AudioFlinger.cpp revision c26741598a840f7c1d12ee457fb26f68fcdbcc70
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: %d\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=%d mSampleRate=%d " 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 %d 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: %d mSampleRate %d", 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=%d, 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() == (int)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 unsigned 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 mFrameCount(0), 4190 mState(IDLE), 4191 mSampleRate(sampleRate), 4192 mFormat(format), 4193 mStepServerFailed(false), 4194 mSessionId(sessionId) 4195 // mChannelCount 4196 // mChannelMask 4197{ 4198 // client == 0 implies sharedBuffer == 0 4199 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 4200 4201 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 4202 sharedBuffer->size()); 4203 4204 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4205 size_t size = sizeof(audio_track_cblk_t); 4206 uint8_t channelCount = popcount(channelMask); 4207 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4208 if (sharedBuffer == 0) { 4209 size += bufferSize; 4210 } 4211 4212 if (client != 0) { 4213 mCblkMemory = client->heap()->allocate(size); 4214 if (mCblkMemory != 0) { 4215 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4216 // can't assume mCblk != NULL 4217 } else { 4218 ALOGE("not enough memory for AudioTrack size=%u", size); 4219 client->heap()->dump("AudioTrack"); 4220 return; 4221 } 4222 } else { 4223 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4224 // assume mCblk != NULL 4225 } 4226 4227 // construct the shared structure in-place. 4228 if (mCblk != NULL) { 4229 new(mCblk) audio_track_cblk_t(); 4230 // clear all buffers 4231 mCblk->frameCount = frameCount; 4232 mCblk->sampleRate = sampleRate; 4233// uncomment the following lines to quickly test 32-bit wraparound 4234// mCblk->user = 0xffff0000; 4235// mCblk->server = 0xffff0000; 4236// mCblk->userBase = 0xffff0000; 4237// mCblk->serverBase = 0xffff0000; 4238 mChannelCount = channelCount; 4239 mChannelMask = channelMask; 4240 if (sharedBuffer == 0) { 4241 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4242 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4243 // Force underrun condition to avoid false underrun callback until first data is 4244 // written to buffer (other flags are cleared) 4245 mCblk->flags = CBLK_UNDERRUN; 4246 } else { 4247 mBuffer = sharedBuffer->pointer(); 4248 } 4249 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4250 } 4251} 4252 4253AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4254{ 4255 if (mCblk != NULL) { 4256 if (mClient == 0) { 4257 delete mCblk; 4258 } else { 4259 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4260 } 4261 } 4262 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4263 if (mClient != 0) { 4264 // Client destructor must run with AudioFlinger mutex locked 4265 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4266 // If the client's reference count drops to zero, the associated destructor 4267 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4268 // relying on the automatic clear() at end of scope. 4269 mClient.clear(); 4270 } 4271} 4272 4273// AudioBufferProvider interface 4274// getNextBuffer() = 0; 4275// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4276void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4277{ 4278 buffer->raw = NULL; 4279 mFrameCount = buffer->frameCount; 4280 // FIXME See note at getNextBuffer() 4281 (void) step(); // ignore return value of step() 4282 buffer->frameCount = 0; 4283} 4284 4285bool AudioFlinger::ThreadBase::TrackBase::step() { 4286 bool result; 4287 audio_track_cblk_t* cblk = this->cblk(); 4288 4289 result = cblk->stepServer(mFrameCount); 4290 if (!result) { 4291 ALOGV("stepServer failed acquiring cblk mutex"); 4292 mStepServerFailed = true; 4293 } 4294 return result; 4295} 4296 4297void AudioFlinger::ThreadBase::TrackBase::reset() { 4298 audio_track_cblk_t* cblk = this->cblk(); 4299 4300 cblk->user = 0; 4301 cblk->server = 0; 4302 cblk->userBase = 0; 4303 cblk->serverBase = 0; 4304 mStepServerFailed = false; 4305 ALOGV("TrackBase::reset"); 4306} 4307 4308int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4309 return (int)mCblk->sampleRate; 4310} 4311 4312void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4313 audio_track_cblk_t* cblk = this->cblk(); 4314 size_t frameSize = cblk->frameSize; 4315 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4316 int8_t *bufferEnd = bufferStart + frames * frameSize; 4317 4318 // Check validity of returned pointer in case the track control block would have been corrupted. 4319 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4320 "TrackBase::getBuffer buffer out of range:\n" 4321 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4322 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4323 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4324 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4325 4326 return bufferStart; 4327} 4328 4329status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4330{ 4331 mSyncEvents.add(event); 4332 return NO_ERROR; 4333} 4334 4335// ---------------------------------------------------------------------------- 4336 4337// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4338AudioFlinger::PlaybackThread::Track::Track( 4339 PlaybackThread *thread, 4340 const sp<Client>& client, 4341 audio_stream_type_t streamType, 4342 uint32_t sampleRate, 4343 audio_format_t format, 4344 audio_channel_mask_t channelMask, 4345 int frameCount, 4346 const sp<IMemory>& sharedBuffer, 4347 int sessionId, 4348 IAudioFlinger::track_flags_t flags) 4349 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 4350 sessionId), 4351 mMute(false), 4352 mFillingUpStatus(FS_INVALID), 4353 // mRetryCount initialized later when needed 4354 mSharedBuffer(sharedBuffer), 4355 mStreamType(streamType), 4356 mName(-1), // see note below 4357 mMainBuffer(thread->mixBuffer()), 4358 mAuxBuffer(NULL), 4359 mAuxEffectId(0), mHasVolumeController(false), 4360 mPresentationCompleteFrames(0), 4361 mFlags(flags), 4362 mFastIndex(-1), 4363 mUnderrunCount(0), 4364 mCachedVolume(1.0) 4365{ 4366 if (mCblk != NULL) { 4367 // NOTE: audio_track_cblk_t::frameSize 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 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : 4370 sizeof(uint8_t); 4371 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4372 mName = thread->getTrackName_l(channelMask, sessionId); 4373 mCblk->mName = mName; 4374 if (mName < 0) { 4375 ALOGE("no more track names available"); 4376 return; 4377 } 4378 // only allocate a fast track index if we were able to allocate a normal track name 4379 if (flags & IAudioFlinger::TRACK_FAST) { 4380 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4381 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4382 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4383 // FIXME This is too eager. We allocate a fast track index before the 4384 // fast track becomes active. Since fast tracks are a scarce resource, 4385 // this means we are potentially denying other more important fast tracks from 4386 // being created. It would be better to allocate the index dynamically. 4387 mFastIndex = i; 4388 mCblk->mName = i; 4389 // Read the initial underruns because this field is never cleared by the fast mixer 4390 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4391 thread->mFastTrackAvailMask &= ~(1 << i); 4392 } 4393 } 4394 ALOGV("Track constructor name %d, calling pid %d", mName, 4395 IPCThreadState::self()->getCallingPid()); 4396} 4397 4398AudioFlinger::PlaybackThread::Track::~Track() 4399{ 4400 ALOGV("PlaybackThread::Track destructor"); 4401} 4402 4403void AudioFlinger::PlaybackThread::Track::destroy() 4404{ 4405 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4406 // by removing it from mTracks vector, so there is a risk that this Tracks's 4407 // destructor is called. As the destructor needs to lock mLock, 4408 // we must acquire a strong reference on this Track before locking mLock 4409 // here so that the destructor is called only when exiting this function. 4410 // On the other hand, as long as Track::destroy() is only called by 4411 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4412 // this Track with its member mTrack. 4413 sp<Track> keep(this); 4414 { // scope for mLock 4415 sp<ThreadBase> thread = mThread.promote(); 4416 if (thread != 0) { 4417 if (!isOutputTrack()) { 4418 if (mState == ACTIVE || mState == RESUMING) { 4419 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4420 4421#ifdef ADD_BATTERY_DATA 4422 // to track the speaker usage 4423 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4424#endif 4425 } 4426 AudioSystem::releaseOutput(thread->id()); 4427 } 4428 Mutex::Autolock _l(thread->mLock); 4429 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4430 playbackThread->destroyTrack_l(this); 4431 } 4432 } 4433} 4434 4435/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4436{ 4437 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate " 4438 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 4439} 4440 4441void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4442{ 4443 uint32_t vlr = mCblk->getVolumeLR(); 4444 if (isFastTrack()) { 4445 sprintf(buffer, " F %2d", mFastIndex); 4446 } else { 4447 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4448 } 4449 track_state state = mState; 4450 char stateChar; 4451 switch (state) { 4452 case IDLE: 4453 stateChar = 'I'; 4454 break; 4455 case TERMINATED: 4456 stateChar = 'T'; 4457 break; 4458 case STOPPING_1: 4459 stateChar = 's'; 4460 break; 4461 case STOPPING_2: 4462 stateChar = '5'; 4463 break; 4464 case STOPPED: 4465 stateChar = 'S'; 4466 break; 4467 case RESUMING: 4468 stateChar = 'R'; 4469 break; 4470 case ACTIVE: 4471 stateChar = 'A'; 4472 break; 4473 case PAUSING: 4474 stateChar = 'p'; 4475 break; 4476 case PAUSED: 4477 stateChar = 'P'; 4478 break; 4479 case FLUSHED: 4480 stateChar = 'F'; 4481 break; 4482 default: 4483 stateChar = '?'; 4484 break; 4485 } 4486 char nowInUnderrun; 4487 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4488 case UNDERRUN_FULL: 4489 nowInUnderrun = ' '; 4490 break; 4491 case UNDERRUN_PARTIAL: 4492 nowInUnderrun = '<'; 4493 break; 4494 case UNDERRUN_EMPTY: 4495 nowInUnderrun = '*'; 4496 break; 4497 default: 4498 nowInUnderrun = '?'; 4499 break; 4500 } 4501 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4502 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4503 (mClient == 0) ? getpid_cached : mClient->pid(), 4504 mStreamType, 4505 mFormat, 4506 mChannelMask, 4507 mSessionId, 4508 mFrameCount, 4509 mCblk->frameCount, 4510 stateChar, 4511 mMute, 4512 mFillingUpStatus, 4513 mCblk->sampleRate, 4514 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4515 20.0 * log10((vlr >> 16) / 4096.0), 4516 mCblk->server, 4517 mCblk->user, 4518 (int)mMainBuffer, 4519 (int)mAuxBuffer, 4520 mCblk->flags, 4521 mUnderrunCount, 4522 nowInUnderrun); 4523} 4524 4525// AudioBufferProvider interface 4526status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4527 AudioBufferProvider::Buffer* buffer, int64_t pts) 4528{ 4529 audio_track_cblk_t* cblk = this->cblk(); 4530 uint32_t framesReady; 4531 uint32_t framesReq = buffer->frameCount; 4532 4533 // Check if last stepServer failed, try to step now 4534 if (mStepServerFailed) { 4535 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4536 // Since the fast mixer is higher priority than client callback thread, 4537 // it does not result in priority inversion for client. 4538 // But a non-blocking solution would be preferable to avoid 4539 // fast mixer being unable to tryLock(), and 4540 // to avoid the extra context switches if the client wakes up, 4541 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4542 if (!step()) goto getNextBuffer_exit; 4543 ALOGV("stepServer recovered"); 4544 mStepServerFailed = false; 4545 } 4546 4547 // FIXME Same as above 4548 framesReady = cblk->framesReady(); 4549 4550 if (CC_LIKELY(framesReady)) { 4551 uint32_t s = cblk->server; 4552 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4553 4554 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4555 if (framesReq > framesReady) { 4556 framesReq = framesReady; 4557 } 4558 if (framesReq > bufferEnd - s) { 4559 framesReq = bufferEnd - s; 4560 } 4561 4562 buffer->raw = getBuffer(s, framesReq); 4563 buffer->frameCount = framesReq; 4564 return NO_ERROR; 4565 } 4566 4567getNextBuffer_exit: 4568 buffer->raw = NULL; 4569 buffer->frameCount = 0; 4570 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4571 return NOT_ENOUGH_DATA; 4572} 4573 4574// Note that framesReady() takes a mutex on the control block using tryLock(). 4575// This could result in priority inversion if framesReady() is called by the normal mixer, 4576// as the normal mixer thread runs at lower 4577// priority than the client's callback thread: there is a short window within framesReady() 4578// during which the normal mixer could be preempted, and the client callback would block. 4579// Another problem can occur if framesReady() is called by the fast mixer: 4580// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4581// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4582size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4583 return mCblk->framesReady(); 4584} 4585 4586// Don't call for fast tracks; the framesReady() could result in priority inversion 4587bool AudioFlinger::PlaybackThread::Track::isReady() const { 4588 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4589 4590 if (framesReady() >= mCblk->frameCount || 4591 (mCblk->flags & CBLK_FORCEREADY)) { 4592 mFillingUpStatus = FS_FILLED; 4593 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4594 return true; 4595 } 4596 return false; 4597} 4598 4599status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4600 int triggerSession) 4601{ 4602 status_t status = NO_ERROR; 4603 ALOGV("start(%d), calling pid %d session %d", 4604 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4605 4606 sp<ThreadBase> thread = mThread.promote(); 4607 if (thread != 0) { 4608 Mutex::Autolock _l(thread->mLock); 4609 track_state state = mState; 4610 // here the track could be either new, or restarted 4611 // in both cases "unstop" the track 4612 if (mState == PAUSED) { 4613 mState = TrackBase::RESUMING; 4614 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4615 } else { 4616 mState = TrackBase::ACTIVE; 4617 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4618 } 4619 4620 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4621 thread->mLock.unlock(); 4622 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4623 thread->mLock.lock(); 4624 4625#ifdef ADD_BATTERY_DATA 4626 // to track the speaker usage 4627 if (status == NO_ERROR) { 4628 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4629 } 4630#endif 4631 } 4632 if (status == NO_ERROR) { 4633 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4634 playbackThread->addTrack_l(this); 4635 } else { 4636 mState = state; 4637 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4638 } 4639 } else { 4640 status = BAD_VALUE; 4641 } 4642 return status; 4643} 4644 4645void AudioFlinger::PlaybackThread::Track::stop() 4646{ 4647 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4648 sp<ThreadBase> thread = mThread.promote(); 4649 if (thread != 0) { 4650 Mutex::Autolock _l(thread->mLock); 4651 track_state state = mState; 4652 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4653 // If the track is not active (PAUSED and buffers full), flush buffers 4654 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4655 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4656 reset(); 4657 mState = STOPPED; 4658 } else if (!isFastTrack()) { 4659 mState = STOPPED; 4660 } else { 4661 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4662 // and then to STOPPED and reset() when presentation is complete 4663 mState = STOPPING_1; 4664 } 4665 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 4666 playbackThread); 4667 } 4668 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4669 thread->mLock.unlock(); 4670 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4671 thread->mLock.lock(); 4672 4673#ifdef ADD_BATTERY_DATA 4674 // to track the speaker usage 4675 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4676#endif 4677 } 4678 } 4679} 4680 4681void AudioFlinger::PlaybackThread::Track::pause() 4682{ 4683 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4684 sp<ThreadBase> thread = mThread.promote(); 4685 if (thread != 0) { 4686 Mutex::Autolock _l(thread->mLock); 4687 if (mState == ACTIVE || mState == RESUMING) { 4688 mState = PAUSING; 4689 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4690 if (!isOutputTrack()) { 4691 thread->mLock.unlock(); 4692 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4693 thread->mLock.lock(); 4694 4695#ifdef ADD_BATTERY_DATA 4696 // to track the speaker usage 4697 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4698#endif 4699 } 4700 } 4701 } 4702} 4703 4704void AudioFlinger::PlaybackThread::Track::flush() 4705{ 4706 ALOGV("flush(%d)", mName); 4707 sp<ThreadBase> thread = mThread.promote(); 4708 if (thread != 0) { 4709 Mutex::Autolock _l(thread->mLock); 4710 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4711 mState != PAUSING) { 4712 return; 4713 } 4714 // No point remaining in PAUSED state after a flush => go to 4715 // FLUSHED state 4716 mState = FLUSHED; 4717 // do not reset the track if it is still in the process of being stopped or paused. 4718 // this will be done by prepareTracks_l() when the track is stopped. 4719 // prepareTracks_l() will see mState == FLUSHED, then 4720 // remove from active track list, reset(), and trigger presentation complete 4721 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4722 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4723 reset(); 4724 } 4725 } 4726} 4727 4728void AudioFlinger::PlaybackThread::Track::reset() 4729{ 4730 // Do not reset twice to avoid discarding data written just after a flush and before 4731 // the audioflinger thread detects the track is stopped. 4732 if (!mResetDone) { 4733 TrackBase::reset(); 4734 // Force underrun condition to avoid false underrun callback until first data is 4735 // written to buffer 4736 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 4737 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 4738 mFillingUpStatus = FS_FILLING; 4739 mResetDone = true; 4740 if (mState == FLUSHED) { 4741 mState = IDLE; 4742 } 4743 } 4744} 4745 4746void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4747{ 4748 mMute = muted; 4749} 4750 4751status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4752{ 4753 status_t status = DEAD_OBJECT; 4754 sp<ThreadBase> thread = mThread.promote(); 4755 if (thread != 0) { 4756 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4757 sp<AudioFlinger> af = mClient->audioFlinger(); 4758 4759 Mutex::Autolock _l(af->mLock); 4760 4761 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4762 4763 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4764 Mutex::Autolock _dl(playbackThread->mLock); 4765 Mutex::Autolock _sl(srcThread->mLock); 4766 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4767 if (chain == 0) { 4768 return INVALID_OPERATION; 4769 } 4770 4771 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4772 if (effect == 0) { 4773 return INVALID_OPERATION; 4774 } 4775 srcThread->removeEffect_l(effect); 4776 playbackThread->addEffect_l(effect); 4777 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4778 if (effect->state() == EffectModule::ACTIVE || 4779 effect->state() == EffectModule::STOPPING) { 4780 effect->start(); 4781 } 4782 4783 sp<EffectChain> dstChain = effect->chain().promote(); 4784 if (dstChain == 0) { 4785 srcThread->addEffect_l(effect); 4786 return INVALID_OPERATION; 4787 } 4788 AudioSystem::unregisterEffect(effect->id()); 4789 AudioSystem::registerEffect(&effect->desc(), 4790 srcThread->id(), 4791 dstChain->strategy(), 4792 AUDIO_SESSION_OUTPUT_MIX, 4793 effect->id()); 4794 } 4795 status = playbackThread->attachAuxEffect(this, EffectId); 4796 } 4797 return status; 4798} 4799 4800void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4801{ 4802 mAuxEffectId = EffectId; 4803 mAuxBuffer = buffer; 4804} 4805 4806bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4807 size_t audioHalFrames) 4808{ 4809 // a track is considered presented when the total number of frames written to audio HAL 4810 // corresponds to the number of frames written when presentationComplete() is called for the 4811 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4812 if (mPresentationCompleteFrames == 0) { 4813 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4814 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4815 mPresentationCompleteFrames, audioHalFrames); 4816 } 4817 if (framesWritten >= mPresentationCompleteFrames) { 4818 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4819 mSessionId, framesWritten); 4820 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4821 return true; 4822 } 4823 return false; 4824} 4825 4826void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4827{ 4828 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4829 if (mSyncEvents[i]->type() == type) { 4830 mSyncEvents[i]->trigger(); 4831 mSyncEvents.removeAt(i); 4832 i--; 4833 } 4834 } 4835} 4836 4837// implement VolumeBufferProvider interface 4838 4839uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4840{ 4841 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4842 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4843 uint32_t vlr = mCblk->getVolumeLR(); 4844 uint32_t vl = vlr & 0xFFFF; 4845 uint32_t vr = vlr >> 16; 