AudioFlinger.cpp revision 57b2dd1e78af53115985f18d31ec5421c9da947e
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 } else { 295 // check a match for the requested module handle 296 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 297 if (audioHwDevice != NULL) { 298 return audioHwDevice; 299 } 300 } 301 // then try to find a module supporting the requested device. 302 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 303 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 304 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 305 if ((dev->get_supported_devices(dev) & devices) == devices) 306 return audioHwDevice; 307 } 308 309 return NULL; 310} 311 312void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 313{ 314 const size_t SIZE = 256; 315 char buffer[SIZE]; 316 String8 result; 317 318 result.append("Clients:\n"); 319 for (size_t i = 0; i < mClients.size(); ++i) { 320 sp<Client> client = mClients.valueAt(i).promote(); 321 if (client != 0) { 322 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 323 result.append(buffer); 324 } 325 } 326 327 result.append("Global session refs:\n"); 328 result.append(" session pid count\n"); 329 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 330 AudioSessionRef *r = mAudioSessionRefs[i]; 331 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 332 result.append(buffer); 333 } 334 write(fd, result.string(), result.size()); 335} 336 337 338void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 339{ 340 const size_t SIZE = 256; 341 char buffer[SIZE]; 342 String8 result; 343 hardware_call_state hardwareStatus = mHardwareStatus; 344 345 snprintf(buffer, SIZE, "Hardware status: %d\n" 346 "Standby Time mSec: %u\n", 347 hardwareStatus, 348 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351} 352 353void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 354{ 355 const size_t SIZE = 256; 356 char buffer[SIZE]; 357 String8 result; 358 snprintf(buffer, SIZE, "Permission Denial: " 359 "can't dump AudioFlinger from pid=%d, uid=%d\n", 360 IPCThreadState::self()->getCallingPid(), 361 IPCThreadState::self()->getCallingUid()); 362 result.append(buffer); 363 write(fd, result.string(), result.size()); 364} 365 366static bool tryLock(Mutex& mutex) 367{ 368 bool locked = false; 369 for (int i = 0; i < kDumpLockRetries; ++i) { 370 if (mutex.tryLock() == NO_ERROR) { 371 locked = true; 372 break; 373 } 374 usleep(kDumpLockSleepUs); 375 } 376 return locked; 377} 378 379status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 380{ 381 if (!dumpAllowed()) { 382 dumpPermissionDenial(fd, args); 383 } else { 384 // get state of hardware lock 385 bool hardwareLocked = tryLock(mHardwareLock); 386 if (!hardwareLocked) { 387 String8 result(kHardwareLockedString); 388 write(fd, result.string(), result.size()); 389 } else { 390 mHardwareLock.unlock(); 391 } 392 393 bool locked = tryLock(mLock); 394 395 // failed to lock - AudioFlinger is probably deadlocked 396 if (!locked) { 397 String8 result(kDeadlockedString); 398 write(fd, result.string(), result.size()); 399 } 400 401 dumpClients(fd, args); 402 dumpInternals(fd, args); 403 404 // dump playback threads 405 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 406 mPlaybackThreads.valueAt(i)->dump(fd, args); 407 } 408 409 // dump record threads 410 for (size_t i = 0; i < mRecordThreads.size(); i++) { 411 mRecordThreads.valueAt(i)->dump(fd, args); 412 } 413 414 // dump all hardware devs 415 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 416 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 417 dev->dump(dev, fd); 418 } 419 if (locked) mLock.unlock(); 420 } 421 return NO_ERROR; 422} 423 424sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 425{ 426 // If pid is already in the mClients wp<> map, then use that entry 427 // (for which promote() is always != 0), otherwise create a new entry and Client. 428 sp<Client> client = mClients.valueFor(pid).promote(); 429 if (client == 0) { 430 client = new Client(this, pid); 431 mClients.add(pid, client); 432 } 433 434 return client; 435} 436 437// IAudioFlinger interface 438 439 440sp<IAudioTrack> AudioFlinger::createTrack( 441 pid_t pid, 442 audio_stream_type_t streamType, 443 uint32_t sampleRate, 444 audio_format_t format, 445 audio_channel_mask_t channelMask, 446 int frameCount, 447 IAudioFlinger::track_flags_t flags, 448 const sp<IMemory>& sharedBuffer, 449 audio_io_handle_t output, 450 pid_t tid, 451 int *sessionId, 452 status_t *status) 453{ 454 sp<PlaybackThread::Track> track; 455 sp<TrackHandle> trackHandle; 456 sp<Client> client; 457 status_t lStatus; 458 int lSessionId; 459 460 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 461 // but if someone uses binder directly they could bypass that and cause us to crash 462 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 463 ALOGE("createTrack() invalid stream type %d", streamType); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 { 469 Mutex::Autolock _l(mLock); 470 PlaybackThread *thread = checkPlaybackThread_l(output); 471 PlaybackThread *effectThread = NULL; 472 if (thread == NULL) { 473 ALOGE("unknown output thread"); 474 lStatus = BAD_VALUE; 475 goto Exit; 476 } 477 478 client = registerPid_l(pid); 479 480 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 481 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 482 // check if an effect chain with the same session ID is present on another 483 // output thread and move it here. 484 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 485 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 486 if (mPlaybackThreads.keyAt(i) != output) { 487 uint32_t sessions = t->hasAudioSession(*sessionId); 488 if (sessions & PlaybackThread::EFFECT_SESSION) { 489 effectThread = t.get(); 490 break; 491 } 492 } 493 } 494 lSessionId = *sessionId; 495 } else { 496 // if no audio session id is provided, create one here 497 lSessionId = nextUniqueId(); 498 if (sessionId != NULL) { 499 *sessionId = lSessionId; 500 } 501 } 502 ALOGV("createTrack() lSessionId: %d", lSessionId); 503 504 track = thread->createTrack_l(client, streamType, sampleRate, format, 505 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 506 507 // move effect chain to this output thread if an effect on same session was waiting 508 // for a track to be created 509 if (lStatus == NO_ERROR && effectThread != NULL) { 510 Mutex::Autolock _dl(thread->mLock); 511 Mutex::Autolock _sl(effectThread->mLock); 512 moveEffectChain_l(lSessionId, effectThread, thread, true); 513 } 514 515 // Look for sync events awaiting for a session to be used. 516 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 517 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 518 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 519 if (lStatus == NO_ERROR) { 520 (void) track->setSyncEvent(mPendingSyncEvents[i]); 521 } else { 522 mPendingSyncEvents[i]->cancel(); 523 } 524 mPendingSyncEvents.removeAt(i); 525 i--; 526 } 527 } 528 } 529 } 530 if (lStatus == NO_ERROR) { 531 trackHandle = new TrackHandle(track); 532 } else { 533 // remove local strong reference to Client before deleting the Track so that the Client 534 // destructor is called by the TrackBase destructor with mLock held 535 client.clear(); 536 track.clear(); 537 } 538 539Exit: 540 if (status != NULL) { 541 *status = lStatus; 542 } 543 return trackHandle; 544} 545 546uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("sampleRate() unknown thread %d", output); 552 return 0; 553 } 554 return thread->sampleRate(); 555} 556 557int AudioFlinger::channelCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("channelCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->channelCount(); 566} 567 568audio_format_t AudioFlinger::format(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("format() unknown thread %d", output); 574 return AUDIO_FORMAT_INVALID; 575 } 576 return thread->format(); 577} 578 579size_t AudioFlinger::frameCount(audio_io_handle_t output) const 580{ 581 Mutex::Autolock _l(mLock); 582 PlaybackThread *thread = checkPlaybackThread_l(output); 583 if (thread == NULL) { 584 ALOGW("frameCount() unknown thread %d", output); 585 return 0; 586 } 587 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 588 // should examine all callers and fix them to handle smaller counts 589 return thread->frameCount(); 590} 591 592uint32_t AudioFlinger::latency(audio_io_handle_t output) const 593{ 594 Mutex::Autolock _l(mLock); 595 PlaybackThread *thread = checkPlaybackThread_l(output); 596 if (thread == NULL) { 597 ALOGW("latency() unknown thread %d", output); 598 return 0; 599 } 600 return thread->latency(); 601} 602 603status_t AudioFlinger::setMasterVolume(float value) 604{ 605 status_t ret = initCheck(); 606 if (ret != NO_ERROR) { 607 return ret; 608 } 609 610 // check calling permissions 611 if (!settingsAllowed()) { 612 return PERMISSION_DENIED; 613 } 614 615 Mutex::Autolock _l(mLock); 616 mMasterVolume = value; 617 618 // Set master volume in the HALs which support it. 619 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 620 AutoMutex lock(mHardwareLock); 621 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 622 623 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 624 if (dev->canSetMasterVolume()) { 625 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 626 } 627 mHardwareStatus = AUDIO_HW_IDLE; 628 } 629 630 // Now set the master volume in each playback thread. Playback threads 631 // assigned to HALs which do not have master volume support will apply 632 // master volume during the mix operation. Threads with HALs which do 633 // support master volume will simply ignore the setting. 634 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 635 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 636 637 return NO_ERROR; 638} 639 640status_t AudioFlinger::setMode(audio_mode_t mode) 641{ 642 status_t ret = initCheck(); 643 if (ret != NO_ERROR) { 644 return ret; 645 } 646 647 // check calling permissions 648 if (!settingsAllowed()) { 649 return PERMISSION_DENIED; 650 } 651 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 652 ALOGW("Illegal value: setMode(%d)", mode); 653 return BAD_VALUE; 654 } 655 656 { // scope for the lock 657 AutoMutex lock(mHardwareLock); 658 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 659 mHardwareStatus = AUDIO_HW_SET_MODE; 660 ret = dev->set_mode(dev, mode); 661 mHardwareStatus = AUDIO_HW_IDLE; 662 } 663 664 if (NO_ERROR == ret) { 665 Mutex::Autolock _l(mLock); 666 mMode = mode; 667 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 668 mPlaybackThreads.valueAt(i)->setMode(mode); 669 } 670 671 return ret; 672} 673 674status_t AudioFlinger::setMicMute(bool state) 675{ 676 status_t ret = initCheck(); 677 if (ret != NO_ERROR) { 678 return ret; 679 } 680 681 // check calling permissions 682 if (!settingsAllowed()) { 683 return PERMISSION_DENIED; 684 } 685 686 AutoMutex lock(mHardwareLock); 687 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 688 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 689 ret = dev->set_mic_mute(dev, state); 690 mHardwareStatus = AUDIO_HW_IDLE; 691 return ret; 692} 693 694bool AudioFlinger::getMicMute() const 695{ 696 status_t ret = initCheck(); 697 if (ret != NO_ERROR) { 698 return false; 699 } 700 701 bool state = AUDIO_MODE_INVALID; 702 AutoMutex lock(mHardwareLock); 703 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 704 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 705 dev->get_mic_mute(dev, &state); 706 mHardwareStatus = AUDIO_HW_IDLE; 707 return state; 708} 709 710status_t AudioFlinger::setMasterMute(bool muted) 711{ 712 status_t ret = initCheck(); 713 if (ret != NO_ERROR) { 714 return ret; 715 } 716 717 // check calling permissions 718 if (!settingsAllowed()) { 719 return PERMISSION_DENIED; 720 } 721 722 Mutex::Autolock _l(mLock); 723 mMasterMute = muted; 724 725 // Set master mute in the HALs which support it. 726 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 727 AutoMutex lock(mHardwareLock); 728 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 729 730 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 731 if (dev->canSetMasterMute()) { 732 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 733 } 734 mHardwareStatus = AUDIO_HW_IDLE; 735 } 736 737 // Now set the master mute in each playback thread. Playback threads 738 // assigned to HALs which do not have master mute support will apply master 739 // mute during the mix operation. Threads with HALs which do support master 740 // mute will simply ignore the setting. 741 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 742 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 743 744 return NO_ERROR; 745} 746 747float AudioFlinger::masterVolume() const 748{ 749 Mutex::Autolock _l(mLock); 750 return masterVolume_l(); 751} 752 753bool AudioFlinger::masterMute() const 754{ 755 Mutex::Autolock _l(mLock); 756 return masterMute_l(); 757} 758 759float AudioFlinger::masterVolume_l() const 760{ 761 return mMasterVolume; 762} 763 764bool AudioFlinger::masterMute_l() const 765{ 766 return mMasterMute; 767} 768 769status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 770 audio_io_handle_t output) 771{ 772 // check calling permissions 773 if (!settingsAllowed()) { 774 return PERMISSION_DENIED; 775 } 776 777 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 778 ALOGE("setStreamVolume() invalid stream %d", stream); 779 return BAD_VALUE; 780 } 781 782 AutoMutex lock(mLock); 783 PlaybackThread *thread = NULL; 784 if (output) { 785 thread = checkPlaybackThread_l(output); 786 if (thread == NULL) { 787 return BAD_VALUE; 788 } 789 } 790 791 mStreamTypes[stream].volume = value; 792 793 if (thread == NULL) { 794 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 795 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 796 } 797 } else { 798 thread->setStreamVolume(stream, value); 799 } 800 801 return NO_ERROR; 802} 803 804status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 805{ 806 // check calling permissions 807 if (!settingsAllowed()) { 808 return PERMISSION_DENIED; 809 } 810 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 812 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 813 ALOGE("setStreamMute() invalid stream %d", stream); 814 return BAD_VALUE; 815 } 816 817 AutoMutex lock(mLock); 818 mStreamTypes[stream].mute = muted; 819 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 820 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 821 822 return NO_ERROR; 823} 824 825float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 826{ 827 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 828 return 0.0f; 829 } 830 831 AutoMutex lock(mLock); 832 float volume; 833 if (output) { 834 PlaybackThread *thread = checkPlaybackThread_l(output); 835 if (thread == NULL) { 836 return 0.0f; 837 } 838 volume = thread->streamVolume(stream); 839 } else { 840 volume = streamVolume_l(stream); 841 } 842 843 return volume; 844} 845 846bool AudioFlinger::streamMute(audio_stream_type_t stream) const 847{ 848 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 849 return true; 850 } 851 852 AutoMutex lock(mLock); 853 return streamMute_l(stream); 854} 855 856status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 857{ 858 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 859 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 860 // check calling permissions 861 if (!settingsAllowed()) { 862 return PERMISSION_DENIED; 863 } 864 865 // ioHandle == 0 means the parameters are global to the audio hardware interface 866 if (ioHandle == 0) { 867 Mutex::Autolock _l(mLock); 868 status_t final_result = NO_ERROR; 869 { 870 AutoMutex lock(mHardwareLock); 871 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 872 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 873 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 874 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 875 final_result = result ?: final_result; 876 } 877 mHardwareStatus = AUDIO_HW_IDLE; 878 } 879 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 880 AudioParameter param = AudioParameter(keyValuePairs); 881 String8 value; 882 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 883 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 884 if (mBtNrecIsOff != btNrecIsOff) { 885 for (size_t i = 0; i < mRecordThreads.size(); i++) { 886 sp<RecordThread> thread = mRecordThreads.valueAt(i); 887 audio_devices_t device = thread->device() & AUDIO_DEVICE_IN_ALL; 888 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 889 // collect all of the thread's session IDs 890 KeyedVector<int, bool> ids = thread->sessionIds(); 891 // suspend effects associated with those session IDs 892 for (size_t j = 0; j < ids.size(); ++j) { 893 int sessionId = ids.keyAt(j); 894 thread->setEffectSuspended(FX_IID_AEC, 895 suspend, 896 sessionId); 897 thread->setEffectSuspended(FX_IID_NS, 898 suspend, 899 sessionId); 900 } 901 } 902 mBtNrecIsOff = btNrecIsOff; 903 } 904 } 905 String8 screenState; 906 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 907 bool isOff = screenState == "off"; 908 if (isOff != (gScreenState & 1)) { 909 gScreenState = ((gScreenState & ~1) + 2) | isOff; 910 } 911 } 912 return final_result; 913 } 914 915 // hold a strong ref on thread in case closeOutput() or closeInput() is called 916 // and the thread is exited once the lock is released 917 sp<ThreadBase> thread; 918 { 919 Mutex::Autolock _l(mLock); 920 thread = checkPlaybackThread_l(ioHandle); 921 if (thread == 0) { 922 thread = checkRecordThread_l(ioHandle); 923 } else if (thread == primaryPlaybackThread_l()) { 924 // indicate output device change to all input threads for pre processing 925 AudioParameter param = AudioParameter(keyValuePairs); 926 int value; 927 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 928 (value != 0)) { 929 for (size_t i = 0; i < mRecordThreads.size(); i++) { 930 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 931 } 932 } 933 } 934 } 935 if (thread != 0) { 936 return thread->setParameters(keyValuePairs); 937 } 938 return BAD_VALUE; 939} 940 941String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 942{ 943// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 944// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 945 946 Mutex::Autolock _l(mLock); 947 948 if (ioHandle == 0) { 949 String8 out_s8; 950 951 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 952 char *s; 953 { 954 AutoMutex lock(mHardwareLock); 955 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 956 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 957 s = dev->get_parameters(dev, keys.string()); 958 mHardwareStatus = AUDIO_HW_IDLE; 959 } 960 out_s8 += String8(s ? s : ""); 961 free(s); 962 } 963 return out_s8; 964 } 965 966 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 967 if (playbackThread != NULL) { 968 return playbackThread->getParameters(keys); 969 } 970 RecordThread *recordThread = checkRecordThread_l(ioHandle); 971 if (recordThread != NULL) { 972 return recordThread->getParameters(keys); 973 } 974 return String8(""); 975} 976 977size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 978 audio_channel_mask_t channelMask) const 979{ 980 status_t ret = initCheck(); 981 if (ret != NO_ERROR) { 982 return 0; 983 } 984 985 AutoMutex lock(mHardwareLock); 986 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 987 struct audio_config config = { 988 sample_rate: sampleRate, 989 channel_mask: channelMask, 990 format: format, 991 }; 992 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 993 size_t size = dev->get_input_buffer_size(dev, &config); 994 mHardwareStatus = AUDIO_HW_IDLE; 995 return size; 996} 997 998unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 999{ 1000 Mutex::Autolock _l(mLock); 1001 1002 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1003 if (recordThread != NULL) { 1004 return recordThread->getInputFramesLost(); 1005 } 1006 return 0; 1007} 1008 1009status_t AudioFlinger::setVoiceVolume(float value) 1010{ 1011 status_t ret = initCheck(); 1012 if (ret != NO_ERROR) { 1013 return ret; 1014 } 1015 1016 // check calling permissions 1017 if (!settingsAllowed()) { 1018 return PERMISSION_DENIED; 1019 } 1020 1021 AutoMutex lock(mHardwareLock); 1022 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1023 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1024 ret = dev->set_voice_volume(dev, value); 1025 mHardwareStatus = AUDIO_HW_IDLE; 1026 1027 return ret; 1028} 1029 1030status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1031 audio_io_handle_t output) const 1032{ 1033 status_t status; 1034 1035 Mutex::Autolock _l(mLock); 1036 1037 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1038 if (playbackThread != NULL) { 1039 return playbackThread->getRenderPosition(halFrames, dspFrames); 1040 } 1041 1042 return BAD_VALUE; 1043} 1044 1045void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1046{ 1047 1048 Mutex::Autolock _l(mLock); 1049 1050 pid_t pid = IPCThreadState::self()->getCallingPid(); 1051 if (mNotificationClients.indexOfKey(pid) < 0) { 1052 sp<NotificationClient> notificationClient = new NotificationClient(this, 1053 client, 1054 pid); 1055 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1056 1057 mNotificationClients.add(pid, notificationClient); 1058 1059 sp<IBinder> binder = client->asBinder(); 1060 binder->linkToDeath(notificationClient); 1061 1062 // the config change is always sent from playback or record threads to avoid deadlock 1063 // with AudioSystem::gLock 1064 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1065 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1066 } 1067 1068 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1069 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1070 } 1071 } 1072} 1073 1074void AudioFlinger::removeNotificationClient(pid_t pid) 1075{ 1076 Mutex::Autolock _l(mLock); 1077 1078 mNotificationClients.removeItem(pid); 1079 1080 ALOGV("%d died, releasing its sessions", pid); 1081 size_t num = mAudioSessionRefs.size(); 1082 bool removed = false; 1083 for (size_t i = 0; i< num; ) { 1084 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1085 ALOGV(" pid %d @ %d", ref->mPid, i); 1086 if (ref->mPid == pid) { 1087 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1088 mAudioSessionRefs.removeAt(i); 1089 delete ref; 1090 removed = true; 1091 num--; 1092 } else { 1093 i++; 1094 } 1095 } 1096 if (removed) { 1097 purgeStaleEffects_l(); 1098 } 1099} 1100 1101// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1102void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1103{ 1104 size_t size = mNotificationClients.size(); 1105 for (size_t i = 0; i < size; i++) { 1106 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1107 param2); 1108 } 1109} 1110 1111// removeClient_l() must be called with AudioFlinger::mLock held 1112void AudioFlinger::removeClient_l(pid_t pid) 1113{ 1114 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1115 mClients.removeItem(pid); 1116} 1117 1118// getEffectThread_l() must be called with AudioFlinger::mLock held 1119sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1120{ 1121 sp<PlaybackThread> thread; 1122 1123 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1124 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1125 ALOG_ASSERT(thread == 0); 1126 thread = mPlaybackThreads.valueAt(i); 1127 } 1128 } 1129 1130 return thread; 1131} 1132 1133// ---------------------------------------------------------------------------- 1134 1135AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1136 audio_devices_t device, type_t type) 1137 : Thread(false /*canCallJava*/), 1138 mType(type), 1139 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1140 // mChannelMask 1141 mChannelCount(0), 1142 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1143 mParamStatus(NO_ERROR), 1144 mStandby(false), mDevice(device), mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 1145 // mName will be set by concrete (non-virtual) subclass 1146 mDeathRecipient(new PMDeathRecipient(this)) 1147{ 1148} 1149 1150AudioFlinger::ThreadBase::~ThreadBase() 1151{ 1152 mParamCond.broadcast(); 1153 // do not lock the mutex in destructor 1154 releaseWakeLock_l(); 1155 if (mPowerManager != 0) { 1156 sp<IBinder> binder = mPowerManager->asBinder(); 1157 binder->unlinkToDeath(mDeathRecipient); 1158 } 1159} 1160 1161void AudioFlinger::ThreadBase::exit() 1162{ 1163 ALOGV("ThreadBase::exit"); 1164 { 1165 // This lock prevents the following race in thread (uniprocessor for illustration): 1166 // if (!exitPending()) { 1167 // // context switch from here to exit() 1168 // // exit() calls requestExit(), what exitPending() observes 1169 // // exit() calls signal(), which is dropped since no waiters 1170 // // context switch back from exit() to here 1171 // mWaitWorkCV.wait(...); 1172 // // now thread is hung 1173 // } 1174 AutoMutex lock(mLock); 1175 requestExit(); 1176 mWaitWorkCV.signal(); 1177 } 1178 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1179 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1180 requestExitAndWait(); 1181} 1182 1183status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1184{ 1185 status_t status; 1186 1187 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1188 Mutex::Autolock _l(mLock); 1189 1190 mNewParameters.add(keyValuePairs); 1191 mWaitWorkCV.signal(); 1192 // wait condition with timeout in case the thread loop has exited 1193 // before the request could be processed 1194 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1195 status = mParamStatus; 1196 mWaitWorkCV.signal(); 1197 } else { 1198 status = TIMED_OUT; 1199 } 1200 return status; 1201} 1202 1203void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1204{ 1205 Mutex::Autolock _l(mLock); 1206 sendConfigEvent_l(event, param); 1207} 1208 1209// sendConfigEvent_l() must be called with ThreadBase::mLock held 1210void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1211{ 1212 ConfigEvent configEvent; 1213 configEvent.mEvent = event; 1214 configEvent.mParam = param; 1215 mConfigEvents.add(configEvent); 1216 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1217 mWaitWorkCV.signal(); 1218} 1219 1220void AudioFlinger::ThreadBase::processConfigEvents() 1221{ 1222 mLock.lock(); 1223 while (!mConfigEvents.isEmpty()) { 1224 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1225 ConfigEvent configEvent = mConfigEvents[0]; 1226 mConfigEvents.removeAt(0); 1227 // release mLock before locking AudioFlinger mLock: lock order is always 1228 // AudioFlinger then ThreadBase to avoid cross deadlock 1229 mLock.unlock(); 1230 mAudioFlinger->mLock.lock(); 1231 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1232 mAudioFlinger->mLock.unlock(); 1233 mLock.lock(); 1234 } 1235 mLock.unlock(); 1236} 1237 1238void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1239{ 1240 const size_t SIZE = 256; 1241 char buffer[SIZE]; 1242 String8 result; 1243 1244 bool locked = tryLock(mLock); 1245 if (!locked) { 1246 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1247 write(fd, buffer, strlen(buffer)); 1248 } 1249 1250 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1251 result.append(buffer); 1252 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1253 result.append(buffer); 1254 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1255 result.append(buffer); 1256 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1257 result.append(buffer); 1258 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1259 result.append(buffer); 1260 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1261 result.append(buffer); 1262 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1263 result.append(buffer); 1264 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1265 result.append(buffer); 1266 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1267 result.append(buffer); 1268 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1269 result.append(buffer); 1270 1271 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1272 result.append(buffer); 1273 result.append(" Index Command"); 1274 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1275 snprintf(buffer, SIZE, "\n %02d ", i); 1276 result.append(buffer); 1277 result.append(mNewParameters[i]); 1278 } 1279 1280 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1281 result.append(buffer); 1282 snprintf(buffer, SIZE, " Index event param\n"); 1283 result.append(buffer); 1284 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1285 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1286 result.append(buffer); 1287 } 1288 result.append("\n"); 1289 1290 write(fd, result.string(), result.size()); 1291 1292 if (locked) { 1293 mLock.unlock(); 1294 } 1295} 1296 1297void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1298{ 1299 const size_t SIZE = 256; 1300 char buffer[SIZE]; 1301 String8 result; 1302 1303 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1304 write(fd, buffer, strlen(buffer)); 1305 1306 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1307 sp<EffectChain> chain = mEffectChains[i]; 1308 if (chain != 0) { 1309 chain->dump(fd, args); 1310 } 1311 } 1312} 1313 1314void AudioFlinger::ThreadBase::acquireWakeLock() 1315{ 1316 Mutex::Autolock _l(mLock); 1317 acquireWakeLock_l(); 1318} 1319 1320void AudioFlinger::ThreadBase::acquireWakeLock_l() 1321{ 1322 if (mPowerManager == 0) { 1323 // use checkService() to avoid blocking if power service is not up yet 1324 sp<IBinder> binder = 1325 defaultServiceManager()->checkService(String16("power")); 1326 if (binder == 0) { 1327 ALOGW("Thread %s cannot connect to the power manager service", mName); 1328 } else { 1329 mPowerManager = interface_cast<IPowerManager>(binder); 1330 binder->linkToDeath(mDeathRecipient); 1331 } 1332 } 1333 if (mPowerManager != 0) { 1334 sp<IBinder> binder = new BBinder(); 1335 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1336 binder, 1337 String16(mName)); 1338 if (status == NO_ERROR) { 1339 mWakeLockToken = binder; 1340 } 1341 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1342 } 1343} 1344 1345void AudioFlinger::ThreadBase::releaseWakeLock() 1346{ 1347 Mutex::Autolock _l(mLock); 1348 releaseWakeLock_l(); 1349} 1350 1351void AudioFlinger::ThreadBase::releaseWakeLock_l() 1352{ 1353 if (mWakeLockToken != 0) { 1354 ALOGV("releaseWakeLock_l() %s", mName); 1355 if (mPowerManager != 0) { 1356 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1357 } 1358 mWakeLockToken.clear(); 1359 } 1360} 1361 1362void AudioFlinger::ThreadBase::clearPowerManager() 1363{ 1364 Mutex::Autolock _l(mLock); 1365 releaseWakeLock_l(); 1366 mPowerManager.clear(); 1367} 1368 1369void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1370{ 1371 sp<ThreadBase> thread = mThread.promote(); 1372 if (thread != 0) { 1373 thread->clearPowerManager(); 1374 } 1375 ALOGW("power manager service died !!!"); 1376} 1377 1378void AudioFlinger::ThreadBase::setEffectSuspended( 1379 const effect_uuid_t *type, bool suspend, int sessionId) 1380{ 1381 Mutex::Autolock _l(mLock); 1382 setEffectSuspended_l(type, suspend, sessionId); 1383} 1384 1385void AudioFlinger::ThreadBase::setEffectSuspended_l( 1386 const effect_uuid_t *type, bool suspend, int sessionId) 1387{ 1388 sp<EffectChain> chain = getEffectChain_l(sessionId); 1389 if (chain != 0) { 1390 if (type != NULL) { 1391 chain->setEffectSuspended_l(type, suspend); 1392 } else { 1393 chain->setEffectSuspendedAll_l(suspend); 1394 } 1395 } 1396 1397 updateSuspendedSessions_l(type, suspend, sessionId); 1398} 1399 1400void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1401{ 1402 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1403 if (index < 0) { 1404 return; 1405 } 1406 1407 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1408 mSuspendedSessions.valueAt(index); 1409 1410 for (size_t i = 0; i < sessionEffects.size(); i++) { 1411 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1412 for (int j = 0; j < desc->mRefCount; j++) { 1413 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1414 chain->setEffectSuspendedAll_l(true); 1415 } else { 1416 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1417 desc->mType.timeLow); 1418 chain->setEffectSuspended_l(&desc->mType, true); 1419 } 1420 } 1421 } 1422} 1423 1424void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1425 bool suspend, 1426 int sessionId) 1427{ 1428 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1429 1430 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1431 1432 if (suspend) { 1433 if (index >= 0) { 1434 sessionEffects = mSuspendedSessions.valueAt(index); 1435 } else { 1436 mSuspendedSessions.add(sessionId, sessionEffects); 1437 } 1438 } else { 1439 if (index < 0) { 1440 return; 1441 } 1442 sessionEffects = mSuspendedSessions.valueAt(index); 1443 } 1444 1445 1446 int key = EffectChain::kKeyForSuspendAll; 1447 if (type != NULL) { 1448 key = type->timeLow; 1449 } 1450 index = sessionEffects.indexOfKey(key); 1451 1452 sp<SuspendedSessionDesc> desc; 1453 if (suspend) { 1454 if (index >= 0) { 1455 desc = sessionEffects.valueAt(index); 1456 } else { 1457 desc = new SuspendedSessionDesc(); 1458 if (type != NULL) { 1459 desc->mType = *type; 1460 } 1461 sessionEffects.add(key, desc); 1462 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1463 } 1464 desc->mRefCount++; 1465 } else { 1466 if (index < 0) { 1467 return; 1468 } 1469 desc = sessionEffects.valueAt(index); 1470 if (--desc->mRefCount == 0) { 1471 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1472 sessionEffects.removeItemsAt(index); 1473 if (sessionEffects.isEmpty()) { 1474 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1475 sessionId); 1476 mSuspendedSessions.removeItem(sessionId); 1477 } 1478 } 1479 } 1480 if (!sessionEffects.isEmpty()) { 1481 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1482 } 1483} 1484 1485void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1486 bool enabled, 1487 int sessionId) 1488{ 1489 Mutex::Autolock _l(mLock); 1490 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1491} 1492 1493void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1494 bool enabled, 1495 int sessionId) 1496{ 1497 if (mType != RECORD) { 1498 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1499 // another session. This gives the priority to well behaved effect control panels 1500 // and applications not using global effects. 1501 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1502 // global effects 1503 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1504 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1505 } 1506 } 1507 1508 sp<EffectChain> chain = getEffectChain_l(sessionId); 1509 if (chain != 0) { 1510 chain->checkSuspendOnEffectEnabled(effect, enabled); 1511 } 1512} 1513 1514// ---------------------------------------------------------------------------- 1515 1516AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1517 AudioStreamOut* output, 1518 audio_io_handle_t id, 1519 audio_devices_t device, 1520 type_t type) 1521 : ThreadBase(audioFlinger, id, device, type), 1522 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1523 // mStreamTypes[] initialized in constructor body 1524 mOutput(output), 1525 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1526 mMixerStatus(MIXER_IDLE), 1527 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1528 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1529 mScreenState(gScreenState), 1530 // index 0 is reserved for normal mixer's submix 1531 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1532{ 1533 snprintf(mName, kNameLength, "AudioOut_%X", id); 1534 1535 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1536 // it would be safer to explicitly pass initial masterVolume/masterMute as 1537 // parameter. 1538 // 1539 // If the HAL we are using has support for master volume or master mute, 1540 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1541 // and the mute set to false). 