatus = effect->addHandle(handle.get()); 7944 if (enabled != NULL) { 7945 *enabled = (int)effect->isEnabled(); 7946 } 7947 } 7948 7949Exit: 7950 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7951 Mutex::Autolock _l(mLock); 7952 if (effectCreated) { 7953 chain->removeEffect_l(effect); 7954 } 7955 if (effectRegistered) { 7956 AudioSystem::unregisterEffect(effect->id()); 7957 } 7958 if (chainCreated) { 7959 removeEffectChain_l(chain); 7960 } 7961 handle.clear(); 7962 } 7963 7964 if (status != NULL) { 7965 *status = lStatus; 7966 } 7967 return handle; 7968} 7969 7970sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7971{ 7972 Mutex::Autolock _l(mLock); 7973 return getEffect_l(sessionId, effectId); 7974} 7975 7976sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7977{ 7978 sp<EffectChain> chain = getEffectChain_l(sessionId); 7979 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7980} 7981 7982// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7983// PlaybackThread::mLock held 7984status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7985{ 7986 // check for existing effect chain with the requested audio session 7987 int sessionId = effect->sessionId(); 7988 sp<EffectChain> chain = getEffectChain_l(sessionId); 7989 bool chainCreated = false; 7990 7991 if (chain == 0) { 7992 // create a new chain for this session 7993 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7994 chain = new EffectChain(this, sessionId); 7995 addEffectChain_l(chain); 7996 chain->setStrategy(getStrategyForSession_l(sessionId)); 7997 chainCreated = true; 7998 } 7999 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 8000 8001 if (chain->getEffectFromId_l(effect->id()) != 0) { 8002 ALOGW("addEffect_l() %p effect %s already present in chain %p", 8003 this, effect->desc().name, chain.get()); 8004 return BAD_VALUE; 8005 } 8006 8007 status_t status = chain->addEffect_l(effect); 8008 if (status != NO_ERROR) { 8009 if (chainCreated) { 8010 removeEffectChain_l(chain); 8011 } 8012 return status; 8013 } 8014 8015 effect->setDevice(mOutDevice); 8016 effect->setDevice(mInDevice); 8017 effect->setMode(mAudioFlinger->getMode()); 8018 effect->setAudioSource(mAudioSource); 8019 return NO_ERROR; 8020} 8021 8022void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 8023 8024 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 8025 effect_descriptor_t desc = effect->desc(); 8026 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8027 detachAuxEffect_l(effect->id()); 8028 } 8029 8030 sp<EffectChain> chain = effect->chain().promote(); 8031 if (chain != 0) { 8032 // remove effect chain if removing last effect 8033 if (chain->removeEffect_l(effect) == 0) { 8034 removeEffectChain_l(chain); 8035 } 8036 } else { 8037 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 8038 } 8039} 8040 8041void AudioFlinger::ThreadBase::lockEffectChains_l( 8042 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8043{ 8044 effectChains = mEffectChains; 8045 for (size_t i = 0; i < mEffectChains.size(); i++) { 8046 mEffectChains[i]->lock(); 8047 } 8048} 8049 8050void AudioFlinger::ThreadBase::unlockEffectChains( 8051 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8052{ 8053 for (size_t i = 0; i < effectChains.size(); i++) { 8054 effectChains[i]->unlock(); 8055 } 8056} 8057 8058sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8059{ 8060 Mutex::Autolock _l(mLock); 8061 return getEffectChain_l(sessionId); 8062} 8063 8064sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8065{ 8066 size_t size = mEffectChains.size(); 8067 for (size_t i = 0; i < size; i++) { 8068 if (mEffectChains[i]->sessionId() == sessionId) { 8069 return mEffectChains[i]; 8070 } 8071 } 8072 return 0; 8073} 8074 8075void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8076{ 8077 Mutex::Autolock _l(mLock); 8078 size_t size = mEffectChains.size(); 8079 for (size_t i = 0; i < size; i++) { 8080 mEffectChains[i]->setMode_l(mode); 8081 } 8082} 8083 8084void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8085 EffectHandle *handle, 8086 bool unpinIfLast) { 8087 8088 Mutex::Autolock _l(mLock); 8089 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8090 // delete the effect module if removing last handle on it 8091 if (effect->removeHandle(handle) == 0) { 8092 if (!effect->isPinned() || unpinIfLast) { 8093 removeEffect_l(effect); 8094 AudioSystem::unregisterEffect(effect->id()); 8095 } 8096 } 8097} 8098 8099status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8100{ 8101 int session = chain->sessionId(); 8102 int16_t *buffer = mMixBuffer; 8103 bool ownsBuffer = false; 8104 8105 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8106 if (session > 0) { 8107 // Only one effect chain can be present in direct output thread and it uses 8108 // the mix buffer as input 8109 if (mType != DIRECT) { 8110 size_t numSamples = mNormalFrameCount * mChannelCount; 8111 buffer = new int16_t[numSamples]; 8112 memset(buffer, 0, numSamples * sizeof(int16_t)); 8113 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8114 ownsBuffer = true; 8115 } 8116 8117 // Attach all tracks with same session ID to this chain. 8118 for (size_t i = 0; i < mTracks.size(); ++i) { 8119 sp<Track> track = mTracks[i]; 8120 if (session == track->sessionId()) { 8121 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 8122 buffer); 8123 track->setMainBuffer(buffer); 8124 chain->incTrackCnt(); 8125 } 8126 } 8127 8128 // indicate all active tracks in the chain 8129 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8130 sp<Track> track = mActiveTracks[i].promote(); 8131 if (track == 0) continue; 8132 if (session == track->sessionId()) { 8133 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8134 chain->incActiveTrackCnt(); 8135 } 8136 } 8137 } 8138 8139 chain->setInBuffer(buffer, ownsBuffer); 8140 chain->setOutBuffer(mMixBuffer); 8141 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8142 // chains list in order to be processed last as it contains output stage effects 8143 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8144 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8145 // after track specific effects and before output stage 8146 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8147 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8148 // Effect chain for other sessions are inserted at beginning of effect 8149 // chains list to be processed before output mix effects. Relative order between other 8150 // sessions is not important 8151 size_t size = mEffectChains.size(); 8152 size_t i = 0; 8153 for (i = 0; i < size; i++) { 8154 if (mEffectChains[i]->sessionId() < session) break; 8155 } 8156 mEffectChains.insertAt(chain, i); 8157 checkSuspendOnAddEffectChain_l(chain); 8158 8159 return NO_ERROR; 8160} 8161 8162size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8163{ 8164 int session = chain->sessionId(); 8165 8166 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8167 8168 for (size_t i = 0; i < mEffectChains.size(); i++) { 8169 if (chain == mEffectChains[i]) { 8170 mEffectChains.removeAt(i); 8171 // detach all active tracks from the chain 8172 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8173 sp<Track> track = mActiveTracks[i].promote(); 8174 if (track == 0) continue; 