4846 // track volumes come from shared memory, so can't be trusted and must be clamped 4847 if (vl > MAX_GAIN_INT) { 4848 vl = MAX_GAIN_INT; 4849 } 4850 if (vr > MAX_GAIN_INT) { 4851 vr = MAX_GAIN_INT; 4852 } 4853 // now apply the cached master volume and stream type volume; 4854 // this is trusted but lacks any synchronization or barrier so may be stale 4855 float v = mCachedVolume; 4856 vl *= v; 4857 vr *= v; 4858 // re-combine into U4.16 4859 vlr = (vr << 16) | (vl & 0xFFFF); 4860 // FIXME look at mute, pause, and stop flags 4861 return vlr; 4862} 4863 4864status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4865{ 4866 if (mState == TERMINATED || mState == PAUSED || 4867 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4868 (mState == STOPPED)))) { 4869 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4870 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4871 event->cancel(); 4872 return INVALID_OPERATION; 4873 } 4874 (void) TrackBase::setSyncEvent(event); 4875 return NO_ERROR; 4876} 4877 4878// timed audio tracks 4879 4880sp<AudioFlinger::PlaybackThread::TimedTrack> 4881AudioFlinger::PlaybackThread::TimedTrack::create( 4882 PlaybackThread *thread, 4883 const sp<Client>& client, 4884 audio_stream_type_t streamType, 4885 uint32_t sampleRate, 4886 audio_format_t format, 4887 audio_channel_mask_t channelMask, 4888 int frameCount, 4889 const sp<IMemory>& sharedBuffer, 4890 int sessionId) { 4891 if (!client->reserveTimedTrack()) 4892 return 0; 4893 4894 return new TimedTrack( 4895 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4896 sharedBuffer, sessionId); 4897} 4898 4899AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4900 PlaybackThread *thread, 4901 const sp<Client>& client, 4902 audio_stream_type_t streamType, 4903 uint32_t sampleRate, 4904 audio_format_t format, 4905 audio_channel_mask_t channelMask, 4906 int frameCount, 4907 const sp<IMemory>& sharedBuffer, 4908 int sessionId) 4909 : Track(thread, client, streamType, sampleRate, format, channelMask, 4910 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4911 mQueueHeadInFlight(false), 4912 mTrimQueueHeadOnRelease(false), 4913 mFramesPendingInQueue(0), 4914 mTimedSilenceBuffer(NULL), 4915 mTimedSilenceBufferSize(0), 4916 mTimedAudioOutputOnTime(false), 4917 mMediaTimeTransformValid(false) 4918{ 4919 LocalClock lc; 4920 mLocalTimeFreq = lc.getLocalFreq(); 4921 4922 mLocalTimeToSampleTransform.a_zero = 0; 4923 mLocalTimeToSampleTransform.b_zero = 0; 4924 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4925 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4926 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4927 &mLocalTimeToSampleTransform.a_to_b_denom); 4928 4929 mMediaTimeToSampleTransform.a_zero = 0; 4930 mMediaTimeToSampleTransform.b_zero = 0; 4931 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4932 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4933 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4934 &mMediaTimeToSampleTransform.a_to_b_denom); 4935} 4936 4937AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4938 mClient->releaseTimedTrack(); 4939 delete [] mTimedSilenceBuffer; 4940} 4941 4942status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4943 size_t size, sp<IMemory>* buffer) { 4944 4945 Mutex::Autolock _l(mTimedBufferQueueLock); 4946 4947 trimTimedBufferQueue_l(); 4948 4949 // lazily initialize the shared memory heap for timed buffers 4950 if (mTimedMemoryDealer == NULL) { 4951 const int kTimedBufferHeapSize = 512 << 10; 4952 4953 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4954 "AudioFlingerTimed"); 4955 if (mTimedMemoryDealer == NULL) 4956 return NO_MEMORY; 4957 } 4958 4959 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4960 if (newBuffer == NULL) { 4961 newBuffer = mTimedMemoryDealer->allocate(size); 4962 if (newBuffer == NULL) 4963 return NO_MEMORY; 4964 } 4965 4966 *buffer = newBuffer; 4967 return NO_ERROR; 4968} 4969 4970// caller must hold mTimedBufferQueueLock 4971void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4972 int64_t mediaTimeNow; 4973 { 4974 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4975 if (!mMediaTimeTransformValid) 4976 return; 4977 4978 int64_t targetTimeNow; 4979 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4980 ? mCCHelper.getCommonTime(&targetTimeNow) 4981 : mCCHelper.getLocalTime(&targetTimeNow); 4982 4983 if (OK != res) 4984 return; 4985 4986 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4987 &mediaTimeNow)) { 4988 return; 4989 } 4990 } 4991 4992 size_t trimEnd; 4993 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4994 int64_t bufEnd; 4995 4996 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4997 // We have a next buffer. Just use its PTS as the PTS of the frame 4998 // following the last frame in this buffer. If the stream is sparse 4999 // (ie, there are deliberate gaps left in the stream which should be 5000 // filled with silence by the TimedAudioTrack), then this can result 5001 // in one extra buffer being left un-trimmed when it could have 5002 // been. In general, this is not typical, and we would rather 5003 // optimized away the TS calculation below for the more common case 5004 // where PTSes are contiguous. 5005 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 5006 } else { 5007 // We have no next buffer. Compute the PTS of the frame following 5008 // the last frame in this buffer by computing the duration of of 5009 // this frame in media time units and adding it to the PTS of the 5010 // buffer. 5011 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 5012 / mCblk->frameSize; 5013 5014 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 5015 &bufEnd)) { 5016 ALOGE("Failed to convert frame count of %lld to media time" 5017 " duration" " (scale factor %d/%u) in %s", 5018 frameCount, 5019 mMediaTimeToSampleTransform.a_to_b_numer, 5020 mMediaTimeToSampleTransform.a_to_b_denom, 5021 __PRETTY_FUNCTION__); 5022 break; 5023 } 5024 bufEnd += mTimedBufferQueue[trimEnd].pts(); 5025 } 5026 5027 if (bufEnd > mediaTimeNow) 5028 break; 5029 5030 // Is the buffer we want to use in the middle of a mix operation right 5031 // now? If so, don't actually trim it. Just wait for the releaseBuffer 5032 // from the mixer which should be coming back shortly. 5033 if (!trimEnd && mQueueHeadInFlight) { 5034 mTrimQueueHeadOnRelease = true; 5035 } 5036 } 5037 5038 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 5039 if (trimStart < trimEnd) { 5040 // Update the bookkeeping for framesReady() 5041 for (size_t i = trimStart; i < trimEnd; ++i) { 5042 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 5043 } 5044 5045 // Now actually remove the buffers from the queue. 5046 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 5047 } 5048} 5049 5050void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 5051 const char* logTag) { 5052 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 5053 "%s called (reason \"%s\"), but timed buffer queue has no" 5054 " elements to trim.", __FUNCTION__, logTag); 5055 5056 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 5057 mTimedBufferQueue.removeAt(0); 5058} 5059 5060void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 5061 const TimedBuffer& buf, 5062 const char* logTag) { 5063 uint32_t bufBytes = buf.buffer()->size(); 5064 uint32_t consumedAlready = buf.position(); 5065 5066 ALOG_ASSERT(consumedAlready <= bufBytes, 5067 "Bad bookkeeping while updating frames pending. Timed buffer is" 5068 " only %u bytes long, but claims to have consumed %u" 5069 " bytes. (update reason: \"%s\")", 5070 bufBytes, consumedAlready, logTag); 5071 5072 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 5073 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 5074 "Bad bookkeeping while updating frames pending. Should have at" 5075 " least %u queued frames, but we think we have only %u. (update" 5076 " reason: \"%s\")", 5077 bufFrames, mFramesPendingInQueue, logTag); 5078 5079 mFramesPendingInQueue -= bufFrames; 5080} 5081 5082status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5083 const sp<IMemory>& buffer, int64_t pts) { 5084 5085 { 5086 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5087 if (!mMediaTimeTransformValid) 5088 return INVALID_OPERATION; 5089 } 5090 5091 Mutex::Autolock _l(mTimedBufferQueueLock); 5092 5093 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5094 mFramesPendingInQueue += bufFrames; 5095 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5096 5097 return NO_ERROR; 5098} 5099 5100status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5101 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5102 5103 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5104 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5105 target); 5106 5107 if (!(target == TimedAudioTrack::LOCAL_TIME || 5108 target == TimedAudioTrack::COMMON_TIME)) { 5109 return BAD_VALUE; 5110 } 5111 5112 Mutex::Autolock lock(mMediaTimeTransformLock); 5113 mMediaTimeTransform = xform; 5114 mMediaTimeTransformTarget = target; 5115 mMediaTimeTransformValid = true; 5116 5117 return NO_ERROR; 5118} 5119 5120#define min(a, b) ((a) < (b) ? (a) : (b)) 5121 5122// implementation of getNextBuffer for tracks whose buffers have timestamps 5123status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5124 AudioBufferProvider::Buffer* buffer, int64_t pts) 5125{ 5126 if (pts == AudioBufferProvider::kInvalidPTS) { 5127 buffer->raw = NULL; 5128 buffer->frameCount = 0; 5129 mTimedAudioOutputOnTime = false; 5130 return INVALID_OPERATION; 5131 } 5132 5133 Mutex::Autolock _l(mTimedBufferQueueLock); 5134 5135 ALOG_ASSERT(!mQueueHeadInFlight, 5136 "getNextBuffer called without releaseBuffer!"); 5137 5138 while (true) { 5139 5140 // if we have no timed buffers, then fail 5141 if (mTimedBufferQueue.isEmpty()) { 5142 buffer->raw = NULL; 5143 buffer->frameCount = 0; 5144 return NOT_ENOUGH_DATA; 5145 } 5146 5147 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5148 5149 // calculate the PTS of the head of the timed buffer queue expressed in 5150 // local time 5151 int64_t headLocalPTS; 5152 { 5153 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5154 5155 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5156 5157 if (mMediaTimeTransform.a_to_b_denom == 0) { 5158 // the transform represents a pause, so yield silence 5159 timedYieldSilence_l(buffer->frameCount, buffer); 5160 return NO_ERROR; 5161 } 5162 5163 int64_t transformedPTS; 5164 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5165 &transformedPTS)) { 5166 // the transform failed. this shouldn't happen, but if it does 5167 // then just drop this buffer 5168 ALOGW("timedGetNextBuffer transform failed"); 5169 buffer->raw = NULL; 5170 buffer->frameCount = 0; 5171 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5172 return NO_ERROR; 5173 } 5174 5175 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5176 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5177 &headLocalPTS)) { 5178 buffer->raw = NULL; 5179 buffer->frameCount = 0; 5180 return INVALID_OPERATION; 5181 } 5182 } else { 5183 headLocalPTS = transformedPTS; 5184 } 5185 } 5186 5187 // adjust the head buffer's PTS to reflect the portion of the head buffer 5188 // that has already been consumed 5189 int64_t effectivePTS = headLocalPTS + 5190 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5191 5192 // Calculate the delta in samples between the head of the input buffer 5193 // queue and the start of the next output buffer that will be written. 5194 // If the transformation fails because of over or underflow, it means 5195 // that the sample's position in the output stream is so far out of 5196 // whack that it should just be dropped. 5197 int64_t sampleDelta; 5198 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5199 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5200 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5201 " mix"); 5202 continue; 5203 } 5204 if (!mLocalTimeToSampleTransform.doForwardTransform( 5205 (effectivePTS - pts) << 32, &sampleDelta)) { 5206 ALOGV("*** too late during sample rate transform: dropped buffer"); 5207 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5208 continue; 5209 } 5210 5211 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5212 " sampleDelta=[%d.%08x]", 5213 head.pts(), head.position(), pts, 5214 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5215 + (sampleDelta >> 32)), 5216 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5217 5218 // if the delta between the ideal placement for the next input sample and 5219 // the current output position is within this threshold, then we will 5220 // concatenate the next input samples to the previous output 5221 const int64_t kSampleContinuityThreshold = 5222 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5223 5224 // if this is the first buffer of audio that we're emitting from this track 5225 // then it should be almost exactly on time. 5226 const int64_t kSampleStartupThreshold = 1LL << 32; 5227 5228 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5229 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5230 // the next input is close enough to being on time, so concatenate it 5231 // with the last output 5232 timedYieldSamples_l(buffer); 5233 5234 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5235 head.position(), buffer->frameCount); 5236 return NO_ERROR; 5237 } 5238 5239 // Looks like our output is not on time. Reset our on timed status. 5240 // Next time we mix samples from our input queue, then should be within 5241 // the StartupThreshold. 5242 mTimedAudioOutputOnTime = false; 5243 if (sampleDelta > 0) { 5244 // the gap between the current output position and the proper start of 5245 // the next input sample is too big, so fill it with silence 5246 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5247 5248 timedYieldSilence_l(framesUntilNextInput, buffer); 5249 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5250 return NO_ERROR; 5251 } else { 5252 // the next input sample is late 5253 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5254 size_t onTimeSamplePosition = 5255 head.position() + lateFrames * mCblk->frameSize; 5256 5257 if (onTimeSamplePosition > head.buffer()->size()) { 5258 // all the remaining samples in the head are too late, so 5259 // drop it and move on 5260 ALOGV("*** too late: dropped buffer"); 5261 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5262 continue; 5263 } else { 5264 // skip over the late samples 5265 head.setPosition(onTimeSamplePosition); 5266 5267 // yield the available samples 5268 timedYieldSamples_l(buffer); 5269 5270 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5271 return NO_ERROR; 5272 } 5273 } 5274 } 5275} 5276 5277// Yield samples from the timed buffer queue head up to the given output 5278// buffer's capacity. 5279// 5280// Caller must hold mTimedBufferQueueLock 5281void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5282 AudioBufferProvider::Buffer* buffer) { 5283 5284 const TimedBuffer& head = mTimedBufferQueue[0]; 5285 5286 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5287 head.position()); 5288 5289 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5290 mCblk->frameSize); 5291 size_t framesRequested = buffer->frameCount; 5292 buffer->frameCount = min(framesLeftInHead, framesRequested); 5293 5294 mQueueHeadInFlight = true; 5295 mTimedAudioOutputOnTime = true; 5296} 5297 5298// Yield samples of silence up to the given output buffer's capacity 5299// 5300// Caller must hold mTimedBufferQueueLock 5301void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5302 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5303 5304 // lazily allocate a buffer filled with silence 5305 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5306 delete [] mTimedSilenceBuffer; 5307 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5308 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5309 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5310 } 5311 5312 buffer->raw = mTimedSilenceBuffer; 5313 size_t framesRequested = buffer->frameCount; 5314 buffer->frameCount = min(numFrames, framesRequested); 5315 5316 mTimedAudioOutputOnTime = false; 5317} 5318 5319// AudioBufferProvider interface 5320void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5321 AudioBufferProvider::Buffer* buffer) { 5322 5323 Mutex::Autolock _l(mTimedBufferQueueLock); 5324 5325 // If the buffer which was just released is part of the buffer at the head 5326 // of the queue, be sure to update the amt of the buffer which has been 5327 // consumed. If the buffer being returned is not part of the head of the 5328 // queue, its either because the buffer is part of the silence buffer, or 5329 // because the head of the timed queue was trimmed after the mixer called 5330 // getNextBuffer but before the mixer called releaseBuffer. 5331 if (buffer->raw == mTimedSilenceBuffer) { 5332 ALOG_ASSERT(!mQueueHeadInFlight, 5333 "Queue head in flight during release of silence buffer!"); 5334 goto done; 5335 } 5336 5337 ALOG_ASSERT(mQueueHeadInFlight, 5338 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5339 " head in flight."); 5340 5341 if (mTimedBufferQueue.size()) { 5342 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5343 5344 void* start = head.buffer()->pointer(); 5345 void* end = reinterpret_cast<void*>( 5346 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5347 + head.buffer()->size()); 5348 5349 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5350 "released buffer not within the head of the timed buffer" 5351 " queue; qHead = [%p, %p], released buffer = %p", 5352 start, end, buffer->raw); 5353 5354 head.setPosition(head.position() + 5355 (buffer->frameCount * mCblk->frameSize)); 5356 mQueueHeadInFlight = false; 5357 5358 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5359 "Bad bookkeeping during releaseBuffer! Should have at" 5360 " least %u queued frames, but we think we have only %u", 5361 buffer->frameCount, mFramesPendingInQueue); 5362 5363 mFramesPendingInQueue -= buffer->frameCount; 5364 5365 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5366 || mTrimQueueHeadOnRelease) { 5367 trimTimedBufferQueueHead_l("releaseBuffer"); 5368 mTrimQueueHeadOnRelease = false; 5369 } 5370 } else { 5371 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5372 " buffers in the timed buffer queue"); 5373 } 5374 5375done: 5376 buffer->raw = 0; 5377 buffer->frameCount = 0; 5378} 5379 5380size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5381 Mutex::Autolock _l(mTimedBufferQueueLock); 5382 return mFramesPendingInQueue; 5383} 5384 5385AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5386 : mPTS(0), mPosition(0) {} 5387 5388AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5389 const sp<IMemory>& buffer, int64_t pts) 5390 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5391 5392// ---------------------------------------------------------------------------- 5393 5394// RecordTrack constructor must be called with AudioFlinger::mLock held 5395AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5396 RecordThread *thread, 5397 const sp<Client>& client, 5398 uint32_t sampleRate, 5399 audio_format_t format, 5400 audio_channel_mask_t channelMask, 5401 int frameCount, 5402 int sessionId) 5403 : TrackBase(thread, client, sampleRate, format, 5404 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5405 mOverflow(false) 5406{ 5407 if (mCblk != NULL) { 5408 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5409 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5410 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5411 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5412 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5413 } else { 5414 mCblk->frameSize = sizeof(int8_t); 5415 } 5416 } 5417} 5418 5419AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5420{ 5421 ALOGV("%s", __func__); 5422} 5423 5424// AudioBufferProvider interface 5425status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 5426 int64_t pts) 5427{ 5428 audio_track_cblk_t* cblk = this->cblk(); 5429 uint32_t framesAvail; 5430 uint32_t framesReq = buffer->frameCount; 5431 5432 // Check if last stepServer failed, try to step now 5433 if (mStepServerFailed) { 5434 if (!step()) goto getNextBuffer_exit; 5435 ALOGV("stepServer recovered"); 5436 mStepServerFailed = false; 5437 } 5438 5439 framesAvail = cblk->framesAvailable_l(); 5440 5441 if (CC_LIKELY(framesAvail)) { 5442 uint32_t s = cblk->server; 5443 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5444 5445 if (framesReq > framesAvail) { 5446 framesReq = framesAvail; 5447 } 5448 if (framesReq > bufferEnd - s) { 5449 framesReq = bufferEnd - s; 5450 } 5451 5452 buffer->raw = getBuffer(s, framesReq); 5453 buffer->frameCount = framesReq; 5454 return NO_ERROR; 5455 } 5456 5457getNextBuffer_exit: 5458 buffer->raw = NULL; 5459 buffer->frameCount = 0; 5460 return NOT_ENOUGH_DATA; 5461} 5462 5463status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5464 int triggerSession) 5465{ 5466 sp<ThreadBase> thread = mThread.promote(); 5467 if (thread != 0) { 5468 RecordThread *recordThread = (RecordThread *)thread.get(); 5469 return recordThread->start(this, event, triggerSession); 5470 } else { 5471 return BAD_VALUE; 5472 } 5473} 5474 5475void AudioFlinger::RecordThread::RecordTrack::stop() 5476{ 5477 sp<ThreadBase> thread = mThread.promote(); 5478 if (thread != 0) { 5479 RecordThread *recordThread = (RecordThread *)thread.get(); 5480 recordThread->mLock.lock(); 5481 bool doStop = recordThread->stop_l(this); 5482 if (doStop) { 5483 TrackBase::reset(); 5484 // Force overrun condition to avoid false overrun callback until first data is 5485 // read from buffer 5486 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 5487 } 5488 recordThread->mLock.unlock(); 5489 if (doStop) { 5490 