1542 mMasterVolume = audioFlinger->masterVolume_l(); 1543 mMasterMute = audioFlinger->masterMute_l(); 1544 if (mOutput && mOutput->audioHwDev) { 1545 if (mOutput->audioHwDev->canSetMasterVolume()) { 1546 mMasterVolume = 1.0; 1547 } 1548 1549 if (mOutput->audioHwDev->canSetMasterMute()) { 1550 mMasterMute = false; 1551 } 1552 } 1553 1554 readOutputParameters(); 1555 1556 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1557 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1558 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1559 stream = (audio_stream_type_t) (stream + 1)) { 1560 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1561 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1562 } 1563 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1564 // because mAudioFlinger doesn't have one to copy from 1565} 1566 1567AudioFlinger::PlaybackThread::~PlaybackThread() 1568{ 1569 delete [] mMixBuffer; 1570} 1571 1572void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1573{ 1574 dumpInternals(fd, args); 1575 dumpTracks(fd, args); 1576 dumpEffectChains(fd, args); 1577} 1578 1579void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1580{ 1581 const size_t SIZE = 256; 1582 char buffer[SIZE]; 1583 String8 result; 1584 1585 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1586 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1587 const stream_type_t *st = &mStreamTypes[i]; 1588 if (i > 0) { 1589 result.appendFormat(", "); 1590 } 1591 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1592 if (st->mute) { 1593 result.append("M"); 1594 } 1595 } 1596 result.append("\n"); 1597 write(fd, result.string(), result.length()); 1598 result.clear(); 1599 1600 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1601 result.append(buffer); 1602 Track::appendDumpHeader(result); 1603 for (size_t i = 0; i < mTracks.size(); ++i) { 1604 sp<Track> track = mTracks[i]; 1605 if (track != 0) { 1606 track->dump(buffer, SIZE); 1607 result.append(buffer); 1608 } 1609 } 1610 1611 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1612 result.append(buffer); 1613 Track::appendDumpHeader(result); 1614 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1615 sp<Track> track = mActiveTracks[i].promote(); 1616 if (track != 0) { 1617 track->dump(buffer, SIZE); 1618 result.append(buffer); 1619 } 1620 } 1621 write(fd, result.string(), result.size()); 1622 1623 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1624 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1625 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1626 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1627} 1628 1629void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1630{ 1631 const size_t SIZE = 256; 1632 char buffer[SIZE]; 1633 String8 result; 1634 1635 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1636 result.append(buffer); 1637 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1638 result.append(buffer); 1639 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1640 result.append(buffer); 1641 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1642 result.append(buffer); 1643 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1644 result.append(buffer); 1645 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1646 result.append(buffer); 1647 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1648 result.append(buffer); 1649 write(fd, result.string(), result.size()); 1650 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1651 1652 dumpBase(fd, args); 1653} 1654 1655// Thread virtuals 1656status_t AudioFlinger::PlaybackThread::readyToRun() 1657{ 1658 status_t status = initCheck(); 1659 if (status == NO_ERROR) { 1660 ALOGI("AudioFlinger's thread %p ready to run", this); 1661 } else { 1662 ALOGE("No working audio driver found."); 1663 } 1664 return status; 1665} 1666 1667void AudioFlinger::PlaybackThread::onFirstRef() 1668{ 1669 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1670} 1671 1672// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1673sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1674 const sp<AudioFlinger::Client>& client, 1675 audio_stream_type_t streamType, 1676 uint32_t sampleRate, 1677 audio_format_t format, 1678 audio_channel_mask_t channelMask, 1679 int frameCount, 1680 const sp<IMemory>& sharedBuffer, 1681 int sessionId, 1682 IAudioFlinger::track_flags_t flags, 1683 pid_t tid, 1684 status_t *status) 1685{ 1686 sp<Track> track; 1687 status_t lStatus; 1688 1689 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1690 1691 // client expresses a preference for FAST, but we get the final say 1692 if (flags & IAudioFlinger::TRACK_FAST) { 1693 if ( 1694 // not timed 1695 (!isTimed) && 1696 // either of these use cases: 1697 ( 1698 // use case 1: shared buffer with any frame count 1699 ( 1700 (sharedBuffer != 0) 1701 ) || 1702 // use case 2: callback handler and frame count is default or at least as large as HAL 1703 ( 1704 (tid != -1) && 1705 ((frameCount == 0) || 1706 (frameCount >= (int) (mFrameCount * kFastTrackMultiplier))) 1707 ) 1708 ) && 1709 // PCM data 1710 audio_is_linear_pcm(format) && 1711 // mono or stereo 1712 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1713 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1714#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1715 // hardware sample rate 1716 (sampleRate == mSampleRate) && 1717#endif 1718 // normal mixer has an associated fast mixer 1719 hasFastMixer() && 1720 // there are sufficient fast track slots available 1721 (mFastTrackAvailMask != 0) 1722 // FIXME test that MixerThread for this fast track has a capable output HAL 1723 // FIXME add a permission test also? 1724 ) { 1725 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1726 if (frameCount == 0) { 1727 frameCount = mFrameCount * kFastTrackMultiplier; 1728 } 1729 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1730 frameCount, mFrameCount); 1731 } else { 1732 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1733 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1734 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1735 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1736 audio_is_linear_pcm(format), 1737 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1738 flags &= ~IAudioFlinger::TRACK_FAST; 1739 // For compatibility with AudioTrack calculation, buffer depth is forced 1740 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1741 // This is probably too conservative, but legacy application code may depend on it. 1742 // If you change this calculation, also review the start threshold which is related. 1743 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1744 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1745 if (minBufCount < 2) { 1746 minBufCount = 2; 1747 } 1748 int minFrameCount = mNormalFrameCount * minBufCount; 1749 if (frameCount < minFrameCount) { 1750 frameCount = minFrameCount; 1751 } 1752 } 1753 } 1754 1755 if (mType == DIRECT) { 1756 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1757 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1758 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1759 "for output %p with format %d", 1760 sampleRate, format, channelMask, mOutput, mFormat); 1761 lStatus = BAD_VALUE; 1762 goto Exit; 1763 } 1764 } 1765 } else { 1766 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1767 if (sampleRate > mSampleRate*2) { 1768 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1769 lStatus = BAD_VALUE; 1770 goto Exit; 1771 } 1772 } 1773 1774 lStatus = initCheck(); 1775 if (lStatus != NO_ERROR) { 1776 ALOGE("Audio driver not initialized."); 1777 goto Exit; 1778 } 1779 1780 { // scope for mLock 1781 Mutex::Autolock _l(mLock); 1782 1783 // all tracks in same audio session must share the same routing strategy otherwise 1784 // conflicts will happen when tracks are moved from one output to another by audio policy 1785 // manager 1786 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1787 for (size_t i = 0; i < mTracks.size(); ++i) { 1788 sp<Track> t = mTracks[i]; 1789 if (t != 0 && !t->isOutputTrack()) { 1790 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1791 if (sessionId == t->sessionId() && strategy != actual) { 1792 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1793 strategy, actual); 1794 lStatus = BAD_VALUE; 1795 goto Exit; 1796 } 1797 } 1798 } 1799 1800 if (!isTimed) { 1801 track = new Track(this, client, streamType, sampleRate, format, 1802 channelMask, frameCount, sharedBuffer, sessionId, flags); 1803 } else { 1804 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1805 channelMask, frameCount, sharedBuffer, sessionId); 1806 } 1807 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1808 lStatus = NO_MEMORY; 1809 goto Exit; 1810 } 1811 mTracks.add(track); 1812 1813 sp<EffectChain> chain = getEffectChain_l(sessionId); 1814 if (chain != 0) { 1815 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1816 track->setMainBuffer(chain->inBuffer()); 1817 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1818 chain->incTrackCnt(); 1819 } 1820 } 1821 1822 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1823 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1824 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1825 // so ask activity manager to do this on our behalf 1826 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1827 if (err != 0) { 1828 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1829 kPriorityAudioApp, callingPid, tid, err); 1830 } 1831 } 1832 1833 lStatus = NO_ERROR; 1834 1835Exit: 1836 if (status) { 1837 *status = lStatus; 1838 } 1839 return track; 1840} 1841 1842uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1843{ 1844 if (mFastMixer != NULL) { 1845 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1846 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1847 } 1848 return latency; 1849} 1850 1851uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1852{ 1853 return latency; 1854} 1855 1856uint32_t AudioFlinger::PlaybackThread::latency() const 1857{ 1858 Mutex::Autolock _l(mLock); 1859 return latency_l(); 1860} 1861uint32_t AudioFlinger::PlaybackThread::latency_l() const 1862{ 1863 if (initCheck() == NO_ERROR) { 1864 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1865 } else { 1866 return 0; 1867 } 1868} 1869 1870void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1871{ 1872 Mutex::Autolock _l(mLock); 1873 // Don't apply master volume in SW if our HAL can do it for us. 1874 if (mOutput && mOutput->audioHwDev && 1875 mOutput->audioHwDev->canSetMasterVolume()) { 1876 mMasterVolume = 1.0; 1877 } else { 1878 mMasterVolume = value; 1879 } 1880} 1881 1882void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1883{ 1884 Mutex::Autolock _l(mLock); 1885 // Don't apply master mute in SW if our HAL can do it for us. 1886 if (mOutput && mOutput->audioHwDev && 1887 mOutput->audioHwDev->canSetMasterMute()) { 1888 mMasterMute = false; 1889 } else { 1890 mMasterMute = muted; 1891 } 1892} 1893 1894void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1895{ 1896 Mutex::Autolock _l(mLock); 1897 mStreamTypes[stream].volume = value; 1898} 1899 1900void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1901{ 1902 Mutex::Autolock _l(mLock); 1903 mStreamTypes[stream].mute = muted; 1904} 1905 1906float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1907{ 1908 Mutex::Autolock _l(mLock); 1909 return mStreamTypes[stream].volume; 1910} 1911 1912// addTrack_l() must be called with ThreadBase::mLock held 1913status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1914{ 1915 status_t status = ALREADY_EXISTS; 1916 1917 // set retry count for buffer fill 1918 track->mRetryCount = kMaxTrackStartupRetries; 1919 if (mActiveTracks.indexOf(track) < 0) { 1920 // the track is newly added, make sure it fills up all its 1921 // buffers before playing. This is to ensure the client will 1922 // effectively get the latency it requested. 1923 track->mFillingUpStatus = Track::FS_FILLING; 1924 track->mResetDone = false; 1925 track->mPresentationCompleteFrames = 0; 1926 mActiveTracks.add(track); 1927 if (track->mainBuffer() != mMixBuffer) { 1928 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1929 if (chain != 0) { 1930 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1931 chain->incActiveTrackCnt(); 1932 } 1933 } 1934 1935 status = NO_ERROR; 1936 } 1937 1938 ALOGV("mWaitWorkCV.broadcast"); 1939 mWaitWorkCV.broadcast(); 1940 1941 return status; 1942} 1943 1944// destroyTrack_l() must be called with ThreadBase::mLock held 1945void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1946{ 1947 track->mState = TrackBase::TERMINATED; 1948 // active tracks are removed by threadLoop() 1949 if (mActiveTracks.indexOf(track) < 0) { 1950 removeTrack_l(track); 1951 } 1952} 1953 1954void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1955{ 1956 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1957 mTracks.remove(track); 1958 deleteTrackName_l(track->name()); 1959 // redundant as track is about to be destroyed, for dumpsys only 1960 track->mName = -1; 1961 if (track->isFastTrack()) { 1962 int index = track->mFastIndex; 1963 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1964 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1965 mFastTrackAvailMask |= 1 << index; 1966 // redundant as track is about to be destroyed, for dumpsys only 1967 track->mFastIndex = -1; 1968 } 1969 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1970 if (chain != 0) { 1971 chain->decTrackCnt(); 1972 } 1973} 1974 1975String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1976{ 1977 String8 out_s8 = String8(""); 1978 char *s; 1979 1980 Mutex::Autolock _l(mLock); 1981 if (initCheck() != NO_ERROR) { 1982 return out_s8; 1983 } 1984 1985 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1986 out_s8 = String8(s); 1987 free(s); 1988 return out_s8; 1989} 1990 1991// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1992void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1993 AudioSystem::OutputDescriptor desc; 1994 void *param2 = NULL; 1995 1996 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1997 1998 switch (event) { 1999 case AudioSystem::OUTPUT_OPENED: 2000 case AudioSystem::OUTPUT_CONFIG_CHANGED: 2001 desc.channels = mChannelMask; 2002 desc.samplingRate = mSampleRate; 2003 desc.format = mFormat; 2004 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 2005 desc.latency = latency(); 2006 param2 = &desc; 2007 break; 2008 2009 case AudioSystem::STREAM_CONFIG_CHANGED: 2010 param2 = ¶m; 2011 case AudioSystem::OUTPUT_CLOSED: 2012 default: 2013 break; 2014 } 2015 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 2016} 2017 2018void AudioFlinger::PlaybackThread::readOutputParameters() 2019{ 2020 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 2021 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 2022 mChannelCount = (uint16_t)popcount(mChannelMask); 2023 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2024 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 2025 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 2026 if (mFrameCount & 15) { 2027 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2028 mFrameCount); 2029 } 2030 2031 // Calculate size of normal mix buffer relative to the HAL output buffer size 2032 double multiplier = 1.0; 2033 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 2034 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 2035 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2036 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2037 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2038 maxNormalFrameCount = maxNormalFrameCount & ~15; 2039 if (maxNormalFrameCount < minNormalFrameCount) { 2040 maxNormalFrameCount = minNormalFrameCount; 2041 } 2042 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2043 if (multiplier <= 1.0) { 2044 multiplier = 1.0; 2045 } else if (multiplier <= 2.0) { 2046 if (2 * mFrameCount <= maxNormalFrameCount) { 2047 multiplier = 2.0; 2048 } else { 2049 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2050 } 2051 } else { 2052 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2053 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2054 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2055 // FIXME this rounding up should not be done if no HAL SRC 2056 uint32_t truncMult = (uint32_t) multiplier; 2057 if ((truncMult & 1)) { 2058 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2059 ++truncMult; 2060 } 2061 } 2062 multiplier = (double) truncMult; 2063 } 2064 } 2065 mNormalFrameCount = multiplier * mFrameCount; 2066 // round up to nearest 16 frames to satisfy AudioMixer 2067 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2068 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2069 2070 delete[] mMixBuffer; 2071 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2072 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2073 2074 // force reconfiguration of effect chains and engines to take new buffer size and audio 2075 // parameters into account 2076 // Note that mLock is not held when readOutputParameters() is called from the constructor 2077 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2078 // matter. 2079 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2080 Vector< sp<EffectChain> > effectChains = mEffectChains; 2081 for (size_t i = 0; i < effectChains.size(); i ++) { 2082 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2083 } 2084} 2085 2086 2087status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2088{ 2089 if (halFrames == NULL || dspFrames == NULL) { 2090 return BAD_VALUE; 2091 } 2092 Mutex::Autolock _l(mLock); 2093 if (initCheck() != NO_ERROR) { 2094 return INVALID_OPERATION; 2095 } 2096 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2097 2098 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2099} 2100 2101uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2102{ 2103 Mutex::Autolock _l(mLock); 2104 uint32_t result = 0; 2105 if (getEffectChain_l(sessionId) != 0) { 2106 result = EFFECT_SESSION; 2107 } 2108 2109 for (size_t i = 0; i < mTracks.size(); ++i) { 2110 sp<Track> track = mTracks[i]; 2111 if (sessionId == track->sessionId() && 2112 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2113 result |= TRACK_SESSION; 2114 break; 2115 } 2116 } 2117 2118 return result; 2119} 2120 2121uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2122{ 2123 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2124 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2125 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2126 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2127 } 2128 for (size_t i = 0; i < mTracks.size(); i++) { 2129 sp<Track> track = mTracks[i]; 2130 if (sessionId == track->sessionId() && 2131 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2132 return AudioSystem::getStrategyForStream(track->streamType()); 2133 } 2134 } 2135 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2136} 2137 2138 2139AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2140{ 2141 Mutex::Autolock _l(mLock); 2142 return mOutput; 2143} 2144 2145AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2146{ 2147 Mutex::Autolock _l(mLock); 2148 AudioStreamOut *output = mOutput; 2149 mOutput = NULL; 2150 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2151 // must push a NULL and wait for ack 2152 mOutputSink.clear(); 2153 mPipeSink.clear(); 2154 mNormalSink.clear(); 2155 return output; 2156} 2157 2158// this method must always be called either with ThreadBase mLock held or inside the thread loop 2159audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2160{ 2161 if (mOutput == NULL) { 2162 return NULL; 2163 } 2164 return &mOutput->stream->common; 2165} 2166 2167uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2168{ 2169 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2170} 2171 2172status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2173{ 2174 if (!isValidSyncEvent(event)) { 2175 return BAD_VALUE; 2176 } 2177 2178 Mutex::Autolock _l(mLock); 2179 2180 for (size_t i = 0; i < mTracks.size(); ++i) { 2181 sp<Track> track = mTracks[i]; 2182 if (event->triggerSession() == track->sessionId()) { 2183 (void) track->setSyncEvent(event); 2184 return NO_ERROR; 2185 } 2186 } 2187 2188 return NAME_NOT_FOUND; 2189} 2190 2191bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2192{ 2193 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2194} 2195 2196void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2197{ 2198 size_t count = tracksToRemove.size(); 2199 if (CC_UNLIKELY(count)) { 2200 for (size_t i = 0 ; i < count ; i++) { 2201 const sp<Track>& track = tracksToRemove.itemAt(i); 2202 if ((track->sharedBuffer() != 0) && 2203 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2204 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2205 } 2206 } 2207 } 2208 2209} 2210 2211// ---------------------------------------------------------------------------- 2212 2213AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2214 audio_io_handle_t id, audio_devices_t device, type_t type) 2215 : PlaybackThread(audioFlinger, output, id, device, type), 2216 // mAudioMixer below 2217 // mFastMixer below 2218 mFastMixerFutex(0) 2219 // mOutputSink below 2220 // mPipeSink below 2221 // mNormalSink below 2222{ 2223 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2224 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2225 "mFrameCount=%d, mNormalFrameCount=%d", 2226 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2227 mNormalFrameCount); 2228 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2229 2230 // FIXME - Current mixer implementation only supports stereo output 2231 if (mChannelCount != FCC_2) { 2232 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2233 } 2234 2235 // create an NBAIO sink for the HAL output stream, and negotiate 2236 mOutputSink = new AudioStreamOutSink(output->stream); 2237 size_t numCounterOffers = 0; 2238 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2239 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2240 ALOG_ASSERT(index == 0); 2241 2242 // initialize fast mixer depending on configuration 2243 bool initFastMixer; 2244 switch (kUseFastMixer) { 2245 case FastMixer_Never: 2246 initFastMixer = false; 2247 break; 2248 case FastMixer_Always: 2249 initFastMixer = true; 2250 break; 2251 case FastMixer_Static: 2252 case FastMixer_Dynamic: 2253 initFastMixer = mFrameCount < mNormalFrameCount; 2254 break; 2255 } 2256 if (initFastMixer) { 2257 2258 // create a MonoPipe to connect our submix to FastMixer 2259 NBAIO_Format format = mOutputSink->format(); 2260 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2261 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2262 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2263 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2264 const NBAIO_Format offers[1] = {format}; 2265 size_t numCounterOffers = 0; 2266 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2267 ALOG_ASSERT(index == 0); 2268 monoPipe->setAvgFrames((mScreenState & 1) ? 2269 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2270 mPipeSink = monoPipe; 2271 2272#ifdef TEE_SINK_FRAMES 2273 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2274 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2275 numCounterOffers = 0; 2276 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2277 ALOG_ASSERT(index == 0); 2278 mTeeSink = teeSink; 2279 PipeReader *teeSource = new PipeReader(*teeSink); 2280 numCounterOffers = 0; 2281 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2282 ALOG_ASSERT(index == 0); 2283 mTeeSource = teeSource; 2284#endif 2285 2286 // create fast mixer and configure it initially with just one fast track for our submix 2287 mFastMixer = new FastMixer(); 2288 FastMixerStateQueue *sq = mFastMixer->sq(); 2289#ifdef STATE_QUEUE_DUMP 2290 sq->setObserverDump(&mStateQueueObserverDump); 2291 sq->setMutatorDump(&mStateQueueMutatorDump); 2292#endif 2293 FastMixerState *state = sq->begin(); 2294 FastTrack *fastTrack = &state->mFastTracks[0]; 2295 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2296 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2297 fastTrack->mVolumeProvider = NULL; 2298 fastTrack->mGeneration++; 2299 state->mFastTracksGen++; 2300 state->mTrackMask = 1; 2301 // fast mixer will use the HAL output sink 2302 state->mOutputSink = mOutputSink.get(); 2303 state->mOutputSinkGen++; 2304 state->mFrameCount = mFrameCount; 2305 state->mCommand = FastMixerState::COLD_IDLE; 2306 // already done in constructor initialization list 2307 //mFastMixerFutex = 0; 2308 state->mColdFutexAddr = &mFastMixerFutex; 2309 state->mColdGen++; 2310 state->mDumpState = &mFastMixerDumpState; 2311 state->mTeeSink = mTeeSink.get(); 2312 sq->end(); 2313 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2314 2315 // start the fast mixer 2316 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2317 pid_t tid = mFastMixer->getTid(); 2318 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2319 if (err != 0) { 2320 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2321 kPriorityFastMixer, getpid_cached, tid, err); 2322 } 2323 2324#ifdef AUDIO_WATCHDOG 2325 // create and start the watchdog 2326 mAudioWatchdog = new AudioWatchdog(); 2327 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2328 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2329 tid = mAudioWatchdog->getTid(); 2330 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2331 if (err != 0) { 2332 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2333 kPriorityFastMixer, getpid_cached, tid, err); 2334 } 2335#endif 2336 2337 } else { 2338 mFastMixer = NULL; 2339 } 2340 2341 switch (kUseFastMixer) { 2342 case FastMixer_Never: 2343 case FastMixer_Dynamic: 2344 mNormalSink = mOutputSink; 2345 break; 2346 case FastMixer_Always: 2347 mNormalSink = mPipeSink; 2348 break; 2349 case FastMixer_Static: 2350 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2351 break; 2352 } 2353} 2354 2355AudioFlinger::MixerThread::~MixerThread() 2356{ 2357 if (mFastMixer != NULL) { 2358 FastMixerStateQueue *sq = mFastMixer->sq(); 2359 FastMixerState *state = sq->begin(); 2360 if (state->mCommand == FastMixerState::COLD_IDLE) { 2361 int32_t old = android_atomic_inc(&mFastMixerFutex); 2362 if (old == -1) { 2363 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2364 } 2365 } 2366 state->mCommand = FastMixerState::EXIT; 2367 sq->end(); 2368 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2369 mFastMixer->join(); 2370 // Though the fast mixer thread has exited, it's state queue is still valid. 2371 // We'll use that extract the final state which contains one remaining fast track 2372 // corresponding to our sub-mix. 2373 state = sq->begin(); 2374 ALOG_ASSERT(state->mTrackMask == 1); 2375 FastTrack *fastTrack = &state->mFastTracks[0]; 2376 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2377 delete fastTrack->mBufferProvider; 2378 sq->end(false /*didModify*/); 2379 delete mFastMixer; 2380 if (mAudioWatchdog != 0) { 2381 mAudioWatchdog->requestExit(); 2382 mAudioWatchdog->requestExitAndWait(); 2383 mAudioWatchdog.clear(); 2384 } 2385 } 2386 delete mAudioMixer; 2387} 2388 2389class CpuStats { 2390public: 2391 CpuStats(); 2392 void sample(const String8 &title); 2393#ifdef DEBUG_CPU_USAGE 2394private: 2395 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2396 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2397 2398 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2399 2400 int mCpuNum; // thread's current CPU number 2401 int mCpukHz; // frequency of thread's current CPU in kHz 2402#endif 2403}; 2404 2405CpuStats::CpuStats() 2406#ifdef DEBUG_CPU_USAGE 2407 : mCpuNum(-1), mCpukHz(-1) 2408#endif 2409{ 2410} 2411 2412void CpuStats::sample(const String8 &title) { 2413#ifdef DEBUG_CPU_USAGE 2414 // get current thread's delta CPU time in wall clock ns 2415 double wcNs; 2416 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2417 2418 // record sample for wall clock statistics 2419 if (valid) { 2420 mWcStats.sample(wcNs); 2421 } 2422 2423 // get the current CPU number 2424 int cpuNum = sched_getcpu(); 2425 2426 // get the current CPU frequency in kHz 2427 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2428 2429 // check if either CPU number or frequency changed 2430 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2431 mCpuNum = cpuNum; 2432 mCpukHz = cpukHz; 2433 // ignore sample for purposes of cycles 2434 valid = false; 2435 } 2436 2437 // if no change in CPU number or frequency, then record sample for cycle statistics 2438 if (valid && mCpukHz > 0) { 2439 double cycles = wcNs * cpukHz * 0.000001; 2440 mHzStats.sample(cycles); 2441 } 2442 2443 unsigned n = mWcStats.n(); 2444 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2445 if ((n & 127) == 1) { 2446 long long elapsed = mCpuUsage.elapsed(); 2447 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2448 double perLoop = elapsed / (double) n; 2449 double perLoop100 = perLoop * 0.01; 2450 double perLoop1k = perLoop * 0.001; 2451 double mean = mWcStats.mean(); 2452 double stddev = mWcStats.stddev(); 2453 double minimum = mWcStats.minimum(); 2454 double maximum = mWcStats.maximum(); 2455 double meanCycles = mHzStats.mean(); 2456 double stddevCycles = mHzStats.stddev(); 2457 double minCycles = mHzStats.minimum(); 2458 double maxCycles = mHzStats.maximum(); 2459 mCpuUsage.resetElapsed(); 2460 mWcStats.reset(); 2461 mHzStats.reset(); 2462 ALOGD("CPU usage for %s over past %.1f secs\n" 2463 " (%u mixer loops at %.1f mean ms per loop):\n" 2464 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2465 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2466 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2467 title.string(), 2468 elapsed * .000000001, n, perLoop * .000001, 2469 mean * .001, 2470 stddev * .001, 2471 minimum * .001, 2472 maximum * .001, 2473 mean / perLoop100, 2474 stddev / perLoop100, 2475 minimum / perLoop100, 2476 maximum / perLoop100, 2477 meanCycles / perLoop1k, 2478 stddevCycles / perLoop1k, 2479 minCycles / perLoop1k, 2480 maxCycles / perLoop1k); 2481 2482 } 2483 } 2484#endif 2485}; 2486 2487void AudioFlinger::PlaybackThread::checkSilentMode_l() 2488{ 2489 if (!mMasterMute) { 2490 char value[PROPERTY_VALUE_MAX]; 2491 if (property_get("ro.audio.silent", value, "0") > 0) { 2492 char *endptr; 2493 unsigned long ul = strtoul(value, &endptr, 0); 2494 if (*endptr == '\0' && ul != 0) { 2495 ALOGD("Silence is golden"); 2496 // The setprop command will not allow a property to be changed after 2497 // the first time it is set, so we don't have to worry about un-muting. 2498 setMasterMute_l(true); 2499 } 2500 } 2501 } 2502} 2503 2504bool AudioFlinger::PlaybackThread::threadLoop() 2505{ 2506 Vector< sp<Track> > tracksToRemove; 2507 2508 standbyTime = systemTime(); 2509 2510 // MIXER 2511 nsecs_t lastWarning = 0; 2512 2513 // DUPLICATING 2514 // FIXME could this be made local to while loop? 2515 writeFrames = 0; 2516 2517 cacheParameters_l(); 2518 sleepTime = idleSleepTime; 2519 2520 if (mType == MIXER) { 2521 sleepTimeShift = 0; 2522 } 2523 2524 CpuStats cpuStats; 2525 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2526 2527 acquireWakeLock(); 2528 2529 while (!exitPending()) 2530 { 2531 cpuStats.sample(myName); 2532 2533 Vector< sp<EffectChain> > effectChains; 2534 2535 processConfigEvents(); 2536 2537 { // scope for mLock 2538 2539 Mutex::Autolock _l(mLock); 2540 2541 if (checkForNewParameters_l()) { 2542 cacheParameters_l(); 2543 } 2544 2545 saveOutputTracks(); 2546 2547 // put audio hardware into standby after short delay 2548 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2549 isSuspended())) { 2550 if (!mStandby) { 2551 2552 threadLoop_standby(); 2553 2554 mStandby = true; 2555 mBytesWritten = 0; 2556 } 2557 2558 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2559 // we're about to wait, flush the binder command buffer 2560 IPCThreadState::self()->flushCommands(); 2561 2562 clearOutputTracks(); 2563 2564 if (exitPending()) break; 2565 2566 releaseWakeLock_l(); 2567 // wait until we have something to do... 2568 ALOGV("%s going to sleep", myName.string()); 2569 mWaitWorkCV.wait(mLock); 2570 ALOGV("%s waking up", myName.string()); 2571 acquireWakeLock_l(); 2572 2573 mMixerStatus = MIXER_IDLE; 2574 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2575 2576 checkSilentMode_l(); 2577 2578 standbyTime = systemTime() + standbyDelay; 2579 sleepTime = idleSleepTime; 2580 if (mType == MIXER) { 2581 sleepTimeShift = 0; 2582 } 2583 2584 continue; 2585 } 2586 } 2587 2588 // mMixerStatusIgnoringFastTracks is also updated internally 2589 mMixerStatus = prepareTracks_l(&tracksToRemove); 2590 2591 // prevent any changes in effect chain list and in each effect chain 2592 // during mixing and effect process as the audio buffers could be deleted 2593 // or modified if an effect is created or deleted 2594 lockEffectChains_l(effectChains); 2595 } 2596 2597 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2598 threadLoop_mix(); 2599 } else { 2600 threadLoop_sleepTime(); 2601 } 2602 2603 if (isSuspended()) { 2604 sleepTime = suspendSleepTimeUs(); 2605 } 2606 2607 // only process effects if we're going to write 2608 if (sleepTime == 0) { 2609 for (size_t i = 0; i < effectChains.size(); i ++) { 2610 effectChains[i]->process_l(); 2611 } 2612 } 2613 2614 // enable changes in effect chain 2615 unlockEffectChains(effectChains); 2616 2617 // sleepTime == 0 means we must write to audio hardware 2618 if (sleepTime == 0) { 2619 2620 threadLoop_write(); 2621 2622if (mType == MIXER) { 2623 // write blocked detection 2624 nsecs_t now = systemTime(); 2625 nsecs_t delta = now - mLastWriteTime; 2626 if (!mStandby && delta > maxPeriod) { 2627 mNumDelayedWrites++; 2628 if ((now - lastWarning) > kWarningThrottleNs) { 2629#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2630 ScopedTrace st(ATRACE_TAG, "underrun"); 2631#endif 2632 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2633 ns2ms(delta), mNumDelayedWrites, this); 2634 lastWarning = now; 2635 } 2636 } 2637} 2638 2639 mStandby = false; 2640 } else { 2641 usleep(sleepTime); 2642 } 2643 2644 // Finally let go of removed track(s), without the lock held 2645 // since we can't guarantee the destructors won't acquire that 2646 // same lock. This will also mutate and push a new fast mixer state. 2647 threadLoop_removeTracks(tracksToRemove); 2648 tracksToRemove.clear(); 2649 2650 // FIXME I don't understand the need for this here; 2651 // it was in the original code but maybe the 2652 // assignment in saveOutputTracks() makes this unnecessary? 2653 clearOutputTracks(); 2654 2655 // Effect chains will be actually deleted here if they were removed from 2656 // mEffectChains list during mixing or effects processing 2657 effectChains.clear(); 2658 2659 // FIXME Note that the above .clear() is no longer necessary since effectChains 2660 // is now local to this block, but will keep it for now (at least until merge done). 2661 } 2662 2663 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2664 if (mType == MIXER || mType == DIRECT) { 2665 // put output stream into standby mode 2666 if (!mStandby) { 2667 mOutput->stream->common.standby(&mOutput->stream->common); 2668 } 2669 } 2670 2671 releaseWakeLock(); 2672 2673 ALOGV("Thread %p type %d exiting", this, mType); 2674 return false; 2675} 2676 2677void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2678{ 2679 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2680} 2681 2682void AudioFlinger::MixerThread::threadLoop_write() 2683{ 2684 // FIXME we should only do one push per cycle; confirm this is true 2685 // Start the fast mixer if it's not already running 2686 if (mFastMixer != NULL) { 2687 FastMixerStateQueue *sq = mFastMixer->sq(); 2688 FastMixerState *state = sq->begin(); 2689 if (state->mCommand != FastMixerState::MIX_WRITE && 2690 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2691 if (state->mCommand == FastMixerState::COLD_IDLE) { 2692 int32_t old = android_atomic_inc(&mFastMixerFutex); 2693 if (old == -1) { 2694 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2695 } 2696 if (mAudioWatchdog != 0) { 2697 mAudioWatchdog->resume(); 2698 } 2699 } 2700 state->mCommand = FastMixerState::MIX_WRITE; 2701 sq->end(); 2702 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2703 if (kUseFastMixer == FastMixer_Dynamic) { 2704 mNormalSink = mPipeSink; 2705 } 2706 } else { 2707 sq->end(false /*didModify*/); 2708 } 2709 } 2710 PlaybackThread::threadLoop_write(); 2711} 2712 2713// shared by MIXER and DIRECT, overridden by DUPLICATING 2714void AudioFlinger::PlaybackThread::threadLoop_write() 2715{ 2716 // FIXME rewrite to reduce number of system calls 2717 mLastWriteTime = systemTime(); 2718 mInWrite = true; 2719 int bytesWritten; 2720 2721 // If an NBAIO sink is present, use it to write the normal mixer's submix 2722 if (mNormalSink != 0) { 2723#define mBitShift 2 // FIXME 2724 size_t count = mixBufferSize >> mBitShift; 2725#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2726 Tracer::traceBegin(ATRACE_TAG, "write"); 2727#endif 2728 // update the setpoint when gScreenState changes 2729 uint32_t screenState = gScreenState; 2730 if (screenState != mScreenState) { 2731 mScreenState = screenState; 2732 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2733 if (pipe != NULL) { 2734 pipe->setAvgFrames((mScreenState & 1) ? 