8175 if (session == track->sessionId()) { 8176 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8177 chain.get(), session); 8178 chain->decActiveTrackCnt(); 8179 } 8180 } 8181 8182 // detach all tracks with same session ID from this chain 8183 for (size_t i = 0; i < mTracks.size(); ++i) { 8184 sp<Track> track = mTracks[i]; 8185 if (session == track->sessionId()) { 8186 track->setMainBuffer(mMixBuffer); 8187 chain->decTrackCnt(); 8188 } 8189 } 8190 break; 8191 } 8192 } 8193 return mEffectChains.size(); 8194} 8195 8196status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8197 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8198{ 8199 Mutex::Autolock _l(mLock); 8200 return attachAuxEffect_l(track, EffectId); 8201} 8202 8203status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8204 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8205{ 8206 status_t status = NO_ERROR; 8207 8208 if (EffectId == 0) { 8209 track->setAuxBuffer(0, NULL); 8210 } else { 8211 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8212 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8213 if (effect != 0) { 8214 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8215 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8216 } else { 8217 status = INVALID_OPERATION; 8218 } 8219 } else { 8220 status = BAD_VALUE; 8221 } 8222 } 8223 return status; 8224} 8225 8226void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8227{ 8228 for (size_t i = 0; i < mTracks.size(); ++i) { 8229 sp<Track> track = mTracks[i]; 8230 if (track->auxEffectId() == effectId) { 8231 attachAuxEffect_l(track, 0); 8232 } 8233 } 8234} 8235 8236status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8237{ 8238 // only one chain per input thread 8239 if (mEffectChains.size() != 0) { 8240 return INVALID_OPERATION; 8241 } 8242 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8243 8244 chain->setInBuffer(NULL); 8245 chain->setOutBuffer(NULL); 8246 8247 checkSuspendOnAddEffectChain_l(chain); 8248 8249 mEffectChains.add(chain); 8250 8251 return NO_ERROR; 8252} 8253 8254size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8255{ 8256 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8257 ALOGW_IF(mEffectChains.size() != 1, 8258 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8259 chain.get(), mEffectChains.size(), this); 8260 if (mEffectChains.size() == 1) { 8261 mEffectChains.removeAt(0); 8262 } 8263 return 0; 8264} 8265 8266// ---------------------------------------------------------------------------- 8267// EffectModule implementation 8268// ---------------------------------------------------------------------------- 8269 8270#undef LOG_TAG 8271#define LOG_TAG "AudioFlinger::EffectModule" 8272 8273AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8274 const wp<AudioFlinger::EffectChain>& chain, 8275 effect_descriptor_t *desc, 8276 int id, 8277 int sessionId) 8278 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8279 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8280 mDescriptor(*desc), 8281 // mConfig is set by configure() and not used before then 8282 mEffectInterface(NULL), 8283 mStatus(NO_INIT), mState(IDLE), 8284 // mMaxDisableWaitCnt is set by configure() and not used before then 8285 // mDisableWaitCnt is set by process() and updateState() and not used before then 8286 mSuspended(false) 8287{ 8288 ALOGV("Constructor %p", this); 8289 int lStatus; 8290 8291 // create effect engine from effect factory 8292 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8293 8294 if (mStatus != NO_ERROR) { 8295 return; 8296 } 8297 lStatus = init(); 8298 if (lStatus < 0) { 8299 mStatus = lStatus; 8300 goto Error; 8301 } 8302 8303 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8304 return; 8305Error: 8306 EffectRelease(mEffectInterface); 8307 mEffectInterface = NULL; 8308 ALOGV("Constructor Error %d", mStatus); 8309} 8310 8311AudioFlinger::EffectModule::~EffectModule() 8312{ 8313 ALOGV("Destructor %p", this); 8314 if (mEffectInterface != NULL) { 8315 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8316 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8317 sp<ThreadBase> thread = mThread.promote(); 8318 if (thread != 0) { 8319 audio_stream_t *stream = thread->stream(); 8320 if (stream != NULL) { 8321 stream->remove_audio_effect(stream, mEffectInterface); 8322 } 8323 } 8324 } 8325 // release effect engine 8326 EffectRelease(mEffectInterface); 8327 } 8328} 8329 8330status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8331{ 8332 status_t status; 8333 8334 Mutex::Autolock _l(mLock); 8335 int priority = handle->priority(); 8336 size_t size = mHandles.size(); 8337 EffectHandle *controlHandle = NULL; 8338 size_t i; 8339 for (i = 0; i < size; i++) { 8340 EffectHandle *h = mHandles[i]; 8341 if (h == NULL || h->destroyed_l()) continue; 8342 // first non destroyed handle is considered in control 8343 if (controlHandle == NULL) 8344 controlHandle = h; 8345 if (h->priority() <= priority) break; 8346 } 8347 // if inserted in first place, move effect control from previous owner to this handle 8348 if (i == 0) { 8349 bool enabled = false; 8350 if (controlHandle != NULL) { 8351 enabled = controlHandle->enabled(); 8352 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8353 } 8354 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8355 status = NO_ERROR; 8356 } else { 8357 status = ALREADY_EXISTS; 8358 } 8359 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8360 mHandles.insertAt(handle, i); 8361 return status; 8362} 8363 8364size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8365{ 8366 Mutex::Autolock _l(mLock); 8367 size_t size = mHandles.size(); 8368 size_t i; 8369 for (i = 0; i < size; i++) { 8370 if (mHandles[i] == handle) break; 8371 } 8372 if (i == size) { 8373 return size; 8374 } 8375 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8376 8377 mHandles.removeAt(i); 8378 // if removed from first place, move effect control from this handle to next in line 8379 if (i == 0) { 8380 EffectHandle *h = controlHandle_l(); 8381 if (h != NULL) { 8382 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8383 } 8384 } 8385 8386 // Prevent calls to process() and other functions on effect interface from now on. 8387 // The effect engine will be released by the destructor when the last strong reference on 8388 // this object is released which can happen after next process is called. 8389 if (mHandles.size() == 0 && !mPinned) { 8390 mState = DESTROYED; 8391 } 8392 8393 return mHandles.size(); 8394} 8395 8396// must be called with EffectModule::mLock held 8397AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8398{ 8399 // the first valid handle in the list has control over the module 8400 for (size_t i = 0; i < mHandles.size(); i++) { 8401 EffectHandle *h = mHandles[i]; 8402 if (h != NULL && !h->destroyed_l()) { 8403 return h; 8404 } 8405 } 8406 8407 return NULL; 8408} 8409 8410size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8411{ 8412 ALOGV("disconnect() %p handle %p", this, handle); 8413 // keep a strong reference on this EffectModule to avoid calling the 8414 // destructor before we exit 8415 sp<EffectModule> keep(this); 8416 { 8417 sp<ThreadBase> thread = mThread.promote(); 8418 if (thread != 0) { 8419 thread->disconnectEffect(keep, handle, unpinIfLast); 8420 } 8421 } 8422 return mHandles.size(); 8423} 8424 8425void AudioFlinger::EffectModule::updateState() { 8426 Mutex::Autolock _l(mLock); 8427 8428 switch (mState) { 8429 case RESTART: 8430 reset_l(); 8431 // FALL THROUGH 8432 8433 case STARTING: 8434 // clear auxiliary effect input buffer for next accumulation 8435 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8436 memset(mConfig.inputCfg.buffer.raw, 8437 0, 8438 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8439 } 8440 start_l(); 8441 mState = ACTIVE; 8442 break; 8443 case STOPPING: 8444 stop_l(); 8445 mDisableWaitCnt = mMaxDisableWaitCnt; 8446 mState = STOPPED; 8447 break; 8448 case STOPPED: 8449 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8450 // turn off sequence. 