AudioSystem::stopInput(recordThread->id()); 5491 } 5492 } 5493} 5494 5495/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5496{ 5497 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User FrameCount\n"); 5498} 5499 5500void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5501{ 5502 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 5503 (mClient == 0) ? getpid_cached : mClient->pid(), 5504 mFormat, 5505 mChannelMask, 5506 mSessionId, 5507 mFrameCount, 5508 mState, 5509 mCblk->sampleRate, 5510 mCblk->server, 5511 mCblk->user, 5512 mCblk->frameCount); 5513} 5514 5515 5516// ---------------------------------------------------------------------------- 5517 5518AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5519 PlaybackThread *playbackThread, 5520 DuplicatingThread *sourceThread, 5521 uint32_t sampleRate, 5522 audio_format_t format, 5523 audio_channel_mask_t channelMask, 5524 int frameCount) 5525 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5526 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5527 mActive(false), mSourceThread(sourceThread) 5528{ 5529 5530 if (mCblk != NULL) { 5531 mCblk->flags |= CBLK_DIRECTION; 5532 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5533 mOutBuffer.frameCount = 0; 5534 playbackThread->mTracks.add(this); 5535 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5536 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5537 mCblk, mBuffer, mCblk->buffers, 5538 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5539 } else { 5540 ALOGW("Error creating output track on thread %p", playbackThread); 5541 } 5542} 5543 5544AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5545{ 5546 clearBufferQueue(); 5547} 5548 5549status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5550 int triggerSession) 5551{ 5552 status_t status = Track::start(event, triggerSession); 5553 if (status != NO_ERROR) { 5554 return status; 5555 } 5556 5557 mActive = true; 5558 mRetryCount = 127; 5559 return status; 5560} 5561 5562void AudioFlinger::PlaybackThread::OutputTrack::stop() 5563{ 5564 Track::stop(); 5565 clearBufferQueue(); 5566 mOutBuffer.frameCount = 0; 5567 mActive = false; 5568} 5569 5570bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5571{ 5572 Buffer *pInBuffer; 5573 Buffer inBuffer; 5574 uint32_t channelCount = mChannelCount; 5575 bool outputBufferFull = false; 5576 inBuffer.frameCount = frames; 5577 inBuffer.i16 = data; 5578 5579 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5580 5581 if (!mActive && frames != 0) { 5582 start(); 5583 sp<ThreadBase> thread = mThread.promote(); 5584 if (thread != 0) { 5585 MixerThread *mixerThread = (MixerThread *)thread.get(); 5586 if (mCblk->frameCount > frames){ 5587 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5588 uint32_t startFrames = (mCblk->frameCount - frames); 5589 pInBuffer = new Buffer; 5590 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5591 pInBuffer->frameCount = startFrames; 5592 pInBuffer->i16 = pInBuffer->mBuffer; 5593 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5594 mBufferQueue.add(pInBuffer); 5595 } else { 5596 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5597 } 5598 } 5599 } 5600 } 5601 5602 while (waitTimeLeftMs) { 5603 // First write pending buffers, then new data 5604 if (mBufferQueue.size()) { 5605 pInBuffer = mBufferQueue.itemAt(0); 5606 } else { 5607 pInBuffer = &inBuffer; 5608 } 5609 5610 if (pInBuffer->frameCount == 0) { 5611 break; 5612 } 5613 5614 if (mOutBuffer.frameCount == 0) { 5615 mOutBuffer.frameCount = pInBuffer->frameCount; 5616 nsecs_t startTime = systemTime(); 5617 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5618 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 5619 mThread.unsafe_get()); 5620 outputBufferFull = true; 5621 break; 5622 } 5623 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5624 if (waitTimeLeftMs >= waitTimeMs) { 5625 waitTimeLeftMs -= waitTimeMs; 5626 } else { 5627 waitTimeLeftMs = 0; 5628 } 5629 } 5630 5631 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 5632 pInBuffer->frameCount; 5633 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5634 mCblk->stepUser(outFrames); 5635 pInBuffer->frameCount -= outFrames; 5636 pInBuffer->i16 += outFrames * channelCount; 5637 mOutBuffer.frameCount -= outFrames; 5638 mOutBuffer.i16 += outFrames * channelCount; 5639 5640 if (pInBuffer->frameCount == 0) { 5641 if (mBufferQueue.size()) { 5642 mBufferQueue.removeAt(0); 5643 delete [] pInBuffer->mBuffer; 5644 delete pInBuffer; 5645 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 5646 mThread.unsafe_get(), mBufferQueue.size()); 5647 } else { 5648 break; 5649 } 5650 } 5651 } 5652 5653 // If we could not write all frames, allocate a buffer and queue it for next time. 5654 if (inBuffer.frameCount) { 5655 sp<ThreadBase> thread = mThread.promote(); 5656 if (thread != 0 && !thread->standby()) { 5657 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5658 pInBuffer = new Buffer; 5659 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5660 pInBuffer->frameCount = inBuffer.frameCount; 5661 pInBuffer->i16 = pInBuffer->mBuffer; 5662 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 5663 sizeof(int16_t)); 5664 mBufferQueue.add(pInBuffer); 5665 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 5666 mThread.unsafe_get(), mBufferQueue.size()); 5667 } else { 5668 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 5669 mThread.unsafe_get(), this); 5670 } 5671 } 5672 } 5673 5674 // Calling write() with a 0 length buffer, means that no more data will be written: 5675 // If no more buffers are pending, fill output track buffer to make sure it is started 5676 // by output mixer. 5677 if (frames == 0 && mBufferQueue.size() == 0) { 5678 if (mCblk->user < mCblk->frameCount) { 5679 frames = mCblk->frameCount - mCblk->user; 5680 pInBuffer = new Buffer; 5681 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5682 pInBuffer->frameCount = frames; 5683 pInBuffer->i16 = pInBuffer->mBuffer; 5684 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5685 mBufferQueue.add(pInBuffer); 5686 } else if (mActive) { 5687 stop(); 5688 } 5689 } 5690 5691 return outputBufferFull; 5692} 5693 5694status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 5695 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5696{ 5697 int active; 5698 status_t result; 5699 audio_track_cblk_t* cblk = mCblk; 5700 uint32_t framesReq = buffer->frameCount; 5701 5702 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5703 buffer->frameCount = 0; 5704 5705 uint32_t framesAvail = cblk->framesAvailable(); 5706 5707 5708 if (framesAvail == 0) { 5709 Mutex::Autolock _l(cblk->lock); 5710 goto start_loop_here; 5711 while (framesAvail == 0) { 5712 active = mActive; 5713 if (CC_UNLIKELY(!active)) { 5714 ALOGV("Not active and NO_MORE_BUFFERS"); 5715 return NO_MORE_BUFFERS; 5716 } 5717 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5718 if (result != NO_ERROR) { 5719 return NO_MORE_BUFFERS; 5720 } 5721 // read the server count again 5722 start_loop_here: 5723 framesAvail = cblk->framesAvailable_l(); 5724 } 5725 } 5726 5727// if (framesAvail < framesReq) { 5728// return NO_MORE_BUFFERS; 5729// } 5730 5731 if (framesReq > framesAvail) { 5732 framesReq = framesAvail; 5733 } 5734 5735 uint32_t u = cblk->user; 5736 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5737 5738 if (framesReq > bufferEnd - u) { 5739 framesReq = bufferEnd - u; 5740 } 5741 5742 buffer->frameCount = framesReq; 5743 buffer->raw = (void *)cblk->buffer(u); 5744 return NO_ERROR; 5745} 5746 5747 5748void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5749{ 5750 size_t size = mBufferQueue.size(); 5751 5752 for (size_t i = 0; i < size; i++) { 5753 Buffer *pBuffer = mBufferQueue.itemAt(i); 5754 delete [] pBuffer->mBuffer; 5755 delete pBuffer; 5756 } 5757 mBufferQueue.clear(); 5758} 5759 5760// ---------------------------------------------------------------------------- 5761 5762AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5763 : RefBase(), 5764 mAudioFlinger(audioFlinger), 5765 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5766 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5767 mPid(pid), 5768 mTimedTrackCount(0) 5769{ 5770 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5771} 5772 5773// Client destructor must be called with AudioFlinger::mLock held 5774AudioFlinger::Client::~Client() 5775{ 5776 mAudioFlinger->removeClient_l(mPid); 5777} 5778 5779sp<MemoryDealer> AudioFlinger::Client::heap() const 5780{ 5781 return mMemoryDealer; 5782} 5783 5784// Reserve one of the limited slots for a timed audio track associated 5785// with this client 5786bool AudioFlinger::Client::reserveTimedTrack() 5787{ 5788 const int kMaxTimedTracksPerClient = 4; 5789 5790 Mutex::Autolock _l(mTimedTrackLock); 5791 5792 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5793 ALOGW("can not create timed track - pid %d has exceeded the limit", 5794 mPid); 5795 return false; 5796 } 5797 5798 mTimedTrackCount++; 5799 return true; 5800} 5801 5802// Release a slot for a timed audio track 5803void AudioFlinger::Client::releaseTimedTrack() 5804{ 5805 Mutex::Autolock _l(mTimedTrackLock); 5806 mTimedTrackCount--; 5807} 5808 5809// ---------------------------------------------------------------------------- 5810 5811AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5812 const sp<IAudioFlingerClient>& client, 5813 pid_t pid) 5814 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5815{ 5816} 5817 5818AudioFlinger::NotificationClient::~NotificationClient() 5819{ 5820} 5821 5822void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5823{ 5824 sp<NotificationClient> keep(this); 5825 mAudioFlinger->removeNotificationClient(mPid); 5826} 5827 5828// ---------------------------------------------------------------------------- 5829 5830AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5831 : BnAudioTrack(), 5832 mTrack(track) 5833{ 5834} 5835 5836AudioFlinger::TrackHandle::~TrackHandle() { 5837 // just stop the track on deletion, associated resources 5838 // will be freed from the main thread once all pending buffers have 5839 // been played. Unless it's not in the active track list, in which 5840 // case we free everything now... 5841 mTrack->destroy(); 5842} 5843 5844sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5845 return mTrack->getCblk(); 5846} 5847 5848status_t AudioFlinger::TrackHandle::start() { 5849 return mTrack->start(); 5850} 5851 5852void AudioFlinger::TrackHandle::stop() { 5853 mTrack->stop(); 5854} 5855 5856void AudioFlinger::TrackHandle::flush() { 5857 mTrack->flush(); 5858} 5859 5860void AudioFlinger::TrackHandle::mute(bool e) { 5861 mTrack->mute(e); 5862} 5863 5864void AudioFlinger::TrackHandle::pause() { 5865 mTrack->pause(); 5866} 5867 5868status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5869{ 5870 return mTrack->attachAuxEffect(EffectId); 5871} 5872 5873status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5874 sp<IMemory>* buffer) { 5875 if (!mTrack->isTimedTrack()) 5876 return INVALID_OPERATION; 5877 5878 PlaybackThread::TimedTrack* tt = 5879 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5880 return tt->allocateTimedBuffer(size, buffer); 5881} 5882 5883status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5884 int64_t pts) { 5885 if (!mTrack->isTimedTrack()) 5886 return INVALID_OPERATION; 5887 5888 PlaybackThread::TimedTrack* tt = 5889 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5890 return tt->queueTimedBuffer(buffer, pts); 5891} 5892 5893status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5894 const LinearTransform& xform, int target) { 5895 5896 if (!mTrack->isTimedTrack()) 5897 return INVALID_OPERATION; 5898 5899 PlaybackThread::TimedTrack* tt = 5900 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5901 return tt->setMediaTimeTransform( 5902 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5903} 5904 5905status_t AudioFlinger::TrackHandle::onTransact( 5906 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5907{ 5908 return BnAudioTrack::onTransact(code, data, reply, flags); 5909} 5910 5911// ---------------------------------------------------------------------------- 5912 5913sp<IAudioRecord> AudioFlinger::openRecord( 5914 pid_t pid, 5915 audio_io_handle_t input, 5916 uint32_t sampleRate, 5917 audio_format_t format, 5918 audio_channel_mask_t channelMask, 5919 int frameCount, 5920 IAudioFlinger::track_flags_t flags, 5921 pid_t tid, 5922 int *sessionId, 5923 status_t *status) 5924{ 5925 sp<RecordThread::RecordTrack> recordTrack; 5926 sp<RecordHandle> recordHandle; 5927 sp<Client> client; 5928 status_t lStatus; 5929 RecordThread *thread; 5930 size_t inFrameCount; 5931 int lSessionId; 5932 5933 // check calling permissions 5934 if (!recordingAllowed()) { 5935 lStatus = PERMISSION_DENIED; 5936 goto Exit; 5937 } 5938 5939 // add client to list 5940 { // scope for mLock 5941 Mutex::Autolock _l(mLock); 5942 thread = checkRecordThread_l(input); 5943 if (thread == NULL) { 5944 lStatus = BAD_VALUE; 5945 goto Exit; 5946 } 5947 5948 client = registerPid_l(pid); 5949 5950 // If no audio session id is provided, create one here 5951 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5952 lSessionId = *sessionId; 5953 } else { 5954 lSessionId = nextUniqueId(); 5955 if (sessionId != NULL) { 5956 *sessionId = lSessionId; 5957 } 5958 } 5959 // create new record track. 5960 // The record track uses one track in mHardwareMixerThread by convention. 5961 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5962 frameCount, lSessionId, flags, tid, &lStatus); 5963 } 5964 if (lStatus != NO_ERROR) { 5965 // remove local strong reference to Client before deleting the RecordTrack so that the 5966 // Client destructor is called by the TrackBase destructor with mLock held 5967 client.clear(); 5968 recordTrack.clear(); 5969 goto Exit; 5970 } 5971 5972 // return to handle to client 5973 recordHandle = new RecordHandle(recordTrack); 5974 lStatus = NO_ERROR; 5975 5976Exit: 5977 if (status) { 5978 *status = lStatus; 5979 } 5980 return recordHandle; 5981} 5982 5983// ---------------------------------------------------------------------------- 5984 5985AudioFlinger::RecordHandle::RecordHandle( 5986 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5987 : BnAudioRecord(), 5988 mRecordTrack(recordTrack) 5989{ 5990} 5991 5992AudioFlinger::RecordHandle::~RecordHandle() { 5993 stop_nonvirtual(); 5994 mRecordTrack->destroy(); 5995} 5996 5997sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5998 return mRecordTrack->getCblk(); 5999} 6000 6001status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 6002 int triggerSession) { 6003 ALOGV("RecordHandle::start()"); 6004 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 6005} 6006 6007void AudioFlinger::RecordHandle::stop() { 6008 stop_nonvirtual(); 6009} 6010 6011void AudioFlinger::RecordHandle::stop_nonvirtual() { 6012 ALOGV("RecordHandle::stop()"); 6013 mRecordTrack->stop(); 6014} 6015 6016status_t AudioFlinger::RecordHandle::onTransact( 6017 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6018{ 6019 return BnAudioRecord::onTransact(code, data, reply, flags); 6020} 6021 6022// ---------------------------------------------------------------------------- 6023 6024AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6025 AudioStreamIn *input, 6026 uint32_t sampleRate, 6027 audio_channel_mask_t channelMask, 6028 audio_io_handle_t id, 6029 audio_devices_t device, 6030 const sp<NBAIO_Sink>& teeSink) : 6031 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 6032 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 6033 // mRsmpInIndex and mInputBytes set by readInputParameters() 6034 mReqChannelCount(popcount(channelMask)), 6035 mReqSampleRate(sampleRate), 6036 // mBytesRead is only meaningful while active, and so is cleared in start() 6037 // (but might be better to also clear here for dump?) 6038 mTeeSink(teeSink) 6039{ 6040 snprintf(mName, kNameLength, "AudioIn_%X", id); 6041 6042 readInputParameters(); 6043 6044} 6045 6046 6047AudioFlinger::RecordThread::~RecordThread() 6048{ 6049 delete[] mRsmpInBuffer; 6050 delete mResampler; 6051 delete[] mRsmpOutBuffer; 6052} 6053 6054void AudioFlinger::RecordThread::onFirstRef() 6055{ 6056 run(mName, PRIORITY_URGENT_AUDIO); 6057} 6058 6059status_t AudioFlinger::RecordThread::readyToRun() 6060{ 6061 status_t status = initCheck(); 6062 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 6063 return status; 6064} 6065 6066bool AudioFlinger::RecordThread::threadLoop() 6067{ 6068 AudioBufferProvider::Buffer buffer; 6069 sp<RecordTrack> activeTrack; 6070 Vector< sp<EffectChain> > effectChains; 6071 6072 nsecs_t lastWarning = 0; 6073 6074 inputStandBy(); 6075 acquireWakeLock(); 6076 6077 // used to verify we've read at least once before evaluating how many bytes were read 6078 bool readOnce = false; 6079 6080 // start recording 6081 while (!exitPending()) { 6082 6083 processConfigEvents(); 6084 6085 { // scope for mLock 6086 Mutex::Autolock _l(mLock); 6087 checkForNewParameters_l(); 6088 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 6089 standby(); 6090 6091 if (exitPending()) break; 6092 6093 releaseWakeLock_l(); 6094 ALOGV("RecordThread: loop stopping"); 6095 // go to sleep 6096 mWaitWorkCV.wait(mLock); 6097 ALOGV("RecordThread: loop starting"); 6098 acquireWakeLock_l(); 6099 continue; 6100 } 6101 if (mActiveTrack != 0) { 6102 if (mActiveTrack->mState == TrackBase::PAUSING) { 6103 standby(); 6104 mActiveTrack.clear(); 6105 mStartStopCond.broadcast(); 6106 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6107 if (mReqChannelCount != mActiveTrack->channelCount()) { 6108 mActiveTrack.clear(); 6109 mStartStopCond.broadcast(); 6110 } else if (readOnce) { 6111 // record start succeeds only if first read from audio input 6112 // succeeds 6113 if (mBytesRead >= 0) { 6114 mActiveTrack->mState = TrackBase::ACTIVE; 6115 } else { 6116 mActiveTrack.clear(); 6117 } 6118 mStartStopCond.broadcast(); 6119 } 6120 mStandby = false; 6121 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6122 removeTrack_l(mActiveTrack); 6123 mActiveTrack.clear(); 6124 } 6125 } 6126 lockEffectChains_l(effectChains); 6127 } 6128 6129 if (mActiveTrack != 0) { 6130 if (mActiveTrack->mState != TrackBase::ACTIVE && 6131 mActiveTrack->mState != TrackBase::RESUMING) { 6132 unlockEffectChains(effectChains); 6133 usleep(kRecordThreadSleepUs); 6134 continue; 6135 } 6136 for (size_t i = 0; i < effectChains.size(); i ++) { 6137 effectChains[i]->process_l(); 6138 } 6139 6140 buffer.frameCount = mFrameCount; 6141 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6142 readOnce = true; 6143 size_t framesOut = buffer.frameCount; 6144 if (mResampler == NULL) { 6145 // no resampling 6146 while (framesOut) { 6147 size_t framesIn = mFrameCount - mRsmpInIndex; 6148 if (framesIn) { 6149 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6150 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 6151 mActiveTrack->mCblk->frameSize; 6152 if (framesIn > framesOut) 6153 framesIn = framesOut; 6154 mRsmpInIndex += framesIn; 6155 framesOut -= framesIn; 6156 if ((int)mChannelCount == mReqChannelCount || 6157 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6158 memcpy(dst, src, framesIn * mFrameSize); 6159 } else { 6160 if (mChannelCount == 1) { 6161 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6162 (int16_t *)src, framesIn); 6163 } else { 6164 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6165 (int16_t *)src, framesIn); 6166 } 6167 } 6168 } 6169 if (framesOut && mFrameCount == mRsmpInIndex) { 6170 void *readInto; 6171 if (framesOut == mFrameCount && 6172 ((int)mChannelCount == mReqChannelCount || 6173 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6174 readInto = buffer.raw; 6175 framesOut = 0; 6176 } else { 6177 readInto = mRsmpInBuffer; 6178 mRsmpInIndex = 0; 6179 } 6180 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 6181 if (mBytesRead <= 0) { 6182 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 6183 { 6184 ALOGE("Error reading audio input"); 6185 // Force input into standby so that it tries to 6186 // recover at next read attempt 6187 inputStandBy(); 6188 usleep(kRecordThreadSleepUs); 6189 } 6190 mRsmpInIndex = mFrameCount; 6191 framesOut = 0; 6192 buffer.frameCount = 0; 6193 } else if (mTeeSink != 0) { 6194 (void) mTeeSink->write(readInto, 6195 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 6196 } 6197 } 6198 } 6199 } else { 6200 // resampling 6201 6202 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6203 // alter output frame count as if we were expecting stereo samples 6204 if (mChannelCount == 1 && mReqChannelCount == 1) { 6205 framesOut >>= 1; 6206 } 6207 mResampler->resample(mRsmpOutBuffer, framesOut, 6208 this /* AudioBufferProvider* */); 6209 // ditherAndClamp() works as long as all buffers returned by 6210 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 6211 if (mChannelCount == 2 && mReqChannelCount == 1) { 6212 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6213 // the resampler always outputs stereo samples: 6214 // do post stereo to mono conversion 6215 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6216 framesOut); 6217 } else { 6218 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6219 } 6220 6221 } 6222 if (mFramestoDrop == 0) { 6223 mActiveTrack->releaseBuffer(&buffer); 6224 } else { 6225 if (mFramestoDrop > 0) { 6226 mFramestoDrop -= buffer.frameCount; 6227 if (mFramestoDrop <= 0) { 6228 clearSyncStartEvent(); 6229 } 6230 } else { 6231 mFramestoDrop += buffer.frameCount; 6232 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6233 mSyncStartEvent->isCancelled()) { 6234 ALOGW("Synced record %s, session %d, trigger session %d", 6235 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6236 mActiveTrack->sessionId(), 6237 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6238 clearSyncStartEvent(); 6239 } 6240 } 6241 } 6242 mActiveTrack->clearOverflow(); 6243 } 6244 // client isn't retrieving buffers fast enough 6245 else { 6246 if (!mActiveTrack->setOverflow()) { 6247 nsecs_t now = systemTime(); 6248 if ((now - lastWarning) > kWarningThrottleNs) { 6249 ALOGW("RecordThread: buffer overflow"); 6250 lastWarning = now; 6251 } 6252 } 6253 // Release the processor for a while before asking for a new buffer. 