2735 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2736 } 2737 } 2738 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2739#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2740 Tracer::traceEnd(ATRACE_TAG); 2741#endif 2742 if (framesWritten > 0) { 2743 bytesWritten = framesWritten << mBitShift; 2744 } else { 2745 bytesWritten = framesWritten; 2746 } 2747 // otherwise use the HAL / AudioStreamOut directly 2748 } else { 2749 // Direct output thread. 2750 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2751 } 2752 2753 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2754 mNumWrites++; 2755 mInWrite = false; 2756} 2757 2758void AudioFlinger::MixerThread::threadLoop_standby() 2759{ 2760 // Idle the fast mixer if it's currently running 2761 if (mFastMixer != NULL) { 2762 FastMixerStateQueue *sq = mFastMixer->sq(); 2763 FastMixerState *state = sq->begin(); 2764 if (!(state->mCommand & FastMixerState::IDLE)) { 2765 state->mCommand = FastMixerState::COLD_IDLE; 2766 state->mColdFutexAddr = &mFastMixerFutex; 2767 state->mColdGen++; 2768 mFastMixerFutex = 0; 2769 sq->end(); 2770 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2771 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2772 if (kUseFastMixer == FastMixer_Dynamic) { 2773 mNormalSink = mOutputSink; 2774 } 2775 if (mAudioWatchdog != 0) { 2776 mAudioWatchdog->pause(); 2777 } 2778 } else { 2779 sq->end(false /*didModify*/); 2780 } 2781 } 2782 PlaybackThread::threadLoop_standby(); 2783} 2784 2785// shared by MIXER and DIRECT, overridden by DUPLICATING 2786void AudioFlinger::PlaybackThread::threadLoop_standby() 2787{ 2788 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2789 mOutput->stream->common.standby(&mOutput->stream->common); 2790} 2791 2792void AudioFlinger::MixerThread::threadLoop_mix() 2793{ 2794 // obtain the presentation timestamp of the next output buffer 2795 int64_t pts; 2796 status_t status = INVALID_OPERATION; 2797 2798 if (mNormalSink != 0) { 2799 status = mNormalSink->getNextWriteTimestamp(&pts); 2800 } else { 2801 status = mOutputSink->getNextWriteTimestamp(&pts); 2802 } 2803 2804 if (status != NO_ERROR) { 2805 pts = AudioBufferProvider::kInvalidPTS; 2806 } 2807 2808 // mix buffers... 2809 mAudioMixer->process(pts); 2810 // increase sleep time progressively when application underrun condition clears. 2811 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2812 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2813 // such that we would underrun the audio HAL. 2814 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2815 sleepTimeShift--; 2816 } 2817 sleepTime = 0; 2818 standbyTime = systemTime() + standbyDelay; 2819 //TODO: delay standby when effects have a tail 2820} 2821 2822void AudioFlinger::MixerThread::threadLoop_sleepTime() 2823{ 2824 // If no tracks are ready, sleep once for the duration of an output 2825 // buffer size, then write 0s to the output 2826 if (sleepTime == 0) { 2827 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2828 sleepTime = activeSleepTime >> sleepTimeShift; 2829 if (sleepTime < kMinThreadSleepTimeUs) { 2830 sleepTime = kMinThreadSleepTimeUs; 2831 } 2832 // reduce sleep time in case of consecutive application underruns to avoid 2833 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2834 // duration we would end up writing less data than needed by the audio HAL if 2835 // the condition persists. 2836 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2837 sleepTimeShift++; 2838 } 2839 } else { 2840 sleepTime = idleSleepTime; 2841 } 2842 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2843 memset (mMixBuffer, 0, mixBufferSize); 2844 sleepTime = 0; 2845 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2846 } 2847 // TODO add standby time extension fct of effect tail 2848} 2849 2850// prepareTracks_l() must be called with ThreadBase::mLock held 2851AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2852 Vector< sp<Track> > *tracksToRemove) 2853{ 2854 2855 mixer_state mixerStatus = MIXER_IDLE; 2856 // find out which tracks need to be processed 2857 size_t count = mActiveTracks.size(); 2858 size_t mixedTracks = 0; 2859 size_t tracksWithEffect = 0; 2860 // counts only _active_ fast tracks 2861 size_t fastTracks = 0; 2862 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2863 2864 float masterVolume = mMasterVolume; 2865 bool masterMute = mMasterMute; 2866 2867 if (masterMute) { 2868 masterVolume = 0; 2869 } 2870 // Delegate master volume control to effect in output mix effect chain if needed 2871 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2872 if (chain != 0) { 2873 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2874 chain->setVolume_l(&v, &v); 2875 masterVolume = (float)((v + (1 << 23)) >> 24); 2876 chain.clear(); 2877 } 2878 2879 // prepare a new state to push 2880 FastMixerStateQueue *sq = NULL; 2881 FastMixerState *state = NULL; 2882 bool didModify = false; 2883 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2884 if (mFastMixer != NULL) { 2885 sq = mFastMixer->sq(); 2886 state = sq->begin(); 2887 } 2888 2889 for (size_t i=0 ; i<count ; i++) { 2890 sp<Track> t = mActiveTracks[i].promote(); 2891 if (t == 0) continue; 2892 2893 // this const just means the local variable doesn't change 2894 Track* const track = t.get(); 2895 2896 // process fast tracks 2897 if (track->isFastTrack()) { 2898 2899 // It's theoretically possible (though unlikely) for a fast track to be created 2900 // and then removed within the same normal mix cycle. This is not a problem, as 2901 // the track never becomes active so it's fast mixer slot is never touched. 2902 // The converse, of removing an (active) track and then creating a new track 2903 // at the identical fast mixer slot within the same normal mix cycle, 2904 // is impossible because the slot isn't marked available until the end of each cycle. 2905 int j = track->mFastIndex; 2906 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2907 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2908 FastTrack *fastTrack = &state->mFastTracks[j]; 2909 2910 // Determine whether the track is currently in underrun condition, 2911 // and whether it had a recent underrun. 2912 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2913 FastTrackUnderruns underruns = ftDump->mUnderruns; 2914 uint32_t recentFull = (underruns.mBitFields.mFull - 2915 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2916 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2917 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2918 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2919 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2920 uint32_t recentUnderruns = recentPartial + recentEmpty; 2921 track->mObservedUnderruns = underruns; 2922 // don't count underruns that occur while stopping or pausing 2923 // or stopped which can occur when flush() is called while active 2924 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2925 track->mUnderrunCount += recentUnderruns; 2926 } 2927 2928 // This is similar to the state machine for normal tracks, 2929 // with a few modifications for fast tracks. 2930 bool isActive = true; 2931 switch (track->mState) { 2932 case TrackBase::STOPPING_1: 2933 // track stays active in STOPPING_1 state until first underrun 2934 if (recentUnderruns > 0) { 2935 track->mState = TrackBase::STOPPING_2; 2936 } 2937 break; 2938 case TrackBase::PAUSING: 2939 // ramp down is not yet implemented 2940 track->setPaused(); 2941 break; 2942 case TrackBase::RESUMING: 2943 // ramp up is not yet implemented 2944 track->mState = TrackBase::ACTIVE; 2945 break; 2946 case TrackBase::ACTIVE: 2947 if (recentFull > 0 || recentPartial > 0) { 2948 // track has provided at least some frames recently: reset retry count 2949 track->mRetryCount = kMaxTrackRetries; 2950 } 2951 if (recentUnderruns == 0) { 2952 // no recent underruns: stay active 2953 break; 2954 } 2955 // there has recently been an underrun of some kind 2956 if (track->sharedBuffer() == 0) { 2957 // were any of the recent underruns "empty" (no frames available)? 2958 if (recentEmpty == 0) { 2959 // no, then ignore the partial underruns as they are allowed indefinitely 2960 break; 2961 } 2962 // there has recently been an "empty" underrun: decrement the retry counter 2963 if (--(track->mRetryCount) > 0) { 2964 break; 2965 } 2966 // indicate to client process that the track was disabled because of underrun; 2967 // it will then automatically call start() when data is available 2968 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2969 // remove from active list, but state remains ACTIVE [confusing but true] 2970 isActive = false; 2971 break; 2972 } 2973 // fall through 2974 case TrackBase::STOPPING_2: 2975 case TrackBase::PAUSED: 2976 case TrackBase::TERMINATED: 2977 case TrackBase::STOPPED: 2978 case TrackBase::FLUSHED: // flush() while active 2979 // Check for presentation complete if track is inactive 2980 // We have consumed all the buffers of this track. 2981 // This would be incomplete if we auto-paused on underrun 2982 { 2983 size_t audioHALFrames = 2984 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2985 size_t framesWritten = 2986 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2987 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2988 // track stays in active list until presentation is complete 2989 break; 2990 } 2991 } 2992 if (track->isStopping_2()) { 2993 track->mState = TrackBase::STOPPED; 2994 } 2995 if (track->isStopped()) { 2996 // Can't reset directly, as fast mixer is still polling this track 2997 // track->reset(); 2998 // So instead mark this track as needing to be reset after push with ack 2999 resetMask |= 1 << i; 3000 } 3001 isActive = false; 3002 break; 3003 case TrackBase::IDLE: 3004 default: 3005 LOG_FATAL("unexpected track state %d", track->mState); 3006 } 3007 3008 if (isActive) { 3009 // was it previously inactive? 3010 if (!(state->mTrackMask & (1 << j))) { 3011 ExtendedAudioBufferProvider *eabp = track; 3012 VolumeProvider *vp = track; 3013 fastTrack->mBufferProvider = eabp; 3014 fastTrack->mVolumeProvider = vp; 3015 fastTrack->mSampleRate = track->mSampleRate; 3016 fastTrack->mChannelMask = track->mChannelMask; 3017 fastTrack->mGeneration++; 3018 state->mTrackMask |= 1 << j; 3019 didModify = true; 3020 // no acknowledgement required for newly active tracks 3021 } 3022 // cache the combined master volume and stream type volume for fast mixer; this 3023 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3024 track->mCachedVolume = track->isMuted() ? 3025 0 : masterVolume * mStreamTypes[track->streamType()].volume; 3026 ++fastTracks; 3027 } else { 3028 // was it previously active? 3029 if (state->mTrackMask & (1 << j)) { 3030 fastTrack->mBufferProvider = NULL; 3031 fastTrack->mGeneration++; 3032 state->mTrackMask &= ~(1 << j); 3033 didModify = true; 3034 // If any fast tracks were removed, we must wait for acknowledgement 3035 // because we're about to decrement the last sp<> on those tracks. 3036 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3037 } else { 3038 LOG_FATAL("fast track %d should have been active", j); 3039 } 3040 tracksToRemove->add(track); 3041 // Avoids a misleading display in dumpsys 3042 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3043 } 3044 continue; 3045 } 3046 3047 { // local variable scope to avoid goto warning 3048 3049 audio_track_cblk_t* cblk = track->cblk(); 3050 3051 // The first time a track is added we wait 3052 // for all its buffers to be filled before processing it 3053 int name = track->name(); 3054 // make sure that we have enough frames to mix one full buffer. 3055 // enforce this condition only once to enable draining the buffer in case the client 3056 // app does not call stop() and relies on underrun to stop: 3057 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3058 // during last round 3059 uint32_t minFrames = 1; 3060 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3061 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3062 if (t->sampleRate() == (int)mSampleRate) { 3063 minFrames = mNormalFrameCount; 3064 } else { 3065 // +1 for rounding and +1 for additional sample needed for interpolation 3066 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3067 // add frames already consumed but not yet released by the resampler 3068 // because cblk->framesReady() will include these frames 3069 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3070 // the minimum track buffer size is normally twice the number of frames necessary 3071 // to fill one buffer and the resampler should not leave more than one buffer worth 3072 // of unreleased frames after each pass, but just in case... 3073 ALOG_ASSERT(minFrames <= cblk->frameCount); 3074 } 3075 } 3076 if ((track->framesReady() >= minFrames) && track->isReady() && 3077 !track->isPaused() && !track->isTerminated()) 3078 { 3079 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3080 3081 mixedTracks++; 3082 3083 // track->mainBuffer() != mMixBuffer means there is an effect chain 3084 // connected to the track 3085 chain.clear(); 3086 if (track->mainBuffer() != mMixBuffer) { 3087 chain = getEffectChain_l(track->sessionId()); 3088 // Delegate volume control to effect in track effect chain if needed 3089 if (chain != 0) { 3090 tracksWithEffect++; 3091 } else { 3092 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3093 name, track->sessionId()); 3094 } 3095 } 3096 3097 3098 int param = AudioMixer::VOLUME; 3099 if (track->mFillingUpStatus == Track::FS_FILLED) { 3100 // no ramp for the first volume setting 3101 track->mFillingUpStatus = Track::FS_ACTIVE; 3102 if (track->mState == TrackBase::RESUMING) { 3103 track->mState = TrackBase::ACTIVE; 3104 param = AudioMixer::RAMP_VOLUME; 3105 } 3106 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3107 } else if (cblk->server != 0) { 3108 // If the track is stopped before the first frame was mixed, 3109 // do not apply ramp 3110 param = AudioMixer::RAMP_VOLUME; 3111 } 3112 3113 // compute volume for this track 3114 uint32_t vl, vr, va; 3115 if (track->isMuted() || track->isPausing() || 3116 mStreamTypes[track->streamType()].mute) { 3117 vl = vr = va = 0; 3118 if (track->isPausing()) { 3119 track->setPaused(); 3120 } 3121 } else { 3122 3123 // read original volumes with volume control 3124 float typeVolume = mStreamTypes[track->streamType()].volume; 3125 float v = masterVolume * typeVolume; 3126 uint32_t vlr = cblk->getVolumeLR(); 3127 vl = vlr & 0xFFFF; 3128 vr = vlr >> 16; 3129 // track volumes come from shared memory, so can't be trusted and must be clamped 3130 if (vl > MAX_GAIN_INT) { 3131 ALOGV("Track left volume out of range: %04X", vl); 3132 vl = MAX_GAIN_INT; 3133 } 3134 if (vr > MAX_GAIN_INT) { 3135 ALOGV("Track right volume out of range: %04X", vr); 3136 vr = MAX_GAIN_INT; 3137 } 3138 // now apply the master volume and stream type volume 3139 vl = (uint32_t)(v * vl) << 12; 3140 vr = (uint32_t)(v * vr) << 12; 3141 // assuming master volume and stream type volume each go up to 1.0, 3142 // vl and vr are now in 8.24 format 3143 3144 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3145 // send level comes from shared memory and so may be corrupt 3146 if (sendLevel > MAX_GAIN_INT) { 3147 ALOGV("Track send level out of range: %04X", sendLevel); 3148 sendLevel = MAX_GAIN_INT; 3149 } 3150 va = (uint32_t)(v * sendLevel); 3151 } 3152 // Delegate volume control to effect in track effect chain if needed 3153 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3154 // Do not ramp volume if volume is controlled by effect 3155 param = AudioMixer::VOLUME; 3156 track->mHasVolumeController = true; 3157 } else { 3158 // force no volume ramp when volume controller was just disabled or removed 3159 // from effect chain to avoid volume spike 3160 if (track->mHasVolumeController) { 3161 param = AudioMixer::VOLUME; 3162 } 3163 track->mHasVolumeController = false; 3164 } 3165 3166 // Convert volumes from 8.24 to 4.12 format 3167 // This additional clamping is needed in case chain->setVolume_l() overshot 3168 vl = (vl + (1 << 11)) >> 12; 3169 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3170 vr = (vr + (1 << 11)) >> 12; 3171 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3172 3173 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3174 3175 // XXX: these things DON'T need to be done each time 3176 mAudioMixer->setBufferProvider(name, track); 3177 mAudioMixer->enable(name); 3178 3179 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3180 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3181 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3182 mAudioMixer->setParameter( 3183 name, 3184 AudioMixer::TRACK, 3185 AudioMixer::FORMAT, (void *)track->format()); 3186 mAudioMixer->setParameter( 3187 name, 3188 AudioMixer::TRACK, 3189 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3190 mAudioMixer->setParameter( 3191 name, 3192 AudioMixer::RESAMPLE, 3193 AudioMixer::SAMPLE_RATE, 3194 (void *)(cblk->sampleRate)); 3195 mAudioMixer->setParameter( 3196 name, 3197 AudioMixer::TRACK, 3198 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3199 mAudioMixer->setParameter( 3200 name, 3201 AudioMixer::TRACK, 3202 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3203 3204 // reset retry count 3205 track->mRetryCount = kMaxTrackRetries; 3206 3207 // If one track is ready, set the mixer ready if: 3208 // - the mixer was not ready during previous round OR 3209 // - no other track is not ready 3210 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3211 mixerStatus != MIXER_TRACKS_ENABLED) { 3212 mixerStatus = MIXER_TRACKS_READY; 3213 } 3214 } else { 3215 // clear effect chain input buffer if an active track underruns to avoid sending 3216 // previous audio buffer again to effects 3217 chain = getEffectChain_l(track->sessionId()); 3218 if (chain != 0) { 3219 chain->clearInputBuffer(); 3220 } 3221 3222 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3223 if ((track->sharedBuffer() != 0) || 3224 track->isStopped() || track->isPaused()) { 3225 // We have consumed all the buffers of this track. 3226 // Remove it from the list of active tracks. 3227 // TODO: use actual buffer filling status instead of latency when available from 3228 // audio HAL 3229 size_t audioHALFrames = 3230 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3231 size_t framesWritten = 3232 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3233 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3234 if (track->isStopped()) { 3235 track->reset(); 3236 } 3237 tracksToRemove->add(track); 3238 } 3239 } else { 3240 track->mUnderrunCount++; 3241 // No buffers for this track. Give it a few chances to 3242 // fill a buffer, then remove it from active list. 3243 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3244 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3245 tracksToRemove->add(track); 3246 // indicate to client process that the track was disabled because of underrun; 3247 // it will then automatically call start() when data is available 3248 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3249 // If one track is not ready, mark the mixer also not ready if: 3250 // - the mixer was ready during previous round OR 3251 // - no other track is ready 3252 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3253 mixerStatus != MIXER_TRACKS_READY) { 3254 mixerStatus = MIXER_TRACKS_ENABLED; 3255 } 3256 } 3257 mAudioMixer->disable(name); 3258 } 3259 3260 } // local variable scope to avoid goto warning 3261track_is_ready: ; 3262 3263 } 3264 3265 // Push the new FastMixer state if necessary 3266 bool pauseAudioWatchdog = false; 3267 if (didModify) { 3268 state->mFastTracksGen++; 3269 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3270 if (kUseFastMixer == FastMixer_Dynamic && 3271 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3272 state->mCommand = FastMixerState::COLD_IDLE; 3273 state->mColdFutexAddr = &mFastMixerFutex; 3274 state->mColdGen++; 3275 mFastMixerFutex = 0; 3276 if (kUseFastMixer == FastMixer_Dynamic) { 3277 mNormalSink = mOutputSink; 3278 } 3279 // If we go into cold idle, need to wait for acknowledgement 3280 // so that fast mixer stops doing I/O. 3281 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3282 pauseAudioWatchdog = true; 3283 } 3284 sq->end(); 3285 } 3286 if (sq != NULL) { 3287 sq->end(didModify); 3288 sq->push(block); 3289 } 3290 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3291 mAudioWatchdog->pause(); 3292 } 3293 3294 // Now perform the deferred reset on fast tracks that have stopped 3295 while (resetMask != 0) { 3296 size_t i = __builtin_ctz(resetMask); 3297 ALOG_ASSERT(i < count); 3298 resetMask &= ~(1 << i); 3299 sp<Track> t = mActiveTracks[i].promote(); 3300 if (t == 0) continue; 3301 Track* track = t.get(); 3302 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3303 track->reset(); 3304 } 3305 3306 // remove all the tracks that need to be... 3307 count = tracksToRemove->size(); 3308 if (CC_UNLIKELY(count)) { 3309 for (size_t i=0 ; i<count ; i++) { 3310 const sp<Track>& track = tracksToRemove->itemAt(i); 3311 mActiveTracks.remove(track); 3312 if (track->mainBuffer() != mMixBuffer) { 3313 chain = getEffectChain_l(track->sessionId()); 3314 if (chain != 0) { 3315 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3316 chain->decActiveTrackCnt(); 3317 } 3318 } 3319 if (track->isTerminated()) { 3320 removeTrack_l(track); 3321 } 3322 } 3323 } 3324 3325 // mix buffer must be cleared if all tracks are connected to an 3326 // effect chain as in this case the mixer will not write to 3327 // mix buffer and track effects will accumulate into it 3328 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3329 // FIXME as a performance optimization, should remember previous zero status 3330 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3331 } 3332 3333 // if any fast tracks, then status is ready 3334 mMixerStatusIgnoringFastTracks = mixerStatus; 3335 if (fastTracks > 0) { 3336 mixerStatus = MIXER_TRACKS_READY; 3337 } 3338 return mixerStatus; 3339} 3340 3341/* 3342The derived values that are cached: 3343 - mixBufferSize from frame count * frame size 3344 - activeSleepTime from activeSleepTimeUs() 3345 - idleSleepTime from idleSleepTimeUs() 3346 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3347 - maxPeriod from frame count and sample rate (MIXER only) 3348 3349The parameters that affect these derived values are: 3350 - frame count 3351 - frame size 3352 - sample rate 3353 - device type: A2DP or not 3354 - device latency 3355 - format: PCM or not 3356 - active sleep time 3357 - idle sleep time 3358*/ 3359 3360void AudioFlinger::PlaybackThread::cacheParameters_l() 3361{ 3362 mixBufferSize = mNormalFrameCount * mFrameSize; 3363 activeSleepTime = activeSleepTimeUs(); 3364 idleSleepTime = idleSleepTimeUs(); 3365} 3366 3367void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3368{ 3369 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3370 this, streamType, mTracks.size()); 3371 Mutex::Autolock _l(mLock); 3372 3373 size_t size = mTracks.size(); 3374 for (size_t i = 0; i < size; i++) { 3375 sp<Track> t = mTracks[i]; 3376 if (t->streamType() == streamType) { 3377 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3378 t->mCblk->cv.signal(); 3379 } 3380 } 3381} 3382 3383// getTrackName_l() must be called with ThreadBase::mLock held 3384int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3385{ 3386 return mAudioMixer->getTrackName(channelMask); 3387} 3388 3389// deleteTrackName_l() must be called with ThreadBase::mLock held 3390void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3391{ 3392 ALOGV("remove track (%d) and delete from mixer", name); 3393 mAudioMixer->deleteTrackName(name); 3394} 3395 3396// checkForNewParameters_l() must be called with ThreadBase::mLock held 3397bool AudioFlinger::MixerThread::checkForNewParameters_l() 3398{ 3399 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3400 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3401 bool reconfig = false; 3402 3403 while (!mNewParameters.isEmpty()) { 3404 3405 if (mFastMixer != NULL) { 3406 FastMixerStateQueue *sq = mFastMixer->sq(); 3407 FastMixerState *state = sq->begin(); 3408 if (!(state->mCommand & FastMixerState::IDLE)) { 3409 previousCommand = state->mCommand; 3410 state->mCommand = FastMixerState::HOT_IDLE; 3411 sq->end(); 3412 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3413 } else { 3414 sq->end(false /*didModify*/); 3415 } 3416 } 3417 3418 status_t status = NO_ERROR; 3419 String8 keyValuePair = mNewParameters[0]; 3420 AudioParameter param = AudioParameter(keyValuePair); 3421 int value; 3422 3423 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3424 reconfig = true; 3425 } 3426 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3427 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3428 status = BAD_VALUE; 3429 } else { 3430 reconfig = true; 3431 } 3432 } 3433 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3434 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3435 status = BAD_VALUE; 3436 } else { 3437 reconfig = true; 3438 } 3439 } 3440 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3441 // do not accept frame count changes if tracks are open as the track buffer 3442 // size depends on frame count and correct behavior would not be guaranteed 3443 // if frame count is changed after track creation 3444 if (!mTracks.isEmpty()) { 3445 status = INVALID_OPERATION; 3446 } else { 3447 reconfig = true; 3448 } 3449 } 3450 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3451#ifdef ADD_BATTERY_DATA 3452 // when changing the audio output device, call addBatteryData to notify 3453 // the change 3454 if (mDevice != value) { 3455 uint32_t params = 0; 3456 // check whether speaker is on 3457 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3458 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3459 } 3460 3461 audio_devices_t deviceWithoutSpeaker 3462 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3463 // check if any other device (except speaker) is on 3464 if (value & deviceWithoutSpeaker ) { 3465 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3466 } 3467 3468 if (params != 0) { 3469 addBatteryData(params); 3470 } 3471 } 3472#endif 3473 3474 // forward device change to effects that have requested to be 3475 // aware of attached audio device. 3476 mDevice = value; 3477 for (size_t i = 0; i < mEffectChains.size(); i++) { 3478 mEffectChains[i]->setDevice_l(mDevice); 3479 } 3480 } 3481 3482 if (status == NO_ERROR) { 3483 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3484 keyValuePair.string()); 3485 if (!mStandby && status == INVALID_OPERATION) { 3486 mOutput->stream->common.standby(&mOutput->stream->common); 3487 mStandby = true; 3488 mBytesWritten = 0; 3489 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3490 keyValuePair.string()); 3491 } 3492 if (status == NO_ERROR && reconfig) { 3493 delete mAudioMixer; 3494 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3495 mAudioMixer = NULL; 3496 readOutputParameters(); 3497 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3498 for (size_t i = 0; i < mTracks.size() ; i++) { 3499 int name = getTrackName_l(mTracks[i]->mChannelMask); 3500 if (name < 0) break; 3501 mTracks[i]->mName = name; 3502 // limit track sample rate to 2 x new output sample rate 3503 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3504 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3505 } 3506 } 3507 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3508 } 3509 } 3510 3511 mNewParameters.removeAt(0); 3512 3513 mParamStatus = status; 3514 mParamCond.signal(); 3515 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3516 // already timed out waiting for the status and will never signal the condition. 3517 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3518 } 3519 3520 if (!(previousCommand & FastMixerState::IDLE)) { 3521 ALOG_ASSERT(mFastMixer != NULL); 3522 FastMixerStateQueue *sq = mFastMixer->sq(); 3523 FastMixerState *state = sq->begin(); 3524 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3525 state->mCommand = previousCommand; 3526 sq->end(); 3527 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3528 } 3529 3530 return reconfig; 3531} 3532 3533void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3534{ 3535 const size_t SIZE = 256; 3536 char buffer[SIZE]; 3537 String8 result; 3538 3539 PlaybackThread::dumpInternals(fd, args); 3540 3541 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3542 result.append(buffer); 3543 write(fd, result.string(), result.size()); 3544 3545 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3546 FastMixerDumpState copy = mFastMixerDumpState; 3547 copy.dump(fd); 3548 3549#ifdef STATE_QUEUE_DUMP 3550 // Similar for state queue 3551 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3552 observerCopy.dump(fd); 3553 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3554 mutatorCopy.dump(fd); 3555#endif 3556 3557 // Write the tee output to a .wav file 3558 NBAIO_Source *teeSource = mTeeSource.get(); 3559 if (teeSource != NULL) { 3560 char teePath[64]; 3561 struct timeval tv; 3562 gettimeofday(&tv, NULL); 3563 struct tm tm; 3564 localtime_r(&tv.tv_sec, &tm); 3565 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3566 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3567 if (teeFd >= 0) { 3568 char wavHeader[44]; 3569 memcpy(wavHeader, 3570 "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", 3571 sizeof(wavHeader)); 3572 NBAIO_Format format = teeSource->format(); 3573 unsigned channelCount = Format_channelCount(format); 3574 ALOG_ASSERT(channelCount <= FCC_2); 3575 unsigned sampleRate = Format_sampleRate(format); 3576 wavHeader[22] = channelCount; // number of channels 3577 wavHeader[24] = sampleRate; // sample rate 3578 wavHeader[25] = sampleRate >> 8; 3579 wavHeader[32] = channelCount * 2; // block alignment 3580 write(teeFd, wavHeader, sizeof(wavHeader)); 3581 size_t total = 0; 3582 bool firstRead = true; 3583 for (;;) { 3584#define TEE_SINK_READ 1024 3585 short buffer[TEE_SINK_READ * FCC_2]; 3586 size_t count = TEE_SINK_READ; 3587 ssize_t actual = teeSource->read(buffer, count, 3588 AudioBufferProvider::kInvalidPTS); 3589 bool wasFirstRead = firstRead; 3590 firstRead = false; 3591 if (actual <= 0) { 3592 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3593 continue; 3594 } 3595 break; 3596 } 3597 ALOG_ASSERT(actual <= (ssize_t)count); 3598 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3599 total += actual; 3600 } 3601 lseek(teeFd, (off_t) 4, SEEK_SET); 3602 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3603 write(teeFd, &temp, sizeof(temp)); 3604 lseek(teeFd, (off_t) 40, SEEK_SET); 3605 temp = total * channelCount * sizeof(short); 3606 write(teeFd, &temp, sizeof(temp)); 3607 close(teeFd); 3608 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3609 } else { 3610 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3611 } 3612 } 3613 3614 if (mAudioWatchdog != 0) { 3615 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3616 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3617 wdCopy.dump(fd); 3618 } 3619} 3620 3621uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3622{ 3623 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3624} 3625 3626uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3627{ 3628 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3629} 3630 3631void AudioFlinger::MixerThread::cacheParameters_l() 3632{ 3633 PlaybackThread::cacheParameters_l(); 3634 3635 // FIXME: Relaxed timing because of a certain device that can't meet latency 3636 // Should be reduced to 2x after the vendor fixes the driver issue 3637 // increase threshold again due to low power audio mode. The way this warning 3638 // threshold is calculated and its usefulness should be reconsidered anyway. 3639 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3640} 3641 3642// ---------------------------------------------------------------------------- 3643AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3644 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3645 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3646 // mLeftVolFloat, mRightVolFloat 3647{ 3648} 3649 3650AudioFlinger::DirectOutputThread::~DirectOutputThread() 3651{ 3652} 3653 3654AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3655 Vector< sp<Track> > *tracksToRemove 3656) 3657{ 3658 sp<Track> trackToRemove; 3659 3660 mixer_state mixerStatus = MIXER_IDLE; 3661 3662 // find out which tracks need to be processed 3663 if (mActiveTracks.size() != 0) { 3664 sp<Track> t = mActiveTracks[0].promote(); 3665 // The track died recently 3666 if (t == 0) return MIXER_IDLE; 3667 3668 Track* const track = t.get(); 3669 audio_track_cblk_t* cblk = track->cblk(); 3670 3671 // The first time a track is added we wait 3672 // for all its buffers to be filled before processing it 3673 uint32_t minFrames; 3674 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3675 minFrames = mNormalFrameCount; 3676 } else { 3677 minFrames = 1; 3678 } 3679 if ((track->framesReady() >= minFrames) && track->isReady() && 3680 !track->isPaused() && !track->isTerminated()) 3681 { 3682 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3683 3684 if (track->mFillingUpStatus == Track::FS_FILLED) { 3685 track->mFillingUpStatus = Track::FS_ACTIVE; 3686 mLeftVolFloat = mRightVolFloat = 0; 3687 if (track->mState == TrackBase::RESUMING) { 3688 track->mState = TrackBase::ACTIVE; 3689 } 3690 } 3691 3692 // compute volume for this track 3693 float left, right; 3694 if (track->isMuted() || mMasterMute || track->isPausing() || 3695 mStreamTypes[track->streamType()].mute) { 3696 left = right = 0; 3697 if (track->isPausing()) { 3698 track->setPaused(); 3699 } 3700 } else { 3701 float typeVolume = mStreamTypes[track->streamType()].volume; 3702 float v = mMasterVolume * typeVolume; 3703 uint32_t vlr = cblk->getVolumeLR(); 3704 float v_clamped = v * (vlr & 0xFFFF); 3705 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3706 left = v_clamped/MAX_GAIN; 3707 v_clamped = v * (vlr >> 16); 3708 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3709 right = v_clamped/MAX_GAIN; 3710 } 3711 3712 if (left != mLeftVolFloat || right != mRightVolFloat) { 3713 mLeftVolFloat = left; 3714 mRightVolFloat = right; 3715 3716 // Convert volumes from float to 8.24 3717 uint32_t vl = (uint32_t)(left * (1 << 24)); 3718 uint32_t vr = (uint32_t)(right * (1 << 24)); 3719 3720 // Delegate volume control to effect in track effect chain if needed 3721 // only one effect chain can be present on DirectOutputThread, so if 3722 // there is one, the track is connected to it 3723 if (!mEffectChains.isEmpty()) { 3724 // Do not ramp volume if volume is controlled by effect 3725 mEffectChains[0]->setVolume_l(&vl, &vr); 3726 left = (float)vl / (1 << 24); 3727 right = (float)vr / (1 << 24); 3728 } 3729 mOutput->stream->set_volume(mOutput->stream, left, right); 3730 } 3731 3732 // reset retry count 3733 track->mRetryCount = kMaxTrackRetriesDirect; 3734 mActiveTrack = t; 3735 mixerStatus = MIXER_TRACKS_READY; 3736 } else { 3737 // clear effect chain input buffer if an active track underruns to avoid sending 3738 // previous audio buffer again to effects 3739 if (!mEffectChains.isEmpty()) { 3740 mEffectChains[0]->clearInputBuffer(); 3741 } 3742 3743 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3744 if ((track->sharedBuffer() != 0) || 3745 track->isStopped() || track->isPaused()) { 3746 // We have consumed all the buffers of this track. 3747 // Remove it from the list of active tracks. 