8451 if (--mDisableWaitCnt == 0) { 8452 reset_l(); 8453 mState = IDLE; 8454 } 8455 break; 8456 default: //IDLE , ACTIVE, DESTROYED 8457 break; 8458 } 8459} 8460 8461void AudioFlinger::EffectModule::process() 8462{ 8463 Mutex::Autolock _l(mLock); 8464 8465 if (mState == DESTROYED || mEffectInterface == NULL || 8466 mConfig.inputCfg.buffer.raw == NULL || 8467 mConfig.outputCfg.buffer.raw == NULL) { 8468 return; 8469 } 8470 8471 if (isProcessEnabled()) { 8472 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8473 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8474 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8475 mConfig.inputCfg.buffer.s32, 8476 mConfig.inputCfg.buffer.frameCount/2); 8477 } 8478 8479 // do the actual processing in the effect engine 8480 int ret = (*mEffectInterface)->process(mEffectInterface, 8481 &mConfig.inputCfg.buffer, 8482 &mConfig.outputCfg.buffer); 8483 8484 // force transition to IDLE state when engine is ready 8485 if (mState == STOPPED && ret == -ENODATA) { 8486 mDisableWaitCnt = 1; 8487 } 8488 8489 // clear auxiliary effect input buffer for next accumulation 8490 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8491 memset(mConfig.inputCfg.buffer.raw, 0, 8492 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8493 } 8494 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8495 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8496 // If an insert effect is idle and input buffer is different from output buffer, 8497 // accumulate input onto output 8498 sp<EffectChain> chain = mChain.promote(); 8499 if (chain != 0 && chain->activeTrackCnt() != 0) { 8500 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8501 int16_t *in = mConfig.inputCfg.buffer.s16; 8502 int16_t *out = mConfig.outputCfg.buffer.s16; 8503 for (size_t i = 0; i < frameCnt; i++) { 8504 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8505 } 8506 } 8507 } 8508} 8509 8510void AudioFlinger::EffectModule::reset_l() 8511{ 8512 if (mEffectInterface == NULL) { 8513 return; 8514 } 8515 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8516} 8517 8518status_t AudioFlinger::EffectModule::configure() 8519{ 8520 if (mEffectInterface == NULL) { 8521 return NO_INIT; 8522 } 8523 8524 sp<ThreadBase> thread = mThread.promote(); 8525 if (thread == 0) { 8526 return DEAD_OBJECT; 8527 } 8528 8529 // TODO: handle configuration of effects replacing track process 8530 audio_channel_mask_t channelMask = thread->channelMask(); 8531 8532 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8533 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8534 } else { 8535 mConfig.inputCfg.channels = channelMask; 8536 } 8537 mConfig.outputCfg.channels = channelMask; 8538 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8539 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8540 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8541 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8542 mConfig.inputCfg.bufferProvider.cookie = NULL; 8543 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8544 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8545 mConfig.outputCfg.bufferProvider.cookie = NULL; 8546 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8547 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8548 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8549 // Insert effect: 8550 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8551 // always overwrites output buffer: input buffer == output buffer 8552 // - in other sessions: 8553 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8554 // other effect: overwrites output buffer: input buffer == output buffer 8555 // Auxiliary effect: 8556 // accumulates in output buffer: input buffer != output buffer 8557 // Therefore: accumulate <=> input buffer != output buffer 8558 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8559 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8560 } else { 8561 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8562 } 8563 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8564 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8565 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8566 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8567 8568 ALOGV("configure() %p thread %p buffer %p framecount %d", 8569 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8570 8571 status_t cmdStatus; 8572 uint32_t size = sizeof(int); 8573 status_t status = (*mEffectInterface)->command(mEffectInterface, 8574 EFFECT_CMD_SET_CONFIG, 8575 sizeof(effect_config_t), 8576 &mConfig, 8577 &size, 8578 &cmdStatus); 8579 if (status == 0) { 8580 status = cmdStatus; 8581 } 8582 8583 if (status == 0 && 8584 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8585 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8586 effect_param_t *p = (effect_param_t *)buf32; 8587 8588 p->psize = sizeof(uint32_t); 8589 p->vsize = sizeof(uint32_t); 8590 size = sizeof(int); 8591 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8592 8593 uint32_t latency = 0; 8594 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8595 if (pbt != NULL) { 8596 latency = pbt->latency_l(); 8597 } 8598 8599 *((int32_t *)p->data + 1)= latency; 8600 (*mEffectInterface)->command(mEffectInterface, 8601 EFFECT_CMD_SET_PARAM, 8602 sizeof(effect_param_t) + 8, 8603 &buf32, 8604 &size, 8605 &cmdStatus); 8606 } 8607 8608 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8609 (1000 * mConfig.outputCfg.buffer.frameCount); 8610 8611 return status; 8612} 8613 8614status_t AudioFlinger::EffectModule::init() 8615{ 8616 Mutex::Autolock _l(mLock); 8617 if (mEffectInterface == NULL) { 8618 return NO_INIT; 8619 } 8620 status_t cmdStatus; 8621 uint32_t size = sizeof(status_t); 8622 status_t status = (*mEffectInterface)->command(mEffectInterface, 8623 EFFECT_CMD_INIT, 8624 0, 8625 NULL, 8626 &size, 8627 &cmdStatus); 8628 if (status == 0) { 8629 status = cmdStatus; 8630 } 8631 return status; 8632} 8633 8634status_t AudioFlinger::EffectModule::start() 8635{ 8636 Mutex::Autolock _l(mLock); 8637 return start_l(); 8638} 8639 8640status_t AudioFlinger::EffectModule::start_l() 8641{ 8642 if (mEffectInterface == NULL) { 8643 return NO_INIT; 8644 } 8645 status_t cmdStatus; 8646 uint32_t size = sizeof(status_t); 8647 status_t status = (*mEffectInterface)->command(mEffectInterface, 8648 EFFECT_CMD_ENABLE, 8649 0, 8650 NULL, 8651 &size, 8652 &cmdStatus); 8653 if (status == 0) { 8654 status = cmdStatus; 8655 } 