6254 // This will give the application more chance to read from the buffer and 6255 // clear the overflow. 6256 usleep(kRecordThreadSleepUs); 6257 } 6258 } 6259 // enable changes in effect chain 6260 unlockEffectChains(effectChains); 6261 effectChains.clear(); 6262 } 6263 6264 standby(); 6265 6266 { 6267 Mutex::Autolock _l(mLock); 6268 mActiveTrack.clear(); 6269 mStartStopCond.broadcast(); 6270 } 6271 6272 releaseWakeLock(); 6273 6274 ALOGV("RecordThread %p exiting", this); 6275 return false; 6276} 6277 6278void AudioFlinger::RecordThread::standby() 6279{ 6280 if (!mStandby) { 6281 inputStandBy(); 6282 mStandby = true; 6283 } 6284} 6285 6286void AudioFlinger::RecordThread::inputStandBy() 6287{ 6288 mInput->stream->common.standby(&mInput->stream->common); 6289} 6290 6291sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6292 const sp<AudioFlinger::Client>& client, 6293 uint32_t sampleRate, 6294 audio_format_t format, 6295 audio_channel_mask_t channelMask, 6296 int frameCount, 6297 int sessionId, 6298 IAudioFlinger::track_flags_t flags, 6299 pid_t tid, 6300 status_t *status) 6301{ 6302 sp<RecordTrack> track; 6303 status_t lStatus; 6304 6305 lStatus = initCheck(); 6306 if (lStatus != NO_ERROR) { 6307 ALOGE("Audio driver not initialized."); 6308 goto Exit; 6309 } 6310 6311 // FIXME use flags and tid similar to createTrack_l() 6312 6313 { // scope for mLock 6314 Mutex::Autolock _l(mLock); 6315 6316 track = new RecordTrack(this, client, sampleRate, 6317 format, channelMask, frameCount, sessionId); 6318 6319 if (track->getCblk() == 0) { 6320 lStatus = NO_MEMORY; 6321 goto Exit; 6322 } 6323 mTracks.add(track); 6324 6325 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6326 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6327 mAudioFlinger->btNrecIsOff(); 6328 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6329 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6330 } 6331 lStatus = NO_ERROR; 6332 6333Exit: 6334 if (status) { 6335 *status = lStatus; 6336 } 6337 return track; 6338} 6339 6340status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6341 AudioSystem::sync_event_t event, 6342 int triggerSession) 6343{ 6344 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6345 sp<ThreadBase> strongMe = this; 6346 status_t status = NO_ERROR; 6347 6348 if (event == AudioSystem::SYNC_EVENT_NONE) { 6349 clearSyncStartEvent(); 6350 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6351 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6352 triggerSession, 6353 recordTrack->sessionId(), 6354 syncStartEventCallback, 6355 this); 6356 // Sync event can be cancelled by the trigger session if the track is not in a 6357 // compatible state in which case we start record immediately 6358 if (mSyncStartEvent->isCancelled()) { 6359 clearSyncStartEvent(); 6360 } else { 6361 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6362 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6363 } 6364 } 6365 6366 { 6367 AutoMutex lock(mLock); 6368 if (mActiveTrack != 0) { 6369 if (recordTrack != mActiveTrack.get()) { 6370 status = -EBUSY; 6371 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6372 mActiveTrack->mState = TrackBase::ACTIVE; 6373 } 6374 return status; 6375 } 6376 6377 recordTrack->mState = TrackBase::IDLE; 6378 mActiveTrack = recordTrack; 6379 mLock.unlock(); 6380 status_t status = AudioSystem::startInput(mId); 6381 mLock.lock(); 6382 if (status != NO_ERROR) { 6383 mActiveTrack.clear(); 6384 clearSyncStartEvent(); 6385 return status; 6386 } 6387 mRsmpInIndex = mFrameCount; 6388 mBytesRead = 0; 6389 if (mResampler != NULL) { 6390 mResampler->reset(); 6391 } 6392 mActiveTrack->mState = TrackBase::RESUMING; 6393 // signal thread to start 6394 ALOGV("Signal record thread"); 6395 mWaitWorkCV.broadcast(); 6396 // do not wait for mStartStopCond if exiting 6397 if (exitPending()) { 6398 mActiveTrack.clear(); 6399 status = INVALID_OPERATION; 6400 goto startError; 6401 } 6402 mStartStopCond.wait(mLock); 6403 if (mActiveTrack == 0) { 6404 ALOGV("Record failed to start"); 6405 status = BAD_VALUE; 6406 goto startError; 6407 } 6408 ALOGV("Record started OK"); 6409 return status; 6410 } 6411startError: 6412 AudioSystem::stopInput(mId); 6413 clearSyncStartEvent(); 6414 return status; 6415} 6416 6417void AudioFlinger::RecordThread::clearSyncStartEvent() 6418{ 6419 if (mSyncStartEvent != 0) { 6420 mSyncStartEvent->cancel(); 6421 } 6422 mSyncStartEvent.clear(); 6423 mFramestoDrop = 0; 6424} 6425 6426void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6427{ 6428 sp<SyncEvent> strongEvent = event.promote(); 6429 6430 if (strongEvent != 0) { 6431 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6432 me->handleSyncStartEvent(strongEvent); 6433 } 6434} 6435 6436void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6437{ 6438 if (event == mSyncStartEvent) { 6439 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6440 // from audio HAL 6441 mFramestoDrop = mFrameCount * 2; 6442 } 6443} 6444 6445bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6446 ALOGV("RecordThread::stop"); 6447 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6448 return false; 6449 } 6450 recordTrack->mState = TrackBase::PAUSING; 6451 // do not wait for mStartStopCond if exiting 6452 if (exitPending()) { 6453 return true; 6454 } 6455 mStartStopCond.wait(mLock); 6456 // if we have been restarted, recordTrack == mActiveTrack.get() here 6457 if (exitPending() || recordTrack != mActiveTrack.get()) { 6458 ALOGV("Record stopped OK"); 6459 return true; 6460 } 6461 return false; 6462} 6463 6464bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6465{ 6466 return false; 6467} 6468 6469status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6470{ 6471#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6472 if (!isValidSyncEvent(event)) { 6473 return BAD_VALUE; 6474 } 6475 6476 int eventSession = event->triggerSession(); 6477 status_t ret = NAME_NOT_FOUND; 6478 6479 Mutex::Autolock _l(mLock); 6480 6481 for (size_t i = 0; i < mTracks.size(); i++) { 6482 sp<RecordTrack> track = mTracks[i]; 6483 if (eventSession == track->sessionId()) { 6484 (void) track->setSyncEvent(event); 6485 ret = NO_ERROR; 6486 } 6487 } 6488 return ret; 6489#else 6490 return BAD_VALUE; 6491#endif 6492} 6493 6494void AudioFlinger::RecordThread::RecordTrack::destroy() 6495{ 6496 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6497 sp<RecordTrack> keep(this); 6498 { 6499 sp<ThreadBase> thread = mThread.promote(); 6500 if (thread != 0) { 6501 if (mState == ACTIVE || mState == RESUMING) { 6502 AudioSystem::stopInput(thread->id()); 6503 } 6504 AudioSystem::releaseInput(thread->id()); 6505 Mutex::Autolock _l(thread->mLock); 6506 RecordThread *recordThread = (RecordThread *) thread.get(); 6507 recordThread->destroyTrack_l(this); 6508 } 6509 } 6510} 6511 6512// destroyTrack_l() must be called with ThreadBase::mLock held 6513void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6514{ 6515 track->mState = TrackBase::TERMINATED; 6516 // active tracks are removed by threadLoop() 6517 if (mActiveTrack != track) { 6518 removeTrack_l(track); 6519 } 6520} 6521 6522void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6523{ 6524 mTracks.remove(track); 6525 // need anything related to effects here? 6526} 6527 6528void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6529{ 6530 dumpInternals(fd, args); 6531 dumpTracks(fd, args); 6532 dumpEffectChains(fd, args); 6533} 6534 6535void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6536{ 6537 const size_t SIZE = 256; 6538 char buffer[SIZE]; 6539 String8 result; 6540 6541 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6542 result.append(buffer); 6543 6544 if (mActiveTrack != 0) { 6545 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6546 result.append(buffer); 6547 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6548 result.append(buffer); 6549 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6550 result.append(buffer); 6551 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6552 result.append(buffer); 6553 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6554 result.append(buffer); 6555 } else { 6556 result.append("No active record client\n"); 6557 } 6558 6559 write(fd, result.string(), result.size()); 6560 6561 dumpBase(fd, args); 6562} 6563 6564void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6565{ 6566 const size_t SIZE = 256; 6567 char buffer[SIZE]; 6568 String8 result; 6569 6570 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6571 result.append(buffer); 6572 RecordTrack::appendDumpHeader(result); 6573 for (size_t i = 0; i < mTracks.size(); ++i) { 6574 sp<RecordTrack> track = mTracks[i]; 6575 if (track != 0) { 6576 track->dump(buffer, SIZE); 6577 result.append(buffer); 6578 } 6579 } 6580 6581 if (mActiveTrack != 0) { 6582 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6583 result.append(buffer); 6584 RecordTrack::appendDumpHeader(result); 6585 mActiveTrack->dump(buffer, SIZE); 6586 result.append(buffer); 6587 6588 } 6589 write(fd, result.string(), result.size()); 6590} 6591 6592// AudioBufferProvider interface 6593status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6594{ 6595 size_t framesReq = buffer->frameCount; 6596 size_t framesReady = mFrameCount - mRsmpInIndex; 6597 int channelCount; 6598 6599 if (framesReady == 0) { 6600 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6601 if (mBytesRead <= 0) { 6602 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 6603 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6604 // Force input into standby so that it tries to 6605 // recover at next read attempt 6606 inputStandBy(); 6607 usleep(kRecordThreadSleepUs); 6608 } 6609 buffer->raw = NULL; 6610 buffer->frameCount = 0; 6611 return NOT_ENOUGH_DATA; 6612 } 6613 mRsmpInIndex = 0; 6614 framesReady = mFrameCount; 6615 } 6616 6617 if (framesReq > framesReady) { 6618 framesReq = framesReady; 6619 } 6620 6621 if (mChannelCount == 1 && mReqChannelCount == 2) { 6622 channelCount = 1; 6623 } else { 6624 channelCount = 2; 6625 } 6626 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6627 buffer->frameCount = framesReq; 6628 return NO_ERROR; 6629} 6630 6631// AudioBufferProvider interface 6632void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6633{ 6634 mRsmpInIndex += buffer->frameCount; 6635 buffer->frameCount = 0; 6636} 6637 6638bool AudioFlinger::RecordThread::checkForNewParameters_l() 6639{ 6640 bool reconfig = false; 6641 6642 while (!mNewParameters.isEmpty()) { 6643 status_t status = NO_ERROR; 6644 String8 keyValuePair = mNewParameters[0]; 6645 AudioParameter param = AudioParameter(keyValuePair); 6646 int value; 6647 audio_format_t reqFormat = mFormat; 6648 int reqSamplingRate = mReqSampleRate; 6649 int reqChannelCount = mReqChannelCount; 6650 6651 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6652 reqSamplingRate = value; 6653 reconfig = true; 6654 } 6655 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6656 reqFormat = (audio_format_t) value; 6657 reconfig = true; 6658 } 6659 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6660 reqChannelCount = popcount(value); 6661 reconfig = true; 6662 } 6663 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6664 // do not accept frame count changes if tracks are open as the track buffer 6665 // size depends on frame count and correct behavior would not be guaranteed 6666 // if frame count is changed after track creation 6667 if (mActiveTrack != 0) { 6668 status = INVALID_OPERATION; 6669 } else { 6670 reconfig = true; 6671 } 6672 } 6673 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6674 // forward device change to effects that have requested to be 6675 // aware of attached audio device. 6676 for (size_t i = 0; i < mEffectChains.size(); i++) { 6677 mEffectChains[i]->setDevice_l(value); 6678 } 6679 6680 // store input device and output device but do not forward output device to audio HAL. 6681 // Note that status is ignored by the caller for output device 6682 // (see AudioFlinger::setParameters() 6683 if (audio_is_output_devices(value)) { 6684 mOutDevice = value; 6685 status = BAD_VALUE; 6686 } else { 6687 mInDevice = value; 6688 // disable AEC and NS if the device is a BT SCO headset supporting those 6689 // pre processings 6690 if (mTracks.size() > 0) { 6691 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6692 mAudioFlinger->btNrecIsOff(); 6693 for (size_t i = 0; i < mTracks.size(); i++) { 6694 sp<RecordTrack> track = mTracks[i]; 6695 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6696 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6697 } 6698 } 6699 } 6700 } 6701 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6702 mAudioSource != (audio_source_t)value) { 6703 // forward device change to effects that have requested to be 6704 // aware of attached audio device. 6705 for (size_t i = 0; i < mEffectChains.size(); i++) { 6706 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6707 } 6708 mAudioSource = (audio_source_t)value; 6709 } 6710 if (status == NO_ERROR) { 6711 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6712 keyValuePair.string()); 6713 if (status == INVALID_OPERATION) { 6714 inputStandBy(); 6715 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6716 keyValuePair.string()); 6717 } 6718 if (reconfig) { 6719 if (status == BAD_VALUE && 6720 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6721 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6722 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 6723 <= (2 * reqSamplingRate)) && 6724 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 6725 <= FCC_2 && 6726 (reqChannelCount <= FCC_2)) { 6727 status = NO_ERROR; 6728 } 6729 if (status == NO_ERROR) { 6730 readInputParameters(); 6731 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6732 } 6733 } 6734 } 6735 6736 mNewParameters.removeAt(0); 6737 6738 mParamStatus = status; 6739 mParamCond.signal(); 6740 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6741 // already timed out waiting for the status and will never signal the condition. 6742 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6743 } 6744 return reconfig; 6745} 6746 6747String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6748{ 6749 char *s; 6750 String8 out_s8 = String8(); 6751 6752 Mutex::Autolock _l(mLock); 6753 if (initCheck() != NO_ERROR) { 6754 return out_s8; 6755 } 6756 6757 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6758 out_s8 = String8(s); 6759 free(s); 6760 return out_s8; 6761} 6762 6763void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6764 AudioSystem::OutputDescriptor desc; 6765 void *param2 = NULL; 6766 6767 switch (event) { 6768 case AudioSystem::INPUT_OPENED: 6769 case AudioSystem::INPUT_CONFIG_CHANGED: 6770 desc.channels = mChannelMask; 6771 desc.samplingRate = mSampleRate; 6772 desc.format = mFormat; 6773 desc.frameCount = mFrameCount; 6774 desc.latency = 0; 6775 param2 = &desc; 6776 break; 6777 6778 case AudioSystem::INPUT_CLOSED: 6779 default: 6780 break; 6781 } 6782 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6783} 6784 6785void AudioFlinger::RecordThread::readInputParameters() 6786{ 6787 delete mRsmpInBuffer; 6788 // mRsmpInBuffer is always assigned a new[] below 6789 delete mRsmpOutBuffer; 6790 mRsmpOutBuffer = NULL; 6791 delete mResampler; 6792 mResampler = NULL; 6793 6794 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6795 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6796 mChannelCount = (uint16_t)popcount(mChannelMask); 6797 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6798 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6799 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6800 mFrameCount = mInputBytes / mFrameSize; 6801 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6802 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6803 6804 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6805 { 6806 int channelCount; 6807 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6808 // stereo to mono post process as the resampler always outputs stereo. 6809 if (mChannelCount == 1 && mReqChannelCount == 2) { 6810 channelCount = 1; 6811 } else { 6812 channelCount = 2; 6813 } 6814 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6815 mResampler->setSampleRate(mSampleRate); 6816 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6817 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6818 6819 // optmization: if mono to mono, alter input frame count as if we were inputing 6820 // stereo samples 6821 if (mChannelCount == 1 && mReqChannelCount == 1) { 6822 mFrameCount >>= 1; 6823 } 6824 6825 } 6826 mRsmpInIndex = mFrameCount; 6827} 6828 6829unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6830{ 6831 Mutex::Autolock _l(mLock); 6832 if (initCheck() != NO_ERROR) { 6833 return 0; 6834 } 6835 6836 return mInput->stream->get_input_frames_lost(mInput->stream); 6837} 6838 6839uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6840{ 6841 Mutex::Autolock _l(mLock); 6842 uint32_t result = 0; 6843 if (getEffectChain_l(sessionId) != 0) { 6844 result = EFFECT_SESSION; 6845 } 6846 6847 for (size_t i = 0; i < mTracks.size(); ++i) { 6848 if (sessionId == mTracks[i]->sessionId()) { 6849 result |= TRACK_SESSION; 6850 break; 6851 } 6852 } 6853 6854 return result; 6855} 6856 6857KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6858{ 6859 KeyedVector<int, bool> ids; 6860 Mutex::Autolock _l(mLock); 6861 for (size_t j = 0; j < mTracks.size(); ++j) { 6862 sp<RecordThread::RecordTrack> track = mTracks[j]; 6863 int sessionId = track->sessionId(); 6864 if (ids.indexOfKey(sessionId) < 0) { 6865 ids.add(sessionId, true); 6866 } 6867 } 6868 return ids; 6869} 6870 6871AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6872{ 6873 Mutex::Autolock _l(mLock); 6874 AudioStreamIn *input = mInput; 6875 mInput = NULL; 6876 return input; 6877} 6878 6879// this method must always be called either with ThreadBase mLock held or inside the thread loop 6880audio_stream_t* AudioFlinger::RecordThread::stream() const 6881{ 6882 if (mInput == NULL) { 6883 return NULL; 6884 } 6885 return &mInput->stream->common; 6886} 6887 6888 6889// ---------------------------------------------------------------------------- 6890 6891audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6892{ 6893 if (!settingsAllowed()) { 6894 return 0; 6895 } 6896 Mutex::Autolock _l(mLock); 6897 return loadHwModule_l(name); 6898} 6899 6900// loadHwModule_l() must be called with AudioFlinger::mLock held 6901audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6902{ 6903 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6904 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6905 ALOGW("loadHwModule() module %s already loaded", name); 6906 return mAudioHwDevs.keyAt(i); 6907 } 6908 } 6909 6910 audio_hw_device_t *dev; 6911 6912 int rc = load_audio_interface(name, &dev); 6913 if (rc) { 6914 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6915 return 0; 6916 } 6917 6918 mHardwareStatus = AUDIO_HW_INIT; 6919 rc = dev->init_check(dev); 6920 mHardwareStatus = AUDIO_HW_IDLE; 6921 if (rc) { 6922 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6923 return 0; 6924 } 6925 6926 // Check and cache this HAL's level of support for master mute and master 6927 // volume. If this is the first HAL opened, and it supports the get 6928 // methods, use the initial values provided by the HAL as the current 6929 // master mute and volume settings. 