3748 // TODO: implement behavior for compressed audio 3749 size_t audioHALFrames = 3750 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3751 size_t framesWritten = 3752 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3753 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3754 if (track->isStopped()) { 3755 track->reset(); 3756 } 3757 trackToRemove = track; 3758 } 3759 } else { 3760 // No buffers for this track. Give it a few chances to 3761 // fill a buffer, then remove it from active list. 3762 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3763 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3764 trackToRemove = track; 3765 } else { 3766 mixerStatus = MIXER_TRACKS_ENABLED; 3767 } 3768 } 3769 } 3770 } 3771 3772 // FIXME merge this with similar code for removing multiple tracks 3773 // remove all the tracks that need to be... 3774 if (CC_UNLIKELY(trackToRemove != 0)) { 3775 tracksToRemove->add(trackToRemove); 3776 mActiveTracks.remove(trackToRemove); 3777 if (!mEffectChains.isEmpty()) { 3778 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3779 trackToRemove->sessionId()); 3780 mEffectChains[0]->decActiveTrackCnt(); 3781 } 3782 if (trackToRemove->isTerminated()) { 3783 removeTrack_l(trackToRemove); 3784 } 3785 } 3786 3787 return mixerStatus; 3788} 3789 3790void AudioFlinger::DirectOutputThread::threadLoop_mix() 3791{ 3792 AudioBufferProvider::Buffer buffer; 3793 size_t frameCount = mFrameCount; 3794 int8_t *curBuf = (int8_t *)mMixBuffer; 3795 // output audio to hardware 3796 while (frameCount) { 3797 buffer.frameCount = frameCount; 3798 mActiveTrack->getNextBuffer(&buffer); 3799 if (CC_UNLIKELY(buffer.raw == NULL)) { 3800 memset(curBuf, 0, frameCount * mFrameSize); 3801 break; 3802 } 3803 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3804 frameCount -= buffer.frameCount; 3805 curBuf += buffer.frameCount * mFrameSize; 3806 mActiveTrack->releaseBuffer(&buffer); 3807 } 3808 sleepTime = 0; 3809 standbyTime = systemTime() + standbyDelay; 3810 mActiveTrack.clear(); 3811 3812} 3813 3814void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3815{ 3816 if (sleepTime == 0) { 3817 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3818 sleepTime = activeSleepTime; 3819 } else { 3820 sleepTime = idleSleepTime; 3821 } 3822 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3823 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3824 sleepTime = 0; 3825 } 3826} 3827 3828// getTrackName_l() must be called with ThreadBase::mLock held 3829int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3830{ 3831 return 0; 3832} 3833 3834// deleteTrackName_l() must be called with ThreadBase::mLock held 3835void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3836{ 3837} 3838 3839// checkForNewParameters_l() must be called with ThreadBase::mLock held 3840bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3841{ 3842 bool reconfig = false; 3843 3844 while (!mNewParameters.isEmpty()) { 3845 status_t status = NO_ERROR; 3846 String8 keyValuePair = mNewParameters[0]; 3847 AudioParameter param = AudioParameter(keyValuePair); 3848 int value; 3849 3850 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3851 // do not accept frame count changes if tracks are open as the track buffer 3852 // size depends on frame count and correct behavior would not be garantied 3853 // if frame count is changed after track creation 3854 if (!mTracks.isEmpty()) { 3855 status = INVALID_OPERATION; 3856 } else { 3857 reconfig = true; 3858 } 3859 } 3860 if (status == NO_ERROR) { 3861 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3862 keyValuePair.string()); 3863 if (!mStandby && status == INVALID_OPERATION) { 3864 mOutput->stream->common.standby(&mOutput->stream->common); 3865 mStandby = true; 3866 mBytesWritten = 0; 3867 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3868 keyValuePair.string()); 3869 } 3870 if (status == NO_ERROR && reconfig) { 3871 readOutputParameters(); 3872 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3873 } 3874 } 3875 3876 mNewParameters.removeAt(0); 3877 3878 mParamStatus = status; 3879 mParamCond.signal(); 3880 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3881 // already timed out waiting for the status and will never signal the condition. 3882 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3883 } 3884 return reconfig; 3885} 3886 3887uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3888{ 3889 uint32_t time; 3890 if (audio_is_linear_pcm(mFormat)) { 3891 time = PlaybackThread::activeSleepTimeUs(); 3892 } else { 3893 time = 10000; 3894 } 3895 return time; 3896} 3897 3898uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3899{ 3900 uint32_t time; 3901 if (audio_is_linear_pcm(mFormat)) { 3902 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3903 } else { 3904 time = 10000; 3905 } 3906 return time; 3907} 3908 3909uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3910{ 3911 uint32_t time; 3912 if (audio_is_linear_pcm(mFormat)) { 3913 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3914 } else { 3915 time = 10000; 3916 } 3917 return time; 3918} 3919 3920void AudioFlinger::DirectOutputThread::cacheParameters_l() 3921{ 3922 PlaybackThread::cacheParameters_l(); 3923 3924 // use shorter standby delay as on normal output to release 3925 // hardware resources as soon as possible 3926 standbyDelay = microseconds(activeSleepTime*2); 3927} 3928 3929// ---------------------------------------------------------------------------- 3930 3931AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3932 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3933 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3934 mWaitTimeMs(UINT_MAX) 3935{ 3936 addOutputTrack(mainThread); 3937} 3938 3939AudioFlinger::DuplicatingThread::~DuplicatingThread() 3940{ 3941 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3942 mOutputTracks[i]->destroy(); 3943 } 3944} 3945 3946void AudioFlinger::DuplicatingThread::threadLoop_mix() 3947{ 3948 // mix buffers... 3949 if (outputsReady(outputTracks)) { 3950 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3951 } else { 3952 memset(mMixBuffer, 0, mixBufferSize); 3953 } 3954 sleepTime = 0; 3955 writeFrames = mNormalFrameCount; 3956 standbyTime = systemTime() + standbyDelay; 3957} 3958 3959void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3960{ 3961 if (sleepTime == 0) { 3962 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3963 sleepTime = activeSleepTime; 3964 } else { 3965 sleepTime = idleSleepTime; 3966 } 3967 } else if (mBytesWritten != 0) { 3968 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3969 writeFrames = mNormalFrameCount; 3970 memset(mMixBuffer, 0, mixBufferSize); 3971 } else { 3972 // flush remaining overflow buffers in output tracks 3973 writeFrames = 0; 3974 } 3975 sleepTime = 0; 3976 } 3977} 3978 3979void AudioFlinger::DuplicatingThread::threadLoop_write() 3980{ 3981 for (size_t i = 0; i < outputTracks.size(); i++) { 3982 outputTracks[i]->write(mMixBuffer, writeFrames); 3983 } 3984 mBytesWritten += mixBufferSize; 3985} 3986 3987void AudioFlinger::DuplicatingThread::threadLoop_standby() 3988{ 3989 // DuplicatingThread implements standby by stopping all tracks 3990 for (size_t i = 0; i < outputTracks.size(); i++) { 3991 outputTracks[i]->stop(); 3992 } 3993} 3994 3995void AudioFlinger::DuplicatingThread::saveOutputTracks() 3996{ 3997 outputTracks = mOutputTracks; 3998} 3999 4000void AudioFlinger::DuplicatingThread::clearOutputTracks() 4001{ 4002 outputTracks.clear(); 4003} 4004 4005void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4006{ 4007 Mutex::Autolock _l(mLock); 4008 // FIXME explain this formula 4009 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4010 OutputTrack *outputTrack = new OutputTrack(thread, 4011 this, 4012 mSampleRate, 4013 mFormat, 4014 mChannelMask, 4015 frameCount); 4016 if (outputTrack->cblk() != NULL) { 4017 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4018 mOutputTracks.add(outputTrack); 4019 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4020 updateWaitTime_l(); 4021 } 4022} 4023 4024void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4025{ 4026 Mutex::Autolock _l(mLock); 4027 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4028 if (mOutputTracks[i]->thread() == thread) { 4029 mOutputTracks[i]->destroy(); 4030 mOutputTracks.removeAt(i); 4031 updateWaitTime_l(); 4032 return; 4033 } 4034 } 4035 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4036} 4037 4038// caller must hold mLock 4039void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4040{ 4041 mWaitTimeMs = UINT_MAX; 4042 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4043 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4044 if (strong != 0) { 4045 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4046 if (waitTimeMs < mWaitTimeMs) { 4047 mWaitTimeMs = waitTimeMs; 4048 } 4049 } 4050 } 4051} 4052 4053 4054bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4055{ 4056 for (size_t i = 0; i < outputTracks.size(); i++) { 4057 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4058 if (thread == 0) { 4059 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4060 return false; 4061 } 4062 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4063 // see note at standby() declaration 4064 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4065 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4066 return false; 4067 } 4068 } 4069 return true; 4070} 4071 4072uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4073{ 4074 return (mWaitTimeMs * 1000) / 2; 4075} 4076 4077void AudioFlinger::DuplicatingThread::cacheParameters_l() 4078{ 4079 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4080 updateWaitTime_l(); 4081 4082 MixerThread::cacheParameters_l(); 4083} 4084 4085// ---------------------------------------------------------------------------- 4086 4087// TrackBase constructor must be called with AudioFlinger::mLock held 4088AudioFlinger::ThreadBase::TrackBase::TrackBase( 4089 ThreadBase *thread, 4090 const sp<Client>& client, 4091 uint32_t sampleRate, 4092 audio_format_t format, 4093 audio_channel_mask_t channelMask, 4094 int frameCount, 4095 const sp<IMemory>& sharedBuffer, 4096 int sessionId) 4097 : RefBase(), 4098 mThread(thread), 4099 mClient(client), 4100 mCblk(NULL), 4101 // mBuffer 4102 // mBufferEnd 4103 mFrameCount(0), 4104 mState(IDLE), 4105 mSampleRate(sampleRate), 4106 mFormat(format), 4107 mStepServerFailed(false), 4108 mSessionId(sessionId) 4109 // mChannelCount 4110 // mChannelMask 4111{ 4112 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4113 4114 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4115 size_t size = sizeof(audio_track_cblk_t); 4116 uint8_t channelCount = popcount(channelMask); 4117 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4118 if (sharedBuffer == 0) { 4119 size += bufferSize; 4120 } 4121 4122 if (client != NULL) { 4123 mCblkMemory = client->heap()->allocate(size); 4124 if (mCblkMemory != 0) { 4125 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4126 if (mCblk != NULL) { // construct the shared structure in-place. 4127 new(mCblk) audio_track_cblk_t(); 4128 // clear all buffers 4129 mCblk->frameCount = frameCount; 4130 mCblk->sampleRate = sampleRate; 4131// uncomment the following lines to quickly test 32-bit wraparound 4132// mCblk->user = 0xffff0000; 4133// mCblk->server = 0xffff0000; 4134// mCblk->userBase = 0xffff0000; 4135// mCblk->serverBase = 0xffff0000; 4136 mChannelCount = channelCount; 4137 mChannelMask = channelMask; 4138 if (sharedBuffer == 0) { 4139 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4140 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4141 // Force underrun condition to avoid false underrun callback until first data is 4142 // written to buffer (other flags are cleared) 4143 mCblk->flags = CBLK_UNDERRUN_ON; 4144 } else { 4145 mBuffer = sharedBuffer->pointer(); 4146 } 4147 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4148 } 4149 } else { 4150 ALOGE("not enough memory for AudioTrack size=%u", size); 4151 client->heap()->dump("AudioTrack"); 4152 return; 4153 } 4154 } else { 4155 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4156 // construct the shared structure in-place. 4157 new(mCblk) audio_track_cblk_t(); 4158 // clear all buffers 4159 mCblk->frameCount = frameCount; 4160 mCblk->sampleRate = sampleRate; 4161// uncomment the following lines to quickly test 32-bit wraparound 4162// mCblk->user = 0xffff0000; 4163// mCblk->server = 0xffff0000; 4164// mCblk->userBase = 0xffff0000; 4165// mCblk->serverBase = 0xffff0000; 4166 mChannelCount = channelCount; 4167 mChannelMask = channelMask; 4168 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4169 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4170 // Force underrun condition to avoid false underrun callback until first data is 4171 // written to buffer (other flags are cleared) 4172 mCblk->flags = CBLK_UNDERRUN_ON; 4173 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4174 } 4175} 4176 4177AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4178{ 4179 if (mCblk != NULL) { 4180 if (mClient == 0) { 4181 delete mCblk; 4182 } else { 4183 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4184 } 4185 } 4186 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4187 if (mClient != 0) { 4188 // Client destructor must run with AudioFlinger mutex locked 4189 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4190 // If the client's reference count drops to zero, the associated destructor 4191 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4192 // relying on the automatic clear() at end of scope. 4193 mClient.clear(); 4194 } 4195} 4196 4197// AudioBufferProvider interface 4198// getNextBuffer() = 0; 4199// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4200void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4201{ 4202 buffer->raw = NULL; 4203 mFrameCount = buffer->frameCount; 4204 // FIXME See note at getNextBuffer() 4205 (void) step(); // ignore return value of step() 4206 buffer->frameCount = 0; 4207} 4208 4209bool AudioFlinger::ThreadBase::TrackBase::step() { 4210 bool result; 4211 audio_track_cblk_t* cblk = this->cblk(); 4212 4213 result = cblk->stepServer(mFrameCount); 4214 if (!result) { 4215 ALOGV("stepServer failed acquiring cblk mutex"); 4216 mStepServerFailed = true; 4217 } 4218 return result; 4219} 4220 4221void AudioFlinger::ThreadBase::TrackBase::reset() { 4222 audio_track_cblk_t* cblk = this->cblk(); 4223 4224 cblk->user = 0; 4225 cblk->server = 0; 4226 cblk->userBase = 0; 4227 cblk->serverBase = 0; 4228 mStepServerFailed = false; 4229 ALOGV("TrackBase::reset"); 4230} 4231 4232int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4233 return (int)mCblk->sampleRate; 4234} 4235 4236void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4237 audio_track_cblk_t* cblk = this->cblk(); 4238 size_t frameSize = cblk->frameSize; 4239 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4240 int8_t *bufferEnd = bufferStart + frames * frameSize; 4241 4242 // Check validity of returned pointer in case the track control block would have been corrupted. 4243 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4244 "TrackBase::getBuffer buffer out of range:\n" 4245 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4246 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4247 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4248 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4249 4250 return bufferStart; 4251} 4252 4253status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4254{ 4255 mSyncEvents.add(event); 4256 return NO_ERROR; 4257} 4258 4259// ---------------------------------------------------------------------------- 4260 4261// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4262AudioFlinger::PlaybackThread::Track::Track( 4263 PlaybackThread *thread, 4264 const sp<Client>& client, 4265 audio_stream_type_t streamType, 4266 uint32_t sampleRate, 4267 audio_format_t format, 4268 audio_channel_mask_t channelMask, 4269 int frameCount, 4270 const sp<IMemory>& sharedBuffer, 4271 int sessionId, 4272 IAudioFlinger::track_flags_t flags) 4273 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4274 mMute(false), 4275 mFillingUpStatus(FS_INVALID), 4276 // mRetryCount initialized later when needed 4277 mSharedBuffer(sharedBuffer), 4278 mStreamType(streamType), 4279 mName(-1), // see note below 4280 mMainBuffer(thread->mixBuffer()), 4281 mAuxBuffer(NULL), 4282 mAuxEffectId(0), mHasVolumeController(false), 4283 mPresentationCompleteFrames(0), 4284 mFlags(flags), 4285 mFastIndex(-1), 4286 mUnderrunCount(0), 4287 mCachedVolume(1.0) 4288{ 4289 if (mCblk != NULL) { 4290 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4291 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4292 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4293 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4294 mName = thread->getTrackName_l(channelMask); 4295 mCblk->mName = mName; 4296 if (mName < 0) { 4297 ALOGE("no more track names available"); 4298 return; 4299 } 4300 // only allocate a fast track index if we were able to allocate a normal track name 4301 if (flags & IAudioFlinger::TRACK_FAST) { 4302 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4303 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4304 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4305 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4306 // FIXME This is too eager. We allocate a fast track index before the 4307 // fast track becomes active. Since fast tracks are a scarce resource, 4308 // this means we are potentially denying other more important fast tracks from 4309 // being created. It would be better to allocate the index dynamically. 4310 mFastIndex = i; 4311 mCblk->mName = i; 4312 // Read the initial underruns because this field is never cleared by the fast mixer 4313 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4314 thread->mFastTrackAvailMask &= ~(1 << i); 4315 } 4316 } 4317 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4318} 4319 4320AudioFlinger::PlaybackThread::Track::~Track() 4321{ 4322 ALOGV("PlaybackThread::Track destructor"); 4323} 4324 4325void AudioFlinger::PlaybackThread::Track::destroy() 4326{ 4327 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4328 // by removing it from mTracks vector, so there is a risk that this Tracks's 4329 // destructor is called. As the destructor needs to lock mLock, 4330 // we must acquire a strong reference on this Track before locking mLock 4331 // here so that the destructor is called only when exiting this function. 4332 // On the other hand, as long as Track::destroy() is only called by 4333 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4334 // this Track with its member mTrack. 4335 sp<Track> keep(this); 4336 { // scope for mLock 4337 sp<ThreadBase> thread = mThread.promote(); 4338 if (thread != 0) { 4339 if (!isOutputTrack()) { 4340 if (mState == ACTIVE || mState == RESUMING) { 4341 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4342 4343#ifdef ADD_BATTERY_DATA 4344 // to track the speaker usage 4345 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4346#endif 4347 } 4348 AudioSystem::releaseOutput(thread->id()); 4349 } 4350 Mutex::Autolock _l(thread->mLock); 4351 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4352 playbackThread->destroyTrack_l(this); 4353 } 4354 } 4355} 4356 4357/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4358{ 4359 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4360 " Server User Main buf Aux Buf Flags Underruns\n"); 4361} 4362 4363void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4364{ 4365 uint32_t vlr = mCblk->getVolumeLR(); 4366 if (isFastTrack()) { 4367 sprintf(buffer, " F %2d", mFastIndex); 4368 } else { 4369 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4370 } 4371 track_state state = mState; 4372 char stateChar; 4373 switch (state) { 4374 case IDLE: 4375 stateChar = 'I'; 4376 break; 4377 case TERMINATED: 4378 stateChar = 'T'; 4379 break; 4380 case STOPPING_1: 4381 stateChar = 's'; 4382 break; 4383 case STOPPING_2: 4384 stateChar = '5'; 4385 break; 4386 case STOPPED: 4387 stateChar = 'S'; 4388 break; 4389 case RESUMING: 4390 stateChar = 'R'; 4391 break; 4392 case ACTIVE: 4393 stateChar = 'A'; 4394 break; 4395 case PAUSING: 4396 stateChar = 'p'; 4397 break; 4398 case PAUSED: 4399 stateChar = 'P'; 4400 break; 4401 case FLUSHED: 4402 stateChar = 'F'; 4403 break; 4404 default: 4405 stateChar = '?'; 4406 break; 4407 } 4408 char nowInUnderrun; 4409 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4410 case UNDERRUN_FULL: 4411 nowInUnderrun = ' '; 4412 break; 4413 case UNDERRUN_PARTIAL: 4414 nowInUnderrun = '<'; 4415 break; 4416 case UNDERRUN_EMPTY: 4417 nowInUnderrun = '*'; 4418 break; 4419 default: 4420 nowInUnderrun = '?'; 4421 break; 4422 } 4423 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4424 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4425 (mClient == 0) ? getpid_cached : mClient->pid(), 4426 mStreamType, 4427 mFormat, 4428 mChannelMask, 4429 mSessionId, 4430 mFrameCount, 4431 mCblk->frameCount, 4432 stateChar, 4433 mMute, 4434 mFillingUpStatus, 4435 mCblk->sampleRate, 4436 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4437 20.0 * log10((vlr >> 16) / 4096.0), 4438 mCblk->server, 4439 mCblk->user, 4440 (int)mMainBuffer, 4441 (int)mAuxBuffer, 4442 mCblk->flags, 4443 mUnderrunCount, 4444 nowInUnderrun); 4445} 4446 4447// AudioBufferProvider interface 4448status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4449 AudioBufferProvider::Buffer* buffer, int64_t pts) 4450{ 4451 audio_track_cblk_t* cblk = this->cblk(); 4452 uint32_t framesReady; 4453 uint32_t framesReq = buffer->frameCount; 4454 4455 // Check if last stepServer failed, try to step now 4456 if (mStepServerFailed) { 4457 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4458 // Since the fast mixer is higher priority than client callback thread, 4459 // it does not result in priority inversion for client. 4460 // But a non-blocking solution would be preferable to avoid 4461 // fast mixer being unable to tryLock(), and 4462 // to avoid the extra context switches if the client wakes up, 4463 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4464 if (!step()) goto getNextBuffer_exit; 4465 ALOGV("stepServer recovered"); 4466 mStepServerFailed = false; 4467 } 4468 4469 // FIXME Same as above 4470 framesReady = cblk->framesReady(); 4471 4472 if (CC_LIKELY(framesReady)) { 4473 uint32_t s = cblk->server; 4474 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4475 4476 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4477 if (framesReq > framesReady) { 4478 framesReq = framesReady; 4479 } 4480 if (framesReq > bufferEnd - s) { 4481 framesReq = bufferEnd - s; 4482 } 4483 4484 buffer->raw = getBuffer(s, framesReq); 4485 buffer->frameCount = framesReq; 4486 return NO_ERROR; 4487 } 4488 4489getNextBuffer_exit: 4490 buffer->raw = NULL; 4491 buffer->frameCount = 0; 4492 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4493 return NOT_ENOUGH_DATA; 4494} 4495 4496// Note that framesReady() takes a mutex on the control block using tryLock(). 4497// This could result in priority inversion if framesReady() is called by the normal mixer, 4498// as the normal mixer thread runs at lower 4499// priority than the client's callback thread: there is a short window within framesReady() 4500// during which the normal mixer could be preempted, and the client callback would block. 4501// Another problem can occur if framesReady() is called by the fast mixer: 4502// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4503// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4504size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4505 return mCblk->framesReady(); 4506} 4507 4508// Don't call for fast tracks; the framesReady() could result in priority inversion 4509bool AudioFlinger::PlaybackThread::Track::isReady() const { 4510 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4511 4512 if (framesReady() >= mCblk->frameCount || 4513 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4514 mFillingUpStatus = FS_FILLED; 4515 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4516 return true; 4517 } 4518 return false; 4519} 4520 4521status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4522 int triggerSession) 4523{ 4524 status_t status = NO_ERROR; 4525 ALOGV("start(%d), calling pid %d session %d", 4526 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4527 4528 sp<ThreadBase> thread = mThread.promote(); 4529 if (thread != 0) { 4530 Mutex::Autolock _l(thread->mLock); 4531 track_state state = mState; 4532 // here the track could be either new, or restarted 4533 // in both cases "unstop" the track 4534 if (mState == PAUSED) { 4535 mState = TrackBase::RESUMING; 4536 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4537 } else { 4538 mState = TrackBase::ACTIVE; 4539 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4540 } 4541 4542 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4543 thread->mLock.unlock(); 4544 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4545 thread->mLock.lock(); 4546 4547#ifdef ADD_BATTERY_DATA 4548 // to track the speaker usage 4549 if (status == NO_ERROR) { 4550 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4551 } 4552#endif 4553 } 4554 if (status == NO_ERROR) { 4555 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4556 playbackThread->addTrack_l(this); 4557 } else { 4558 mState = state; 4559 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4560 } 4561 } else { 4562 status = BAD_VALUE; 4563 } 4564 return status; 4565} 4566 4567void AudioFlinger::PlaybackThread::Track::stop() 4568{ 4569 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4570 sp<ThreadBase> thread = mThread.promote(); 4571 if (thread != 0) { 4572 Mutex::Autolock _l(thread->mLock); 4573 track_state state = mState; 4574 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4575 // If the track is not active (PAUSED and buffers full), flush buffers 4576 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4577 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4578 reset(); 4579 mState = STOPPED; 4580 } else if (!isFastTrack()) { 4581 mState = STOPPED; 4582 } else { 4583 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4584 // and then to STOPPED and reset() when presentation is complete 4585 mState = STOPPING_1; 4586 } 4587 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4588 } 4589 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4590 thread->mLock.unlock(); 4591 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4592 thread->mLock.lock(); 4593 4594#ifdef ADD_BATTERY_DATA 4595 // to track the speaker usage 4596 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4597#endif 4598 } 4599 } 4600} 4601 4602void AudioFlinger::PlaybackThread::Track::pause() 4603{ 4604 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4605 sp<ThreadBase> thread = mThread.promote(); 4606 if (thread != 0) { 4607 Mutex::Autolock _l(thread->mLock); 4608 if (mState == ACTIVE || mState == RESUMING) { 4609 mState = PAUSING; 4610 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4611 if (!isOutputTrack()) { 4612 thread->mLock.unlock(); 4613 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4614 thread->mLock.lock(); 4615 4616#ifdef ADD_BATTERY_DATA 4617 // to track the speaker usage 4618 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4619#endif 4620 } 4621 } 4622 } 4623} 4624 4625void AudioFlinger::PlaybackThread::Track::flush() 4626{ 4627 ALOGV("flush(%d)", mName); 4628 sp<ThreadBase> thread = mThread.promote(); 4629 if (thread != 0) { 4630 Mutex::Autolock _l(thread->mLock); 4631 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4632 mState != PAUSING) { 4633 return; 4634 } 4635 // No point remaining in PAUSED state after a flush => go to 4636 // FLUSHED state 4637 mState = FLUSHED; 4638 // do not reset the track if it is still in the process of being stopped or paused. 4639 // this will be done by prepareTracks_l() when the track is stopped. 4640 // prepareTracks_l() will see mState == FLUSHED, then 4641 // remove from active track list, reset(), and trigger presentation complete 4642 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4643 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4644 reset(); 4645 } 4646 } 4647} 4648 4649void AudioFlinger::PlaybackThread::Track::reset() 4650{ 4651 // Do not reset twice to avoid discarding data written just after a flush and before 4652 // the audioflinger thread detects the track is stopped. 4653 if (!mResetDone) { 4654 TrackBase::reset(); 4655 // Force underrun condition to avoid false underrun callback until first data is 4656 // written to buffer 4657 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4658 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4659 mFillingUpStatus = FS_FILLING; 4660 mResetDone = true; 4661 if (mState == FLUSHED) { 4662 mState = IDLE; 4663 } 4664 } 4665} 4666 4667void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4668{ 4669 mMute = muted; 4670} 4671 4672status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4673{ 4674 status_t status = DEAD_OBJECT; 4675 sp<ThreadBase> thread = mThread.promote(); 4676 if (thread != 0) { 4677 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4678 sp<AudioFlinger> af = mClient->audioFlinger(); 4679 4680 Mutex::Autolock _l(af->mLock); 4681 4682 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4683 4684 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4685 Mutex::Autolock _dl(playbackThread->mLock); 4686 Mutex::Autolock _sl(srcThread->mLock); 4687 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4688 if (chain == 0) { 4689 return INVALID_OPERATION; 4690 } 4691 4692 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4693 if (effect == 0) { 4694 return INVALID_OPERATION; 4695 } 4696 srcThread->removeEffect_l(effect); 4697 playbackThread->addEffect_l(effect); 4698 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4699 if (effect->state() == EffectModule::ACTIVE || 4700 effect->state() == EffectModule::STOPPING) { 4701 effect->start(); 4702 } 4703 4704 sp<EffectChain> dstChain = effect->chain().promote(); 4705 if (dstChain == 0) { 4706 srcThread->addEffect_l(effect); 4707 return INVALID_OPERATION; 4708 } 4709 AudioSystem::unregisterEffect(effect->id()); 4710 AudioSystem::registerEffect(&effect->desc(), 4711 srcThread->id(), 4712 dstChain->strategy(), 4713 AUDIO_SESSION_OUTPUT_MIX, 4714 effect->id()); 4715 } 4716 status = playbackThread->attachAuxEffect(this, EffectId); 4717 } 4718 return status; 4719} 4720 4721void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4722{ 4723 mAuxEffectId = EffectId; 4724 mAuxBuffer = buffer; 4725} 4726 4727bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4728 size_t audioHalFrames) 4729{ 4730 // a track is considered presented when the total number of frames written to audio HAL 4731 // corresponds to the number of frames written when presentationComplete() is called for the 4732 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4733 if (mPresentationCompleteFrames == 0) { 4734 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4735 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4736 mPresentationCompleteFrames, audioHalFrames); 4737 } 4738 if (framesWritten >= mPresentationCompleteFrames) { 4739 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4740 mSessionId, framesWritten); 4741 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4742 return true; 4743 } 4744 return false; 4745} 4746 4747void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4748{ 4749 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4750 if (mSyncEvents[i]->type() == type) { 4751 mSyncEvents[i]->trigger(); 4752 mSyncEvents.removeAt(i); 4753 i--; 4754 } 4755 } 4756} 4757 4758// implement VolumeBufferProvider interface 4759 4760uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4761{ 4762 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4763 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4764 uint32_t vlr = mCblk->getVolumeLR(); 4765 uint32_t vl = vlr & 0xFFFF; 4766 uint32_t vr = vlr >> 16; 4767 // track volumes come from shared memory, so can't be trusted and must be clamped 4768 if (vl > MAX_GAIN_INT) { 4769 vl = MAX_GAIN_INT; 4770 } 4771 if (vr > MAX_GAIN_INT) { 4772 vr = MAX_GAIN_INT; 4773 } 4774 // now apply the cached master volume and stream type volume; 4775 // this is trusted but lacks any synchronization or barrier so may be stale 4776 float v = mCachedVolume; 4777 vl *= v; 4778 vr *= v; 4779 // re-combine into U4.16 4780 vlr = (vr << 16) | (vl & 0xFFFF); 4781 // FIXME look at mute, pause, and stop flags 4782 return vlr; 4783} 4784 4785status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4786{ 4787 if (mState == TERMINATED || mState == PAUSED || 4788 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4789 (mState == STOPPED)))) { 4790 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4791 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4792 event->cancel(); 4793 return INVALID_OPERATION; 4794 } 4795 (void) TrackBase::setSyncEvent(event); 4796 return NO_ERROR; 4797} 4798 4799// timed audio tracks 4800 4801sp<AudioFlinger::PlaybackThread::TimedTrack> 4802AudioFlinger::PlaybackThread::TimedTrack::create( 4803 PlaybackThread *thread, 4804 const sp<Client>& client, 4805 audio_stream_type_t streamType, 4806 uint32_t sampleRate, 4807 audio_format_t format, 4808 audio_channel_mask_t channelMask, 4809 int frameCount, 4810 const sp<IMemory>& sharedBuffer, 4811 int sessionId) { 4812 if (!client->reserveTimedTrack()) 4813 return 0; 4814 4815 return new TimedTrack( 4816 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4817 sharedBuffer, sessionId); 4818} 4819 4820AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4821 PlaybackThread *thread, 4822 const sp<Client>& client, 4823 audio_stream_type_t streamType, 4824 uint32_t sampleRate, 4825 audio_format_t format, 4826 audio_channel_mask_t channelMask, 4827 int frameCount, 4828 const sp<IMemory>& sharedBuffer, 4829 int sessionId) 4830 : Track(thread, client, streamType, sampleRate, format, channelMask, 4831 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4832 mQueueHeadInFlight(false), 4833 mTrimQueueHeadOnRelease(false), 4834 mFramesPendingInQueue(0), 4835 mTimedSilenceBuffer(NULL), 4836 mTimedSilenceBufferSize(0), 4837 mTimedAudioOutputOnTime(false), 4838 mMediaTimeTransformValid(false) 4839{ 4840 LocalClock lc; 4841 mLocalTimeFreq = lc.getLocalFreq(); 4842 4843 mLocalTimeToSampleTransform.a_zero = 0; 4844 mLocalTimeToSampleTransform.b_zero = 0; 4845 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4846 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4847 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4848 &mLocalTimeToSampleTransform.a_to_b_denom); 4849 4850 mMediaTimeToSampleTransform.a_zero = 0; 4851 mMediaTimeToSampleTransform.b_zero = 0; 4852 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4853 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4854 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4855 &mMediaTimeToSampleTransform.a_to_b_denom); 4856} 4857 4858AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4859 mClient->releaseTimedTrack(); 4860 delete [] mTimedSilenceBuffer; 4861} 4862 4863status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4864 size_t size, sp<IMemory>* buffer) { 4865 4866 Mutex::Autolock _l(mTimedBufferQueueLock); 4867 4868 trimTimedBufferQueue_l(); 4869 4870 // lazily initialize the shared memory heap for timed buffers 4871 if (mTimedMemoryDealer == NULL) { 4872 const int kTimedBufferHeapSize = 512 << 10; 4873 4874 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4875 "AudioFlingerTimed"); 4876 if (mTimedMemoryDealer == NULL) 4877 return NO_MEMORY; 4878 } 4879 4880 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4881 if (newBuffer == NULL) { 4882 newBuffer = mTimedMemoryDealer->allocate(size); 4883 if (newBuffer == NULL) 4884 return NO_MEMORY; 4885 } 4886 4887 *buffer = newBuffer; 4888 return NO_ERROR; 4889} 4890 4891// caller must hold mTimedBufferQueueLock 4892void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4893 int64_t mediaTimeNow; 4894 { 4895 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4896 if (!mMediaTimeTransformValid) 4897 return; 4898 4899 int64_t targetTimeNow; 4900 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4901 ? mCCHelper.getCommonTime(&targetTimeNow) 4902 : mCCHelper.getLocalTime(&targetTimeNow); 4903 4904 if (OK != res) 4905 return; 4906 4907 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4908 &mediaTimeNow)) { 4909 return; 4910 } 4911 } 4912 4913 size_t trimEnd; 4914 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4915 int64_t bufEnd; 4916 4917 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4918 // We have a next buffer. Just use its PTS as the PTS of the frame 4919 // following the last frame in this buffer. If the stream is sparse 4920 // (ie, there are deliberate gaps left in the stream which should be 4921 // filled with silence by the TimedAudioTrack), then this can result 4922 // in one extra buffer being left un-trimmed when it could have 4923 // been. In general, this is not typical, and we would rather 4924 // optimized away the TS calculation below for the more common case 4925 // where PTSes are contiguous. 