8656 if (status == 0 && 8657 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8658 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8659 sp<ThreadBase> thread = mThread.promote(); 8660 if (thread != 0) { 8661 audio_stream_t *stream = thread->stream(); 8662 if (stream != NULL) { 8663 stream->add_audio_effect(stream, mEffectInterface); 8664 } 8665 } 8666 } 8667 return status; 8668} 8669 8670status_t AudioFlinger::EffectModule::stop() 8671{ 8672 Mutex::Autolock _l(mLock); 8673 return stop_l(); 8674} 8675 8676status_t AudioFlinger::EffectModule::stop_l() 8677{ 8678 if (mEffectInterface == NULL) { 8679 return NO_INIT; 8680 } 8681 status_t cmdStatus; 8682 uint32_t size = sizeof(status_t); 8683 status_t status = (*mEffectInterface)->command(mEffectInterface, 8684 EFFECT_CMD_DISABLE, 8685 0, 8686 NULL, 8687 &size, 8688 &cmdStatus); 8689 if (status == 0) { 8690 status = cmdStatus; 8691 } 8692 if (status == 0 && 8693 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8694 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8695 sp<ThreadBase> thread = mThread.promote(); 8696 if (thread != 0) { 8697 audio_stream_t *stream = thread->stream(); 8698 if (stream != NULL) { 8699 stream->remove_audio_effect(stream, mEffectInterface); 8700 } 8701 } 8702 } 8703 return status; 8704} 8705 8706status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8707 uint32_t cmdSize, 8708 void *pCmdData, 8709 uint32_t *replySize, 8710 void *pReplyData) 8711{ 8712 Mutex::Autolock _l(mLock); 8713 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8714 8715 if (mState == DESTROYED || mEffectInterface == NULL) { 8716 return NO_INIT; 8717 } 8718 status_t status = (*mEffectInterface)->command(mEffectInterface, 8719 cmdCode, 8720 cmdSize, 8721 pCmdData, 8722 replySize, 8723 pReplyData); 8724 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8725 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8726 for (size_t i = 1; i < mHandles.size(); i++) { 8727 EffectHandle *h = mHandles[i]; 8728 if (h != NULL && !h->destroyed_l()) { 8729 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8730 } 8731 } 8732 } 8733 return status; 8734} 8735 8736status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8737{ 8738 Mutex::Autolock _l(mLock); 8739 return setEnabled_l(enabled); 8740} 8741 8742// must be called with EffectModule::mLock held 8743status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8744{ 8745 8746 ALOGV("setEnabled %p enabled %d", this, enabled); 8747 8748 if (enabled != isEnabled()) { 8749 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8750 if (enabled && status != NO_ERROR) { 8751 return status; 8752 } 8753 8754 switch (mState) { 8755 // going from disabled to enabled 8756 case IDLE: 8757 mState = STARTING; 8758 break; 8759 case STOPPED: 8760 mState = RESTART; 8761 break; 8762 case STOPPING: 8763 mState = ACTIVE; 8764 break; 8765 8766 // going from enabled to disabled 8767 case RESTART: 8768 mState = STOPPED; 8769 break; 8770 case STARTING: 8771 mState = IDLE; 8772 break; 8773 case ACTIVE: 8774 mState = STOPPING; 8775 break; 8776 case DESTROYED: 8777 return NO_ERROR; // simply ignore as we are being destroyed 8778 } 8779 for (size_t i = 1; i < mHandles.size(); i++) { 8780 EffectHandle *h = mHandles[i]; 8781 if (h != NULL && !h->destroyed_l()) { 8782 h->setEnabled(enabled); 8783 } 8784 } 8785 } 8786 return NO_ERROR; 8787} 8788 8789bool AudioFlinger::EffectModule::isEnabled() const 8790{ 8791 switch (mState) { 8792 case RESTART: 8793 case STARTING: 8794 case ACTIVE: 8795 return true; 8796 case IDLE: 8797 case STOPPING: 8798 case STOPPED: 8799 case DESTROYED: 8800 default: 8801 return false; 8802 } 8803} 8804 8805bool AudioFlinger::EffectModule::isProcessEnabled() const 8806{ 8807 switch (mState) { 8808 case RESTART: 8809 case ACTIVE: 8810 case STOPPING: 8811 case STOPPED: 8812 return true; 8813 case IDLE: 8814 case STARTING: 8815 case DESTROYED: 8816 default: 8817 return false; 8818 } 8819} 8820 8821status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8822{ 8823 Mutex::Autolock _l(mLock); 8824 status_t status = NO_ERROR; 8825 8826 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8827 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8828 if (isProcessEnabled() && 8829 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8830 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8831 status_t cmdStatus; 8832 uint32_t volume[2]; 8833 uint32_t *pVolume = NULL; 8834 uint32_t size = sizeof(volume); 8835 volume[0] = *left; 8836 volume[1] = *right; 8837 if (controller) { 8838 pVolume = volume; 8839 } 8840 status = (*mEffectInterface)->command(mEffectInterface, 8841 EFFECT_CMD_SET_VOLUME, 8842 size, 8843 volume, 8844 &size, 8845 pVolume); 8846 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8847 *left = volume[0]; 8848 *right = volume[1]; 8849 } 8850 } 8851 return status; 8852} 8853 8854status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8855{ 8856 if (device == AUDIO_DEVICE_NONE) { 8857 return NO_ERROR; 8858 } 8859 8860 Mutex::Autolock _l(mLock); 8861 status_t status = NO_ERROR; 8862 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8863 status_t cmdStatus; 8864 uint32_t size = sizeof(status_t); 8865 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8866 EFFECT_CMD_SET_INPUT_DEVICE; 8867 status = (*mEffectInterface)->command(mEffectInterface, 8868 cmd, 8869 sizeof(uint32_t), 8870 &device, 8871 &size, 8872 &cmdStatus); 8873 } 8874 return status; 8875} 8876 8877status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8878{ 8879 Mutex::Autolock _l(mLock); 8880 status_t status = NO_ERROR; 8881 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8882 status_t cmdStatus; 8883 uint32_t size = sizeof(status_t); 8884 status = (*mEffectInterface)->command(mEffectInterface, 8885 EFFECT_CMD_SET_AUDIO_MODE, 8886 sizeof(audio_mode_t), 8887 &mode, 8888 &size, 8889 &cmdStatus); 8890 if (status == NO_ERROR) { 8891 status = cmdStatus; 8892 } 8893 } 8894 return status; 8895} 8896 8897status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8898{ 8899 Mutex::Autolock _l(mLock); 8900 status_t status = NO_ERROR; 8901 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8902 uint32_t size = 0; 8903 status = (*mEffectInterface)->command(mEffectInterface, 8904 EFFECT_CMD_SET_AUDIO_SOURCE, 8905 sizeof(audio_source_t), 8906 &source, 8907 &size, 8908 NULL); 8909 } 8910 return status; 8911} 8912 8913void AudioFlinger::EffectModule::setSuspended(bool suspended) 8914{ 8915 Mutex::Autolock _l(mLock); 8916 mSuspended = suspended; 8917} 8918 8919bool AudioFlinger::EffectModule::suspended() const 8920{ 8921 Mutex::Autolock _l(mLock); 8922 return mSuspended; 8923} 8924 8925bool AudioFlinger::EffectModule::purgeHandles() 8926{ 8927 bool enabled = false; 8928 Mutex::Autolock _l(mLock); 8929 for (size_t i = 0; i < mHandles.size(); i++) { 8930 EffectHandle *handle = mHandles[i]; 8931 if (handle != NULL && !handle->destroyed_l()) { 8932 handle->effect().clear(); 8933 if (handle->hasControl()) { 8934 enabled = handle->enabled(); 8935 } 8936 } 8937 } 8938 return enabled; 8939} 8940 8941void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8942{ 8943 const size_t SIZE = 256; 8944 char buffer[SIZE]; 8945 String8 result; 8946 8947 