6930 6931 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6932 { // scope for auto-lock pattern 6933 AutoMutex lock(mHardwareLock); 6934 6935 if (0 == mAudioHwDevs.size()) { 6936 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6937 if (NULL != dev->get_master_volume) { 6938 float mv; 6939 if (OK == dev->get_master_volume(dev, &mv)) { 6940 mMasterVolume = mv; 6941 } 6942 } 6943 6944 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6945 if (NULL != dev->get_master_mute) { 6946 bool mm; 6947 if (OK == dev->get_master_mute(dev, &mm)) { 6948 mMasterMute = mm; 6949 } 6950 } 6951 } 6952 6953 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6954 if ((NULL != dev->set_master_volume) && 6955 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6956 flags = static_cast<AudioHwDevice::Flags>(flags | 6957 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6958 } 6959 6960 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6961 if ((NULL != dev->set_master_mute) && 6962 (OK == dev->set_master_mute(dev, mMasterMute))) { 6963 flags = static_cast<AudioHwDevice::Flags>(flags | 6964 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6965 } 6966 6967 mHardwareStatus = AUDIO_HW_IDLE; 6968 } 6969 6970 audio_module_handle_t handle = nextUniqueId(); 6971 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6972 6973 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6974 name, dev->common.module->name, dev->common.module->id, handle); 6975 6976 return handle; 6977 6978} 6979 6980// ---------------------------------------------------------------------------- 6981 6982int32_t AudioFlinger::getPrimaryOutputSamplingRate() 6983{ 6984 Mutex::Autolock _l(mLock); 6985 PlaybackThread *thread = primaryPlaybackThread_l(); 6986 return thread != NULL ? thread->sampleRate() : 0; 6987} 6988 6989int32_t AudioFlinger::getPrimaryOutputFrameCount() 6990{ 6991 Mutex::Autolock _l(mLock); 6992 PlaybackThread *thread = primaryPlaybackThread_l(); 6993 return thread != NULL ? thread->frameCountHAL() : 0; 6994} 6995 6996// ---------------------------------------------------------------------------- 6997 6998audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6999 audio_devices_t *pDevices, 7000 uint32_t *pSamplingRate, 7001 audio_format_t *pFormat, 7002 audio_channel_mask_t *pChannelMask, 7003 uint32_t *pLatencyMs, 7004 audio_output_flags_t flags) 7005{ 7006 status_t status; 7007 PlaybackThread *thread = NULL; 7008 struct audio_config config = { 7009 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7010 channel_mask: pChannelMask ? *pChannelMask : 0, 7011 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7012 }; 7013 audio_stream_out_t *outStream = NULL; 7014 AudioHwDevice *outHwDev; 7015 7016 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 7017 module, 7018 (pDevices != NULL) ? *pDevices : 0, 7019 config.sample_rate, 7020 config.format, 7021 config.channel_mask, 7022 flags); 7023 7024 if (pDevices == NULL || *pDevices == 0) { 7025 return 0; 7026 } 7027 7028 Mutex::Autolock _l(mLock); 7029 7030 outHwDev = findSuitableHwDev_l(module, *pDevices); 7031 if (outHwDev == NULL) 7032 return 0; 7033 7034 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 7035 audio_io_handle_t id = nextUniqueId(); 7036 7037 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 7038 7039 status = hwDevHal->open_output_stream(hwDevHal, 7040 id, 7041 *pDevices, 7042 (audio_output_flags_t)flags, 7043 &config, 7044 &outStream); 7045 7046 mHardwareStatus = AUDIO_HW_IDLE; 7047 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, " 7048 "Channels %x, status %d", 7049 outStream, 7050 config.sample_rate, 7051 config.format, 7052 config.channel_mask, 7053 status); 7054 7055 if (status == NO_ERROR && outStream != NULL) { 7056 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 7057 7058 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 7059 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 7060 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 7061 thread = new DirectOutputThread(this, output, id, *pDevices); 7062 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 7063 } else { 7064 thread = new MixerThread(this, output, id, *pDevices); 7065 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 7066 } 7067 mPlaybackThreads.add(id, thread); 7068 7069 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 7070 if (pFormat != NULL) *pFormat = config.format; 7071 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 7072 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 7073 7074 // notify client processes of the new output creation 7075 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7076 7077 // the first primary output opened designates the primary hw device 7078 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 7079 ALOGI("Using module %d has the primary audio interface", module); 7080 mPrimaryHardwareDev = outHwDev; 7081 7082 AutoMutex lock(mHardwareLock); 7083 mHardwareStatus = AUDIO_HW_SET_MODE; 7084 hwDevHal->set_mode(hwDevHal, mMode); 7085 mHardwareStatus = AUDIO_HW_IDLE; 7086 } 7087 return id; 7088 } 7089 7090 return 0; 7091} 7092 7093audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 7094 audio_io_handle_t output2) 7095{ 7096 Mutex::Autolock _l(mLock); 7097 MixerThread *thread1 = checkMixerThread_l(output1); 7098 MixerThread *thread2 = checkMixerThread_l(output2); 7099 7100 if (thread1 == NULL || thread2 == NULL) { 7101 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 7102 output2); 7103 return 0; 7104 } 7105 7106 audio_io_handle_t id = nextUniqueId(); 7107 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 7108 thread->addOutputTrack(thread2); 7109 mPlaybackThreads.add(id, thread); 7110 // notify client processes of the new output creation 7111 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 7112 return id; 7113} 7114 7115status_t AudioFlinger::closeOutput(audio_io_handle_t output) 7116{ 7117 return closeOutput_nonvirtual(output); 7118} 7119 7120status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 7121{ 7122 // keep strong reference on the playback thread so that 7123 // it is not destroyed while exit() is executed 7124 sp<PlaybackThread> thread; 7125 { 7126 Mutex::Autolock _l(mLock); 7127 thread = checkPlaybackThread_l(output); 7128 if (thread == NULL) { 7129 return BAD_VALUE; 7130 } 7131 7132 ALOGV("closeOutput() %d", output); 7133 7134 if (thread->type() == ThreadBase::MIXER) { 7135 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7136 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7137 DuplicatingThread *dupThread = 7138 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7139 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7140 } 7141 } 7142 } 7143 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7144 mPlaybackThreads.removeItem(output); 7145 } 7146 thread->exit(); 7147 // The thread entity (active unit of execution) is no longer running here, 7148 // but the ThreadBase container still exists. 7149 7150 if (thread->type() != ThreadBase::DUPLICATING) { 7151 AudioStreamOut *out = thread->clearOutput(); 7152 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7153 // from now on thread->mOutput is NULL 7154 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7155 delete out; 7156 } 7157 return NO_ERROR; 7158} 7159 7160status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7161{ 7162 Mutex::Autolock _l(mLock); 7163 PlaybackThread *thread = checkPlaybackThread_l(output); 7164 7165 if (thread == NULL) { 7166 return BAD_VALUE; 7167 } 7168 7169 ALOGV("suspendOutput() %d", output); 7170 thread->suspend(); 7171 7172 return NO_ERROR; 7173} 7174 7175status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7176{ 7177 Mutex::Autolock _l(mLock); 7178 PlaybackThread *thread = checkPlaybackThread_l(output); 7179 7180 if (thread == NULL) { 7181 return BAD_VALUE; 7182 } 7183 7184 ALOGV("restoreOutput() %d", output); 7185 7186 thread->restore(); 7187 7188 return NO_ERROR; 7189} 7190 7191audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7192 audio_devices_t *pDevices, 7193 uint32_t *pSamplingRate, 7194 audio_format_t *pFormat, 7195 audio_channel_mask_t *pChannelMask) 7196{ 7197 status_t status; 7198 RecordThread *thread = NULL; 7199 struct audio_config config = { 7200 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7201 channel_mask: pChannelMask ? *pChannelMask : 0, 7202 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7203 }; 7204 uint32_t reqSamplingRate = config.sample_rate; 7205 audio_format_t reqFormat = config.format; 7206 audio_channel_mask_t reqChannels = config.channel_mask; 7207 audio_stream_in_t *inStream = NULL; 7208 AudioHwDevice *inHwDev; 7209 7210 if (pDevices == NULL || *pDevices == 0) { 7211 return 0; 7212 } 7213 7214 Mutex::Autolock _l(mLock); 7215 7216 inHwDev = findSuitableHwDev_l(module, *pDevices); 7217 if (inHwDev == NULL) 7218 return 0; 7219 7220 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7221 audio_io_handle_t id = nextUniqueId(); 7222 7223 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7224 &inStream); 7225 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 7226 "status %d", 7227 inStream, 7228 config.sample_rate, 7229 config.format, 7230 config.channel_mask, 7231 status); 7232 7233 // If the input could not be opened with the requested parameters and we can handle the 7234 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 7235 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 7236 if (status == BAD_VALUE && 7237 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7238 (config.sample_rate <= 2 * reqSamplingRate) && 7239 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7240 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7241 inStream = NULL; 7242 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7243 } 7244 7245 if (status == NO_ERROR && inStream != NULL) { 7246 7247 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 7248 // or (re-)create if current Pipe is idle and does not match the new format 7249 sp<NBAIO_Sink> teeSink; 7250#ifdef TEE_SINK_INPUT_FRAMES 7251 enum { 7252 TEE_SINK_NO, // don't copy input 7253 TEE_SINK_NEW, // copy input using a new pipe 7254 TEE_SINK_OLD, // copy input using an existing pipe 7255 } kind; 7256 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 7257 popcount(inStream->common.get_channels(&inStream->common))); 7258 if (format == Format_Invalid) { 7259 kind = TEE_SINK_NO; 7260 } else if (mRecordTeeSink == 0) { 7261 kind = TEE_SINK_NEW; 7262 } else if (mRecordTeeSink->getStrongCount() != 1) { 7263 kind = TEE_SINK_NO; 7264 } else if (format == mRecordTeeSink->format()) { 7265 kind = TEE_SINK_OLD; 7266 } else { 7267 kind = TEE_SINK_NEW; 7268 } 7269 switch (kind) { 7270 case TEE_SINK_NEW: { 7271 Pipe *pipe = new Pipe(TEE_SINK_INPUT_FRAMES, format); 7272 size_t numCounterOffers = 0; 7273 const NBAIO_Format offers[1] = {format}; 7274 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 7275 ALOG_ASSERT(index == 0); 7276 PipeReader *pipeReader = new PipeReader(*pipe); 7277 numCounterOffers = 0; 7278 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 7279 ALOG_ASSERT(index == 0); 7280 mRecordTeeSink = pipe; 7281 mRecordTeeSource = pipeReader; 7282 teeSink = pipe; 7283 } 7284 break; 7285 case TEE_SINK_OLD: 7286 teeSink = mRecordTeeSink; 7287 break; 7288 case TEE_SINK_NO: 7289 default: 7290 break; 7291 } 7292#endif 7293 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7294 7295 // Start record thread 7296 // RecorThread require both input and output device indication to forward to audio 7297 // pre processing modules 7298 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7299 7300 thread = new RecordThread(this, 7301 input, 7302 reqSamplingRate, 7303 reqChannels, 7304 id, 7305 device, teeSink); 7306 mRecordThreads.add(id, thread); 7307 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7308 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7309 if (pFormat != NULL) *pFormat = config.format; 7310 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7311 7312 // notify client processes of the new input creation 7313 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7314 return id; 7315 } 7316 7317 return 0; 7318} 7319 7320status_t AudioFlinger::closeInput(audio_io_handle_t input) 7321{ 7322 return closeInput_nonvirtual(input); 7323} 7324 7325status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7326{ 7327 // keep strong reference on the record thread so that 7328 // it is not destroyed while exit() is executed 7329 sp<RecordThread> thread; 7330 { 7331 Mutex::Autolock _l(mLock); 7332 thread = checkRecordThread_l(input); 7333 if (thread == 0) { 7334 return BAD_VALUE; 7335 } 7336 7337 ALOGV("closeInput() %d", input); 7338 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7339 mRecordThreads.removeItem(input); 7340 } 7341 thread->exit(); 7342 // The thread entity (active unit of execution) is no longer running here, 7343 // but the ThreadBase container still exists. 7344 7345 AudioStreamIn *in = thread->clearInput(); 7346 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7347 // from now on thread->mInput is NULL 7348 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7349 delete in; 7350 7351 return NO_ERROR; 7352} 7353 7354status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7355{ 7356 Mutex::Autolock _l(mLock); 7357 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7358 7359 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7360 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7361 thread->invalidateTracks(stream); 7362 } 7363 7364 return NO_ERROR; 7365} 7366 7367 7368int AudioFlinger::newAudioSessionId() 7369{ 7370 return nextUniqueId(); 7371} 7372 7373void AudioFlinger::acquireAudioSessionId(int audioSession) 7374{ 7375 Mutex::Autolock _l(mLock); 7376 pid_t caller = IPCThreadState::self()->getCallingPid(); 7377 ALOGV("acquiring %d from %d", audioSession, caller); 7378 size_t num = mAudioSessionRefs.size(); 7379 for (size_t i = 0; i< num; i++) { 7380 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7381 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7382 ref->mCnt++; 7383 ALOGV(" incremented refcount to %d", ref->mCnt); 7384 return; 7385 } 7386 } 7387 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7388 ALOGV(" added new entry for %d", audioSession); 7389} 7390 7391void AudioFlinger::releaseAudioSessionId(int audioSession) 7392{ 7393 Mutex::Autolock _l(mLock); 7394 pid_t caller = IPCThreadState::self()->getCallingPid(); 7395 ALOGV("releasing %d from %d", audioSession, caller); 7396 size_t num = mAudioSessionRefs.size(); 7397 for (size_t i = 0; i< num; i++) { 7398 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7399 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7400 ref->mCnt--; 7401 ALOGV(" decremented refcount to %d", ref->mCnt); 7402 if (ref->mCnt == 0) { 7403 mAudioSessionRefs.removeAt(i); 7404 delete ref; 7405 purgeStaleEffects_l(); 7406 } 7407 return; 7408 } 7409 } 7410 ALOGW("session id %d not found for pid %d", audioSession, caller); 7411} 7412 7413void AudioFlinger::purgeStaleEffects_l() { 7414 7415 ALOGV("purging stale effects"); 7416 7417 Vector< sp<EffectChain> > chains; 7418 7419 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7420 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7421 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7422 sp<EffectChain> ec = t->mEffectChains[j]; 7423 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7424 chains.push(ec); 7425 } 7426 } 7427 } 7428 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7429 sp<RecordThread> t = mRecordThreads.valueAt(i); 7430 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7431 sp<EffectChain> ec = t->mEffectChains[j]; 7432 chains.push(ec); 7433 } 7434 } 7435 7436 for (size_t i = 0; i < chains.size(); i++) { 7437 sp<EffectChain> ec = chains[i]; 7438 int sessionid = ec->sessionId(); 7439 sp<ThreadBase> t = ec->mThread.promote(); 7440 if (t == 0) { 7441 continue; 7442 } 7443 size_t numsessionrefs = mAudioSessionRefs.size(); 7444 bool found = false; 7445 for (size_t k = 0; k < numsessionrefs; k++) { 7446 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7447 if (ref->mSessionid == sessionid) { 7448 ALOGV(" session %d still exists for %d with %d refs", 7449 sessionid, ref->mPid, ref->mCnt); 7450 found = true; 7451 break; 7452 } 7453 } 7454 if (!found) { 7455 Mutex::Autolock _l (t->mLock); 7456 // remove all effects from the chain 7457 while (ec->mEffects.size()) { 7458 sp<EffectModule> effect = ec->mEffects[0]; 7459 effect->unPin(); 7460 t->removeEffect_l(effect); 7461 if (effect->purgeHandles()) { 7462 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7463 } 7464 AudioSystem::unregisterEffect(effect->id()); 7465 } 7466 } 7467 } 7468 return; 7469} 7470 7471// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7472AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7473{ 7474 return mPlaybackThreads.valueFor(output).get(); 7475} 7476 7477// checkMixerThread_l() must be called with AudioFlinger::mLock held 7478AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7479{ 7480 PlaybackThread *thread = checkPlaybackThread_l(output); 7481 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7482} 7483 7484// checkRecordThread_l() must be called with AudioFlinger::mLock held 7485AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7486{ 7487 return mRecordThreads.valueFor(input).get(); 7488} 7489 7490uint32_t AudioFlinger::nextUniqueId() 7491{ 7492 return android_atomic_inc(&mNextUniqueId); 7493} 7494 7495AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7496{ 7497 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7498 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7499 AudioStreamOut *output = thread->getOutput(); 7500 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7501 return thread; 7502 } 7503 } 7504 return NULL; 7505} 7506 7507audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7508{ 7509 PlaybackThread *thread = primaryPlaybackThread_l(); 7510 7511 if (thread == NULL) { 7512 return 0; 7513 } 7514 7515 return thread->outDevice(); 7516} 7517 7518sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7519 int triggerSession, 7520 int listenerSession, 7521 sync_event_callback_t callBack, 7522 void *cookie) 7523{ 7524 Mutex::Autolock _l(mLock); 7525 7526 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7527 status_t playStatus = NAME_NOT_FOUND; 7528 status_t recStatus = NAME_NOT_FOUND; 7529 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7530 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7531 if (playStatus == NO_ERROR) { 7532 return event; 7533 } 7534 } 7535 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7536 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7537 if (recStatus == NO_ERROR) { 7538 return event; 7539 } 7540 } 7541 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7542 mPendingSyncEvents.add(event); 7543 } else { 7544 ALOGV("createSyncEvent() invalid event %d", event->type()); 7545 event.clear(); 7546 } 7547 return event; 7548} 7549 7550// ---------------------------------------------------------------------------- 7551// Effect management 7552// ---------------------------------------------------------------------------- 7553 7554 7555status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7556{ 7557 Mutex::Autolock _l(mLock); 7558 return EffectQueryNumberEffects(numEffects); 7559} 7560 7561status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7562{ 7563 Mutex::Autolock _l(mLock); 7564 return EffectQueryEffect(index, descriptor); 7565} 7566 7567status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7568 effect_descriptor_t *descriptor) const 7569{ 7570 Mutex::Autolock _l(mLock); 7571 return EffectGetDescriptor(pUuid, descriptor); 7572} 7573 7574 7575sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7576 effect_descriptor_t *pDesc, 7577 const sp<IEffectClient>& effectClient, 7578 int32_t priority, 7579 audio_io_handle_t io, 7580 int sessionId, 7581 status_t *status, 7582 int *id, 7583 int *enabled) 7584{ 7585 status_t lStatus = NO_ERROR; 7586 sp<EffectHandle> handle; 7587 effect_descriptor_t desc; 7588 7589 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7590 pid, effectClient.get(), priority, sessionId, io); 7591 7592 if (pDesc == NULL) { 7593 lStatus = BAD_VALUE; 7594 goto Exit; 7595 } 7596 7597 // check audio settings permission for global effects 7598 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7599 lStatus = PERMISSION_DENIED; 7600 goto Exit; 7601 } 7602 7603 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7604 // that can only be created by audio policy manager (running in same process) 7605 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7606 lStatus = PERMISSION_DENIED; 7607 goto Exit; 7608 } 7609 7610 if (io == 0) { 7611 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7612 // output must be specified by AudioPolicyManager when using session 7613 // AUDIO_SESSION_OUTPUT_STAGE 7614 lStatus = BAD_VALUE; 7615 goto Exit; 7616 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7617 // if the output returned by getOutputForEffect() is removed before we lock the 7618 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7619 // and we will exit safely 7620 io = AudioSystem::getOutputForEffect(&desc); 7621 } 7622 } 7623 7624 { 7625 Mutex::Autolock _l(mLock); 7626 7627 7628 if (!EffectIsNullUuid(&pDesc->uuid)) { 7629 // if uuid is specified, request effect descriptor 7630 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7631 if (lStatus < 0) { 7632 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7633 goto Exit; 7634 } 7635 } else { 7636 // if uuid is not specified, look for an available implementation 7637 // of the required type in effect factory 7638 if (EffectIsNullUuid(&pDesc->type)) { 7639 ALOGW("createEffect() no effect type"); 7640 lStatus = BAD_VALUE; 7641 goto Exit; 7642 } 7643 uint32_t numEffects = 0; 7644 effect_descriptor_t d; 7645 d.flags = 0; // prevent compiler warning 7646 bool found = false; 7647 7648 lStatus = EffectQueryNumberEffects(&numEffects); 7649 if (lStatus < 0) { 7650 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7651 goto Exit; 7652 } 7653 for (uint32_t i = 0; i < numEffects; i++) { 7654 lStatus = EffectQueryEffect(i, &desc); 7655 if (lStatus < 0) { 7656 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7657 continue; 7658 } 7659 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7660 // If matching type found save effect descriptor. If the session is 7661 // 0 and the effect is not auxiliary, continue enumeration in case 7662 // an auxiliary version of this effect type is available 7663 found = true; 7664 d = desc; 7665 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7666 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7667 break; 7668 } 7669 } 7670 } 7671 if (!found) { 7672 lStatus = BAD_VALUE; 7673 ALOGW("createEffect() effect not found"); 7674 goto Exit; 7675 } 7676 // For same effect type, chose auxiliary version over insert version if 7677 // connect to output mix (Compliance to OpenSL ES) 7678 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7679 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7680 desc = d; 7681 } 7682 } 7683 7684 // Do not allow auxiliary effects on a session different from 0 (output mix) 7685 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7686 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7687 lStatus = INVALID_OPERATION; 7688 goto Exit; 7689 } 7690 7691 // check recording permission for visualizer 7692 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7693 !recordingAllowed()) { 7694 lStatus = PERMISSION_DENIED; 7695 goto Exit; 7696 } 7697 7698 // return effect descriptor 7699 *pDesc = desc; 7700 7701 // If output is not specified try to find a matching audio session ID in one of the 7702 // output threads. 