4926 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4927 } else { 4928 // We have no next buffer. Compute the PTS of the frame following 4929 // the last frame in this buffer by computing the duration of of 4930 // this frame in media time units and adding it to the PTS of the 4931 // buffer. 4932 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4933 / mCblk->frameSize; 4934 4935 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4936 &bufEnd)) { 4937 ALOGE("Failed to convert frame count of %lld to media time" 4938 " duration" " (scale factor %d/%u) in %s", 4939 frameCount, 4940 mMediaTimeToSampleTransform.a_to_b_numer, 4941 mMediaTimeToSampleTransform.a_to_b_denom, 4942 __PRETTY_FUNCTION__); 4943 break; 4944 } 4945 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4946 } 4947 4948 if (bufEnd > mediaTimeNow) 4949 break; 4950 4951 // Is the buffer we want to use in the middle of a mix operation right 4952 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4953 // from the mixer which should be coming back shortly. 4954 if (!trimEnd && mQueueHeadInFlight) { 4955 mTrimQueueHeadOnRelease = true; 4956 } 4957 } 4958 4959 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4960 if (trimStart < trimEnd) { 4961 // Update the bookkeeping for framesReady() 4962 for (size_t i = trimStart; i < trimEnd; ++i) { 4963 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4964 } 4965 4966 // Now actually remove the buffers from the queue. 4967 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4968 } 4969} 4970 4971void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4972 const char* logTag) { 4973 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4974 "%s called (reason \"%s\"), but timed buffer queue has no" 4975 " elements to trim.", __FUNCTION__, logTag); 4976 4977 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4978 mTimedBufferQueue.removeAt(0); 4979} 4980 4981void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4982 const TimedBuffer& buf, 4983 const char* logTag) { 4984 uint32_t bufBytes = buf.buffer()->size(); 4985 uint32_t consumedAlready = buf.position(); 4986 4987 ALOG_ASSERT(consumedAlready <= bufBytes, 4988 "Bad bookkeeping while updating frames pending. Timed buffer is" 4989 " only %u bytes long, but claims to have consumed %u" 4990 " bytes. (update reason: \"%s\")", 4991 bufBytes, consumedAlready, logTag); 4992 4993 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4994 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4995 "Bad bookkeeping while updating frames pending. Should have at" 4996 " least %u queued frames, but we think we have only %u. (update" 4997 " reason: \"%s\")", 4998 bufFrames, mFramesPendingInQueue, logTag); 4999 5000 mFramesPendingInQueue -= bufFrames; 5001} 5002 5003status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 5004 const sp<IMemory>& buffer, int64_t pts) { 5005 5006 { 5007 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5008 if (!mMediaTimeTransformValid) 5009 return INVALID_OPERATION; 5010 } 5011 5012 Mutex::Autolock _l(mTimedBufferQueueLock); 5013 5014 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 5015 mFramesPendingInQueue += bufFrames; 5016 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 5017 5018 return NO_ERROR; 5019} 5020 5021status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 5022 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 5023 5024 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5025 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5026 target); 5027 5028 if (!(target == TimedAudioTrack::LOCAL_TIME || 5029 target == TimedAudioTrack::COMMON_TIME)) { 5030 return BAD_VALUE; 5031 } 5032 5033 Mutex::Autolock lock(mMediaTimeTransformLock); 5034 mMediaTimeTransform = xform; 5035 mMediaTimeTransformTarget = target; 5036 mMediaTimeTransformValid = true; 5037 5038 return NO_ERROR; 5039} 5040 5041#define min(a, b) ((a) < (b) ? (a) : (b)) 5042 5043// implementation of getNextBuffer for tracks whose buffers have timestamps 5044status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5045 AudioBufferProvider::Buffer* buffer, int64_t pts) 5046{ 5047 if (pts == AudioBufferProvider::kInvalidPTS) { 5048 buffer->raw = NULL; 5049 buffer->frameCount = 0; 5050 mTimedAudioOutputOnTime = false; 5051 return INVALID_OPERATION; 5052 } 5053 5054 Mutex::Autolock _l(mTimedBufferQueueLock); 5055 5056 ALOG_ASSERT(!mQueueHeadInFlight, 5057 "getNextBuffer called without releaseBuffer!"); 5058 5059 while (true) { 5060 5061 // if we have no timed buffers, then fail 5062 if (mTimedBufferQueue.isEmpty()) { 5063 buffer->raw = NULL; 5064 buffer->frameCount = 0; 5065 return NOT_ENOUGH_DATA; 5066 } 5067 5068 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5069 5070 // calculate the PTS of the head of the timed buffer queue expressed in 5071 // local time 5072 int64_t headLocalPTS; 5073 { 5074 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5075 5076 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5077 5078 if (mMediaTimeTransform.a_to_b_denom == 0) { 5079 // the transform represents a pause, so yield silence 5080 timedYieldSilence_l(buffer->frameCount, buffer); 5081 return NO_ERROR; 5082 } 5083 5084 int64_t transformedPTS; 5085 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5086 &transformedPTS)) { 5087 // the transform failed. this shouldn't happen, but if it does 5088 // then just drop this buffer 5089 ALOGW("timedGetNextBuffer transform failed"); 5090 buffer->raw = NULL; 5091 buffer->frameCount = 0; 5092 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5093 return NO_ERROR; 5094 } 5095 5096 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5097 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5098 &headLocalPTS)) { 5099 buffer->raw = NULL; 5100 buffer->frameCount = 0; 5101 return INVALID_OPERATION; 5102 } 5103 } else { 5104 headLocalPTS = transformedPTS; 5105 } 5106 } 5107 5108 // adjust the head buffer's PTS to reflect the portion of the head buffer 5109 // that has already been consumed 5110 int64_t effectivePTS = headLocalPTS + 5111 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5112 5113 // Calculate the delta in samples between the head of the input buffer 5114 // queue and the start of the next output buffer that will be written. 5115 // If the transformation fails because of over or underflow, it means 5116 // that the sample's position in the output stream is so far out of 5117 // whack that it should just be dropped. 5118 int64_t sampleDelta; 5119 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5120 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5121 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5122 " mix"); 5123 continue; 5124 } 5125 if (!mLocalTimeToSampleTransform.doForwardTransform( 5126 (effectivePTS - pts) << 32, &sampleDelta)) { 5127 ALOGV("*** too late during sample rate transform: dropped buffer"); 5128 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5129 continue; 5130 } 5131 5132 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5133 " sampleDelta=[%d.%08x]", 5134 head.pts(), head.position(), pts, 5135 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5136 + (sampleDelta >> 32)), 5137 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5138 5139 // if the delta between the ideal placement for the next input sample and 5140 // the current output position is within this threshold, then we will 5141 // concatenate the next input samples to the previous output 5142 const int64_t kSampleContinuityThreshold = 5143 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5144 5145 // if this is the first buffer of audio that we're emitting from this track 5146 // then it should be almost exactly on time. 5147 const int64_t kSampleStartupThreshold = 1LL << 32; 5148 5149 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5150 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5151 // the next input is close enough to being on time, so concatenate it 5152 // with the last output 5153 timedYieldSamples_l(buffer); 5154 5155 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5156 head.position(), buffer->frameCount); 5157 return NO_ERROR; 5158 } 5159 5160 // Looks like our output is not on time. Reset our on timed status. 5161 // Next time we mix samples from our input queue, then should be within 5162 // the StartupThreshold. 5163 mTimedAudioOutputOnTime = false; 5164 if (sampleDelta > 0) { 5165 // the gap between the current output position and the proper start of 5166 // the next input sample is too big, so fill it with silence 5167 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5168 5169 timedYieldSilence_l(framesUntilNextInput, buffer); 5170 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5171 return NO_ERROR; 5172 } else { 5173 // the next input sample is late 5174 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5175 size_t onTimeSamplePosition = 5176 head.position() + lateFrames * mCblk->frameSize; 5177 5178 if (onTimeSamplePosition > head.buffer()->size()) { 5179 // all the remaining samples in the head are too late, so 5180 // drop it and move on 5181 ALOGV("*** too late: dropped buffer"); 5182 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5183 continue; 5184 } else { 5185 // skip over the late samples 5186 head.setPosition(onTimeSamplePosition); 5187 5188 // yield the available samples 5189 timedYieldSamples_l(buffer); 5190 5191 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5192 return NO_ERROR; 5193 } 5194 } 5195 } 5196} 5197 5198// Yield samples from the timed buffer queue head up to the given output 5199// buffer's capacity. 5200// 5201// Caller must hold mTimedBufferQueueLock 5202void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5203 AudioBufferProvider::Buffer* buffer) { 5204 5205 const TimedBuffer& head = mTimedBufferQueue[0]; 5206 5207 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5208 head.position()); 5209 5210 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5211 mCblk->frameSize); 5212 size_t framesRequested = buffer->frameCount; 5213 buffer->frameCount = min(framesLeftInHead, framesRequested); 5214 5215 mQueueHeadInFlight = true; 5216 mTimedAudioOutputOnTime = true; 5217} 5218 5219// Yield samples of silence up to the given output buffer's capacity 5220// 5221// Caller must hold mTimedBufferQueueLock 5222void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5223 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5224 5225 // lazily allocate a buffer filled with silence 5226 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5227 delete [] mTimedSilenceBuffer; 5228 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5229 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5230 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5231 } 5232 5233 buffer->raw = mTimedSilenceBuffer; 5234 size_t framesRequested = buffer->frameCount; 5235 buffer->frameCount = min(numFrames, framesRequested); 5236 5237 mTimedAudioOutputOnTime = false; 5238} 5239 5240// AudioBufferProvider interface 5241void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5242 AudioBufferProvider::Buffer* buffer) { 5243 5244 Mutex::Autolock _l(mTimedBufferQueueLock); 5245 5246 // If the buffer which was just released is part of the buffer at the head 5247 // of the queue, be sure to update the amt of the buffer which has been 5248 // consumed. If the buffer being returned is not part of the head of the 5249 // queue, its either because the buffer is part of the silence buffer, or 5250 // because the head of the timed queue was trimmed after the mixer called 5251 // getNextBuffer but before the mixer called releaseBuffer. 5252 if (buffer->raw == mTimedSilenceBuffer) { 5253 ALOG_ASSERT(!mQueueHeadInFlight, 5254 "Queue head in flight during release of silence buffer!"); 5255 goto done; 5256 } 5257 5258 ALOG_ASSERT(mQueueHeadInFlight, 5259 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5260 " head in flight."); 5261 5262 if (mTimedBufferQueue.size()) { 5263 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5264 5265 void* start = head.buffer()->pointer(); 5266 void* end = reinterpret_cast<void*>( 5267 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5268 + head.buffer()->size()); 5269 5270 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5271 "released buffer not within the head of the timed buffer" 5272 " queue; qHead = [%p, %p], released buffer = %p", 5273 start, end, buffer->raw); 5274 5275 head.setPosition(head.position() + 5276 (buffer->frameCount * mCblk->frameSize)); 5277 mQueueHeadInFlight = false; 5278 5279 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5280 "Bad bookkeeping during releaseBuffer! Should have at" 5281 " least %u queued frames, but we think we have only %u", 5282 buffer->frameCount, mFramesPendingInQueue); 5283 5284 mFramesPendingInQueue -= buffer->frameCount; 5285 5286 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5287 || mTrimQueueHeadOnRelease) { 5288 trimTimedBufferQueueHead_l("releaseBuffer"); 5289 mTrimQueueHeadOnRelease = false; 5290 } 5291 } else { 5292 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5293 " buffers in the timed buffer queue"); 5294 } 5295 5296done: 5297 buffer->raw = 0; 5298 buffer->frameCount = 0; 5299} 5300 5301size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5302 Mutex::Autolock _l(mTimedBufferQueueLock); 5303 return mFramesPendingInQueue; 5304} 5305 5306AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5307 : mPTS(0), mPosition(0) {} 5308 5309AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5310 const sp<IMemory>& buffer, int64_t pts) 5311 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5312 5313// ---------------------------------------------------------------------------- 5314 5315// RecordTrack constructor must be called with AudioFlinger::mLock held 5316AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5317 RecordThread *thread, 5318 const sp<Client>& client, 5319 uint32_t sampleRate, 5320 audio_format_t format, 5321 audio_channel_mask_t channelMask, 5322 int frameCount, 5323 int sessionId) 5324 : TrackBase(thread, client, sampleRate, format, 5325 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5326 mOverflow(false) 5327{ 5328 if (mCblk != NULL) { 5329 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5330 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5331 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5332 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5333 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5334 } else { 5335 mCblk->frameSize = sizeof(int8_t); 5336 } 5337 } 5338} 5339 5340AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5341{ 5342 ALOGV("%s", __func__); 5343} 5344 5345// AudioBufferProvider interface 5346status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5347{ 5348 audio_track_cblk_t* cblk = this->cblk(); 5349 uint32_t framesAvail; 5350 uint32_t framesReq = buffer->frameCount; 5351 5352 // Check if last stepServer failed, try to step now 5353 if (mStepServerFailed) { 5354 if (!step()) goto getNextBuffer_exit; 5355 ALOGV("stepServer recovered"); 5356 mStepServerFailed = false; 5357 } 5358 5359 framesAvail = cblk->framesAvailable_l(); 5360 5361 if (CC_LIKELY(framesAvail)) { 5362 uint32_t s = cblk->server; 5363 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5364 5365 if (framesReq > framesAvail) { 5366 framesReq = framesAvail; 5367 } 5368 if (framesReq > bufferEnd - s) { 5369 framesReq = bufferEnd - s; 5370 } 5371 5372 buffer->raw = getBuffer(s, framesReq); 5373 buffer->frameCount = framesReq; 5374 return NO_ERROR; 5375 } 5376 5377getNextBuffer_exit: 5378 buffer->raw = NULL; 5379 buffer->frameCount = 0; 5380 return NOT_ENOUGH_DATA; 5381} 5382 5383status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5384 int triggerSession) 5385{ 5386 sp<ThreadBase> thread = mThread.promote(); 5387 if (thread != 0) { 5388 RecordThread *recordThread = (RecordThread *)thread.get(); 5389 return recordThread->start(this, event, triggerSession); 5390 } else { 5391 return BAD_VALUE; 5392 } 5393} 5394 5395void AudioFlinger::RecordThread::RecordTrack::stop() 5396{ 5397 sp<ThreadBase> thread = mThread.promote(); 5398 if (thread != 0) { 5399 RecordThread *recordThread = (RecordThread *)thread.get(); 5400 recordThread->mLock.lock(); 5401 bool doStop = recordThread->stop_l(this); 5402 if (doStop) { 5403 TrackBase::reset(); 5404 // Force overrun condition to avoid false overrun callback until first data is 5405 // read from buffer 5406 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5407 } 5408 recordThread->mLock.unlock(); 5409 if (doStop) { 5410 AudioSystem::stopInput(recordThread->id()); 5411 } 5412 } 5413} 5414 5415/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5416{ 5417 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5418} 5419 5420void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5421{ 5422 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5423 (mClient == 0) ? getpid_cached : mClient->pid(), 5424 mFormat, 5425 mChannelMask, 5426 mSessionId, 5427 mFrameCount, 5428 mState, 5429 mCblk->sampleRate, 5430 mCblk->server, 5431 mCblk->user); 5432} 5433 5434 5435// ---------------------------------------------------------------------------- 5436 5437AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5438 PlaybackThread *playbackThread, 5439 DuplicatingThread *sourceThread, 5440 uint32_t sampleRate, 5441 audio_format_t format, 5442 audio_channel_mask_t channelMask, 5443 int frameCount) 5444 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5445 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5446 mActive(false), mSourceThread(sourceThread) 5447{ 5448 5449 if (mCblk != NULL) { 5450 mCblk->flags |= CBLK_DIRECTION_OUT; 5451 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5452 mOutBuffer.frameCount = 0; 5453 playbackThread->mTracks.add(this); 5454 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5455 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5456 mCblk, mBuffer, mCblk->buffers, 5457 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5458 } else { 5459 ALOGW("Error creating output track on thread %p", playbackThread); 5460 } 5461} 5462 5463AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5464{ 5465 clearBufferQueue(); 5466} 5467 5468status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5469 int triggerSession) 5470{ 5471 status_t status = Track::start(event, triggerSession); 5472 if (status != NO_ERROR) { 5473 return status; 5474 } 5475 5476 mActive = true; 5477 mRetryCount = 127; 5478 return status; 5479} 5480 5481void AudioFlinger::PlaybackThread::OutputTrack::stop() 5482{ 5483 Track::stop(); 5484 clearBufferQueue(); 5485 mOutBuffer.frameCount = 0; 5486 mActive = false; 5487} 5488 5489bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5490{ 5491 Buffer *pInBuffer; 5492 Buffer inBuffer; 5493 uint32_t channelCount = mChannelCount; 5494 bool outputBufferFull = false; 5495 inBuffer.frameCount = frames; 5496 inBuffer.i16 = data; 5497 5498 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5499 5500 if (!mActive && frames != 0) { 5501 start(); 5502 sp<ThreadBase> thread = mThread.promote(); 5503 if (thread != 0) { 5504 MixerThread *mixerThread = (MixerThread *)thread.get(); 5505 if (mCblk->frameCount > frames){ 5506 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5507 uint32_t startFrames = (mCblk->frameCount - frames); 5508 pInBuffer = new Buffer; 5509 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5510 pInBuffer->frameCount = startFrames; 5511 pInBuffer->i16 = pInBuffer->mBuffer; 5512 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5513 mBufferQueue.add(pInBuffer); 5514 } else { 5515 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5516 } 5517 } 5518 } 5519 } 5520 5521 while (waitTimeLeftMs) { 5522 // First write pending buffers, then new data 5523 if (mBufferQueue.size()) { 5524 pInBuffer = mBufferQueue.itemAt(0); 5525 } else { 5526 pInBuffer = &inBuffer; 5527 } 5528 5529 if (pInBuffer->frameCount == 0) { 5530 break; 5531 } 5532 5533 if (mOutBuffer.frameCount == 0) { 5534 mOutBuffer.frameCount = pInBuffer->frameCount; 5535 nsecs_t startTime = systemTime(); 5536 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5537 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5538 outputBufferFull = true; 5539 break; 5540 } 5541 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5542 if (waitTimeLeftMs >= waitTimeMs) { 5543 waitTimeLeftMs -= waitTimeMs; 5544 } else { 5545 waitTimeLeftMs = 0; 5546 } 5547 } 5548 5549 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5550 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5551 mCblk->stepUser(outFrames); 5552 pInBuffer->frameCount -= outFrames; 5553 pInBuffer->i16 += outFrames * channelCount; 5554 mOutBuffer.frameCount -= outFrames; 5555 mOutBuffer.i16 += outFrames * channelCount; 5556 5557 if (pInBuffer->frameCount == 0) { 5558 if (mBufferQueue.size()) { 5559 mBufferQueue.removeAt(0); 5560 delete [] pInBuffer->mBuffer; 5561 delete pInBuffer; 5562 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5563 } else { 5564 break; 5565 } 5566 } 5567 } 5568 5569 // If we could not write all frames, allocate a buffer and queue it for next time. 5570 if (inBuffer.frameCount) { 5571 sp<ThreadBase> thread = mThread.promote(); 5572 if (thread != 0 && !thread->standby()) { 5573 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5574 pInBuffer = new Buffer; 5575 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5576 pInBuffer->frameCount = inBuffer.frameCount; 5577 pInBuffer->i16 = pInBuffer->mBuffer; 5578 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5579 mBufferQueue.add(pInBuffer); 5580 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5581 } else { 5582 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5583 } 5584 } 5585 } 5586 5587 // Calling write() with a 0 length buffer, means that no more data will be written: 5588 // If no more buffers are pending, fill output track buffer to make sure it is started 5589 // by output mixer. 5590 if (frames == 0 && mBufferQueue.size() == 0) { 5591 if (mCblk->user < mCblk->frameCount) { 5592 frames = mCblk->frameCount - mCblk->user; 5593 pInBuffer = new Buffer; 5594 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5595 pInBuffer->frameCount = frames; 5596 pInBuffer->i16 = pInBuffer->mBuffer; 5597 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5598 mBufferQueue.add(pInBuffer); 5599 } else if (mActive) { 5600 stop(); 5601 } 5602 } 5603 5604 return outputBufferFull; 5605} 5606 5607status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5608{ 5609 int active; 5610 status_t result; 5611 audio_track_cblk_t* cblk = mCblk; 5612 uint32_t framesReq = buffer->frameCount; 5613 5614// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5615 buffer->frameCount = 0; 5616 5617 uint32_t framesAvail = cblk->framesAvailable(); 5618 5619 5620 if (framesAvail == 0) { 5621 Mutex::Autolock _l(cblk->lock); 5622 goto start_loop_here; 5623 while (framesAvail == 0) { 5624 active = mActive; 5625 if (CC_UNLIKELY(!active)) { 5626 ALOGV("Not active and NO_MORE_BUFFERS"); 5627 return NO_MORE_BUFFERS; 5628 } 5629 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5630 if (result != NO_ERROR) { 5631 return NO_MORE_BUFFERS; 5632 } 5633 // read the server count again 5634 start_loop_here: 5635 framesAvail = cblk->framesAvailable_l(); 5636 } 5637 } 5638 5639// if (framesAvail < framesReq) { 5640// return NO_MORE_BUFFERS; 5641// } 5642 5643 if (framesReq > framesAvail) { 5644 framesReq = framesAvail; 5645 } 5646 5647 uint32_t u = cblk->user; 5648 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5649 5650 if (framesReq > bufferEnd - u) { 5651 framesReq = bufferEnd - u; 5652 } 5653 5654 buffer->frameCount = framesReq; 5655 buffer->raw = (void *)cblk->buffer(u); 5656 return NO_ERROR; 5657} 5658 5659 5660void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5661{ 5662 size_t size = mBufferQueue.size(); 5663 5664 for (size_t i = 0; i < size; i++) { 5665 Buffer *pBuffer = mBufferQueue.itemAt(i); 5666 delete [] pBuffer->mBuffer; 5667 delete pBuffer; 5668 } 5669 mBufferQueue.clear(); 5670} 5671 5672// ---------------------------------------------------------------------------- 5673 5674AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5675 : RefBase(), 5676 mAudioFlinger(audioFlinger), 5677 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5678 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5679 mPid(pid), 5680 mTimedTrackCount(0) 5681{ 5682 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5683} 5684 5685// Client destructor must be called with AudioFlinger::mLock held 5686AudioFlinger::Client::~Client() 5687{ 5688 mAudioFlinger->removeClient_l(mPid); 5689} 5690 5691sp<MemoryDealer> AudioFlinger::Client::heap() const 5692{ 5693 return mMemoryDealer; 5694} 5695 5696// Reserve one of the limited slots for a timed audio track associated 5697// with this client 5698bool AudioFlinger::Client::reserveTimedTrack() 5699{ 5700 const int kMaxTimedTracksPerClient = 4; 5701 5702 Mutex::Autolock _l(mTimedTrackLock); 5703 5704 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5705 ALOGW("can not create timed track - pid %d has exceeded the limit", 5706 mPid); 5707 return false; 5708 } 5709 5710 mTimedTrackCount++; 5711 return true; 5712} 5713 5714// Release a slot for a timed audio track 5715void AudioFlinger::Client::releaseTimedTrack() 5716{ 5717 Mutex::Autolock _l(mTimedTrackLock); 5718 mTimedTrackCount--; 5719} 5720 5721// ---------------------------------------------------------------------------- 5722 5723AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5724 const sp<IAudioFlingerClient>& client, 5725 pid_t pid) 5726 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5727{ 5728} 5729 5730AudioFlinger::NotificationClient::~NotificationClient() 5731{ 5732} 5733 5734void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5735{ 5736 sp<NotificationClient> keep(this); 5737 mAudioFlinger->removeNotificationClient(mPid); 5738} 5739 5740// ---------------------------------------------------------------------------- 5741 5742AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5743 : BnAudioTrack(), 5744 mTrack(track) 5745{ 5746} 5747 5748AudioFlinger::TrackHandle::~TrackHandle() { 5749 // just stop the track on deletion, associated resources 5750 // will be freed from the main thread once all pending buffers have 5751 // been played. Unless it's not in the active track list, in which 5752 // case we free everything now... 5753 mTrack->destroy(); 5754} 5755 5756sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5757 return mTrack->getCblk(); 5758} 5759 5760status_t AudioFlinger::TrackHandle::start() { 5761 return mTrack->start(); 5762} 5763 5764void AudioFlinger::TrackHandle::stop() { 5765 mTrack->stop(); 5766} 5767 5768void AudioFlinger::TrackHandle::flush() { 5769 mTrack->flush(); 5770} 5771 5772void AudioFlinger::TrackHandle::mute(bool e) { 5773 mTrack->mute(e); 5774} 5775 5776void AudioFlinger::TrackHandle::pause() { 5777 mTrack->pause(); 5778} 5779 5780status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5781{ 5782 return mTrack->attachAuxEffect(EffectId); 5783} 5784 5785status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5786 sp<IMemory>* buffer) { 5787 if (!mTrack->isTimedTrack()) 5788 return INVALID_OPERATION; 5789 5790 PlaybackThread::TimedTrack* tt = 5791 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5792 return tt->allocateTimedBuffer(size, buffer); 5793} 5794 5795status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5796 int64_t pts) { 5797 if (!mTrack->isTimedTrack()) 5798 return INVALID_OPERATION; 5799 5800 PlaybackThread::TimedTrack* tt = 5801 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5802 return tt->queueTimedBuffer(buffer, pts); 5803} 5804 5805status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5806 const LinearTransform& xform, int target) { 5807 5808 if (!mTrack->isTimedTrack()) 5809 return INVALID_OPERATION; 5810 5811 PlaybackThread::TimedTrack* tt = 5812 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5813 return tt->setMediaTimeTransform( 5814 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5815} 5816 5817status_t AudioFlinger::TrackHandle::onTransact( 5818 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5819{ 5820 return BnAudioTrack::onTransact(code, data, reply, flags); 5821} 5822 5823// ---------------------------------------------------------------------------- 5824 5825sp<IAudioRecord> AudioFlinger::openRecord( 5826 pid_t pid, 5827 audio_io_handle_t input, 5828 uint32_t sampleRate, 5829 audio_format_t format, 5830 audio_channel_mask_t channelMask, 5831 int frameCount, 5832 IAudioFlinger::track_flags_t flags, 5833 pid_t tid, 5834 int *sessionId, 5835 status_t *status) 5836{ 5837 sp<RecordThread::RecordTrack> recordTrack; 5838 sp<RecordHandle> recordHandle; 5839 sp<Client> client; 5840 status_t lStatus; 5841 RecordThread *thread; 5842 size_t inFrameCount; 5843 int lSessionId; 5844 5845 // check calling permissions 5846 if (!recordingAllowed()) { 5847 lStatus = PERMISSION_DENIED; 5848 goto Exit; 5849 } 5850 5851 // add client to list 5852 { // scope for mLock 5853 Mutex::Autolock _l(mLock); 5854 thread = checkRecordThread_l(input); 5855 if (thread == NULL) { 5856 lStatus = BAD_VALUE; 5857 goto Exit; 5858 } 5859 5860 client = registerPid_l(pid); 5861 5862 // If no audio session id is provided, create one here 5863 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5864 lSessionId = *sessionId; 5865 } else { 5866 lSessionId = nextUniqueId(); 5867 if (sessionId != NULL) { 5868 *sessionId = lSessionId; 5869 } 5870 } 5871 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5872 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5873 frameCount, lSessionId, flags, tid, &lStatus); 5874 } 5875 if (lStatus != NO_ERROR) { 5876 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5877 // destructor is called by the TrackBase destructor with mLock held 5878 client.clear(); 5879 recordTrack.clear(); 5880 goto Exit; 5881 } 5882 5883 // return to handle to client 5884 recordHandle = new RecordHandle(recordTrack); 5885 lStatus = NO_ERROR; 5886 5887Exit: 5888 if (status) { 5889 *status = lStatus; 5890 } 5891 return recordHandle; 5892} 5893 5894// ---------------------------------------------------------------------------- 5895 5896AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5897 : BnAudioRecord(), 5898 mRecordTrack(recordTrack) 5899{ 5900} 5901 5902AudioFlinger::RecordHandle::~RecordHandle() { 5903 stop_nonvirtual(); 5904 mRecordTrack->destroy(); 5905} 5906 5907sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5908 return mRecordTrack->getCblk(); 5909} 5910 5911status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { 5912 ALOGV("RecordHandle::start()"); 5913 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5914} 5915 5916void AudioFlinger::RecordHandle::stop() { 5917 stop_nonvirtual(); 5918} 5919 5920void AudioFlinger::RecordHandle::stop_nonvirtual() { 5921 ALOGV("RecordHandle::stop()"); 5922 mRecordTrack->stop(); 5923} 5924 5925status_t AudioFlinger::RecordHandle::onTransact( 5926 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5927{ 5928 return BnAudioRecord::onTransact(code, data, reply, flags); 5929} 5930 5931// ---------------------------------------------------------------------------- 5932 5933AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5934 AudioStreamIn *input, 5935 uint32_t sampleRate, 5936 audio_channel_mask_t channelMask, 5937 audio_io_handle_t id, 5938 audio_devices_t device) : 5939 ThreadBase(audioFlinger, id, device, RECORD), 5940 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5941 // mRsmpInIndex and mInputBytes set by readInputParameters() 5942 mReqChannelCount(popcount(channelMask)), 5943 mReqSampleRate(sampleRate) 5944 // mBytesRead is only meaningful while active, and so is cleared in start() 5945 // (but might be better to also clear here for dump?) 5946{ 5947 snprintf(mName, kNameLength, "AudioIn_%X", id); 5948 5949 readInputParameters(); 5950} 5951 5952 5953AudioFlinger::RecordThread::~RecordThread() 5954{ 5955 delete[] mRsmpInBuffer; 5956 delete mResampler; 5957 delete[] mRsmpOutBuffer; 5958} 5959 5960void AudioFlinger::RecordThread::onFirstRef() 5961{ 5962 run(mName, PRIORITY_URGENT_AUDIO); 5963} 5964 5965status_t AudioFlinger::RecordThread::readyToRun() 5966{ 5967 status_t status = initCheck(); 5968 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5969 return status; 5970} 5971 5972bool AudioFlinger::RecordThread::threadLoop() 5973{ 5974 AudioBufferProvider::Buffer buffer; 5975 sp<RecordTrack> activeTrack; 5976 Vector< sp<EffectChain> > effectChains; 5977 5978 nsecs_t lastWarning = 0; 5979 5980 inputStandBy(); 5981 acquireWakeLock(); 5982 5983 // start recording 5984 while (!exitPending()) { 5985 5986 processConfigEvents(); 5987 5988 { // scope for mLock 5989 Mutex::Autolock _l(mLock); 5990 checkForNewParameters_l(); 5991 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5992 standby(); 5993 5994 if (exitPending()) break; 5995 5996 releaseWakeLock_l(); 5997 ALOGV("RecordThread: loop stopping"); 5998 // go to sleep 5999 mWaitWorkCV.wait(mLock); 6000 ALOGV("RecordThread: loop starting"); 6001 acquireWakeLock_l(); 6002 continue; 6003 } 6004 if (mActiveTrack != 0) { 6005 if (mActiveTrack->mState == TrackBase::PAUSING) { 6006 standby(); 6007 mActiveTrack.clear(); 6008 mStartStopCond.broadcast(); 6009 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 6010 if (mReqChannelCount != mActiveTrack->channelCount()) { 6011 mActiveTrack.clear(); 6012 mStartStopCond.broadcast(); 6013 } else if (mBytesRead != 0) { 6014 // record start succeeds only if first read from audio input 6015 // succeeds 6016 if (mBytesRead > 0) { 6017 mActiveTrack->mState = TrackBase::ACTIVE; 6018 } else { 6019 mActiveTrack.clear(); 6020 } 6021 mStartStopCond.broadcast(); 6022 } 6023 mStandby = false; 6024 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 6025 removeTrack_l(mActiveTrack); 6026 mActiveTrack.clear(); 6027 } 6028 } 6029 lockEffectChains_l(effectChains); 6030 } 6031 6032 if (mActiveTrack != 0) { 6033 if (mActiveTrack->mState != TrackBase::ACTIVE && 6034 mActiveTrack->mState != TrackBase::RESUMING) { 6035 unlockEffectChains(effectChains); 6036 usleep(kRecordThreadSleepUs); 6037 continue; 6038 } 6039 for (size_t i = 0; i < effectChains.size(); i ++) { 6040 effectChains[i]->process_l(); 6041 } 6042 6043 buffer.frameCount = mFrameCount; 6044 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6045 size_t framesOut = buffer.frameCount; 6046 if (mResampler == NULL) { 6047 // no resampling 6048 while (framesOut) { 6049 size_t framesIn = mFrameCount - mRsmpInIndex; 6050 if (framesIn) { 6051 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6052 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6053 if (framesIn > framesOut) 6054 framesIn = framesOut; 6055 mRsmpInIndex += framesIn; 6056 framesOut -= framesIn; 6057 if ((int)mChannelCount == mReqChannelCount || 6058 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6059 memcpy(dst, src, framesIn * mFrameSize); 6060 } else { 6061 if (mChannelCount == 1) { 6062 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6063 (int16_t *)src, framesIn); 6064 } else { 6065 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6066 (int16_t *)src, framesIn); 6067 } 6068 } 6069 } 6070 if (framesOut && mFrameCount == mRsmpInIndex) { 6071 if (framesOut == mFrameCount && 6072 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6073 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6074 framesOut = 0; 6075 } else { 6076 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6077 mRsmpInIndex = 0; 6078 } 6079 if (mBytesRead < 0) { 6080 ALOGE("Error reading audio input"); 6081 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6082 // Force input into standby so that it tries to 6083 // recover at next read attempt 6084 inputStandBy(); 6085 usleep(kRecordThreadSleepUs); 6086 } 6087 mRsmpInIndex = mFrameCount; 6088 framesOut = 0; 6089 buffer.frameCount = 0; 6090 } 6091 } 6092 } 6093 } else { 6094 // resampling 6095 6096 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6097 // alter output frame count as if we were expecting stereo samples 6098 if (mChannelCount == 1 && mReqChannelCount == 1) { 6099 framesOut >>= 1; 6100 } 6101 mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */); 6102 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6103 // are 32 bit aligned which should be always true. 6104 if (mChannelCount == 2 && mReqChannelCount == 1) { 6105 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6106 // the resampler always outputs stereo samples: do post stereo to mono conversion 6107 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6108 framesOut); 6109 } else { 6110 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6111 } 6112 6113 } 6114 if (mFramestoDrop == 0) { 6115 mActiveTrack->releaseBuffer(&buffer); 6116 } else { 6117 if (mFramestoDrop > 0) { 6118 mFramestoDrop -= buffer.frameCount; 6119 if (mFramestoDrop <= 0) { 6120 clearSyncStartEvent(); 6121 } 6122 } else { 6123 mFramestoDrop += buffer.frameCount; 6124 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6125 mSyncStartEvent->isCancelled()) { 6126 ALOGW("Synced record %s, session %d, trigger session %d", 6127 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6128 mActiveTrack->sessionId(), 6129 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6130 clearSyncStartEvent(); 6131 } 6132 } 6133 } 6134 mActiveTrack->clearOverflow(); 6135 } 6136 // client isn't retrieving buffers fast enough 6137 else { 6138 if (!mActiveTrack->setOverflow()) { 6139 nsecs_t now = systemTime(); 6140 if ((now - lastWarning) > kWarningThrottleNs) { 6141 ALOGW("RecordThread: buffer overflow"); 6142 lastWarning = now; 6143 } 6144 } 6145 // Release the processor for a while before asking for a new buffer. 