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8948 result.append(buffer); 8949 8950 bool locked = tryLock(mLock); 8951 // failed to lock - AudioFlinger is probably deadlocked 8952 if (!locked) { 8953 result.append("\t\tCould not lock Fx mutex:\n"); 8954 } 8955 8956 result.append("\t\tSession Status State Engine:\n"); 8957 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8958 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8959 result.append(buffer); 8960 8961 result.append("\t\tDescriptor:\n"); 8962 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8963 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8964 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], 8965 mDescriptor.uuid.node[2], 8966 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8967 result.append(buffer); 8968 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8969 mDescriptor.type.timeLow, mDescriptor.type.timeMid, 8970 mDescriptor.type.timeHiAndVersion, 8971 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], 8972 mDescriptor.type.node[2], 8973 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8974 result.append(buffer); 8975 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8976 mDescriptor.apiVersion, 8977 mDescriptor.flags); 8978 result.append(buffer); 8979 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8980 mDescriptor.name); 8981 result.append(buffer); 8982 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8983 mDescriptor.implementor); 8984 result.append(buffer); 8985 8986 result.append("\t\t- Input configuration:\n"); 8987 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8988 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8989 (uint32_t)mConfig.inputCfg.buffer.raw, 8990 mConfig.inputCfg.buffer.frameCount, 8991 mConfig.inputCfg.samplingRate, 8992 mConfig.inputCfg.channels, 8993 mConfig.inputCfg.format); 8994 result.append(buffer); 8995 8996 result.append("\t\t- Output configuration:\n"); 8997 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8998 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8999 (uint32_t)mConfig.outputCfg.buffer.raw, 9000 mConfig.outputCfg.buffer.frameCount, 9001 mConfig.outputCfg.samplingRate, 9002 mConfig.outputCfg.channels, 9003 mConfig.outputCfg.format); 9004 result.append(buffer); 9005 9006 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 9007 result.append(buffer); 9008 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 9009 for (size_t i = 0; i < mHandles.size(); ++i) { 9010 EffectHandle *handle = mHandles[i]; 9011 if (handle != NULL && !handle->destroyed_l()) { 9012 handle->dump(buffer, SIZE); 9013 result.append(buffer); 9014 } 9015 } 9016 9017 result.append("\n"); 9018 9019 write(fd, result.string(), result.length()); 9020 9021 if (locked) { 9022 mLock.unlock(); 9023 } 9024} 9025 9026// ---------------------------------------------------------------------------- 9027// EffectHandle implementation 9028// ---------------------------------------------------------------------------- 9029 9030#undef LOG_TAG 9031#define LOG_TAG "AudioFlinger::EffectHandle" 9032 9033AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 9034 const sp<AudioFlinger::Client>& client, 9035 const sp<IEffectClient>& effectClient, 9036 int32_t priority) 9037 : BnEffect(), 9038 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 9039 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 9040{ 9041 ALOGV("constructor %p", this); 9042 9043 if (client == 0) { 9044 return; 9045 } 9046 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 9047 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 9048 if (mCblkMemory != 0) { 9049 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9050 9051 if (mCblk != NULL) { 9052 new(mCblk) effect_param_cblk_t(); 9053 mBuffer = (uint8_t *)mCblk + bufOffset; 9054 } 9055 } else { 9056 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + 9057 sizeof(effect_param_cblk_t)); 9058 return; 9059 } 9060} 9061 9062AudioFlinger::EffectHandle::~EffectHandle() 9063{ 9064 ALOGV("Destructor %p", this); 9065 9066 if (mEffect == 0) { 9067 mDestroyed = true; 9068 return; 9069 } 9070 mEffect->lock(); 9071 mDestroyed = true; 9072 mEffect->unlock(); 9073 disconnect(false); 9074} 9075 9076status_t AudioFlinger::EffectHandle::enable() 9077{ 9078 ALOGV("enable %p", this); 9079 if (!mHasControl) return INVALID_OPERATION; 9080 if (mEffect == 0) return DEAD_OBJECT; 9081 9082 if (mEnabled) { 9083 return NO_ERROR; 9084 } 9085 9086 mEnabled = true; 9087 9088 sp<ThreadBase> thread = mEffect->thread().promote(); 9089 if (thread != 0) { 9090 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9091 } 9092 9093 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9094 if (mEffect->suspended()) { 9095 return NO_ERROR; 9096 } 9097 9098 status_t status = mEffect->setEnabled(true); 9099 if (status != NO_ERROR) { 9100 if (thread != 0) { 9101 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9102 } 9103 mEnabled = false; 9104 } 9105 return status; 9106} 9107 9108status_t AudioFlinger::EffectHandle::disable() 9109{ 9110 ALOGV("disable %p", this); 9111 if (!mHasControl) return INVALID_OPERATION; 9112 if (mEffect == 0) return DEAD_OBJECT; 9113 9114 if (!mEnabled) { 9115 return NO_ERROR; 9116 } 9117 mEnabled = false; 9118 9119 if (mEffect->suspended()) { 9120 return NO_ERROR; 9121 } 9122 9123 status_t status = mEffect->setEnabled(false); 9124 9125 sp<ThreadBase> thread = mEffect->thread().promote(); 9126 if (thread != 0) { 9127 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9128 } 9129 9130 return status; 9131} 9132 9133void AudioFlinger::EffectHandle::disconnect() 9134{ 9135 disconnect(true); 9136} 9137 9138void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9139{ 9140 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9141 if (mEffect == 0) { 9142 return; 9143 } 9144 // restore suspended effects if the disconnected handle was enabled and the last one. 9145 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9146 sp<ThreadBase> thread = mEffect->thread().promote(); 9147 if (thread != 0) { 9148 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9149 } 9150 } 9151 9152 // release sp on module => module destructor can be called now 9153 mEffect.clear(); 9154 if (mClient != 0) { 9155 if (mCblk != NULL) { 9156 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9157 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9158 } 9159 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9160 // Client destructor must run with AudioFlinger mutex locked 9161 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9162 mClient.clear(); 9163 } 9164} 9165 9166status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9167 uint32_t cmdSize, 9168 void *pCmdData, 9169 uint32_t *replySize, 9170 void *pReplyData) 9171{ 9172 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9173 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9174 9175 // only get parameter command is permitted for applications not controlling the effect 9176 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9177 return INVALID_OPERATION; 9178 } 9179 if (mEffect == 0) return DEAD_OBJECT; 9180 if (mClient == 0) return INVALID_OPERATION; 9181 9182 // handle commands that are not forwarded transparently to effect engine 9183 