7703 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7704 // because of code checking output when entering the function. 7705 // Note: io is never 0 when creating an effect on an input 7706 if (io == 0) { 7707 // look for the thread where the specified audio session is present 7708 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7709 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7710 io = mPlaybackThreads.keyAt(i); 7711 break; 7712 } 7713 } 7714 if (io == 0) { 7715 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7716 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7717 io = mRecordThreads.keyAt(i); 7718 break; 7719 } 7720 } 7721 } 7722 // If no output thread contains the requested session ID, default to 7723 // first output. The effect chain will be moved to the correct output 7724 // thread when a track with the same session ID is created 7725 if (io == 0 && mPlaybackThreads.size()) { 7726 io = mPlaybackThreads.keyAt(0); 7727 } 7728 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7729 } 7730 ThreadBase *thread = checkRecordThread_l(io); 7731 if (thread == NULL) { 7732 thread = checkPlaybackThread_l(io); 7733 if (thread == NULL) { 7734 ALOGE("createEffect() unknown output thread"); 7735 lStatus = BAD_VALUE; 7736 goto Exit; 7737 } 7738 } 7739 7740 sp<Client> client = registerPid_l(pid); 7741 7742 // create effect on selected output thread 7743 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7744 &desc, enabled, &lStatus); 7745 if (handle != 0 && id != NULL) { 7746 *id = handle->id(); 7747 } 7748 } 7749 7750Exit: 7751 if (status != NULL) { 7752 *status = lStatus; 7753 } 7754 return handle; 7755} 7756 7757status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7758 audio_io_handle_t dstOutput) 7759{ 7760 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7761 sessionId, srcOutput, dstOutput); 7762 Mutex::Autolock _l(mLock); 7763 if (srcOutput == dstOutput) { 7764 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7765 return NO_ERROR; 7766 } 7767 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7768 if (srcThread == NULL) { 7769 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7770 return BAD_VALUE; 7771 } 7772 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7773 if (dstThread == NULL) { 7774 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7775 return BAD_VALUE; 7776 } 7777 7778 Mutex::Autolock _dl(dstThread->mLock); 7779 Mutex::Autolock _sl(srcThread->mLock); 7780 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7781 7782 return NO_ERROR; 7783} 7784 7785// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7786status_t AudioFlinger::moveEffectChain_l(int sessionId, 7787 AudioFlinger::PlaybackThread *srcThread, 7788 AudioFlinger::PlaybackThread *dstThread, 7789 bool reRegister) 7790{ 7791 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7792 sessionId, srcThread, dstThread); 7793 7794 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7795 if (chain == 0) { 7796 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7797 sessionId, srcThread); 7798 return INVALID_OPERATION; 7799 } 7800 7801 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7802 // so that a new chain is created with correct parameters when first effect is added. This is 7803 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7804 // removed. 7805 srcThread->removeEffectChain_l(chain); 7806 7807 // transfer all effects one by one so that new effect chain is created on new thread with 7808 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7809 audio_io_handle_t dstOutput = dstThread->id(); 7810 sp<EffectChain> dstChain; 7811 uint32_t strategy = 0; // prevent compiler warning 7812 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7813 while (effect != 0) { 7814 srcThread->removeEffect_l(effect); 7815 dstThread->addEffect_l(effect); 7816 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7817 if (effect->state() == EffectModule::ACTIVE || 7818 effect->state() == EffectModule::STOPPING) { 7819 effect->start(); 7820 } 7821 // if the move request is not received from audio policy manager, the effect must be 7822 // re-registered with the new strategy and output 7823 if (dstChain == 0) { 7824 dstChain = effect->chain().promote(); 7825 if (dstChain == 0) { 7826 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7827 srcThread->addEffect_l(effect); 7828 return NO_INIT; 7829 } 7830 strategy = dstChain->strategy(); 7831 } 7832 if (reRegister) { 7833 AudioSystem::unregisterEffect(effect->id()); 7834 AudioSystem::registerEffect(&effect->desc(), 7835 dstOutput, 7836 strategy, 7837 sessionId, 7838 effect->id()); 7839 } 7840 effect = chain->getEffectFromId_l(0); 7841 } 7842 7843 return NO_ERROR; 7844} 7845 7846 7847// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7848sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7849 const sp<AudioFlinger::Client>& client, 7850 const sp<IEffectClient>& effectClient, 7851 int32_t priority, 7852 int sessionId, 7853 effect_descriptor_t *desc, 7854 int *enabled, 7855 status_t *status 7856 ) 7857{ 7858 sp<EffectModule> effect; 7859 sp<EffectHandle> handle; 7860 status_t lStatus; 7861 sp<EffectChain> chain; 7862 bool chainCreated = false; 7863 bool effectCreated = false; 7864 bool effectRegistered = false; 7865 7866 lStatus = initCheck(); 7867 if (lStatus != NO_ERROR) { 7868 ALOGW("createEffect_l() Audio driver not initialized."); 7869 goto Exit; 7870 } 7871 7872 // Do not allow effects with session ID 0 on direct output or duplicating threads 7873 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7874 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7875 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7876 desc->name, sessionId); 7877 lStatus = BAD_VALUE; 7878 goto Exit; 7879 } 7880 // Only Pre processor effects are allowed on input threads and only on input threads 7881 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7882 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7883 desc->name, desc->flags, mType); 7884 lStatus = BAD_VALUE; 7885 goto Exit; 7886 } 7887 7888 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7889 7890 { // scope for mLock 7891 Mutex::Autolock _l(mLock); 7892 7893 // check for existing effect chain with the requested audio session 7894 chain = getEffectChain_l(sessionId); 7895 if (chain == 0) { 7896 // create a new chain for this session 7897 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7898 chain = new EffectChain(this, sessionId); 7899 addEffectChain_l(chain); 7900 chain->setStrategy(getStrategyForSession_l(sessionId)); 7901 chainCreated = true; 7902 } else { 7903 effect = chain->getEffectFromDesc_l(desc); 7904 } 7905 7906 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7907 7908 if (effect == 0) { 7909 int id = mAudioFlinger->nextUniqueId(); 7910 // Check CPU and memory usage 7911 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7912 if (lStatus != NO_ERROR) { 7913 goto Exit; 7914 } 7915 effectRegistered = true; 7916 // create a new effect module if none present in the chain 7917 effect = new EffectModule(this, chain, desc, id, sessionId); 7918 lStatus = effect->status(); 7919 if (lStatus != NO_ERROR) { 7920 goto Exit; 7921 } 7922 lStatus = chain->addEffect_l(effect); 7923 if (lStatus != NO_ERROR) { 7924 goto Exit; 7925 } 7926 effectCreated = true; 7927 7928 effect->setDevice(mOutDevice); 7929 effect->setDevice(mInDevice); 7930 effect->setMode(mAudioFlinger->getMode()); 7931 effect->setAudioSource(mAudioSource); 7932 } 7933 // create effect handle and connect it to effect module 7934 handle = new EffectHandle(effect, client, effectClient, priority); 7935 lStatus = effect->addHandle(handle.get()); 7936 if (enabled != NULL) { 7937 *enabled = (int)effect->isEnabled(); 7938 } 7939 } 7940 7941Exit: 7942 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7943 Mutex::Autolock _l(mLock); 7944 if (effectCreated) { 7945 chain->removeEffect_l(effect); 7946 } 7947 if (effectRegistered) { 7948 AudioSystem::unregisterEffect(effect->id()); 7949 } 7950 if (chainCreated) { 7951 removeEffectChain_l(chain); 7952 } 7953 handle.clear(); 7954 } 7955 7956 if (status != NULL) { 7957 *status = lStatus; 7958 } 7959 return handle; 7960} 7961 7962sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7963{ 7964 Mutex::Autolock _l(mLock); 7965 return getEffect_l(sessionId, effectId); 7966} 7967 7968sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7969{ 7970 sp<EffectChain> chain = getEffectChain_l(sessionId); 7971 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7972} 7973 7974// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7975// PlaybackThread::mLock held 7976status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7977{ 7978 // check for existing effect chain with the requested audio session 7979 int sessionId = effect->sessionId(); 7980 sp<EffectChain> chain = getEffectChain_l(sessionId); 7981 bool chainCreated = false; 7982 7983 if (chain == 0) { 7984 // create a new chain for this session 7985 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7986 chain = new EffectChain(this, sessionId); 7987 addEffectChain_l(chain); 7988 chain->setStrategy(getStrategyForSession_l(sessionId)); 7989 chainCreated = true; 7990 } 7991 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7992 7993 if (chain->getEffectFromId_l(effect->id()) != 0) { 7994 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7995 this, effect->desc().name, chain.get()); 7996 return BAD_VALUE; 7997 } 7998 7999 status_t status = chain->addEffect_l(effect); 8000 if (status != NO_ERROR) { 8001 if (chainCreated) { 8002 removeEffectChain_l(chain); 8003 } 8004 return status; 8005 } 8006 8007 effect->setDevice(mOutDevice); 8008 effect->setDevice(mInDevice); 8009 effect->setMode(mAudioFlinger->getMode()); 8010 effect->setAudioSource(mAudioSource); 8011 return NO_ERROR; 8012} 8013 8014void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 8015 8016 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 8017 effect_descriptor_t desc = effect->desc(); 8018 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8019 detachAuxEffect_l(effect->id()); 8020 } 8021 8022 sp<EffectChain> chain = effect->chain().promote(); 8023 if (chain != 0) { 8024 // remove effect chain if removing last effect 8025 if (chain->removeEffect_l(effect) == 0) { 8026 removeEffectChain_l(chain); 8027 } 8028 } else { 8029 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 8030 } 8031} 8032 8033void AudioFlinger::ThreadBase::lockEffectChains_l( 8034 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8035{ 8036 effectChains = mEffectChains; 8037 for (size_t i = 0; i < mEffectChains.size(); i++) { 8038 mEffectChains[i]->lock(); 8039 } 8040} 8041 8042void AudioFlinger::ThreadBase::unlockEffectChains( 8043 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 8044{ 8045 for (size_t i = 0; i < effectChains.size(); i++) { 8046 effectChains[i]->unlock(); 8047 } 8048} 8049 8050sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 8051{ 8052 Mutex::Autolock _l(mLock); 8053 return getEffectChain_l(sessionId); 8054} 8055 8056sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 8057{ 8058 size_t size = mEffectChains.size(); 8059 for (size_t i = 0; i < size; i++) { 8060 if (mEffectChains[i]->sessionId() == sessionId) { 8061 return mEffectChains[i]; 8062 } 8063 } 8064 return 0; 8065} 8066 8067void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 8068{ 8069 Mutex::Autolock _l(mLock); 8070 size_t size = mEffectChains.size(); 8071 for (size_t i = 0; i < size; i++) { 8072 mEffectChains[i]->setMode_l(mode); 8073 } 8074} 8075 8076void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 8077 EffectHandle *handle, 8078 bool unpinIfLast) { 8079 8080 Mutex::Autolock _l(mLock); 8081 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 8082 // delete the effect module if removing last handle on it 8083 if (effect->removeHandle(handle) == 0) { 8084 if (!effect->isPinned() || unpinIfLast) { 8085 removeEffect_l(effect); 8086 AudioSystem::unregisterEffect(effect->id()); 8087 } 8088 } 8089} 8090 8091status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 8092{ 8093 int session = chain->sessionId(); 8094 int16_t *buffer = mMixBuffer; 8095 bool ownsBuffer = false; 8096 8097 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8098 if (session > 0) { 8099 // Only one effect chain can be present in direct output thread and it uses 8100 // the mix buffer as input 8101 if (mType != DIRECT) { 8102 size_t numSamples = mNormalFrameCount * mChannelCount; 8103 buffer = new int16_t[numSamples]; 8104 memset(buffer, 0, numSamples * sizeof(int16_t)); 8105 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 8106 ownsBuffer = true; 8107 } 8108 8109 // Attach all tracks with same session ID to this chain. 8110 for (size_t i = 0; i < mTracks.size(); ++i) { 8111 sp<Track> track = mTracks[i]; 8112 if (session == track->sessionId()) { 8113 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 8114 buffer); 8115 track->setMainBuffer(buffer); 8116 chain->incTrackCnt(); 8117 } 8118 } 8119 8120 // indicate all active tracks in the chain 8121 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8122 sp<Track> track = mActiveTracks[i].promote(); 8123 if (track == 0) continue; 8124 if (session == track->sessionId()) { 8125 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 8126 chain->incActiveTrackCnt(); 8127 } 8128 } 8129 } 8130 8131 chain->setInBuffer(buffer, ownsBuffer); 8132 chain->setOutBuffer(mMixBuffer); 8133 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 8134 // chains list in order to be processed last as it contains output stage effects 8135 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 8136 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 8137 // after track specific effects and before output stage 8138 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 8139 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 8140 // Effect chain for other sessions are inserted at beginning of effect 8141 // chains list to be processed before output mix effects. Relative order between other 8142 // sessions is not important 8143 size_t size = mEffectChains.size(); 8144 size_t i = 0; 8145 for (i = 0; i < size; i++) { 8146 if (mEffectChains[i]->sessionId() < session) break; 8147 } 8148 mEffectChains.insertAt(chain, i); 8149 checkSuspendOnAddEffectChain_l(chain); 8150 8151 return NO_ERROR; 8152} 8153 8154size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 8155{ 8156 int session = chain->sessionId(); 8157 8158 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8159 8160 for (size_t i = 0; i < mEffectChains.size(); i++) { 8161 if (chain == mEffectChains[i]) { 8162 mEffectChains.removeAt(i); 8163 // detach all active tracks from the chain 8164 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 8165 sp<Track> track = mActiveTracks[i].promote(); 8166 if (track == 0) continue; 8167 if (session == track->sessionId()) { 8168 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 8169 chain.get(), session); 8170 chain->decActiveTrackCnt(); 8171 } 8172 } 8173 8174 // detach all tracks with same session ID from this chain 8175 for (size_t i = 0; i < mTracks.size(); ++i) { 8176 sp<Track> track = mTracks[i]; 8177 if (session == track->sessionId()) { 8178 track->setMainBuffer(mMixBuffer); 8179 chain->decTrackCnt(); 8180 } 8181 } 8182 break; 8183 } 8184 } 8185 return mEffectChains.size(); 8186} 8187 8188status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8189 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8190{ 8191 Mutex::Autolock _l(mLock); 8192 return attachAuxEffect_l(track, EffectId); 8193} 8194 8195status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8196 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8197{ 8198 status_t status = NO_ERROR; 8199 8200 if (EffectId == 0) { 8201 track->setAuxBuffer(0, NULL); 8202 } else { 8203 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8204 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8205 if (effect != 0) { 8206 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8207 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8208 } else { 8209 status = INVALID_OPERATION; 8210 } 8211 } else { 8212 status = BAD_VALUE; 8213 } 8214 } 8215 return status; 8216} 8217 8218void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8219{ 8220 for (size_t i = 0; i < mTracks.size(); ++i) { 8221 sp<Track> track = mTracks[i]; 8222 if (track->auxEffectId() == effectId) { 8223 attachAuxEffect_l(track, 0); 8224 } 8225 } 8226} 8227 8228status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8229{ 8230 // only one chain per input thread 8231 if (mEffectChains.size() != 0) { 8232 return INVALID_OPERATION; 8233 } 8234 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8235 8236 chain->setInBuffer(NULL); 8237 chain->setOutBuffer(NULL); 8238 8239 checkSuspendOnAddEffectChain_l(chain); 8240 8241 mEffectChains.add(chain); 8242 8243 return NO_ERROR; 8244} 8245 8246size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8247{ 8248 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8249 ALOGW_IF(mEffectChains.size() != 1, 8250 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8251 chain.get(), mEffectChains.size(), this); 8252 if (mEffectChains.size() == 1) { 8253 mEffectChains.removeAt(0); 8254 } 8255 return 0; 8256} 8257 8258// ---------------------------------------------------------------------------- 8259// EffectModule implementation 8260// ---------------------------------------------------------------------------- 8261 8262#undef LOG_TAG 8263#define LOG_TAG "AudioFlinger::EffectModule" 8264 8265AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8266 const wp<AudioFlinger::EffectChain>& chain, 8267 effect_descriptor_t *desc, 8268 int id, 8269 int sessionId) 8270 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8271 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8272 mDescriptor(*desc), 8273 // mConfig is set by configure() and not used before then 8274 mEffectInterface(NULL), 8275 mStatus(NO_INIT), mState(IDLE), 8276 // mMaxDisableWaitCnt is set by configure() and not used before then 8277 // mDisableWaitCnt is set by process() and updateState() and not used before then 8278 mSuspended(false) 8279{ 8280 ALOGV("Constructor %p", this); 8281 int lStatus; 8282 8283 // create effect engine from effect factory 8284 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8285 8286 if (mStatus != NO_ERROR) { 8287 return; 8288 } 8289 lStatus = init(); 8290 if (lStatus < 0) { 8291 mStatus = lStatus; 8292 goto Error; 8293 } 8294 8295 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8296 return; 8297Error: 8298 EffectRelease(mEffectInterface); 8299 mEffectInterface = NULL; 8300 ALOGV("Constructor Error %d", mStatus); 8301} 8302 8303AudioFlinger::EffectModule::~EffectModule() 8304{ 8305 ALOGV("Destructor %p", this); 8306 if (mEffectInterface != NULL) { 8307 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8308 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8309 sp<ThreadBase> thread = mThread.promote(); 8310 if (thread != 0) { 8311 audio_stream_t *stream = thread->stream(); 8312 if (stream != NULL) { 8313 stream->remove_audio_effect(stream, mEffectInterface); 8314 } 8315 } 8316 } 8317 // release effect engine 8318 EffectRelease(mEffectInterface); 8319 } 8320} 8321 8322status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8323{ 8324 status_t status; 8325 8326 Mutex::Autolock _l(mLock); 8327 int priority = handle->priority(); 8328 size_t size = mHandles.size(); 8329 EffectHandle *controlHandle = NULL; 8330 size_t i; 8331 for (i = 0; i < size; i++) { 8332 EffectHandle *h = mHandles[i]; 8333 if (h == NULL || h->destroyed_l()) continue; 8334 // first non destroyed handle is considered in control 8335 if (controlHandle == NULL) 8336 controlHandle = h; 8337 if (h->priority() <= priority) break; 8338 } 8339 // if inserted in first place, move effect control from previous owner to this handle 8340 if (i == 0) { 8341 bool enabled = false; 8342 if (controlHandle != NULL) { 8343 enabled = controlHandle->enabled(); 8344 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8345 } 8346 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8347 status = NO_ERROR; 8348 } else { 8349 status = ALREADY_EXISTS; 8350 } 8351 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8352 mHandles.insertAt(handle, i); 8353 return status; 8354} 8355 8356size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8357{ 8358 Mutex::Autolock _l(mLock); 8359 size_t size = mHandles.size(); 8360 size_t i; 8361 for (i = 0; i < size; i++) { 8362 if (mHandles[i] == handle) break; 8363 } 8364 if (i == size) { 8365 return size; 8366 } 8367 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8368 8369 mHandles.removeAt(i); 8370 // if removed from first place, move effect control from this handle to next in line 8371 if (i == 0) { 8372 EffectHandle *h = controlHandle_l(); 8373 if (h != NULL) { 8374 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8375 } 8376 } 8377 8378 // Prevent calls to process() and other functions on effect interface from now on. 8379 // The effect engine will be released by the destructor when the last strong reference on 8380 // this object is released which can happen after next process is called. 