6146 // This will give the application more chance to read from the buffer and 6147 // clear the overflow. 6148 usleep(kRecordThreadSleepUs); 6149 } 6150 } 6151 // enable changes in effect chain 6152 unlockEffectChains(effectChains); 6153 effectChains.clear(); 6154 } 6155 6156 standby(); 6157 6158 { 6159 Mutex::Autolock _l(mLock); 6160 mActiveTrack.clear(); 6161 mStartStopCond.broadcast(); 6162 } 6163 6164 releaseWakeLock(); 6165 6166 ALOGV("RecordThread %p exiting", this); 6167 return false; 6168} 6169 6170void AudioFlinger::RecordThread::standby() 6171{ 6172 if (!mStandby) { 6173 inputStandBy(); 6174 mStandby = true; 6175 } 6176} 6177 6178void AudioFlinger::RecordThread::inputStandBy() 6179{ 6180 mInput->stream->common.standby(&mInput->stream->common); 6181} 6182 6183sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6184 const sp<AudioFlinger::Client>& client, 6185 uint32_t sampleRate, 6186 audio_format_t format, 6187 audio_channel_mask_t channelMask, 6188 int frameCount, 6189 int sessionId, 6190 IAudioFlinger::track_flags_t flags, 6191 pid_t tid, 6192 status_t *status) 6193{ 6194 sp<RecordTrack> track; 6195 status_t lStatus; 6196 6197 lStatus = initCheck(); 6198 if (lStatus != NO_ERROR) { 6199 ALOGE("Audio driver not initialized."); 6200 goto Exit; 6201 } 6202 6203 // FIXME use flags and tid similar to createTrack_l() 6204 6205 { // scope for mLock 6206 Mutex::Autolock _l(mLock); 6207 6208 track = new RecordTrack(this, client, sampleRate, 6209 format, channelMask, frameCount, sessionId); 6210 6211 if (track->getCblk() == 0) { 6212 lStatus = NO_MEMORY; 6213 goto Exit; 6214 } 6215 mTracks.add(track); 6216 6217 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6218 bool suspend = audio_is_bluetooth_sco_device(mDevice & AUDIO_DEVICE_IN_ALL) && 6219 mAudioFlinger->btNrecIsOff(); 6220 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6221 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6222 } 6223 lStatus = NO_ERROR; 6224 6225Exit: 6226 if (status) { 6227 *status = lStatus; 6228 } 6229 return track; 6230} 6231 6232status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6233 AudioSystem::sync_event_t event, 6234 int triggerSession) 6235{ 6236 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6237 sp<ThreadBase> strongMe = this; 6238 status_t status = NO_ERROR; 6239 6240 if (event == AudioSystem::SYNC_EVENT_NONE) { 6241 clearSyncStartEvent(); 6242 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6243 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6244 triggerSession, 6245 recordTrack->sessionId(), 6246 syncStartEventCallback, 6247 this); 6248 // Sync event can be cancelled by the trigger session if the track is not in a 6249 // compatible state in which case we start record immediately 6250 if (mSyncStartEvent->isCancelled()) { 6251 clearSyncStartEvent(); 6252 } else { 6253 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6254 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6255 } 6256 } 6257 6258 { 6259 AutoMutex lock(mLock); 6260 if (mActiveTrack != 0) { 6261 if (recordTrack != mActiveTrack.get()) { 6262 status = -EBUSY; 6263 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6264 mActiveTrack->mState = TrackBase::ACTIVE; 6265 } 6266 return status; 6267 } 6268 6269 recordTrack->mState = TrackBase::IDLE; 6270 mActiveTrack = recordTrack; 6271 mLock.unlock(); 6272 status_t status = AudioSystem::startInput(mId); 6273 mLock.lock(); 6274 if (status != NO_ERROR) { 6275 mActiveTrack.clear(); 6276 clearSyncStartEvent(); 6277 return status; 6278 } 6279 mRsmpInIndex = mFrameCount; 6280 mBytesRead = 0; 6281 if (mResampler != NULL) { 6282 mResampler->reset(); 6283 } 6284 mActiveTrack->mState = TrackBase::RESUMING; 6285 // signal thread to start 6286 ALOGV("Signal record thread"); 6287 mWaitWorkCV.signal(); 6288 // do not wait for mStartStopCond if exiting 6289 if (exitPending()) { 6290 mActiveTrack.clear(); 6291 status = INVALID_OPERATION; 6292 goto startError; 6293 } 6294 mStartStopCond.wait(mLock); 6295 if (mActiveTrack == 0) { 6296 ALOGV("Record failed to start"); 6297 status = BAD_VALUE; 6298 goto startError; 6299 } 6300 ALOGV("Record started OK"); 6301 return status; 6302 } 6303startError: 6304 AudioSystem::stopInput(mId); 6305 clearSyncStartEvent(); 6306 return status; 6307} 6308 6309void AudioFlinger::RecordThread::clearSyncStartEvent() 6310{ 6311 if (mSyncStartEvent != 0) { 6312 mSyncStartEvent->cancel(); 6313 } 6314 mSyncStartEvent.clear(); 6315 mFramestoDrop = 0; 6316} 6317 6318void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6319{ 6320 sp<SyncEvent> strongEvent = event.promote(); 6321 6322 if (strongEvent != 0) { 6323 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6324 me->handleSyncStartEvent(strongEvent); 6325 } 6326} 6327 6328void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6329{ 6330 if (event == mSyncStartEvent) { 6331 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6332 // from audio HAL 6333 mFramestoDrop = mFrameCount * 2; 6334 } 6335} 6336 6337bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6338 ALOGV("RecordThread::stop"); 6339 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6340 return false; 6341 } 6342 recordTrack->mState = TrackBase::PAUSING; 6343 // do not wait for mStartStopCond if exiting 6344 if (exitPending()) { 6345 return true; 6346 } 6347 mStartStopCond.wait(mLock); 6348 // if we have been restarted, recordTrack == mActiveTrack.get() here 6349 if (exitPending() || recordTrack != mActiveTrack.get()) { 6350 ALOGV("Record stopped OK"); 6351 return true; 6352 } 6353 return false; 6354} 6355 6356bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 6357{ 6358 return false; 6359} 6360 6361status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6362{ 6363#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6364 if (!isValidSyncEvent(event)) { 6365 return BAD_VALUE; 6366 } 6367 6368 int eventSession = event->triggerSession(); 6369 status_t ret = NAME_NOT_FOUND; 6370 6371 Mutex::Autolock _l(mLock); 6372 6373 for (size_t i = 0; i < mTracks.size(); i++) { 6374 sp<RecordTrack> track = mTracks[i]; 6375 if (eventSession == track->sessionId()) { 6376 (void) track->setSyncEvent(event); 6377 ret = NO_ERROR; 6378 } 6379 } 6380 return ret; 6381#else 6382 return BAD_VALUE; 6383#endif 6384} 6385 6386void AudioFlinger::RecordThread::RecordTrack::destroy() 6387{ 6388 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6389 sp<RecordTrack> keep(this); 6390 { 6391 sp<ThreadBase> thread = mThread.promote(); 6392 if (thread != 0) { 6393 if (mState == ACTIVE || mState == RESUMING) { 6394 AudioSystem::stopInput(thread->id()); 6395 } 6396 AudioSystem::releaseInput(thread->id()); 6397 Mutex::Autolock _l(thread->mLock); 6398 RecordThread *recordThread = (RecordThread *) thread.get(); 6399 recordThread->destroyTrack_l(this); 6400 } 6401 } 6402} 6403 6404// destroyTrack_l() must be called with ThreadBase::mLock held 6405void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6406{ 6407 track->mState = TrackBase::TERMINATED; 6408 // active tracks are removed by threadLoop() 6409 if (mActiveTrack != track) { 6410 removeTrack_l(track); 6411 } 6412} 6413 6414void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6415{ 6416 mTracks.remove(track); 6417 // need anything related to effects here? 6418} 6419 6420void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6421{ 6422 dumpInternals(fd, args); 6423 dumpTracks(fd, args); 6424 dumpEffectChains(fd, args); 6425} 6426 6427void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6428{ 6429 const size_t SIZE = 256; 6430 char buffer[SIZE]; 6431 String8 result; 6432 6433 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6434 result.append(buffer); 6435 6436 if (mActiveTrack != 0) { 6437 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6438 result.append(buffer); 6439 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6440 result.append(buffer); 6441 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6442 result.append(buffer); 6443 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6444 result.append(buffer); 6445 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6446 result.append(buffer); 6447 } else { 6448 result.append("No active record client\n"); 6449 } 6450 6451 write(fd, result.string(), result.size()); 6452 6453 dumpBase(fd, args); 6454} 6455 6456void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6457{ 6458 const size_t SIZE = 256; 6459 char buffer[SIZE]; 6460 String8 result; 6461 6462 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6463 result.append(buffer); 6464 RecordTrack::appendDumpHeader(result); 6465 for (size_t i = 0; i < mTracks.size(); ++i) { 6466 sp<RecordTrack> track = mTracks[i]; 6467 if (track != 0) { 6468 track->dump(buffer, SIZE); 6469 result.append(buffer); 6470 } 6471 } 6472 6473 if (mActiveTrack != 0) { 6474 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6475 result.append(buffer); 6476 RecordTrack::appendDumpHeader(result); 6477 mActiveTrack->dump(buffer, SIZE); 6478 result.append(buffer); 6479 6480 } 6481 write(fd, result.string(), result.size()); 6482} 6483 6484// AudioBufferProvider interface 6485status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6486{ 6487 size_t framesReq = buffer->frameCount; 6488 size_t framesReady = mFrameCount - mRsmpInIndex; 6489 int channelCount; 6490 6491 if (framesReady == 0) { 6492 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6493 if (mBytesRead < 0) { 6494 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6495 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6496 // Force input into standby so that it tries to 6497 // recover at next read attempt 6498 inputStandBy(); 6499 usleep(kRecordThreadSleepUs); 6500 } 6501 buffer->raw = NULL; 6502 buffer->frameCount = 0; 6503 return NOT_ENOUGH_DATA; 6504 } 6505 mRsmpInIndex = 0; 6506 framesReady = mFrameCount; 6507 } 6508 6509 if (framesReq > framesReady) { 6510 framesReq = framesReady; 6511 } 6512 6513 if (mChannelCount == 1 && mReqChannelCount == 2) { 6514 channelCount = 1; 6515 } else { 6516 channelCount = 2; 6517 } 6518 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6519 buffer->frameCount = framesReq; 6520 return NO_ERROR; 6521} 6522 6523// AudioBufferProvider interface 6524void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6525{ 6526 mRsmpInIndex += buffer->frameCount; 6527 buffer->frameCount = 0; 6528} 6529 6530bool AudioFlinger::RecordThread::checkForNewParameters_l() 6531{ 6532 bool reconfig = false; 6533 6534 while (!mNewParameters.isEmpty()) { 6535 status_t status = NO_ERROR; 6536 String8 keyValuePair = mNewParameters[0]; 6537 AudioParameter param = AudioParameter(keyValuePair); 6538 int value; 6539 audio_format_t reqFormat = mFormat; 6540 int reqSamplingRate = mReqSampleRate; 6541 int reqChannelCount = mReqChannelCount; 6542 6543 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6544 reqSamplingRate = value; 6545 reconfig = true; 6546 } 6547 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6548 reqFormat = (audio_format_t) value; 6549 reconfig = true; 6550 } 6551 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6552 reqChannelCount = popcount(value); 6553 reconfig = true; 6554 } 6555 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6556 // do not accept frame count changes if tracks are open as the track buffer 6557 // size depends on frame count and correct behavior would not be guaranteed 6558 // if frame count is changed after track creation 6559 if (mActiveTrack != 0) { 6560 status = INVALID_OPERATION; 6561 } else { 6562 reconfig = true; 6563 } 6564 } 6565 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6566 // forward device change to effects that have requested to be 6567 // aware of attached audio device. 6568 for (size_t i = 0; i < mEffectChains.size(); i++) { 6569 mEffectChains[i]->setDevice_l(value); 6570 } 6571 // store input device and output device but do not forward output device to audio HAL. 6572 // Note that status is ignored by the caller for output device 6573 // (see AudioFlinger::setParameters() 6574 audio_devices_t newDevice = mDevice; 6575 if (value & AUDIO_DEVICE_OUT_ALL) { 6576 newDevice &= ~(value & AUDIO_DEVICE_OUT_ALL); 6577 status = BAD_VALUE; 6578 } else { 6579 newDevice &= ~(value & AUDIO_DEVICE_IN_ALL); 6580 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6581 if (mTracks.size() > 0) { 6582 bool suspend = audio_is_bluetooth_sco_device( 6583 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6584 for (size_t i = 0; i < mTracks.size(); i++) { 6585 sp<RecordTrack> track = mTracks[i]; 6586 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6587 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6588 } 6589 } 6590 } 6591 newDevice |= value; 6592 mDevice = newDevice; // since mDevice is read by other threads, only write to it once 6593 } 6594 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6595 mAudioSource != (audio_source_t)value) { 6596 // forward device change to effects that have requested to be 6597 // aware of attached audio device. 6598 for (size_t i = 0; i < mEffectChains.size(); i++) { 6599 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6600 } 6601 mAudioSource = (audio_source_t)value; 6602 } 6603 if (status == NO_ERROR) { 6604 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6605 if (status == INVALID_OPERATION) { 6606 inputStandBy(); 6607 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6608 keyValuePair.string()); 6609 } 6610 if (reconfig) { 6611 if (status == BAD_VALUE && 6612 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6613 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6614 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6615 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6616 (reqChannelCount <= FCC_2)) { 6617 status = NO_ERROR; 6618 } 6619 if (status == NO_ERROR) { 6620 readInputParameters(); 6621 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6622 } 6623 } 6624 } 6625 6626 mNewParameters.removeAt(0); 6627 6628 mParamStatus = status; 6629 mParamCond.signal(); 6630 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6631 // already timed out waiting for the status and will never signal the condition. 6632 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6633 } 6634 return reconfig; 6635} 6636 6637String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6638{ 6639 char *s; 6640 String8 out_s8 = String8(); 6641 6642 Mutex::Autolock _l(mLock); 6643 if (initCheck() != NO_ERROR) { 6644 return out_s8; 6645 } 6646 6647 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6648 out_s8 = String8(s); 6649 free(s); 6650 return out_s8; 6651} 6652 6653void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6654 AudioSystem::OutputDescriptor desc; 6655 void *param2 = NULL; 6656 6657 switch (event) { 6658 case AudioSystem::INPUT_OPENED: 6659 case AudioSystem::INPUT_CONFIG_CHANGED: 6660 desc.channels = mChannelMask; 6661 desc.samplingRate = mSampleRate; 6662 desc.format = mFormat; 6663 desc.frameCount = mFrameCount; 6664 desc.latency = 0; 6665 param2 = &desc; 6666 break; 6667 6668 case AudioSystem::INPUT_CLOSED: 6669 default: 6670 break; 6671 } 6672 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6673} 6674 6675void AudioFlinger::RecordThread::readInputParameters() 6676{ 6677 delete mRsmpInBuffer; 6678 // mRsmpInBuffer is always assigned a new[] below 6679 delete mRsmpOutBuffer; 6680 mRsmpOutBuffer = NULL; 6681 delete mResampler; 6682 mResampler = NULL; 6683 6684 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6685 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6686 mChannelCount = (uint16_t)popcount(mChannelMask); 6687 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6688 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6689 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6690 mFrameCount = mInputBytes / mFrameSize; 6691 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6692 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6693 6694 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6695 { 6696 int channelCount; 6697 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6698 // stereo to mono post process as the resampler always outputs stereo. 6699 if (mChannelCount == 1 && mReqChannelCount == 2) { 6700 channelCount = 1; 6701 } else { 6702 channelCount = 2; 6703 } 6704 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6705 mResampler->setSampleRate(mSampleRate); 6706 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6707 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6708 6709 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6710 if (mChannelCount == 1 && mReqChannelCount == 1) { 6711 mFrameCount >>= 1; 6712 } 6713 6714 } 6715 mRsmpInIndex = mFrameCount; 6716} 6717 6718unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6719{ 6720 Mutex::Autolock _l(mLock); 6721 if (initCheck() != NO_ERROR) { 6722 return 0; 6723 } 6724 6725 return mInput->stream->get_input_frames_lost(mInput->stream); 6726} 6727 6728uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6729{ 6730 Mutex::Autolock _l(mLock); 6731 uint32_t result = 0; 6732 if (getEffectChain_l(sessionId) != 0) { 6733 result = EFFECT_SESSION; 6734 } 6735 6736 for (size_t i = 0; i < mTracks.size(); ++i) { 6737 if (sessionId == mTracks[i]->sessionId()) { 6738 result |= TRACK_SESSION; 6739 break; 6740 } 6741 } 6742 6743 return result; 6744} 6745 6746KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6747{ 6748 KeyedVector<int, bool> ids; 6749 Mutex::Autolock _l(mLock); 6750 for (size_t j = 0; j < mTracks.size(); ++j) { 6751 sp<RecordThread::RecordTrack> track = mTracks[j]; 6752 int sessionId = track->sessionId(); 6753 if (ids.indexOfKey(sessionId) < 0) { 6754 ids.add(sessionId, true); 6755 } 6756 } 6757 return ids; 6758} 6759 6760AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6761{ 6762 Mutex::Autolock _l(mLock); 6763 AudioStreamIn *input = mInput; 6764 mInput = NULL; 6765 return input; 6766} 6767 6768// this method must always be called either with ThreadBase mLock held or inside the thread loop 6769audio_stream_t* AudioFlinger::RecordThread::stream() const 6770{ 6771 if (mInput == NULL) { 6772 return NULL; 6773 } 6774 return &mInput->stream->common; 6775} 6776 6777 6778// ---------------------------------------------------------------------------- 6779 6780audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6781{ 6782 if (!settingsAllowed()) { 6783 return 0; 6784 } 6785 Mutex::Autolock _l(mLock); 6786 return loadHwModule_l(name); 6787} 6788 6789// loadHwModule_l() must be called with AudioFlinger::mLock held 6790audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6791{ 6792 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6793 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6794 ALOGW("loadHwModule() module %s already loaded", name); 6795 return mAudioHwDevs.keyAt(i); 6796 } 6797 } 6798 6799 audio_hw_device_t *dev; 6800 6801 int rc = load_audio_interface(name, &dev); 6802 if (rc) { 6803 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6804 return 0; 6805 } 6806 6807 mHardwareStatus = AUDIO_HW_INIT; 6808 rc = dev->init_check(dev); 6809 mHardwareStatus = AUDIO_HW_IDLE; 6810 if (rc) { 6811 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6812 return 0; 6813 } 6814 6815 // Check and cache this HAL's level of support for master mute and master 6816 // volume. If this is the first HAL opened, and it supports the get 6817 // methods, use the initial values provided by the HAL as the current 6818 // master mute and volume settings. 6819 6820 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 6821 { // scope for auto-lock pattern 6822 AutoMutex lock(mHardwareLock); 6823 6824 if (0 == mAudioHwDevs.size()) { 6825 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6826 if (NULL != dev->get_master_volume) { 6827 float mv; 6828 if (OK == dev->get_master_volume(dev, &mv)) { 6829 mMasterVolume = mv; 6830 } 6831 } 6832 6833 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 6834 if (NULL != dev->get_master_mute) { 6835 bool mm; 6836 if (OK == dev->get_master_mute(dev, &mm)) { 6837 mMasterMute = mm; 6838 } 6839 } 6840 } 6841 6842 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6843 if ((NULL != dev->set_master_volume) && 6844 (OK == dev->set_master_volume(dev, mMasterVolume))) { 6845 flags = static_cast<AudioHwDevice::Flags>(flags | 6846 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 6847 } 6848 6849 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 6850 if ((NULL != dev->set_master_mute) && 6851 (OK == dev->set_master_mute(dev, mMasterMute))) { 6852 flags = static_cast<AudioHwDevice::Flags>(flags | 6853 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 6854 } 6855 6856 mHardwareStatus = AUDIO_HW_IDLE; 6857 } 6858 6859 audio_module_handle_t handle = nextUniqueId(); 6860 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 6861 6862 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6863 name, dev->common.module->name, dev->common.module->id, handle); 6864 6865 return handle; 6866 6867} 6868 6869audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6870 audio_devices_t *pDevices, 6871 uint32_t *pSamplingRate, 6872 audio_format_t *pFormat, 6873 audio_channel_mask_t *pChannelMask, 6874 uint32_t *pLatencyMs, 6875 audio_output_flags_t flags) 6876{ 6877 status_t status; 6878 PlaybackThread *thread = NULL; 6879 struct audio_config config = { 6880 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6881 channel_mask: pChannelMask ? *pChannelMask : 0, 6882 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6883 }; 6884 audio_stream_out_t *outStream = NULL; 6885 AudioHwDevice *outHwDev; 6886 6887 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6888 module, 6889 (pDevices != NULL) ? *pDevices : 0, 6890 config.sample_rate, 6891 config.format, 6892 config.channel_mask, 6893 flags); 6894 6895 if (pDevices == NULL || *pDevices == 0) { 6896 return 0; 6897 } 6898 6899 Mutex::Autolock _l(mLock); 6900 6901 outHwDev = findSuitableHwDev_l(module, *pDevices); 6902 if (outHwDev == NULL) 6903 return 0; 6904 6905 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 6906 audio_io_handle_t id = nextUniqueId(); 6907 6908 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6909 6910 status = hwDevHal->open_output_stream(hwDevHal, 6911 id, 6912 *pDevices, 6913 (audio_output_flags_t)flags, 6914 &config, 6915 &outStream); 6916 6917 mHardwareStatus = AUDIO_HW_IDLE; 6918 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6919 outStream, 6920 config.sample_rate, 6921 config.format, 6922 config.channel_mask, 6923 status); 6924 6925 if (status == NO_ERROR && outStream != NULL) { 6926 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6927 6928 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6929 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6930 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6931 thread = new DirectOutputThread(this, output, id, *pDevices); 6932 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6933 } else { 6934 thread = new MixerThread(this, output, id, *pDevices); 6935 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6936 } 6937 mPlaybackThreads.add(id, thread); 6938 6939 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6940 if (pFormat != NULL) *pFormat = config.format; 6941 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6942 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6943 6944 // notify client processes of the new output creation 6945 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6946 6947 // the first primary output opened designates the primary hw device 6948 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6949 ALOGI("Using module %d has the primary audio interface", module); 6950 mPrimaryHardwareDev = outHwDev; 6951 6952 AutoMutex lock(mHardwareLock); 6953 mHardwareStatus = AUDIO_HW_SET_MODE; 6954 hwDevHal->set_mode(hwDevHal, mMode); 6955 mHardwareStatus = AUDIO_HW_IDLE; 6956 } 6957 return id; 6958 } 6959 6960 return 0; 6961} 6962 6963audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6964 audio_io_handle_t output2) 6965{ 6966 Mutex::Autolock _l(mLock); 6967 MixerThread *thread1 = checkMixerThread_l(output1); 6968 MixerThread *thread2 = checkMixerThread_l(output2); 6969 6970 if (thread1 == NULL || thread2 == NULL) { 6971 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6972 return 0; 6973 } 6974 6975 audio_io_handle_t id = nextUniqueId(); 6976 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6977 thread->addOutputTrack(thread2); 6978 mPlaybackThreads.add(id, thread); 6979 // notify client processes of the new output creation 6980 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6981 return id; 6982} 6983 6984status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6985{ 6986 return closeOutput_nonvirtual(output); 6987} 6988 6989status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6990{ 6991 // keep strong reference on the playback thread so that 6992 // it is not destroyed while exit() is executed 6993 sp<PlaybackThread> thread; 6994 { 6995 Mutex::Autolock _l(mLock); 6996 thread = checkPlaybackThread_l(output); 6997 if (thread == NULL) { 6998 return BAD_VALUE; 6999 } 7000 7001 ALOGV("closeOutput() %d", output); 7002 7003 if (thread->type() == ThreadBase::MIXER) { 7004 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7005 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 7006 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 7007 dupThread->removeOutputTrack((MixerThread *)thread.get()); 7008 } 7009 } 7010 } 7011 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 7012 mPlaybackThreads.removeItem(output); 7013 } 7014 thread->exit(); 7015 // The thread entity (active unit of execution) is no longer running here, 7016 // but the ThreadBase container still exists. 7017 7018 if (thread->type() != ThreadBase::DUPLICATING) { 7019 AudioStreamOut *out = thread->clearOutput(); 7020 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 7021 // from now on thread->mOutput is NULL 7022 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 7023 delete out; 7024 } 7025 return NO_ERROR; 7026} 7027 7028status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 7029{ 7030 Mutex::Autolock _l(mLock); 7031 PlaybackThread *thread = checkPlaybackThread_l(output); 7032 7033 if (thread == NULL) { 7034 return BAD_VALUE; 7035 } 7036 7037 ALOGV("suspendOutput() %d", output); 7038 thread->suspend(); 7039 7040 return NO_ERROR; 7041} 7042 7043status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 7044{ 7045 Mutex::Autolock _l(mLock); 7046 PlaybackThread *thread = checkPlaybackThread_l(output); 7047 7048 if (thread == NULL) { 7049 return BAD_VALUE; 7050 } 7051 7052 ALOGV("restoreOutput() %d", output); 7053 7054 thread->restore(); 7055 7056 return NO_ERROR; 7057} 7058 7059audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7060 audio_devices_t *pDevices, 7061 uint32_t *pSamplingRate, 7062 audio_format_t *pFormat, 7063 audio_channel_mask_t *pChannelMask) 7064{ 7065 status_t status; 7066 RecordThread *thread = NULL; 7067 struct audio_config config = { 7068 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7069 channel_mask: pChannelMask ? *pChannelMask : 0, 7070 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7071 }; 7072 uint32_t reqSamplingRate = config.sample_rate; 7073 audio_format_t reqFormat = config.format; 7074 audio_channel_mask_t reqChannels = config.channel_mask; 7075 audio_stream_in_t *inStream = NULL; 7076 AudioHwDevice *inHwDev; 7077 7078 if (pDevices == NULL || *pDevices == 0) { 7079 return 0; 7080 } 7081 7082 Mutex::Autolock _l(mLock); 7083 7084 inHwDev = findSuitableHwDev_l(module, *pDevices); 7085 if (inHwDev == NULL) 7086 return 0; 7087 7088 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 7089 audio_io_handle_t id = nextUniqueId(); 7090 7091 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 7092 &inStream); 7093 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7094 inStream, 7095 config.sample_rate, 7096 config.format, 7097 config.channel_mask, 7098 status); 7099 7100 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7101 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7102 // or stereo to mono conversions on 16 bit PCM inputs. 7103 if (status == BAD_VALUE && 7104 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7105 (config.sample_rate <= 2 * reqSamplingRate) && 7106 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7107 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7108 inStream = NULL; 7109 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 7110 } 7111 7112 if (status == NO_ERROR && inStream != NULL) { 7113 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7114 7115 // Start record thread 7116 // RecorThread require both input and output device indication to forward to audio 7117 // pre processing modules 7118 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7119 thread = new RecordThread(this, 7120 input, 7121 reqSamplingRate, 7122 reqChannels, 7123 id, 7124 device); 7125 mRecordThreads.add(id, thread); 7126 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7127 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7128 if (pFormat != NULL) *pFormat = config.format; 7129 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7130 7131 // notify client processes of the new input creation 7132 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7133 return id; 7134 } 7135 7136 return 0; 7137} 7138 7139status_t AudioFlinger::closeInput(audio_io_handle_t input) 7140{ 7141 return closeInput_nonvirtual(input); 7142} 7143 7144status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7145{ 7146 // keep strong reference on the record thread so that 7147 // it is not destroyed while exit() is executed 7148 sp<RecordThread> thread; 7149 { 7150 Mutex::Autolock _l(mLock); 7151 thread = checkRecordThread_l(input); 7152 if (thread == 0) { 7153 return BAD_VALUE; 7154 } 7155 7156 ALOGV("closeInput() %d", input); 7157 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7158 mRecordThreads.removeItem(input); 7159 } 7160 thread->exit(); 7161 // The thread entity (active unit of execution) is no longer running here, 7162 // but the ThreadBase container still exists. 