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9184 // No need to trylock() here as this function is executed in the binder thread serving a 9185 // particular client process: no risk to block the whole media server process or mixer 9186 // threads if we are stuck here 9187 Mutex::Autolock _l(mCblk->lock); 9188 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9189 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9190 mCblk->serverIndex = 0; 9191 mCblk->clientIndex = 0; 9192 return BAD_VALUE; 9193 } 9194 status_t status = NO_ERROR; 9195 while (mCblk->serverIndex < mCblk->clientIndex) { 9196 int reply; 9197 uint32_t rsize = sizeof(int); 9198 int *p = (int *)(mBuffer + mCblk->serverIndex); 9199 int size = *p++; 9200 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9201 ALOGW("command(): invalid parameter block size"); 9202 break; 9203 } 9204 effect_param_t *param = (effect_param_t *)p; 9205 if (param->psize == 0 || param->vsize == 0) { 9206 ALOGW("command(): null parameter or value size"); 9207 mCblk->serverIndex += size; 9208 continue; 9209 } 9210 uint32_t psize = sizeof(effect_param_t) + 9211 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9212 param->vsize; 9213 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9214 psize, 9215 p, 9216 &rsize, 9217 &reply); 9218 // stop at first error encountered 9219 if (ret != NO_ERROR) { 9220 status = ret; 9221 *(int *)pReplyData = reply; 9222 break; 9223 } else if (reply != NO_ERROR) { 9224 *(int *)pReplyData = reply; 9225 break; 9226 } 9227 mCblk->serverIndex += size; 9228 } 9229 mCblk->serverIndex = 0; 9230 mCblk->clientIndex = 0; 9231 return status; 9232 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9233 *(int *)pReplyData = NO_ERROR; 9234 return enable(); 9235 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9236 *(int *)pReplyData = NO_ERROR; 9237 return disable(); 9238 } 9239 9240 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9241} 9242 9243void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9244{ 9245 ALOGV("setControl %p control %d", this, hasControl); 9246 9247 mHasControl = hasControl; 9248 mEnabled = enabled; 9249 9250 if (signal && mEffectClient != 0) { 9251 mEffectClient->controlStatusChanged(hasControl); 9252 } 9253} 9254 9255void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9256 uint32_t cmdSize, 9257 void *pCmdData, 9258 uint32_t replySize, 9259 void *pReplyData) 9260{ 9261 if (mEffectClient != 0) { 9262 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9263 } 9264} 9265 9266 9267 9268void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9269{ 9270 if (mEffectClient != 0) { 9271 mEffectClient->enableStatusChanged(enabled); 9272 } 9273} 9274 9275status_t AudioFlinger::EffectHandle::onTransact( 9276 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9277{ 9278 return BnEffect::onTransact(code, data, reply, flags); 9279} 9280 9281 9282void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9283{ 9284 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9285 9286 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9287 (mClient == 0) ? getpid_cached : mClient->pid(), 9288 mPriority, 9289 mHasControl, 9290 !locked, 9291 mCblk ? mCblk->clientIndex : 0, 9292 mCblk ? mCblk->serverIndex : 0 9293 ); 9294 9295 if (locked) { 9296 mCblk->lock.unlock(); 9297 } 9298} 9299 9300#undef LOG_TAG 9301#define LOG_TAG "AudioFlinger::EffectChain" 9302 9303AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9304 int sessionId) 9305 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9306 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9307 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9308{ 9309 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9310 if (thread == NULL) { 9311 return; 9312 } 9313 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9314 thread->frameCount(); 9315} 9316 9317AudioFlinger::EffectChain::~EffectChain() 9318{ 9319 if (mOwnInBuffer) { 9320 delete mInBuffer; 9321 } 9322 9323} 9324 9325// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9326sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( 9327 effect_descriptor_t *descriptor) 9328{ 9329 size_t size = mEffects.size(); 9330 9331 for (size_t i = 0; i < size; i++) { 9332 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9333 return mEffects[i]; 9334 } 9335 } 9336 return 0; 9337} 9338 9339// getEffectFromId_l() must be called with ThreadBase::mLock held 9340sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9341{ 9342 size_t size = mEffects.size(); 9343 9344 for (size_t i = 0; i < size; i++) { 9345 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9346 if (id == 0 || mEffects[i]->id() == id) { 9347 return mEffects[i]; 9348 } 9349 } 9350 return 0; 9351} 9352 9353// getEffectFromType_l() must be called with ThreadBase::mLock held 9354sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9355 const effect_uuid_t *type) 9356{ 9357 size_t size = mEffects.size(); 9358 9359 for (size_t i = 0; i < size; i++) { 9360 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9361 return mEffects[i]; 9362 } 9363 } 9364 return 0; 9365} 9366 9367void AudioFlinger::EffectChain::clearInputBuffer() 9368{ 9369 Mutex::Autolock _l(mLock); 9370 sp<ThreadBase> thread = mThread.promote(); 9371 if (thread == 0) { 9372 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9373 return; 9374 } 9375 clearInputBuffer_l(thread); 9376} 9377 9378// Must be called with EffectChain::mLock locked 9379void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9380{ 9381 size_t numSamples = thread->frameCount() * thread->channelCount(); 9382 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9383 9384} 9385 9386// Must be called with EffectChain::mLock locked 9387void AudioFlinger::EffectChain::process_l() 9388{ 9389 sp<ThreadBase> thread = mThread.promote(); 9390 if (thread == 0) { 9391 ALOGW("process_l(): cannot promote mixer thread"); 9392 return; 9393 } 9394 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9395 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9396 // always process effects unless no more tracks are on the session and the effect tail 9397 // has been rendered 9398 bool doProcess = true; 9399 if (!isGlobalSession) { 9400 bool tracksOnSession = (trackCnt() != 0); 9401 9402 if (!tracksOnSession && mTailBufferCount == 0) { 9403 doProcess = false; 9404 } 9405 9406 if (activeTrackCnt() == 0) { 9407 // if no track is active and the effect tail has not been rendered, 9408 // the input buffer must be cleared here as the mixer process will not do it 9409 if (tracksOnSession || mTailBufferCount > 0) { 9410 clearInputBuffer_l(thread); 9411 if (mTailBufferCount > 0) { 9412 mTailBufferCount--; 9413 } 9414 } 9415 } 9416 } 9417 9418 size_t size = mEffects.size(); 9419 if (doProcess) { 9420 for (size_t i = 0; i < size; i++) { 9421 mEffects[i]->process(); 9422 } 9423 } 9424 for (size_t i = 0; i < size; i++) { 9425 mEffects[i]->updateState(); 9426 } 9427} 9428 9429// addEffect_l() must be called with PlaybackThread::mLock held 9430status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9431{ 9432 effect_descriptor_t desc = effect->desc(); 9433 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9434 9435 Mutex::Autolock _l(mLock); 9436 effect->setChain(this); 