8381 if (mHandles.size() == 0 && !mPinned) { 8382 mState = DESTROYED; 8383 } 8384 8385 return mHandles.size(); 8386} 8387 8388// must be called with EffectModule::mLock held 8389AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8390{ 8391 // the first valid handle in the list has control over the module 8392 for (size_t i = 0; i < mHandles.size(); i++) { 8393 EffectHandle *h = mHandles[i]; 8394 if (h != NULL && !h->destroyed_l()) { 8395 return h; 8396 } 8397 } 8398 8399 return NULL; 8400} 8401 8402size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8403{ 8404 ALOGV("disconnect() %p handle %p", this, handle); 8405 // keep a strong reference on this EffectModule to avoid calling the 8406 // destructor before we exit 8407 sp<EffectModule> keep(this); 8408 { 8409 sp<ThreadBase> thread = mThread.promote(); 8410 if (thread != 0) { 8411 thread->disconnectEffect(keep, handle, unpinIfLast); 8412 } 8413 } 8414 return mHandles.size(); 8415} 8416 8417void AudioFlinger::EffectModule::updateState() { 8418 Mutex::Autolock _l(mLock); 8419 8420 switch (mState) { 8421 case RESTART: 8422 reset_l(); 8423 // FALL THROUGH 8424 8425 case STARTING: 8426 // clear auxiliary effect input buffer for next accumulation 8427 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8428 memset(mConfig.inputCfg.buffer.raw, 8429 0, 8430 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8431 } 8432 start_l(); 8433 mState = ACTIVE; 8434 break; 8435 case STOPPING: 8436 stop_l(); 8437 mDisableWaitCnt = mMaxDisableWaitCnt; 8438 mState = STOPPED; 8439 break; 8440 case STOPPED: 8441 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8442 // turn off sequence. 8443 if (--mDisableWaitCnt == 0) { 8444 reset_l(); 8445 mState = IDLE; 8446 } 8447 break; 8448 default: //IDLE , ACTIVE, DESTROYED 8449 break; 8450 } 8451} 8452 8453void AudioFlinger::EffectModule::process() 8454{ 8455 Mutex::Autolock _l(mLock); 8456 8457 if (mState == DESTROYED || mEffectInterface == NULL || 8458 mConfig.inputCfg.buffer.raw == NULL || 8459 mConfig.outputCfg.buffer.raw == NULL) { 8460 return; 8461 } 8462 8463 if (isProcessEnabled()) { 8464 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8465 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8466 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8467 mConfig.inputCfg.buffer.s32, 8468 mConfig.inputCfg.buffer.frameCount/2); 8469 } 8470 8471 // do the actual processing in the effect engine 8472 int ret = (*mEffectInterface)->process(mEffectInterface, 8473 &mConfig.inputCfg.buffer, 8474 &mConfig.outputCfg.buffer); 8475 8476 // force transition to IDLE state when engine is ready 8477 if (mState == STOPPED && ret == -ENODATA) { 8478 mDisableWaitCnt = 1; 8479 } 8480 8481 // clear auxiliary effect input buffer for next accumulation 8482 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8483 memset(mConfig.inputCfg.buffer.raw, 0, 8484 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8485 } 8486 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8487 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8488 // If an insert effect is idle and input buffer is different from output buffer, 8489 // accumulate input onto output 8490 sp<EffectChain> chain = mChain.promote(); 8491 if (chain != 0 && chain->activeTrackCnt() != 0) { 8492 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8493 int16_t *in = mConfig.inputCfg.buffer.s16; 8494 int16_t *out = mConfig.outputCfg.buffer.s16; 8495 for (size_t i = 0; i < frameCnt; i++) { 8496 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8497 } 8498 } 8499 } 8500} 8501 8502void AudioFlinger::EffectModule::reset_l() 8503{ 8504 if (mEffectInterface == NULL) { 8505 return; 8506 } 8507 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8508} 8509 8510status_t AudioFlinger::EffectModule::configure() 8511{ 8512 if (mEffectInterface == NULL) { 8513 return NO_INIT; 8514 } 8515 8516 sp<ThreadBase> thread = mThread.promote(); 8517 if (thread == 0) { 8518 return DEAD_OBJECT; 8519 } 8520 8521 // TODO: handle configuration of effects replacing track process 8522 audio_channel_mask_t channelMask = thread->channelMask(); 8523 8524 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8525 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8526 } else { 8527 mConfig.inputCfg.channels = channelMask; 8528 } 8529 mConfig.outputCfg.channels = channelMask; 8530 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8531 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8532 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8533 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8534 mConfig.inputCfg.bufferProvider.cookie = NULL; 8535 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8536 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8537 mConfig.outputCfg.bufferProvider.cookie = NULL; 8538 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8539 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8540 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8541 // Insert effect: 8542 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8543 // always overwrites output buffer: input buffer == output buffer 8544 // - in other sessions: 8545 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8546 // other effect: overwrites output buffer: input buffer == output buffer 8547 // Auxiliary effect: 8548 // accumulates in output buffer: input buffer != output buffer 8549 // Therefore: accumulate <=> input buffer != output buffer 8550 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8551 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8552 } else { 8553 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8554 } 8555 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8556 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8557 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8558 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8559 8560 ALOGV("configure() %p thread %p buffer %p framecount %d", 8561 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8562 8563 status_t cmdStatus; 8564 uint32_t size = sizeof(int); 8565 status_t status = (*mEffectInterface)->command(mEffectInterface, 8566 EFFECT_CMD_SET_CONFIG, 8567 sizeof(effect_config_t), 8568 &mConfig, 8569 &size, 8570 &cmdStatus); 8571 if (status == 0) { 8572 status = cmdStatus; 8573 } 8574 8575 if (status == 0 && 8576 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8577 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8578 effect_param_t *p = (effect_param_t *)buf32; 8579 8580 p->psize = sizeof(uint32_t); 8581 p->vsize = sizeof(uint32_t); 8582 size = sizeof(int); 8583 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8584 8585 uint32_t latency = 0; 8586 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8587 if (pbt != NULL) { 8588 latency = pbt->latency_l(); 8589 } 8590 8591 *((int32_t *)p->data + 1)= latency; 8592 (*mEffectInterface)->command(mEffectInterface, 8593 EFFECT_CMD_SET_PARAM, 8594 sizeof(effect_param_t) + 8, 8595 &buf32, 8596 &size, 8597 &cmdStatus); 8598 } 8599 8600 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8601 (1000 * mConfig.outputCfg.buffer.frameCount); 8602 8603 return status; 8604} 8605 8606status_t AudioFlinger::EffectModule::init() 8607{ 8608 Mutex::Autolock _l(mLock); 8609 if (mEffectInterface == NULL) { 8610 return NO_INIT; 8611 } 8612 status_t cmdStatus; 8613 uint32_t size = sizeof(status_t); 8614 status_t status = (*mEffectInterface)->command(mEffectInterface, 8615 EFFECT_CMD_INIT, 8616 0, 8617 NULL, 8618 &size, 8619 &cmdStatus); 8620 if (status == 0) { 8621 status = cmdStatus; 8622 } 8623 return status; 8624} 8625 8626status_t AudioFlinger::EffectModule::start() 8627{ 8628 Mutex::Autolock _l(mLock); 8629 return start_l(); 8630} 8631 8632status_t AudioFlinger::EffectModule::start_l() 8633{ 8634 if (mEffectInterface == NULL) { 8635 return NO_INIT; 8636 } 8637 status_t cmdStatus; 8638 uint32_t size = sizeof(status_t); 8639 status_t status = (*mEffectInterface)->command(mEffectInterface, 8640 EFFECT_CMD_ENABLE, 8641 0, 8642 NULL, 8643 &size, 8644 &cmdStatus); 8645 if (status == 0) { 8646 status = cmdStatus; 8647 } 8648 if (status == 0 && 8649 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8650 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8651 sp<ThreadBase> thread = mThread.promote(); 8652 if (thread != 0) { 8653 audio_stream_t *stream = thread->stream(); 8654 if (stream != NULL) { 8655 stream->add_audio_effect(stream, mEffectInterface); 8656 } 8657 } 8658 } 8659 return status; 8660} 8661 8662status_t AudioFlinger::EffectModule::stop() 8663{ 8664 Mutex::Autolock _l(mLock); 8665 return stop_l(); 8666} 8667 8668status_t AudioFlinger::EffectModule::stop_l() 8669{ 8670 if (mEffectInterface == NULL) { 8671 return NO_INIT; 8672 } 8673 status_t cmdStatus; 8674 uint32_t size = sizeof(status_t); 8675 status_t status = (*mEffectInterface)->command(mEffectInterface, 8676 EFFECT_CMD_DISABLE, 8677 0, 8678 NULL, 8679 &size, 8680 &cmdStatus); 8681 if (status == 0) { 8682 status = cmdStatus; 8683 } 8684 if (status == 0 && 8685 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8686 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8687 sp<ThreadBase> thread = mThread.promote(); 8688 if (thread != 0) { 8689 audio_stream_t *stream = thread->stream(); 8690 if (stream != NULL) { 8691 stream->remove_audio_effect(stream, mEffectInterface); 8692 } 8693 } 8694 } 8695 return status; 8696} 8697 8698status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8699 uint32_t cmdSize, 8700 void *pCmdData, 8701 uint32_t *replySize, 8702 void *pReplyData) 8703{ 8704 Mutex::Autolock _l(mLock); 8705 ALOGVV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8706 8707 if (mState == DESTROYED || mEffectInterface == NULL) { 8708 return NO_INIT; 8709 } 8710 status_t status = (*mEffectInterface)->command(mEffectInterface, 8711 cmdCode, 8712 cmdSize, 8713 pCmdData, 8714 replySize, 8715 pReplyData); 8716 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8717 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8718 for (size_t i = 1; i < mHandles.size(); i++) { 8719 EffectHandle *h = mHandles[i]; 8720 if (h != NULL && !h->destroyed_l()) { 8721 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8722 } 8723 } 8724 } 8725 return status; 8726} 8727 8728status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8729{ 8730 Mutex::Autolock _l(mLock); 8731 return setEnabled_l(enabled); 8732} 8733 8734// must be called with EffectModule::mLock held 8735status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8736{ 8737 8738 ALOGV("setEnabled %p enabled %d", this, enabled); 8739 8740 if (enabled != isEnabled()) { 8741 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8742 if (enabled && status != NO_ERROR) { 8743 return status; 8744 } 8745 8746 switch (mState) { 8747 // going from disabled to enabled 8748 case IDLE: 8749 mState = STARTING; 8750 break; 8751 case STOPPED: 8752 mState = RESTART; 8753 break; 8754 case STOPPING: 8755 mState = ACTIVE; 8756 break; 8757 8758 // going from enabled to disabled 8759 case RESTART: 8760 mState = STOPPED; 8761 break; 8762 case STARTING: 8763 mState = IDLE; 8764 break; 8765 case ACTIVE: 8766 mState = STOPPING; 8767 break; 8768 case DESTROYED: 8769 return NO_ERROR; // simply ignore as we are being destroyed 8770 } 8771 for (size_t i = 1; i < mHandles.size(); i++) { 8772 EffectHandle *h = mHandles[i]; 8773 if (h != NULL && !h->destroyed_l()) { 8774 h->setEnabled(enabled); 8775 } 8776 } 8777 } 8778 return NO_ERROR; 8779} 8780 8781bool AudioFlinger::EffectModule::isEnabled() const 8782{ 8783 switch (mState) { 8784 case RESTART: 8785 case STARTING: 8786 case ACTIVE: 8787 return true; 8788 case IDLE: 8789 case STOPPING: 8790 case STOPPED: 8791 case DESTROYED: 8792 default: 8793 return false; 8794 } 8795} 8796 8797bool AudioFlinger::EffectModule::isProcessEnabled() const 8798{ 8799 switch (mState) { 8800 case RESTART: 8801 case ACTIVE: 8802 case STOPPING: 8803 case STOPPED: 8804 return true; 8805 case IDLE: 8806 case STARTING: 8807 case DESTROYED: 8808 default: 8809 return false; 8810 } 8811} 8812 8813status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8814{ 8815 Mutex::Autolock _l(mLock); 8816 status_t status = NO_ERROR; 8817 8818 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8819 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8820 if (isProcessEnabled() && 8821 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8822 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8823 status_t cmdStatus; 8824 uint32_t volume[2]; 8825 uint32_t *pVolume = NULL; 8826 uint32_t size = sizeof(volume); 8827 volume[0] = *left; 8828 volume[1] = *right; 8829 if (controller) { 8830 pVolume = volume; 8831 } 8832 status = (*mEffectInterface)->command(mEffectInterface, 8833 EFFECT_CMD_SET_VOLUME, 8834 size, 8835 volume, 8836 &size, 8837 pVolume); 8838 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8839 *left = volume[0]; 8840 *right = volume[1]; 8841 } 8842 } 8843 return status; 8844} 8845 8846status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8847{ 8848 if (device == AUDIO_DEVICE_NONE) { 8849 return NO_ERROR; 8850 } 8851 8852 Mutex::Autolock _l(mLock); 8853 status_t status = NO_ERROR; 8854 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8855 status_t cmdStatus; 8856 uint32_t size = sizeof(status_t); 8857 uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE : 8858 EFFECT_CMD_SET_INPUT_DEVICE; 8859 status = (*mEffectInterface)->command(mEffectInterface, 8860 cmd, 8861 sizeof(uint32_t), 8862 &device, 8863 &size, 8864 &cmdStatus); 8865 } 8866 return status; 8867} 8868 8869status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8870{ 8871 Mutex::Autolock _l(mLock); 8872 status_t status = NO_ERROR; 8873 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8874 status_t cmdStatus; 8875 uint32_t size = sizeof(status_t); 8876 status = (*mEffectInterface)->command(mEffectInterface, 8877 EFFECT_CMD_SET_AUDIO_MODE, 8878 sizeof(audio_mode_t), 8879 &mode, 8880 &size, 8881 &cmdStatus); 8882 if (status == NO_ERROR) { 8883 status = cmdStatus; 8884 } 8885 } 8886 return status; 8887} 8888 8889status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8890{ 8891 Mutex::Autolock _l(mLock); 8892 status_t status = NO_ERROR; 8893 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8894 uint32_t size = 0; 8895 status = (*mEffectInterface)->command(mEffectInterface, 8896 EFFECT_CMD_SET_AUDIO_SOURCE, 8897 sizeof(audio_source_t), 8898 &source, 8899 &size, 8900 NULL); 8901 } 8902 return status; 8903} 8904 8905void AudioFlinger::EffectModule::setSuspended(bool suspended) 8906{ 8907 Mutex::Autolock _l(mLock); 8908 mSuspended = suspended; 8909} 8910 8911bool AudioFlinger::EffectModule::suspended() const 8912{ 8913 Mutex::Autolock _l(mLock); 8914 return mSuspended; 8915} 8916 8917bool AudioFlinger::EffectModule::purgeHandles() 8918{ 8919 bool enabled = false; 8920 Mutex::Autolock _l(mLock); 8921 for (size_t i = 0; i < mHandles.size(); i++) { 8922 EffectHandle *handle = mHandles[i]; 8923 if (handle != NULL && !handle->destroyed_l()) { 8924 handle->effect().clear(); 8925 if (handle->hasControl()) { 8926 enabled = handle->enabled(); 8927 } 8928 } 8929 } 8930 return enabled; 8931} 8932 8933void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8934{ 8935 const size_t SIZE = 256; 8936 char buffer[SIZE]; 8937 String8 result; 8938 8939 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8940 result.append(buffer); 8941 8942 bool locked = tryLock(mLock); 8943 // failed to lock - AudioFlinger is probably deadlocked 8944 if (!locked) { 8945 result.append("\t\tCould not lock Fx mutex:\n"); 8946 } 8947 8948 result.append("\t\tSession Status State Engine:\n"); 8949 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8950 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8951 result.append(buffer); 8952 8953 result.append("\t\tDescriptor:\n"); 8954 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8955 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8956 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1], 8957 mDescriptor.uuid.node[2], 8958 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8959 result.append(buffer); 8960 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8961 mDescriptor.type.timeLow, mDescriptor.type.timeMid, 8962 mDescriptor.type.timeHiAndVersion, 8963 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1], 8964 mDescriptor.type.node[2], 8965 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8966 result.append(buffer); 8967 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8968 mDescriptor.apiVersion, 8969 mDescriptor.flags); 8970 result.append(buffer); 8971 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8972 mDescriptor.name); 8973 result.append(buffer); 8974 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8975 mDescriptor.implementor); 8976 result.append(buffer); 8977 8978 result.append("\t\t- Input configuration:\n"); 8979 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8980 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8981 (uint32_t)mConfig.inputCfg.buffer.raw, 8982 mConfig.inputCfg.buffer.frameCount, 8983 mConfig.inputCfg.samplingRate, 8984 mConfig.inputCfg.channels, 8985 mConfig.inputCfg.format); 8986 result.append(buffer); 8987 8988 result.append("\t\t- Output configuration:\n"); 8989 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8990 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8991 (uint32_t)mConfig.outputCfg.buffer.raw, 8992 mConfig.outputCfg.buffer.frameCount, 8993 mConfig.outputCfg.samplingRate, 8994 mConfig.outputCfg.channels, 8995 mConfig.outputCfg.format); 8996 result.append(buffer); 8997 8998 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8999 result.append(buffer); 9000 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 9001 for (size_t i = 0; i < mHandles.size(); ++i) { 9002 EffectHandle *handle = mHandles[i]; 9003 if (handle != NULL && !handle->destroyed_l()) { 9004 handle->dump(buffer, SIZE); 9005 result.append(buffer); 9006 } 9007 } 9008 9009 result.append("\n"); 9010 9011 write(fd, result.string(), result.length()); 9012 9013 if (locked) { 9014 mLock.unlock(); 9015 } 9016} 9017 9018// ---------------------------------------------------------------------------- 9019// EffectHandle implementation 9020// ---------------------------------------------------------------------------- 9021 9022#undef LOG_TAG 9023#define LOG_TAG "AudioFlinger::EffectHandle" 9024 9025AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 9026 const sp<AudioFlinger::Client>& client, 9027 const sp<IEffectClient>& effectClient, 9028 int32_t priority) 9029 : BnEffect(), 9030 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 9031 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 9032{ 9033 ALOGV("constructor %p", this); 9034 9035 if (client == 0) { 9036 return; 9037 } 9038 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 9039 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 9040 if (mCblkMemory != 0) { 9041 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 9042 9043 if (mCblk != NULL) { 9044 new(mCblk) effect_param_cblk_t(); 9045 mBuffer = (uint8_t *)mCblk + bufOffset; 9046 } 9047 } else { 9048 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + 9049 sizeof(effect_param_cblk_t)); 9050 return; 9051 } 9052} 9053 9054AudioFlinger::EffectHandle::~EffectHandle() 9055{ 9056 ALOGV("Destructor %p", this); 9057 9058 if (mEffect == 0) { 9059 mDestroyed = true; 9060 return; 9061 } 9062 mEffect->lock(); 9063 mDestroyed = true; 9064 mEffect->unlock(); 9065 disconnect(false); 9066} 9067 9068status_t AudioFlinger::EffectHandle::enable() 9069{ 9070 ALOGV("enable %p", this); 9071 if (!mHasControl) return INVALID_OPERATION; 9072 if (mEffect == 0) return DEAD_OBJECT; 9073 9074 if (mEnabled) { 9075 return NO_ERROR; 9076 } 9077 9078 mEnabled = true; 9079 9080 sp<ThreadBase> thread = mEffect->thread().promote(); 9081 if (thread != 0) { 9082 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 9083 } 9084 9085 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 9086 if (mEffect->suspended()) { 9087 return NO_ERROR; 9088 } 9089 9090 status_t status = mEffect->setEnabled(true); 9091 if (status != NO_ERROR) { 9092 if (thread != 0) { 9093 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9094 } 9095 mEnabled = false; 9096 } 9097 return status; 9098} 9099 9100status_t AudioFlinger::EffectHandle::disable() 9101{ 9102 ALOGV("disable %p", this); 9103 if (!mHasControl) return INVALID_OPERATION; 9104 if (mEffect == 0) return DEAD_OBJECT; 9105 9106 if (!mEnabled) { 9107 return NO_ERROR; 9108 } 9109 mEnabled = false; 9110 9111 if (mEffect->suspended()) { 9112 return NO_ERROR; 9113 } 9114 9115 status_t status = mEffect->setEnabled(false); 9116 9117 sp<ThreadBase> thread = mEffect->thread().promote(); 9118 if (thread != 0) { 9119 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9120 } 9121 9122 return status; 9123} 9124 9125void AudioFlinger::EffectHandle::disconnect() 9126{ 9127 disconnect(true); 9128} 9129 9130void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 9131{ 9132 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 9133 if (mEffect == 0) { 9134 return; 9135 } 9136 // restore suspended effects if the disconnected handle was enabled and the last one. 9137 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 9138 sp<ThreadBase> thread = mEffect->thread().promote(); 9139 if (thread != 0) { 9140 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 9141 } 9142 } 9143 9144 // release sp on module => module destructor can be called now 9145 mEffect.clear(); 9146 if (mClient != 0) { 9147 if (mCblk != NULL) { 9148 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 9149 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 9150 } 9151 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 9152 // Client destructor must run with AudioFlinger mutex locked 9153 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 9154 mClient.clear(); 9155 } 9156} 9157 9158status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 9159 uint32_t cmdSize, 9160 void *pCmdData, 9161 uint32_t *replySize, 9162 void *pReplyData) 9163{ 9164 ALOGVV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 9165 cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 9166 9167 // only get parameter command is permitted for applications not controlling the effect 9168 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9169 return INVALID_OPERATION; 9170 } 9171 if (mEffect == 0) return DEAD_OBJECT; 