7163 7164 AudioStreamIn *in = thread->clearInput(); 7165 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7166 // from now on thread->mInput is NULL 7167 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 7168 delete in; 7169 7170 return NO_ERROR; 7171} 7172 7173status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7174{ 7175 Mutex::Autolock _l(mLock); 7176 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7177 7178 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7179 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7180 thread->invalidateTracks(stream); 7181 } 7182 7183 return NO_ERROR; 7184} 7185 7186 7187int AudioFlinger::newAudioSessionId() 7188{ 7189 return nextUniqueId(); 7190} 7191 7192void AudioFlinger::acquireAudioSessionId(int audioSession) 7193{ 7194 Mutex::Autolock _l(mLock); 7195 pid_t caller = IPCThreadState::self()->getCallingPid(); 7196 ALOGV("acquiring %d from %d", audioSession, caller); 7197 size_t num = mAudioSessionRefs.size(); 7198 for (size_t i = 0; i< num; i++) { 7199 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7200 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7201 ref->mCnt++; 7202 ALOGV(" incremented refcount to %d", ref->mCnt); 7203 return; 7204 } 7205 } 7206 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7207 ALOGV(" added new entry for %d", audioSession); 7208} 7209 7210void AudioFlinger::releaseAudioSessionId(int audioSession) 7211{ 7212 Mutex::Autolock _l(mLock); 7213 pid_t caller = IPCThreadState::self()->getCallingPid(); 7214 ALOGV("releasing %d from %d", audioSession, caller); 7215 size_t num = mAudioSessionRefs.size(); 7216 for (size_t i = 0; i< num; i++) { 7217 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7218 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7219 ref->mCnt--; 7220 ALOGV(" decremented refcount to %d", ref->mCnt); 7221 if (ref->mCnt == 0) { 7222 mAudioSessionRefs.removeAt(i); 7223 delete ref; 7224 purgeStaleEffects_l(); 7225 } 7226 return; 7227 } 7228 } 7229 ALOGW("session id %d not found for pid %d", audioSession, caller); 7230} 7231 7232void AudioFlinger::purgeStaleEffects_l() { 7233 7234 ALOGV("purging stale effects"); 7235 7236 Vector< sp<EffectChain> > chains; 7237 7238 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7239 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7240 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7241 sp<EffectChain> ec = t->mEffectChains[j]; 7242 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7243 chains.push(ec); 7244 } 7245 } 7246 } 7247 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7248 sp<RecordThread> t = mRecordThreads.valueAt(i); 7249 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7250 sp<EffectChain> ec = t->mEffectChains[j]; 7251 chains.push(ec); 7252 } 7253 } 7254 7255 for (size_t i = 0; i < chains.size(); i++) { 7256 sp<EffectChain> ec = chains[i]; 7257 int sessionid = ec->sessionId(); 7258 sp<ThreadBase> t = ec->mThread.promote(); 7259 if (t == 0) { 7260 continue; 7261 } 7262 size_t numsessionrefs = mAudioSessionRefs.size(); 7263 bool found = false; 7264 for (size_t k = 0; k < numsessionrefs; k++) { 7265 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7266 if (ref->mSessionid == sessionid) { 7267 ALOGV(" session %d still exists for %d with %d refs", 7268 sessionid, ref->mPid, ref->mCnt); 7269 found = true; 7270 break; 7271 } 7272 } 7273 if (!found) { 7274 Mutex::Autolock _l (t->mLock); 7275 // remove all effects from the chain 7276 while (ec->mEffects.size()) { 7277 sp<EffectModule> effect = ec->mEffects[0]; 7278 effect->unPin(); 7279 t->removeEffect_l(effect); 7280 if (effect->purgeHandles()) { 7281 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7282 } 7283 AudioSystem::unregisterEffect(effect->id()); 7284 } 7285 } 7286 } 7287 return; 7288} 7289 7290// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7291AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7292{ 7293 return mPlaybackThreads.valueFor(output).get(); 7294} 7295 7296// checkMixerThread_l() must be called with AudioFlinger::mLock held 7297AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7298{ 7299 PlaybackThread *thread = checkPlaybackThread_l(output); 7300 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7301} 7302 7303// checkRecordThread_l() must be called with AudioFlinger::mLock held 7304AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7305{ 7306 return mRecordThreads.valueFor(input).get(); 7307} 7308 7309uint32_t AudioFlinger::nextUniqueId() 7310{ 7311 return android_atomic_inc(&mNextUniqueId); 7312} 7313 7314AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7315{ 7316 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7317 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7318 AudioStreamOut *output = thread->getOutput(); 7319 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 7320 return thread; 7321 } 7322 } 7323 return NULL; 7324} 7325 7326audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7327{ 7328 PlaybackThread *thread = primaryPlaybackThread_l(); 7329 7330 if (thread == NULL) { 7331 return 0; 7332 } 7333 7334 return thread->device(); 7335} 7336 7337sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7338 int triggerSession, 7339 int listenerSession, 7340 sync_event_callback_t callBack, 7341 void *cookie) 7342{ 7343 Mutex::Autolock _l(mLock); 7344 7345 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7346 status_t playStatus = NAME_NOT_FOUND; 7347 status_t recStatus = NAME_NOT_FOUND; 7348 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7349 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7350 if (playStatus == NO_ERROR) { 7351 return event; 7352 } 7353 } 7354 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7355 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7356 if (recStatus == NO_ERROR) { 7357 return event; 7358 } 7359 } 7360 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7361 mPendingSyncEvents.add(event); 7362 } else { 7363 ALOGV("createSyncEvent() invalid event %d", event->type()); 7364 event.clear(); 7365 } 7366 return event; 7367} 7368 7369// ---------------------------------------------------------------------------- 7370// Effect management 7371// ---------------------------------------------------------------------------- 7372 7373 7374status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7375{ 7376 Mutex::Autolock _l(mLock); 7377 return EffectQueryNumberEffects(numEffects); 7378} 7379 7380status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7381{ 7382 Mutex::Autolock _l(mLock); 7383 return EffectQueryEffect(index, descriptor); 7384} 7385 7386status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7387 effect_descriptor_t *descriptor) const 7388{ 7389 Mutex::Autolock _l(mLock); 7390 return EffectGetDescriptor(pUuid, descriptor); 7391} 7392 7393 7394sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7395 effect_descriptor_t *pDesc, 7396 const sp<IEffectClient>& effectClient, 7397 int32_t priority, 7398 audio_io_handle_t io, 7399 int sessionId, 7400 status_t *status, 7401 int *id, 7402 int *enabled) 7403{ 7404 status_t lStatus = NO_ERROR; 7405 sp<EffectHandle> handle; 7406 effect_descriptor_t desc; 7407 7408 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7409 pid, effectClient.get(), priority, sessionId, io); 7410 7411 if (pDesc == NULL) { 7412 lStatus = BAD_VALUE; 7413 goto Exit; 7414 } 7415 7416 // check audio settings permission for global effects 7417 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7418 lStatus = PERMISSION_DENIED; 7419 goto Exit; 7420 } 7421 7422 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7423 // that can only be created by audio policy manager (running in same process) 7424 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7425 lStatus = PERMISSION_DENIED; 7426 goto Exit; 7427 } 7428 7429 if (io == 0) { 7430 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7431 // output must be specified by AudioPolicyManager when using session 7432 // AUDIO_SESSION_OUTPUT_STAGE 7433 lStatus = BAD_VALUE; 7434 goto Exit; 7435 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7436 // if the output returned by getOutputForEffect() is removed before we lock the 7437 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7438 // and we will exit safely 7439 io = AudioSystem::getOutputForEffect(&desc); 7440 } 7441 } 7442 7443 { 7444 Mutex::Autolock _l(mLock); 7445 7446 7447 if (!EffectIsNullUuid(&pDesc->uuid)) { 7448 // if uuid is specified, request effect descriptor 7449 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7450 if (lStatus < 0) { 7451 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7452 goto Exit; 7453 } 7454 } else { 7455 // if uuid is not specified, look for an available implementation 7456 // of the required type in effect factory 7457 if (EffectIsNullUuid(&pDesc->type)) { 7458 ALOGW("createEffect() no effect type"); 7459 lStatus = BAD_VALUE; 7460 goto Exit; 7461 } 7462 uint32_t numEffects = 0; 7463 effect_descriptor_t d; 7464 d.flags = 0; // prevent compiler warning 7465 bool found = false; 7466 7467 lStatus = EffectQueryNumberEffects(&numEffects); 7468 if (lStatus < 0) { 7469 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7470 goto Exit; 7471 } 7472 for (uint32_t i = 0; i < numEffects; i++) { 7473 lStatus = EffectQueryEffect(i, &desc); 7474 if (lStatus < 0) { 7475 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7476 continue; 7477 } 7478 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7479 // If matching type found save effect descriptor. If the session is 7480 // 0 and the effect is not auxiliary, continue enumeration in case 7481 // an auxiliary version of this effect type is available 7482 found = true; 7483 d = desc; 7484 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7485 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7486 break; 7487 } 7488 } 7489 } 7490 if (!found) { 7491 lStatus = BAD_VALUE; 7492 ALOGW("createEffect() effect not found"); 7493 goto Exit; 7494 } 7495 // For same effect type, chose auxiliary version over insert version if 7496 // connect to output mix (Compliance to OpenSL ES) 7497 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7498 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7499 desc = d; 7500 } 7501 } 7502 7503 // Do not allow auxiliary effects on a session different from 0 (output mix) 7504 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7505 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7506 lStatus = INVALID_OPERATION; 7507 goto Exit; 7508 } 7509 7510 // check recording permission for visualizer 7511 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7512 !recordingAllowed()) { 7513 lStatus = PERMISSION_DENIED; 7514 goto Exit; 7515 } 7516 7517 // return effect descriptor 7518 *pDesc = desc; 7519 7520 // If output is not specified try to find a matching audio session ID in one of the 7521 // output threads. 7522 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7523 // because of code checking output when entering the function. 7524 // Note: io is never 0 when creating an effect on an input 7525 if (io == 0) { 7526 // look for the thread where the specified audio session is present 7527 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7528 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7529 io = mPlaybackThreads.keyAt(i); 7530 break; 7531 } 7532 } 7533 if (io == 0) { 7534 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7535 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7536 io = mRecordThreads.keyAt(i); 7537 break; 7538 } 7539 } 7540 } 7541 // If no output thread contains the requested session ID, default to 7542 // first output. The effect chain will be moved to the correct output 7543 // thread when a track with the same session ID is created 7544 if (io == 0 && mPlaybackThreads.size()) { 7545 io = mPlaybackThreads.keyAt(0); 7546 } 7547 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7548 } 7549 ThreadBase *thread = checkRecordThread_l(io); 7550 if (thread == NULL) { 7551 thread = checkPlaybackThread_l(io); 7552 if (thread == NULL) { 7553 ALOGE("createEffect() unknown output thread"); 7554 lStatus = BAD_VALUE; 7555 goto Exit; 7556 } 7557 } 7558 7559 sp<Client> client = registerPid_l(pid); 7560 7561 // create effect on selected output thread 7562 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7563 &desc, enabled, &lStatus); 7564 if (handle != 0 && id != NULL) { 7565 *id = handle->id(); 7566 } 7567 } 7568 7569Exit: 7570 if (status != NULL) { 7571 *status = lStatus; 7572 } 7573 return handle; 7574} 7575 7576status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7577 audio_io_handle_t dstOutput) 7578{ 7579 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7580 sessionId, srcOutput, dstOutput); 7581 Mutex::Autolock _l(mLock); 7582 if (srcOutput == dstOutput) { 7583 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7584 return NO_ERROR; 7585 } 7586 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7587 if (srcThread == NULL) { 7588 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7589 return BAD_VALUE; 7590 } 7591 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7592 if (dstThread == NULL) { 7593 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7594 return BAD_VALUE; 7595 } 7596 7597 Mutex::Autolock _dl(dstThread->mLock); 7598 Mutex::Autolock _sl(srcThread->mLock); 7599 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7600 7601 return NO_ERROR; 7602} 7603 7604// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7605status_t AudioFlinger::moveEffectChain_l(int sessionId, 7606 AudioFlinger::PlaybackThread *srcThread, 7607 AudioFlinger::PlaybackThread *dstThread, 7608 bool reRegister) 7609{ 7610 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7611 sessionId, srcThread, dstThread); 7612 7613 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7614 if (chain == 0) { 7615 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7616 sessionId, srcThread); 7617 return INVALID_OPERATION; 7618 } 7619 7620 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7621 // so that a new chain is created with correct parameters when first effect is added. This is 7622 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7623 // removed. 7624 srcThread->removeEffectChain_l(chain); 7625 7626 // transfer all effects one by one so that new effect chain is created on new thread with 7627 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7628 audio_io_handle_t dstOutput = dstThread->id(); 7629 sp<EffectChain> dstChain; 7630 uint32_t strategy = 0; // prevent compiler warning 7631 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7632 while (effect != 0) { 7633 srcThread->removeEffect_l(effect); 7634 dstThread->addEffect_l(effect); 7635 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7636 if (effect->state() == EffectModule::ACTIVE || 7637 effect->state() == EffectModule::STOPPING) { 7638 effect->start(); 7639 } 7640 // if the move request is not received from audio policy manager, the effect must be 7641 // re-registered with the new strategy and output 7642 if (dstChain == 0) { 7643 dstChain = effect->chain().promote(); 7644 if (dstChain == 0) { 7645 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7646 srcThread->addEffect_l(effect); 7647 return NO_INIT; 7648 } 7649 strategy = dstChain->strategy(); 7650 } 7651 if (reRegister) { 7652 AudioSystem::unregisterEffect(effect->id()); 7653 AudioSystem::registerEffect(&effect->desc(), 7654 dstOutput, 7655 strategy, 7656 sessionId, 7657 effect->id()); 7658 } 7659 effect = chain->getEffectFromId_l(0); 7660 } 7661 7662 return NO_ERROR; 7663} 7664 7665 7666// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7667sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7668 const sp<AudioFlinger::Client>& client, 7669 const sp<IEffectClient>& effectClient, 7670 int32_t priority, 7671 int sessionId, 7672 effect_descriptor_t *desc, 7673 int *enabled, 7674 status_t *status 7675 ) 7676{ 7677 sp<EffectModule> effect; 7678 sp<EffectHandle> handle; 7679 status_t lStatus; 7680 sp<EffectChain> chain; 7681 bool chainCreated = false; 7682 bool effectCreated = false; 7683 bool effectRegistered = false; 7684 7685 lStatus = initCheck(); 7686 if (lStatus != NO_ERROR) { 7687 ALOGW("createEffect_l() Audio driver not initialized."); 7688 goto Exit; 7689 } 7690 7691 // Do not allow effects with session ID 0 on direct output or duplicating threads 7692 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7693 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7694 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7695 desc->name, sessionId); 7696 lStatus = BAD_VALUE; 7697 goto Exit; 7698 } 7699 // Only Pre processor effects are allowed on input threads and only on input threads 7700 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7701 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7702 desc->name, desc->flags, mType); 7703 lStatus = BAD_VALUE; 7704 goto Exit; 7705 } 7706 7707 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7708 7709 { // scope for mLock 7710 Mutex::Autolock _l(mLock); 7711 7712 // check for existing effect chain with the requested audio session 7713 chain = getEffectChain_l(sessionId); 7714 if (chain == 0) { 7715 // create a new chain for this session 7716 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7717 chain = new EffectChain(this, sessionId); 7718 addEffectChain_l(chain); 7719 chain->setStrategy(getStrategyForSession_l(sessionId)); 7720 chainCreated = true; 7721 } else { 7722 effect = chain->getEffectFromDesc_l(desc); 7723 } 7724 7725 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7726 7727 if (effect == 0) { 7728 int id = mAudioFlinger->nextUniqueId(); 7729 // Check CPU and memory usage 7730 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7731 if (lStatus != NO_ERROR) { 7732 goto Exit; 7733 } 7734 effectRegistered = true; 7735 // create a new effect module if none present in the chain 7736 effect = new EffectModule(this, chain, desc, id, sessionId); 7737 lStatus = effect->status(); 7738 if (lStatus != NO_ERROR) { 7739 goto Exit; 7740 } 7741 lStatus = chain->addEffect_l(effect); 7742 if (lStatus != NO_ERROR) { 7743 goto Exit; 7744 } 7745 effectCreated = true; 7746 7747 effect->setDevice(mDevice); 7748 effect->setMode(mAudioFlinger->getMode()); 7749 effect->setAudioSource(mAudioSource); 7750 } 7751 // create effect handle and connect it to effect module 7752 handle = new EffectHandle(effect, client, effectClient, priority); 7753 lStatus = effect->addHandle(handle.get()); 7754 if (enabled != NULL) { 7755 *enabled = (int)effect->isEnabled(); 7756 } 7757 } 7758 7759Exit: 7760 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7761 Mutex::Autolock _l(mLock); 7762 if (effectCreated) { 7763 chain->removeEffect_l(effect); 7764 } 7765 if (effectRegistered) { 7766 AudioSystem::unregisterEffect(effect->id()); 7767 } 7768 if (chainCreated) { 7769 removeEffectChain_l(chain); 7770 } 7771 handle.clear(); 7772 } 7773 7774 if (status != NULL) { 7775 *status = lStatus; 7776 } 7777 return handle; 7778} 7779 7780sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7781{ 7782 Mutex::Autolock _l(mLock); 7783 return getEffect_l(sessionId, effectId); 7784} 7785 7786sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7787{ 7788 sp<EffectChain> chain = getEffectChain_l(sessionId); 7789 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7790} 7791 7792// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7793// PlaybackThread::mLock held 7794status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7795{ 7796 // check for existing effect chain with the requested audio session 7797 int sessionId = effect->sessionId(); 7798 sp<EffectChain> chain = getEffectChain_l(sessionId); 7799 bool chainCreated = false; 7800 7801 if (chain == 0) { 7802 // create a new chain for this session 7803 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7804 chain = new EffectChain(this, sessionId); 7805 addEffectChain_l(chain); 7806 chain->setStrategy(getStrategyForSession_l(sessionId)); 7807 chainCreated = true; 7808 } 7809 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7810 7811 if (chain->getEffectFromId_l(effect->id()) != 0) { 7812 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7813 this, effect->desc().name, chain.get()); 7814 return BAD_VALUE; 7815 } 7816 7817 status_t status = chain->addEffect_l(effect); 7818 if (status != NO_ERROR) { 7819 if (chainCreated) { 7820 removeEffectChain_l(chain); 7821 } 7822 return status; 7823 } 7824 7825 effect->setDevice(mDevice); 7826 effect->setMode(mAudioFlinger->getMode()); 7827 effect->setAudioSource(mAudioSource); 7828 return NO_ERROR; 7829} 7830 7831void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7832 7833 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7834 effect_descriptor_t desc = effect->desc(); 7835 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7836 detachAuxEffect_l(effect->id()); 7837 } 7838 7839 sp<EffectChain> chain = effect->chain().promote(); 7840 if (chain != 0) { 7841 // remove effect chain if removing last effect 7842 if (chain->removeEffect_l(effect) == 0) { 7843 removeEffectChain_l(chain); 7844 } 7845 } else { 7846 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7847 } 7848} 7849 7850void AudioFlinger::ThreadBase::lockEffectChains_l( 7851 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7852{ 7853 effectChains = mEffectChains; 7854 for (size_t i = 0; i < mEffectChains.size(); i++) { 7855 mEffectChains[i]->lock(); 7856 } 7857} 7858 7859void AudioFlinger::ThreadBase::unlockEffectChains( 7860 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7861{ 7862 for (size_t i = 0; i < effectChains.size(); i++) { 7863 effectChains[i]->unlock(); 7864 } 7865} 7866 7867sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7868{ 7869 Mutex::Autolock _l(mLock); 7870 return getEffectChain_l(sessionId); 7871} 7872 7873sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 7874{ 7875 size_t size = mEffectChains.size(); 7876 for (size_t i = 0; i < size; i++) { 7877 if (mEffectChains[i]->sessionId() == sessionId) { 7878 return mEffectChains[i]; 7879 } 7880 } 7881 return 0; 7882} 7883 7884void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7885{ 7886 Mutex::Autolock _l(mLock); 7887 size_t size = mEffectChains.size(); 7888 for (size_t i = 0; i < size; i++) { 7889 mEffectChains[i]->setMode_l(mode); 7890 } 7891} 7892 7893void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7894 EffectHandle *handle, 7895 bool unpinIfLast) { 7896 7897 Mutex::Autolock _l(mLock); 7898 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7899 // delete the effect module if removing last handle on it 7900 if (effect->removeHandle(handle) == 0) { 7901 if (!effect->isPinned() || unpinIfLast) { 7902 removeEffect_l(effect); 7903 AudioSystem::unregisterEffect(effect->id()); 7904 } 7905 } 7906} 7907 7908status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7909{ 7910 int session = chain->sessionId(); 7911 int16_t *buffer = mMixBuffer; 7912 bool ownsBuffer = false; 7913 7914 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7915 if (session > 0) { 7916 // Only one effect chain can be present in direct output thread and it uses 7917 // the mix buffer as input 7918 if (mType != DIRECT) { 7919 size_t numSamples = mNormalFrameCount * mChannelCount; 7920 buffer = new int16_t[numSamples]; 7921 memset(buffer, 0, numSamples * sizeof(int16_t)); 7922 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7923 ownsBuffer = true; 7924 } 7925 7926 // Attach all tracks with same session ID to this chain. 7927 for (size_t i = 0; i < mTracks.size(); ++i) { 7928 sp<Track> track = mTracks[i]; 7929 if (session == track->sessionId()) { 7930 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7931 track->setMainBuffer(buffer); 7932 chain->incTrackCnt(); 7933 } 7934 } 7935 7936 // indicate all active tracks in the chain 7937 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7938 sp<Track> track = mActiveTracks[i].promote(); 7939 if (track == 0) continue; 7940 if (session == track->sessionId()) { 7941 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7942 chain->incActiveTrackCnt(); 7943 } 7944 } 7945 } 7946 7947 chain->setInBuffer(buffer, ownsBuffer); 7948 chain->setOutBuffer(mMixBuffer); 7949 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7950 // chains list in order to be processed last as it contains output stage effects 7951 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7952 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7953 // after track specific effects and before output stage 7954 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7955 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7956 // Effect chain for other sessions are inserted at beginning of effect 7957 // chains list to be processed before output mix effects. Relative order between other 7958 // sessions is not important 7959 size_t size = mEffectChains.size(); 7960 size_t i = 0; 7961 for (i = 0; i < size; i++) { 7962 if (mEffectChains[i]->sessionId() < session) break; 7963 } 7964 mEffectChains.insertAt(chain, i); 7965 checkSuspendOnAddEffectChain_l(chain); 7966 7967 return NO_ERROR; 7968} 7969 7970size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7971{ 7972 int session = chain->sessionId(); 7973 7974 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7975 7976 for (size_t i = 0; i < mEffectChains.size(); i++) { 7977 if (chain == mEffectChains[i]) { 7978 mEffectChains.removeAt(i); 7979 // detach all active tracks from the chain 7980 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7981 sp<Track> track = mActiveTracks[i].promote(); 7982 if (track == 0) continue; 7983 if (session == track->sessionId()) { 7984 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7985 chain.get(), session); 7986 chain->decActiveTrackCnt(); 7987 } 7988 } 7989 7990 // detach all tracks with same session ID from this chain 7991 for (size_t i = 0; i < mTracks.size(); ++i) { 7992 sp<Track> track = mTracks[i]; 7993 if (session == track->sessionId()) { 7994 track->setMainBuffer(mMixBuffer); 7995 chain->decTrackCnt(); 7996 } 7997 } 7998 break; 7999 } 8000 } 8001 return mEffectChains.size(); 8002} 8003 8004status_t AudioFlinger::PlaybackThread::attachAuxEffect( 8005 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8006{ 8007 Mutex::Autolock _l(mLock); 8008 return attachAuxEffect_l(track, EffectId); 8009} 8010 8011status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 8012 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 8013{ 8014 status_t status = NO_ERROR; 8015 8016 if (EffectId == 0) { 8017 track->setAuxBuffer(0, NULL); 8018 } else { 8019 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 8020 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 8021 if (effect != 0) { 8022 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8023 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 8024 } else { 8025 status = INVALID_OPERATION; 8026 } 8027 } else { 8028 status = BAD_VALUE; 8029 } 8030 } 8031 return status; 8032} 8033 8034void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 8035{ 8036 for (size_t i = 0; i < mTracks.size(); ++i) { 8037 sp<Track> track = mTracks[i]; 8038 if (track->auxEffectId() == effectId) { 8039 attachAuxEffect_l(track, 0); 8040 } 8041 } 8042} 8043 8044status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 8045{ 8046 // only one chain per input thread 8047 if (mEffectChains.size() != 0) { 8048 return INVALID_OPERATION; 8049 } 8050 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 8051 8052 chain->setInBuffer(NULL); 8053 chain->setOutBuffer(NULL); 8054 8055 checkSuspendOnAddEffectChain_l(chain); 8056 8057 mEffectChains.add(chain); 8058 8059 return NO_ERROR; 8060} 8061 8062size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8063{ 8064 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8065 ALOGW_IF(mEffectChains.size() != 1, 8066 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8067 chain.get(), mEffectChains.size(), this); 8068 if (mEffectChains.size() == 1) { 8069 mEffectChains.removeAt(0); 8070 } 8071 return 0; 8072} 8073 8074// ---------------------------------------------------------------------------- 8075// EffectModule implementation 8076// ---------------------------------------------------------------------------- 8077 8078#undef LOG_TAG 8079#define LOG_TAG "AudioFlinger::EffectModule" 8080 8081AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8082 const wp<AudioFlinger::EffectChain>& chain, 8083 effect_descriptor_t *desc, 8084 int id, 8085 int sessionId) 8086 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8087 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8088 mDescriptor(*desc), 8089 // mConfig is set by configure() and not used before then 8090 mEffectInterface(NULL), 8091 mStatus(NO_INIT), mState(IDLE), 8092 // mMaxDisableWaitCnt is set by configure() and not used before then 8093 // mDisableWaitCnt is set by process() and updateState() and not used before then 8094 mSuspended(false) 8095{ 8096 ALOGV("Constructor %p", this); 8097 int lStatus; 8098 8099 // create effect engine from effect factory 8100 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8101 8102 if (mStatus != NO_ERROR) { 8103 return; 8104 } 8105 lStatus = init(); 8106 if (lStatus < 0) { 8107 mStatus = lStatus; 8108 goto Error; 8109 } 8110 8111 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8112 return; 8113Error: 8114 EffectRelease(mEffectInterface); 8115 mEffectInterface = NULL; 8116 ALOGV("Constructor Error %d", mStatus); 8117} 8118 8119AudioFlinger::EffectModule::~EffectModule() 8120{ 8121 ALOGV("Destructor %p", this); 8122 if (mEffectInterface != NULL) { 8123 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8124 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8125 sp<ThreadBase> thread = mThread.promote(); 8126 if (thread != 0) { 8127 audio_stream_t *stream = thread->stream(); 8128 if (stream != NULL) { 8129 stream->remove_audio_effect(stream, mEffectInterface); 8130 } 8131 } 8132 } 8133 // release effect engine 8134 EffectRelease(mEffectInterface); 8135 } 8136} 8137 8138status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8139{ 8140 status_t status; 8141 8142 Mutex::Autolock _l(mLock); 8143 int priority = handle->priority(); 8144 size_t size = mHandles.size(); 8145 EffectHandle *controlHandle = NULL; 8146 size_t i; 8147 for (i = 0; i < size; i++) { 8148 EffectHandle *h = mHandles[i]; 8149 if (h == NULL || h->destroyed_l()) continue; 8150 // first non destroyed handle is considered in control 8151 if (controlHandle == NULL) 8152 controlHandle = h; 8153 if (h->priority() <= priority) break; 8154 } 8155 // if inserted in first place, move effect control from previous owner to this handle 8156 if (i == 0) { 8157 bool enabled = false; 8158 if (controlHandle != NULL) { 8159 enabled = controlHandle->enabled(); 8160 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8161 } 8162 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8163 status = NO_ERROR; 8164 } else { 8165 status = ALREADY_EXISTS; 8166 } 8167 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8168 mHandles.insertAt(handle, i); 8169 return status; 8170} 8171 8172size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8173{ 8174 Mutex::Autolock _l(mLock); 8175 size_t size = mHandles.size(); 8176 size_t i; 8177 for (i = 0; i < size; i++) { 8178 if (mHandles[i] == handle) break; 8179 } 8180 if (i == size) { 8181 return size; 8182 } 8183 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8184 8185 mHandles.removeAt(i); 8186 // if removed from first place, move effect control from this handle to next in line 8187 if (i == 0) { 8188 EffectHandle *h = controlHandle_l(); 8189 if (h != NULL) { 8190 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8191 } 8192 } 8193 8194 // Prevent calls to process() and other functions on effect interface from now on. 8195 // The effect engine will be released by the destructor when the last strong reference on 8196 // this object is released which can happen after next process is called. 8197 if (mHandles.size() == 0 && !mPinned) { 8198 mState = DESTROYED; 8199 } 8200 8201 return mHandles.size(); 8202} 8203 8204// must be called with EffectModule::mLock held 8205AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8206{ 8207 // the first valid handle in the list has control over the module 8208 for (size_t i = 0; i < mHandles.size(); i++) { 8209 EffectHandle *h = mHandles[i]; 8210 if (h != NULL && !h->destroyed_l()) { 8211 return h; 8212 } 8213 } 8214 8215 return NULL; 8216} 8217 8218size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8219{ 8220 ALOGV("disconnect() %p handle %p", this, handle); 8221 // keep a strong reference on this EffectModule to avoid calling the 8222 // destructor before we exit 8223 sp<EffectModule> keep(this); 8224 { 8225 sp<ThreadBase> thread = mThread.promote(); 8226 if (thread != 0) { 8227 thread->disconnectEffect(keep, handle, unpinIfLast); 8228 } 8229 } 8230 return mHandles.size(); 8231} 8232 8233void AudioFlinger::EffectModule::updateState() { 8234 Mutex::Autolock _l(mLock); 8235 8236 switch (mState) { 8237 case RESTART: 8238 reset_l(); 8239 // FALL THROUGH 8240 8241 case STARTING: 8242 // clear auxiliary effect input buffer for next accumulation 8243 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8244 memset(mConfig.inputCfg.buffer.raw, 8245 0, 8246 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8247 } 8248 start_l(); 8249 mState = ACTIVE; 8250 break; 8251 case STOPPING: 8252 stop_l(); 8253 mDisableWaitCnt = mMaxDisableWaitCnt; 8254 mState = STOPPED; 8255 break; 8256 case STOPPED: 8257 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8258 // turn off sequence. 8259 if (--mDisableWaitCnt == 0) { 8260 reset_l(); 8261 mState = IDLE; 8262 } 8263 break; 8264 default: //IDLE , ACTIVE, DESTROYED 8265 break; 8266 } 8267} 8268 8269void AudioFlinger::EffectModule::process() 8270{ 8271 Mutex::Autolock _l(mLock); 8272 8273 if (mState == DESTROYED || mEffectInterface == NULL || 8274 mConfig.inputCfg.buffer.raw == NULL || 8275 mConfig.outputCfg.buffer.raw == NULL) { 8276 return; 8277 } 8278 8279 if (isProcessEnabled()) { 8280 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8281 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8282 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8283 mConfig.inputCfg.buffer.s32, 8284 mConfig.inputCfg.buffer.frameCount/2); 8285 } 8286 8287 // do the actual processing in the effect engine 8288 int ret = (*mEffectInterface)->process(mEffectInterface, 8289 &mConfig.inputCfg.buffer, 8290 &mConfig.outputCfg.buffer); 8291 8292 // force transition to IDLE state when engine is ready 8293 if (mState == STOPPED && ret == -ENODATA) { 8294 mDisableWaitCnt = 1; 8295 } 8296 8297 // clear auxiliary effect input buffer for next accumulation 8298 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8299 memset(mConfig.inputCfg.buffer.raw, 0, 8300 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8301 } 8302 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8303 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8304 // If an insert effect is idle and input buffer is different from output buffer, 8305 // accumulate input onto output 8306 sp<EffectChain> chain = mChain.promote(); 8307 if (chain != 0 && chain->activeTrackCnt() != 0) { 8308 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8309 int16_t *in = mConfig.inputCfg.buffer.s16; 8310 int16_t *out = mConfig.outputCfg.buffer.s16; 8311 for (size_t i = 0; i < frameCnt; i++) { 8312 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8313 } 8314 } 8315 } 8316} 8317 8318void AudioFlinger::EffectModule::reset_l() 8319{ 8320 if (mEffectInterface == NULL) { 8321 return; 8322 } 8323 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8324} 8325 8326status_t AudioFlinger::EffectModule::configure() 8327{ 8328 if (mEffectInterface == NULL) { 8329 return NO_INIT; 8330 } 8331 8332 sp<ThreadBase> thread = mThread.promote(); 8333 if (thread == 0) { 8334 return DEAD_OBJECT; 8335 } 8336 8337 // TODO: handle configuration of effects replacing track process 8338 audio_channel_mask_t channelMask = thread->channelMask(); 8339 8340 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8341 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8342 } else { 8343 mConfig.inputCfg.channels = channelMask; 8344 } 8345 mConfig.outputCfg.channels = channelMask; 8346 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8347 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8348 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8349 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8350 mConfig.inputCfg.bufferProvider.cookie = NULL; 8351 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8352 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8353 mConfig.outputCfg.bufferProvider.cookie = NULL; 8354 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8355 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8356 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8357 // Insert effect: 8358 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8359 // always overwrites output buffer: input buffer == output buffer 8360 // - in other sessions: 8361 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8362 // other effect: overwrites output buffer: input buffer == output buffer 8363 // Auxiliary effect: 8364 // accumulates in output buffer: input buffer != output buffer 8365 // Therefore: accumulate <=> input buffer != output buffer 8366 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8367 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8368 } else { 8369 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8370 } 8371 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8372 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8373 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8374 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8375 8376 ALOGV("configure() %p thread %p buffer %p framecount %d", 8377 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8378 8379 status_t cmdStatus; 8380 uint32_t size = sizeof(int); 8381 status_t status = (*mEffectInterface)->command(mEffectInterface, 8382 EFFECT_CMD_SET_CONFIG, 8383 sizeof(effect_config_t), 8384 &mConfig, 8385 &size, 8386 &cmdStatus); 8387 if (status == 0) { 8388 status = cmdStatus; 8389 } 8390 8391 if (status == 0 && 8392 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8393 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8394 effect_param_t *p = (effect_param_t *)buf32; 8395 8396 p->psize = sizeof(uint32_t); 8397 p->vsize = sizeof(uint32_t); 8398 size = sizeof(int); 8399 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8400 8401 uint32_t latency = 0; 8402 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8403 if (pbt != NULL) { 8404 latency = pbt->latency_l(); 8405 } 8406 8407 *((int32_t *)p->data + 1)= latency; 8408 (*mEffectInterface)->command(mEffectInterface, 8409 EFFECT_CMD_SET_PARAM, 8410 sizeof(effect_param_t) + 8, 8411 &buf32, 8412 &size, 8413 &cmdStatus); 8414 } 8415 8416 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8417 (1000 * mConfig.outputCfg.buffer.frameCount); 8418 8419 return status; 8420} 8421 8422status_t AudioFlinger::EffectModule::init() 8423{ 8424 Mutex::Autolock _l(mLock); 8425 if (mEffectInterface == NULL) { 8426 return NO_INIT; 8427 } 8428 status_t cmdStatus; 8429 uint32_t size = sizeof(status_t); 8430 status_t status = (*mEffectInterface)->command(mEffectInterface, 8431 EFFECT_CMD_INIT, 8432 0, 8433 NULL, 8434 &size, 8435 &cmdStatus); 8436 if (status == 0) { 8437 status = cmdStatus; 8438 } 8439 return status; 8440} 8441 8442status_t AudioFlinger::EffectModule::start() 8443{ 8444 Mutex::Autolock _l(mLock); 8445 return start_l(); 8446} 8447 8448status_t AudioFlinger::EffectModule::start_l() 8449{ 8450 if (mEffectInterface == NULL) { 8451 return NO_INIT; 8452 } 8453 status_t cmdStatus; 8454 uint32_t size = sizeof(status_t); 8455 status_t status = (*mEffectInterface)->command(mEffectInterface, 8456 EFFECT_CMD_ENABLE, 8457 0, 8458 NULL, 8459 &size, 8460 &cmdStatus); 8461 if (status == 0) { 8462 status = cmdStatus; 8463 } 8464 if (status == 0 && 8465 