9437 sp<ThreadBase> thread = mThread.promote(); 9438 if (thread == 0) { 9439 return NO_INIT; 9440 } 9441 effect->setThread(thread); 9442 9443 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9444 // Auxiliary effects are inserted at the beginning of mEffects vector as 9445 // they are processed first and accumulated in chain input buffer 9446 mEffects.insertAt(effect, 0); 9447 9448 // the input buffer for auxiliary effect contains mono samples in 9449 // 32 bit format. This is to avoid saturation in AudoMixer 9450 // accumulation stage. Saturation is done in EffectModule::process() before 9451 // calling the process in effect engine 9452 size_t numSamples = thread->frameCount(); 9453 int32_t *buffer = new int32_t[numSamples]; 9454 memset(buffer, 0, numSamples * sizeof(int32_t)); 9455 effect->setInBuffer((int16_t *)buffer); 9456 // auxiliary effects output samples to chain input buffer for further processing 9457 // by insert effects 9458 effect->setOutBuffer(mInBuffer); 9459 } else { 9460 // Insert effects are inserted at the end of mEffects vector as they are processed 9461 // after track and auxiliary effects. 9462 // Insert effect order as a function of indicated preference: 9463 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9464 // another effect is present 9465 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9466 // last effect claiming first position 9467 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9468 // first effect claiming last position 9469 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9470 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9471 // already present 9472 9473 size_t size = mEffects.size(); 9474 size_t idx_insert = size; 9475 ssize_t idx_insert_first = -1; 9476 ssize_t idx_insert_last = -1; 9477 9478 for (size_t i = 0; i < size; i++) { 9479 effect_descriptor_t d = mEffects[i]->desc(); 9480 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9481 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9482 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9483 // check invalid effect chaining combinations 9484 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9485 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9486 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", 9487 desc.name, d.name); 9488 return INVALID_OPERATION; 9489 } 9490 // remember position of first insert effect and by default 9491 // select this as insert position for new effect 9492 if (idx_insert == size) { 9493 idx_insert = i; 9494 } 9495 // remember position of last insert effect claiming 9496 // first position 9497 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9498 idx_insert_first = i; 9499 } 9500 // remember position of first insert effect claiming 9501 // last position 9502 if (iPref == EFFECT_FLAG_INSERT_LAST && 9503 idx_insert_last == -1) { 9504 idx_insert_last = i; 9505 } 9506 } 9507 } 9508 9509 // modify idx_insert from first position if needed 9510 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9511 if (idx_insert_last != -1) { 9512 idx_insert = idx_insert_last; 9513 } else { 9514 idx_insert = size; 9515 } 9516 } else { 9517 if (idx_insert_first != -1) { 9518 idx_insert = idx_insert_first + 1; 9519 } 9520 } 9521 9522 // always read samples from chain input buffer 9523 effect->setInBuffer(mInBuffer); 9524 9525 // if last effect in the chain, output samples to chain 9526 // output buffer, otherwise to chain input buffer 9527 if (idx_insert == size) { 9528 if (idx_insert != 0) { 9529 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9530 mEffects[idx_insert-1]->configure(); 9531 } 9532 effect->setOutBuffer(mOutBuffer); 9533 } else { 9534 effect->setOutBuffer(mInBuffer); 9535 } 9536 mEffects.insertAt(effect, idx_insert); 9537 9538 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, 9539 idx_insert); 9540 } 9541 effect->configure(); 9542 return NO_ERROR; 9543} 9544 9545// removeEffect_l() must be called with PlaybackThread::mLock held 9546size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9547{ 9548 Mutex::Autolock _l(mLock); 9549 size_t size = mEffects.size(); 9550 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9551 9552 for (size_t i = 0; i < size; i++) { 9553 if (effect == mEffects[i]) { 9554 // calling stop here will remove pre-processing effect from the audio HAL. 9555 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9556 // the middle of a read from audio HAL 9557 if (mEffects[i]->state() == EffectModule::ACTIVE || 9558 mEffects[i]->state() == EffectModule::STOPPING) { 9559 mEffects[i]->stop(); 9560 } 9561 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9562 delete[] effect->inBuffer(); 9563 } else { 9564 if (i == size - 1 && i != 0) { 9565 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9566 mEffects[i - 1]->configure(); 9567 } 9568 } 9569 mEffects.removeAt(i); 9570 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), 9571 this, i); 9572 break; 9573 } 9574 } 9575 9576 return mEffects.size(); 9577} 9578 9579// setDevice_l() must be called with PlaybackThread::mLock held 9580void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9581{ 9582 size_t size = mEffects.size(); 9583 for (size_t i = 0; i < size; i++) { 9584 mEffects[i]->setDevice(device); 9585 } 9586} 9587 9588// setMode_l() must be called with PlaybackThread::mLock held 9589void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9590{ 9591 size_t size = mEffects.size(); 9592 for (size_t i = 0; i < size; i++) { 9593 mEffects[i]->setMode(mode); 9594 } 9595} 9596 9597// setAudioSource_l() must be called with PlaybackThread::mLock held 9598void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9599{ 9600 size_t size = mEffects.size(); 9601 for (size_t i = 0; i < size; i++) { 9602 mEffects[i]->setAudioSource(source); 9603 } 9604} 9605 9606// setVolume_l() must be called with PlaybackThread::mLock held 9607bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9608{ 9609 uint32_t newLeft = *left; 9610 uint32_t newRight = *right; 9611 bool hasControl = false; 9612 int ctrlIdx = -1; 9613 size_t size = mEffects.size(); 9614 9615 // first update volume controller 9616 for (size_t i = size; i > 0; i--) { 9617 if (mEffects[i - 1]->isProcessEnabled() && 9618 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9619 ctrlIdx = i - 1; 9620 hasControl = true; 9621 break; 9622 } 9623 } 9624 9625 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9626 if (hasControl) { 9627 *left = mNewLeftVolume; 9628 *right = mNewRightVolume; 9629 } 9630 return hasControl; 9631 } 9632 9633 mVolumeCtrlIdx = ctrlIdx; 9634 mLeftVolume = newLeft; 9635 mRightVolume = newRight; 9636 9637 // second get volume update from volume controller 9638 if (ctrlIdx >= 0) { 9639 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9640 mNewLeftVolume = newLeft; 9641 mNewRightVolume = newRight; 9642 } 9643 // then indicate volume to all other effects in chain. 