9172 if (mClient == 0) return INVALID_OPERATION; 9173 9174 // handle commands that are not forwarded transparently to effect engine 9175 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9176 // No need to trylock() here as this function is executed in the binder thread serving a 9177 // particular client process: no risk to block the whole media server process or mixer 9178 // threads if we are stuck here 9179 Mutex::Autolock _l(mCblk->lock); 9180 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9181 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9182 mCblk->serverIndex = 0; 9183 mCblk->clientIndex = 0; 9184 return BAD_VALUE; 9185 } 9186 status_t status = NO_ERROR; 9187 while (mCblk->serverIndex < mCblk->clientIndex) { 9188 int reply; 9189 uint32_t rsize = sizeof(int); 9190 int *p = (int *)(mBuffer + mCblk->serverIndex); 9191 int size = *p++; 9192 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9193 ALOGW("command(): invalid parameter block size"); 9194 break; 9195 } 9196 effect_param_t *param = (effect_param_t *)p; 9197 if (param->psize == 0 || param->vsize == 0) { 9198 ALOGW("command(): null parameter or value size"); 9199 mCblk->serverIndex += size; 9200 continue; 9201 } 9202 uint32_t psize = sizeof(effect_param_t) + 9203 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9204 param->vsize; 9205 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9206 psize, 9207 p, 9208 &rsize, 9209 &reply); 9210 // stop at first error encountered 9211 if (ret != NO_ERROR) { 9212 status = ret; 9213 *(int *)pReplyData = reply; 9214 break; 9215 } else if (reply != NO_ERROR) { 9216 *(int *)pReplyData = reply; 9217 break; 9218 } 9219 mCblk->serverIndex += size; 9220 } 9221 mCblk->serverIndex = 0; 9222 mCblk->clientIndex = 0; 9223 return status; 9224 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9225 *(int *)pReplyData = NO_ERROR; 9226 return enable(); 9227 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9228 *(int *)pReplyData = NO_ERROR; 9229 return disable(); 9230 } 9231 9232 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9233} 9234 9235void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9236{ 9237 ALOGV("setControl %p control %d", this, hasControl); 9238 9239 mHasControl = hasControl; 9240 mEnabled = enabled; 9241 9242 if (signal && mEffectClient != 0) { 9243 mEffectClient->controlStatusChanged(hasControl); 9244 } 9245} 9246 9247void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9248 uint32_t cmdSize, 9249 void *pCmdData, 9250 uint32_t replySize, 9251 void *pReplyData) 9252{ 9253 if (mEffectClient != 0) { 9254 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9255 } 9256} 9257 9258 9259 9260void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9261{ 9262 if (mEffectClient != 0) { 9263 mEffectClient->enableStatusChanged(enabled); 9264 } 9265} 9266 9267status_t AudioFlinger::EffectHandle::onTransact( 9268 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9269{ 9270 return BnEffect::onTransact(code, data, reply, flags); 9271} 9272 9273 9274void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9275{ 9276 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9277 9278 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9279 (mClient == 0) ? getpid_cached : mClient->pid(), 9280 mPriority, 9281 mHasControl, 9282 !locked, 9283 mCblk ? mCblk->clientIndex : 0, 9284 mCblk ? mCblk->serverIndex : 0 9285 ); 9286 9287 if (locked) { 9288 mCblk->lock.unlock(); 9289 } 9290} 9291 9292#undef LOG_TAG 9293#define LOG_TAG "AudioFlinger::EffectChain" 9294 9295AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9296 int sessionId) 9297 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9298 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9299 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9300{ 9301 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9302 if (thread == NULL) { 9303 return; 9304 } 9305 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9306 thread->frameCount(); 9307} 9308 9309AudioFlinger::EffectChain::~EffectChain() 9310{ 9311 if (mOwnInBuffer) { 9312 delete mInBuffer; 9313 } 9314 9315} 9316 9317// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9318sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l( 9319 effect_descriptor_t *descriptor) 9320{ 9321 size_t size = mEffects.size(); 9322 9323 for (size_t i = 0; i < size; i++) { 9324 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9325 return mEffects[i]; 9326 } 9327 } 9328 return 0; 9329} 9330 9331// getEffectFromId_l() must be called with ThreadBase::mLock held 9332sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9333{ 9334 size_t size = mEffects.size(); 9335 9336 for (size_t i = 0; i < size; i++) { 9337 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9338 if (id == 0 || mEffects[i]->id() == id) { 9339 return mEffects[i]; 9340 } 9341 } 9342 return 0; 9343} 9344 9345// getEffectFromType_l() must be called with ThreadBase::mLock held 9346sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9347 const effect_uuid_t *type) 9348{ 9349 size_t size = mEffects.size(); 9350 9351 for (size_t i = 0; i < size; i++) { 9352 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9353 return mEffects[i]; 9354 } 9355 } 9356 return 0; 9357} 9358 9359void AudioFlinger::EffectChain::clearInputBuffer() 9360{ 9361 Mutex::Autolock _l(mLock); 9362 sp<ThreadBase> thread = mThread.promote(); 9363 if (thread == 0) { 9364 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9365 return; 9366 } 9367 clearInputBuffer_l(thread); 9368} 9369 9370// Must be called with EffectChain::mLock locked 9371void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9372{ 9373 size_t numSamples = thread->frameCount() * thread->channelCount(); 9374 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9375 9376} 9377 9378// Must be called with EffectChain::mLock locked 9379void AudioFlinger::EffectChain::process_l() 9380{ 9381 sp<ThreadBase> thread = mThread.promote(); 9382 if (thread == 0) { 9383 ALOGW("process_l(): cannot promote mixer thread"); 9384 return; 9385 } 9386 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9387 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9388 // always process effects unless no more tracks are on the session and the effect tail 9389 // has been rendered 9390 bool doProcess = true; 9391 if (!isGlobalSession) { 9392 bool tracksOnSession = (trackCnt() != 0); 9393 9394 if (!tracksOnSession && mTailBufferCount == 0) { 9395 doProcess = false; 9396 } 9397 9398 if (activeTrackCnt() == 0) { 9399 // if no track is active and the effect tail has not been rendered, 9400 // the input buffer must be cleared here as the mixer process will not do it 9401 if (tracksOnSession || mTailBufferCount > 0) { 9402 clearInputBuffer_l(thread); 9403 if (mTailBufferCount > 0) { 9404 mTailBufferCount--; 9405 } 9406 } 9407 } 9408 } 9409 9410 size_t size = mEffects.size(); 9411 if (doProcess) { 9412 for (size_t i = 0; i < size; i++) { 9413 mEffects[i]->process(); 9414 } 9415 } 9416 for (size_t i = 0; i < size; i++) { 9417 mEffects[i]->updateState(); 9418 } 9419} 9420 9421// addEffect_l() must be called with PlaybackThread::mLock held 9422status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9423{ 9424 effect_descriptor_t desc = effect->desc(); 9425 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9426 9427 Mutex::Autolock _l(mLock); 9428 effect->setChain(this); 9429 sp<ThreadBase> thread = mThread.promote(); 9430 if (thread == 0) { 9431 return NO_INIT; 9432 } 9433 effect->setThread(thread); 9434 9435 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9436 // Auxiliary effects are inserted at the beginning of mEffects vector as 9437 // they are processed first and accumulated in chain input buffer 9438 mEffects.insertAt(effect, 0); 9439 9440 // the input buffer for auxiliary effect contains mono samples in 9441 // 32 bit format. This is to avoid saturation in AudoMixer 9442 // accumulation stage. Saturation is done in EffectModule::process() before 9443 // calling the process in effect engine 9444 size_t numSamples = thread->frameCount(); 9445 int32_t *buffer = new int32_t[numSamples]; 9446 memset(buffer, 0, numSamples * sizeof(int32_t)); 9447 effect->setInBuffer((int16_t *)buffer); 9448 // auxiliary effects output samples to chain input buffer for further processing 9449 // by insert effects 9450 effect->setOutBuffer(mInBuffer); 9451 } else { 9452 // Insert effects are inserted at the end of mEffects vector as they are processed 9453 // after track and auxiliary effects. 9454 // Insert effect order as a function of indicated preference: 9455 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9456 // another effect is present 9457 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9458 // last effect claiming first position 9459 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9460 // first effect claiming last position 9461 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9462 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9463 // already present 9464 9465 size_t size = mEffects.size(); 9466 size_t idx_insert = size; 9467 ssize_t idx_insert_first = -1; 9468 ssize_t idx_insert_last = -1; 9469 9470 for (size_t i = 0; i < size; i++) { 9471 effect_descriptor_t d = mEffects[i]->desc(); 9472 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9473 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9474 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9475 // check invalid effect chaining combinations 9476 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9477 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9478 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", 9479 desc.name, d.name); 9480 return INVALID_OPERATION; 9481 } 9482 // remember position of first insert effect and by default 9483 // select this as insert position for new effect 9484 if (idx_insert == size) { 9485 idx_insert = i; 9486 } 9487 // remember position of last insert effect claiming 9488 // first position 9489 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9490 idx_insert_first = i; 9491 } 9492 // remember position of first insert effect claiming 9493 // last position 9494 if (iPref == EFFECT_FLAG_INSERT_LAST && 9495 idx_insert_last == -1) { 9496 idx_insert_last = i; 9497 } 9498 } 9499 } 9500 9501 // modify idx_insert from first position if needed 9502 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9503 if (idx_insert_last != -1) { 9504 idx_insert = idx_insert_last; 9505 } else { 9506 idx_insert = size; 9507 } 9508 } else { 9509 if (idx_insert_first != -1) { 9510 idx_insert = idx_insert_first + 1; 9511 } 9512 } 9513 9514 // always read samples from chain input buffer 9515 effect->setInBuffer(mInBuffer); 9516 9517 // if last effect in the chain, output samples to chain 9518 // output buffer, otherwise to chain input buffer 9519 if (idx_insert == size) { 9520 if (idx_insert != 0) { 9521 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9522 mEffects[idx_insert-1]->configure(); 9523 } 9524 effect->setOutBuffer(mOutBuffer); 9525 } else { 9526 effect->setOutBuffer(mInBuffer); 9527 } 9528 mEffects.insertAt(effect, idx_insert); 9529 9530 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, 9531 idx_insert); 9532 } 9533 effect->configure(); 9534 return NO_ERROR; 9535} 9536 9537// removeEffect_l() must be called with PlaybackThread::mLock held 9538size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9539{ 9540 Mutex::Autolock _l(mLock); 9541 size_t size = mEffects.size(); 9542 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9543 9544 for (size_t i = 0; i < size; i++) { 9545 if (effect == mEffects[i]) { 9546 // calling stop here will remove pre-processing effect from the audio HAL. 9547 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9548 // the middle of a read from audio HAL 9549 if (mEffects[i]->state() == EffectModule::ACTIVE || 9550 mEffects[i]->state() == EffectModule::STOPPING) { 9551 mEffects[i]->stop(); 9552 } 9553 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9554 delete[] effect->inBuffer(); 9555 } else { 9556 if (i == size - 1 && i != 0) { 9557 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9558 mEffects[i - 1]->configure(); 9559 } 9560 } 9561 mEffects.removeAt(i); 9562 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), 9563 this, i); 9564 break; 9565 } 9566 } 9567 9568 return mEffects.size(); 9569} 9570 9571// setDevice_l() must be called with PlaybackThread::mLock held 9572void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9573{ 9574 size_t size = mEffects.size(); 9575 for (size_t i = 0; i < size; i++) { 9576 mEffects[i]->setDevice(device); 9577 } 9578} 9579 9580// setMode_l() must be called with PlaybackThread::mLock held 9581void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9582{ 9583 size_t size = mEffects.size(); 9584 for (size_t i = 0; i < size; i++) { 9585 mEffects[i]->setMode(mode); 9586 } 9587} 9588 9589// setAudioSource_l() must be called with PlaybackThread::mLock held 9590void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9591{ 9592 size_t size = mEffects.size(); 9593 for (size_t i = 0; i < size; i++) { 9594 mEffects[i]->setAudioSource(source); 9595 } 9596} 9597 9598// setVolume_l() must be called with PlaybackThread::mLock held 9599bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9600{ 9601 uint32_t newLeft = *left; 9602 uint32_t newRight = *right; 9603 bool hasControl = false; 9604 int ctrlIdx = -1; 9605 size_t size = mEffects.size(); 9606 9607 // first update volume controller 9608 for (size_t i = size; i > 0; i--) { 9609 if (mEffects[i - 1]->isProcessEnabled() && 9610 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9611 ctrlIdx = i - 1; 9612 hasControl = true; 9613 break; 9614 } 9615 } 9616 9617 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9618 if (hasControl) { 9619 *left = mNewLeftVolume; 9620 *right = mNewRightVolume; 9621 } 9622 return hasControl; 9623 } 9624 9625 mVolumeCtrlIdx = ctrlIdx; 9626 mLeftVolume = newLeft; 9627 mRightVolume = newRight; 9628 9629 // second get volume update from volume controller 9630 if (ctrlIdx >= 0) { 9631 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9632 mNewLeftVolume = newLeft; 9633 mNewRightVolume = newRight; 9634 } 9635 // then indicate volume to all other effects in chain. 9636 // Pass altered volume to effects before volume controller 9637 // and requested volume to effects after controller 9638 uint32_t lVol = newLeft; 9639 uint32_t rVol = newRight; 9640 9641 for (size_t i = 0; i < size; i++) { 9642 if ((int)i == ctrlIdx) continue; 9643 // this also works for ctrlIdx == -1 when there is no volume controller 9644 if ((int)i > ctrlIdx) { 9645 lVol = *left; 9646 rVol = *right; 9647 } 9648 mEffects[i]->setVolume(&lVol, &rVol, false); 9649 } 9650 *left = newLeft; 9651 *right = newRight; 9652 9653 return hasControl; 9654} 9655 9656void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9657{ 9658 const size_t SIZE = 256; 9659 char buffer[SIZE]; 9660 String8 result; 9661 9662 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9663 result.append(buffer); 9664 9665 bool locked = tryLock(mLock); 9666 // failed to lock - AudioFlinger is probably deadlocked 9667 if (!locked) { 9668 result.append("\tCould not lock mutex:\n"); 9669 } 9670 9671 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9672 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9673 mEffects.size(), 9674 (uint32_t)mInBuffer, 9675 (uint32_t)mOutBuffer, 9676 mActiveTrackCnt); 9677 result.append(buffer); 9678 write(fd, result.string(), result.size()); 9679 9680 for (size_t i = 0; i < mEffects.size(); ++i) { 9681 sp<EffectModule> effect = mEffects[i]; 9682 if (effect != 0) { 9683 effect->dump(fd, args); 9684 } 9685 } 9686 9687 if (locked) { 9688 mLock.unlock(); 9689 } 9690} 9691 9692// must be called with ThreadBase::mLock held 9693void AudioFlinger::EffectChain::setEffectSuspended_l( 9694 const effect_uuid_t *type, bool suspend) 9695{ 9696 sp<SuspendedEffectDesc> desc; 9697 // use effect type UUID timelow as key as there is no real risk of identical 9698 // timeLow fields among effect type UUIDs. 9699 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9700 if (suspend) { 9701 if (index >= 0) { 9702 desc = mSuspendedEffects.valueAt(index); 9703 } else { 9704 desc = new SuspendedEffectDesc(); 9705 desc->mType = *type; 9706 mSuspendedEffects.add(type->timeLow, desc); 9707 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9708 } 9709 if (desc->mRefCount++ == 0) { 9710 sp<EffectModule> effect = getEffectIfEnabled(type); 9711 if (effect != 0) { 9712 desc->mEffect = effect; 9713 effect->setSuspended(true); 9714 effect->setEnabled(false); 9715 } 9716 } 9717 } else { 9718 if (index < 0) { 9719 return; 9720 } 9721 desc = mSuspendedEffects.valueAt(index); 9722 if (desc->mRefCount <= 0) { 9723 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9724 desc->mRefCount = 1; 9725 } 9726 if (--desc->mRefCount == 0) { 9727 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9728 if (desc->mEffect != 0) { 9729 sp<EffectModule> effect = desc->mEffect.promote(); 9730 if (effect != 0) { 9731 effect->setSuspended(false); 9732 effect->lock(); 9733 EffectHandle *handle = effect->controlHandle_l(); 9734 if (handle != NULL && !handle->destroyed_l()) { 9735 effect->setEnabled_l(handle->enabled()); 9736 } 9737 effect->unlock(); 9738 } 9739 desc->mEffect.clear(); 9740 } 9741 mSuspendedEffects.removeItemsAt(index); 9742 } 9743 } 9744} 9745 9746// must be called with ThreadBase::mLock held 9747void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9748{ 9749 sp<SuspendedEffectDesc> desc; 9750 9751 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9752 if (suspend) { 9753 if (index >= 0) { 9754 desc = mSuspendedEffects.valueAt(index); 9755 } else { 9756 desc = new SuspendedEffectDesc(); 9757 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9758 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9759 } 9760 if (desc->mRefCount++ == 0) { 9761 Vector< sp<EffectModule> > effects; 9762 getSuspendEligibleEffects(effects); 9763 for (size_t i = 0; i < effects.size(); i++) { 9764 setEffectSuspended_l(&effects[i]->desc().type, true); 9765 } 9766 } 9767 } else { 9768 if (index < 0) { 9769 return; 9770 } 9771 desc = mSuspendedEffects.valueAt(index); 9772 if (desc->mRefCount <= 0) { 9773 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9774 desc->mRefCount = 1; 9775 } 9776 if (--desc->mRefCount == 0) { 9777 Vector<const effect_uuid_t *> types; 9778 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9779 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9780 continue; 9781 } 9782 types.add(&mSuspendedEffects.valueAt(i)->mType); 9783 } 9784 for (size_t i = 0; i < types.size(); i++) { 9785 setEffectSuspended_l(types[i], false); 9786 } 9787 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", 9788 mSuspendedEffects.keyAt(index)); 9789 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9790 } 9791 } 9792} 9793 9794 9795// The volume effect is used for automated tests only 9796#ifndef OPENSL_ES_H_ 9797static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9798 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9799const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9800#endif //OPENSL_ES_H_ 9801 9802bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9803{ 9804 // auxiliary effects and visualizer are never suspended on output mix 9805 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9806 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9807 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9808 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9809 return false; 9810 } 9811 return true; 9812} 9813 9814void AudioFlinger::EffectChain::getSuspendEligibleEffects( 9815 Vector< sp<AudioFlinger::EffectModule> > &effects) 9816{ 9817 effects.clear(); 9818 for (size_t i = 0; i < mEffects.size(); i++) { 9819 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9820 effects.add(mEffects[i]); 9821 } 9822 } 9823} 9824 9825sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9826 const effect_uuid_t *type) 9827{ 9828 sp<EffectModule> effect = getEffectFromType_l(type); 9829 return effect != 0 && effect->isEnabled() ? effect : 0; 9830} 9831 9832void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9833 bool enabled) 9834{ 9835 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9836 if (enabled) { 9837 if (index < 0) { 9838 // if the effect is not suspend check if all effects are suspended 9839 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9840 if (index < 0) { 9841 return; 9842 } 9843 if (!isEffectEligibleForSuspend(effect->desc())) { 9844 return; 9845 } 9846 setEffectSuspended_l(&effect->desc().type, enabled); 9847 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9848 if (index < 0) { 9849 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9850 return; 9851 } 9852 } 9853 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9854 effect->desc().type.timeLow); 9855 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9856 // if effect is requested to suspended but was not yet enabled, supend it now. 9857 if (desc->mEffect == 0) { 9858 desc->mEffect = effect; 9859 effect->setEnabled(false); 9860 effect->setSuspended(true); 9861 } 9862 } else { 9863 if (index < 0) { 9864 return; 9865 } 9866 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9867 effect->desc().type.timeLow); 9868 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9869 desc->mEffect.clear(); 9870 effect->setSuspended(false); 9871 } 9872} 9873 9874#undef LOG_TAG 9875#define LOG_TAG "AudioFlinger" 9876 9877// ---------------------------------------------------------------------------- 9878 9879status_t AudioFlinger::onTransact( 9880 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9881{ 9882 return BnAudioFlinger::onTransact(code, data, reply, flags); 9883} 9884 9885}; // namespace android 9886