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8466 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8467 sp<ThreadBase> thread = mThread.promote(); 8468 if (thread != 0) { 8469 audio_stream_t *stream = thread->stream(); 8470 if (stream != NULL) { 8471 stream->add_audio_effect(stream, mEffectInterface); 8472 } 8473 } 8474 } 8475 return status; 8476} 8477 8478status_t AudioFlinger::EffectModule::stop() 8479{ 8480 Mutex::Autolock _l(mLock); 8481 return stop_l(); 8482} 8483 8484status_t AudioFlinger::EffectModule::stop_l() 8485{ 8486 if (mEffectInterface == NULL) { 8487 return NO_INIT; 8488 } 8489 status_t cmdStatus; 8490 uint32_t size = sizeof(status_t); 8491 status_t status = (*mEffectInterface)->command(mEffectInterface, 8492 EFFECT_CMD_DISABLE, 8493 0, 8494 NULL, 8495 &size, 8496 &cmdStatus); 8497 if (status == 0) { 8498 status = cmdStatus; 8499 } 8500 if (status == 0 && 8501 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8502 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8503 sp<ThreadBase> thread = mThread.promote(); 8504 if (thread != 0) { 8505 audio_stream_t *stream = thread->stream(); 8506 if (stream != NULL) { 8507 stream->remove_audio_effect(stream, mEffectInterface); 8508 } 8509 } 8510 } 8511 return status; 8512} 8513 8514status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8515 uint32_t cmdSize, 8516 void *pCmdData, 8517 uint32_t *replySize, 8518 void *pReplyData) 8519{ 8520 Mutex::Autolock _l(mLock); 8521// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8522 8523 if (mState == DESTROYED || mEffectInterface == NULL) { 8524 return NO_INIT; 8525 } 8526 status_t status = (*mEffectInterface)->command(mEffectInterface, 8527 cmdCode, 8528 cmdSize, 8529 pCmdData, 8530 replySize, 8531 pReplyData); 8532 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8533 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8534 for (size_t i = 1; i < mHandles.size(); i++) { 8535 EffectHandle *h = mHandles[i]; 8536 if (h != NULL && !h->destroyed_l()) { 8537 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8538 } 8539 } 8540 } 8541 return status; 8542} 8543 8544status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8545{ 8546 Mutex::Autolock _l(mLock); 8547 return setEnabled_l(enabled); 8548} 8549 8550// must be called with EffectModule::mLock held 8551status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8552{ 8553 8554 ALOGV("setEnabled %p enabled %d", this, enabled); 8555 8556 if (enabled != isEnabled()) { 8557 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8558 if (enabled && status != NO_ERROR) { 8559 return status; 8560 } 8561 8562 switch (mState) { 8563 // going from disabled to enabled 8564 case IDLE: 8565 mState = STARTING; 8566 break; 8567 case STOPPED: 8568 mState = RESTART; 8569 break; 8570 case STOPPING: 8571 mState = ACTIVE; 8572 break; 8573 8574 // going from enabled to disabled 8575 case RESTART: 8576 mState = STOPPED; 8577 break; 8578 case STARTING: 8579 mState = IDLE; 8580 break; 8581 case ACTIVE: 8582 mState = STOPPING; 8583 break; 8584 case DESTROYED: 8585 return NO_ERROR; // simply ignore as we are being destroyed 8586 } 8587 for (size_t i = 1; i < mHandles.size(); i++) { 8588 EffectHandle *h = mHandles[i]; 8589 if (h != NULL && !h->destroyed_l()) { 8590 h->setEnabled(enabled); 8591 } 8592 } 8593 } 8594 return NO_ERROR; 8595} 8596 8597bool AudioFlinger::EffectModule::isEnabled() const 8598{ 8599 switch (mState) { 8600 case RESTART: 8601 case STARTING: 8602 case ACTIVE: 8603 return true; 8604 case IDLE: 8605 case STOPPING: 8606 case STOPPED: 8607 case DESTROYED: 8608 default: 8609 return false; 8610 } 8611} 8612 8613bool AudioFlinger::EffectModule::isProcessEnabled() const 8614{ 8615 switch (mState) { 8616 case RESTART: 8617 case ACTIVE: 8618 case STOPPING: 8619 case STOPPED: 8620 return true; 8621 case IDLE: 8622 case STARTING: 8623 case DESTROYED: 8624 default: 8625 return false; 8626 } 8627} 8628 8629status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8630{ 8631 Mutex::Autolock _l(mLock); 8632 status_t status = NO_ERROR; 8633 8634 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8635 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8636 if (isProcessEnabled() && 8637 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8638 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8639 status_t cmdStatus; 8640 uint32_t volume[2]; 8641 uint32_t *pVolume = NULL; 8642 uint32_t size = sizeof(volume); 8643 volume[0] = *left; 8644 volume[1] = *right; 8645 if (controller) { 8646 pVolume = volume; 8647 } 8648 status = (*mEffectInterface)->command(mEffectInterface, 8649 EFFECT_CMD_SET_VOLUME, 8650 size, 8651 volume, 8652 &size, 8653 pVolume); 8654 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8655 *left = volume[0]; 8656 *right = volume[1]; 8657 } 8658 } 8659 return status; 8660} 8661 8662status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8663{ 8664 Mutex::Autolock _l(mLock); 8665 status_t status = NO_ERROR; 8666 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8667 // audio pre processing modules on RecordThread can receive both output and 8668 // input device indication in the same call 8669 audio_devices_t dev = device & AUDIO_DEVICE_OUT_ALL; 8670 if (dev) { 8671 status_t cmdStatus; 8672 uint32_t size = sizeof(status_t); 8673 8674 status = (*mEffectInterface)->command(mEffectInterface, 8675 EFFECT_CMD_SET_DEVICE, 8676 sizeof(uint32_t), 8677 &dev, 8678 &size, 8679 &cmdStatus); 8680 if (status == NO_ERROR) { 8681 status = cmdStatus; 8682 } 8683 } 8684 dev = device & AUDIO_DEVICE_IN_ALL; 8685 if (dev) { 8686 status_t cmdStatus; 8687 uint32_t size = sizeof(status_t); 8688 8689 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8690 EFFECT_CMD_SET_INPUT_DEVICE, 8691 sizeof(uint32_t), 8692 &dev, 8693 &size, 8694 &cmdStatus); 8695 if (status2 == NO_ERROR) { 8696 status2 = cmdStatus; 8697 } 8698 if (status == NO_ERROR) { 8699 status = status2; 8700 } 8701 } 8702 } 8703 return status; 8704} 8705 8706status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8707{ 8708 Mutex::Autolock _l(mLock); 8709 status_t status = NO_ERROR; 8710 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8711 status_t cmdStatus; 8712 uint32_t size = sizeof(status_t); 8713 status = (*mEffectInterface)->command(mEffectInterface, 8714 EFFECT_CMD_SET_AUDIO_MODE, 8715 sizeof(audio_mode_t), 8716 &mode, 8717 &size, 8718 &cmdStatus); 8719 if (status == NO_ERROR) { 8720 status = cmdStatus; 8721 } 8722 } 8723 return status; 8724} 8725 8726status_t AudioFlinger::EffectModule::setAudioSource(audio_source_t source) 8727{ 8728 Mutex::Autolock _l(mLock); 8729 status_t status = NO_ERROR; 8730 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_SOURCE_MASK) == EFFECT_FLAG_AUDIO_SOURCE_IND) { 8731 uint32_t size = 0; 8732 status = (*mEffectInterface)->command(mEffectInterface, 8733 EFFECT_CMD_SET_AUDIO_SOURCE, 8734 sizeof(audio_source_t), 8735 &source, 8736 &size, 8737 NULL); 8738 } 8739 return status; 8740} 8741 8742void AudioFlinger::EffectModule::setSuspended(bool suspended) 8743{ 8744 Mutex::Autolock _l(mLock); 8745 mSuspended = suspended; 8746} 8747 8748bool AudioFlinger::EffectModule::suspended() const 8749{ 8750 Mutex::Autolock _l(mLock); 8751 return mSuspended; 8752} 8753 8754bool AudioFlinger::EffectModule::purgeHandles() 8755{ 8756 bool enabled = false; 8757 Mutex::Autolock _l(mLock); 8758 for (size_t i = 0; i < mHandles.size(); i++) { 8759 EffectHandle *handle = mHandles[i]; 8760 if (handle != NULL && !handle->destroyed_l()) { 8761 handle->effect().clear(); 8762 if (handle->hasControl()) { 8763 enabled = handle->enabled(); 8764 } 8765 } 8766 } 8767 return enabled; 8768} 8769 8770void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8771{ 8772 const size_t SIZE = 256; 8773 char buffer[SIZE]; 8774 String8 result; 8775 8776 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8777 result.append(buffer); 8778 8779 bool locked = tryLock(mLock); 8780 // failed to lock - AudioFlinger is probably deadlocked 8781 if (!locked) { 8782 result.append("\t\tCould not lock Fx mutex:\n"); 8783 } 8784 8785 result.append("\t\tSession Status State Engine:\n"); 8786 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8787 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8788 result.append(buffer); 8789 8790 result.append("\t\tDescriptor:\n"); 8791 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8792 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8793 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8794 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8795 result.append(buffer); 8796 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8797 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8798 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8799 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8800 result.append(buffer); 8801 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8802 mDescriptor.apiVersion, 8803 mDescriptor.flags); 8804 result.append(buffer); 8805 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8806 mDescriptor.name); 8807 result.append(buffer); 8808 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8809 mDescriptor.implementor); 8810 result.append(buffer); 8811 8812 result.append("\t\t- Input configuration:\n"); 8813 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8814 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8815 (uint32_t)mConfig.inputCfg.buffer.raw, 8816 mConfig.inputCfg.buffer.frameCount, 8817 mConfig.inputCfg.samplingRate, 8818 mConfig.inputCfg.channels, 8819 mConfig.inputCfg.format); 8820 result.append(buffer); 8821 8822 result.append("\t\t- Output configuration:\n"); 8823 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8824 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8825 (uint32_t)mConfig.outputCfg.buffer.raw, 8826 mConfig.outputCfg.buffer.frameCount, 8827 mConfig.outputCfg.samplingRate, 8828 mConfig.outputCfg.channels, 8829 mConfig.outputCfg.format); 8830 result.append(buffer); 8831 8832 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8833 result.append(buffer); 8834 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8835 for (size_t i = 0; i < mHandles.size(); ++i) { 8836 EffectHandle *handle = mHandles[i]; 8837 if (handle != NULL && !handle->destroyed_l()) { 8838 handle->dump(buffer, SIZE); 8839 result.append(buffer); 8840 } 8841 } 8842 8843 result.append("\n"); 8844 8845 write(fd, result.string(), result.length()); 8846 8847 if (locked) { 8848 mLock.unlock(); 8849 } 8850} 8851 8852// ---------------------------------------------------------------------------- 8853// EffectHandle implementation 8854// ---------------------------------------------------------------------------- 8855 8856#undef LOG_TAG 8857#define LOG_TAG "AudioFlinger::EffectHandle" 8858 8859AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8860 const sp<AudioFlinger::Client>& client, 8861 const sp<IEffectClient>& effectClient, 8862 int32_t priority) 8863 : BnEffect(), 8864 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8865 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8866{ 8867 ALOGV("constructor %p", this); 8868 8869 if (client == 0) { 8870 return; 8871 } 8872 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8873 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8874 if (mCblkMemory != 0) { 8875 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8876 8877 if (mCblk != NULL) { 8878 new(mCblk) effect_param_cblk_t(); 8879 mBuffer = (uint8_t *)mCblk + bufOffset; 8880 } 8881 } else { 8882 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8883 return; 8884 } 8885} 8886 8887AudioFlinger::EffectHandle::~EffectHandle() 8888{ 8889 ALOGV("Destructor %p", this); 8890 8891 if (mEffect == 0) { 8892 mDestroyed = true; 8893 return; 8894 } 8895 mEffect->lock(); 8896 mDestroyed = true; 8897 mEffect->unlock(); 8898 disconnect(false); 8899} 8900 8901status_t AudioFlinger::EffectHandle::enable() 8902{ 8903 ALOGV("enable %p", this); 8904 if (!mHasControl) return INVALID_OPERATION; 8905 if (mEffect == 0) return DEAD_OBJECT; 8906 8907 if (mEnabled) { 8908 return NO_ERROR; 8909 } 8910 8911 mEnabled = true; 8912 8913 sp<ThreadBase> thread = mEffect->thread().promote(); 8914 if (thread != 0) { 8915 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8916 } 8917 8918 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8919 if (mEffect->suspended()) { 8920 return NO_ERROR; 8921 } 8922 8923 status_t status = mEffect->setEnabled(true); 8924 if (status != NO_ERROR) { 8925 if (thread != 0) { 8926 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8927 } 8928 mEnabled = false; 8929 } 8930 return status; 8931} 8932 8933status_t AudioFlinger::EffectHandle::disable() 8934{ 8935 ALOGV("disable %p", this); 8936 if (!mHasControl) return INVALID_OPERATION; 8937 if (mEffect == 0) return DEAD_OBJECT; 8938 8939 if (!mEnabled) { 8940 return NO_ERROR; 8941 } 8942 mEnabled = false; 8943 8944 if (mEffect->suspended()) { 8945 return NO_ERROR; 8946 } 8947 8948 status_t status = mEffect->setEnabled(false); 8949 8950 sp<ThreadBase> thread = mEffect->thread().promote(); 8951 if (thread != 0) { 8952 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8953 } 8954 8955 return status; 8956} 8957 8958void AudioFlinger::EffectHandle::disconnect() 8959{ 8960 disconnect(true); 8961} 8962 8963void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8964{ 8965 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8966 if (mEffect == 0) { 8967 return; 8968 } 8969 // restore suspended effects if the disconnected handle was enabled and the last one. 8970 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8971 sp<ThreadBase> thread = mEffect->thread().promote(); 8972 if (thread != 0) { 8973 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8974 } 8975 } 8976 8977 // release sp on module => module destructor can be called now 8978 mEffect.clear(); 8979 if (mClient != 0) { 8980 if (mCblk != NULL) { 8981 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8982 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8983 } 8984 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8985 // Client destructor must run with AudioFlinger mutex locked 8986 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8987 mClient.clear(); 8988 } 8989} 8990 8991status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8992 uint32_t cmdSize, 8993 void *pCmdData, 8994 uint32_t *replySize, 8995 void *pReplyData) 8996{ 8997// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8998// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8999 9000 // only get parameter command is permitted for applications not controlling the effect 9001 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 9002 return INVALID_OPERATION; 9003 } 9004 if (mEffect == 0) return DEAD_OBJECT; 9005 if (mClient == 0) return INVALID_OPERATION; 9006 9007 // handle commands that are not forwarded transparently to effect engine 9008 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 9009 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 9010 // no risk to block the whole media server process or mixer threads is we are stuck here 9011 Mutex::Autolock _l(mCblk->lock); 9012 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 9013 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 9014 mCblk->serverIndex = 0; 9015 mCblk->clientIndex = 0; 9016 return BAD_VALUE; 9017 } 9018 status_t status = NO_ERROR; 9019 while (mCblk->serverIndex < mCblk->clientIndex) { 9020 int reply; 9021 uint32_t rsize = sizeof(int); 9022 int *p = (int *)(mBuffer + mCblk->serverIndex); 9023 int size = *p++; 9024 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 9025 ALOGW("command(): invalid parameter block size"); 9026 break; 9027 } 9028 effect_param_t *param = (effect_param_t *)p; 9029 if (param->psize == 0 || param->vsize == 0) { 9030 ALOGW("command(): null parameter or value size"); 9031 mCblk->serverIndex += size; 9032 continue; 9033 } 9034 uint32_t psize = sizeof(effect_param_t) + 9035 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 9036 param->vsize; 9037 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 9038 psize, 9039 p, 9040 &rsize, 9041 &reply); 9042 // stop at first error encountered 9043 if (ret != NO_ERROR) { 9044 status = ret; 9045 *(int *)pReplyData = reply; 9046 break; 9047 } else if (reply != NO_ERROR) { 9048 *(int *)pReplyData = reply; 9049 break; 9050 } 9051 mCblk->serverIndex += size; 9052 } 9053 mCblk->serverIndex = 0; 9054 mCblk->clientIndex = 0; 9055 return status; 9056 } else if (cmdCode == EFFECT_CMD_ENABLE) { 9057 *(int *)pReplyData = NO_ERROR; 9058 return enable(); 9059 } else if (cmdCode == EFFECT_CMD_DISABLE) { 9060 *(int *)pReplyData = NO_ERROR; 9061 return disable(); 9062 } 9063 9064 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9065} 9066 9067void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 9068{ 9069 ALOGV("setControl %p control %d", this, hasControl); 9070 9071 mHasControl = hasControl; 9072 mEnabled = enabled; 9073 9074 if (signal && mEffectClient != 0) { 9075 mEffectClient->controlStatusChanged(hasControl); 9076 } 9077} 9078 9079void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9080 uint32_t cmdSize, 9081 void *pCmdData, 9082 uint32_t replySize, 9083 void *pReplyData) 9084{ 9085 if (mEffectClient != 0) { 9086 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9087 } 9088} 9089 9090 9091 9092void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9093{ 9094 if (mEffectClient != 0) { 9095 mEffectClient->enableStatusChanged(enabled); 9096 } 9097} 9098 9099status_t AudioFlinger::EffectHandle::onTransact( 9100 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9101{ 9102 return BnEffect::onTransact(code, data, reply, flags); 9103} 9104 9105 9106void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9107{ 9108 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9109 9110 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9111 (mClient == 0) ? getpid_cached : mClient->pid(), 9112 mPriority, 9113 mHasControl, 9114 !locked, 9115 mCblk ? mCblk->clientIndex : 0, 9116 mCblk ? mCblk->serverIndex : 0 9117 ); 9118 9119 if (locked) { 9120 mCblk->lock.unlock(); 9121 } 9122} 9123 9124#undef LOG_TAG 9125#define LOG_TAG "AudioFlinger::EffectChain" 9126 9127AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9128 int sessionId) 9129 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9130 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9131 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9132{ 9133 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9134 if (thread == NULL) { 9135 return; 9136 } 9137 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9138 thread->frameCount(); 9139} 9140 9141AudioFlinger::EffectChain::~EffectChain() 9142{ 9143 if (mOwnInBuffer) { 9144 delete mInBuffer; 9145 } 9146 9147} 9148 9149// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9150sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9151{ 9152 size_t size = mEffects.size(); 9153 9154 for (size_t i = 0; i < size; i++) { 9155 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9156 return mEffects[i]; 9157 } 9158 } 9159 return 0; 9160} 9161 9162// getEffectFromId_l() must be called with ThreadBase::mLock held 9163sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9164{ 9165 size_t size = mEffects.size(); 9166 9167 for (size_t i = 0; i < size; i++) { 9168 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9169 if (id == 0 || mEffects[i]->id() == id) { 9170 return mEffects[i]; 9171 } 9172 } 9173 return 0; 9174} 9175 9176// getEffectFromType_l() must be called with ThreadBase::mLock held 9177sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9178 const effect_uuid_t *type) 9179{ 9180 size_t size = mEffects.size(); 9181 9182 for (size_t i = 0; i < size; i++) { 9183 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9184 return mEffects[i]; 9185 } 9186 } 9187 return 0; 9188} 9189 9190void AudioFlinger::EffectChain::clearInputBuffer() 9191{ 9192 Mutex::Autolock _l(mLock); 9193 sp<ThreadBase> thread = mThread.promote(); 9194 if (thread == 0) { 9195 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9196 return; 9197 } 9198 clearInputBuffer_l(thread); 9199} 9200 9201// Must be called with EffectChain::mLock locked 9202void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9203{ 9204 size_t numSamples = thread->frameCount() * thread->channelCount(); 9205 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9206 9207} 9208 9209// Must be called with EffectChain::mLock locked 9210void AudioFlinger::EffectChain::process_l() 9211{ 9212 sp<ThreadBase> thread = mThread.promote(); 9213 if (thread == 0) { 9214 ALOGW("process_l(): cannot promote mixer thread"); 9215 return; 9216 } 9217 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9218 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9219 // always process effects unless no more tracks are on the session and the effect tail 9220 // has been rendered 9221 bool doProcess = true; 9222 if (!isGlobalSession) { 9223 bool tracksOnSession = (trackCnt() != 0); 9224 9225 if (!tracksOnSession && mTailBufferCount == 0) { 9226 doProcess = false; 9227 } 9228 9229 if (activeTrackCnt() == 0) { 9230 // if no track is active and the effect tail has not been rendered, 9231 // the input buffer must be cleared here as the mixer process will not do it 9232 if (tracksOnSession || mTailBufferCount > 0) { 9233 clearInputBuffer_l(thread); 9234 if (mTailBufferCount > 0) { 9235 mTailBufferCount--; 9236 } 9237 } 9238 } 9239 } 9240 9241 size_t size = mEffects.size(); 9242 if (doProcess) { 9243 for (size_t i = 0; i < size; i++) { 9244 mEffects[i]->process(); 9245 } 9246 } 9247 for (size_t i = 0; i < size; i++) { 9248 mEffects[i]->updateState(); 9249 } 9250} 9251 9252// addEffect_l() must be called with PlaybackThread::mLock held 9253status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9254{ 9255 effect_descriptor_t desc = effect->desc(); 9256 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9257 9258 Mutex::Autolock _l(mLock); 9259 effect->setChain(this); 9260 sp<ThreadBase> thread = mThread.promote(); 9261 if (thread == 0) { 9262 return NO_INIT; 9263 } 9264 effect->setThread(thread); 9265 9266 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9267 // Auxiliary effects are inserted at the beginning of mEffects vector as 9268 // they are processed first and accumulated in chain input buffer 9269 mEffects.insertAt(effect, 0); 9270 9271 // the input buffer for auxiliary effect contains mono samples in 9272 // 32 bit format. This is to avoid saturation in AudoMixer 9273 // accumulation stage. Saturation is done in EffectModule::process() before 9274 // calling the process in effect engine 9275 size_t numSamples = thread->frameCount(); 9276 int32_t *buffer = new int32_t[numSamples]; 9277 memset(buffer, 0, numSamples * sizeof(int32_t)); 9278 effect->setInBuffer((int16_t *)buffer); 9279 // auxiliary effects output samples to chain input buffer for further processing 9280 // by insert effects 9281 effect->setOutBuffer(mInBuffer); 9282 } else { 9283 // Insert effects are inserted at the end of mEffects vector as they are processed 9284 // after track and auxiliary effects. 9285 // Insert effect order as a function of indicated preference: 9286 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9287 // another effect is present 9288 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9289 // last effect claiming first position 9290 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9291 // first effect claiming last position 9292 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9293 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9294 // already present 9295 9296 size_t size = mEffects.size(); 9297 size_t idx_insert = size; 9298 ssize_t idx_insert_first = -1; 9299 ssize_t idx_insert_last = -1; 9300 9301 for (size_t i = 0; i < size; i++) { 9302 effect_descriptor_t d = mEffects[i]->desc(); 9303 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9304 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9305 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9306 // check invalid effect chaining combinations 9307 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9308 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9309 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9310 return INVALID_OPERATION; 9311 } 9312 // remember position of first insert effect and by default 9313 // select this as insert position for new effect 9314 if (idx_insert == size) { 9315 idx_insert = i; 9316 } 9317 // remember position of last insert effect claiming 9318 // first position 9319 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9320 idx_insert_first = i; 9321 } 9322 // remember position of first insert effect claiming 9323 // last position 9324 if (iPref == EFFECT_FLAG_INSERT_LAST && 9325 idx_insert_last == -1) { 9326 idx_insert_last = i; 9327 } 9328 } 9329 } 9330 9331 // modify idx_insert from first position if needed 9332 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9333 if (idx_insert_last != -1) { 9334 idx_insert = idx_insert_last; 9335 } else { 9336 idx_insert = size; 9337 } 9338 } else { 9339 if (idx_insert_first != -1) { 9340 idx_insert = idx_insert_first + 1; 9341 } 9342 } 9343 9344 // always read samples from chain input buffer 9345 effect->setInBuffer(mInBuffer); 9346 9347 // if last effect in the chain, output samples to chain 9348 // output buffer, otherwise to chain input buffer 9349 if (idx_insert == size) { 9350 if (idx_insert != 0) { 9351 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9352 mEffects[idx_insert-1]->configure(); 9353 } 9354 effect->setOutBuffer(mOutBuffer); 9355 } else { 9356 effect->setOutBuffer(mInBuffer); 9357 } 9358 mEffects.insertAt(effect, idx_insert); 9359 9360 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9361 } 9362 effect->configure(); 9363 return NO_ERROR; 9364} 9365 9366// removeEffect_l() must be called with PlaybackThread::mLock held 9367size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9368{ 9369 Mutex::Autolock _l(mLock); 9370 size_t size = mEffects.size(); 9371 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9372 9373 for (size_t i = 0; i < size; i++) { 9374 if (effect == mEffects[i]) { 9375 // calling stop here will remove pre-processing effect from the audio HAL. 9376 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9377 // the middle of a read from audio HAL 9378 if (mEffects[i]->state() == EffectModule::ACTIVE || 9379 mEffects[i]->state() == EffectModule::STOPPING) { 9380 mEffects[i]->stop(); 9381 } 9382 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9383 delete[] effect->inBuffer(); 9384 } else { 9385 if (i == size - 1 && i != 0) { 9386 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9387 mEffects[i - 1]->configure(); 9388 } 9389 } 9390 mEffects.removeAt(i); 9391 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9392 break; 9393 } 9394 } 9395 9396 return mEffects.size(); 9397} 9398 9399// setDevice_l() must be called with PlaybackThread::mLock held 9400void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9401{ 9402 size_t size = mEffects.size(); 9403 for (size_t i = 0; i < size; i++) { 9404 mEffects[i]->setDevice(device); 9405 } 9406} 9407 9408// setMode_l() must be called with PlaybackThread::mLock held 9409void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9410{ 9411 size_t size = mEffects.size(); 9412 for (size_t i = 0; i < size; i++) { 9413 mEffects[i]->setMode(mode); 9414 } 9415} 9416 9417// setAudioSource_l() must be called with PlaybackThread::mLock held 9418void AudioFlinger::EffectChain::setAudioSource_l(audio_source_t source) 9419{ 9420 size_t size = mEffects.size(); 9421 for (size_t i = 0; i < size; i++) { 9422 mEffects[i]->setAudioSource(source); 9423 } 9424} 9425 9426// setVolume_l() must be called with PlaybackThread::mLock held 9427bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9428{ 9429 uint32_t newLeft = *left; 9430 uint32_t newRight = *right; 9431 bool hasControl = false; 9432 int ctrlIdx = -1; 9433 size_t size = mEffects.size(); 9434 9435 // first update volume controller 9436 for (size_t i = size; i > 0; i--) { 9437 if (mEffects[i - 1]->isProcessEnabled() && 9438 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9439 ctrlIdx = i - 1; 9440 hasControl = true; 9441 break; 9442 } 9443 } 9444 9445 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9446 if (hasControl) { 9447 *left = mNewLeftVolume; 9448 *right = mNewRightVolume; 9449 } 9450 return hasControl; 9451 } 9452 9453 mVolumeCtrlIdx = ctrlIdx; 9454 mLeftVolume = newLeft; 9455 mRightVolume = newRight; 9456 9457 // second get volume update from volume controller 9458 if (ctrlIdx >= 0) { 9459 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9460 mNewLeftVolume = newLeft; 9461 mNewRightVolume = newRight; 9462 } 9463 // then indicate volume to all other effects in chain. 9464 // Pass altered volume to effects before volume controller 9465 // and requested volume to effects after controller 9466 uint32_t lVol = newLeft; 9467 uint32_t rVol = newRight; 9468 9469 for (size_t i = 0; i < size; i++) { 9470 if ((int)i == ctrlIdx) continue; 9471 // this also works for ctrlIdx == -1 when there is no volume controller 9472 if ((int)i > ctrlIdx) { 9473 lVol = *left; 9474 rVol = *right; 9475 } 9476 mEffects[i]->setVolume(&lVol, &rVol, false); 9477 } 9478 *left = newLeft; 9479 *right = newRight; 9480 9481 return hasControl; 9482} 9483 9484void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9485{ 9486 const size_t SIZE = 256; 9487 char buffer[SIZE]; 9488 String8 result; 9489 9490 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9491 result.append(buffer); 9492 9493 bool locked = tryLock(mLock); 9494 // failed to lock - AudioFlinger is probably deadlocked 9495 if (!locked) { 9496 result.append("\tCould not lock mutex:\n"); 9497 } 9498 9499 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9500 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9501 mEffects.size(), 9502 (uint32_t)mInBuffer, 9503 (uint32_t)mOutBuffer, 9504 mActiveTrackCnt); 9505 result.append(buffer); 9506 write(fd, result.string(), result.size()); 9507 9508 for (size_t i = 0; i < mEffects.size(); ++i) { 9509 sp<EffectModule> effect = mEffects[i]; 9510 if (effect != 0) { 9511 effect->dump(fd, args); 9512 } 9513 } 9514 9515 if (locked) { 9516 mLock.unlock(); 9517 } 9518} 9519 9520// must be called with ThreadBase::mLock held 9521void AudioFlinger::EffectChain::setEffectSuspended_l( 9522 const effect_uuid_t *type, bool suspend) 9523{ 9524 sp<SuspendedEffectDesc> desc; 9525 // use effect type UUID timelow as key as there is no real risk of identical 9526 // timeLow fields among effect type UUIDs. 9527 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9528 if (suspend) { 9529 if (index >= 0) { 9530 desc = mSuspendedEffects.valueAt(index); 9531 } else { 9532 desc = new SuspendedEffectDesc(); 9533 desc->mType = *type; 9534 mSuspendedEffects.add(type->timeLow, desc); 9535 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9536 } 9537 if (desc->mRefCount++ == 0) { 9538 sp<EffectModule> effect = getEffectIfEnabled(type); 9539 if (effect != 0) { 9540 desc->mEffect = effect; 9541 effect->setSuspended(true); 9542 effect->setEnabled(false); 9543 } 9544 } 9545 } else { 9546 if (index < 0) { 9547 return; 9548 } 9549 desc = mSuspendedEffects.valueAt(index); 9550 if (desc->mRefCount <= 0) { 9551 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9552 desc->mRefCount = 1; 9553 } 9554 if (--desc->mRefCount == 0) { 9555 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9556 if (desc->mEffect != 0) { 9557 sp<EffectModule> effect = desc->mEffect.promote(); 9558 if (effect != 0) { 9559 effect->setSuspended(false); 9560 effect->lock(); 9561 EffectHandle *handle = effect->controlHandle_l(); 9562 if (handle != NULL && !handle->destroyed_l()) { 9563 effect->setEnabled_l(handle->enabled()); 9564 } 9565 effect->unlock(); 9566 } 9567 desc->mEffect.clear(); 9568 } 9569 mSuspendedEffects.removeItemsAt(index); 9570 } 9571 } 9572} 9573 9574// must be called with ThreadBase::mLock held 9575void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9576{ 9577 sp<SuspendedEffectDesc> desc; 9578 9579 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9580 if (suspend) { 9581 if (index >= 0) { 9582 desc = mSuspendedEffects.valueAt(index); 9583 } else { 9584 desc = new SuspendedEffectDesc(); 9585 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9586 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9587 } 9588 if (desc->mRefCount++ == 0) { 9589 Vector< sp<EffectModule> > effects; 9590 getSuspendEligibleEffects(effects); 9591 for (size_t i = 0; i < effects.size(); i++) { 9592 setEffectSuspended_l(&effects[i]->desc().type, true); 9593 } 9594 } 9595 } else { 9596 if (index < 0) { 9597 return; 9598 } 9599 desc = mSuspendedEffects.valueAt(index); 9600 if (desc->mRefCount <= 0) { 9601 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9602 desc->mRefCount = 1; 9603 } 9604 if (--desc->mRefCount == 0) { 9605 Vector<const effect_uuid_t *> types; 9606 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9607 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9608 continue; 9609 } 9610 types.add(&mSuspendedEffects.valueAt(i)->mType); 9611 } 9612 for (size_t i = 0; i < types.size(); i++) { 9613 setEffectSuspended_l(types[i], false); 9614 } 9615 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9616 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9617 } 9618 } 9619} 9620 9621 9622// The volume effect is used for automated tests only 9623#ifndef OPENSL_ES_H_ 9624static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9625 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9626const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9627#endif //OPENSL_ES_H_ 9628 9629bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9630{ 9631 // auxiliary effects and visualizer are never suspended on output mix 9632 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9633 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9634 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9635 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9636 return false; 9637 } 9638 return true; 9639} 9640 9641void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9642{ 9643 effects.clear(); 9644 for (size_t i = 0; i < mEffects.size(); i++) { 9645 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9646 effects.add(mEffects[i]); 9647 } 9648 } 9649} 9650 9651sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9652 const effect_uuid_t *type) 9653{ 9654 sp<EffectModule> effect = getEffectFromType_l(type); 9655 return effect != 0 && effect->isEnabled() ? effect : 0; 9656} 9657 9658void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9659 bool enabled) 9660{ 9661 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9662 if (enabled) { 9663 if (index < 0) { 9664 // if the effect is not suspend check if all effects are suspended 9665 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9666 if (index < 0) { 9667 return; 9668 } 9669 if (!isEffectEligibleForSuspend(effect->desc())) { 9670 return; 9671 } 9672 setEffectSuspended_l(&effect->desc().type, enabled); 9673 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9674 if (index < 0) { 9675 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9676 return; 9677 } 9678 } 9679 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9680 effect->desc().type.timeLow); 9681 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9682 // if effect is requested to suspended but was not yet enabled, supend it now. 9683 if (desc->mEffect == 0) { 9684 desc->mEffect = effect; 9685 effect->setEnabled(false); 9686 effect->setSuspended(true); 9687 } 9688 } else { 9689 if (index < 0) { 9690 return; 9691 } 9692 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9693 effect->desc().type.timeLow); 9694 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9695 desc->mEffect.clear(); 9696 effect->setSuspended(false); 9697 } 9698} 9699 9700#undef LOG_TAG 9701#define LOG_TAG "AudioFlinger" 9702 9703// ---------------------------------------------------------------------------- 9704 9705status_t AudioFlinger::onTransact( 9706 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9707{ 9708 return BnAudioFlinger::onTransact(code, data, reply, flags); 9709} 9710 9711}; // namespace android 9712