9644 // Pass altered volume to effects before volume controller 9645 // and requested volume to effects after controller 9646 uint32_t lVol = newLeft; 9647 uint32_t rVol = newRight; 9648 9649 for (size_t i = 0; i < size; i++) { 9650 if ((int)i == ctrlIdx) continue; 9651 // this also works for ctrlIdx == -1 when there is no volume controller 9652 if ((int)i > ctrlIdx) { 9653 lVol = *left; 9654 rVol = *right; 9655 } 9656 mEffects[i]->setVolume(&lVol, &rVol, false); 9657 } 9658 *left = newLeft; 9659 *right = newRight; 9660 9661 return hasControl; 9662} 9663 9664void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9665{ 9666 const size_t SIZE = 256; 9667 char buffer[SIZE]; 9668 String8 result; 9669 9670 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9671 result.append(buffer); 9672 9673 bool locked = tryLock(mLock); 9674 // failed to lock - AudioFlinger is probably deadlocked 9675 if (!locked) { 9676 result.append("\tCould not lock mutex:\n"); 9677 } 9678 9679 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9680 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9681 mEffects.size(), 9682 (uint32_t)mInBuffer, 9683 (uint32_t)mOutBuffer, 9684 mActiveTrackCnt); 9685 result.append(buffer); 9686 write(fd, result.string(), result.size()); 9687 9688 for (size_t i = 0; i < mEffects.size(); ++i) { 9689 sp<EffectModule> effect = mEffects[i]; 9690 if (effect != 0) { 9691 effect->dump(fd, args); 9692 } 9693 } 9694 9695 if (locked) { 9696 mLock.unlock(); 9697 } 9698} 9699 9700// must be called with ThreadBase::mLock held 9701void AudioFlinger::EffectChain::setEffectSuspended_l( 9702 const effect_uuid_t *type, bool suspend) 9703{ 9704 sp<SuspendedEffectDesc> desc; 9705 // use effect type UUID timelow as key as there is no real risk of identical 9706 // timeLow fields among effect type UUIDs. 9707 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9708 if (suspend) { 9709 if (index >= 0) { 9710 desc = mSuspendedEffects.valueAt(index); 9711 } else { 9712 desc = new SuspendedEffectDesc(); 9713 desc->mType = *type; 9714 mSuspendedEffects.add(type->timeLow, desc); 9715 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9716 } 9717 if (desc->mRefCount++ == 0) { 9718 sp<EffectModule> effect = getEffectIfEnabled(type); 9719 if (effect != 0) { 9720 desc->mEffect = effect; 9721 effect->setSuspended(true); 9722 effect->setEnabled(false); 9723 } 9724 } 9725 } else { 9726 if (index < 0) { 9727 return; 9728 } 9729 desc = mSuspendedEffects.valueAt(index); 9730 if (desc->mRefCount <= 0) { 9731 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9732 desc->mRefCount = 1; 9733 } 9734 if (--desc->mRefCount == 0) { 9735 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9736 if (desc->mEffect != 0) { 9737 sp<EffectModule> effect = desc->mEffect.promote(); 9738 if (effect != 0) { 9739 effect->setSuspended(false); 9740 effect->lock(); 9741 EffectHandle *handle = effect->controlHandle_l(); 9742 if (handle != NULL && !handle->destroyed_l()) { 9743 effect->setEnabled_l(handle->enabled()); 9744 } 9745 effect->unlock(); 9746 } 9747 desc->mEffect.clear(); 9748 } 9749 mSuspendedEffects.removeItemsAt(index); 9750 } 9751 } 9752} 9753 9754// must be called with ThreadBase::mLock held 9755void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9756{ 9757 sp<SuspendedEffectDesc> desc; 9758 9759 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9760 if (suspend) { 9761 if (index >= 0) { 9762 desc = mSuspendedEffects.valueAt(index); 9763 } else { 9764 desc = new SuspendedEffectDesc(); 9765 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9766 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9767 } 9768 if (desc->mRefCount++ == 0) { 9769 Vector< sp<EffectModule> > effects; 9770 getSuspendEligibleEffects(effects); 9771 for (size_t i = 0; i < effects.size(); i++) { 9772 setEffectSuspended_l(&effects[i]->desc().type, true); 9773 } 9774 } 9775 } else { 9776 if (index < 0) { 9777 return; 9778 } 9779 desc = mSuspendedEffects.valueAt(index); 9780 if (desc->mRefCount <= 0) { 9781 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9782 desc->mRefCount = 1; 9783 } 9784 if (--desc->mRefCount == 0) { 9785 Vector<const effect_uuid_t *> types; 9786 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9787 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9788 continue; 9789 } 9790 types.add(&mSuspendedEffects.valueAt(i)->mType); 9791 } 9792 for (size_t i = 0; i < types.size(); i++) { 9793 setEffectSuspended_l(types[i], false); 9794 } 9795 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", 9796 mSuspendedEffects.keyAt(index)); 9797 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9798 } 9799 } 9800} 9801 9802 9803// The volume effect is used for automated tests only 9804#ifndef OPENSL_ES_H_ 9805static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9806 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9807const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9808#endif //OPENSL_ES_H_ 9809 9810bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9811{ 9812 // auxiliary effects and visualizer are never suspended on output mix 9813 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9814 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9815 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9816 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9817 return false; 9818 } 9819 return true; 9820} 9821 9822void AudioFlinger::EffectChain::getSuspendEligibleEffects( 9823 Vector< sp<AudioFlinger::EffectModule> > &effects) 9824{ 9825 effects.clear(); 9826 for (size_t i = 0; i < mEffects.size(); i++) { 9827 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9828 effects.add(mEffects[i]); 9829 } 9830 } 9831} 9832 9833sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9834 const effect_uuid_t *type) 9835{ 9836 sp<EffectModule> effect = getEffectFromType_l(type); 9837 return effect != 0 && effect->isEnabled() ? effect : 0; 9838} 9839 9840void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9841 bool enabled) 9842{ 9843 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9844 if (enabled) { 9845 if (index < 0) { 9846 // if the effect is not suspend check if all effects are suspended 9847 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9848 if (index < 0) { 9849 return; 9850 } 9851 if (!isEffectEligibleForSuspend(effect->desc())) { 9852 return; 9853 } 9854 setEffectSuspended_l(&effect->desc().type, enabled); 9855 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9856 if (index < 0) { 9857 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9858 return; 9859 } 9860 } 9861 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9862 effect->desc().type.timeLow); 9863 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9864 // if effect is requested to suspended but was not yet enabled, supend it now. 9865 if (desc->mEffect == 0) { 9866 desc->mEffect = effect; 9867 effect->setEnabled(false); 9868 effect->setSuspended(true); 9869 } 9870 } else { 9871 if (index < 0) { 9872 return; 9873 } 9874 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9875 effect->desc().type.timeLow); 9876 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9877 desc->mEffect.clear(); 9878 effect->setSuspended(false); 9879 } 9880} 9881 9882#undef LOG_TAG 9883#define LOG_TAG "AudioFlinger" 9884 9885// ---------------------------------------------------------------------------- 9886 9887status_t AudioFlinger::onTransact( 9888 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9889{ 9890 return BnAudioFlinger::onTransact(code, data, reply, flags); 9891} 9892 9893}; // namespace android 9894